diff options
Diffstat (limited to 'third_party/libwebrtc/examples')
264 files changed, 30134 insertions, 0 deletions
diff --git a/third_party/libwebrtc/examples/BUILD.gn b/third_party/libwebrtc/examples/BUILD.gn new file mode 100644 index 0000000000..18cb6ecc67 --- /dev/null +++ b/third_party/libwebrtc/examples/BUILD.gn @@ -0,0 +1,953 @@ +# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. + +import("../webrtc.gni") + +if (is_android) { + import("//build/config/android/config.gni") + import("//build/config/android/rules.gni") +} else if (is_mac) { + import("//build/config/mac/rules.gni") +} else if (is_ios) { + import("//build/config/ios/rules.gni") +} else if (is_linux || is_chromeos) { + import("//build/config/linux/pkg_config.gni") +} + +group("examples") { + # This target shall build all targets in examples. + testonly = true + deps = [] + + if (is_android) { + deps += [ + ":AppRTCMobile", + ":AppRTCMobile_test_apk", + ":libwebrtc_unity", + "androidvoip", + ] + + # TODO(sakal): We include some code from the tests. Remove this dependency + # and remove this if-clause. + if (rtc_include_tests) { + deps += [ "androidnativeapi" ] + } + } + + if (!build_with_chromium) { + deps += [ ":stun_prober" ] + } + + if (is_ios || (is_mac && target_cpu != "x86")) { + deps += [ ":AppRTCMobile" ] + } + + if (is_linux || is_chromeos || is_win) { + deps += [ + ":peerconnection_server", + ":stunserver", + ":turnserver", + ] + if (current_os != "winuwp") { + deps += [ ":peerconnection_client" ] + } + } + + if (is_android || is_win) { + deps += [ ":webrtc_unity_plugin" ] + } +} + +rtc_library("read_auth_file") { + testonly = true + sources = [ + "turnserver/read_auth_file.cc", + "turnserver/read_auth_file.h", + ] + deps = [ + "../api:array_view", + "../rtc_base:stringutils", + ] + absl_deps = [ "//third_party/abseil-cpp/absl/strings:strings" ] +} + +if (rtc_include_tests) { + rtc_test("examples_unittests") { + testonly = true + sources = [ "turnserver/read_auth_file_unittest.cc" ] + deps = [ + ":read_auth_file", + "../test:test_main", + "//test:test_support", + "//testing/gtest", + ] + } +} + +if (is_android) { + rtc_android_apk("AppRTCMobile") { + testonly = true + apk_name = "AppRTCMobile" + android_manifest = "androidapp/AndroidManifest.xml" + min_sdk_version = 21 + target_sdk_version = 31 + + deps = [ + ":AppRTCMobile_javalib", + ":AppRTCMobile_resources", + "../rtc_base:base_java", + ] + + shared_libraries = [ "../sdk/android:libjingle_peerconnection_so" ] + } + + rtc_android_library("AppRTCMobile_javalib") { + testonly = true + android_manifest = "androidapp/AndroidManifest.xml" + + sources = [ + "androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java", + "androidapp/src/org/appspot/apprtc/AppRTCBluetoothManager.java", + "androidapp/src/org/appspot/apprtc/AppRTCClient.java", + "androidapp/src/org/appspot/apprtc/AppRTCProximitySensor.java", + "androidapp/src/org/appspot/apprtc/CallActivity.java", + "androidapp/src/org/appspot/apprtc/CallFragment.java", + "androidapp/src/org/appspot/apprtc/CaptureQualityController.java", + "androidapp/src/org/appspot/apprtc/ConnectActivity.java", + "androidapp/src/org/appspot/apprtc/CpuMonitor.java", + "androidapp/src/org/appspot/apprtc/DirectRTCClient.java", + "androidapp/src/org/appspot/apprtc/HudFragment.java", + "androidapp/src/org/appspot/apprtc/PeerConnectionClient.java", + "androidapp/src/org/appspot/apprtc/RecordedAudioToFileController.java", + "androidapp/src/org/appspot/apprtc/RoomParametersFetcher.java", + "androidapp/src/org/appspot/apprtc/RtcEventLog.java", + "androidapp/src/org/appspot/apprtc/SettingsActivity.java", + "androidapp/src/org/appspot/apprtc/SettingsFragment.java", + "androidapp/src/org/appspot/apprtc/TCPChannelClient.java", + "androidapp/src/org/appspot/apprtc/UnhandledExceptionHandler.java", + "androidapp/src/org/appspot/apprtc/WebSocketChannelClient.java", + "androidapp/src/org/appspot/apprtc/WebSocketRTCClient.java", + "androidapp/src/org/appspot/apprtc/util/AppRTCUtils.java", + "androidapp/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java", + ] + + resources_package = "org.appspot.apprtc" + deps = [ + ":AppRTCMobile_resources", + "../rtc_base:base_java", + "../sdk/android:audio_api_java", + "../sdk/android:base_java", + "../sdk/android:camera_java", + "../sdk/android:default_video_codec_factory_java", + "../sdk/android:filevideo_java", + "../sdk/android:hwcodecs_java", + "../sdk/android:java_audio_device_module_java", + "../sdk/android:libjingle_peerconnection_java", + "../sdk/android:libjingle_peerconnection_metrics_default_java", + "../sdk/android:peerconnection_java", + "../sdk/android:screencapturer_java", + "../sdk/android:surfaceviewrenderer_java", + "../sdk/android:swcodecs_java", + "../sdk/android:video_api_java", + "../sdk/android:video_java", + "androidapp/third_party/autobanh:autobanh_java", + "//third_party/androidx:androidx_annotation_annotation_java", + ] + } + + android_resources("AppRTCMobile_resources") { + testonly = true + sources = [ + "androidapp/res/drawable-hdpi/disconnect.png", + "androidapp/res/drawable-hdpi/ic_action_full_screen.png", + "androidapp/res/drawable-hdpi/ic_action_return_from_full_screen.png", + "androidapp/res/drawable-hdpi/ic_launcher.png", + "androidapp/res/drawable-hdpi/ic_loopback_call.png", + "androidapp/res/drawable-ldpi/disconnect.png", + "androidapp/res/drawable-ldpi/ic_action_full_screen.png", + "androidapp/res/drawable-ldpi/ic_action_return_from_full_screen.png", + "androidapp/res/drawable-ldpi/ic_launcher.png", + "androidapp/res/drawable-ldpi/ic_loopback_call.png", + "androidapp/res/drawable-mdpi/disconnect.png", + "androidapp/res/drawable-mdpi/ic_action_full_screen.png", + "androidapp/res/drawable-mdpi/ic_action_return_from_full_screen.png", + "androidapp/res/drawable-mdpi/ic_launcher.png", + "androidapp/res/drawable-mdpi/ic_loopback_call.png", + "androidapp/res/drawable-xhdpi/disconnect.png", + "androidapp/res/drawable-xhdpi/ic_action_full_screen.png", + "androidapp/res/drawable-xhdpi/ic_action_return_from_full_screen.png", + "androidapp/res/drawable-xhdpi/ic_launcher.png", + "androidapp/res/drawable-xhdpi/ic_loopback_call.png", + "androidapp/res/layout/activity_call.xml", + "androidapp/res/layout/activity_connect.xml", + "androidapp/res/layout/fragment_call.xml", + "androidapp/res/layout/fragment_hud.xml", + "androidapp/res/menu/connect_menu.xml", + "androidapp/res/values-v17/styles.xml", + "androidapp/res/values-v21/styles.xml", + "androidapp/res/values/arrays.xml", + "androidapp/res/values/strings.xml", + "androidapp/res/xml/preferences.xml", + ] + + # Needed for Bazel converter. + custom_package = "org.appspot.apprtc" + resource_dirs = [ "androidapp/res" ] + assert(resource_dirs != []) # Mark as used. + } + + rtc_instrumentation_test_apk("AppRTCMobile_test_apk") { + apk_name = "AppRTCMobileTest" + android_manifest = "androidtests/AndroidManifest.xml" + min_sdk_version = 21 + target_sdk_version = 31 + + sources = [ + "androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java", + ] + + apk_under_test = ":AppRTCMobile" + + deps = [ + ":AppRTCMobile_javalib", + "../sdk/android:base_java", + "../sdk/android:camera_java", + "../sdk/android:libjingle_peerconnection_java", + "../sdk/android:peerconnection_java", + "../sdk/android:video_api_java", + "../sdk/android:video_java", + "//third_party/android_support_test_runner:runner_java", + "//third_party/androidx:androidx_test_runner_java", + "//third_party/junit", + ] + } +} + +if (is_ios || (is_mac && target_cpu != "x86")) { + config("apprtc_common_config") { + include_dirs = [ "objc/AppRTCMobile/common" ] + } + + rtc_library("apprtc_common") { + testonly = true + sources = [ + "objc/AppRTCMobile/common/ARDUtilities.h", + "objc/AppRTCMobile/common/ARDUtilities.m", + ] + public_configs = [ ":apprtc_common_config" ] + deps = [ "../sdk:base_objc" ] + } + + config("apprtc_signaling_config") { + include_dirs = [ "objc/AppRTCMobile" ] + } + + rtc_library("apprtc_signaling") { + testonly = true + sources = [ + "objc/AppRTCMobile/ARDAppClient+Internal.h", + "objc/AppRTCMobile/ARDAppClient.h", + "objc/AppRTCMobile/ARDAppClient.m", + "objc/AppRTCMobile/ARDAppEngineClient.h", + "objc/AppRTCMobile/ARDAppEngineClient.m", + "objc/AppRTCMobile/ARDCaptureController.h", + "objc/AppRTCMobile/ARDCaptureController.m", + "objc/AppRTCMobile/ARDExternalSampleCapturer.h", + "objc/AppRTCMobile/ARDExternalSampleCapturer.m", + "objc/AppRTCMobile/ARDJoinResponse+Internal.h", + "objc/AppRTCMobile/ARDJoinResponse.h", + "objc/AppRTCMobile/ARDJoinResponse.m", + "objc/AppRTCMobile/ARDMessageResponse+Internal.h", + "objc/AppRTCMobile/ARDMessageResponse.h", + "objc/AppRTCMobile/ARDMessageResponse.m", + "objc/AppRTCMobile/ARDRoomServerClient.h", + "objc/AppRTCMobile/ARDSettingsModel+Private.h", + "objc/AppRTCMobile/ARDSettingsModel.h", + "objc/AppRTCMobile/ARDSettingsModel.m", + "objc/AppRTCMobile/ARDSettingsStore.h", + "objc/AppRTCMobile/ARDSettingsStore.m", + "objc/AppRTCMobile/ARDSignalingChannel.h", + "objc/AppRTCMobile/ARDSignalingMessage.h", + "objc/AppRTCMobile/ARDSignalingMessage.m", + "objc/AppRTCMobile/ARDStatsBuilder.h", + "objc/AppRTCMobile/ARDStatsBuilder.m", + "objc/AppRTCMobile/ARDTURNClient+Internal.h", + "objc/AppRTCMobile/ARDTURNClient.h", + "objc/AppRTCMobile/ARDTURNClient.m", + "objc/AppRTCMobile/ARDWebSocketChannel.h", + "objc/AppRTCMobile/ARDWebSocketChannel.m", + "objc/AppRTCMobile/RTCIceCandidate+JSON.h", + "objc/AppRTCMobile/RTCIceCandidate+JSON.m", + "objc/AppRTCMobile/RTCIceServer+JSON.h", + "objc/AppRTCMobile/RTCIceServer+JSON.m", + "objc/AppRTCMobile/RTCSessionDescription+JSON.h", + "objc/AppRTCMobile/RTCSessionDescription+JSON.m", + ] + public_configs = [ ":apprtc_signaling_config" ] + deps = [ + ":apprtc_common", + ":socketrocket", + "../sdk:base_objc", + "../sdk:default_codec_factory_objc", + "../sdk:file_logger_objc", + "../sdk:helpers_objc", + "../sdk:mediaconstraints_objc", + "../sdk:peerconnectionfactory_base_objc", + "../sdk:videocapture_objc", + "../sdk:videoframebuffer_objc", + "../sdk:videosource_objc", + ] + frameworks = [ + "CoreMedia.framework", + "QuartzCore.framework", + ] + } + + if (is_ios) { + rtc_library("AppRTCMobile_lib") { + # iOS must use WebRTC.framework which is dynamically linked. + testonly = true + sources = [ + "objc/AppRTCMobile/ios/ARDAppDelegate.h", + "objc/AppRTCMobile/ios/ARDAppDelegate.m", + "objc/AppRTCMobile/ios/ARDFileCaptureController.h", + "objc/AppRTCMobile/ios/ARDFileCaptureController.m", + "objc/AppRTCMobile/ios/ARDMainView.h", + "objc/AppRTCMobile/ios/ARDMainView.m", + "objc/AppRTCMobile/ios/ARDMainViewController.h", + "objc/AppRTCMobile/ios/ARDMainViewController.m", + "objc/AppRTCMobile/ios/ARDSettingsViewController.h", + "objc/AppRTCMobile/ios/ARDSettingsViewController.m", + "objc/AppRTCMobile/ios/ARDStatsView.h", + "objc/AppRTCMobile/ios/ARDStatsView.m", + "objc/AppRTCMobile/ios/ARDVideoCallView.h", + "objc/AppRTCMobile/ios/ARDVideoCallView.m", + "objc/AppRTCMobile/ios/ARDVideoCallViewController.h", + "objc/AppRTCMobile/ios/ARDVideoCallViewController.m", + "objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.h", + "objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.m", + "objc/AppRTCMobile/ios/UIImage+ARDUtilities.h", + "objc/AppRTCMobile/ios/UIImage+ARDUtilities.m", + ] + + configs += [ "..:common_objc" ] + + deps = [ + ":apprtc_common", + ":apprtc_signaling", + "../sdk:audio_session_objc", + "../sdk:base_objc", + "../sdk:helpers_objc", + "../sdk:mediaconstraints_objc", + "../sdk:metal_objc", + "../sdk:peerconnectionfactory_base_objc", + "../sdk:peerconnectionfactory_base_objc", + "../sdk:videocapture_objc", + "../sdk:videocodec_objc", + ] + if (rtc_ios_macos_use_opengl_rendering) { + deps += [ "../sdk:opengl_ui_objc" ] + } + + frameworks = [ "AVFoundation.framework" ] + } + + ios_app_bundle("AppRTCMobile") { + testonly = true + sources = [ "objc/AppRTCMobile/ios/main.m" ] + + info_plist = "objc/AppRTCMobile/ios/Info.plist" + + configs += [ "..:common_config" ] + public_configs = [ "..:common_inherited_config" ] + + deps = [ + ":AppRTCMobile_ios_bundle_data", + ":AppRTCMobile_lib", + "../sdk:framework_objc", + "../sdk:ios_framework_bundle", + ] + + if (rtc_apprtcmobile_broadcast_extension) { + deps += [ + ":AppRTCMobileBroadcastSetupUI_extension_bundle", + ":AppRTCMobileBroadcastUpload_extension_bundle", + ] + } + + if (target_cpu == "x86") { + deps += [ "//testing/iossim" ] + } + } + + if (rtc_apprtcmobile_broadcast_extension) { + bundle_data("AppRTCMobileBroadcastUpload_extension_bundle") { + testonly = true + public_deps = [ # no-presubmit-check TODO(webrtc:8603) + ":AppRTCMobileBroadcastUpload", # prevent code format + ] + sources = [ "$root_out_dir/AppRTCMobileBroadcastUpload.appex" ] + outputs = [ "{{bundle_contents_dir}}/Plugins/{{source_file_part}}" ] + } + + bundle_data("AppRTCMobileBroadcastSetupUI_extension_bundle") { + testonly = true + public_deps = [ # no-presubmit-check TODO(webrtc:8603) + ":AppRTCMobileBroadcastSetupUI", # prevent code format + ] + sources = [ "$root_out_dir/AppRTCMobileBroadcastSetupUI.appex" ] + outputs = [ "{{bundle_contents_dir}}/Plugins/{{source_file_part}}" ] + } + + rtc_library("AppRTCMobileBroadcastUpload_lib") { + testonly = true + sources = [ + "objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.h", + "objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.m", + ] + + deps = [ + ":apprtc_signaling", + "../sdk:framework_objc+link", + "../sdk:ios_framework_bundle", + ] + + frameworks = [ "ReplayKit.framework" ] + } + + ios_appex_bundle("AppRTCMobileBroadcastUpload") { + testonly = true + configs += [ "..:common_config" ] + public_configs = [ "..:common_inherited_config" ] + + info_plist = "objc/AppRTCMobile/ios/broadcast_extension/BroadcastUploadInfo.plist" + + deps = [ + ":AppRTCMobileBroadcastUpload_lib", + "../sdk:framework_objc", + ] + } + + ios_appex_bundle("AppRTCMobileBroadcastSetupUI") { + sources = [ + "objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.h", + "objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.m", + ] + + info_plist = "objc/AppRTCMobile/ios/broadcast_extension/BroadcastSetupUIInfo.plist" + + frameworks = [ "ReplayKit.framework" ] + + deps = [ ":AppRTCMobile_ios_bundle_data" ] + } + } + + bundle_data("AppRTCMobile_ios_bundle_data") { + sources = [ + "objc/AppRTCMobile/ios/resources/Roboto-Regular.ttf", + + # Sample video taken from https://media.xiph.org/video/derf/ + "objc/AppRTCMobile/ios/resources/foreman.mp4", + "objc/AppRTCMobile/ios/resources/iPhone5@2x.png", + "objc/AppRTCMobile/ios/resources/iPhone6@2x.png", + "objc/AppRTCMobile/ios/resources/iPhone6p@3x.png", + "objc/AppRTCMobile/ios/resources/ic_call_end_black_24dp.png", + "objc/AppRTCMobile/ios/resources/ic_call_end_black_24dp@2x.png", + "objc/AppRTCMobile/ios/resources/ic_clear_black_24dp.png", + "objc/AppRTCMobile/ios/resources/ic_clear_black_24dp@2x.png", + "objc/AppRTCMobile/ios/resources/ic_settings_black_24dp.png", + "objc/AppRTCMobile/ios/resources/ic_settings_black_24dp@2x.png", + "objc/AppRTCMobile/ios/resources/ic_surround_sound_black_24dp.png", + "objc/AppRTCMobile/ios/resources/ic_surround_sound_black_24dp@2x.png", + "objc/AppRTCMobile/ios/resources/ic_switch_video_black_24dp.png", + "objc/AppRTCMobile/ios/resources/ic_switch_video_black_24dp@2x.png", + "objc/AppRTCMobile/ios/resources/mozart.mp3", + "objc/Icon-120.png", + "objc/Icon-180.png", + "objc/Icon.png", + ] + outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ] + } + + rtc_library("ObjCNativeAPIDemo_lib") { + testonly = true + sources = [ + "objcnativeapi/objc/NADAppDelegate.h", + "objcnativeapi/objc/NADAppDelegate.m", + "objcnativeapi/objc/NADViewController.h", + "objcnativeapi/objc/NADViewController.mm", + "objcnativeapi/objc/objc_call_client.h", + "objcnativeapi/objc/objc_call_client.mm", + ] + + deps = [ + "../api:libjingle_peerconnection_api", + "../api:scoped_refptr", + "../api:sequence_checker", + "../api/audio_codecs:builtin_audio_decoder_factory", + "../api/audio_codecs:builtin_audio_encoder_factory", + "../api/rtc_event_log:rtc_event_log_factory", + "../api/task_queue:default_task_queue_factory", + "../media:rtc_audio_video", + "../modules/audio_processing", + "../modules/audio_processing:api", + "../pc:libjingle_peerconnection", + "../rtc_base/synchronization:mutex", + "../sdk:base_objc", + "../sdk:default_codec_factory_objc", + "../sdk:helpers_objc", + "../sdk:metal_objc", + "../sdk:native_api", + "../sdk:videocapture_objc", + "../sdk:videotoolbox_objc", + ] + + if (rtc_ios_macos_use_opengl_rendering) { + deps += [ "../sdk:opengl_ui_objc" ] + } + } + + ios_app_bundle("ObjCNativeAPIDemo") { + testonly = true + sources = [ "objcnativeapi/objc/main.m" ] + + info_plist = "objcnativeapi/Info.plist" + + configs += [ "..:common_config" ] + public_configs = [ "..:common_inherited_config" ] + + deps = [ ":ObjCNativeAPIDemo_lib" ] + + if (target_cpu == "x86") { + deps += [ "//testing/iossim" ] + } + } + } + + if (is_mac) { + rtc_library("AppRTCMobile_lib") { + testonly = true + sources = [ + "objc/AppRTCMobile/mac/APPRTCAppDelegate.h", + "objc/AppRTCMobile/mac/APPRTCAppDelegate.m", + "objc/AppRTCMobile/mac/APPRTCViewController.h", + "objc/AppRTCMobile/mac/APPRTCViewController.m", + ] + configs += [ "..:common_objc" ] + deps = [ + ":apprtc_common", + ":apprtc_signaling", + "../sdk:base_objc", + "../sdk:helpers_objc", + "../sdk:mediaconstraints_objc", + "../sdk:metal_objc", + "../sdk:opengl_ui_objc", + "../sdk:peerconnectionfactory_base_objc", + "../sdk:peerconnectionfactory_base_objc", + "../sdk:videocapture_objc", + "../sdk:videocodec_objc", + ] + } + + mac_app_bundle("AppRTCMobile") { + testonly = true + output_name = "AppRTCMobile" + + sources = [ "objc/AppRTCMobile/mac/main.m" ] + + public_configs = [ "..:common_inherited_config" ] + + info_plist = "objc/AppRTCMobile/mac/Info.plist" + + frameworks = [ "AppKit.framework" ] + + ldflags = [ + "-rpath", + "@executable_path/../Frameworks", + ] + + deps = [ + ":AppRTCMobile_lib", + "../sdk:mac_framework_bundle", + "../sdk:mac_framework_objc+link", + ] + } + } + + config("socketrocket_include_config") { + include_dirs = [ "objc/AppRTCMobile/third_party/SocketRocket" ] + } + + config("socketrocket_warning_config") { + # GN orders flags on a target before flags from configs. The default config + # adds these flags so to cancel them out they need to come from a config and + # cannot be on the target directly. + cflags = [ + "-Wno-deprecated-declarations", + "-Wno-nonnull", + "-Wno-semicolon-before-method-body", + "-Wno-unused-variable", + ] + + cflags_objc = [ + # Enabled for cflags_objc in build/config/compiler/BUILD.gn. + "-Wno-objc-missing-property-synthesis", + ] + } + + rtc_library("socketrocket") { + testonly = true + sources = [ + "objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.h", + "objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m", + ] + configs += [ ":socketrocket_warning_config" ] + public_configs = [ ":socketrocket_include_config" ] + + libs = [ "icucore" ] + frameworks = [ + "CFNetwork.framework", + "Security.framework", + ] + } + + if (rtc_include_tests) { + # TODO(kthelgason): compile xctests on mac when chromium supports it. + if (is_ios) { + rtc_library("apprtcmobile_test_sources") { + # iOS must use WebRTC.framework which is dynamically linked. + testonly = true + include_dirs = [ + "objc/AppRTCMobile", + "objc/AppRTCMobile/ios", + ] + sources = [ + "objc/AppRTCMobile/tests/ARDAppClient_xctest.mm", + "objc/AppRTCMobile/tests/ARDFileCaptureController_xctest.mm", + "objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm", + ] + deps = [ + ":AppRTCMobile_lib", + ":apprtc_signaling", + "../rtc_base:ssl", + "../sdk:mediaconstraints_objc", + "../sdk:peerconnectionfactory_base_objc", + "../sdk:videocapture_objc", + "//build/config/ios:xctest", + "//third_party/ocmock", + ] + } + + rtc_test("apprtcmobile_tests") { + is_xctest = true + info_plist = "objc/AppRTCMobile/ios/Info.plist" + sources = [ "objc/AppRTCMobile/tests/main.mm" ] + deps = [ + ":AppRTCMobile_lib", + ":apprtcmobile_test_sources", + "../sdk:framework_objc", + "//test:test_support", + ] + ldflags = [ "-all_load" ] + } + } + } +} + +if (is_linux || is_chromeos || is_win) { + if (is_linux || is_chromeos) { + pkg_config("gtk_config") { + packages = [ + # Gtk requires gmodule, but it does not list it as a dependency in some + # misconfigured systems. + "gmodule-2.0", + "gthread-2.0", + "gtk+-3.0", + ] + } + } + + rtc_executable("peerconnection_client") { + testonly = true + sources = [ + "peerconnection/client/conductor.cc", + "peerconnection/client/conductor.h", + "peerconnection/client/defaults.cc", + "peerconnection/client/defaults.h", + "peerconnection/client/peer_connection_client.cc", + "peerconnection/client/peer_connection_client.h", + ] + + deps = [ + "../api:audio_options_api", + "../api:create_peerconnection_factory", + "../api:libjingle_peerconnection_api", + "../api:media_stream_interface", + "../api:rtp_sender_interface", + "../api:scoped_refptr", + "../api/audio:audio_mixer_api", + "../api/audio_codecs:audio_codecs_api", + "../api/task_queue:pending_task_safety_flag", + "../api/units:time_delta", + "../api/video:video_frame", + "../api/video:video_rtp_headers", + "../api/video_codecs:video_codecs_api", + "../media:media_channel", + "../media:rtc_media_base", + "../p2p:rtc_p2p", + "../pc:video_track_source", + "../rtc_base:checks", + "../rtc_base:logging", + "../rtc_base:macromagic", + "../rtc_base:net_helpers", + "../rtc_base:refcount", + "../rtc_base:rtc_certificate_generator", + "../rtc_base:ssl", + "../rtc_base:stringutils", + "../rtc_base:threading", + "../rtc_base/third_party/sigslot", + "../system_wrappers:field_trial", + "../test:field_trial", + "../test:platform_video_capturer", + "../test:rtp_test_utils", + "//third_party/abseil-cpp/absl/memory", + "//third_party/abseil-cpp/absl/types:optional", + ] + if (is_win) { + sources += [ + "peerconnection/client/flag_defs.h", + "peerconnection/client/main.cc", + "peerconnection/client/main_wnd.cc", + "peerconnection/client/main_wnd.h", + ] + configs += [ "//build/config/win:windowed" ] + deps += [ + "../media:rtc_media_base", + "../rtc_base:win32", + "../rtc_base:win32_socket_init", + ] + } + if (is_linux || is_chromeos) { + sources += [ + "peerconnection/client/linux/main.cc", + "peerconnection/client/linux/main_wnd.cc", + "peerconnection/client/linux/main_wnd.h", + ] + cflags = [ "-Wno-deprecated-declarations" ] + libs = [ + "X11", + "Xcomposite", + "Xext", + "Xrender", + ] + configs += [ ":gtk_config" ] + } + + deps += [ + "../api:libjingle_peerconnection_api", + "../api/audio_codecs:builtin_audio_decoder_factory", + "../api/audio_codecs:builtin_audio_encoder_factory", + "../api/video:video_frame", + "../api/video:video_rtp_headers", + "../api/video_codecs:builtin_video_decoder_factory", + "../api/video_codecs:builtin_video_encoder_factory", + "../media:rtc_audio_video", + "../modules/audio_device", + "../modules/audio_processing", + "../modules/audio_processing:api", + "../modules/video_capture:video_capture_module", + "../pc:libjingle_peerconnection", + "../rtc_base:rtc_json", + "../test:video_test_common", + "//third_party/abseil-cpp/absl/flags:flag", + "//third_party/abseil-cpp/absl/flags:parse", + "//third_party/libyuv", + ] + } + + rtc_executable("peerconnection_server") { + testonly = true + sources = [ + "peerconnection/server/data_socket.cc", + "peerconnection/server/data_socket.h", + "peerconnection/server/main.cc", + "peerconnection/server/peer_channel.cc", + "peerconnection/server/peer_channel.h", + "peerconnection/server/utils.cc", + "peerconnection/server/utils.h", + ] + deps = [ + "../rtc_base:checks", + "../rtc_base:stringutils", + "../system_wrappers:field_trial", + "../test:field_trial", + "//third_party/abseil-cpp/absl/flags:flag", + "//third_party/abseil-cpp/absl/flags:parse", + "//third_party/abseil-cpp/absl/flags:usage", + ] + } + rtc_executable("turnserver") { + testonly = true + sources = [ "turnserver/turnserver_main.cc" ] + deps = [ + ":read_auth_file", + "../p2p:p2p_server_utils", + "../p2p:rtc_p2p", + "../pc:rtc_pc", + "../rtc_base:async_udp_socket", + "../rtc_base:ip_address", + "../rtc_base:socket_address", + "../rtc_base:socket_server", + "../rtc_base:threading", + "//third_party/abseil-cpp/absl/strings:strings", + ] + } + rtc_executable("stunserver") { + testonly = true + sources = [ "stunserver/stunserver_main.cc" ] + deps = [ + "../p2p:p2p_server_utils", + "../p2p:rtc_p2p", + "../pc:rtc_pc", + "../rtc_base:async_udp_socket", + "../rtc_base:socket_address", + "../rtc_base:socket_server", + "../rtc_base:threading", + ] + } +} + +if (is_win || is_android) { + rtc_shared_library("webrtc_unity_plugin") { + testonly = true + sources = [ + "unityplugin/simple_peer_connection.cc", + "unityplugin/simple_peer_connection.h", + "unityplugin/unity_plugin_apis.cc", + "unityplugin/unity_plugin_apis.h", + "unityplugin/video_observer.cc", + "unityplugin/video_observer.h", + ] + + if (is_android) { + sources += [ + "unityplugin/class_reference_holder.cc", + "unityplugin/class_reference_holder.h", + "unityplugin/jni_onload.cc", + ] + suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ] + } + + if (is_win) { + configs += [ "//build/config/win:windowed" ] + } + deps = [ + "../api:create_peerconnection_factory", + "../api:libjingle_peerconnection_api", + "../api:media_stream_interface", + "../api/audio_codecs:builtin_audio_decoder_factory", + "../api/audio_codecs:builtin_audio_encoder_factory", + "../api/video:video_frame", + "../api/video:video_rtp_headers", + "../media:rtc_audio_video", + "../media:rtc_internal_video_codecs", + "../media:rtc_media", + "../media:rtc_media_base", + "../modules/audio_device", + "../modules/audio_processing", + "../modules/audio_processing:api", + "../modules/video_capture:video_capture_module", + "../pc:libjingle_peerconnection", + "../pc:video_track_source", + "../rtc_base:ssl", + "../test:platform_video_capturer", + "../test:video_test_common", + "//third_party/abseil-cpp/absl/memory", + ] + if (is_android) { + deps += [ + "../modules/utility", + "../sdk/android:libjingle_peerconnection_jni", + "../sdk/android:native_api_jni", + ] + } + } +} + +if (is_android) { + rtc_android_library("webrtc_unity_java") { + sources = [ "unityplugin/java/src/org/webrtc/UnityUtility.java" ] + deps = [ + "../rtc_base:base_java", + "../sdk/android:camera_java", + "../sdk/android:libjingle_peerconnection_java", + "../sdk/android:peerconnection_java", + "../sdk/android:video_api_java", + "../sdk/android:video_java", + "//third_party/androidx:androidx_annotation_annotation_java", + ] + } + + dist_jar("libwebrtc_unity") { + _target_dir_name = get_label_info(":$target_name", "dir") + output = "${root_out_dir}/lib.java${_target_dir_name}/${target_name}.jar" + direct_deps_only = false + use_interface_jars = false + use_unprocessed_jars = false + requires_android = true + deps = [ + ":webrtc_unity_java", + "../rtc_base:base_java", + "../sdk/android:libjingle_peerconnection_java", + "../sdk/android:libjingle_peerconnection_metrics_default_java", + "//third_party/androidx:androidx_annotation_annotation_java", + ] + } + + robolectric_binary("android_examples_junit_tests") { + sources = [ + "androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java", + "androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java", + "androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java", + ] + + deps = [ + ":AppRTCMobile_javalib", + "../sdk/android:peerconnection_java", + "//third_party/androidx:androidx_test_core_java", + "//third_party/google-truth:google_truth_java", + ] + + additional_jar_files = [ [ + "../sdk/android/tests/resources/robolectric.properties", + "robolectric.properties", + ] ] + } +} + +if (!build_with_chromium) { + # Doesn't build within Chrome on Win. + rtc_executable("stun_prober") { + testonly = true + sources = [ "stunprober/main.cc" ] + deps = [ + "../p2p:libstunprober", + "../p2p:rtc_p2p", + "../rtc_base:checks", + "../rtc_base:logging", + "../rtc_base:network", + "../rtc_base:socket_address", + "../rtc_base:ssl", + "../rtc_base:threading", + "../rtc_base:timeutils", + "../test:scoped_key_value_config", + "//third_party/abseil-cpp/absl/flags:flag", + "//third_party/abseil-cpp/absl/flags:parse", + ] + } +} diff --git a/third_party/libwebrtc/examples/DEPS b/third_party/libwebrtc/examples/DEPS new file mode 100644 index 0000000000..114cda384b --- /dev/null +++ b/third_party/libwebrtc/examples/DEPS @@ -0,0 +1,13 @@ +include_rules = [ + "+common_video", + "+logging/rtc_event_log/rtc_event_log_factory.h", + "+media", + "+modules/audio_device", + "+modules/video_capture", + "+modules/audio_processing", + "+p2p", + "+pc", + "+sdk/objc", + "+system_wrappers/include", + "+third_party/libyuv", +] diff --git a/third_party/libwebrtc/examples/OWNERS b/third_party/libwebrtc/examples/OWNERS new file mode 100644 index 0000000000..ff1f425462 --- /dev/null +++ b/third_party/libwebrtc/examples/OWNERS @@ -0,0 +1,4 @@ +magjed@webrtc.org +perkj@webrtc.org +tkchin@webrtc.org +kthelgason@webrtc.org diff --git a/third_party/libwebrtc/examples/aarproject/.gitignore b/third_party/libwebrtc/examples/aarproject/.gitignore new file mode 100644 index 0000000000..e93eb885a9 --- /dev/null +++ b/third_party/libwebrtc/examples/aarproject/.gitignore @@ -0,0 +1,16 @@ +# Default ignores by Android Studio +*.iml +.gradle +# We want to specify our own SDK. +# /local.properties +/.idea/workspace.xml +/.idea/libraries +.DS_Store +/build +/captures +.externalNativeBuild + +# Additional ignores +/gradlew +/gradlew.bat +/gradle diff --git a/third_party/libwebrtc/examples/aarproject/OWNERS b/third_party/libwebrtc/examples/aarproject/OWNERS new file mode 100644 index 0000000000..cf092a316a --- /dev/null +++ b/third_party/libwebrtc/examples/aarproject/OWNERS @@ -0,0 +1 @@ +xalep@webrtc.org diff --git a/third_party/libwebrtc/examples/aarproject/app/.gitignore b/third_party/libwebrtc/examples/aarproject/app/.gitignore new file mode 100644 index 0000000000..796b96d1c4 --- /dev/null +++ b/third_party/libwebrtc/examples/aarproject/app/.gitignore @@ -0,0 +1 @@ +/build diff --git a/third_party/libwebrtc/examples/aarproject/app/build.gradle b/third_party/libwebrtc/examples/aarproject/app/build.gradle new file mode 100644 index 0000000000..94fa851af2 --- /dev/null +++ b/third_party/libwebrtc/examples/aarproject/app/build.gradle @@ -0,0 +1,53 @@ +apply plugin: 'com.android.application' + +android { + compileSdkVersion 31 + defaultConfig { + applicationId "org.appspot.apprtc" + minSdkVersion 21 + targetSdkVersion 31 + versionCode 1 + versionName "1.0" + testInstrumentationRunner "android.support.test.runner.AndroidJUnitRunner" + } + buildTypes { + release { + minifyEnabled false + proguardFiles getDefaultProguardFile('proguard-android.txt'), 'proguard-rules.pro' + } + } + compileOptions { + sourceCompatibility JavaVersion.VERSION_1_8 + targetCompatibility JavaVersion.VERSION_1_8 + } + sourceSets { + main { + manifest.srcFile "../../androidapp/AndroidManifest.xml" + java.srcDirs = [ + "../../androidapp/src" + ] + res.srcDirs = [ + "../../androidapp/res" + ] + } + androidTest { + manifest.srcFile "../../androidtests/AndroidManifest.xml" + java.srcDirs = [ + "../../androidtests/src" + ] + // This test doesn't work in Android Studio. + java.exclude('**/CallActivityStubbedInputOutputTest.java') + } + } +} + +dependencies { + if (project.hasProperty('aarDir')) { + implementation fileTree(dir: project.aarDir, include: ['google-webrtc-*.aar']) + } + implementation fileTree(dir: '../../androidapp/third_party/autobanh/lib', include: ['autobanh.jar']) + implementation 'androidx.annotation:annotation:1.2.0' + testImplementation 'junit:junit:4.12' + androidTestImplementation 'com.android.support.test:runner:1.0.1' + androidTestImplementation 'com.android.support.test.espresso:espresso-core:3.0.1' +} diff --git a/third_party/libwebrtc/examples/aarproject/app/proguard-rules.pro b/third_party/libwebrtc/examples/aarproject/app/proguard-rules.pro new file mode 100644 index 0000000000..d6cc4c5eba --- /dev/null +++ b/third_party/libwebrtc/examples/aarproject/app/proguard-rules.pro @@ -0,0 +1,25 @@ +# Add project specific ProGuard rules here. +# By default, the flags in this file are appended to flags specified +# in /usr/local/google/home/sakal/Android/Sdk/tools/proguard/proguard-android.txt +# You can edit the include path and order by changing the proguardFiles +# directive in build.gradle. +# +# For more details, see +# http://developer.android.com/guide/developing/tools/proguard.html + +# Add any project specific keep options here: + +# If your project uses WebView with JS, uncomment the following +# and specify the fully qualified class name to the JavaScript interface +# class: +#-keepclassmembers class fqcn.of.javascript.interface.for.webview { +# public *; +#} + +# Uncomment this to preserve the line number information for +# debugging stack traces. +#-keepattributes SourceFile,LineNumberTable + +# If you keep the line number information, uncomment this to +# hide the original source file name. +#-renamesourcefileattribute SourceFile diff --git a/third_party/libwebrtc/examples/aarproject/build.gradle b/third_party/libwebrtc/examples/aarproject/build.gradle new file mode 100644 index 0000000000..969effd3ed --- /dev/null +++ b/third_party/libwebrtc/examples/aarproject/build.gradle @@ -0,0 +1,27 @@ +// Top-level build file where you can add configuration options common to all sub-projects/modules. + +buildscript { + + repositories { + google() + mavenCentral() + } + dependencies { + classpath "com.android.tools.build:gradle:7.0.3" + + + // NOTE: Do not place your application dependencies here; they belong + // in the individual module build.gradle files + } +} + +allprojects { + repositories { + google() + mavenCentral() + } +} + +task clean(type: Delete) { + delete rootProject.buildDir +} diff --git a/third_party/libwebrtc/examples/aarproject/gradle.properties b/third_party/libwebrtc/examples/aarproject/gradle.properties new file mode 100644 index 0000000000..2e87c52f83 --- /dev/null +++ b/third_party/libwebrtc/examples/aarproject/gradle.properties @@ -0,0 +1,22 @@ +# Project-wide Gradle settings. + +# IDE (e.g. Android Studio) users: +# Gradle settings configured through the IDE *will override* +# any settings specified in this file. + +# For more details on how to configure your build environment visit +# http://www.gradle.org/docs/current/userguide/build_environment.html + +# Specifies the JVM arguments used for the daemon process. +# The setting is particularly useful for tweaking memory settings. +org.gradle.jvmargs=-Xmx1536m + +# When configured, Gradle will run in incubating parallel mode. +# This option should only be used with decoupled projects. More details, visit +# http://www.gradle.org/docs/current/userguide/multi_project_builds.html#sec:decoupled_projects +# org.gradle.parallel=true + +# AndroidX package structure to make it clearer which packages are bundled with the +# Android operating system, and which are packaged with your app's APK +# https://developer.android.com/topic/libraries/support-library/androidx-rn +android.useAndroidX=true diff --git a/third_party/libwebrtc/examples/aarproject/local.properties b/third_party/libwebrtc/examples/aarproject/local.properties new file mode 100644 index 0000000000..99cdcd2674 --- /dev/null +++ b/third_party/libwebrtc/examples/aarproject/local.properties @@ -0,0 +1,2 @@ +# Use Android SDK from third_party/android_sdk/public +sdk.dir=../../third_party/android_sdk/public diff --git a/third_party/libwebrtc/examples/aarproject/settings.gradle b/third_party/libwebrtc/examples/aarproject/settings.gradle new file mode 100644 index 0000000000..e7b4def49c --- /dev/null +++ b/third_party/libwebrtc/examples/aarproject/settings.gradle @@ -0,0 +1 @@ +include ':app' diff --git a/third_party/libwebrtc/examples/androidapp/AndroidManifest.xml b/third_party/libwebrtc/examples/androidapp/AndroidManifest.xml new file mode 100644 index 0000000000..05f1bd3da3 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/AndroidManifest.xml @@ -0,0 +1,60 @@ +<?xml version="1.0" encoding="utf-8"?> +<manifest xmlns:android="http://schemas.android.com/apk/res/android" + xmlns:tools="http://schemas.android.com/tools" + package="org.appspot.apprtc" + android:versionCode="1" + android:versionName="1.0"> + + <uses-feature android:name="android.hardware.camera" /> + <uses-feature android:name="android.hardware.camera.autofocus" /> + <uses-feature android:glEsVersion="0x00020000" android:required="true" /> + + <uses-permission android:name="android.permission.CAMERA" /> + <uses-permission android:name="android.permission.CHANGE_NETWORK_STATE" /> + <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" /> + <uses-permission android:name="android.permission.RECORD_AUDIO" /> + <uses-permission android:name="android.permission.BLUETOOTH" /> + <uses-permission android:name="android.permission.INTERNET" /> + <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/> + <uses-permission android:name="android.permission.ACCESS_NETWORK_STATE" /> + + <!-- This is a test application that should always be debuggable. --> + <application android:label="@string/app_name" + android:icon="@drawable/ic_launcher" + android:allowBackup="false" + android:debuggable="true" + android:supportsRtl="false" + tools:ignore="HardcodedDebugMode"> + + <activity android:name="ConnectActivity" + android:label="@string/app_name" + android:windowSoftInputMode="adjustPan" + android:theme="@style/AppTheme" + android:exported="true"> + <intent-filter> + <action android:name="android.intent.action.MAIN"/> + <category android:name="android.intent.category.LAUNCHER"/> + </intent-filter> + + <intent-filter> + <action android:name="android.intent.action.VIEW"/> + <category android:name="android.intent.category.DEFAULT"/> + <category android:name="android.intent.category.BROWSABLE"/> + <data android:scheme="https" android:host="appr.tc"/> + <data android:scheme="http" android:host="appr.tc"/> + </intent-filter> + </activity> + + <activity android:name="SettingsActivity" + android:label="@string/settings_name" + android:theme="@style/AppTheme"> + </activity> + + <activity android:name="CallActivity" + android:label="@string/app_name" + android:screenOrientation="fullUser" + android:configChanges="orientation|smallestScreenSize|screenSize|screenLayout" + android:theme="@style/CallActivityTheme"> + </activity> + </application> +</manifest> diff --git a/third_party/libwebrtc/examples/androidapp/OWNERS b/third_party/libwebrtc/examples/androidapp/OWNERS new file mode 100644 index 0000000000..109bea2725 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/OWNERS @@ -0,0 +1,2 @@ +magjed@webrtc.org +xalep@webrtc.org diff --git a/third_party/libwebrtc/examples/androidapp/README b/third_party/libwebrtc/examples/androidapp/README new file mode 100644 index 0000000000..97e609117c --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/README @@ -0,0 +1,23 @@ +This directory contains an example Android client for https://appr.tc + +Prerequisites: +- "Getting the code", "Compiling", and "Using the Bundled Android SDK/NDK" + on http://www.webrtc.org/native-code/android + +Example of building & using the app: + +cd <path/to/webrtc>/src +ninja -C out/Default AppRTCMobile +adb install -r out/Default/apks/AppRTCMobile.apk + +In desktop chrome, navigate to https://appr.tc and note the r=<NNN> room +this redirects to or navigate directly to https://appr.tc/r/<NNN> with +your own room number. Launch AppRTC on the device and add same <NNN> into the room name list. + +You can also run application from a command line to connect to the first room in a list: +adb shell am start -n org.appspot.apprtc/.ConnectActivity -a android.intent.action.VIEW +This should result in the app launching on Android and connecting to the 3-dot-apprtc +page displayed in the desktop browser. +To run loopback test execute following command: +adb shell am start -n org.appspot.apprtc/.ConnectActivity -a android.intent.action.VIEW --ez "org.appspot.apprtc.LOOPBACK" true + diff --git a/third_party/libwebrtc/examples/androidapp/ant.properties b/third_party/libwebrtc/examples/androidapp/ant.properties new file mode 100644 index 0000000000..b0971e891e --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/ant.properties @@ -0,0 +1,17 @@ +# This file is used to override default values used by the Ant build system. +# +# This file must be checked into Version Control Systems, as it is +# integral to the build system of your project. + +# This file is only used by the Ant script. + +# You can use this to override default values such as +# 'source.dir' for the location of your java source folder and +# 'out.dir' for the location of your output folder. + +# You can also use it define how the release builds are signed by declaring +# the following properties: +# 'key.store' for the location of your keystore and +# 'key.alias' for the name of the key to use. +# The password will be asked during the build when you use the 'release' target. + diff --git a/third_party/libwebrtc/examples/androidapp/build.xml b/third_party/libwebrtc/examples/androidapp/build.xml new file mode 100644 index 0000000000..aa1db6db79 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/build.xml @@ -0,0 +1,92 @@ +<?xml version="1.0" encoding="UTF-8"?> +<project name="AppRTCMobile" default="help"> + + <!-- The local.properties file is created and updated by the 'android' tool. + It contains the path to the SDK. It should *NOT* be checked into + Version Control Systems. --> + <property file="local.properties" /> + + <!-- The ant.properties file can be created by you. It is only edited by the + 'android' tool to add properties to it. + This is the place to change some Ant specific build properties. + Here are some properties you may want to change/update: + + source.dir + The name of the source directory. Default is 'src'. + out.dir + The name of the output directory. Default is 'bin'. + + For other overridable properties, look at the beginning of the rules + files in the SDK, at tools/ant/build.xml + + Properties related to the SDK location or the project target should + be updated using the 'android' tool with the 'update' action. + + This file is an integral part of the build system for your + application and should be checked into Version Control Systems. + + --> + <property file="ant.properties" /> + + <!-- if sdk.dir was not set from one of the property file, then + get it from the ANDROID_HOME env var. + This must be done before we load project.properties since + the proguard config can use sdk.dir --> + <property environment="env" /> + <condition property="sdk.dir" value="${env.ANDROID_SDK_ROOT}"> + <isset property="env.ANDROID_SDK_ROOT" /> + </condition> + + <!-- The project.properties file is created and updated by the 'android' + tool, as well as ADT. + + This contains project specific properties such as project target, and library + dependencies. Lower level build properties are stored in ant.properties + (or in .classpath for Eclipse projects). + + This file is an integral part of the build system for your + application and should be checked into Version Control Systems. --> + <loadproperties srcFile="project.properties" /> + + <!-- quick check on sdk.dir --> + <fail + message="sdk.dir is missing. Make sure to generate local.properties using 'android update project' or to inject it through the ANDROID_HOME environment variable." + unless="sdk.dir" + /> + + <!-- + Import per project custom build rules if present at the root of the project. + This is the place to put custom intermediary targets such as: + -pre-build + -pre-compile + -post-compile (This is typically used for code obfuscation. + Compiled code location: ${out.classes.absolute.dir} + If this is not done in place, override ${out.dex.input.absolute.dir}) + -post-package + -post-build + -pre-clean + --> + <import file="custom_rules.xml" optional="true" /> + + <!-- Import the actual build file. + + To customize existing targets, there are two options: + - Customize only one target: + - copy/paste the target into this file, *before* the + <import> task. + - customize it to your needs. + - Customize the whole content of build.xml + - copy/paste the content of the rules files (minus the top node) + into this file, replacing the <import> task. + - customize to your needs. + + *********************** + ****** IMPORTANT ****** + *********************** + In all cases you must update the value of version-tag below to read 'custom' instead of an integer, + in order to avoid having your file be overridden by tools such as "android update project" + --> + <!-- version-tag: 1 --> + <import file="${sdk.dir}/tools/ant/build.xml" /> + +</project> diff --git a/third_party/libwebrtc/examples/androidapp/project.properties b/third_party/libwebrtc/examples/androidapp/project.properties new file mode 100644 index 0000000000..a6ca533fe3 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/project.properties @@ -0,0 +1,16 @@ +# This file is automatically generated by Android Tools. +# Do not modify this file -- YOUR CHANGES WILL BE ERASED! +# +# This file must be checked in Version Control Systems. +# +# To customize properties used by the Ant build system edit +# "ant.properties", and override values to adapt the script to your +# project structure. +# +# To enable ProGuard to shrink and obfuscate your code, uncomment this (available properties: sdk.dir, user.home): +#proguard.config=${sdk.dir}/tools/proguard/proguard-android.txt:proguard-project.txt + +# Project target. +target=android-22 + +java.compilerargs=-Xlint:all -Werror diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/disconnect.png b/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/disconnect.png Binary files differnew file mode 100644 index 0000000000..be36174c24 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/disconnect.png diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/ic_action_full_screen.png b/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/ic_action_full_screen.png Binary files differnew file mode 100644 index 0000000000..22f30d31ca --- /dev/null +++ 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b/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_launcher.png Binary files differnew file mode 100644 index 0000000000..a3cd45890c --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_launcher.png diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_loopback_call.png b/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_loopback_call.png Binary files differnew file mode 100644 index 0000000000..39311853b3 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_loopback_call.png diff --git a/third_party/libwebrtc/examples/androidapp/res/layout/activity_call.xml b/third_party/libwebrtc/examples/androidapp/res/layout/activity_call.xml new file mode 100644 index 0000000000..bf811426f3 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/res/layout/activity_call.xml @@ -0,0 +1,34 @@ +<?xml version="1.0" encoding="utf-8"?> + +<!-- tools:ignore is needed because lint thinks this can be replaced with a merge. Replacing this + with a merge causes the fullscreen SurfaceView not to be centered. --> +<FrameLayout + xmlns:android="http://schemas.android.com/apk/res/android" + xmlns:tools="http://schemas.android.com/tools" + android:layout_width="match_parent" + android:layout_height="match_parent" + tools:ignore="MergeRootFrame"> + + <org.webrtc.SurfaceViewRenderer + android:id="@+id/fullscreen_video_view" + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_gravity="center" /> + + <org.webrtc.SurfaceViewRenderer + android:id="@+id/pip_video_view" + android:layout_height="144dp" + android:layout_width="wrap_content" + android:layout_gravity="bottom|end" + android:layout_margin="16dp"/> + + <FrameLayout + android:id="@+id/call_fragment_container" + android:layout_width="match_parent" + android:layout_height="match_parent" /> + <FrameLayout + android:id="@+id/hud_fragment_container" + android:layout_width="match_parent" + android:layout_height="match_parent" /> + +</FrameLayout> diff --git a/third_party/libwebrtc/examples/androidapp/res/layout/activity_connect.xml b/third_party/libwebrtc/examples/androidapp/res/layout/activity_connect.xml new file mode 100644 index 0000000000..017e5cabff --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/res/layout/activity_connect.xml @@ -0,0 +1,80 @@ +<?xml version="1.0" encoding="utf-8"?> +<LinearLayout + xmlns:android="http://schemas.android.com/apk/res/android" + xmlns:tools="http://schemas.android.com/tools" + android:layout_margin="16dp" + android:layout_width="match_parent" + android:layout_height="match_parent" + android:orientation="vertical" + android:weightSum="1" + android:layout_centerHorizontal="true"> + + <TextView + android:id="@+id/room_edittext_description" + android:layout_width="match_parent" + android:layout_height="wrap_content" + android:text="@string/room_description"/> + + <LinearLayout + android:orientation="horizontal" + android:layout_width="match_parent" + android:layout_height="wrap_content" + android:gravity="center" + android:layout_marginBottom="8dp"> + + <!-- TODO(crbug.com/900912): Fix and remove lint ignore --> + <EditText + tools:ignore="LabelFor,Autofill" + android:id="@+id/room_edittext" + android:layout_width="0dp" + android:layout_height="wrap_content" + android:layout_weight="1" + android:maxLines="1" + android:imeOptions="actionDone" + android:inputType="text"/> + + <ImageButton + android:id="@+id/connect_button" + android:layout_width="48dp" + android:layout_height="48dp" + android:contentDescription="@string/connect_description" + android:background="@android:drawable/sym_action_call" /> + + <ImageButton + android:id="@+id/add_favorite_button" + android:layout_width="48dp" + android:layout_height="48dp" + android:contentDescription="@string/add_favorite_description" + android:background="@android:drawable/ic_input_add" /> + </LinearLayout> + + <TextView + android:id="@+id/room_listview_description" + android:layout_width="match_parent" + android:layout_height="48dp" + android:layout_marginTop="8dp" + android:lines="1" + android:maxLines="1" + android:textAppearance="?android:attr/textAppearanceMedium" + android:text="@string/favorites" + android:gravity="center_vertical"/> + + <FrameLayout + android:layout_width="match_parent" + android:layout_height="0dp" + android:layout_weight="1"> + + <ListView + android:id="@+id/room_listview" + android:layout_width="match_parent" + android:layout_height="match_parent" + android:drawSelectorOnTop="false" /> + + <TextView + android:id="@android:id/empty" + android:layout_width="match_parent" + android:layout_height="match_parent" + android:gravity="center" + android:text="@string/no_favorites" /> + </FrameLayout> +</LinearLayout> diff --git a/third_party/libwebrtc/examples/androidapp/res/layout/fragment_call.xml b/third_party/libwebrtc/examples/androidapp/res/layout/fragment_call.xml new file mode 100644 index 0000000000..90b1e9ca0e --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/res/layout/fragment_call.xml @@ -0,0 +1,77 @@ +<?xml version="1.0" encoding="utf-8"?> + +<RelativeLayout + xmlns:android="http://schemas.android.com/apk/res/android" + android:layout_width="match_parent" + android:layout_height="match_parent"> + + <TextView + android:id="@+id/contact_name_call" + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_centerHorizontal="true" + android:layout_above="@+id/buttons_call_container" + android:textSize="24sp" + android:layout_margin="8dp"/> + + <LinearLayout + android:id="@+id/buttons_call_container" + android:orientation="horizontal" + android:layout_above="@+id/capture_format_text_call" + android:layout_alignWithParentIfMissing="true" + android:layout_marginBottom="32dp" + android:layout_centerHorizontal="true" + android:layout_width="wrap_content" + android:layout_height="wrap_content"> + + <ImageButton + android:id="@+id/button_call_disconnect" + android:background="@drawable/disconnect" + android:contentDescription="@string/disconnect_call" + android:layout_marginEnd="16dp" + android:layout_width="48dp" + android:layout_height="48dp"/> + + <ImageButton + android:id="@+id/button_call_switch_camera" + android:background="@android:drawable/ic_menu_camera" + android:contentDescription="@string/switch_camera" + android:layout_marginEnd="8dp" + android:layout_width="48dp" + android:layout_height="48dp"/> + + <ImageButton + android:id="@+id/button_call_scaling_mode" + android:background="@drawable/ic_action_return_from_full_screen" + android:contentDescription="@string/disconnect_call" + android:layout_width="48dp" + android:layout_height="48dp"/> + + <ImageButton + android:id="@+id/button_call_toggle_mic" + android:background="@android:drawable/ic_btn_speak_now" + android:contentDescription="@string/toggle_mic" + android:layout_marginEnd="8dp" + android:layout_width="48dp" + android:layout_height="48dp"/> + </LinearLayout> + + <TextView + android:id="@+id/capture_format_text_call" + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_centerHorizontal="true" + android:layout_above="@+id/capture_format_slider_call" + android:textSize="16sp" + android:text="@string/capture_format_change_text"/> + + <SeekBar + android:id="@+id/capture_format_slider_call" + android:layout_width="match_parent" + android:layout_height="wrap_content" + android:layout_centerHorizontal="true" + android:layout_alignParentBottom="true" + android:progress="50" + android:layout_margin="8dp"/> + +</RelativeLayout> diff --git a/third_party/libwebrtc/examples/androidapp/res/layout/fragment_hud.xml b/third_party/libwebrtc/examples/androidapp/res/layout/fragment_hud.xml new file mode 100644 index 0000000000..483e7ba456 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/res/layout/fragment_hud.xml @@ -0,0 +1,27 @@ +<?xml version="1.0" encoding="utf-8"?> + +<RelativeLayout + xmlns:android="http://schemas.android.com/apk/res/android" + android:layout_width="match_parent" + android:layout_height="match_parent"> + + <ImageButton + android:id="@+id/button_toggle_debug" + android:background="@android:drawable/ic_menu_info_details" + android:contentDescription="@string/toggle_debug" + android:layout_alignParentBottom="true" + android:layout_alignParentStart="true" + android:layout_width="48dp" + android:layout_height="48dp"/> + + <TextView + android:id="@+id/hud_stat_call" + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_alignParentEnd="true" + android:textStyle="bold" + android:textColor="#C000FF00" + android:textSize="12sp" + android:layout_margin="8dp"/> + +</RelativeLayout> diff --git a/third_party/libwebrtc/examples/androidapp/res/menu/connect_menu.xml b/third_party/libwebrtc/examples/androidapp/res/menu/connect_menu.xml new file mode 100644 index 0000000000..a723f54941 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/res/menu/connect_menu.xml @@ -0,0 +1,13 @@ +<menu xmlns:android="http://schemas.android.com/apk/res/android"> + <item + android:id="@+id/action_loopback" + android:icon="@drawable/ic_loopback_call" + android:showAsAction="always" + android:title="@string/action_loopback"/> + <item + android:id="@+id/action_settings" + android:orderInCategory="100" + android:icon="@android:drawable/ic_menu_preferences" + android:showAsAction="ifRoom" + android:title="@string/action_settings"/> +</menu> diff --git a/third_party/libwebrtc/examples/androidapp/res/values-v17/styles.xml b/third_party/libwebrtc/examples/androidapp/res/values-v17/styles.xml new file mode 100644 index 0000000000..969b5012e9 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/res/values-v17/styles.xml @@ -0,0 +1,10 @@ +<?xml version="1.0" encoding="utf-8"?> +<resources> + <style name="AppTheme" parent="android:Theme.Holo" /> + + <style name="CallActivityTheme" parent="android:Theme.Black"> + <item name="android:windowActionBar">false</item> + <item name="android:windowFullscreen">true</item> + <item name="android:windowNoTitle">true</item> + </style> +</resources> diff --git a/third_party/libwebrtc/examples/androidapp/res/values-v21/styles.xml b/third_party/libwebrtc/examples/androidapp/res/values-v21/styles.xml new file mode 100644 index 0000000000..b19af7e38f --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/res/values-v21/styles.xml @@ -0,0 +1,4 @@ +<?xml version="1.0" encoding="utf-8"?> +<resources> + <style name="AppTheme" parent="android:Theme.Material" /> +</resources> diff --git a/third_party/libwebrtc/examples/androidapp/res/values/arrays.xml b/third_party/libwebrtc/examples/androidapp/res/values/arrays.xml new file mode 100644 index 0000000000..4a2948c875 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/res/values/arrays.xml @@ -0,0 +1,61 @@ +<?xml version="1.0" encoding="utf-8"?> +<resources> + <string-array name="videoResolutions"> + <item>Default</item> + <item>4K (3840 x 2160)</item> + <item>Full HD (1920 x 1080)</item> + <item>HD (1280 x 720)</item> + <item>VGA (640 x 480)</item> + <item>QVGA (320 x 240)</item> + </string-array> + + <string-array name="videoResolutionsValues"> + <item>Default</item> + <item>3840 x 2160</item> + <item>1920 x 1080</item> + <item>1280 x 720</item> + <item>640 x 480</item> + <item>320 x 240</item> + </string-array> + + <string-array name="cameraFps"> + <item>Default</item> + <item>30 fps</item> + <item>15 fps</item> + </string-array> + + <string-array name="startBitrate"> + <item>Default</item> + <item>Manual</item> + </string-array> + + <string-array name="videoCodecs"> + <item>VP8</item> + <item>VP9</item> + <item>H264 Baseline</item> + <item>H264 High</item> + <item>AV1</item> + </string-array> + + <string-array name="audioCodecs"> + <item>OPUS</item> + <item>ISAC</item> + </string-array> + + <string-array name="speakerphone"> + <item>Auto (proximity sensor)</item> + <item>Enabled</item> + <item>Disabled</item> + </string-array> + + <string-array name="speakerphoneValues"> + <item>auto</item> + <item>true</item> + <item>false</item> + </string-array> + + <string-array name="roomListContextMenu"> + <item>Remove favorite</item> + </string-array> + +</resources> diff --git a/third_party/libwebrtc/examples/androidapp/res/values/strings.xml b/third_party/libwebrtc/examples/androidapp/res/values/strings.xml new file mode 100644 index 0000000000..814966f200 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/res/values/strings.xml @@ -0,0 +1,224 @@ +<?xml version="1.0" encoding="utf-8"?> +<resources> + <string name="app_name" translatable="false">AppRTC</string> + <string name="settings_name" translatable="false">AppRTC Settings</string> + <string name="disconnect_call">Disconnect Call</string> + <string name="room_description"> + Please enter a room name. Room names are shared with everyone, so think + of something unique and send it to a friend. + </string> + <string name="favorites">Favorites</string> + <string name="no_favorites">No favorites</string> + <string name="invalid_url_title">Invalid URL</string> + <string name="invalid_url_text">The URL or room name you entered resulted in an invalid URL: %1$s + </string> + <string name="channel_error_title">Connection error</string> + <string name="connecting_to">Connecting to: %1$s</string> + <string name="missing_url">FATAL ERROR: Missing URL to connect to.</string> + <string name="camera2_texture_only_error">Camera2 only supports capturing to texture. Either disable Camera2 or enable capturing to texture in the options.</string> + <string name="ok">OK</string> + <string name="switch_camera">Switch front/back camera</string> + <string name="capture_format_change_text">Slide to change capture format</string> + <string name="muted">Muted</string> + <string name="toggle_debug">Toggle debug view</string> + <string name="toggle_mic">Toggle microphone on/off</string> + <string name="action_settings">Settings</string> + <string name="action_loopback">Loopback connection</string> + <string name="connect_description">Connect to the room</string> + <string name="add_favorite_description">Add favorite</string> + <string name="format_description">%1$dx%2$d @ %3$d fps</string> + <string name="missing_permissions_try_again">The application is missing permissions. It might not work correctly. Do you want to try again?</string> + <string name="yes">Yes</string> + <string name="no">No</string> + + <!-- Settings strings. --> + <string name="pref_room_key">room_preference</string> + <string name="pref_room_list_key">room_list_preference</string> + + <string name="pref_videosettings_key">video_settings_key</string> + <string name="pref_videosettings_title">WebRTC video settings.</string> + + <string name="pref_videocall_key">videocall_preference</string> + <string name="pref_videocall_title">Video call.</string> + <string name="pref_videocall_dlg">Enable video in a call.</string> + <string name="pref_videocall_default">true</string> + + <string name="pref_screencapture_key">screencapture_preference</string> + <string name="pref_screencapture_title">Use screencapture.</string> + <string name="pref_screencapture_default">false</string> + + <string name="pref_camera2_key">camera2_preference</string> + <string name="pref_camera2_title">Use Camera2.</string> + <string name="pref_camera2_default">true</string> + <string name="pref_camera2_not_supported">Not supported on this device.</string> + + <string name="pref_resolution_key">resolution_preference</string> + <string name="pref_resolution_title">Video resolution.</string> + <string name="pref_resolution_dlg">Enter AppRTC local video resolution.</string> + <string name="pref_resolution_default">Default</string> + + <string name="pref_fps_key">fps_preference</string> + <string name="pref_fps_title">Camera fps.</string> + <string name="pref_fps_dlg">Enter local camera fps.</string> + <string name="pref_fps_default">Default</string> + + <string name="pref_capturequalityslider_key">capturequalityslider_preference</string> + <string name="pref_capturequalityslider_title">Capture quality slider.</string> + <string name="pref_capturequalityslider_dlg">Enable slider for changing capture quality.</string> + <string name="pref_capturequalityslider_default">false</string> + + <string name="pref_maxvideobitrate_key">maxvideobitrate_preference</string> + <string name="pref_maxvideobitrate_title">Maximum video bitrate setting.</string> + <string name="pref_maxvideobitrate_dlg">Maximum video bitrate setting.</string> + <string name="pref_maxvideobitrate_default">Default</string> + + <string name="pref_maxvideobitratevalue_key">maxvideobitratevalue_preference</string> + <string name="pref_maxvideobitratevalue_title">Video encoder maximum bitrate.</string> + <string name="pref_maxvideobitratevalue_dlg">Enter video encoder maximum bitrate in kbps.</string> + <string name="pref_maxvideobitratevalue_default">1700</string> + + <string name="pref_videocodec_key">videocodec_preference</string> + <string name="pref_videocodec_title">Default video codec.</string> + <string name="pref_videocodec_dlg">Select default video codec.</string> + <string name="pref_videocodec_default">VP8</string> + + <string name="pref_hwcodec_key">hwcodec_preference</string> + <string name="pref_hwcodec_title">Video codec hardware acceleration.</string> + <string name="pref_hwcodec_dlg">Use hardware accelerated video codec (if available).</string> + <string name="pref_hwcodec_default">true</string> + + <string name="pref_capturetotexture_key">capturetotexture_preference</string> + <string name="pref_capturetotexture_title">Video capture to surface texture.</string> + <string name="pref_capturetotexture_dlg">Capture video to textures (if available).</string> + <string name="pref_capturetotexture_default">true</string> + + <string name="pref_flexfec_key">flexfec_preference</string> + <string name="pref_flexfec_title">Codec-agnostic Flexible FEC.</string> + <string name="pref_flexfec_dlg">Enable FlexFEC.</string> + <string name="pref_flexfec_default">false</string> + + <string name="pref_value_enabled">Enabled</string> + <string name="pref_value_disabled">Disabled</string> + + <string name="pref_audiosettings_key">audio_settings_key</string> + <string name="pref_audiosettings_title">WebRTC audio settings.</string> + + <string name="pref_startaudiobitrate_key">startaudiobitrate_preference</string> + <string name="pref_startaudiobitrate_title">Audio bitrate setting.</string> + <string name="pref_startaudiobitrate_dlg">Audio bitrate setting.</string> + <string name="pref_startaudiobitrate_default">Default</string> + + <string name="pref_startaudiobitratevalue_key">startaudiobitratevalue_preference</string> + <string name="pref_startaudiobitratevalue_title">Audio codec bitrate.</string> + <string name="pref_startaudiobitratevalue_dlg">Enter audio codec bitrate in kbps.</string> + <string name="pref_startaudiobitratevalue_default">32</string> + + <string name="pref_audiocodec_key">audiocodec_preference</string> + <string name="pref_audiocodec_title">Default audio codec.</string> + <string name="pref_audiocodec_dlg">Select default audio codec.</string> + <string name="pref_audiocodec_default">OPUS</string> + + <string name="pref_noaudioprocessing_key">audioprocessing_preference</string> + <string name="pref_noaudioprocessing_title">Disable audio processing.</string> + <string name="pref_noaudioprocessing_dlg">Disable audio processing pipeline.</string> + <string name="pref_noaudioprocessing_default">false</string> + + <string name="pref_aecdump_key">aecdump_preference</string> + <string name="pref_aecdump_title">Create aecdump.</string> + <string name="pref_aecdump_dlg">Enable diagnostic audio recordings.</string> + <string name="pref_aecdump_default">false</string> + + <string name="pref_enable_save_input_audio_to_file_key">enable_key</string> + <string name="pref_enable_save_input_audio_to_file_title">Save input audio to file.</string> + <string name="pref_enable_save_input_audio_to_file_dlg">Save input audio to file.</string> + <string name="pref_enable_save_input_audio_to_file_default">false</string> + + <string name="pref_opensles_key">opensles_preference</string> + <string name="pref_opensles_title">Use OpenSL ES for audio playback.</string> + <string name="pref_opensles_dlg">Use OpenSL ES for audio playback.</string> + <string name="pref_opensles_default">false</string> + + <string name="pref_disable_built_in_aec_key">disable_built_in_aec_preference</string> + <string name="pref_disable_built_in_aec_title">Disable hardware AEC.</string> + <string name="pref_disable_built_in_aec_dlg">Disable hardware AEC.</string> + <string name="pref_disable_built_in_aec_default">false</string> + <string name="pref_built_in_aec_not_available">Hardware AEC is not available</string> + + <string name="pref_disable_built_in_agc_key">disable_built_in_agc_preference</string> + <string name="pref_disable_built_in_agc_title">Disable hardware AGC.</string> + <string name="pref_disable_built_in_agc_dlg">Disable hardware AGC.</string> + <string name="pref_disable_built_in_agc_default">false</string> + <string name="pref_built_in_agc_not_available">Hardware AGC is not available</string> + + <string name="pref_disable_built_in_ns_key">disable_built_in_ns_preference</string> + <string name="pref_disable_built_in_ns_title">Disable hardware NS.</string> + <string name="pref_disable_built_in_ns_dlg">Disable hardware NS.</string> + <string name="pref_disable_built_in_ns_default">false</string> + <string name="pref_built_in_ns_not_available">Hardware NS is not available</string> + + <string name="pref_disable_webrtc_agc_and_hpf_key">disable_webrtc_agc_and_hpf_preference</string> + <string name="pref_disable_webrtc_agc_and_hpf_title">Disable WebRTC AGC and HPF.</string> + <string name="pref_disable_webrtc_agc_default">false</string> + + <string name="pref_speakerphone_key">speakerphone_preference</string> + <string name="pref_speakerphone_title">Speakerphone.</string> + <string name="pref_speakerphone_dlg">Speakerphone.</string> + <string name="pref_speakerphone_default">auto</string> + + <string name="pref_datasettings_key">data_settings_key</string> + <string name="pref_datasettings_title">WebRTC data channel settings.</string> + + <string name="pref_enable_datachannel_key">enable_datachannel_preference</string> + <string name="pref_enable_datachannel_title">Enable datachannel.</string> + <string name="pref_enable_datachannel_default" translatable="false">true</string> + + <string name="pref_ordered_key">ordered_preference</string> + <string name="pref_ordered_title">Order messages.</string> + <string name="pref_ordered_default" translatable="false">true</string> + + <string name="pref_data_protocol_key">Subprotocol</string> + <string name="pref_data_protocol_title">Subprotocol.</string> + <string name="pref_data_protocol_dlg">Enter subprotocol.</string> + <string name="pref_data_protocol_default" translatable="false"></string> + + <string name="pref_negotiated_key">negotiated_preference</string> + <string name="pref_negotiated_title">Negotiated.</string> + <string name="pref_negotiated_default" translatable="false">false</string> + + <string name="pref_max_retransmit_time_ms_key">max_retransmit_time_ms_preference</string> + <string name="pref_max_retransmit_time_ms_title">Max delay to retransmit.</string> + <string name="pref_max_retransmit_time_ms_dlg">Enter max delay to retransmit (in ms).</string> + <string name="pref_max_retransmit_time_ms_default" translatable="false">-1</string> + + <string name="pref_max_retransmits_key">max_retransmits_preference</string> + <string name="pref_max_retransmits_title">Max attempts to retransmit.</string> + <string name="pref_max_retransmits_dlg">Enter max attempts to retransmit.</string> + <string name="pref_max_retransmits_default" translatable="false">-1</string> + + <string name="pref_data_id_key">data_id_preference</string> + <string name="pref_data_id_title">Data id.</string> + <string name="pref_data_id_dlg">Enter data channel id.</string> + <string name="pref_data_id_default" translatable="false">-1</string> + + <string name="pref_miscsettings_key">misc_settings_key</string> + <string name="pref_miscsettings_title">Miscellaneous settings.</string> + + <string name="pref_room_server_url_key">room_server_url_preference</string> + <string name="pref_room_server_url_title">Room server URL.</string> + <string name="pref_room_server_url_dlg">Enter a room server URL.</string> + <string name="pref_room_server_url_default" translatable="false">https://appr.tc</string> + + <string name="pref_displayhud_key">displayhud_preference</string> + <string name="pref_displayhud_title">Display call statistics.</string> + <string name="pref_displayhud_dlg">Display call statistics.</string> + <string name="pref_displayhud_default" translatable="false">false</string> + + <string name="pref_tracing_key">tracing_preference</string> + <string name="pref_tracing_title">Debug performance tracing.</string> + <string name="pref_tracing_dlg">Debug performance tracing.</string> + <string name="pref_tracing_default" translatable="false">false</string> + + <string name="pref_enable_rtceventlog_key">enable_rtceventlog_key</string> + <string name="pref_enable_rtceventlog_title">Enable RtcEventLog.</string> + <string name="pref_enable_rtceventlog_default">false</string> +</resources> diff --git a/third_party/libwebrtc/examples/androidapp/res/xml/preferences.xml b/third_party/libwebrtc/examples/androidapp/res/xml/preferences.xml new file mode 100644 index 0000000000..14e74d5c0b --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/res/xml/preferences.xml @@ -0,0 +1,247 @@ +<?xml version="1.0" encoding="utf-8"?> +<PreferenceScreen xmlns:android="http://schemas.android.com/apk/res/android"> + <PreferenceCategory + android:key="@string/pref_videosettings_key" + android:title="@string/pref_videosettings_title"> + + <CheckBoxPreference + android:key="@string/pref_videocall_key" + android:title="@string/pref_videocall_title" + android:dialogTitle="@string/pref_videocall_dlg" + android:defaultValue="@string/pref_videocall_default" /> + + <CheckBoxPreference + android:key="@string/pref_screencapture_key" + android:title="@string/pref_screencapture_title" + android:defaultValue="@string/pref_screencapture_default" /> + + <CheckBoxPreference + android:key="@string/pref_camera2_key" + android:title="@string/pref_camera2_title" + android:defaultValue="@string/pref_camera2_default" /> + + <ListPreference + android:key="@string/pref_resolution_key" + android:title="@string/pref_resolution_title" + android:defaultValue="@string/pref_resolution_default" + android:dialogTitle="@string/pref_resolution_dlg" + android:entries="@array/videoResolutions" + android:entryValues="@array/videoResolutionsValues" /> + + <ListPreference + android:key="@string/pref_fps_key" + android:title="@string/pref_fps_title" + android:defaultValue="@string/pref_fps_default" + android:dialogTitle="@string/pref_fps_dlg" + android:entries="@array/cameraFps" + android:entryValues="@array/cameraFps" /> + + <CheckBoxPreference + android:key="@string/pref_capturequalityslider_key" + android:title="@string/pref_capturequalityslider_title" + android:dialogTitle="@string/pref_capturequalityslider_dlg" + android:defaultValue="@string/pref_capturequalityslider_default" /> + + <ListPreference + android:key="@string/pref_maxvideobitrate_key" + android:title="@string/pref_maxvideobitrate_title" + android:defaultValue="@string/pref_maxvideobitrate_default" + android:dialogTitle="@string/pref_maxvideobitrate_dlg" + android:entries="@array/startBitrate" + android:entryValues="@array/startBitrate" /> + + <EditTextPreference + android:key="@string/pref_maxvideobitratevalue_key" + android:title="@string/pref_maxvideobitratevalue_title" + android:inputType="number" + android:defaultValue="@string/pref_maxvideobitratevalue_default" + android:dialogTitle="@string/pref_maxvideobitratevalue_dlg" /> + + <ListPreference + android:key="@string/pref_videocodec_key" + android:title="@string/pref_videocodec_title" + android:defaultValue="@string/pref_videocodec_default" + android:dialogTitle="@string/pref_videocodec_dlg" + android:entries="@array/videoCodecs" + android:entryValues="@array/videoCodecs" /> + + <CheckBoxPreference + android:key="@string/pref_hwcodec_key" + android:title="@string/pref_hwcodec_title" + android:dialogTitle="@string/pref_hwcodec_dlg" + android:defaultValue="@string/pref_hwcodec_default" /> + + <CheckBoxPreference + android:key="@string/pref_capturetotexture_key" + android:title="@string/pref_capturetotexture_title" + android:dialogTitle="@string/pref_capturetotexture_dlg" + android:defaultValue="@string/pref_capturetotexture_default" /> + + <CheckBoxPreference + android:key="@string/pref_flexfec_key" + android:title="@string/pref_flexfec_title" + android:dialogTitle="@string/pref_flexfec_dlg" + android:defaultValue="@string/pref_flexfec_default" /> + </PreferenceCategory> + + <PreferenceCategory + android:key="@string/pref_audiosettings_key" + android:title="@string/pref_audiosettings_title"> + + <ListPreference + android:key="@string/pref_startaudiobitrate_key" + android:title="@string/pref_startaudiobitrate_title" + android:defaultValue="@string/pref_startaudiobitrate_default" + android:dialogTitle="@string/pref_startaudiobitrate_dlg" + android:entries="@array/startBitrate" + android:entryValues="@array/startBitrate" /> + + <EditTextPreference + android:key="@string/pref_startaudiobitratevalue_key" + android:title="@string/pref_startaudiobitratevalue_title" + android:inputType="number" + android:defaultValue="@string/pref_startaudiobitratevalue_default" + android:dialogTitle="@string/pref_startaudiobitratevalue_dlg" /> + + <ListPreference + android:key="@string/pref_audiocodec_key" + android:title="@string/pref_audiocodec_title" + android:defaultValue="@string/pref_audiocodec_default" + android:dialogTitle="@string/pref_audiocodec_dlg" + android:entries="@array/audioCodecs" + android:entryValues="@array/audioCodecs" /> + + <CheckBoxPreference + android:key="@string/pref_noaudioprocessing_key" + android:title="@string/pref_noaudioprocessing_title" + android:dialogTitle="@string/pref_noaudioprocessing_dlg" + android:defaultValue="@string/pref_noaudioprocessing_default" /> + + <CheckBoxPreference + android:key="@string/pref_aecdump_key" + android:title="@string/pref_aecdump_title" + android:dialogTitle="@string/pref_aecdump_dlg" + android:defaultValue="@string/pref_aecdump_default" /> + + <CheckBoxPreference + android:key="@string/pref_enable_save_input_audio_to_file_key" + android:title="@string/pref_enable_save_input_audio_to_file_title" + android:dialogTitle="@string/pref_enable_save_input_audio_to_file_dlg" + android:defaultValue="@string/pref_enable_save_input_audio_to_file_default" /> + + <CheckBoxPreference + android:key="@string/pref_opensles_key" + android:title="@string/pref_opensles_title" + android:dialogTitle="@string/pref_opensles_dlg" + android:defaultValue="@string/pref_opensles_default" /> + + <CheckBoxPreference + android:key="@string/pref_disable_built_in_aec_key" + android:title="@string/pref_disable_built_in_aec_title" + android:dialogTitle="@string/pref_disable_built_in_aec_dlg" + android:defaultValue="@string/pref_disable_built_in_aec_default" /> + + <CheckBoxPreference + android:key="@string/pref_disable_built_in_agc_key" + android:title="@string/pref_disable_built_in_agc_title" + android:dialogTitle="@string/pref_disable_built_in_agc_dlg" + android:defaultValue="@string/pref_disable_built_in_agc_default" /> + + <CheckBoxPreference + android:key="@string/pref_disable_built_in_ns_key" + android:title="@string/pref_disable_built_in_ns_title" + android:dialogTitle="@string/pref_disable_built_in_ns_dlg" + android:defaultValue="@string/pref_disable_built_in_ns_default" /> + + <CheckBoxPreference + android:key="@string/pref_disable_webrtc_agc_and_hpf_key" + android:title="@string/pref_disable_webrtc_agc_and_hpf_title" + android:defaultValue="@string/pref_disable_webrtc_agc_default" /> + + <ListPreference + android:key="@string/pref_speakerphone_key" + android:title="@string/pref_speakerphone_title" + android:defaultValue="@string/pref_speakerphone_default" + android:dialogTitle="@string/pref_speakerphone_dlg" + android:entries="@array/speakerphone" + android:entryValues="@array/speakerphoneValues" /> + </PreferenceCategory> + + <PreferenceCategory + android:key="@string/pref_datasettings_key" + android:title="@string/pref_datasettings_title"> + + <CheckBoxPreference + android:key="@string/pref_enable_datachannel_key" + android:title="@string/pref_enable_datachannel_title" + android:defaultValue="@string/pref_enable_datachannel_default" /> + + <CheckBoxPreference + android:key="@string/pref_ordered_key" + android:title="@string/pref_ordered_title" + android:defaultValue="@string/pref_ordered_default" /> + + <EditTextPreference + android:key="@string/pref_data_protocol_key" + android:title="@string/pref_data_protocol_title" + android:inputType="text" + android:defaultValue="@string/pref_data_protocol_default" + android:dialogTitle="@string/pref_data_protocol_dlg" /> + + <CheckBoxPreference + android:key="@string/pref_negotiated_key" + android:title="@string/pref_negotiated_title" + android:defaultValue="@string/pref_negotiated_default" /> + + <EditTextPreference + android:key="@string/pref_max_retransmit_time_ms_key" + android:title="@string/pref_max_retransmit_time_ms_title" + android:inputType="number" + android:defaultValue="@string/pref_max_retransmit_time_ms_default" + android:dialogTitle="@string/pref_max_retransmit_time_ms_dlg" /> + + <EditTextPreference + android:key="@string/pref_max_retransmits_key" + android:title="@string/pref_max_retransmits_title" + android:inputType="number" + android:defaultValue="@string/pref_max_retransmits_default" + android:dialogTitle="@string/pref_max_retransmits_dlg" /> + + <EditTextPreference + android:key="@string/pref_data_id_key" + android:title="@string/pref_data_id_title" + android:inputType="number" + android:defaultValue="@string/pref_data_id_default" + android:dialogTitle="@string/pref_data_id_dlg" /> + </PreferenceCategory> + + <PreferenceCategory + android:key="@string/pref_miscsettings_key" + android:title="@string/pref_miscsettings_title"> + + <EditTextPreference + android:key="@string/pref_room_server_url_key" + android:title="@string/pref_room_server_url_title" + android:inputType="text" + android:defaultValue="@string/pref_room_server_url_default" + android:dialogTitle="@string/pref_room_server_url_dlg" /> + + <CheckBoxPreference + android:key="@string/pref_displayhud_key" + android:title="@string/pref_displayhud_title" + android:dialogTitle="@string/pref_displayhud_dlg" + android:defaultValue="@string/pref_displayhud_default" /> + + <CheckBoxPreference + android:key="@string/pref_tracing_key" + android:title="@string/pref_tracing_title" + android:dialogTitle="@string/pref_tracing_dlg" + android:defaultValue="@string/pref_tracing_default" /> + + <CheckBoxPreference + android:key="@string/pref_enable_rtceventlog_key" + android:title="@string/pref_enable_rtceventlog_title" + android:defaultValue="@string/pref_enable_rtceventlog_default"/> + </PreferenceCategory> + +</PreferenceScreen> diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java new file mode 100644 index 0000000000..2536b131a1 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java @@ -0,0 +1,594 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.content.BroadcastReceiver; +import android.content.Context; +import android.content.Intent; +import android.content.IntentFilter; +import android.content.SharedPreferences; +import android.content.pm.PackageManager; +import android.media.AudioDeviceInfo; +import android.media.AudioManager; +import android.os.Build; +import android.preference.PreferenceManager; +import android.util.Log; +import androidx.annotation.Nullable; +import java.util.Collections; +import java.util.HashSet; +import java.util.Set; +import org.appspot.apprtc.util.AppRTCUtils; +import org.webrtc.ThreadUtils; + +/** + * AppRTCAudioManager manages all audio related parts of the AppRTC demo. + */ +public class AppRTCAudioManager { + private static final String TAG = "AppRTCAudioManager"; + private static final String SPEAKERPHONE_AUTO = "auto"; + private static final String SPEAKERPHONE_TRUE = "true"; + private static final String SPEAKERPHONE_FALSE = "false"; + + /** + * AudioDevice is the names of possible audio devices that we currently + * support. + */ + public enum AudioDevice { SPEAKER_PHONE, WIRED_HEADSET, EARPIECE, BLUETOOTH, NONE } + + /** AudioManager state. */ + public enum AudioManagerState { + UNINITIALIZED, + PREINITIALIZED, + RUNNING, + } + + /** Selected audio device change event. */ + public interface AudioManagerEvents { + // Callback fired once audio device is changed or list of available audio devices changed. + void onAudioDeviceChanged( + AudioDevice selectedAudioDevice, Set<AudioDevice> availableAudioDevices); + } + + private final Context apprtcContext; + @Nullable + private AudioManager audioManager; + + @Nullable + private AudioManagerEvents audioManagerEvents; + private AudioManagerState amState; + private int savedAudioMode = AudioManager.MODE_INVALID; + private boolean savedIsSpeakerPhoneOn; + private boolean savedIsMicrophoneMute; + private boolean hasWiredHeadset; + + // Default audio device; speaker phone for video calls or earpiece for audio + // only calls. + private AudioDevice defaultAudioDevice; + + // Contains the currently selected audio device. + // This device is changed automatically using a certain scheme where e.g. + // a wired headset "wins" over speaker phone. It is also possible for a + // user to explicitly select a device (and overrid any predefined scheme). + // See `userSelectedAudioDevice` for details. + private AudioDevice selectedAudioDevice; + + // Contains the user-selected audio device which overrides the predefined + // selection scheme. + // TODO(henrika): always set to AudioDevice.NONE today. Add support for + // explicit selection based on choice by userSelectedAudioDevice. + private AudioDevice userSelectedAudioDevice; + + // Contains speakerphone setting: auto, true or false + @Nullable private final String useSpeakerphone; + + // Proximity sensor object. It measures the proximity of an object in cm + // relative to the view screen of a device and can therefore be used to + // assist device switching (close to ear <=> use headset earpiece if + // available, far from ear <=> use speaker phone). + @Nullable private AppRTCProximitySensor proximitySensor; + + // Handles all tasks related to Bluetooth headset devices. + private final AppRTCBluetoothManager bluetoothManager; + + // Contains a list of available audio devices. A Set collection is used to + // avoid duplicate elements. + private Set<AudioDevice> audioDevices = new HashSet<>(); + + // Broadcast receiver for wired headset intent broadcasts. + private BroadcastReceiver wiredHeadsetReceiver; + + // Callback method for changes in audio focus. + @Nullable + private AudioManager.OnAudioFocusChangeListener audioFocusChangeListener; + + /** + * This method is called when the proximity sensor reports a state change, + * e.g. from "NEAR to FAR" or from "FAR to NEAR". + */ + private void onProximitySensorChangedState() { + if (!useSpeakerphone.equals(SPEAKERPHONE_AUTO)) { + return; + } + + // The proximity sensor should only be activated when there are exactly two + // available audio devices. + if (audioDevices.size() == 2 && audioDevices.contains(AppRTCAudioManager.AudioDevice.EARPIECE) + && audioDevices.contains(AppRTCAudioManager.AudioDevice.SPEAKER_PHONE)) { + if (proximitySensor.sensorReportsNearState()) { + // Sensor reports that a "handset is being held up to a person's ear", + // or "something is covering the light sensor". + setAudioDeviceInternal(AppRTCAudioManager.AudioDevice.EARPIECE); + } else { + // Sensor reports that a "handset is removed from a person's ear", or + // "the light sensor is no longer covered". + setAudioDeviceInternal(AppRTCAudioManager.AudioDevice.SPEAKER_PHONE); + } + } + } + + /* Receiver which handles changes in wired headset availability. */ + private class WiredHeadsetReceiver extends BroadcastReceiver { + private static final int STATE_UNPLUGGED = 0; + private static final int STATE_PLUGGED = 1; + private static final int HAS_NO_MIC = 0; + private static final int HAS_MIC = 1; + + @Override + public void onReceive(Context context, Intent intent) { + int state = intent.getIntExtra("state", STATE_UNPLUGGED); + int microphone = intent.getIntExtra("microphone", HAS_NO_MIC); + String name = intent.getStringExtra("name"); + Log.d(TAG, "WiredHeadsetReceiver.onReceive" + AppRTCUtils.getThreadInfo() + ": " + + "a=" + intent.getAction() + ", s=" + + (state == STATE_UNPLUGGED ? "unplugged" : "plugged") + ", m=" + + (microphone == HAS_MIC ? "mic" : "no mic") + ", n=" + name + ", sb=" + + isInitialStickyBroadcast()); + hasWiredHeadset = (state == STATE_PLUGGED); + updateAudioDeviceState(); + } + } + + /** Construction. */ + static AppRTCAudioManager create(Context context) { + return new AppRTCAudioManager(context); + } + + private AppRTCAudioManager(Context context) { + Log.d(TAG, "ctor"); + ThreadUtils.checkIsOnMainThread(); + apprtcContext = context; + audioManager = ((AudioManager) context.getSystemService(Context.AUDIO_SERVICE)); + bluetoothManager = AppRTCBluetoothManager.create(context, this); + wiredHeadsetReceiver = new WiredHeadsetReceiver(); + amState = AudioManagerState.UNINITIALIZED; + + SharedPreferences sharedPreferences = PreferenceManager.getDefaultSharedPreferences(context); + useSpeakerphone = sharedPreferences.getString(context.getString(R.string.pref_speakerphone_key), + context.getString(R.string.pref_speakerphone_default)); + Log.d(TAG, "useSpeakerphone: " + useSpeakerphone); + if (useSpeakerphone.equals(SPEAKERPHONE_FALSE)) { + defaultAudioDevice = AudioDevice.EARPIECE; + } else { + defaultAudioDevice = AudioDevice.SPEAKER_PHONE; + } + + // Create and initialize the proximity sensor. + // Tablet devices (e.g. Nexus 7) does not support proximity sensors. + // Note that, the sensor will not be active until start() has been called. + proximitySensor = AppRTCProximitySensor.create(context, + // This method will be called each time a state change is detected. + // Example: user holds their hand over the device (closer than ~5 cm), + // or removes their hand from the device. + this ::onProximitySensorChangedState); + + Log.d(TAG, "defaultAudioDevice: " + defaultAudioDevice); + AppRTCUtils.logDeviceInfo(TAG); + } + + @SuppressWarnings("deprecation") // TODO(henrika): audioManager.requestAudioFocus() is deprecated. + public void start(AudioManagerEvents audioManagerEvents) { + Log.d(TAG, "start"); + ThreadUtils.checkIsOnMainThread(); + if (amState == AudioManagerState.RUNNING) { + Log.e(TAG, "AudioManager is already active"); + return; + } + // TODO(henrika): perhaps call new method called preInitAudio() here if UNINITIALIZED. + + Log.d(TAG, "AudioManager starts..."); + this.audioManagerEvents = audioManagerEvents; + amState = AudioManagerState.RUNNING; + + // Store current audio state so we can restore it when stop() is called. + savedAudioMode = audioManager.getMode(); + savedIsSpeakerPhoneOn = audioManager.isSpeakerphoneOn(); + savedIsMicrophoneMute = audioManager.isMicrophoneMute(); + hasWiredHeadset = hasWiredHeadset(); + + // Create an AudioManager.OnAudioFocusChangeListener instance. + audioFocusChangeListener = new AudioManager.OnAudioFocusChangeListener() { + // Called on the listener to notify if the audio focus for this listener has been changed. + // The `focusChange` value indicates whether the focus was gained, whether the focus was lost, + // and whether that loss is transient, or whether the new focus holder will hold it for an + // unknown amount of time. + // TODO(henrika): possibly extend support of handling audio-focus changes. Only contains + // logging for now. + @Override + public void onAudioFocusChange(int focusChange) { + final String typeOfChange; + switch (focusChange) { + case AudioManager.AUDIOFOCUS_GAIN: + typeOfChange = "AUDIOFOCUS_GAIN"; + break; + case AudioManager.AUDIOFOCUS_GAIN_TRANSIENT: + typeOfChange = "AUDIOFOCUS_GAIN_TRANSIENT"; + break; + case AudioManager.AUDIOFOCUS_GAIN_TRANSIENT_EXCLUSIVE: + typeOfChange = "AUDIOFOCUS_GAIN_TRANSIENT_EXCLUSIVE"; + break; + case AudioManager.AUDIOFOCUS_GAIN_TRANSIENT_MAY_DUCK: + typeOfChange = "AUDIOFOCUS_GAIN_TRANSIENT_MAY_DUCK"; + break; + case AudioManager.AUDIOFOCUS_LOSS: + typeOfChange = "AUDIOFOCUS_LOSS"; + break; + case AudioManager.AUDIOFOCUS_LOSS_TRANSIENT: + typeOfChange = "AUDIOFOCUS_LOSS_TRANSIENT"; + break; + case AudioManager.AUDIOFOCUS_LOSS_TRANSIENT_CAN_DUCK: + typeOfChange = "AUDIOFOCUS_LOSS_TRANSIENT_CAN_DUCK"; + break; + default: + typeOfChange = "AUDIOFOCUS_INVALID"; + break; + } + Log.d(TAG, "onAudioFocusChange: " + typeOfChange); + } + }; + + // Request audio playout focus (without ducking) and install listener for changes in focus. + int result = audioManager.requestAudioFocus(audioFocusChangeListener, + AudioManager.STREAM_VOICE_CALL, AudioManager.AUDIOFOCUS_GAIN_TRANSIENT); + if (result == AudioManager.AUDIOFOCUS_REQUEST_GRANTED) { + Log.d(TAG, "Audio focus request granted for VOICE_CALL streams"); + } else { + Log.e(TAG, "Audio focus request failed"); + } + + // Start by setting MODE_IN_COMMUNICATION as default audio mode. It is + // required to be in this mode when playout and/or recording starts for + // best possible VoIP performance. + audioManager.setMode(AudioManager.MODE_IN_COMMUNICATION); + + // Always disable microphone mute during a WebRTC call. + setMicrophoneMute(false); + + // Set initial device states. + userSelectedAudioDevice = AudioDevice.NONE; + selectedAudioDevice = AudioDevice.NONE; + audioDevices.clear(); + + // Initialize and start Bluetooth if a BT device is available or initiate + // detection of new (enabled) BT devices. + bluetoothManager.start(); + + // Do initial selection of audio device. This setting can later be changed + // either by adding/removing a BT or wired headset or by covering/uncovering + // the proximity sensor. + updateAudioDeviceState(); + + // Register receiver for broadcast intents related to adding/removing a + // wired headset. + registerReceiver(wiredHeadsetReceiver, new IntentFilter(Intent.ACTION_HEADSET_PLUG)); + Log.d(TAG, "AudioManager started"); + } + + @SuppressWarnings("deprecation") // TODO(henrika): audioManager.abandonAudioFocus() is deprecated. + public void stop() { + Log.d(TAG, "stop"); + ThreadUtils.checkIsOnMainThread(); + if (amState != AudioManagerState.RUNNING) { + Log.e(TAG, "Trying to stop AudioManager in incorrect state: " + amState); + return; + } + amState = AudioManagerState.UNINITIALIZED; + + unregisterReceiver(wiredHeadsetReceiver); + + bluetoothManager.stop(); + + // Restore previously stored audio states. + setSpeakerphoneOn(savedIsSpeakerPhoneOn); + setMicrophoneMute(savedIsMicrophoneMute); + audioManager.setMode(savedAudioMode); + + // Abandon audio focus. Gives the previous focus owner, if any, focus. + audioManager.abandonAudioFocus(audioFocusChangeListener); + audioFocusChangeListener = null; + Log.d(TAG, "Abandoned audio focus for VOICE_CALL streams"); + + if (proximitySensor != null) { + proximitySensor.stop(); + proximitySensor = null; + } + + audioManagerEvents = null; + Log.d(TAG, "AudioManager stopped"); + } + + /** Changes selection of the currently active audio device. */ + private void setAudioDeviceInternal(AudioDevice device) { + Log.d(TAG, "setAudioDeviceInternal(device=" + device + ")"); + AppRTCUtils.assertIsTrue(audioDevices.contains(device)); + + switch (device) { + case SPEAKER_PHONE: + setSpeakerphoneOn(true); + break; + case EARPIECE: + setSpeakerphoneOn(false); + break; + case WIRED_HEADSET: + setSpeakerphoneOn(false); + break; + case BLUETOOTH: + setSpeakerphoneOn(false); + break; + default: + Log.e(TAG, "Invalid audio device selection"); + break; + } + selectedAudioDevice = device; + } + + /** + * Changes default audio device. + * TODO(henrika): add usage of this method in the AppRTCMobile client. + */ + public void setDefaultAudioDevice(AudioDevice defaultDevice) { + ThreadUtils.checkIsOnMainThread(); + switch (defaultDevice) { + case SPEAKER_PHONE: + defaultAudioDevice = defaultDevice; + break; + case EARPIECE: + if (hasEarpiece()) { + defaultAudioDevice = defaultDevice; + } else { + defaultAudioDevice = AudioDevice.SPEAKER_PHONE; + } + break; + default: + Log.e(TAG, "Invalid default audio device selection"); + break; + } + Log.d(TAG, "setDefaultAudioDevice(device=" + defaultAudioDevice + ")"); + updateAudioDeviceState(); + } + + /** Changes selection of the currently active audio device. */ + public void selectAudioDevice(AudioDevice device) { + ThreadUtils.checkIsOnMainThread(); + if (!audioDevices.contains(device)) { + Log.e(TAG, "Can not select " + device + " from available " + audioDevices); + } + userSelectedAudioDevice = device; + updateAudioDeviceState(); + } + + /** Returns current set of available/selectable audio devices. */ + public Set<AudioDevice> getAudioDevices() { + ThreadUtils.checkIsOnMainThread(); + return Collections.unmodifiableSet(new HashSet<>(audioDevices)); + } + + /** Returns the currently selected audio device. */ + public AudioDevice getSelectedAudioDevice() { + ThreadUtils.checkIsOnMainThread(); + return selectedAudioDevice; + } + + /** Helper method for receiver registration. */ + private void registerReceiver(BroadcastReceiver receiver, IntentFilter filter) { + apprtcContext.registerReceiver(receiver, filter); + } + + /** Helper method for unregistration of an existing receiver. */ + private void unregisterReceiver(BroadcastReceiver receiver) { + apprtcContext.unregisterReceiver(receiver); + } + + /** Sets the speaker phone mode. */ + private void setSpeakerphoneOn(boolean on) { + boolean wasOn = audioManager.isSpeakerphoneOn(); + if (wasOn == on) { + return; + } + audioManager.setSpeakerphoneOn(on); + } + + /** Sets the microphone mute state. */ + private void setMicrophoneMute(boolean on) { + boolean wasMuted = audioManager.isMicrophoneMute(); + if (wasMuted == on) { + return; + } + audioManager.setMicrophoneMute(on); + } + + /** Gets the current earpiece state. */ + private boolean hasEarpiece() { + return apprtcContext.getPackageManager().hasSystemFeature(PackageManager.FEATURE_TELEPHONY); + } + + /** + * Checks whether a wired headset is connected or not. + * This is not a valid indication that audio playback is actually over + * the wired headset as audio routing depends on other conditions. We + * only use it as an early indicator (during initialization) of an attached + * wired headset. + */ + @Deprecated + private boolean hasWiredHeadset() { + if (Build.VERSION.SDK_INT < Build.VERSION_CODES.M) { + return audioManager.isWiredHeadsetOn(); + } else { + final AudioDeviceInfo[] devices = audioManager.getDevices(AudioManager.GET_DEVICES_ALL); + for (AudioDeviceInfo device : devices) { + final int type = device.getType(); + if (type == AudioDeviceInfo.TYPE_WIRED_HEADSET) { + Log.d(TAG, "hasWiredHeadset: found wired headset"); + return true; + } else if (type == AudioDeviceInfo.TYPE_USB_DEVICE) { + Log.d(TAG, "hasWiredHeadset: found USB audio device"); + return true; + } + } + return false; + } + } + + /** + * Updates list of possible audio devices and make new device selection. + * TODO(henrika): add unit test to verify all state transitions. + */ + public void updateAudioDeviceState() { + ThreadUtils.checkIsOnMainThread(); + Log.d(TAG, "--- updateAudioDeviceState: " + + "wired headset=" + hasWiredHeadset + ", " + + "BT state=" + bluetoothManager.getState()); + Log.d(TAG, "Device status: " + + "available=" + audioDevices + ", " + + "selected=" + selectedAudioDevice + ", " + + "user selected=" + userSelectedAudioDevice); + + // Check if any Bluetooth headset is connected. The internal BT state will + // change accordingly. + // TODO(henrika): perhaps wrap required state into BT manager. + if (bluetoothManager.getState() == AppRTCBluetoothManager.State.HEADSET_AVAILABLE + || bluetoothManager.getState() == AppRTCBluetoothManager.State.HEADSET_UNAVAILABLE + || bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_DISCONNECTING) { + bluetoothManager.updateDevice(); + } + + // Update the set of available audio devices. + Set<AudioDevice> newAudioDevices = new HashSet<>(); + + if (bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_CONNECTED + || bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_CONNECTING + || bluetoothManager.getState() == AppRTCBluetoothManager.State.HEADSET_AVAILABLE) { + newAudioDevices.add(AudioDevice.BLUETOOTH); + } + + if (hasWiredHeadset) { + // If a wired headset is connected, then it is the only possible option. + newAudioDevices.add(AudioDevice.WIRED_HEADSET); + } else { + // No wired headset, hence the audio-device list can contain speaker + // phone (on a tablet), or speaker phone and earpiece (on mobile phone). + newAudioDevices.add(AudioDevice.SPEAKER_PHONE); + if (hasEarpiece()) { + newAudioDevices.add(AudioDevice.EARPIECE); + } + } + // Store state which is set to true if the device list has changed. + boolean audioDeviceSetUpdated = !audioDevices.equals(newAudioDevices); + // Update the existing audio device set. + audioDevices = newAudioDevices; + // Correct user selected audio devices if needed. + if (bluetoothManager.getState() == AppRTCBluetoothManager.State.HEADSET_UNAVAILABLE + && userSelectedAudioDevice == AudioDevice.BLUETOOTH) { + // If BT is not available, it can't be the user selection. + userSelectedAudioDevice = AudioDevice.NONE; + } + if (hasWiredHeadset && userSelectedAudioDevice == AudioDevice.SPEAKER_PHONE) { + // If user selected speaker phone, but then plugged wired headset then make + // wired headset as user selected device. + userSelectedAudioDevice = AudioDevice.WIRED_HEADSET; + } + if (!hasWiredHeadset && userSelectedAudioDevice == AudioDevice.WIRED_HEADSET) { + // If user selected wired headset, but then unplugged wired headset then make + // speaker phone as user selected device. + userSelectedAudioDevice = AudioDevice.SPEAKER_PHONE; + } + + // Need to start Bluetooth if it is available and user either selected it explicitly or + // user did not select any output device. + boolean needBluetoothAudioStart = + bluetoothManager.getState() == AppRTCBluetoothManager.State.HEADSET_AVAILABLE + && (userSelectedAudioDevice == AudioDevice.NONE + || userSelectedAudioDevice == AudioDevice.BLUETOOTH); + + // Need to stop Bluetooth audio if user selected different device and + // Bluetooth SCO connection is established or in the process. + boolean needBluetoothAudioStop = + (bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_CONNECTED + || bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_CONNECTING) + && (userSelectedAudioDevice != AudioDevice.NONE + && userSelectedAudioDevice != AudioDevice.BLUETOOTH); + + if (bluetoothManager.getState() == AppRTCBluetoothManager.State.HEADSET_AVAILABLE + || bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_CONNECTING + || bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_CONNECTED) { + Log.d(TAG, "Need BT audio: start=" + needBluetoothAudioStart + ", " + + "stop=" + needBluetoothAudioStop + ", " + + "BT state=" + bluetoothManager.getState()); + } + + // Start or stop Bluetooth SCO connection given states set earlier. + if (needBluetoothAudioStop) { + bluetoothManager.stopScoAudio(); + bluetoothManager.updateDevice(); + } + + if (needBluetoothAudioStart && !needBluetoothAudioStop) { + // Attempt to start Bluetooth SCO audio (takes a few second to start). + if (!bluetoothManager.startScoAudio()) { + // Remove BLUETOOTH from list of available devices since SCO failed. + audioDevices.remove(AudioDevice.BLUETOOTH); + audioDeviceSetUpdated = true; + } + } + + // Update selected audio device. + final AudioDevice newAudioDevice; + + if (bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_CONNECTED) { + // If a Bluetooth is connected, then it should be used as output audio + // device. Note that it is not sufficient that a headset is available; + // an active SCO channel must also be up and running. + newAudioDevice = AudioDevice.BLUETOOTH; + } else if (hasWiredHeadset) { + // If a wired headset is connected, but Bluetooth is not, then wired headset is used as + // audio device. + newAudioDevice = AudioDevice.WIRED_HEADSET; + } else { + // No wired headset and no Bluetooth, hence the audio-device list can contain speaker + // phone (on a tablet), or speaker phone and earpiece (on mobile phone). + // `defaultAudioDevice` contains either AudioDevice.SPEAKER_PHONE or AudioDevice.EARPIECE + // depending on the user's selection. + newAudioDevice = defaultAudioDevice; + } + // Switch to new device but only if there has been any changes. + if (newAudioDevice != selectedAudioDevice || audioDeviceSetUpdated) { + // Do the required device switch. + setAudioDeviceInternal(newAudioDevice); + Log.d(TAG, "New device status: " + + "available=" + audioDevices + ", " + + "selected=" + newAudioDevice); + if (audioManagerEvents != null) { + // Notify a listening client that audio device has been changed. + audioManagerEvents.onAudioDeviceChanged(selectedAudioDevice, audioDevices); + } + } + Log.d(TAG, "--- updateAudioDeviceState done"); + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCBluetoothManager.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCBluetoothManager.java new file mode 100644 index 0000000000..e9077d8bd6 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCBluetoothManager.java @@ -0,0 +1,532 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.annotation.SuppressLint; +import android.bluetooth.BluetoothAdapter; +import android.bluetooth.BluetoothDevice; +import android.bluetooth.BluetoothHeadset; +import android.bluetooth.BluetoothProfile; +import android.content.BroadcastReceiver; +import android.content.Context; +import android.content.Intent; +import android.content.IntentFilter; +import android.content.pm.PackageManager; +import android.media.AudioManager; +import android.os.Handler; +import android.os.Looper; +import android.os.Process; +import android.util.Log; +import androidx.annotation.Nullable; +import java.util.List; +import java.util.Set; +import org.appspot.apprtc.util.AppRTCUtils; +import org.webrtc.ThreadUtils; + +/** + * AppRTCProximitySensor manages functions related to Bluetoth devices in the + * AppRTC demo. + */ +public class AppRTCBluetoothManager { + private static final String TAG = "AppRTCBluetoothManager"; + + // Timeout interval for starting or stopping audio to a Bluetooth SCO device. + private static final int BLUETOOTH_SCO_TIMEOUT_MS = 4000; + // Maximum number of SCO connection attempts. + private static final int MAX_SCO_CONNECTION_ATTEMPTS = 2; + + // Bluetooth connection state. + public enum State { + // Bluetooth is not available; no adapter or Bluetooth is off. + UNINITIALIZED, + // Bluetooth error happened when trying to start Bluetooth. + ERROR, + // Bluetooth proxy object for the Headset profile exists, but no connected headset devices, + // SCO is not started or disconnected. + HEADSET_UNAVAILABLE, + // Bluetooth proxy object for the Headset profile connected, connected Bluetooth headset + // present, but SCO is not started or disconnected. + HEADSET_AVAILABLE, + // Bluetooth audio SCO connection with remote device is closing. + SCO_DISCONNECTING, + // Bluetooth audio SCO connection with remote device is initiated. + SCO_CONNECTING, + // Bluetooth audio SCO connection with remote device is established. + SCO_CONNECTED + } + + private final Context apprtcContext; + private final AppRTCAudioManager apprtcAudioManager; + @Nullable + private final AudioManager audioManager; + private final Handler handler; + + int scoConnectionAttempts; + private State bluetoothState; + private final BluetoothProfile.ServiceListener bluetoothServiceListener; + @Nullable + private BluetoothAdapter bluetoothAdapter; + @Nullable + private BluetoothHeadset bluetoothHeadset; + @Nullable + private BluetoothDevice bluetoothDevice; + private final BroadcastReceiver bluetoothHeadsetReceiver; + + // Runs when the Bluetooth timeout expires. We use that timeout after calling + // startScoAudio() or stopScoAudio() because we're not guaranteed to get a + // callback after those calls. + private final Runnable bluetoothTimeoutRunnable = new Runnable() { + @Override + public void run() { + bluetoothTimeout(); + } + }; + + /** + * Implementation of an interface that notifies BluetoothProfile IPC clients when they have been + * connected to or disconnected from the service. + */ + private class BluetoothServiceListener implements BluetoothProfile.ServiceListener { + @Override + // Called to notify the client when the proxy object has been connected to the service. + // Once we have the profile proxy object, we can use it to monitor the state of the + // connection and perform other operations that are relevant to the headset profile. + public void onServiceConnected(int profile, BluetoothProfile proxy) { + if (profile != BluetoothProfile.HEADSET || bluetoothState == State.UNINITIALIZED) { + return; + } + Log.d(TAG, "BluetoothServiceListener.onServiceConnected: BT state=" + bluetoothState); + // Android only supports one connected Bluetooth Headset at a time. + bluetoothHeadset = (BluetoothHeadset) proxy; + updateAudioDeviceState(); + Log.d(TAG, "onServiceConnected done: BT state=" + bluetoothState); + } + + @Override + /** Notifies the client when the proxy object has been disconnected from the service. */ + public void onServiceDisconnected(int profile) { + if (profile != BluetoothProfile.HEADSET || bluetoothState == State.UNINITIALIZED) { + return; + } + Log.d(TAG, "BluetoothServiceListener.onServiceDisconnected: BT state=" + bluetoothState); + stopScoAudio(); + bluetoothHeadset = null; + bluetoothDevice = null; + bluetoothState = State.HEADSET_UNAVAILABLE; + updateAudioDeviceState(); + Log.d(TAG, "onServiceDisconnected done: BT state=" + bluetoothState); + } + } + + // Intent broadcast receiver which handles changes in Bluetooth device availability. + // Detects headset changes and Bluetooth SCO state changes. + private class BluetoothHeadsetBroadcastReceiver extends BroadcastReceiver { + @Override + public void onReceive(Context context, Intent intent) { + if (bluetoothState == State.UNINITIALIZED) { + return; + } + final String action = intent.getAction(); + // Change in connection state of the Headset profile. Note that the + // change does not tell us anything about whether we're streaming + // audio to BT over SCO. Typically received when user turns on a BT + // headset while audio is active using another audio device. + if (action.equals(BluetoothHeadset.ACTION_CONNECTION_STATE_CHANGED)) { + final int state = + intent.getIntExtra(BluetoothHeadset.EXTRA_STATE, BluetoothHeadset.STATE_DISCONNECTED); + Log.d(TAG, "BluetoothHeadsetBroadcastReceiver.onReceive: " + + "a=ACTION_CONNECTION_STATE_CHANGED, " + + "s=" + stateToString(state) + ", " + + "sb=" + isInitialStickyBroadcast() + ", " + + "BT state: " + bluetoothState); + if (state == BluetoothHeadset.STATE_CONNECTED) { + scoConnectionAttempts = 0; + updateAudioDeviceState(); + } else if (state == BluetoothHeadset.STATE_CONNECTING) { + // No action needed. + } else if (state == BluetoothHeadset.STATE_DISCONNECTING) { + // No action needed. + } else if (state == BluetoothHeadset.STATE_DISCONNECTED) { + // Bluetooth is probably powered off during the call. + stopScoAudio(); + updateAudioDeviceState(); + } + // Change in the audio (SCO) connection state of the Headset profile. + // Typically received after call to startScoAudio() has finalized. + } else if (action.equals(BluetoothHeadset.ACTION_AUDIO_STATE_CHANGED)) { + final int state = intent.getIntExtra( + BluetoothHeadset.EXTRA_STATE, BluetoothHeadset.STATE_AUDIO_DISCONNECTED); + Log.d(TAG, "BluetoothHeadsetBroadcastReceiver.onReceive: " + + "a=ACTION_AUDIO_STATE_CHANGED, " + + "s=" + stateToString(state) + ", " + + "sb=" + isInitialStickyBroadcast() + ", " + + "BT state: " + bluetoothState); + if (state == BluetoothHeadset.STATE_AUDIO_CONNECTED) { + cancelTimer(); + if (bluetoothState == State.SCO_CONNECTING) { + Log.d(TAG, "+++ Bluetooth audio SCO is now connected"); + bluetoothState = State.SCO_CONNECTED; + scoConnectionAttempts = 0; + updateAudioDeviceState(); + } else { + Log.w(TAG, "Unexpected state BluetoothHeadset.STATE_AUDIO_CONNECTED"); + } + } else if (state == BluetoothHeadset.STATE_AUDIO_CONNECTING) { + Log.d(TAG, "+++ Bluetooth audio SCO is now connecting..."); + } else if (state == BluetoothHeadset.STATE_AUDIO_DISCONNECTED) { + Log.d(TAG, "+++ Bluetooth audio SCO is now disconnected"); + if (isInitialStickyBroadcast()) { + Log.d(TAG, "Ignore STATE_AUDIO_DISCONNECTED initial sticky broadcast."); + return; + } + updateAudioDeviceState(); + } + } + Log.d(TAG, "onReceive done: BT state=" + bluetoothState); + } + } + + /** Construction. */ + static AppRTCBluetoothManager create(Context context, AppRTCAudioManager audioManager) { + Log.d(TAG, "create" + AppRTCUtils.getThreadInfo()); + return new AppRTCBluetoothManager(context, audioManager); + } + + protected AppRTCBluetoothManager(Context context, AppRTCAudioManager audioManager) { + Log.d(TAG, "ctor"); + ThreadUtils.checkIsOnMainThread(); + apprtcContext = context; + apprtcAudioManager = audioManager; + this.audioManager = getAudioManager(context); + bluetoothState = State.UNINITIALIZED; + bluetoothServiceListener = new BluetoothServiceListener(); + bluetoothHeadsetReceiver = new BluetoothHeadsetBroadcastReceiver(); + handler = new Handler(Looper.getMainLooper()); + } + + /** Returns the internal state. */ + public State getState() { + ThreadUtils.checkIsOnMainThread(); + return bluetoothState; + } + + /** + * Activates components required to detect Bluetooth devices and to enable + * BT SCO (audio is routed via BT SCO) for the headset profile. The end + * state will be HEADSET_UNAVAILABLE but a state machine has started which + * will start a state change sequence where the final outcome depends on + * if/when the BT headset is enabled. + * Example of state change sequence when start() is called while BT device + * is connected and enabled: + * UNINITIALIZED --> HEADSET_UNAVAILABLE --> HEADSET_AVAILABLE --> + * SCO_CONNECTING --> SCO_CONNECTED <==> audio is now routed via BT SCO. + * Note that the AppRTCAudioManager is also involved in driving this state + * change. + */ + public void start() { + ThreadUtils.checkIsOnMainThread(); + Log.d(TAG, "start"); + if (!hasPermission(apprtcContext, android.Manifest.permission.BLUETOOTH)) { + Log.w(TAG, "Process (pid=" + Process.myPid() + ") lacks BLUETOOTH permission"); + return; + } + if (bluetoothState != State.UNINITIALIZED) { + Log.w(TAG, "Invalid BT state"); + return; + } + bluetoothHeadset = null; + bluetoothDevice = null; + scoConnectionAttempts = 0; + // Get a handle to the default local Bluetooth adapter. + bluetoothAdapter = BluetoothAdapter.getDefaultAdapter(); + if (bluetoothAdapter == null) { + Log.w(TAG, "Device does not support Bluetooth"); + return; + } + // Ensure that the device supports use of BT SCO audio for off call use cases. + if (!audioManager.isBluetoothScoAvailableOffCall()) { + Log.e(TAG, "Bluetooth SCO audio is not available off call"); + return; + } + logBluetoothAdapterInfo(bluetoothAdapter); + // Establish a connection to the HEADSET profile (includes both Bluetooth Headset and + // Hands-Free) proxy object and install a listener. + if (!getBluetoothProfileProxy( + apprtcContext, bluetoothServiceListener, BluetoothProfile.HEADSET)) { + Log.e(TAG, "BluetoothAdapter.getProfileProxy(HEADSET) failed"); + return; + } + // Register receivers for BluetoothHeadset change notifications. + IntentFilter bluetoothHeadsetFilter = new IntentFilter(); + // Register receiver for change in connection state of the Headset profile. + bluetoothHeadsetFilter.addAction(BluetoothHeadset.ACTION_CONNECTION_STATE_CHANGED); + // Register receiver for change in audio connection state of the Headset profile. + bluetoothHeadsetFilter.addAction(BluetoothHeadset.ACTION_AUDIO_STATE_CHANGED); + registerReceiver(bluetoothHeadsetReceiver, bluetoothHeadsetFilter); + Log.d(TAG, "HEADSET profile state: " + + stateToString(bluetoothAdapter.getProfileConnectionState(BluetoothProfile.HEADSET))); + Log.d(TAG, "Bluetooth proxy for headset profile has started"); + bluetoothState = State.HEADSET_UNAVAILABLE; + Log.d(TAG, "start done: BT state=" + bluetoothState); + } + + /** Stops and closes all components related to Bluetooth audio. */ + public void stop() { + ThreadUtils.checkIsOnMainThread(); + Log.d(TAG, "stop: BT state=" + bluetoothState); + if (bluetoothAdapter == null) { + return; + } + // Stop BT SCO connection with remote device if needed. + stopScoAudio(); + // Close down remaining BT resources. + if (bluetoothState == State.UNINITIALIZED) { + return; + } + unregisterReceiver(bluetoothHeadsetReceiver); + cancelTimer(); + if (bluetoothHeadset != null) { + bluetoothAdapter.closeProfileProxy(BluetoothProfile.HEADSET, bluetoothHeadset); + bluetoothHeadset = null; + } + bluetoothAdapter = null; + bluetoothDevice = null; + bluetoothState = State.UNINITIALIZED; + Log.d(TAG, "stop done: BT state=" + bluetoothState); + } + + /** + * Starts Bluetooth SCO connection with remote device. + * Note that the phone application always has the priority on the usage of the SCO connection + * for telephony. If this method is called while the phone is in call it will be ignored. + * Similarly, if a call is received or sent while an application is using the SCO connection, + * the connection will be lost for the application and NOT returned automatically when the call + * ends. Also note that: up to and including API version JELLY_BEAN_MR1, this method initiates a + * virtual voice call to the Bluetooth headset. After API version JELLY_BEAN_MR2 only a raw SCO + * audio connection is established. + * TODO(henrika): should we add support for virtual voice call to BT headset also for JBMR2 and + * higher. It might be required to initiates a virtual voice call since many devices do not + * accept SCO audio without a "call". + */ + public boolean startScoAudio() { + ThreadUtils.checkIsOnMainThread(); + Log.d(TAG, "startSco: BT state=" + bluetoothState + ", " + + "attempts: " + scoConnectionAttempts + ", " + + "SCO is on: " + isScoOn()); + if (scoConnectionAttempts >= MAX_SCO_CONNECTION_ATTEMPTS) { + Log.e(TAG, "BT SCO connection fails - no more attempts"); + return false; + } + if (bluetoothState != State.HEADSET_AVAILABLE) { + Log.e(TAG, "BT SCO connection fails - no headset available"); + return false; + } + // Start BT SCO channel and wait for ACTION_AUDIO_STATE_CHANGED. + Log.d(TAG, "Starting Bluetooth SCO and waits for ACTION_AUDIO_STATE_CHANGED..."); + // The SCO connection establishment can take several seconds, hence we cannot rely on the + // connection to be available when the method returns but instead register to receive the + // intent ACTION_SCO_AUDIO_STATE_UPDATED and wait for the state to be SCO_AUDIO_STATE_CONNECTED. + bluetoothState = State.SCO_CONNECTING; + audioManager.startBluetoothSco(); + audioManager.setBluetoothScoOn(true); + scoConnectionAttempts++; + startTimer(); + Log.d(TAG, "startScoAudio done: BT state=" + bluetoothState + ", " + + "SCO is on: " + isScoOn()); + return true; + } + + /** Stops Bluetooth SCO connection with remote device. */ + public void stopScoAudio() { + ThreadUtils.checkIsOnMainThread(); + Log.d(TAG, "stopScoAudio: BT state=" + bluetoothState + ", " + + "SCO is on: " + isScoOn()); + if (bluetoothState != State.SCO_CONNECTING && bluetoothState != State.SCO_CONNECTED) { + return; + } + cancelTimer(); + audioManager.stopBluetoothSco(); + audioManager.setBluetoothScoOn(false); + bluetoothState = State.SCO_DISCONNECTING; + Log.d(TAG, "stopScoAudio done: BT state=" + bluetoothState + ", " + + "SCO is on: " + isScoOn()); + } + + /** + * Use the BluetoothHeadset proxy object (controls the Bluetooth Headset + * Service via IPC) to update the list of connected devices for the HEADSET + * profile. The internal state will change to HEADSET_UNAVAILABLE or to + * HEADSET_AVAILABLE and `bluetoothDevice` will be mapped to the connected + * device if available. + */ + public void updateDevice() { + if (bluetoothState == State.UNINITIALIZED || bluetoothHeadset == null) { + return; + } + Log.d(TAG, "updateDevice"); + // Get connected devices for the headset profile. Returns the set of + // devices which are in state STATE_CONNECTED. The BluetoothDevice class + // is just a thin wrapper for a Bluetooth hardware address. + List<BluetoothDevice> devices = bluetoothHeadset.getConnectedDevices(); + if (devices.isEmpty()) { + bluetoothDevice = null; + bluetoothState = State.HEADSET_UNAVAILABLE; + Log.d(TAG, "No connected bluetooth headset"); + } else { + // Always use first device in list. Android only supports one device. + bluetoothDevice = devices.get(0); + bluetoothState = State.HEADSET_AVAILABLE; + Log.d(TAG, "Connected bluetooth headset: " + + "name=" + bluetoothDevice.getName() + ", " + + "state=" + stateToString(bluetoothHeadset.getConnectionState(bluetoothDevice)) + + ", SCO audio=" + bluetoothHeadset.isAudioConnected(bluetoothDevice)); + } + Log.d(TAG, "updateDevice done: BT state=" + bluetoothState); + } + + /** + * Stubs for test mocks. + */ + @Nullable + protected AudioManager getAudioManager(Context context) { + return (AudioManager) context.getSystemService(Context.AUDIO_SERVICE); + } + + protected void registerReceiver(BroadcastReceiver receiver, IntentFilter filter) { + apprtcContext.registerReceiver(receiver, filter); + } + + protected void unregisterReceiver(BroadcastReceiver receiver) { + apprtcContext.unregisterReceiver(receiver); + } + + protected boolean getBluetoothProfileProxy( + Context context, BluetoothProfile.ServiceListener listener, int profile) { + return bluetoothAdapter.getProfileProxy(context, listener, profile); + } + + protected boolean hasPermission(Context context, String permission) { + return apprtcContext.checkPermission(permission, Process.myPid(), Process.myUid()) + == PackageManager.PERMISSION_GRANTED; + } + + /** Logs the state of the local Bluetooth adapter. */ + @SuppressLint("HardwareIds") + protected void logBluetoothAdapterInfo(BluetoothAdapter localAdapter) { + Log.d(TAG, "BluetoothAdapter: " + + "enabled=" + localAdapter.isEnabled() + ", " + + "state=" + stateToString(localAdapter.getState()) + ", " + + "name=" + localAdapter.getName() + ", " + + "address=" + localAdapter.getAddress()); + // Log the set of BluetoothDevice objects that are bonded (paired) to the local adapter. + Set<BluetoothDevice> pairedDevices = localAdapter.getBondedDevices(); + if (!pairedDevices.isEmpty()) { + Log.d(TAG, "paired devices:"); + for (BluetoothDevice device : pairedDevices) { + Log.d(TAG, " name=" + device.getName() + ", address=" + device.getAddress()); + } + } + } + + /** Ensures that the audio manager updates its list of available audio devices. */ + private void updateAudioDeviceState() { + ThreadUtils.checkIsOnMainThread(); + Log.d(TAG, "updateAudioDeviceState"); + apprtcAudioManager.updateAudioDeviceState(); + } + + /** Starts timer which times out after BLUETOOTH_SCO_TIMEOUT_MS milliseconds. */ + private void startTimer() { + ThreadUtils.checkIsOnMainThread(); + Log.d(TAG, "startTimer"); + handler.postDelayed(bluetoothTimeoutRunnable, BLUETOOTH_SCO_TIMEOUT_MS); + } + + /** Cancels any outstanding timer tasks. */ + private void cancelTimer() { + ThreadUtils.checkIsOnMainThread(); + Log.d(TAG, "cancelTimer"); + handler.removeCallbacks(bluetoothTimeoutRunnable); + } + + /** + * Called when start of the BT SCO channel takes too long time. Usually + * happens when the BT device has been turned on during an ongoing call. + */ + private void bluetoothTimeout() { + ThreadUtils.checkIsOnMainThread(); + if (bluetoothState == State.UNINITIALIZED || bluetoothHeadset == null) { + return; + } + Log.d(TAG, "bluetoothTimeout: BT state=" + bluetoothState + ", " + + "attempts: " + scoConnectionAttempts + ", " + + "SCO is on: " + isScoOn()); + if (bluetoothState != State.SCO_CONNECTING) { + return; + } + // Bluetooth SCO should be connecting; check the latest result. + boolean scoConnected = false; + List<BluetoothDevice> devices = bluetoothHeadset.getConnectedDevices(); + if (devices.size() > 0) { + bluetoothDevice = devices.get(0); + if (bluetoothHeadset.isAudioConnected(bluetoothDevice)) { + Log.d(TAG, "SCO connected with " + bluetoothDevice.getName()); + scoConnected = true; + } else { + Log.d(TAG, "SCO is not connected with " + bluetoothDevice.getName()); + } + } + if (scoConnected) { + // We thought BT had timed out, but it's actually on; updating state. + bluetoothState = State.SCO_CONNECTED; + scoConnectionAttempts = 0; + } else { + // Give up and "cancel" our request by calling stopBluetoothSco(). + Log.w(TAG, "BT failed to connect after timeout"); + stopScoAudio(); + } + updateAudioDeviceState(); + Log.d(TAG, "bluetoothTimeout done: BT state=" + bluetoothState); + } + + /** Checks whether audio uses Bluetooth SCO. */ + private boolean isScoOn() { + return audioManager.isBluetoothScoOn(); + } + + /** Converts BluetoothAdapter states into local string representations. */ + private String stateToString(int state) { + switch (state) { + case BluetoothAdapter.STATE_DISCONNECTED: + return "DISCONNECTED"; + case BluetoothAdapter.STATE_CONNECTED: + return "CONNECTED"; + case BluetoothAdapter.STATE_CONNECTING: + return "CONNECTING"; + case BluetoothAdapter.STATE_DISCONNECTING: + return "DISCONNECTING"; + case BluetoothAdapter.STATE_OFF: + return "OFF"; + case BluetoothAdapter.STATE_ON: + return "ON"; + case BluetoothAdapter.STATE_TURNING_OFF: + // Indicates the local Bluetooth adapter is turning off. Local clients should immediately + // attempt graceful disconnection of any remote links. + return "TURNING_OFF"; + case BluetoothAdapter.STATE_TURNING_ON: + // Indicates the local Bluetooth adapter is turning on. However local clients should wait + // for STATE_ON before attempting to use the adapter. + return "TURNING_ON"; + default: + return "INVALID"; + } + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCClient.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCClient.java new file mode 100644 index 0000000000..d5b7b4338e --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCClient.java @@ -0,0 +1,137 @@ +/* + * Copyright 2013 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import org.webrtc.IceCandidate; +import org.webrtc.PeerConnection; +import org.webrtc.SessionDescription; + +import java.util.List; + +/** + * AppRTCClient is the interface representing an AppRTC client. + */ +public interface AppRTCClient { + /** + * Struct holding the connection parameters of an AppRTC room. + */ + class RoomConnectionParameters { + public final String roomUrl; + public final String roomId; + public final boolean loopback; + public final String urlParameters; + public RoomConnectionParameters( + String roomUrl, String roomId, boolean loopback, String urlParameters) { + this.roomUrl = roomUrl; + this.roomId = roomId; + this.loopback = loopback; + this.urlParameters = urlParameters; + } + public RoomConnectionParameters(String roomUrl, String roomId, boolean loopback) { + this(roomUrl, roomId, loopback, null /* urlParameters */); + } + } + + /** + * Asynchronously connect to an AppRTC room URL using supplied connection + * parameters. Once connection is established onConnectedToRoom() + * callback with room parameters is invoked. + */ + void connectToRoom(RoomConnectionParameters connectionParameters); + + /** + * Send offer SDP to the other participant. + */ + void sendOfferSdp(final SessionDescription sdp); + + /** + * Send answer SDP to the other participant. + */ + void sendAnswerSdp(final SessionDescription sdp); + + /** + * Send Ice candidate to the other participant. + */ + void sendLocalIceCandidate(final IceCandidate candidate); + + /** + * Send removed ICE candidates to the other participant. + */ + void sendLocalIceCandidateRemovals(final IceCandidate[] candidates); + + /** + * Disconnect from room. + */ + void disconnectFromRoom(); + + /** + * Struct holding the signaling parameters of an AppRTC room. + */ + class SignalingParameters { + public final List<PeerConnection.IceServer> iceServers; + public final boolean initiator; + public final String clientId; + public final String wssUrl; + public final String wssPostUrl; + public final SessionDescription offerSdp; + public final List<IceCandidate> iceCandidates; + + public SignalingParameters(List<PeerConnection.IceServer> iceServers, boolean initiator, + String clientId, String wssUrl, String wssPostUrl, SessionDescription offerSdp, + List<IceCandidate> iceCandidates) { + this.iceServers = iceServers; + this.initiator = initiator; + this.clientId = clientId; + this.wssUrl = wssUrl; + this.wssPostUrl = wssPostUrl; + this.offerSdp = offerSdp; + this.iceCandidates = iceCandidates; + } + } + + /** + * Callback interface for messages delivered on signaling channel. + * + * <p>Methods are guaranteed to be invoked on the UI thread of `activity`. + */ + interface SignalingEvents { + /** + * Callback fired once the room's signaling parameters + * SignalingParameters are extracted. + */ + void onConnectedToRoom(final SignalingParameters params); + + /** + * Callback fired once remote SDP is received. + */ + void onRemoteDescription(final SessionDescription sdp); + + /** + * Callback fired once remote Ice candidate is received. + */ + void onRemoteIceCandidate(final IceCandidate candidate); + + /** + * Callback fired once remote Ice candidate removals are received. + */ + void onRemoteIceCandidatesRemoved(final IceCandidate[] candidates); + + /** + * Callback fired once channel is closed. + */ + void onChannelClose(); + + /** + * Callback fired once channel error happened. + */ + void onChannelError(final String description); + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCProximitySensor.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCProximitySensor.java new file mode 100644 index 0000000000..604e2863d9 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCProximitySensor.java @@ -0,0 +1,158 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.content.Context; +import android.hardware.Sensor; +import android.hardware.SensorEvent; +import android.hardware.SensorEventListener; +import android.hardware.SensorManager; +import android.os.Build; +import android.util.Log; +import androidx.annotation.Nullable; +import org.appspot.apprtc.util.AppRTCUtils; +import org.webrtc.ThreadUtils; + +/** + * AppRTCProximitySensor manages functions related to the proximity sensor in + * the AppRTC demo. + * On most device, the proximity sensor is implemented as a boolean-sensor. + * It returns just two values "NEAR" or "FAR". Thresholding is done on the LUX + * value i.e. the LUX value of the light sensor is compared with a threshold. + * A LUX-value more than the threshold means the proximity sensor returns "FAR". + * Anything less than the threshold value and the sensor returns "NEAR". + */ +public class AppRTCProximitySensor implements SensorEventListener { + private static final String TAG = "AppRTCProximitySensor"; + + // This class should be created, started and stopped on one thread + // (e.g. the main thread). We use `nonThreadSafe` to ensure that this is + // the case. Only active when `DEBUG` is set to true. + private final ThreadUtils.ThreadChecker threadChecker = new ThreadUtils.ThreadChecker(); + + private final Runnable onSensorStateListener; + private final SensorManager sensorManager; + @Nullable private Sensor proximitySensor; + private boolean lastStateReportIsNear; + + /** Construction */ + static AppRTCProximitySensor create(Context context, Runnable sensorStateListener) { + return new AppRTCProximitySensor(context, sensorStateListener); + } + + private AppRTCProximitySensor(Context context, Runnable sensorStateListener) { + Log.d(TAG, "AppRTCProximitySensor" + AppRTCUtils.getThreadInfo()); + onSensorStateListener = sensorStateListener; + sensorManager = ((SensorManager) context.getSystemService(Context.SENSOR_SERVICE)); + } + + /** + * Activate the proximity sensor. Also do initialization if called for the + * first time. + */ + public boolean start() { + threadChecker.checkIsOnValidThread(); + Log.d(TAG, "start" + AppRTCUtils.getThreadInfo()); + if (!initDefaultSensor()) { + // Proximity sensor is not supported on this device. + return false; + } + sensorManager.registerListener(this, proximitySensor, SensorManager.SENSOR_DELAY_NORMAL); + return true; + } + + /** Deactivate the proximity sensor. */ + public void stop() { + threadChecker.checkIsOnValidThread(); + Log.d(TAG, "stop" + AppRTCUtils.getThreadInfo()); + if (proximitySensor == null) { + return; + } + sensorManager.unregisterListener(this, proximitySensor); + } + + /** Getter for last reported state. Set to true if "near" is reported. */ + public boolean sensorReportsNearState() { + threadChecker.checkIsOnValidThread(); + return lastStateReportIsNear; + } + + @Override + public final void onAccuracyChanged(Sensor sensor, int accuracy) { + threadChecker.checkIsOnValidThread(); + AppRTCUtils.assertIsTrue(sensor.getType() == Sensor.TYPE_PROXIMITY); + if (accuracy == SensorManager.SENSOR_STATUS_UNRELIABLE) { + Log.e(TAG, "The values returned by this sensor cannot be trusted"); + } + } + + @Override + public final void onSensorChanged(SensorEvent event) { + threadChecker.checkIsOnValidThread(); + AppRTCUtils.assertIsTrue(event.sensor.getType() == Sensor.TYPE_PROXIMITY); + // As a best practice; do as little as possible within this method and + // avoid blocking. + float distanceInCentimeters = event.values[0]; + if (distanceInCentimeters < proximitySensor.getMaximumRange()) { + Log.d(TAG, "Proximity sensor => NEAR state"); + lastStateReportIsNear = true; + } else { + Log.d(TAG, "Proximity sensor => FAR state"); + lastStateReportIsNear = false; + } + + // Report about new state to listening client. Client can then call + // sensorReportsNearState() to query the current state (NEAR or FAR). + if (onSensorStateListener != null) { + onSensorStateListener.run(); + } + + Log.d(TAG, "onSensorChanged" + AppRTCUtils.getThreadInfo() + ": " + + "accuracy=" + event.accuracy + ", timestamp=" + event.timestamp + ", distance=" + + event.values[0]); + } + + /** + * Get default proximity sensor if it exists. Tablet devices (e.g. Nexus 7) + * does not support this type of sensor and false will be returned in such + * cases. + */ + private boolean initDefaultSensor() { + if (proximitySensor != null) { + return true; + } + proximitySensor = sensorManager.getDefaultSensor(Sensor.TYPE_PROXIMITY); + if (proximitySensor == null) { + return false; + } + logProximitySensorInfo(); + return true; + } + + /** Helper method for logging information about the proximity sensor. */ + private void logProximitySensorInfo() { + if (proximitySensor == null) { + return; + } + StringBuilder info = new StringBuilder("Proximity sensor: "); + info.append("name=").append(proximitySensor.getName()); + info.append(", vendor: ").append(proximitySensor.getVendor()); + info.append(", power: ").append(proximitySensor.getPower()); + info.append(", resolution: ").append(proximitySensor.getResolution()); + info.append(", max range: ").append(proximitySensor.getMaximumRange()); + info.append(", min delay: ").append(proximitySensor.getMinDelay()); + info.append(", type: ").append(proximitySensor.getStringType()); + info.append(", max delay: ").append(proximitySensor.getMaxDelay()); + info.append(", reporting mode: ").append(proximitySensor.getReportingMode()); + info.append(", isWakeUpSensor: ").append(proximitySensor.isWakeUpSensor()); + Log.d(TAG, info.toString()); + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java new file mode 100644 index 0000000000..eb5ee8289e --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java @@ -0,0 +1,962 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.annotation.TargetApi; +import android.app.Activity; +import android.app.AlertDialog; +import android.app.FragmentTransaction; +import android.content.Context; +import android.content.DialogInterface; +import android.content.Intent; +import android.content.pm.PackageManager; +import android.media.projection.MediaProjection; +import android.media.projection.MediaProjectionManager; +import android.net.Uri; +import android.os.Build; +import android.os.Bundle; +import android.os.Handler; +import android.util.DisplayMetrics; +import android.util.Log; +import android.view.View; +import android.view.Window; +import android.view.WindowManager; +import android.view.WindowManager.LayoutParams; +import android.widget.Toast; +import androidx.annotation.Nullable; +import java.io.IOException; +import java.lang.RuntimeException; +import java.util.ArrayList; +import java.util.List; +import java.util.Set; +import org.appspot.apprtc.AppRTCAudioManager.AudioDevice; +import org.appspot.apprtc.AppRTCAudioManager.AudioManagerEvents; +import org.appspot.apprtc.AppRTCClient.RoomConnectionParameters; +import org.appspot.apprtc.AppRTCClient.SignalingParameters; +import org.appspot.apprtc.PeerConnectionClient.DataChannelParameters; +import org.appspot.apprtc.PeerConnectionClient.PeerConnectionParameters; +import org.webrtc.Camera1Enumerator; +import org.webrtc.Camera2Enumerator; +import org.webrtc.CameraEnumerator; +import org.webrtc.EglBase; +import org.webrtc.FileVideoCapturer; +import org.webrtc.IceCandidate; +import org.webrtc.Logging; +import org.webrtc.PeerConnectionFactory; +import org.webrtc.RTCStatsReport; +import org.webrtc.RendererCommon.ScalingType; +import org.webrtc.ScreenCapturerAndroid; +import org.webrtc.SessionDescription; +import org.webrtc.SurfaceViewRenderer; +import org.webrtc.VideoCapturer; +import org.webrtc.VideoFileRenderer; +import org.webrtc.VideoFrame; +import org.webrtc.VideoSink; + +/** + * Activity for peer connection call setup, call waiting + * and call view. + */ +public class CallActivity extends Activity implements AppRTCClient.SignalingEvents, + PeerConnectionClient.PeerConnectionEvents, + CallFragment.OnCallEvents { + private static final String TAG = "CallRTCClient"; + + public static final String EXTRA_ROOMID = "org.appspot.apprtc.ROOMID"; + public static final String EXTRA_URLPARAMETERS = "org.appspot.apprtc.URLPARAMETERS"; + public static final String EXTRA_LOOPBACK = "org.appspot.apprtc.LOOPBACK"; + public static final String EXTRA_VIDEO_CALL = "org.appspot.apprtc.VIDEO_CALL"; + public static final String EXTRA_SCREENCAPTURE = "org.appspot.apprtc.SCREENCAPTURE"; + public static final String EXTRA_CAMERA2 = "org.appspot.apprtc.CAMERA2"; + public static final String EXTRA_VIDEO_WIDTH = "org.appspot.apprtc.VIDEO_WIDTH"; + public static final String EXTRA_VIDEO_HEIGHT = "org.appspot.apprtc.VIDEO_HEIGHT"; + public static final String EXTRA_VIDEO_FPS = "org.appspot.apprtc.VIDEO_FPS"; + public static final String EXTRA_VIDEO_CAPTUREQUALITYSLIDER_ENABLED = + "org.appsopt.apprtc.VIDEO_CAPTUREQUALITYSLIDER"; + public static final String EXTRA_VIDEO_BITRATE = "org.appspot.apprtc.VIDEO_BITRATE"; + public static final String EXTRA_VIDEOCODEC = "org.appspot.apprtc.VIDEOCODEC"; + public static final String EXTRA_HWCODEC_ENABLED = "org.appspot.apprtc.HWCODEC"; + public static final String EXTRA_CAPTURETOTEXTURE_ENABLED = "org.appspot.apprtc.CAPTURETOTEXTURE"; + public static final String EXTRA_FLEXFEC_ENABLED = "org.appspot.apprtc.FLEXFEC"; + public static final String EXTRA_AUDIO_BITRATE = "org.appspot.apprtc.AUDIO_BITRATE"; + public static final String EXTRA_AUDIOCODEC = "org.appspot.apprtc.AUDIOCODEC"; + public static final String EXTRA_NOAUDIOPROCESSING_ENABLED = + "org.appspot.apprtc.NOAUDIOPROCESSING"; + public static final String EXTRA_AECDUMP_ENABLED = "org.appspot.apprtc.AECDUMP"; + public static final String EXTRA_SAVE_INPUT_AUDIO_TO_FILE_ENABLED = + "org.appspot.apprtc.SAVE_INPUT_AUDIO_TO_FILE"; + public static final String EXTRA_OPENSLES_ENABLED = "org.appspot.apprtc.OPENSLES"; + public static final String EXTRA_DISABLE_BUILT_IN_AEC = "org.appspot.apprtc.DISABLE_BUILT_IN_AEC"; + public static final String EXTRA_DISABLE_BUILT_IN_AGC = "org.appspot.apprtc.DISABLE_BUILT_IN_AGC"; + public static final String EXTRA_DISABLE_BUILT_IN_NS = "org.appspot.apprtc.DISABLE_BUILT_IN_NS"; + public static final String EXTRA_DISABLE_WEBRTC_AGC_AND_HPF = + "org.appspot.apprtc.DISABLE_WEBRTC_GAIN_CONTROL"; + public static final String EXTRA_DISPLAY_HUD = "org.appspot.apprtc.DISPLAY_HUD"; + public static final String EXTRA_TRACING = "org.appspot.apprtc.TRACING"; + public static final String EXTRA_CMDLINE = "org.appspot.apprtc.CMDLINE"; + public static final String EXTRA_RUNTIME = "org.appspot.apprtc.RUNTIME"; + public static final String EXTRA_VIDEO_FILE_AS_CAMERA = "org.appspot.apprtc.VIDEO_FILE_AS_CAMERA"; + public static final String EXTRA_SAVE_REMOTE_VIDEO_TO_FILE = + "org.appspot.apprtc.SAVE_REMOTE_VIDEO_TO_FILE"; + public static final String EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_WIDTH = + "org.appspot.apprtc.SAVE_REMOTE_VIDEO_TO_FILE_WIDTH"; + public static final String EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_HEIGHT = + "org.appspot.apprtc.SAVE_REMOTE_VIDEO_TO_FILE_HEIGHT"; + public static final String EXTRA_USE_VALUES_FROM_INTENT = + "org.appspot.apprtc.USE_VALUES_FROM_INTENT"; + public static final String EXTRA_DATA_CHANNEL_ENABLED = "org.appspot.apprtc.DATA_CHANNEL_ENABLED"; + public static final String EXTRA_ORDERED = "org.appspot.apprtc.ORDERED"; + public static final String EXTRA_MAX_RETRANSMITS_MS = "org.appspot.apprtc.MAX_RETRANSMITS_MS"; + public static final String EXTRA_MAX_RETRANSMITS = "org.appspot.apprtc.MAX_RETRANSMITS"; + public static final String EXTRA_PROTOCOL = "org.appspot.apprtc.PROTOCOL"; + public static final String EXTRA_NEGOTIATED = "org.appspot.apprtc.NEGOTIATED"; + public static final String EXTRA_ID = "org.appspot.apprtc.ID"; + public static final String EXTRA_ENABLE_RTCEVENTLOG = "org.appspot.apprtc.ENABLE_RTCEVENTLOG"; + + private static final int CAPTURE_PERMISSION_REQUEST_CODE = 1; + + // List of mandatory application permissions. + private static final String[] MANDATORY_PERMISSIONS = {"android.permission.MODIFY_AUDIO_SETTINGS", + "android.permission.RECORD_AUDIO", "android.permission.INTERNET"}; + + // Peer connection statistics callback period in ms. + private static final int STAT_CALLBACK_PERIOD = 1000; + + private static class ProxyVideoSink implements VideoSink { + private VideoSink target; + + @Override + synchronized public void onFrame(VideoFrame frame) { + if (target == null) { + Logging.d(TAG, "Dropping frame in proxy because target is null."); + return; + } + + target.onFrame(frame); + } + + synchronized public void setTarget(VideoSink target) { + this.target = target; + } + } + + private final ProxyVideoSink remoteProxyRenderer = new ProxyVideoSink(); + private final ProxyVideoSink localProxyVideoSink = new ProxyVideoSink(); + @Nullable private PeerConnectionClient peerConnectionClient; + @Nullable + private AppRTCClient appRtcClient; + @Nullable + private SignalingParameters signalingParameters; + @Nullable private AppRTCAudioManager audioManager; + @Nullable + private SurfaceViewRenderer pipRenderer; + @Nullable + private SurfaceViewRenderer fullscreenRenderer; + @Nullable + private VideoFileRenderer videoFileRenderer; + private final List<VideoSink> remoteSinks = new ArrayList<>(); + private Toast logToast; + private boolean commandLineRun; + private boolean activityRunning; + private RoomConnectionParameters roomConnectionParameters; + @Nullable + private PeerConnectionParameters peerConnectionParameters; + private boolean connected; + private boolean isError; + private boolean callControlFragmentVisible = true; + private long callStartedTimeMs; + private boolean micEnabled = true; + private boolean screencaptureEnabled; + private static Intent mediaProjectionPermissionResultData; + private static int mediaProjectionPermissionResultCode; + // True if local view is in the fullscreen renderer. + private boolean isSwappedFeeds; + + // Controls + private CallFragment callFragment; + private HudFragment hudFragment; + private CpuMonitor cpuMonitor; + + @Override + // TODO(bugs.webrtc.org/8580): LayoutParams.FLAG_TURN_SCREEN_ON and + // LayoutParams.FLAG_SHOW_WHEN_LOCKED are deprecated. + @SuppressWarnings("deprecation") + public void onCreate(Bundle savedInstanceState) { + super.onCreate(savedInstanceState); + Thread.setDefaultUncaughtExceptionHandler(new UnhandledExceptionHandler(this)); + + // Set window styles for fullscreen-window size. Needs to be done before + // adding content. + requestWindowFeature(Window.FEATURE_NO_TITLE); + getWindow().addFlags(LayoutParams.FLAG_FULLSCREEN | LayoutParams.FLAG_KEEP_SCREEN_ON + | LayoutParams.FLAG_SHOW_WHEN_LOCKED | LayoutParams.FLAG_TURN_SCREEN_ON); + getWindow().getDecorView().setSystemUiVisibility(getSystemUiVisibility()); + setContentView(R.layout.activity_call); + + connected = false; + signalingParameters = null; + + // Create UI controls. + pipRenderer = findViewById(R.id.pip_video_view); + fullscreenRenderer = findViewById(R.id.fullscreen_video_view); + callFragment = new CallFragment(); + hudFragment = new HudFragment(); + + // Show/hide call control fragment on view click. + View.OnClickListener listener = new View.OnClickListener() { + @Override + public void onClick(View view) { + toggleCallControlFragmentVisibility(); + } + }; + + // Swap feeds on pip view click. + pipRenderer.setOnClickListener(new View.OnClickListener() { + @Override + public void onClick(View view) { + setSwappedFeeds(!isSwappedFeeds); + } + }); + + fullscreenRenderer.setOnClickListener(listener); + remoteSinks.add(remoteProxyRenderer); + + final Intent intent = getIntent(); + final EglBase eglBase = EglBase.create(); + + // Create video renderers. + pipRenderer.init(eglBase.getEglBaseContext(), null); + pipRenderer.setScalingType(ScalingType.SCALE_ASPECT_FIT); + String saveRemoteVideoToFile = intent.getStringExtra(EXTRA_SAVE_REMOTE_VIDEO_TO_FILE); + + // When saveRemoteVideoToFile is set we save the video from the remote to a file. + if (saveRemoteVideoToFile != null) { + int videoOutWidth = intent.getIntExtra(EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_WIDTH, 0); + int videoOutHeight = intent.getIntExtra(EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_HEIGHT, 0); + try { + videoFileRenderer = new VideoFileRenderer( + saveRemoteVideoToFile, videoOutWidth, videoOutHeight, eglBase.getEglBaseContext()); + remoteSinks.add(videoFileRenderer); + } catch (IOException e) { + throw new RuntimeException( + "Failed to open video file for output: " + saveRemoteVideoToFile, e); + } + } + fullscreenRenderer.init(eglBase.getEglBaseContext(), null); + fullscreenRenderer.setScalingType(ScalingType.SCALE_ASPECT_FILL); + + pipRenderer.setZOrderMediaOverlay(true); + pipRenderer.setEnableHardwareScaler(true /* enabled */); + fullscreenRenderer.setEnableHardwareScaler(false /* enabled */); + // Start with local feed in fullscreen and swap it to the pip when the call is connected. + setSwappedFeeds(true /* isSwappedFeeds */); + + // Check for mandatory permissions. + for (String permission : MANDATORY_PERMISSIONS) { + if (checkCallingOrSelfPermission(permission) != PackageManager.PERMISSION_GRANTED) { + logAndToast("Permission " + permission + " is not granted"); + setResult(RESULT_CANCELED); + finish(); + return; + } + } + + Uri roomUri = intent.getData(); + if (roomUri == null) { + logAndToast(getString(R.string.missing_url)); + Log.e(TAG, "Didn't get any URL in intent!"); + setResult(RESULT_CANCELED); + finish(); + return; + } + + // Get Intent parameters. + String roomId = intent.getStringExtra(EXTRA_ROOMID); + Log.d(TAG, "Room ID: " + roomId); + if (roomId == null || roomId.length() == 0) { + logAndToast(getString(R.string.missing_url)); + Log.e(TAG, "Incorrect room ID in intent!"); + setResult(RESULT_CANCELED); + finish(); + return; + } + + boolean loopback = intent.getBooleanExtra(EXTRA_LOOPBACK, false); + boolean tracing = intent.getBooleanExtra(EXTRA_TRACING, false); + + int videoWidth = intent.getIntExtra(EXTRA_VIDEO_WIDTH, 0); + int videoHeight = intent.getIntExtra(EXTRA_VIDEO_HEIGHT, 0); + + screencaptureEnabled = intent.getBooleanExtra(EXTRA_SCREENCAPTURE, false); + // If capturing format is not specified for screencapture, use screen resolution. + if (screencaptureEnabled && videoWidth == 0 && videoHeight == 0) { + DisplayMetrics displayMetrics = getDisplayMetrics(); + videoWidth = displayMetrics.widthPixels; + videoHeight = displayMetrics.heightPixels; + } + DataChannelParameters dataChannelParameters = null; + if (intent.getBooleanExtra(EXTRA_DATA_CHANNEL_ENABLED, false)) { + dataChannelParameters = new DataChannelParameters(intent.getBooleanExtra(EXTRA_ORDERED, true), + intent.getIntExtra(EXTRA_MAX_RETRANSMITS_MS, -1), + intent.getIntExtra(EXTRA_MAX_RETRANSMITS, -1), intent.getStringExtra(EXTRA_PROTOCOL), + intent.getBooleanExtra(EXTRA_NEGOTIATED, false), intent.getIntExtra(EXTRA_ID, -1)); + } + peerConnectionParameters = + new PeerConnectionParameters(intent.getBooleanExtra(EXTRA_VIDEO_CALL, true), loopback, + tracing, videoWidth, videoHeight, intent.getIntExtra(EXTRA_VIDEO_FPS, 0), + intent.getIntExtra(EXTRA_VIDEO_BITRATE, 0), intent.getStringExtra(EXTRA_VIDEOCODEC), + intent.getBooleanExtra(EXTRA_HWCODEC_ENABLED, true), + intent.getBooleanExtra(EXTRA_FLEXFEC_ENABLED, false), + intent.getIntExtra(EXTRA_AUDIO_BITRATE, 0), intent.getStringExtra(EXTRA_AUDIOCODEC), + intent.getBooleanExtra(EXTRA_NOAUDIOPROCESSING_ENABLED, false), + intent.getBooleanExtra(EXTRA_AECDUMP_ENABLED, false), + intent.getBooleanExtra(EXTRA_SAVE_INPUT_AUDIO_TO_FILE_ENABLED, false), + intent.getBooleanExtra(EXTRA_OPENSLES_ENABLED, false), + intent.getBooleanExtra(EXTRA_DISABLE_BUILT_IN_AEC, false), + intent.getBooleanExtra(EXTRA_DISABLE_BUILT_IN_AGC, false), + intent.getBooleanExtra(EXTRA_DISABLE_BUILT_IN_NS, false), + intent.getBooleanExtra(EXTRA_DISABLE_WEBRTC_AGC_AND_HPF, false), + intent.getBooleanExtra(EXTRA_ENABLE_RTCEVENTLOG, false), dataChannelParameters); + commandLineRun = intent.getBooleanExtra(EXTRA_CMDLINE, false); + int runTimeMs = intent.getIntExtra(EXTRA_RUNTIME, 0); + + Log.d(TAG, "VIDEO_FILE: '" + intent.getStringExtra(EXTRA_VIDEO_FILE_AS_CAMERA) + "'"); + + // Create connection client. Use DirectRTCClient if room name is an IP otherwise use the + // standard WebSocketRTCClient. + if (loopback || !DirectRTCClient.IP_PATTERN.matcher(roomId).matches()) { + appRtcClient = new WebSocketRTCClient(this); + } else { + Log.i(TAG, "Using DirectRTCClient because room name looks like an IP."); + appRtcClient = new DirectRTCClient(this); + } + // Create connection parameters. + String urlParameters = intent.getStringExtra(EXTRA_URLPARAMETERS); + roomConnectionParameters = + new RoomConnectionParameters(roomUri.toString(), roomId, loopback, urlParameters); + + // Create CPU monitor + if (CpuMonitor.isSupported()) { + cpuMonitor = new CpuMonitor(this); + hudFragment.setCpuMonitor(cpuMonitor); + } + + // Send intent arguments to fragments. + callFragment.setArguments(intent.getExtras()); + hudFragment.setArguments(intent.getExtras()); + // Activate call and HUD fragments and start the call. + FragmentTransaction ft = getFragmentManager().beginTransaction(); + ft.add(R.id.call_fragment_container, callFragment); + ft.add(R.id.hud_fragment_container, hudFragment); + ft.commit(); + + // For command line execution run connection for <runTimeMs> and exit. + if (commandLineRun && runTimeMs > 0) { + (new Handler()).postDelayed(new Runnable() { + @Override + public void run() { + disconnect(); + } + }, runTimeMs); + } + + // Create peer connection client. + peerConnectionClient = new PeerConnectionClient( + getApplicationContext(), eglBase, peerConnectionParameters, CallActivity.this); + PeerConnectionFactory.Options options = new PeerConnectionFactory.Options(); + if (loopback) { + options.networkIgnoreMask = 0; + } + peerConnectionClient.createPeerConnectionFactory(options); + + if (screencaptureEnabled) { + startScreenCapture(); + } else { + startCall(); + } + } + + private DisplayMetrics getDisplayMetrics() { + DisplayMetrics displayMetrics = new DisplayMetrics(); + WindowManager windowManager = + (WindowManager) getApplication().getSystemService(Context.WINDOW_SERVICE); + windowManager.getDefaultDisplay().getRealMetrics(displayMetrics); + return displayMetrics; + } + + private static int getSystemUiVisibility() { + return View.SYSTEM_UI_FLAG_HIDE_NAVIGATION | View.SYSTEM_UI_FLAG_FULLSCREEN + | View.SYSTEM_UI_FLAG_IMMERSIVE_STICKY; + } + + private void startScreenCapture() { + MediaProjectionManager mediaProjectionManager = + (MediaProjectionManager) getApplication().getSystemService( + Context.MEDIA_PROJECTION_SERVICE); + startActivityForResult( + mediaProjectionManager.createScreenCaptureIntent(), CAPTURE_PERMISSION_REQUEST_CODE); + } + + @Override + public void onActivityResult(int requestCode, int resultCode, Intent data) { + if (requestCode != CAPTURE_PERMISSION_REQUEST_CODE) + return; + mediaProjectionPermissionResultCode = resultCode; + mediaProjectionPermissionResultData = data; + startCall(); + } + + private boolean useCamera2() { + return Camera2Enumerator.isSupported(this) && getIntent().getBooleanExtra(EXTRA_CAMERA2, true); + } + + private boolean captureToTexture() { + return getIntent().getBooleanExtra(EXTRA_CAPTURETOTEXTURE_ENABLED, false); + } + + private @Nullable VideoCapturer createCameraCapturer(CameraEnumerator enumerator) { + final String[] deviceNames = enumerator.getDeviceNames(); + + // First, try to find front facing camera + Logging.d(TAG, "Looking for front facing cameras."); + for (String deviceName : deviceNames) { + if (enumerator.isFrontFacing(deviceName)) { + Logging.d(TAG, "Creating front facing camera capturer."); + VideoCapturer videoCapturer = enumerator.createCapturer(deviceName, null); + + if (videoCapturer != null) { + return videoCapturer; + } + } + } + + // Front facing camera not found, try something else + Logging.d(TAG, "Looking for other cameras."); + for (String deviceName : deviceNames) { + if (!enumerator.isFrontFacing(deviceName)) { + Logging.d(TAG, "Creating other camera capturer."); + VideoCapturer videoCapturer = enumerator.createCapturer(deviceName, null); + + if (videoCapturer != null) { + return videoCapturer; + } + } + } + + return null; + } + + private @Nullable VideoCapturer createScreenCapturer() { + if (mediaProjectionPermissionResultCode != Activity.RESULT_OK) { + reportError("User didn't give permission to capture the screen."); + return null; + } + return new ScreenCapturerAndroid( + mediaProjectionPermissionResultData, new MediaProjection.Callback() { + @Override + public void onStop() { + reportError("User revoked permission to capture the screen."); + } + }); + } + + // Activity interfaces + @Override + public void onStop() { + super.onStop(); + activityRunning = false; + // Don't stop the video when using screencapture to allow user to show other apps to the remote + // end. + if (peerConnectionClient != null && !screencaptureEnabled) { + peerConnectionClient.stopVideoSource(); + } + if (cpuMonitor != null) { + cpuMonitor.pause(); + } + } + + @Override + public void onStart() { + super.onStart(); + activityRunning = true; + // Video is not paused for screencapture. See onPause. + if (peerConnectionClient != null && !screencaptureEnabled) { + peerConnectionClient.startVideoSource(); + } + if (cpuMonitor != null) { + cpuMonitor.resume(); + } + } + + @Override + protected void onDestroy() { + Thread.setDefaultUncaughtExceptionHandler(null); + disconnect(); + if (logToast != null) { + logToast.cancel(); + } + activityRunning = false; + super.onDestroy(); + } + + // CallFragment.OnCallEvents interface implementation. + @Override + public void onCallHangUp() { + disconnect(); + } + + @Override + public void onCameraSwitch() { + if (peerConnectionClient != null) { + peerConnectionClient.switchCamera(); + } + } + + @Override + public void onVideoScalingSwitch(ScalingType scalingType) { + fullscreenRenderer.setScalingType(scalingType); + } + + @Override + public void onCaptureFormatChange(int width, int height, int framerate) { + if (peerConnectionClient != null) { + peerConnectionClient.changeCaptureFormat(width, height, framerate); + } + } + + @Override + public boolean onToggleMic() { + if (peerConnectionClient != null) { + micEnabled = !micEnabled; + peerConnectionClient.setAudioEnabled(micEnabled); + } + return micEnabled; + } + + // Helper functions. + private void toggleCallControlFragmentVisibility() { + if (!connected || !callFragment.isAdded()) { + return; + } + // Show/hide call control fragment + callControlFragmentVisible = !callControlFragmentVisible; + FragmentTransaction ft = getFragmentManager().beginTransaction(); + if (callControlFragmentVisible) { + ft.show(callFragment); + ft.show(hudFragment); + } else { + ft.hide(callFragment); + ft.hide(hudFragment); + } + ft.setTransition(FragmentTransaction.TRANSIT_FRAGMENT_FADE); + ft.commit(); + } + + private void startCall() { + if (appRtcClient == null) { + Log.e(TAG, "AppRTC client is not allocated for a call."); + return; + } + callStartedTimeMs = System.currentTimeMillis(); + + // Start room connection. + logAndToast(getString(R.string.connecting_to, roomConnectionParameters.roomUrl)); + appRtcClient.connectToRoom(roomConnectionParameters); + + // Create and audio manager that will take care of audio routing, + // audio modes, audio device enumeration etc. + audioManager = AppRTCAudioManager.create(getApplicationContext()); + // Store existing audio settings and change audio mode to + // MODE_IN_COMMUNICATION for best possible VoIP performance. + Log.d(TAG, "Starting the audio manager..."); + audioManager.start(new AudioManagerEvents() { + // This method will be called each time the number of available audio + // devices has changed. + @Override + public void onAudioDeviceChanged( + AudioDevice audioDevice, Set<AudioDevice> availableAudioDevices) { + onAudioManagerDevicesChanged(audioDevice, availableAudioDevices); + } + }); + } + + // Should be called from UI thread + private void callConnected() { + final long delta = System.currentTimeMillis() - callStartedTimeMs; + Log.i(TAG, "Call connected: delay=" + delta + "ms"); + if (peerConnectionClient == null || isError) { + Log.w(TAG, "Call is connected in closed or error state"); + return; + } + // Enable statistics callback. + peerConnectionClient.enableStatsEvents(true, STAT_CALLBACK_PERIOD); + setSwappedFeeds(false /* isSwappedFeeds */); + } + + // This method is called when the audio manager reports audio device change, + // e.g. from wired headset to speakerphone. + private void onAudioManagerDevicesChanged( + final AudioDevice device, final Set<AudioDevice> availableDevices) { + Log.d(TAG, "onAudioManagerDevicesChanged: " + availableDevices + ", " + + "selected: " + device); + // TODO(henrika): add callback handler. + } + + // Disconnect from remote resources, dispose of local resources, and exit. + private void disconnect() { + activityRunning = false; + remoteProxyRenderer.setTarget(null); + localProxyVideoSink.setTarget(null); + if (appRtcClient != null) { + appRtcClient.disconnectFromRoom(); + appRtcClient = null; + } + if (pipRenderer != null) { + pipRenderer.release(); + pipRenderer = null; + } + if (videoFileRenderer != null) { + videoFileRenderer.release(); + videoFileRenderer = null; + } + if (fullscreenRenderer != null) { + fullscreenRenderer.release(); + fullscreenRenderer = null; + } + if (peerConnectionClient != null) { + peerConnectionClient.close(); + peerConnectionClient = null; + } + if (audioManager != null) { + audioManager.stop(); + audioManager = null; + } + if (connected && !isError) { + setResult(RESULT_OK); + } else { + setResult(RESULT_CANCELED); + } + finish(); + } + + private void disconnectWithErrorMessage(final String errorMessage) { + if (commandLineRun || !activityRunning) { + Log.e(TAG, "Critical error: " + errorMessage); + disconnect(); + } else { + new AlertDialog.Builder(this) + .setTitle(getText(R.string.channel_error_title)) + .setMessage(errorMessage) + .setCancelable(false) + .setNeutralButton(R.string.ok, + new DialogInterface.OnClickListener() { + @Override + public void onClick(DialogInterface dialog, int id) { + dialog.cancel(); + disconnect(); + } + }) + .create() + .show(); + } + } + + // Log `msg` and Toast about it. + private void logAndToast(String msg) { + Log.d(TAG, msg); + if (logToast != null) { + logToast.cancel(); + } + logToast = Toast.makeText(this, msg, Toast.LENGTH_SHORT); + logToast.show(); + } + + private void reportError(final String description) { + runOnUiThread(new Runnable() { + @Override + public void run() { + if (!isError) { + isError = true; + disconnectWithErrorMessage(description); + } + } + }); + } + + private @Nullable VideoCapturer createVideoCapturer() { + final VideoCapturer videoCapturer; + String videoFileAsCamera = getIntent().getStringExtra(EXTRA_VIDEO_FILE_AS_CAMERA); + if (videoFileAsCamera != null) { + try { + videoCapturer = new FileVideoCapturer(videoFileAsCamera); + } catch (IOException e) { + reportError("Failed to open video file for emulated camera"); + return null; + } + } else if (screencaptureEnabled) { + return createScreenCapturer(); + } else if (useCamera2()) { + if (!captureToTexture()) { + reportError(getString(R.string.camera2_texture_only_error)); + return null; + } + + Logging.d(TAG, "Creating capturer using camera2 API."); + videoCapturer = createCameraCapturer(new Camera2Enumerator(this)); + } else { + Logging.d(TAG, "Creating capturer using camera1 API."); + videoCapturer = createCameraCapturer(new Camera1Enumerator(captureToTexture())); + } + if (videoCapturer == null) { + reportError("Failed to open camera"); + return null; + } + return videoCapturer; + } + + private void setSwappedFeeds(boolean isSwappedFeeds) { + Logging.d(TAG, "setSwappedFeeds: " + isSwappedFeeds); + this.isSwappedFeeds = isSwappedFeeds; + localProxyVideoSink.setTarget(isSwappedFeeds ? fullscreenRenderer : pipRenderer); + remoteProxyRenderer.setTarget(isSwappedFeeds ? pipRenderer : fullscreenRenderer); + fullscreenRenderer.setMirror(isSwappedFeeds); + pipRenderer.setMirror(!isSwappedFeeds); + } + + // -----Implementation of AppRTCClient.AppRTCSignalingEvents --------------- + // All callbacks are invoked from websocket signaling looper thread and + // are routed to UI thread. + private void onConnectedToRoomInternal(final SignalingParameters params) { + final long delta = System.currentTimeMillis() - callStartedTimeMs; + + signalingParameters = params; + logAndToast("Creating peer connection, delay=" + delta + "ms"); + VideoCapturer videoCapturer = null; + if (peerConnectionParameters.videoCallEnabled) { + videoCapturer = createVideoCapturer(); + } + peerConnectionClient.createPeerConnection( + localProxyVideoSink, remoteSinks, videoCapturer, signalingParameters); + + if (signalingParameters.initiator) { + logAndToast("Creating OFFER..."); + // Create offer. Offer SDP will be sent to answering client in + // PeerConnectionEvents.onLocalDescription event. + peerConnectionClient.createOffer(); + } else { + if (params.offerSdp != null) { + peerConnectionClient.setRemoteDescription(params.offerSdp); + logAndToast("Creating ANSWER..."); + // Create answer. Answer SDP will be sent to offering client in + // PeerConnectionEvents.onLocalDescription event. + peerConnectionClient.createAnswer(); + } + if (params.iceCandidates != null) { + // Add remote ICE candidates from room. + for (IceCandidate iceCandidate : params.iceCandidates) { + peerConnectionClient.addRemoteIceCandidate(iceCandidate); + } + } + } + } + + @Override + public void onConnectedToRoom(final SignalingParameters params) { + runOnUiThread(new Runnable() { + @Override + public void run() { + onConnectedToRoomInternal(params); + } + }); + } + + @Override + public void onRemoteDescription(final SessionDescription desc) { + final long delta = System.currentTimeMillis() - callStartedTimeMs; + runOnUiThread(new Runnable() { + @Override + public void run() { + if (peerConnectionClient == null) { + Log.e(TAG, "Received remote SDP for non-initilized peer connection."); + return; + } + logAndToast("Received remote " + desc.type + ", delay=" + delta + "ms"); + peerConnectionClient.setRemoteDescription(desc); + if (!signalingParameters.initiator) { + logAndToast("Creating ANSWER..."); + // Create answer. Answer SDP will be sent to offering client in + // PeerConnectionEvents.onLocalDescription event. + peerConnectionClient.createAnswer(); + } + } + }); + } + + @Override + public void onRemoteIceCandidate(final IceCandidate candidate) { + runOnUiThread(new Runnable() { + @Override + public void run() { + if (peerConnectionClient == null) { + Log.e(TAG, "Received ICE candidate for a non-initialized peer connection."); + return; + } + peerConnectionClient.addRemoteIceCandidate(candidate); + } + }); + } + + @Override + public void onRemoteIceCandidatesRemoved(final IceCandidate[] candidates) { + runOnUiThread(new Runnable() { + @Override + public void run() { + if (peerConnectionClient == null) { + Log.e(TAG, "Received ICE candidate removals for a non-initialized peer connection."); + return; + } + peerConnectionClient.removeRemoteIceCandidates(candidates); + } + }); + } + + @Override + public void onChannelClose() { + runOnUiThread(new Runnable() { + @Override + public void run() { + logAndToast("Remote end hung up; dropping PeerConnection"); + disconnect(); + } + }); + } + + @Override + public void onChannelError(final String description) { + reportError(description); + } + + // -----Implementation of PeerConnectionClient.PeerConnectionEvents.--------- + // Send local peer connection SDP and ICE candidates to remote party. + // All callbacks are invoked from peer connection client looper thread and + // are routed to UI thread. + @Override + public void onLocalDescription(final SessionDescription desc) { + final long delta = System.currentTimeMillis() - callStartedTimeMs; + runOnUiThread(new Runnable() { + @Override + public void run() { + if (appRtcClient != null) { + logAndToast("Sending " + desc.type + ", delay=" + delta + "ms"); + if (signalingParameters.initiator) { + appRtcClient.sendOfferSdp(desc); + } else { + appRtcClient.sendAnswerSdp(desc); + } + } + if (peerConnectionParameters.videoMaxBitrate > 0) { + Log.d(TAG, "Set video maximum bitrate: " + peerConnectionParameters.videoMaxBitrate); + peerConnectionClient.setVideoMaxBitrate(peerConnectionParameters.videoMaxBitrate); + } + } + }); + } + + @Override + public void onIceCandidate(final IceCandidate candidate) { + runOnUiThread(new Runnable() { + @Override + public void run() { + if (appRtcClient != null) { + appRtcClient.sendLocalIceCandidate(candidate); + } + } + }); + } + + @Override + public void onIceCandidatesRemoved(final IceCandidate[] candidates) { + runOnUiThread(new Runnable() { + @Override + public void run() { + if (appRtcClient != null) { + appRtcClient.sendLocalIceCandidateRemovals(candidates); + } + } + }); + } + + @Override + public void onIceConnected() { + final long delta = System.currentTimeMillis() - callStartedTimeMs; + runOnUiThread(new Runnable() { + @Override + public void run() { + logAndToast("ICE connected, delay=" + delta + "ms"); + } + }); + } + + @Override + public void onIceDisconnected() { + runOnUiThread(new Runnable() { + @Override + public void run() { + logAndToast("ICE disconnected"); + } + }); + } + + @Override + public void onConnected() { + final long delta = System.currentTimeMillis() - callStartedTimeMs; + runOnUiThread(new Runnable() { + @Override + public void run() { + logAndToast("DTLS connected, delay=" + delta + "ms"); + connected = true; + callConnected(); + } + }); + } + + @Override + public void onDisconnected() { + runOnUiThread(new Runnable() { + @Override + public void run() { + logAndToast("DTLS disconnected"); + connected = false; + disconnect(); + } + }); + } + + @Override + public void onPeerConnectionClosed() {} + + @Override + public void onPeerConnectionStatsReady(final RTCStatsReport report) { + runOnUiThread(new Runnable() { + @Override + public void run() { + if (!isError && connected) { + hudFragment.updateEncoderStatistics(report); + } + } + }); + } + + @Override + public void onPeerConnectionError(final String description) { + reportError(description); + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CallFragment.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CallFragment.java new file mode 100644 index 0000000000..0d8bdaa06f --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CallFragment.java @@ -0,0 +1,137 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.app.Activity; +import android.app.Fragment; +import android.os.Bundle; +import android.view.LayoutInflater; +import android.view.View; +import android.view.ViewGroup; +import android.widget.ImageButton; +import android.widget.SeekBar; +import android.widget.TextView; + +import org.webrtc.RendererCommon.ScalingType; + +/** + * Fragment for call control. + */ +public class CallFragment extends Fragment { + private TextView contactView; + private ImageButton cameraSwitchButton; + private ImageButton videoScalingButton; + private ImageButton toggleMuteButton; + private TextView captureFormatText; + private SeekBar captureFormatSlider; + private OnCallEvents callEvents; + private ScalingType scalingType; + private boolean videoCallEnabled = true; + + /** + * Call control interface for container activity. + */ + public interface OnCallEvents { + void onCallHangUp(); + void onCameraSwitch(); + void onVideoScalingSwitch(ScalingType scalingType); + void onCaptureFormatChange(int width, int height, int framerate); + boolean onToggleMic(); + } + + @Override + public View onCreateView( + LayoutInflater inflater, ViewGroup container, Bundle savedInstanceState) { + View controlView = inflater.inflate(R.layout.fragment_call, container, false); + + // Create UI controls. + contactView = controlView.findViewById(R.id.contact_name_call); + ImageButton disconnectButton = controlView.findViewById(R.id.button_call_disconnect); + cameraSwitchButton = controlView.findViewById(R.id.button_call_switch_camera); + videoScalingButton = controlView.findViewById(R.id.button_call_scaling_mode); + toggleMuteButton = controlView.findViewById(R.id.button_call_toggle_mic); + captureFormatText = controlView.findViewById(R.id.capture_format_text_call); + captureFormatSlider = controlView.findViewById(R.id.capture_format_slider_call); + + // Add buttons click events. + disconnectButton.setOnClickListener(new View.OnClickListener() { + @Override + public void onClick(View view) { + callEvents.onCallHangUp(); + } + }); + + cameraSwitchButton.setOnClickListener(new View.OnClickListener() { + @Override + public void onClick(View view) { + callEvents.onCameraSwitch(); + } + }); + + videoScalingButton.setOnClickListener(new View.OnClickListener() { + @Override + public void onClick(View view) { + if (scalingType == ScalingType.SCALE_ASPECT_FILL) { + videoScalingButton.setBackgroundResource(R.drawable.ic_action_full_screen); + scalingType = ScalingType.SCALE_ASPECT_FIT; + } else { + videoScalingButton.setBackgroundResource(R.drawable.ic_action_return_from_full_screen); + scalingType = ScalingType.SCALE_ASPECT_FILL; + } + callEvents.onVideoScalingSwitch(scalingType); + } + }); + scalingType = ScalingType.SCALE_ASPECT_FILL; + + toggleMuteButton.setOnClickListener(new View.OnClickListener() { + @Override + public void onClick(View view) { + boolean enabled = callEvents.onToggleMic(); + toggleMuteButton.setAlpha(enabled ? 1.0f : 0.3f); + } + }); + + return controlView; + } + + @Override + public void onStart() { + super.onStart(); + + boolean captureSliderEnabled = false; + Bundle args = getArguments(); + if (args != null) { + String contactName = args.getString(CallActivity.EXTRA_ROOMID); + contactView.setText(contactName); + videoCallEnabled = args.getBoolean(CallActivity.EXTRA_VIDEO_CALL, true); + captureSliderEnabled = videoCallEnabled + && args.getBoolean(CallActivity.EXTRA_VIDEO_CAPTUREQUALITYSLIDER_ENABLED, false); + } + if (!videoCallEnabled) { + cameraSwitchButton.setVisibility(View.INVISIBLE); + } + if (captureSliderEnabled) { + captureFormatSlider.setOnSeekBarChangeListener( + new CaptureQualityController(captureFormatText, callEvents)); + } else { + captureFormatText.setVisibility(View.GONE); + captureFormatSlider.setVisibility(View.GONE); + } + } + + // TODO(sakal): Replace with onAttach(Context) once we only support API level 23+. + @SuppressWarnings("deprecation") + @Override + public void onAttach(Activity activity) { + super.onAttach(activity); + callEvents = (OnCallEvents) activity; + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CaptureQualityController.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CaptureQualityController.java new file mode 100644 index 0000000000..8a783eca9c --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CaptureQualityController.java @@ -0,0 +1,110 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.widget.SeekBar; +import android.widget.TextView; +import java.util.Arrays; +import java.util.Collections; +import java.util.Comparator; +import java.util.List; +import org.webrtc.CameraEnumerationAndroid.CaptureFormat; + +/** + * Control capture format based on a seekbar listener. + */ +public class CaptureQualityController implements SeekBar.OnSeekBarChangeListener { + private final List<CaptureFormat> formats = + Arrays.asList(new CaptureFormat(1280, 720, 0, 30000), new CaptureFormat(960, 540, 0, 30000), + new CaptureFormat(640, 480, 0, 30000), new CaptureFormat(480, 360, 0, 30000), + new CaptureFormat(320, 240, 0, 30000), new CaptureFormat(256, 144, 0, 30000)); + // Prioritize framerate below this threshold and resolution above the threshold. + private static final int FRAMERATE_THRESHOLD = 15; + private TextView captureFormatText; + private CallFragment.OnCallEvents callEvents; + private int width; + private int height; + private int framerate; + private double targetBandwidth; + + public CaptureQualityController( + TextView captureFormatText, CallFragment.OnCallEvents callEvents) { + this.captureFormatText = captureFormatText; + this.callEvents = callEvents; + } + + private final Comparator<CaptureFormat> compareFormats = new Comparator<CaptureFormat>() { + @Override + public int compare(CaptureFormat first, CaptureFormat second) { + int firstFps = calculateFramerate(targetBandwidth, first); + int secondFps = calculateFramerate(targetBandwidth, second); + + if ((firstFps >= FRAMERATE_THRESHOLD && secondFps >= FRAMERATE_THRESHOLD) + || firstFps == secondFps) { + // Compare resolution. + return first.width * first.height - second.width * second.height; + } else { + // Compare fps. + return firstFps - secondFps; + } + } + }; + + @Override + public void onProgressChanged(SeekBar seekBar, int progress, boolean fromUser) { + if (progress == 0) { + width = 0; + height = 0; + framerate = 0; + captureFormatText.setText(R.string.muted); + return; + } + + // Extract max bandwidth (in millipixels / second). + long maxCaptureBandwidth = java.lang.Long.MIN_VALUE; + for (CaptureFormat format : formats) { + maxCaptureBandwidth = + Math.max(maxCaptureBandwidth, (long) format.width * format.height * format.framerate.max); + } + + // Fraction between 0 and 1. + double bandwidthFraction = (double) progress / 100.0; + // Make a log-scale transformation, still between 0 and 1. + final double kExpConstant = 3.0; + bandwidthFraction = + (Math.exp(kExpConstant * bandwidthFraction) - 1) / (Math.exp(kExpConstant) - 1); + targetBandwidth = bandwidthFraction * maxCaptureBandwidth; + + // Choose the best format given a target bandwidth. + final CaptureFormat bestFormat = Collections.max(formats, compareFormats); + width = bestFormat.width; + height = bestFormat.height; + framerate = calculateFramerate(targetBandwidth, bestFormat); + captureFormatText.setText( + String.format(captureFormatText.getContext().getString(R.string.format_description), width, + height, framerate)); + } + + @Override + public void onStartTrackingTouch(SeekBar seekBar) {} + + @Override + public void onStopTrackingTouch(SeekBar seekBar) { + callEvents.onCaptureFormatChange(width, height, framerate); + } + + // Return the highest frame rate possible based on bandwidth and format. + private int calculateFramerate(double bandwidth, CaptureFormat format) { + return (int) Math.round( + Math.min(format.framerate.max, (int) Math.round(bandwidth / (format.width * format.height))) + / 1000.0); + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/ConnectActivity.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/ConnectActivity.java new file mode 100644 index 0000000000..7206c88498 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/ConnectActivity.java @@ -0,0 +1,666 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.annotation.TargetApi; +import android.app.Activity; +import android.app.AlertDialog; +import android.content.DialogInterface; +import android.content.Intent; +import android.content.SharedPreferences; +import android.content.pm.PackageInfo; +import android.content.pm.PackageManager; +import android.net.Uri; +import android.os.Build; +import android.os.Bundle; +import android.preference.PreferenceManager; +import android.util.Log; +import android.view.ContextMenu; +import android.view.KeyEvent; +import android.view.Menu; +import android.view.MenuItem; +import android.view.View; +import android.view.View.OnClickListener; +import android.view.inputmethod.EditorInfo; +import android.webkit.URLUtil; +import android.widget.AdapterView; +import android.widget.ArrayAdapter; +import android.widget.EditText; +import android.widget.ImageButton; +import android.widget.ListView; +import android.widget.TextView; +import androidx.annotation.Nullable; +import java.util.ArrayList; +import java.util.Random; +import org.json.JSONArray; +import org.json.JSONException; + +/** + * Handles the initial setup where the user selects which room to join. + */ +public class ConnectActivity extends Activity { + private static final String TAG = "ConnectActivity"; + private static final int CONNECTION_REQUEST = 1; + private static final int PERMISSION_REQUEST = 2; + private static final int REMOVE_FAVORITE_INDEX = 0; + private static boolean commandLineRun; + + private ImageButton addFavoriteButton; + private EditText roomEditText; + private ListView roomListView; + private SharedPreferences sharedPref; + private String keyprefResolution; + private String keyprefFps; + private String keyprefVideoBitrateType; + private String keyprefVideoBitrateValue; + private String keyprefAudioBitrateType; + private String keyprefAudioBitrateValue; + private String keyprefRoomServerUrl; + private String keyprefRoom; + private String keyprefRoomList; + private ArrayList<String> roomList; + private ArrayAdapter<String> adapter; + + @Override + public void onCreate(Bundle savedInstanceState) { + super.onCreate(savedInstanceState); + + // Get setting keys. + PreferenceManager.setDefaultValues(this, R.xml.preferences, false); + sharedPref = PreferenceManager.getDefaultSharedPreferences(this); + keyprefResolution = getString(R.string.pref_resolution_key); + keyprefFps = getString(R.string.pref_fps_key); + keyprefVideoBitrateType = getString(R.string.pref_maxvideobitrate_key); + keyprefVideoBitrateValue = getString(R.string.pref_maxvideobitratevalue_key); + keyprefAudioBitrateType = getString(R.string.pref_startaudiobitrate_key); + keyprefAudioBitrateValue = getString(R.string.pref_startaudiobitratevalue_key); + keyprefRoomServerUrl = getString(R.string.pref_room_server_url_key); + keyprefRoom = getString(R.string.pref_room_key); + keyprefRoomList = getString(R.string.pref_room_list_key); + + setContentView(R.layout.activity_connect); + + roomEditText = findViewById(R.id.room_edittext); + roomEditText.setOnEditorActionListener(new TextView.OnEditorActionListener() { + @Override + public boolean onEditorAction(TextView textView, int i, KeyEvent keyEvent) { + if (i == EditorInfo.IME_ACTION_DONE) { + addFavoriteButton.performClick(); + return true; + } + return false; + } + }); + roomEditText.requestFocus(); + + roomListView = findViewById(R.id.room_listview); + roomListView.setEmptyView(findViewById(android.R.id.empty)); + roomListView.setOnItemClickListener(roomListClickListener); + registerForContextMenu(roomListView); + ImageButton connectButton = findViewById(R.id.connect_button); + connectButton.setOnClickListener(connectListener); + addFavoriteButton = findViewById(R.id.add_favorite_button); + addFavoriteButton.setOnClickListener(addFavoriteListener); + + requestPermissions(); + } + + @Override + public boolean onCreateOptionsMenu(Menu menu) { + getMenuInflater().inflate(R.menu.connect_menu, menu); + return true; + } + + @Override + public void onCreateContextMenu(ContextMenu menu, View v, ContextMenu.ContextMenuInfo menuInfo) { + if (v.getId() == R.id.room_listview) { + AdapterView.AdapterContextMenuInfo info = (AdapterView.AdapterContextMenuInfo) menuInfo; + menu.setHeaderTitle(roomList.get(info.position)); + String[] menuItems = getResources().getStringArray(R.array.roomListContextMenu); + for (int i = 0; i < menuItems.length; i++) { + menu.add(Menu.NONE, i, i, menuItems[i]); + } + } else { + super.onCreateContextMenu(menu, v, menuInfo); + } + } + + @Override + public boolean onContextItemSelected(MenuItem item) { + if (item.getItemId() == REMOVE_FAVORITE_INDEX) { + AdapterView.AdapterContextMenuInfo info = + (AdapterView.AdapterContextMenuInfo) item.getMenuInfo(); + roomList.remove(info.position); + adapter.notifyDataSetChanged(); + return true; + } + + return super.onContextItemSelected(item); + } + + @Override + public boolean onOptionsItemSelected(MenuItem item) { + // Handle presses on the action bar items. + if (item.getItemId() == R.id.action_settings) { + Intent intent = new Intent(this, SettingsActivity.class); + startActivity(intent); + return true; + } else if (item.getItemId() == R.id.action_loopback) { + connectToRoom(null, false, true, false, 0); + return true; + } else { + return super.onOptionsItemSelected(item); + } + } + + @Override + public void onPause() { + super.onPause(); + String room = roomEditText.getText().toString(); + String roomListJson = new JSONArray(roomList).toString(); + SharedPreferences.Editor editor = sharedPref.edit(); + editor.putString(keyprefRoom, room); + editor.putString(keyprefRoomList, roomListJson); + editor.commit(); + } + + @Override + public void onResume() { + super.onResume(); + String room = sharedPref.getString(keyprefRoom, ""); + roomEditText.setText(room); + roomList = new ArrayList<>(); + String roomListJson = sharedPref.getString(keyprefRoomList, null); + if (roomListJson != null) { + try { + JSONArray jsonArray = new JSONArray(roomListJson); + for (int i = 0; i < jsonArray.length(); i++) { + roomList.add(jsonArray.get(i).toString()); + } + } catch (JSONException e) { + Log.e(TAG, "Failed to load room list: " + e.toString()); + } + } + adapter = new ArrayAdapter<>(this, android.R.layout.simple_list_item_1, roomList); + roomListView.setAdapter(adapter); + if (adapter.getCount() > 0) { + roomListView.requestFocus(); + roomListView.setItemChecked(0, true); + } + } + + @Override + protected void onActivityResult(int requestCode, int resultCode, Intent data) { + if (requestCode == CONNECTION_REQUEST && commandLineRun) { + Log.d(TAG, "Return: " + resultCode); + setResult(resultCode); + commandLineRun = false; + finish(); + } + } + + @Override + public void onRequestPermissionsResult( + int requestCode, String[] permissions, int[] grantResults) { + if (requestCode == PERMISSION_REQUEST) { + String[] missingPermissions = getMissingPermissions(); + if (missingPermissions.length != 0) { + // User didn't grant all the permissions. Warn that the application might not work + // correctly. + new AlertDialog.Builder(this) + .setMessage(R.string.missing_permissions_try_again) + .setPositiveButton(R.string.yes, + (dialog, id) -> { + // User wants to try giving the permissions again. + dialog.cancel(); + requestPermissions(); + }) + .setNegativeButton(R.string.no, + (dialog, id) -> { + // User doesn't want to give the permissions. + dialog.cancel(); + onPermissionsGranted(); + }) + .show(); + } else { + // All permissions granted. + onPermissionsGranted(); + } + } + } + + private void onPermissionsGranted() { + // If an implicit VIEW intent is launching the app, go directly to that URL. + final Intent intent = getIntent(); + if ("android.intent.action.VIEW".equals(intent.getAction()) && !commandLineRun) { + boolean loopback = intent.getBooleanExtra(CallActivity.EXTRA_LOOPBACK, false); + int runTimeMs = intent.getIntExtra(CallActivity.EXTRA_RUNTIME, 0); + boolean useValuesFromIntent = + intent.getBooleanExtra(CallActivity.EXTRA_USE_VALUES_FROM_INTENT, false); + String room = sharedPref.getString(keyprefRoom, ""); + connectToRoom(room, true, loopback, useValuesFromIntent, runTimeMs); + } + } + + @TargetApi(Build.VERSION_CODES.M) + private void requestPermissions() { + if (Build.VERSION.SDK_INT < Build.VERSION_CODES.M) { + // Dynamic permissions are not required before Android M. + onPermissionsGranted(); + return; + } + + String[] missingPermissions = getMissingPermissions(); + if (missingPermissions.length != 0) { + requestPermissions(missingPermissions, PERMISSION_REQUEST); + } else { + onPermissionsGranted(); + } + } + + @TargetApi(Build.VERSION_CODES.M) + private String[] getMissingPermissions() { + if (Build.VERSION.SDK_INT < Build.VERSION_CODES.M) { + return new String[0]; + } + + PackageInfo info; + try { + info = getPackageManager().getPackageInfo(getPackageName(), PackageManager.GET_PERMISSIONS); + } catch (PackageManager.NameNotFoundException e) { + Log.w(TAG, "Failed to retrieve permissions."); + return new String[0]; + } + + if (info.requestedPermissions == null) { + Log.w(TAG, "No requested permissions."); + return new String[0]; + } + + ArrayList<String> missingPermissions = new ArrayList<>(); + for (int i = 0; i < info.requestedPermissions.length; i++) { + if ((info.requestedPermissionsFlags[i] & PackageInfo.REQUESTED_PERMISSION_GRANTED) == 0) { + missingPermissions.add(info.requestedPermissions[i]); + } + } + Log.d(TAG, "Missing permissions: " + missingPermissions); + + return missingPermissions.toArray(new String[missingPermissions.size()]); + } + + /** + * Get a value from the shared preference or from the intent, if it does not + * exist the default is used. + */ + @Nullable + private String sharedPrefGetString( + int attributeId, String intentName, int defaultId, boolean useFromIntent) { + String defaultValue = getString(defaultId); + if (useFromIntent) { + String value = getIntent().getStringExtra(intentName); + if (value != null) { + return value; + } + return defaultValue; + } else { + String attributeName = getString(attributeId); + return sharedPref.getString(attributeName, defaultValue); + } + } + + /** + * Get a value from the shared preference or from the intent, if it does not + * exist the default is used. + */ + private boolean sharedPrefGetBoolean( + int attributeId, String intentName, int defaultId, boolean useFromIntent) { + boolean defaultValue = Boolean.parseBoolean(getString(defaultId)); + if (useFromIntent) { + return getIntent().getBooleanExtra(intentName, defaultValue); + } else { + String attributeName = getString(attributeId); + return sharedPref.getBoolean(attributeName, defaultValue); + } + } + + /** + * Get a value from the shared preference or from the intent, if it does not + * exist the default is used. + */ + private int sharedPrefGetInteger( + int attributeId, String intentName, int defaultId, boolean useFromIntent) { + String defaultString = getString(defaultId); + int defaultValue = Integer.parseInt(defaultString); + if (useFromIntent) { + return getIntent().getIntExtra(intentName, defaultValue); + } else { + String attributeName = getString(attributeId); + String value = sharedPref.getString(attributeName, defaultString); + try { + return Integer.parseInt(value); + } catch (NumberFormatException e) { + Log.e(TAG, "Wrong setting for: " + attributeName + ":" + value); + return defaultValue; + } + } + } + + @SuppressWarnings("StringSplitter") + private void connectToRoom(String roomId, boolean commandLineRun, boolean loopback, + boolean useValuesFromIntent, int runTimeMs) { + ConnectActivity.commandLineRun = commandLineRun; + + // roomId is random for loopback. + if (loopback) { + roomId = Integer.toString((new Random()).nextInt(100000000)); + } + + String roomUrl = sharedPref.getString( + keyprefRoomServerUrl, getString(R.string.pref_room_server_url_default)); + + // Video call enabled flag. + boolean videoCallEnabled = sharedPrefGetBoolean(R.string.pref_videocall_key, + CallActivity.EXTRA_VIDEO_CALL, R.string.pref_videocall_default, useValuesFromIntent); + + // Use screencapture option. + boolean useScreencapture = sharedPrefGetBoolean(R.string.pref_screencapture_key, + CallActivity.EXTRA_SCREENCAPTURE, R.string.pref_screencapture_default, useValuesFromIntent); + + // Use Camera2 option. + boolean useCamera2 = sharedPrefGetBoolean(R.string.pref_camera2_key, CallActivity.EXTRA_CAMERA2, + R.string.pref_camera2_default, useValuesFromIntent); + + // Get default codecs. + String videoCodec = sharedPrefGetString(R.string.pref_videocodec_key, + CallActivity.EXTRA_VIDEOCODEC, R.string.pref_videocodec_default, useValuesFromIntent); + String audioCodec = sharedPrefGetString(R.string.pref_audiocodec_key, + CallActivity.EXTRA_AUDIOCODEC, R.string.pref_audiocodec_default, useValuesFromIntent); + + // Check HW codec flag. + boolean hwCodec = sharedPrefGetBoolean(R.string.pref_hwcodec_key, + CallActivity.EXTRA_HWCODEC_ENABLED, R.string.pref_hwcodec_default, useValuesFromIntent); + + // Check Capture to texture. + boolean captureToTexture = sharedPrefGetBoolean(R.string.pref_capturetotexture_key, + CallActivity.EXTRA_CAPTURETOTEXTURE_ENABLED, R.string.pref_capturetotexture_default, + useValuesFromIntent); + + // Check FlexFEC. + boolean flexfecEnabled = sharedPrefGetBoolean(R.string.pref_flexfec_key, + CallActivity.EXTRA_FLEXFEC_ENABLED, R.string.pref_flexfec_default, useValuesFromIntent); + + // Check Disable Audio Processing flag. + boolean noAudioProcessing = sharedPrefGetBoolean(R.string.pref_noaudioprocessing_key, + CallActivity.EXTRA_NOAUDIOPROCESSING_ENABLED, R.string.pref_noaudioprocessing_default, + useValuesFromIntent); + + boolean aecDump = sharedPrefGetBoolean(R.string.pref_aecdump_key, + CallActivity.EXTRA_AECDUMP_ENABLED, R.string.pref_aecdump_default, useValuesFromIntent); + + boolean saveInputAudioToFile = + sharedPrefGetBoolean(R.string.pref_enable_save_input_audio_to_file_key, + CallActivity.EXTRA_SAVE_INPUT_AUDIO_TO_FILE_ENABLED, + R.string.pref_enable_save_input_audio_to_file_default, useValuesFromIntent); + + // Check OpenSL ES enabled flag. + boolean useOpenSLES = sharedPrefGetBoolean(R.string.pref_opensles_key, + CallActivity.EXTRA_OPENSLES_ENABLED, R.string.pref_opensles_default, useValuesFromIntent); + + // Check Disable built-in AEC flag. + boolean disableBuiltInAEC = sharedPrefGetBoolean(R.string.pref_disable_built_in_aec_key, + CallActivity.EXTRA_DISABLE_BUILT_IN_AEC, R.string.pref_disable_built_in_aec_default, + useValuesFromIntent); + + // Check Disable built-in AGC flag. + boolean disableBuiltInAGC = sharedPrefGetBoolean(R.string.pref_disable_built_in_agc_key, + CallActivity.EXTRA_DISABLE_BUILT_IN_AGC, R.string.pref_disable_built_in_agc_default, + useValuesFromIntent); + + // Check Disable built-in NS flag. + boolean disableBuiltInNS = sharedPrefGetBoolean(R.string.pref_disable_built_in_ns_key, + CallActivity.EXTRA_DISABLE_BUILT_IN_NS, R.string.pref_disable_built_in_ns_default, + useValuesFromIntent); + + // Check Disable gain control + boolean disableWebRtcAGCAndHPF = sharedPrefGetBoolean( + R.string.pref_disable_webrtc_agc_and_hpf_key, CallActivity.EXTRA_DISABLE_WEBRTC_AGC_AND_HPF, + R.string.pref_disable_webrtc_agc_and_hpf_key, useValuesFromIntent); + + // Get video resolution from settings. + int videoWidth = 0; + int videoHeight = 0; + if (useValuesFromIntent) { + videoWidth = getIntent().getIntExtra(CallActivity.EXTRA_VIDEO_WIDTH, 0); + videoHeight = getIntent().getIntExtra(CallActivity.EXTRA_VIDEO_HEIGHT, 0); + } + if (videoWidth == 0 && videoHeight == 0) { + String resolution = + sharedPref.getString(keyprefResolution, getString(R.string.pref_resolution_default)); + String[] dimensions = resolution.split("[ x]+"); + if (dimensions.length == 2) { + try { + videoWidth = Integer.parseInt(dimensions[0]); + videoHeight = Integer.parseInt(dimensions[1]); + } catch (NumberFormatException e) { + videoWidth = 0; + videoHeight = 0; + Log.e(TAG, "Wrong video resolution setting: " + resolution); + } + } + } + + // Get camera fps from settings. + int cameraFps = 0; + if (useValuesFromIntent) { + cameraFps = getIntent().getIntExtra(CallActivity.EXTRA_VIDEO_FPS, 0); + } + if (cameraFps == 0) { + String fps = sharedPref.getString(keyprefFps, getString(R.string.pref_fps_default)); + String[] fpsValues = fps.split("[ x]+"); + if (fpsValues.length == 2) { + try { + cameraFps = Integer.parseInt(fpsValues[0]); + } catch (NumberFormatException e) { + cameraFps = 0; + Log.e(TAG, "Wrong camera fps setting: " + fps); + } + } + } + + // Check capture quality slider flag. + boolean captureQualitySlider = sharedPrefGetBoolean(R.string.pref_capturequalityslider_key, + CallActivity.EXTRA_VIDEO_CAPTUREQUALITYSLIDER_ENABLED, + R.string.pref_capturequalityslider_default, useValuesFromIntent); + + // Get video and audio start bitrate. + int videoStartBitrate = 0; + if (useValuesFromIntent) { + videoStartBitrate = getIntent().getIntExtra(CallActivity.EXTRA_VIDEO_BITRATE, 0); + } + if (videoStartBitrate == 0) { + String bitrateTypeDefault = getString(R.string.pref_maxvideobitrate_default); + String bitrateType = sharedPref.getString(keyprefVideoBitrateType, bitrateTypeDefault); + if (!bitrateType.equals(bitrateTypeDefault)) { + String bitrateValue = sharedPref.getString( + keyprefVideoBitrateValue, getString(R.string.pref_maxvideobitratevalue_default)); + videoStartBitrate = Integer.parseInt(bitrateValue); + } + } + + int audioStartBitrate = 0; + if (useValuesFromIntent) { + audioStartBitrate = getIntent().getIntExtra(CallActivity.EXTRA_AUDIO_BITRATE, 0); + } + if (audioStartBitrate == 0) { + String bitrateTypeDefault = getString(R.string.pref_startaudiobitrate_default); + String bitrateType = sharedPref.getString(keyprefAudioBitrateType, bitrateTypeDefault); + if (!bitrateType.equals(bitrateTypeDefault)) { + String bitrateValue = sharedPref.getString( + keyprefAudioBitrateValue, getString(R.string.pref_startaudiobitratevalue_default)); + audioStartBitrate = Integer.parseInt(bitrateValue); + } + } + + // Check statistics display option. + boolean displayHud = sharedPrefGetBoolean(R.string.pref_displayhud_key, + CallActivity.EXTRA_DISPLAY_HUD, R.string.pref_displayhud_default, useValuesFromIntent); + + boolean tracing = sharedPrefGetBoolean(R.string.pref_tracing_key, CallActivity.EXTRA_TRACING, + R.string.pref_tracing_default, useValuesFromIntent); + + // Check Enable RtcEventLog. + boolean rtcEventLogEnabled = sharedPrefGetBoolean(R.string.pref_enable_rtceventlog_key, + CallActivity.EXTRA_ENABLE_RTCEVENTLOG, R.string.pref_enable_rtceventlog_default, + useValuesFromIntent); + + // Get datachannel options + boolean dataChannelEnabled = sharedPrefGetBoolean(R.string.pref_enable_datachannel_key, + CallActivity.EXTRA_DATA_CHANNEL_ENABLED, R.string.pref_enable_datachannel_default, + useValuesFromIntent); + boolean ordered = sharedPrefGetBoolean(R.string.pref_ordered_key, CallActivity.EXTRA_ORDERED, + R.string.pref_ordered_default, useValuesFromIntent); + boolean negotiated = sharedPrefGetBoolean(R.string.pref_negotiated_key, + CallActivity.EXTRA_NEGOTIATED, R.string.pref_negotiated_default, useValuesFromIntent); + int maxRetrMs = sharedPrefGetInteger(R.string.pref_max_retransmit_time_ms_key, + CallActivity.EXTRA_MAX_RETRANSMITS_MS, R.string.pref_max_retransmit_time_ms_default, + useValuesFromIntent); + int maxRetr = + sharedPrefGetInteger(R.string.pref_max_retransmits_key, CallActivity.EXTRA_MAX_RETRANSMITS, + R.string.pref_max_retransmits_default, useValuesFromIntent); + int id = sharedPrefGetInteger(R.string.pref_data_id_key, CallActivity.EXTRA_ID, + R.string.pref_data_id_default, useValuesFromIntent); + String protocol = sharedPrefGetString(R.string.pref_data_protocol_key, + CallActivity.EXTRA_PROTOCOL, R.string.pref_data_protocol_default, useValuesFromIntent); + + // Start AppRTCMobile activity. + Log.d(TAG, "Connecting to room " + roomId + " at URL " + roomUrl); + if (validateUrl(roomUrl)) { + Uri uri = Uri.parse(roomUrl); + Intent intent = new Intent(this, CallActivity.class); + intent.setData(uri); + intent.putExtra(CallActivity.EXTRA_ROOMID, roomId); + intent.putExtra(CallActivity.EXTRA_LOOPBACK, loopback); + intent.putExtra(CallActivity.EXTRA_VIDEO_CALL, videoCallEnabled); + intent.putExtra(CallActivity.EXTRA_SCREENCAPTURE, useScreencapture); + intent.putExtra(CallActivity.EXTRA_CAMERA2, useCamera2); + intent.putExtra(CallActivity.EXTRA_VIDEO_WIDTH, videoWidth); + intent.putExtra(CallActivity.EXTRA_VIDEO_HEIGHT, videoHeight); + intent.putExtra(CallActivity.EXTRA_VIDEO_FPS, cameraFps); + intent.putExtra(CallActivity.EXTRA_VIDEO_CAPTUREQUALITYSLIDER_ENABLED, captureQualitySlider); + intent.putExtra(CallActivity.EXTRA_VIDEO_BITRATE, videoStartBitrate); + intent.putExtra(CallActivity.EXTRA_VIDEOCODEC, videoCodec); + intent.putExtra(CallActivity.EXTRA_HWCODEC_ENABLED, hwCodec); + intent.putExtra(CallActivity.EXTRA_CAPTURETOTEXTURE_ENABLED, captureToTexture); + intent.putExtra(CallActivity.EXTRA_FLEXFEC_ENABLED, flexfecEnabled); + intent.putExtra(CallActivity.EXTRA_NOAUDIOPROCESSING_ENABLED, noAudioProcessing); + intent.putExtra(CallActivity.EXTRA_AECDUMP_ENABLED, aecDump); + intent.putExtra(CallActivity.EXTRA_SAVE_INPUT_AUDIO_TO_FILE_ENABLED, saveInputAudioToFile); + intent.putExtra(CallActivity.EXTRA_OPENSLES_ENABLED, useOpenSLES); + intent.putExtra(CallActivity.EXTRA_DISABLE_BUILT_IN_AEC, disableBuiltInAEC); + intent.putExtra(CallActivity.EXTRA_DISABLE_BUILT_IN_AGC, disableBuiltInAGC); + intent.putExtra(CallActivity.EXTRA_DISABLE_BUILT_IN_NS, disableBuiltInNS); + intent.putExtra(CallActivity.EXTRA_DISABLE_WEBRTC_AGC_AND_HPF, disableWebRtcAGCAndHPF); + intent.putExtra(CallActivity.EXTRA_AUDIO_BITRATE, audioStartBitrate); + intent.putExtra(CallActivity.EXTRA_AUDIOCODEC, audioCodec); + intent.putExtra(CallActivity.EXTRA_DISPLAY_HUD, displayHud); + intent.putExtra(CallActivity.EXTRA_TRACING, tracing); + intent.putExtra(CallActivity.EXTRA_ENABLE_RTCEVENTLOG, rtcEventLogEnabled); + intent.putExtra(CallActivity.EXTRA_CMDLINE, commandLineRun); + intent.putExtra(CallActivity.EXTRA_RUNTIME, runTimeMs); + intent.putExtra(CallActivity.EXTRA_DATA_CHANNEL_ENABLED, dataChannelEnabled); + + if (dataChannelEnabled) { + intent.putExtra(CallActivity.EXTRA_ORDERED, ordered); + intent.putExtra(CallActivity.EXTRA_MAX_RETRANSMITS_MS, maxRetrMs); + intent.putExtra(CallActivity.EXTRA_MAX_RETRANSMITS, maxRetr); + intent.putExtra(CallActivity.EXTRA_PROTOCOL, protocol); + intent.putExtra(CallActivity.EXTRA_NEGOTIATED, negotiated); + intent.putExtra(CallActivity.EXTRA_ID, id); + } + + if (useValuesFromIntent) { + if (getIntent().hasExtra(CallActivity.EXTRA_VIDEO_FILE_AS_CAMERA)) { + String videoFileAsCamera = + getIntent().getStringExtra(CallActivity.EXTRA_VIDEO_FILE_AS_CAMERA); + intent.putExtra(CallActivity.EXTRA_VIDEO_FILE_AS_CAMERA, videoFileAsCamera); + } + + if (getIntent().hasExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE)) { + String saveRemoteVideoToFile = + getIntent().getStringExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE); + intent.putExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE, saveRemoteVideoToFile); + } + + if (getIntent().hasExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_WIDTH)) { + int videoOutWidth = + getIntent().getIntExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_WIDTH, 0); + intent.putExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_WIDTH, videoOutWidth); + } + + if (getIntent().hasExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_HEIGHT)) { + int videoOutHeight = + getIntent().getIntExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_HEIGHT, 0); + intent.putExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_HEIGHT, videoOutHeight); + } + } + + startActivityForResult(intent, CONNECTION_REQUEST); + } + } + + private boolean validateUrl(String url) { + if (URLUtil.isHttpsUrl(url) || URLUtil.isHttpUrl(url)) { + return true; + } + + new AlertDialog.Builder(this) + .setTitle(getText(R.string.invalid_url_title)) + .setMessage(getString(R.string.invalid_url_text, url)) + .setCancelable(false) + .setNeutralButton(R.string.ok, + new DialogInterface.OnClickListener() { + @Override + public void onClick(DialogInterface dialog, int id) { + dialog.cancel(); + } + }) + .create() + .show(); + return false; + } + + private final AdapterView.OnItemClickListener roomListClickListener = + new AdapterView.OnItemClickListener() { + @Override + public void onItemClick(AdapterView<?> adapterView, View view, int i, long l) { + String roomId = ((TextView) view).getText().toString(); + connectToRoom(roomId, false, false, false, 0); + } + }; + + private final OnClickListener addFavoriteListener = new OnClickListener() { + @Override + public void onClick(View view) { + String newRoom = roomEditText.getText().toString(); + if (newRoom.length() > 0 && !roomList.contains(newRoom)) { + adapter.add(newRoom); + adapter.notifyDataSetChanged(); + } + } + }; + + private final OnClickListener connectListener = new OnClickListener() { + @Override + public void onClick(View view) { + connectToRoom(roomEditText.getText().toString(), false, false, false, 0); + } + }; +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CpuMonitor.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CpuMonitor.java new file mode 100644 index 0000000000..1c64621864 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CpuMonitor.java @@ -0,0 +1,521 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.annotation.TargetApi; +import android.content.Context; +import android.content.Intent; +import android.content.IntentFilter; +import android.os.BatteryManager; +import android.os.Build; +import android.os.SystemClock; +import android.util.Log; +import androidx.annotation.Nullable; +import java.io.BufferedReader; +import java.io.FileInputStream; +import java.io.FileNotFoundException; +import java.io.IOException; +import java.io.InputStreamReader; +import java.nio.charset.Charset; +import java.util.Arrays; +import java.util.Scanner; +import java.util.concurrent.Executors; +import java.util.concurrent.Future; +import java.util.concurrent.ScheduledExecutorService; +import java.util.concurrent.TimeUnit; + +/** + * Simple CPU monitor. The caller creates a CpuMonitor object which can then + * be used via sampleCpuUtilization() to collect the percentual use of the + * cumulative CPU capacity for all CPUs running at their nominal frequency. 3 + * values are generated: (1) getCpuCurrent() returns the use since the last + * sampleCpuUtilization(), (2) getCpuAvg3() returns the use since 3 prior + * calls, and (3) getCpuAvgAll() returns the use over all SAMPLE_SAVE_NUMBER + * calls. + * + * <p>CPUs in Android are often "offline", and while this of course means 0 Hz + * as current frequency, in this state we cannot even get their nominal + * frequency. We therefore tread carefully, and allow any CPU to be missing. + * Missing CPUs are assumed to have the same nominal frequency as any close + * lower-numbered CPU, but as soon as it is online, we'll get their proper + * frequency and remember it. (Since CPU 0 in practice always seem to be + * online, this unidirectional frequency inheritance should be no problem in + * practice.) + * + * <p>Caveats: + * o No provision made for zany "turbo" mode, common in the x86 world. + * o No provision made for ARM big.LITTLE; if CPU n can switch behind our + * back, we might get incorrect estimates. + * o This is not thread-safe. To call asynchronously, create different + * CpuMonitor objects. + * + * <p>If we can gather enough info to generate a sensible result, + * sampleCpuUtilization returns true. It is designed to never throw an + * exception. + * + * <p>sampleCpuUtilization should not be called too often in its present form, + * since then deltas would be small and the percent values would fluctuate and + * be unreadable. If it is desirable to call it more often than say once per + * second, one would need to increase SAMPLE_SAVE_NUMBER and probably use + * Queue<Integer> to avoid copying overhead. + * + * <p>Known problems: + * 1. Nexus 7 devices running Kitkat have a kernel which often output an + * incorrect 'idle' field in /proc/stat. The value is close to twice the + * correct value, and then returns to back to correct reading. Both when + * jumping up and back down we might create faulty CPU load readings. + */ +class CpuMonitor { + private static final String TAG = "CpuMonitor"; + private static final int MOVING_AVERAGE_SAMPLES = 5; + + private static final int CPU_STAT_SAMPLE_PERIOD_MS = 2000; + private static final int CPU_STAT_LOG_PERIOD_MS = 6000; + + private final Context appContext; + // User CPU usage at current frequency. + private final MovingAverage userCpuUsage; + // System CPU usage at current frequency. + private final MovingAverage systemCpuUsage; + // Total CPU usage relative to maximum frequency. + private final MovingAverage totalCpuUsage; + // CPU frequency in percentage from maximum. + private final MovingAverage frequencyScale; + + @Nullable + private ScheduledExecutorService executor; + private long lastStatLogTimeMs; + private long[] cpuFreqMax; + private int cpusPresent; + private int actualCpusPresent; + private boolean initialized; + private boolean cpuOveruse; + private String[] maxPath; + private String[] curPath; + private double[] curFreqScales; + @Nullable + private ProcStat lastProcStat; + + private static class ProcStat { + final long userTime; + final long systemTime; + final long idleTime; + + ProcStat(long userTime, long systemTime, long idleTime) { + this.userTime = userTime; + this.systemTime = systemTime; + this.idleTime = idleTime; + } + } + + private static class MovingAverage { + private final int size; + private double sum; + private double currentValue; + private double[] circBuffer; + private int circBufferIndex; + + public MovingAverage(int size) { + if (size <= 0) { + throw new AssertionError("Size value in MovingAverage ctor should be positive."); + } + this.size = size; + circBuffer = new double[size]; + } + + public void reset() { + Arrays.fill(circBuffer, 0); + circBufferIndex = 0; + sum = 0; + currentValue = 0; + } + + public void addValue(double value) { + sum -= circBuffer[circBufferIndex]; + circBuffer[circBufferIndex++] = value; + currentValue = value; + sum += value; + if (circBufferIndex >= size) { + circBufferIndex = 0; + } + } + + public double getCurrent() { + return currentValue; + } + + public double getAverage() { + return sum / (double) size; + } + } + + public static boolean isSupported() { + return Build.VERSION.SDK_INT < Build.VERSION_CODES.N; + } + + public CpuMonitor(Context context) { + if (!isSupported()) { + throw new RuntimeException("CpuMonitor is not supported on this Android version."); + } + + Log.d(TAG, "CpuMonitor ctor."); + appContext = context.getApplicationContext(); + userCpuUsage = new MovingAverage(MOVING_AVERAGE_SAMPLES); + systemCpuUsage = new MovingAverage(MOVING_AVERAGE_SAMPLES); + totalCpuUsage = new MovingAverage(MOVING_AVERAGE_SAMPLES); + frequencyScale = new MovingAverage(MOVING_AVERAGE_SAMPLES); + lastStatLogTimeMs = SystemClock.elapsedRealtime(); + + scheduleCpuUtilizationTask(); + } + + public void pause() { + if (executor != null) { + Log.d(TAG, "pause"); + executor.shutdownNow(); + executor = null; + } + } + + public void resume() { + Log.d(TAG, "resume"); + resetStat(); + scheduleCpuUtilizationTask(); + } + + // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression. + @SuppressWarnings("NoSynchronizedMethodCheck") + public synchronized void reset() { + if (executor != null) { + Log.d(TAG, "reset"); + resetStat(); + cpuOveruse = false; + } + } + + // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression. + @SuppressWarnings("NoSynchronizedMethodCheck") + public synchronized int getCpuUsageCurrent() { + return doubleToPercent(userCpuUsage.getCurrent() + systemCpuUsage.getCurrent()); + } + + // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression. + @SuppressWarnings("NoSynchronizedMethodCheck") + public synchronized int getCpuUsageAverage() { + return doubleToPercent(userCpuUsage.getAverage() + systemCpuUsage.getAverage()); + } + + // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression. + @SuppressWarnings("NoSynchronizedMethodCheck") + public synchronized int getFrequencyScaleAverage() { + return doubleToPercent(frequencyScale.getAverage()); + } + + private void scheduleCpuUtilizationTask() { + if (executor != null) { + executor.shutdownNow(); + executor = null; + } + + executor = Executors.newSingleThreadScheduledExecutor(); + @SuppressWarnings("unused") // Prevent downstream linter warnings. + Future<?> possiblyIgnoredError = executor.scheduleAtFixedRate(new Runnable() { + @Override + public void run() { + cpuUtilizationTask(); + } + }, 0, CPU_STAT_SAMPLE_PERIOD_MS, TimeUnit.MILLISECONDS); + } + + private void cpuUtilizationTask() { + boolean cpuMonitorAvailable = sampleCpuUtilization(); + if (cpuMonitorAvailable + && SystemClock.elapsedRealtime() - lastStatLogTimeMs >= CPU_STAT_LOG_PERIOD_MS) { + lastStatLogTimeMs = SystemClock.elapsedRealtime(); + String statString = getStatString(); + Log.d(TAG, statString); + } + } + + private void init() { + try (FileInputStream fin = new FileInputStream("/sys/devices/system/cpu/present"); + InputStreamReader streamReader = new InputStreamReader(fin, Charset.forName("UTF-8")); + BufferedReader reader = new BufferedReader(streamReader); + Scanner scanner = new Scanner(reader).useDelimiter("[-\n]");) { + scanner.nextInt(); // Skip leading number 0. + cpusPresent = 1 + scanner.nextInt(); + scanner.close(); + } catch (FileNotFoundException e) { + Log.e(TAG, "Cannot do CPU stats since /sys/devices/system/cpu/present is missing"); + } catch (IOException e) { + Log.e(TAG, "Error closing file"); + } catch (Exception e) { + Log.e(TAG, "Cannot do CPU stats due to /sys/devices/system/cpu/present parsing problem"); + } + + cpuFreqMax = new long[cpusPresent]; + maxPath = new String[cpusPresent]; + curPath = new String[cpusPresent]; + curFreqScales = new double[cpusPresent]; + for (int i = 0; i < cpusPresent; i++) { + cpuFreqMax[i] = 0; // Frequency "not yet determined". + curFreqScales[i] = 0; + maxPath[i] = "/sys/devices/system/cpu/cpu" + i + "/cpufreq/cpuinfo_max_freq"; + curPath[i] = "/sys/devices/system/cpu/cpu" + i + "/cpufreq/scaling_cur_freq"; + } + + lastProcStat = new ProcStat(0, 0, 0); + resetStat(); + + initialized = true; + } + + private synchronized void resetStat() { + userCpuUsage.reset(); + systemCpuUsage.reset(); + totalCpuUsage.reset(); + frequencyScale.reset(); + lastStatLogTimeMs = SystemClock.elapsedRealtime(); + } + + private int getBatteryLevel() { + // Use sticky broadcast with null receiver to read battery level once only. + Intent intent = appContext.registerReceiver( + null /* receiver */, new IntentFilter(Intent.ACTION_BATTERY_CHANGED)); + + int batteryLevel = 0; + int batteryScale = intent.getIntExtra(BatteryManager.EXTRA_SCALE, 100); + if (batteryScale > 0) { + batteryLevel = + (int) (100f * intent.getIntExtra(BatteryManager.EXTRA_LEVEL, 0) / batteryScale); + } + return batteryLevel; + } + + /** + * Re-measure CPU use. Call this method at an interval of around 1/s. + * This method returns true on success. The fields + * cpuCurrent, cpuAvg3, and cpuAvgAll are updated on success, and represents: + * cpuCurrent: The CPU use since the last sampleCpuUtilization call. + * cpuAvg3: The average CPU over the last 3 calls. + * cpuAvgAll: The average CPU over the last SAMPLE_SAVE_NUMBER calls. + */ + private synchronized boolean sampleCpuUtilization() { + long lastSeenMaxFreq = 0; + long cpuFreqCurSum = 0; + long cpuFreqMaxSum = 0; + + if (!initialized) { + init(); + } + if (cpusPresent == 0) { + return false; + } + + actualCpusPresent = 0; + for (int i = 0; i < cpusPresent; i++) { + /* + * For each CPU, attempt to first read its max frequency, then its + * current frequency. Once as the max frequency for a CPU is found, + * save it in cpuFreqMax[]. + */ + + curFreqScales[i] = 0; + if (cpuFreqMax[i] == 0) { + // We have never found this CPU's max frequency. Attempt to read it. + long cpufreqMax = readFreqFromFile(maxPath[i]); + if (cpufreqMax > 0) { + Log.d(TAG, "Core " + i + ". Max frequency: " + cpufreqMax); + lastSeenMaxFreq = cpufreqMax; + cpuFreqMax[i] = cpufreqMax; + maxPath[i] = null; // Kill path to free its memory. + } + } else { + lastSeenMaxFreq = cpuFreqMax[i]; // A valid, previously read value. + } + + long cpuFreqCur = readFreqFromFile(curPath[i]); + if (cpuFreqCur == 0 && lastSeenMaxFreq == 0) { + // No current frequency information for this CPU core - ignore it. + continue; + } + if (cpuFreqCur > 0) { + actualCpusPresent++; + } + cpuFreqCurSum += cpuFreqCur; + + /* Here, lastSeenMaxFreq might come from + * 1. cpuFreq[i], or + * 2. a previous iteration, or + * 3. a newly read value, or + * 4. hypothetically from the pre-loop dummy. + */ + cpuFreqMaxSum += lastSeenMaxFreq; + if (lastSeenMaxFreq > 0) { + curFreqScales[i] = (double) cpuFreqCur / lastSeenMaxFreq; + } + } + + if (cpuFreqCurSum == 0 || cpuFreqMaxSum == 0) { + Log.e(TAG, "Could not read max or current frequency for any CPU"); + return false; + } + + /* + * Since the cycle counts are for the period between the last invocation + * and this present one, we average the percentual CPU frequencies between + * now and the beginning of the measurement period. This is significantly + * incorrect only if the frequencies have peeked or dropped in between the + * invocations. + */ + double currentFrequencyScale = cpuFreqCurSum / (double) cpuFreqMaxSum; + if (frequencyScale.getCurrent() > 0) { + currentFrequencyScale = (frequencyScale.getCurrent() + currentFrequencyScale) * 0.5; + } + + ProcStat procStat = readProcStat(); + if (procStat == null) { + return false; + } + + long diffUserTime = procStat.userTime - lastProcStat.userTime; + long diffSystemTime = procStat.systemTime - lastProcStat.systemTime; + long diffIdleTime = procStat.idleTime - lastProcStat.idleTime; + long allTime = diffUserTime + diffSystemTime + diffIdleTime; + + if (currentFrequencyScale == 0 || allTime == 0) { + return false; + } + + // Update statistics. + frequencyScale.addValue(currentFrequencyScale); + + double currentUserCpuUsage = diffUserTime / (double) allTime; + userCpuUsage.addValue(currentUserCpuUsage); + + double currentSystemCpuUsage = diffSystemTime / (double) allTime; + systemCpuUsage.addValue(currentSystemCpuUsage); + + double currentTotalCpuUsage = + (currentUserCpuUsage + currentSystemCpuUsage) * currentFrequencyScale; + totalCpuUsage.addValue(currentTotalCpuUsage); + + // Save new measurements for next round's deltas. + lastProcStat = procStat; + + return true; + } + + private int doubleToPercent(double d) { + return (int) (d * 100 + 0.5); + } + + private synchronized String getStatString() { + StringBuilder stat = new StringBuilder(); + stat.append("CPU User: ") + .append(doubleToPercent(userCpuUsage.getCurrent())) + .append("/") + .append(doubleToPercent(userCpuUsage.getAverage())) + .append(". System: ") + .append(doubleToPercent(systemCpuUsage.getCurrent())) + .append("/") + .append(doubleToPercent(systemCpuUsage.getAverage())) + .append(". Freq: ") + .append(doubleToPercent(frequencyScale.getCurrent())) + .append("/") + .append(doubleToPercent(frequencyScale.getAverage())) + .append(". Total usage: ") + .append(doubleToPercent(totalCpuUsage.getCurrent())) + .append("/") + .append(doubleToPercent(totalCpuUsage.getAverage())) + .append(". Cores: ") + .append(actualCpusPresent); + stat.append("( "); + for (int i = 0; i < cpusPresent; i++) { + stat.append(doubleToPercent(curFreqScales[i])).append(" "); + } + stat.append("). Battery: ").append(getBatteryLevel()); + if (cpuOveruse) { + stat.append(". Overuse."); + } + return stat.toString(); + } + + /** + * Read a single integer value from the named file. Return the read value + * or if an error occurs return 0. + */ + private long readFreqFromFile(String fileName) { + long number = 0; + try (FileInputStream stream = new FileInputStream(fileName); + InputStreamReader streamReader = new InputStreamReader(stream, Charset.forName("UTF-8")); + BufferedReader reader = new BufferedReader(streamReader)) { + String line = reader.readLine(); + number = parseLong(line); + } catch (FileNotFoundException e) { + // CPU core is off, so file with its scaling frequency .../cpufreq/scaling_cur_freq + // is not present. This is not an error. + } catch (IOException e) { + // CPU core is off, so file with its scaling frequency .../cpufreq/scaling_cur_freq + // is empty. This is not an error. + } + return number; + } + + private static long parseLong(String value) { + long number = 0; + try { + number = Long.parseLong(value); + } catch (NumberFormatException e) { + Log.e(TAG, "parseLong error.", e); + } + return number; + } + + /* + * Read the current utilization of all CPUs using the cumulative first line + * of /proc/stat. + */ + @SuppressWarnings("StringSplitter") + private @Nullable ProcStat readProcStat() { + long userTime = 0; + long systemTime = 0; + long idleTime = 0; + try (FileInputStream stream = new FileInputStream("/proc/stat"); + InputStreamReader streamReader = new InputStreamReader(stream, Charset.forName("UTF-8")); + BufferedReader reader = new BufferedReader(streamReader)) { + // line should contain something like this: + // cpu 5093818 271838 3512830 165934119 101374 447076 272086 0 0 0 + // user nice system idle iowait irq softirq + String line = reader.readLine(); + String[] lines = line.split("\\s+"); + int length = lines.length; + if (length >= 5) { + userTime = parseLong(lines[1]); // user + userTime += parseLong(lines[2]); // nice + systemTime = parseLong(lines[3]); // system + idleTime = parseLong(lines[4]); // idle + } + if (length >= 8) { + userTime += parseLong(lines[5]); // iowait + systemTime += parseLong(lines[6]); // irq + systemTime += parseLong(lines[7]); // softirq + } + } catch (FileNotFoundException e) { + Log.e(TAG, "Cannot open /proc/stat for reading", e); + return null; + } catch (Exception e) { + Log.e(TAG, "Problems parsing /proc/stat", e); + return null; + } + return new ProcStat(userTime, systemTime, idleTime); + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/DirectRTCClient.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/DirectRTCClient.java new file mode 100644 index 0000000000..1b113e1398 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/DirectRTCClient.java @@ -0,0 +1,346 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.util.Log; +import androidx.annotation.Nullable; +import java.util.ArrayList; +import java.util.concurrent.ExecutorService; +import java.util.concurrent.Executors; +import java.util.regex.Matcher; +import java.util.regex.Pattern; +import org.json.JSONArray; +import org.json.JSONException; +import org.json.JSONObject; +import org.webrtc.IceCandidate; +import org.webrtc.SessionDescription; + +/** + * Implementation of AppRTCClient that uses direct TCP connection as the signaling channel. + * This eliminates the need for an external server. This class does not support loopback + * connections. + */ +public class DirectRTCClient implements AppRTCClient, TCPChannelClient.TCPChannelEvents { + private static final String TAG = "DirectRTCClient"; + private static final int DEFAULT_PORT = 8888; + + // Regex pattern used for checking if room id looks like an IP. + static final Pattern IP_PATTERN = Pattern.compile("(" + // IPv4 + + "((\\d+\\.){3}\\d+)|" + // IPv6 + + "\\[((([0-9a-fA-F]{1,4}:)*[0-9a-fA-F]{1,4})?::" + + "(([0-9a-fA-F]{1,4}:)*[0-9a-fA-F]{1,4})?)\\]|" + + "\\[(([0-9a-fA-F]{1,4}:){7}[0-9a-fA-F]{1,4})\\]|" + // IPv6 without [] + + "((([0-9a-fA-F]{1,4}:)*[0-9a-fA-F]{1,4})?::(([0-9a-fA-F]{1,4}:)*[0-9a-fA-F]{1,4})?)|" + + "(([0-9a-fA-F]{1,4}:){7}[0-9a-fA-F]{1,4})|" + // Literals + + "localhost" + + ")" + // Optional port number + + "(:(\\d+))?"); + + private final ExecutorService executor; + private final SignalingEvents events; + @Nullable + private TCPChannelClient tcpClient; + private RoomConnectionParameters connectionParameters; + + private enum ConnectionState { NEW, CONNECTED, CLOSED, ERROR } + + // All alterations of the room state should be done from inside the looper thread. + private ConnectionState roomState; + + public DirectRTCClient(SignalingEvents events) { + this.events = events; + + executor = Executors.newSingleThreadExecutor(); + roomState = ConnectionState.NEW; + } + + /** + * Connects to the room, roomId in connectionsParameters is required. roomId must be a valid + * IP address matching IP_PATTERN. + */ + @Override + public void connectToRoom(RoomConnectionParameters connectionParameters) { + this.connectionParameters = connectionParameters; + + if (connectionParameters.loopback) { + reportError("Loopback connections aren't supported by DirectRTCClient."); + } + + executor.execute(new Runnable() { + @Override + public void run() { + connectToRoomInternal(); + } + }); + } + + @Override + public void disconnectFromRoom() { + executor.execute(new Runnable() { + @Override + public void run() { + disconnectFromRoomInternal(); + } + }); + } + + /** + * Connects to the room. + * + * Runs on the looper thread. + */ + private void connectToRoomInternal() { + this.roomState = ConnectionState.NEW; + + String endpoint = connectionParameters.roomId; + + Matcher matcher = IP_PATTERN.matcher(endpoint); + if (!matcher.matches()) { + reportError("roomId must match IP_PATTERN for DirectRTCClient."); + return; + } + + String ip = matcher.group(1); + String portStr = matcher.group(matcher.groupCount()); + int port; + + if (portStr != null) { + try { + port = Integer.parseInt(portStr); + } catch (NumberFormatException e) { + reportError("Invalid port number: " + portStr); + return; + } + } else { + port = DEFAULT_PORT; + } + + tcpClient = new TCPChannelClient(executor, this, ip, port); + } + + /** + * Disconnects from the room. + * + * Runs on the looper thread. + */ + private void disconnectFromRoomInternal() { + roomState = ConnectionState.CLOSED; + + if (tcpClient != null) { + tcpClient.disconnect(); + tcpClient = null; + } + executor.shutdown(); + } + + @Override + public void sendOfferSdp(final SessionDescription sdp) { + executor.execute(new Runnable() { + @Override + public void run() { + if (roomState != ConnectionState.CONNECTED) { + reportError("Sending offer SDP in non connected state."); + return; + } + JSONObject json = new JSONObject(); + jsonPut(json, "sdp", sdp.description); + jsonPut(json, "type", "offer"); + sendMessage(json.toString()); + } + }); + } + + @Override + public void sendAnswerSdp(final SessionDescription sdp) { + executor.execute(new Runnable() { + @Override + public void run() { + JSONObject json = new JSONObject(); + jsonPut(json, "sdp", sdp.description); + jsonPut(json, "type", "answer"); + sendMessage(json.toString()); + } + }); + } + + @Override + public void sendLocalIceCandidate(final IceCandidate candidate) { + executor.execute(new Runnable() { + @Override + public void run() { + JSONObject json = new JSONObject(); + jsonPut(json, "type", "candidate"); + jsonPut(json, "label", candidate.sdpMLineIndex); + jsonPut(json, "id", candidate.sdpMid); + jsonPut(json, "candidate", candidate.sdp); + + if (roomState != ConnectionState.CONNECTED) { + reportError("Sending ICE candidate in non connected state."); + return; + } + sendMessage(json.toString()); + } + }); + } + + /** Send removed Ice candidates to the other participant. */ + @Override + public void sendLocalIceCandidateRemovals(final IceCandidate[] candidates) { + executor.execute(new Runnable() { + @Override + public void run() { + JSONObject json = new JSONObject(); + jsonPut(json, "type", "remove-candidates"); + JSONArray jsonArray = new JSONArray(); + for (final IceCandidate candidate : candidates) { + jsonArray.put(toJsonCandidate(candidate)); + } + jsonPut(json, "candidates", jsonArray); + + if (roomState != ConnectionState.CONNECTED) { + reportError("Sending ICE candidate removals in non connected state."); + return; + } + sendMessage(json.toString()); + } + }); + } + + // ------------------------------------------------------------------- + // TCPChannelClient event handlers + + /** + * If the client is the server side, this will trigger onConnectedToRoom. + */ + @Override + public void onTCPConnected(boolean isServer) { + if (isServer) { + roomState = ConnectionState.CONNECTED; + + SignalingParameters parameters = new SignalingParameters( + // Ice servers are not needed for direct connections. + new ArrayList<>(), + isServer, // Server side acts as the initiator on direct connections. + null, // clientId + null, // wssUrl + null, // wwsPostUrl + null, // offerSdp + null // iceCandidates + ); + events.onConnectedToRoom(parameters); + } + } + + @Override + public void onTCPMessage(String msg) { + try { + JSONObject json = new JSONObject(msg); + String type = json.optString("type"); + if (type.equals("candidate")) { + events.onRemoteIceCandidate(toJavaCandidate(json)); + } else if (type.equals("remove-candidates")) { + JSONArray candidateArray = json.getJSONArray("candidates"); + IceCandidate[] candidates = new IceCandidate[candidateArray.length()]; + for (int i = 0; i < candidateArray.length(); ++i) { + candidates[i] = toJavaCandidate(candidateArray.getJSONObject(i)); + } + events.onRemoteIceCandidatesRemoved(candidates); + } else if (type.equals("answer")) { + SessionDescription sdp = new SessionDescription( + SessionDescription.Type.fromCanonicalForm(type), json.getString("sdp")); + events.onRemoteDescription(sdp); + } else if (type.equals("offer")) { + SessionDescription sdp = new SessionDescription( + SessionDescription.Type.fromCanonicalForm(type), json.getString("sdp")); + + SignalingParameters parameters = new SignalingParameters( + // Ice servers are not needed for direct connections. + new ArrayList<>(), + false, // This code will only be run on the client side. So, we are not the initiator. + null, // clientId + null, // wssUrl + null, // wssPostUrl + sdp, // offerSdp + null // iceCandidates + ); + roomState = ConnectionState.CONNECTED; + events.onConnectedToRoom(parameters); + } else { + reportError("Unexpected TCP message: " + msg); + } + } catch (JSONException e) { + reportError("TCP message JSON parsing error: " + e.toString()); + } + } + + @Override + public void onTCPError(String description) { + reportError("TCP connection error: " + description); + } + + @Override + public void onTCPClose() { + events.onChannelClose(); + } + + // -------------------------------------------------------------------- + // Helper functions. + private void reportError(final String errorMessage) { + Log.e(TAG, errorMessage); + executor.execute(new Runnable() { + @Override + public void run() { + if (roomState != ConnectionState.ERROR) { + roomState = ConnectionState.ERROR; + events.onChannelError(errorMessage); + } + } + }); + } + + private void sendMessage(final String message) { + executor.execute(new Runnable() { + @Override + public void run() { + tcpClient.send(message); + } + }); + } + + // Put a `key`->`value` mapping in `json`. + private static void jsonPut(JSONObject json, String key, Object value) { + try { + json.put(key, value); + } catch (JSONException e) { + throw new RuntimeException(e); + } + } + + // Converts a Java candidate to a JSONObject. + private static JSONObject toJsonCandidate(final IceCandidate candidate) { + JSONObject json = new JSONObject(); + jsonPut(json, "label", candidate.sdpMLineIndex); + jsonPut(json, "id", candidate.sdpMid); + jsonPut(json, "candidate", candidate.sdp); + return json; + } + + // Converts a JSON candidate to a Java object. + private static IceCandidate toJavaCandidate(JSONObject json) throws JSONException { + return new IceCandidate( + json.getString("id"), json.getInt("label"), json.getString("candidate")); + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/HudFragment.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/HudFragment.java new file mode 100644 index 0000000000..94ca05549a --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/HudFragment.java @@ -0,0 +1,102 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.app.Fragment; +import android.os.Bundle; +import android.view.LayoutInflater; +import android.view.View; +import android.view.ViewGroup; +import android.widget.ImageButton; +import android.widget.TextView; +import org.webrtc.RTCStats; +import org.webrtc.RTCStatsReport; + +/** + * Fragment for HUD statistics display. + */ +public class HudFragment extends Fragment { + private TextView statView; + private ImageButton toggleDebugButton; + private boolean displayHud; + private volatile boolean isRunning; + private CpuMonitor cpuMonitor; + + @Override + public View onCreateView( + LayoutInflater inflater, ViewGroup container, Bundle savedInstanceState) { + View controlView = inflater.inflate(R.layout.fragment_hud, container, false); + + // Create UI controls. + statView = controlView.findViewById(R.id.hud_stat_call); + toggleDebugButton = controlView.findViewById(R.id.button_toggle_debug); + + toggleDebugButton.setOnClickListener(new View.OnClickListener() { + @Override + public void onClick(View view) { + if (displayHud) { + statView.setVisibility( + statView.getVisibility() == View.VISIBLE ? View.INVISIBLE : View.VISIBLE); + } + } + }); + + return controlView; + } + + @Override + public void onStart() { + super.onStart(); + + Bundle args = getArguments(); + if (args != null) { + displayHud = args.getBoolean(CallActivity.EXTRA_DISPLAY_HUD, false); + } + int visibility = displayHud ? View.VISIBLE : View.INVISIBLE; + statView.setVisibility(View.INVISIBLE); + toggleDebugButton.setVisibility(visibility); + isRunning = true; + } + + @Override + public void onStop() { + isRunning = false; + super.onStop(); + } + + public void setCpuMonitor(CpuMonitor cpuMonitor) { + this.cpuMonitor = cpuMonitor; + } + + public void updateEncoderStatistics(final RTCStatsReport report) { + if (!isRunning || !displayHud) { + return; + } + + StringBuilder sb = new StringBuilder(); + + if (cpuMonitor != null) { + sb.append("CPU%: ") + .append(cpuMonitor.getCpuUsageCurrent()) + .append("/") + .append(cpuMonitor.getCpuUsageAverage()) + .append(". Freq: ") + .append(cpuMonitor.getFrequencyScaleAverage()) + .append("\n"); + } + + for (RTCStats stat : report.getStatsMap().values()) { + sb.append(stat.toString()).append("\n"); + } + + statView.setText(sb.toString()); + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java new file mode 100644 index 0000000000..7bdce00b2f --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java @@ -0,0 +1,1402 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.content.Context; +import android.os.Environment; +import android.os.ParcelFileDescriptor; +import android.util.Log; +import androidx.annotation.Nullable; +import java.io.File; +import java.io.IOException; +import java.nio.ByteBuffer; +import java.nio.charset.Charset; +import java.text.DateFormat; +import java.text.SimpleDateFormat; +import java.util.ArrayList; +import java.util.Arrays; +import java.util.Collections; +import java.util.Date; +import java.util.Iterator; +import java.util.List; +import java.util.Locale; +import java.util.Timer; +import java.util.TimerTask; +import java.util.concurrent.ExecutorService; +import java.util.concurrent.Executors; +import java.util.regex.Matcher; +import java.util.regex.Pattern; +import org.appspot.apprtc.AppRTCClient.SignalingParameters; +import org.appspot.apprtc.RecordedAudioToFileController; +import org.webrtc.AddIceObserver; +import org.webrtc.AudioSource; +import org.webrtc.AudioTrack; +import org.webrtc.CameraVideoCapturer; +import org.webrtc.CandidatePairChangeEvent; +import org.webrtc.DataChannel; +import org.webrtc.DefaultVideoDecoderFactory; +import org.webrtc.DefaultVideoEncoderFactory; +import org.webrtc.EglBase; +import org.webrtc.IceCandidate; +import org.webrtc.IceCandidateErrorEvent; +import org.webrtc.Logging; +import org.webrtc.MediaConstraints; +import org.webrtc.MediaStream; +import org.webrtc.MediaStreamTrack; +import org.webrtc.PeerConnection; +import org.webrtc.PeerConnection.IceConnectionState; +import org.webrtc.PeerConnection.PeerConnectionState; +import org.webrtc.PeerConnectionFactory; +import org.webrtc.RTCStatsCollectorCallback; +import org.webrtc.RTCStatsReport; +import org.webrtc.RtpParameters; +import org.webrtc.RtpReceiver; +import org.webrtc.RtpSender; +import org.webrtc.RtpTransceiver; +import org.webrtc.SdpObserver; +import org.webrtc.SessionDescription; +import org.webrtc.SoftwareVideoDecoderFactory; +import org.webrtc.SoftwareVideoEncoderFactory; +import org.webrtc.SurfaceTextureHelper; +import org.webrtc.VideoCapturer; +import org.webrtc.VideoDecoderFactory; +import org.webrtc.VideoEncoderFactory; +import org.webrtc.VideoSink; +import org.webrtc.VideoSource; +import org.webrtc.VideoTrack; +import org.webrtc.audio.AudioDeviceModule; +import org.webrtc.audio.JavaAudioDeviceModule; +import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordErrorCallback; +import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordStateCallback; +import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackErrorCallback; +import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackStateCallback; + +/** + * Peer connection client implementation. + * + * <p>All public methods are routed to local looper thread. + * All PeerConnectionEvents callbacks are invoked from the same looper thread. + * This class is a singleton. + */ +public class PeerConnectionClient { + public static final String VIDEO_TRACK_ID = "ARDAMSv0"; + public static final String AUDIO_TRACK_ID = "ARDAMSa0"; + public static final String VIDEO_TRACK_TYPE = "video"; + private static final String TAG = "PCRTCClient"; + private static final String VIDEO_CODEC_VP8 = "VP8"; + private static final String VIDEO_CODEC_VP9 = "VP9"; + private static final String VIDEO_CODEC_H264 = "H264"; + private static final String VIDEO_CODEC_H264_BASELINE = "H264 Baseline"; + private static final String VIDEO_CODEC_H264_HIGH = "H264 High"; + private static final String VIDEO_CODEC_AV1 = "AV1"; + private static final String AUDIO_CODEC_OPUS = "opus"; + private static final String AUDIO_CODEC_ISAC = "ISAC"; + private static final String VIDEO_CODEC_PARAM_START_BITRATE = "x-google-start-bitrate"; + private static final String VIDEO_FLEXFEC_FIELDTRIAL = + "WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/"; + private static final String VIDEO_VP8_INTEL_HW_ENCODER_FIELDTRIAL = "WebRTC-IntelVP8/Enabled/"; + private static final String DISABLE_WEBRTC_AGC_FIELDTRIAL = + "WebRTC-Audio-MinimizeResamplingOnMobile/Enabled/"; + private static final String AUDIO_CODEC_PARAM_BITRATE = "maxaveragebitrate"; + private static final String AUDIO_ECHO_CANCELLATION_CONSTRAINT = "googEchoCancellation"; + private static final String AUDIO_AUTO_GAIN_CONTROL_CONSTRAINT = "googAutoGainControl"; + private static final String AUDIO_HIGH_PASS_FILTER_CONSTRAINT = "googHighpassFilter"; + private static final String AUDIO_NOISE_SUPPRESSION_CONSTRAINT = "googNoiseSuppression"; + private static final String DTLS_SRTP_KEY_AGREEMENT_CONSTRAINT = "DtlsSrtpKeyAgreement"; + private static final int HD_VIDEO_WIDTH = 1280; + private static final int HD_VIDEO_HEIGHT = 720; + private static final int BPS_IN_KBPS = 1000; + private static final String RTCEVENTLOG_OUTPUT_DIR_NAME = "rtc_event_log"; + + // Executor thread is started once in private ctor and is used for all + // peer connection API calls to ensure new peer connection factory is + // created on the same thread as previously destroyed factory. + private static final ExecutorService executor = Executors.newSingleThreadExecutor(); + + private final PCObserver pcObserver = new PCObserver(); + private final SDPObserver sdpObserver = new SDPObserver(); + private final Timer statsTimer = new Timer(); + private final EglBase rootEglBase; + private final Context appContext; + private final PeerConnectionParameters peerConnectionParameters; + private final PeerConnectionEvents events; + + @Nullable + private PeerConnectionFactory factory; + @Nullable + private PeerConnection peerConnection; + @Nullable + private AudioSource audioSource; + @Nullable private SurfaceTextureHelper surfaceTextureHelper; + @Nullable private VideoSource videoSource; + private boolean preferIsac; + private boolean videoCapturerStopped; + private boolean isError; + @Nullable + private VideoSink localRender; + @Nullable private List<VideoSink> remoteSinks; + private SignalingParameters signalingParameters; + private int videoWidth; + private int videoHeight; + private int videoFps; + private MediaConstraints audioConstraints; + private MediaConstraints sdpMediaConstraints; + // Queued remote ICE candidates are consumed only after both local and + // remote descriptions are set. Similarly local ICE candidates are sent to + // remote peer after both local and remote description are set. + @Nullable + private List<IceCandidate> queuedRemoteCandidates; + private boolean isInitiator; + @Nullable private SessionDescription localDescription; // either offer or answer description + @Nullable + private VideoCapturer videoCapturer; + // enableVideo is set to true if video should be rendered and sent. + private boolean renderVideo = true; + @Nullable + private VideoTrack localVideoTrack; + @Nullable + private VideoTrack remoteVideoTrack; + @Nullable + private RtpSender localVideoSender; + // enableAudio is set to true if audio should be sent. + private boolean enableAudio = true; + @Nullable + private AudioTrack localAudioTrack; + @Nullable + private DataChannel dataChannel; + private final boolean dataChannelEnabled; + // Enable RtcEventLog. + @Nullable + private RtcEventLog rtcEventLog; + // Implements the WebRtcAudioRecordSamplesReadyCallback interface and writes + // recorded audio samples to an output file. + @Nullable private RecordedAudioToFileController saveRecordedAudioToFile; + + /** + * Peer connection parameters. + */ + public static class DataChannelParameters { + public final boolean ordered; + public final int maxRetransmitTimeMs; + public final int maxRetransmits; + public final String protocol; + public final boolean negotiated; + public final int id; + + public DataChannelParameters(boolean ordered, int maxRetransmitTimeMs, int maxRetransmits, + String protocol, boolean negotiated, int id) { + this.ordered = ordered; + this.maxRetransmitTimeMs = maxRetransmitTimeMs; + this.maxRetransmits = maxRetransmits; + this.protocol = protocol; + this.negotiated = negotiated; + this.id = id; + } + } + + /** + * Peer connection parameters. + */ + public static class PeerConnectionParameters { + public final boolean videoCallEnabled; + public final boolean loopback; + public final boolean tracing; + public final int videoWidth; + public final int videoHeight; + public final int videoFps; + public final int videoMaxBitrate; + public final String videoCodec; + public final boolean videoCodecHwAcceleration; + public final boolean videoFlexfecEnabled; + public final int audioStartBitrate; + public final String audioCodec; + public final boolean noAudioProcessing; + public final boolean aecDump; + public final boolean saveInputAudioToFile; + public final boolean useOpenSLES; + public final boolean disableBuiltInAEC; + public final boolean disableBuiltInAGC; + public final boolean disableBuiltInNS; + public final boolean disableWebRtcAGCAndHPF; + public final boolean enableRtcEventLog; + private final DataChannelParameters dataChannelParameters; + + public PeerConnectionParameters(boolean videoCallEnabled, boolean loopback, boolean tracing, + int videoWidth, int videoHeight, int videoFps, int videoMaxBitrate, String videoCodec, + boolean videoCodecHwAcceleration, boolean videoFlexfecEnabled, int audioStartBitrate, + String audioCodec, boolean noAudioProcessing, boolean aecDump, boolean saveInputAudioToFile, + boolean useOpenSLES, boolean disableBuiltInAEC, boolean disableBuiltInAGC, + boolean disableBuiltInNS, boolean disableWebRtcAGCAndHPF, boolean enableRtcEventLog, + DataChannelParameters dataChannelParameters) { + this.videoCallEnabled = videoCallEnabled; + this.loopback = loopback; + this.tracing = tracing; + this.videoWidth = videoWidth; + this.videoHeight = videoHeight; + this.videoFps = videoFps; + this.videoMaxBitrate = videoMaxBitrate; + this.videoCodec = videoCodec; + this.videoFlexfecEnabled = videoFlexfecEnabled; + this.videoCodecHwAcceleration = videoCodecHwAcceleration; + this.audioStartBitrate = audioStartBitrate; + this.audioCodec = audioCodec; + this.noAudioProcessing = noAudioProcessing; + this.aecDump = aecDump; + this.saveInputAudioToFile = saveInputAudioToFile; + this.useOpenSLES = useOpenSLES; + this.disableBuiltInAEC = disableBuiltInAEC; + this.disableBuiltInAGC = disableBuiltInAGC; + this.disableBuiltInNS = disableBuiltInNS; + this.disableWebRtcAGCAndHPF = disableWebRtcAGCAndHPF; + this.enableRtcEventLog = enableRtcEventLog; + this.dataChannelParameters = dataChannelParameters; + } + } + + /** + * Peer connection events. + */ + public interface PeerConnectionEvents { + /** + * Callback fired once local SDP is created and set. + */ + void onLocalDescription(final SessionDescription sdp); + + /** + * Callback fired once local Ice candidate is generated. + */ + void onIceCandidate(final IceCandidate candidate); + + /** + * Callback fired once local ICE candidates are removed. + */ + void onIceCandidatesRemoved(final IceCandidate[] candidates); + + /** + * Callback fired once connection is established (IceConnectionState is + * CONNECTED). + */ + void onIceConnected(); + + /** + * Callback fired once connection is disconnected (IceConnectionState is + * DISCONNECTED). + */ + void onIceDisconnected(); + + /** + * Callback fired once DTLS connection is established (PeerConnectionState + * is CONNECTED). + */ + void onConnected(); + + /** + * Callback fired once DTLS connection is disconnected (PeerConnectionState + * is DISCONNECTED). + */ + void onDisconnected(); + + /** + * Callback fired once peer connection is closed. + */ + void onPeerConnectionClosed(); + + /** + * Callback fired once peer connection statistics is ready. + */ + void onPeerConnectionStatsReady(final RTCStatsReport report); + + /** + * Callback fired once peer connection error happened. + */ + void onPeerConnectionError(final String description); + } + + /** + * Create a PeerConnectionClient with the specified parameters. PeerConnectionClient takes + * ownership of `eglBase`. + */ + public PeerConnectionClient(Context appContext, EglBase eglBase, + PeerConnectionParameters peerConnectionParameters, PeerConnectionEvents events) { + this.rootEglBase = eglBase; + this.appContext = appContext; + this.events = events; + this.peerConnectionParameters = peerConnectionParameters; + this.dataChannelEnabled = peerConnectionParameters.dataChannelParameters != null; + + Log.d(TAG, "Preferred video codec: " + getSdpVideoCodecName(peerConnectionParameters)); + + final String fieldTrials = getFieldTrials(peerConnectionParameters); + executor.execute(() -> { + Log.d(TAG, "Initialize WebRTC. Field trials: " + fieldTrials); + PeerConnectionFactory.initialize( + PeerConnectionFactory.InitializationOptions.builder(appContext) + .setFieldTrials(fieldTrials) + .setEnableInternalTracer(true) + .createInitializationOptions()); + }); + } + + /** + * This function should only be called once. + */ + public void createPeerConnectionFactory(PeerConnectionFactory.Options options) { + if (factory != null) { + throw new IllegalStateException("PeerConnectionFactory has already been constructed"); + } + executor.execute(() -> createPeerConnectionFactoryInternal(options)); + } + + public void createPeerConnection(final VideoSink localRender, final VideoSink remoteSink, + final VideoCapturer videoCapturer, final SignalingParameters signalingParameters) { + if (peerConnectionParameters.videoCallEnabled && videoCapturer == null) { + Log.w(TAG, "Video call enabled but no video capturer provided."); + } + createPeerConnection( + localRender, Collections.singletonList(remoteSink), videoCapturer, signalingParameters); + } + + public void createPeerConnection(final VideoSink localRender, final List<VideoSink> remoteSinks, + final VideoCapturer videoCapturer, final SignalingParameters signalingParameters) { + if (peerConnectionParameters == null) { + Log.e(TAG, "Creating peer connection without initializing factory."); + return; + } + this.localRender = localRender; + this.remoteSinks = remoteSinks; + this.videoCapturer = videoCapturer; + this.signalingParameters = signalingParameters; + executor.execute(() -> { + try { + createMediaConstraintsInternal(); + createPeerConnectionInternal(); + maybeCreateAndStartRtcEventLog(); + } catch (Exception e) { + reportError("Failed to create peer connection: " + e.getMessage()); + throw e; + } + }); + } + + public void close() { + executor.execute(this ::closeInternal); + } + + private boolean isVideoCallEnabled() { + return peerConnectionParameters.videoCallEnabled && videoCapturer != null; + } + + private void createPeerConnectionFactoryInternal(PeerConnectionFactory.Options options) { + isError = false; + + if (peerConnectionParameters.tracing) { + PeerConnectionFactory.startInternalTracingCapture( + Environment.getExternalStorageDirectory().getAbsolutePath() + File.separator + + "webrtc-trace.txt"); + } + + // Check if ISAC is used by default. + preferIsac = peerConnectionParameters.audioCodec != null + && peerConnectionParameters.audioCodec.equals(AUDIO_CODEC_ISAC); + + // It is possible to save a copy in raw PCM format on a file by checking + // the "Save input audio to file" checkbox in the Settings UI. A callback + // interface is set when this flag is enabled. As a result, a copy of recorded + // audio samples are provided to this client directly from the native audio + // layer in Java. + if (peerConnectionParameters.saveInputAudioToFile) { + if (!peerConnectionParameters.useOpenSLES) { + Log.d(TAG, "Enable recording of microphone input audio to file"); + saveRecordedAudioToFile = new RecordedAudioToFileController(executor); + } else { + // TODO(henrika): ensure that the UI reflects that if OpenSL ES is selected, + // then the "Save inut audio to file" option shall be grayed out. + Log.e(TAG, "Recording of input audio is not supported for OpenSL ES"); + } + } + + final AudioDeviceModule adm = createJavaAudioDevice(); + + // Create peer connection factory. + if (options != null) { + Log.d(TAG, "Factory networkIgnoreMask option: " + options.networkIgnoreMask); + } + final boolean enableH264HighProfile = + VIDEO_CODEC_H264_HIGH.equals(peerConnectionParameters.videoCodec); + final VideoEncoderFactory encoderFactory; + final VideoDecoderFactory decoderFactory; + + if (peerConnectionParameters.videoCodecHwAcceleration) { + encoderFactory = new DefaultVideoEncoderFactory( + rootEglBase.getEglBaseContext(), true /* enableIntelVp8Encoder */, enableH264HighProfile); + decoderFactory = new DefaultVideoDecoderFactory(rootEglBase.getEglBaseContext()); + } else { + encoderFactory = new SoftwareVideoEncoderFactory(); + decoderFactory = new SoftwareVideoDecoderFactory(); + } + + // Disable encryption for loopback calls. + if (peerConnectionParameters.loopback) { + options.disableEncryption = true; + } + factory = PeerConnectionFactory.builder() + .setOptions(options) + .setAudioDeviceModule(adm) + .setVideoEncoderFactory(encoderFactory) + .setVideoDecoderFactory(decoderFactory) + .createPeerConnectionFactory(); + Log.d(TAG, "Peer connection factory created."); + adm.release(); + } + + AudioDeviceModule createJavaAudioDevice() { + // Enable/disable OpenSL ES playback. + if (!peerConnectionParameters.useOpenSLES) { + Log.w(TAG, "External OpenSLES ADM not implemented yet."); + // TODO(magjed): Add support for external OpenSLES ADM. + } + + // Set audio record error callbacks. + AudioRecordErrorCallback audioRecordErrorCallback = new AudioRecordErrorCallback() { + @Override + public void onWebRtcAudioRecordInitError(String errorMessage) { + Log.e(TAG, "onWebRtcAudioRecordInitError: " + errorMessage); + reportError(errorMessage); + } + + @Override + public void onWebRtcAudioRecordStartError( + JavaAudioDeviceModule.AudioRecordStartErrorCode errorCode, String errorMessage) { + Log.e(TAG, "onWebRtcAudioRecordStartError: " + errorCode + ". " + errorMessage); + reportError(errorMessage); + } + + @Override + public void onWebRtcAudioRecordError(String errorMessage) { + Log.e(TAG, "onWebRtcAudioRecordError: " + errorMessage); + reportError(errorMessage); + } + }; + + AudioTrackErrorCallback audioTrackErrorCallback = new AudioTrackErrorCallback() { + @Override + public void onWebRtcAudioTrackInitError(String errorMessage) { + Log.e(TAG, "onWebRtcAudioTrackInitError: " + errorMessage); + reportError(errorMessage); + } + + @Override + public void onWebRtcAudioTrackStartError( + JavaAudioDeviceModule.AudioTrackStartErrorCode errorCode, String errorMessage) { + Log.e(TAG, "onWebRtcAudioTrackStartError: " + errorCode + ". " + errorMessage); + reportError(errorMessage); + } + + @Override + public void onWebRtcAudioTrackError(String errorMessage) { + Log.e(TAG, "onWebRtcAudioTrackError: " + errorMessage); + reportError(errorMessage); + } + }; + + // Set audio record state callbacks. + AudioRecordStateCallback audioRecordStateCallback = new AudioRecordStateCallback() { + @Override + public void onWebRtcAudioRecordStart() { + Log.i(TAG, "Audio recording starts"); + } + + @Override + public void onWebRtcAudioRecordStop() { + Log.i(TAG, "Audio recording stops"); + } + }; + + // Set audio track state callbacks. + AudioTrackStateCallback audioTrackStateCallback = new AudioTrackStateCallback() { + @Override + public void onWebRtcAudioTrackStart() { + Log.i(TAG, "Audio playout starts"); + } + + @Override + public void onWebRtcAudioTrackStop() { + Log.i(TAG, "Audio playout stops"); + } + }; + + return JavaAudioDeviceModule.builder(appContext) + .setSamplesReadyCallback(saveRecordedAudioToFile) + .setUseHardwareAcousticEchoCanceler(!peerConnectionParameters.disableBuiltInAEC) + .setUseHardwareNoiseSuppressor(!peerConnectionParameters.disableBuiltInNS) + .setAudioRecordErrorCallback(audioRecordErrorCallback) + .setAudioTrackErrorCallback(audioTrackErrorCallback) + .setAudioRecordStateCallback(audioRecordStateCallback) + .setAudioTrackStateCallback(audioTrackStateCallback) + .createAudioDeviceModule(); + } + + private void createMediaConstraintsInternal() { + // Create video constraints if video call is enabled. + if (isVideoCallEnabled()) { + videoWidth = peerConnectionParameters.videoWidth; + videoHeight = peerConnectionParameters.videoHeight; + videoFps = peerConnectionParameters.videoFps; + + // If video resolution is not specified, default to HD. + if (videoWidth == 0 || videoHeight == 0) { + videoWidth = HD_VIDEO_WIDTH; + videoHeight = HD_VIDEO_HEIGHT; + } + + // If fps is not specified, default to 30. + if (videoFps == 0) { + videoFps = 30; + } + Logging.d(TAG, "Capturing format: " + videoWidth + "x" + videoHeight + "@" + videoFps); + } + + // Create audio constraints. + audioConstraints = new MediaConstraints(); + // added for audio performance measurements + if (peerConnectionParameters.noAudioProcessing) { + Log.d(TAG, "Disabling audio processing"); + audioConstraints.mandatory.add( + new MediaConstraints.KeyValuePair(AUDIO_ECHO_CANCELLATION_CONSTRAINT, "false")); + audioConstraints.mandatory.add( + new MediaConstraints.KeyValuePair(AUDIO_AUTO_GAIN_CONTROL_CONSTRAINT, "false")); + audioConstraints.mandatory.add( + new MediaConstraints.KeyValuePair(AUDIO_HIGH_PASS_FILTER_CONSTRAINT, "false")); + audioConstraints.mandatory.add( + new MediaConstraints.KeyValuePair(AUDIO_NOISE_SUPPRESSION_CONSTRAINT, "false")); + } + // Create SDP constraints. + sdpMediaConstraints = new MediaConstraints(); + sdpMediaConstraints.mandatory.add( + new MediaConstraints.KeyValuePair("OfferToReceiveAudio", "true")); + sdpMediaConstraints.mandatory.add(new MediaConstraints.KeyValuePair( + "OfferToReceiveVideo", Boolean.toString(isVideoCallEnabled()))); + } + + private void createPeerConnectionInternal() { + if (factory == null || isError) { + Log.e(TAG, "Peerconnection factory is not created"); + return; + } + Log.d(TAG, "Create peer connection."); + + queuedRemoteCandidates = new ArrayList<>(); + + PeerConnection.RTCConfiguration rtcConfig = + new PeerConnection.RTCConfiguration(signalingParameters.iceServers); + // TCP candidates are only useful when connecting to a server that supports + // ICE-TCP. + rtcConfig.tcpCandidatePolicy = PeerConnection.TcpCandidatePolicy.DISABLED; + rtcConfig.bundlePolicy = PeerConnection.BundlePolicy.MAXBUNDLE; + rtcConfig.rtcpMuxPolicy = PeerConnection.RtcpMuxPolicy.REQUIRE; + rtcConfig.continualGatheringPolicy = PeerConnection.ContinualGatheringPolicy.GATHER_CONTINUALLY; + // Use ECDSA encryption. + rtcConfig.keyType = PeerConnection.KeyType.ECDSA; + rtcConfig.sdpSemantics = PeerConnection.SdpSemantics.UNIFIED_PLAN; + + peerConnection = factory.createPeerConnection(rtcConfig, pcObserver); + + if (dataChannelEnabled) { + DataChannel.Init init = new DataChannel.Init(); + init.ordered = peerConnectionParameters.dataChannelParameters.ordered; + init.negotiated = peerConnectionParameters.dataChannelParameters.negotiated; + init.maxRetransmits = peerConnectionParameters.dataChannelParameters.maxRetransmits; + init.maxRetransmitTimeMs = peerConnectionParameters.dataChannelParameters.maxRetransmitTimeMs; + init.id = peerConnectionParameters.dataChannelParameters.id; + init.protocol = peerConnectionParameters.dataChannelParameters.protocol; + dataChannel = peerConnection.createDataChannel("ApprtcDemo data", init); + } + isInitiator = false; + + // Set INFO libjingle logging. + // NOTE: this _must_ happen while `factory` is alive! + Logging.enableLogToDebugOutput(Logging.Severity.LS_INFO); + + List<String> mediaStreamLabels = Collections.singletonList("ARDAMS"); + if (isVideoCallEnabled()) { + peerConnection.addTrack(createVideoTrack(videoCapturer), mediaStreamLabels); + // We can add the renderers right away because we don't need to wait for an + // answer to get the remote track. + remoteVideoTrack = getRemoteVideoTrack(); + remoteVideoTrack.setEnabled(renderVideo); + for (VideoSink remoteSink : remoteSinks) { + remoteVideoTrack.addSink(remoteSink); + } + } + peerConnection.addTrack(createAudioTrack(), mediaStreamLabels); + if (isVideoCallEnabled()) { + findVideoSender(); + } + + if (peerConnectionParameters.aecDump) { + try { + ParcelFileDescriptor aecDumpFileDescriptor = + ParcelFileDescriptor.open(new File(Environment.getExternalStorageDirectory().getPath() + + File.separator + "Download/audio.aecdump"), + ParcelFileDescriptor.MODE_READ_WRITE | ParcelFileDescriptor.MODE_CREATE + | ParcelFileDescriptor.MODE_TRUNCATE); + factory.startAecDump(aecDumpFileDescriptor.detachFd(), -1); + } catch (IOException e) { + Log.e(TAG, "Can not open aecdump file", e); + } + } + + if (saveRecordedAudioToFile != null) { + if (saveRecordedAudioToFile.start()) { + Log.d(TAG, "Recording input audio to file is activated"); + } + } + Log.d(TAG, "Peer connection created."); + } + + private File createRtcEventLogOutputFile() { + DateFormat dateFormat = new SimpleDateFormat("yyyyMMdd_hhmm_ss", Locale.getDefault()); + Date date = new Date(); + final String outputFileName = "event_log_" + dateFormat.format(date) + ".log"; + return new File( + appContext.getDir(RTCEVENTLOG_OUTPUT_DIR_NAME, Context.MODE_PRIVATE), outputFileName); + } + + private void maybeCreateAndStartRtcEventLog() { + if (appContext == null || peerConnection == null) { + return; + } + if (!peerConnectionParameters.enableRtcEventLog) { + Log.d(TAG, "RtcEventLog is disabled."); + return; + } + rtcEventLog = new RtcEventLog(peerConnection); + rtcEventLog.start(createRtcEventLogOutputFile()); + } + + private void closeInternal() { + if (factory != null && peerConnectionParameters.aecDump) { + factory.stopAecDump(); + } + Log.d(TAG, "Closing peer connection."); + statsTimer.cancel(); + if (dataChannel != null) { + dataChannel.dispose(); + dataChannel = null; + } + if (rtcEventLog != null) { + // RtcEventLog should stop before the peer connection is disposed. + rtcEventLog.stop(); + rtcEventLog = null; + } + if (peerConnection != null) { + peerConnection.dispose(); + peerConnection = null; + } + Log.d(TAG, "Closing audio source."); + if (audioSource != null) { + audioSource.dispose(); + audioSource = null; + } + Log.d(TAG, "Stopping capture."); + if (videoCapturer != null) { + try { + videoCapturer.stopCapture(); + } catch (InterruptedException e) { + throw new RuntimeException(e); + } + videoCapturerStopped = true; + videoCapturer.dispose(); + videoCapturer = null; + } + Log.d(TAG, "Closing video source."); + if (videoSource != null) { + videoSource.dispose(); + videoSource = null; + } + if (surfaceTextureHelper != null) { + surfaceTextureHelper.dispose(); + surfaceTextureHelper = null; + } + if (saveRecordedAudioToFile != null) { + Log.d(TAG, "Closing audio file for recorded input audio."); + saveRecordedAudioToFile.stop(); + saveRecordedAudioToFile = null; + } + localRender = null; + remoteSinks = null; + Log.d(TAG, "Closing peer connection factory."); + if (factory != null) { + factory.dispose(); + factory = null; + } + rootEglBase.release(); + Log.d(TAG, "Closing peer connection done."); + events.onPeerConnectionClosed(); + PeerConnectionFactory.stopInternalTracingCapture(); + PeerConnectionFactory.shutdownInternalTracer(); + } + + public boolean isHDVideo() { + return isVideoCallEnabled() && videoWidth * videoHeight >= 1280 * 720; + } + + private void getStats() { + if (peerConnection == null || isError) { + return; + } + peerConnection.getStats(new RTCStatsCollectorCallback() { + @Override + public void onStatsDelivered(RTCStatsReport report) { + events.onPeerConnectionStatsReady(report); + } + }); + } + + public void enableStatsEvents(boolean enable, int periodMs) { + if (enable) { + try { + statsTimer.schedule(new TimerTask() { + @Override + public void run() { + executor.execute(() -> getStats()); + } + }, 0, periodMs); + } catch (Exception e) { + Log.e(TAG, "Can not schedule statistics timer", e); + } + } else { + statsTimer.cancel(); + } + } + + public void setAudioEnabled(final boolean enable) { + executor.execute(() -> { + enableAudio = enable; + if (localAudioTrack != null) { + localAudioTrack.setEnabled(enableAudio); + } + }); + } + + public void setVideoEnabled(final boolean enable) { + executor.execute(() -> { + renderVideo = enable; + if (localVideoTrack != null) { + localVideoTrack.setEnabled(renderVideo); + } + if (remoteVideoTrack != null) { + remoteVideoTrack.setEnabled(renderVideo); + } + }); + } + + public void createOffer() { + executor.execute(() -> { + if (peerConnection != null && !isError) { + Log.d(TAG, "PC Create OFFER"); + isInitiator = true; + peerConnection.createOffer(sdpObserver, sdpMediaConstraints); + } + }); + } + + public void createAnswer() { + executor.execute(() -> { + if (peerConnection != null && !isError) { + Log.d(TAG, "PC create ANSWER"); + isInitiator = false; + peerConnection.createAnswer(sdpObserver, sdpMediaConstraints); + } + }); + } + + public void addRemoteIceCandidate(final IceCandidate candidate) { + executor.execute(() -> { + if (peerConnection != null && !isError) { + if (queuedRemoteCandidates != null) { + queuedRemoteCandidates.add(candidate); + } else { + peerConnection.addIceCandidate(candidate, new AddIceObserver() { + @Override + public void onAddSuccess() { + Log.d(TAG, "Candidate " + candidate + " successfully added."); + } + @Override + public void onAddFailure(String error) { + Log.d(TAG, "Candidate " + candidate + " addition failed: " + error); + } + }); + } + } + }); + } + + public void removeRemoteIceCandidates(final IceCandidate[] candidates) { + executor.execute(() -> { + if (peerConnection == null || isError) { + return; + } + // Drain the queued remote candidates if there is any so that + // they are processed in the proper order. + drainCandidates(); + peerConnection.removeIceCandidates(candidates); + }); + } + + public void setRemoteDescription(final SessionDescription desc) { + executor.execute(() -> { + if (peerConnection == null || isError) { + return; + } + String sdp = desc.description; + if (preferIsac) { + sdp = preferCodec(sdp, AUDIO_CODEC_ISAC, true); + } + if (isVideoCallEnabled()) { + sdp = preferCodec(sdp, getSdpVideoCodecName(peerConnectionParameters), false); + } + if (peerConnectionParameters.audioStartBitrate > 0) { + sdp = setStartBitrate( + AUDIO_CODEC_OPUS, false, sdp, peerConnectionParameters.audioStartBitrate); + } + Log.d(TAG, "Set remote SDP."); + SessionDescription sdpRemote = new SessionDescription(desc.type, sdp); + peerConnection.setRemoteDescription(sdpObserver, sdpRemote); + }); + } + + public void stopVideoSource() { + executor.execute(() -> { + if (videoCapturer != null && !videoCapturerStopped) { + Log.d(TAG, "Stop video source."); + try { + videoCapturer.stopCapture(); + } catch (InterruptedException e) { + } + videoCapturerStopped = true; + } + }); + } + + public void startVideoSource() { + executor.execute(() -> { + if (videoCapturer != null && videoCapturerStopped) { + Log.d(TAG, "Restart video source."); + videoCapturer.startCapture(videoWidth, videoHeight, videoFps); + videoCapturerStopped = false; + } + }); + } + + public void setVideoMaxBitrate(@Nullable final Integer maxBitrateKbps) { + executor.execute(() -> { + if (peerConnection == null || localVideoSender == null || isError) { + return; + } + Log.d(TAG, "Requested max video bitrate: " + maxBitrateKbps); + if (localVideoSender == null) { + Log.w(TAG, "Sender is not ready."); + return; + } + + RtpParameters parameters = localVideoSender.getParameters(); + if (parameters.encodings.size() == 0) { + Log.w(TAG, "RtpParameters are not ready."); + return; + } + + for (RtpParameters.Encoding encoding : parameters.encodings) { + // Null value means no limit. + encoding.maxBitrateBps = maxBitrateKbps == null ? null : maxBitrateKbps * BPS_IN_KBPS; + } + if (!localVideoSender.setParameters(parameters)) { + Log.e(TAG, "RtpSender.setParameters failed."); + } + Log.d(TAG, "Configured max video bitrate to: " + maxBitrateKbps); + }); + } + + private void reportError(final String errorMessage) { + Log.e(TAG, "Peerconnection error: " + errorMessage); + executor.execute(() -> { + if (!isError) { + events.onPeerConnectionError(errorMessage); + isError = true; + } + }); + } + + @Nullable + private AudioTrack createAudioTrack() { + audioSource = factory.createAudioSource(audioConstraints); + localAudioTrack = factory.createAudioTrack(AUDIO_TRACK_ID, audioSource); + localAudioTrack.setEnabled(enableAudio); + return localAudioTrack; + } + + @Nullable + private VideoTrack createVideoTrack(VideoCapturer capturer) { + surfaceTextureHelper = + SurfaceTextureHelper.create("CaptureThread", rootEglBase.getEglBaseContext()); + videoSource = factory.createVideoSource(capturer.isScreencast()); + capturer.initialize(surfaceTextureHelper, appContext, videoSource.getCapturerObserver()); + capturer.startCapture(videoWidth, videoHeight, videoFps); + + localVideoTrack = factory.createVideoTrack(VIDEO_TRACK_ID, videoSource); + localVideoTrack.setEnabled(renderVideo); + localVideoTrack.addSink(localRender); + return localVideoTrack; + } + + private void findVideoSender() { + for (RtpSender sender : peerConnection.getSenders()) { + if (sender.track() != null) { + String trackType = sender.track().kind(); + if (trackType.equals(VIDEO_TRACK_TYPE)) { + Log.d(TAG, "Found video sender."); + localVideoSender = sender; + } + } + } + } + + // Returns the remote VideoTrack, assuming there is only one. + private @Nullable VideoTrack getRemoteVideoTrack() { + for (RtpTransceiver transceiver : peerConnection.getTransceivers()) { + MediaStreamTrack track = transceiver.getReceiver().track(); + if (track instanceof VideoTrack) { + return (VideoTrack) track; + } + } + return null; + } + + private static String getSdpVideoCodecName(PeerConnectionParameters parameters) { + switch (parameters.videoCodec) { + case VIDEO_CODEC_VP8: + return VIDEO_CODEC_VP8; + case VIDEO_CODEC_VP9: + return VIDEO_CODEC_VP9; + case VIDEO_CODEC_AV1: + return VIDEO_CODEC_AV1; + case VIDEO_CODEC_H264_HIGH: + case VIDEO_CODEC_H264_BASELINE: + return VIDEO_CODEC_H264; + default: + return VIDEO_CODEC_VP8; + } + } + + private static String getFieldTrials(PeerConnectionParameters peerConnectionParameters) { + String fieldTrials = ""; + if (peerConnectionParameters.videoFlexfecEnabled) { + fieldTrials += VIDEO_FLEXFEC_FIELDTRIAL; + Log.d(TAG, "Enable FlexFEC field trial."); + } + fieldTrials += VIDEO_VP8_INTEL_HW_ENCODER_FIELDTRIAL; + if (peerConnectionParameters.disableWebRtcAGCAndHPF) { + fieldTrials += DISABLE_WEBRTC_AGC_FIELDTRIAL; + Log.d(TAG, "Disable WebRTC AGC field trial."); + } + return fieldTrials; + } + + @SuppressWarnings("StringSplitter") + private static String setStartBitrate( + String codec, boolean isVideoCodec, String sdp, int bitrateKbps) { + String[] lines = sdp.split("\r\n"); + int rtpmapLineIndex = -1; + boolean sdpFormatUpdated = false; + String codecRtpMap = null; + // Search for codec rtpmap in format + // a=rtpmap:<payload type> <encoding name>/<clock rate> [/<encoding parameters>] + String regex = "^a=rtpmap:(\\d+) " + codec + "(/\\d+)+[\r]?$"; + Pattern codecPattern = Pattern.compile(regex); + for (int i = 0; i < lines.length; i++) { + Matcher codecMatcher = codecPattern.matcher(lines[i]); + if (codecMatcher.matches()) { + codecRtpMap = codecMatcher.group(1); + rtpmapLineIndex = i; + break; + } + } + if (codecRtpMap == null) { + Log.w(TAG, "No rtpmap for " + codec + " codec"); + return sdp; + } + Log.d(TAG, "Found " + codec + " rtpmap " + codecRtpMap + " at " + lines[rtpmapLineIndex]); + + // Check if a=fmtp string already exist in remote SDP for this codec and + // update it with new bitrate parameter. + regex = "^a=fmtp:" + codecRtpMap + " \\w+=\\d+.*[\r]?$"; + codecPattern = Pattern.compile(regex); + for (int i = 0; i < lines.length; i++) { + Matcher codecMatcher = codecPattern.matcher(lines[i]); + if (codecMatcher.matches()) { + Log.d(TAG, "Found " + codec + " " + lines[i]); + if (isVideoCodec) { + lines[i] += "; " + VIDEO_CODEC_PARAM_START_BITRATE + "=" + bitrateKbps; + } else { + lines[i] += "; " + AUDIO_CODEC_PARAM_BITRATE + "=" + (bitrateKbps * 1000); + } + Log.d(TAG, "Update remote SDP line: " + lines[i]); + sdpFormatUpdated = true; + break; + } + } + + StringBuilder newSdpDescription = new StringBuilder(); + for (int i = 0; i < lines.length; i++) { + newSdpDescription.append(lines[i]).append("\r\n"); + // Append new a=fmtp line if no such line exist for a codec. + if (!sdpFormatUpdated && i == rtpmapLineIndex) { + String bitrateSet; + if (isVideoCodec) { + bitrateSet = + "a=fmtp:" + codecRtpMap + " " + VIDEO_CODEC_PARAM_START_BITRATE + "=" + bitrateKbps; + } else { + bitrateSet = "a=fmtp:" + codecRtpMap + " " + AUDIO_CODEC_PARAM_BITRATE + "=" + + (bitrateKbps * 1000); + } + Log.d(TAG, "Add remote SDP line: " + bitrateSet); + newSdpDescription.append(bitrateSet).append("\r\n"); + } + } + return newSdpDescription.toString(); + } + + /** Returns the line number containing "m=audio|video", or -1 if no such line exists. */ + private static int findMediaDescriptionLine(boolean isAudio, String[] sdpLines) { + final String mediaDescription = isAudio ? "m=audio " : "m=video "; + for (int i = 0; i < sdpLines.length; ++i) { + if (sdpLines[i].startsWith(mediaDescription)) { + return i; + } + } + return -1; + } + + private static String joinString( + Iterable<? extends CharSequence> s, String delimiter, boolean delimiterAtEnd) { + Iterator<? extends CharSequence> iter = s.iterator(); + if (!iter.hasNext()) { + return ""; + } + StringBuilder buffer = new StringBuilder(iter.next()); + while (iter.hasNext()) { + buffer.append(delimiter).append(iter.next()); + } + if (delimiterAtEnd) { + buffer.append(delimiter); + } + return buffer.toString(); + } + + private static @Nullable String movePayloadTypesToFront( + List<String> preferredPayloadTypes, String mLine) { + // The format of the media description line should be: m=<media> <port> <proto> <fmt> ... + final List<String> origLineParts = Arrays.asList(mLine.split(" ")); + if (origLineParts.size() <= 3) { + Log.e(TAG, "Wrong SDP media description format: " + mLine); + return null; + } + final List<String> header = origLineParts.subList(0, 3); + final List<String> unpreferredPayloadTypes = + new ArrayList<>(origLineParts.subList(3, origLineParts.size())); + unpreferredPayloadTypes.removeAll(preferredPayloadTypes); + // Reconstruct the line with `preferredPayloadTypes` moved to the beginning of the payload + // types. + final List<String> newLineParts = new ArrayList<>(); + newLineParts.addAll(header); + newLineParts.addAll(preferredPayloadTypes); + newLineParts.addAll(unpreferredPayloadTypes); + return joinString(newLineParts, " ", false /* delimiterAtEnd */); + } + + private static String preferCodec(String sdp, String codec, boolean isAudio) { + final String[] lines = sdp.split("\r\n"); + final int mLineIndex = findMediaDescriptionLine(isAudio, lines); + if (mLineIndex == -1) { + Log.w(TAG, "No mediaDescription line, so can't prefer " + codec); + return sdp; + } + // A list with all the payload types with name `codec`. The payload types are integers in the + // range 96-127, but they are stored as strings here. + final List<String> codecPayloadTypes = new ArrayList<>(); + // a=rtpmap:<payload type> <encoding name>/<clock rate> [/<encoding parameters>] + final Pattern codecPattern = Pattern.compile("^a=rtpmap:(\\d+) " + codec + "(/\\d+)+[\r]?$"); + for (String line : lines) { + Matcher codecMatcher = codecPattern.matcher(line); + if (codecMatcher.matches()) { + codecPayloadTypes.add(codecMatcher.group(1)); + } + } + if (codecPayloadTypes.isEmpty()) { + Log.w(TAG, "No payload types with name " + codec); + return sdp; + } + + final String newMLine = movePayloadTypesToFront(codecPayloadTypes, lines[mLineIndex]); + if (newMLine == null) { + return sdp; + } + Log.d(TAG, "Change media description from: " + lines[mLineIndex] + " to " + newMLine); + lines[mLineIndex] = newMLine; + return joinString(Arrays.asList(lines), "\r\n", true /* delimiterAtEnd */); + } + + private void drainCandidates() { + if (queuedRemoteCandidates != null) { + Log.d(TAG, "Add " + queuedRemoteCandidates.size() + " remote candidates"); + for (IceCandidate candidate : queuedRemoteCandidates) { + peerConnection.addIceCandidate(candidate, new AddIceObserver() { + @Override + public void onAddSuccess() { + Log.d(TAG, "Candidate " + candidate + " successfully added."); + } + @Override + public void onAddFailure(String error) { + Log.d(TAG, "Candidate " + candidate + " addition failed: " + error); + } + }); + } + queuedRemoteCandidates = null; + } + } + + private void switchCameraInternal() { + if (videoCapturer instanceof CameraVideoCapturer) { + if (!isVideoCallEnabled() || isError) { + Log.e(TAG, + "Failed to switch camera. Video: " + isVideoCallEnabled() + ". Error : " + isError); + return; // No video is sent or only one camera is available or error happened. + } + Log.d(TAG, "Switch camera"); + CameraVideoCapturer cameraVideoCapturer = (CameraVideoCapturer) videoCapturer; + cameraVideoCapturer.switchCamera(null); + } else { + Log.d(TAG, "Will not switch camera, video caputurer is not a camera"); + } + } + + public void switchCamera() { + executor.execute(this ::switchCameraInternal); + } + + public void changeCaptureFormat(final int width, final int height, final int framerate) { + executor.execute(() -> changeCaptureFormatInternal(width, height, framerate)); + } + + private void changeCaptureFormatInternal(int width, int height, int framerate) { + if (!isVideoCallEnabled() || isError || videoCapturer == null) { + Log.e(TAG, + "Failed to change capture format. Video: " + isVideoCallEnabled() + + ". Error : " + isError); + return; + } + Log.d(TAG, "changeCaptureFormat: " + width + "x" + height + "@" + framerate); + videoSource.adaptOutputFormat(width, height, framerate); + } + + // Implementation detail: observe ICE & stream changes and react accordingly. + private class PCObserver implements PeerConnection.Observer { + @Override + public void onIceCandidate(final IceCandidate candidate) { + executor.execute(() -> events.onIceCandidate(candidate)); + } + + @Override + public void onIceCandidateError(final IceCandidateErrorEvent event) { + Log.d(TAG, + "IceCandidateError address: " + event.address + ", port: " + event.port + ", url: " + + event.url + ", errorCode: " + event.errorCode + ", errorText: " + event.errorText); + } + + @Override + public void onIceCandidatesRemoved(final IceCandidate[] candidates) { + executor.execute(() -> events.onIceCandidatesRemoved(candidates)); + } + + @Override + public void onSignalingChange(PeerConnection.SignalingState newState) { + Log.d(TAG, "SignalingState: " + newState); + } + + @Override + public void onIceConnectionChange(final PeerConnection.IceConnectionState newState) { + executor.execute(() -> { + Log.d(TAG, "IceConnectionState: " + newState); + if (newState == IceConnectionState.CONNECTED) { + events.onIceConnected(); + } else if (newState == IceConnectionState.DISCONNECTED) { + events.onIceDisconnected(); + } else if (newState == IceConnectionState.FAILED) { + reportError("ICE connection failed."); + } + }); + } + + @Override + public void onConnectionChange(final PeerConnection.PeerConnectionState newState) { + executor.execute(() -> { + Log.d(TAG, "PeerConnectionState: " + newState); + if (newState == PeerConnectionState.CONNECTED) { + events.onConnected(); + } else if (newState == PeerConnectionState.DISCONNECTED) { + events.onDisconnected(); + } else if (newState == PeerConnectionState.FAILED) { + reportError("DTLS connection failed."); + } + }); + } + + @Override + public void onIceGatheringChange(PeerConnection.IceGatheringState newState) { + Log.d(TAG, "IceGatheringState: " + newState); + } + + @Override + public void onIceConnectionReceivingChange(boolean receiving) { + Log.d(TAG, "IceConnectionReceiving changed to " + receiving); + } + + @Override + public void onSelectedCandidatePairChanged(CandidatePairChangeEvent event) { + Log.d(TAG, "Selected candidate pair changed because: " + event); + } + + @Override + public void onAddStream(final MediaStream stream) {} + + @Override + public void onRemoveStream(final MediaStream stream) {} + + @Override + public void onDataChannel(final DataChannel dc) { + Log.d(TAG, "New Data channel " + dc.label()); + + if (!dataChannelEnabled) + return; + + dc.registerObserver(new DataChannel.Observer() { + @Override + public void onBufferedAmountChange(long previousAmount) { + Log.d(TAG, "Data channel buffered amount changed: " + dc.label() + ": " + dc.state()); + } + + @Override + public void onStateChange() { + Log.d(TAG, "Data channel state changed: " + dc.label() + ": " + dc.state()); + } + + @Override + public void onMessage(final DataChannel.Buffer buffer) { + if (buffer.binary) { + Log.d(TAG, "Received binary msg over " + dc); + return; + } + ByteBuffer data = buffer.data; + final byte[] bytes = new byte[data.capacity()]; + data.get(bytes); + String strData = new String(bytes, Charset.forName("UTF-8")); + Log.d(TAG, "Got msg: " + strData + " over " + dc); + } + }); + } + + @Override + public void onRenegotiationNeeded() { + // No need to do anything; AppRTC follows a pre-agreed-upon + // signaling/negotiation protocol. + } + + @Override + public void onAddTrack(final RtpReceiver receiver, final MediaStream[] mediaStreams) {} + + @Override + public void onRemoveTrack(final RtpReceiver receiver) {} + } + + // Implementation detail: handle offer creation/signaling and answer setting, + // as well as adding remote ICE candidates once the answer SDP is set. + private class SDPObserver implements SdpObserver { + @Override + public void onCreateSuccess(final SessionDescription desc) { + if (localDescription != null) { + reportError("Multiple SDP create."); + return; + } + String sdp = desc.description; + if (preferIsac) { + sdp = preferCodec(sdp, AUDIO_CODEC_ISAC, true); + } + if (isVideoCallEnabled()) { + sdp = preferCodec(sdp, getSdpVideoCodecName(peerConnectionParameters), false); + } + final SessionDescription newDesc = new SessionDescription(desc.type, sdp); + localDescription = newDesc; + executor.execute(() -> { + if (peerConnection != null && !isError) { + Log.d(TAG, "Set local SDP from " + desc.type); + peerConnection.setLocalDescription(sdpObserver, newDesc); + } + }); + } + + @Override + public void onSetSuccess() { + executor.execute(() -> { + if (peerConnection == null || isError) { + return; + } + if (isInitiator) { + // For offering peer connection we first create offer and set + // local SDP, then after receiving answer set remote SDP. + if (peerConnection.getRemoteDescription() == null) { + // We've just set our local SDP so time to send it. + Log.d(TAG, "Local SDP set succesfully"); + events.onLocalDescription(localDescription); + } else { + // We've just set remote description, so drain remote + // and send local ICE candidates. + Log.d(TAG, "Remote SDP set succesfully"); + drainCandidates(); + } + } else { + // For answering peer connection we set remote SDP and then + // create answer and set local SDP. + if (peerConnection.getLocalDescription() != null) { + // We've just set our local SDP so time to send it, drain + // remote and send local ICE candidates. + Log.d(TAG, "Local SDP set succesfully"); + events.onLocalDescription(localDescription); + drainCandidates(); + } else { + // We've just set remote SDP - do nothing for now - + // answer will be created soon. + Log.d(TAG, "Remote SDP set succesfully"); + } + } + }); + } + + @Override + public void onCreateFailure(final String error) { + reportError("createSDP error: " + error); + } + + @Override + public void onSetFailure(final String error) { + reportError("setSDP error: " + error); + } + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RecordedAudioToFileController.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RecordedAudioToFileController.java new file mode 100644 index 0000000000..9787852feb --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RecordedAudioToFileController.java @@ -0,0 +1,143 @@ +/* + * Copyright 2018 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.media.AudioFormat; +import android.os.Environment; +import android.util.Log; +import androidx.annotation.Nullable; +import java.io.File; +import java.io.FileNotFoundException; +import java.io.FileOutputStream; +import java.io.IOException; +import java.io.OutputStream; +import java.util.concurrent.ExecutorService; +import org.webrtc.audio.JavaAudioDeviceModule; +import org.webrtc.audio.JavaAudioDeviceModule.SamplesReadyCallback; + +/** + * Implements the AudioRecordSamplesReadyCallback interface and writes + * recorded raw audio samples to an output file. + */ +public class RecordedAudioToFileController implements SamplesReadyCallback { + private static final String TAG = "RecordedAudioToFile"; + private static final long MAX_FILE_SIZE_IN_BYTES = 58348800L; + + private final Object lock = new Object(); + private final ExecutorService executor; + @Nullable private OutputStream rawAudioFileOutputStream; + private boolean isRunning; + private long fileSizeInBytes; + + public RecordedAudioToFileController(ExecutorService executor) { + Log.d(TAG, "ctor"); + this.executor = executor; + } + + /** + * Should be called on the same executor thread as the one provided at + * construction. + */ + public boolean start() { + Log.d(TAG, "start"); + if (!isExternalStorageWritable()) { + Log.e(TAG, "Writing to external media is not possible"); + return false; + } + synchronized (lock) { + isRunning = true; + } + return true; + } + + /** + * Should be called on the same executor thread as the one provided at + * construction. + */ + public void stop() { + Log.d(TAG, "stop"); + synchronized (lock) { + isRunning = false; + if (rawAudioFileOutputStream != null) { + try { + rawAudioFileOutputStream.close(); + } catch (IOException e) { + Log.e(TAG, "Failed to close file with saved input audio: " + e); + } + rawAudioFileOutputStream = null; + } + fileSizeInBytes = 0; + } + } + + // Checks if external storage is available for read and write. + private boolean isExternalStorageWritable() { + String state = Environment.getExternalStorageState(); + if (Environment.MEDIA_MOUNTED.equals(state)) { + return true; + } + return false; + } + + // Utilizes audio parameters to create a file name which contains sufficient + // information so that the file can be played using an external file player. + // Example: /sdcard/recorded_audio_16bits_48000Hz_mono.pcm. + private void openRawAudioOutputFile(int sampleRate, int channelCount) { + final String fileName = Environment.getExternalStorageDirectory().getPath() + File.separator + + "recorded_audio_16bits_" + String.valueOf(sampleRate) + "Hz" + + ((channelCount == 1) ? "_mono" : "_stereo") + ".pcm"; + final File outputFile = new File(fileName); + try { + rawAudioFileOutputStream = new FileOutputStream(outputFile); + } catch (FileNotFoundException e) { + Log.e(TAG, "Failed to open audio output file: " + e.getMessage()); + } + Log.d(TAG, "Opened file for recording: " + fileName); + } + + // Called when new audio samples are ready. + @Override + public void onWebRtcAudioRecordSamplesReady(JavaAudioDeviceModule.AudioSamples samples) { + // The native audio layer on Android should use 16-bit PCM format. + if (samples.getAudioFormat() != AudioFormat.ENCODING_PCM_16BIT) { + Log.e(TAG, "Invalid audio format"); + return; + } + synchronized (lock) { + // Abort early if stop() has been called. + if (!isRunning) { + return; + } + // Open a new file for the first callback only since it allows us to add audio parameters to + // the file name. + if (rawAudioFileOutputStream == null) { + openRawAudioOutputFile(samples.getSampleRate(), samples.getChannelCount()); + fileSizeInBytes = 0; + } + } + // Append the recorded 16-bit audio samples to the open output file. + executor.execute(() -> { + if (rawAudioFileOutputStream != null) { + try { + // Set a limit on max file size. 58348800 bytes corresponds to + // approximately 10 minutes of recording in mono at 48kHz. + if (fileSizeInBytes < MAX_FILE_SIZE_IN_BYTES) { + // Writes samples.getData().length bytes to output stream. + rawAudioFileOutputStream.write(samples.getData()); + fileSizeInBytes += samples.getData().length; + } + } catch (IOException e) { + Log.e(TAG, "Failed to write audio to file: " + e.getMessage()); + } + } + }); + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RoomParametersFetcher.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RoomParametersFetcher.java new file mode 100644 index 0000000000..6a0f235528 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RoomParametersFetcher.java @@ -0,0 +1,226 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.util.Log; +import java.io.IOException; +import java.io.InputStream; +import java.net.HttpURLConnection; +import java.net.URL; +import java.util.ArrayList; +import java.util.Scanner; +import java.util.List; +import org.appspot.apprtc.AppRTCClient.SignalingParameters; +import org.appspot.apprtc.util.AsyncHttpURLConnection; +import org.appspot.apprtc.util.AsyncHttpURLConnection.AsyncHttpEvents; +import org.json.JSONArray; +import org.json.JSONException; +import org.json.JSONObject; +import org.webrtc.IceCandidate; +import org.webrtc.PeerConnection; +import org.webrtc.SessionDescription; + +/** + * AsyncTask that converts an AppRTC room URL into the set of signaling + * parameters to use with that room. + */ +public class RoomParametersFetcher { + private static final String TAG = "RoomRTCClient"; + private static final int TURN_HTTP_TIMEOUT_MS = 5000; + private final RoomParametersFetcherEvents events; + private final String roomUrl; + private final String roomMessage; + + /** + * Room parameters fetcher callbacks. + */ + public interface RoomParametersFetcherEvents { + /** + * Callback fired once the room's signaling parameters + * SignalingParameters are extracted. + */ + void onSignalingParametersReady(final SignalingParameters params); + + /** + * Callback for room parameters extraction error. + */ + void onSignalingParametersError(final String description); + } + + public RoomParametersFetcher( + String roomUrl, String roomMessage, final RoomParametersFetcherEvents events) { + this.roomUrl = roomUrl; + this.roomMessage = roomMessage; + this.events = events; + } + + public void makeRequest() { + Log.d(TAG, "Connecting to room: " + roomUrl); + AsyncHttpURLConnection httpConnection = + new AsyncHttpURLConnection("POST", roomUrl, roomMessage, new AsyncHttpEvents() { + @Override + public void onHttpError(String errorMessage) { + Log.e(TAG, "Room connection error: " + errorMessage); + events.onSignalingParametersError(errorMessage); + } + + @Override + public void onHttpComplete(String response) { + roomHttpResponseParse(response); + } + }); + httpConnection.send(); + } + + private void roomHttpResponseParse(String response) { + Log.d(TAG, "Room response: " + response); + try { + List<IceCandidate> iceCandidates = null; + SessionDescription offerSdp = null; + JSONObject roomJson = new JSONObject(response); + + String result = roomJson.getString("result"); + if (!result.equals("SUCCESS")) { + events.onSignalingParametersError("Room response error: " + result); + return; + } + response = roomJson.getString("params"); + roomJson = new JSONObject(response); + String roomId = roomJson.getString("room_id"); + String clientId = roomJson.getString("client_id"); + String wssUrl = roomJson.getString("wss_url"); + String wssPostUrl = roomJson.getString("wss_post_url"); + boolean initiator = (roomJson.getBoolean("is_initiator")); + if (!initiator) { + iceCandidates = new ArrayList<>(); + String messagesString = roomJson.getString("messages"); + JSONArray messages = new JSONArray(messagesString); + for (int i = 0; i < messages.length(); ++i) { + String messageString = messages.getString(i); + JSONObject message = new JSONObject(messageString); + String messageType = message.getString("type"); + Log.d(TAG, "GAE->C #" + i + " : " + messageString); + if (messageType.equals("offer")) { + offerSdp = new SessionDescription( + SessionDescription.Type.fromCanonicalForm(messageType), message.getString("sdp")); + } else if (messageType.equals("candidate")) { + IceCandidate candidate = new IceCandidate( + message.getString("id"), message.getInt("label"), message.getString("candidate")); + iceCandidates.add(candidate); + } else { + Log.e(TAG, "Unknown message: " + messageString); + } + } + } + Log.d(TAG, "RoomId: " + roomId + ". ClientId: " + clientId); + Log.d(TAG, "Initiator: " + initiator); + Log.d(TAG, "WSS url: " + wssUrl); + Log.d(TAG, "WSS POST url: " + wssPostUrl); + + List<PeerConnection.IceServer> iceServers = + iceServersFromPCConfigJSON(roomJson.getString("pc_config")); + boolean isTurnPresent = false; + for (PeerConnection.IceServer server : iceServers) { + Log.d(TAG, "IceServer: " + server); + for (String uri : server.urls) { + if (uri.startsWith("turn:")) { + isTurnPresent = true; + break; + } + } + } + // Request TURN servers. + if (!isTurnPresent && !roomJson.optString("ice_server_url").isEmpty()) { + List<PeerConnection.IceServer> turnServers = + requestTurnServers(roomJson.getString("ice_server_url")); + for (PeerConnection.IceServer turnServer : turnServers) { + Log.d(TAG, "TurnServer: " + turnServer); + iceServers.add(turnServer); + } + } + + SignalingParameters params = new SignalingParameters( + iceServers, initiator, clientId, wssUrl, wssPostUrl, offerSdp, iceCandidates); + events.onSignalingParametersReady(params); + } catch (JSONException e) { + events.onSignalingParametersError("Room JSON parsing error: " + e.toString()); + } catch (IOException e) { + events.onSignalingParametersError("Room IO error: " + e.toString()); + } + } + + // Requests & returns a TURN ICE Server based on a request URL. Must be run + // off the main thread! + @SuppressWarnings("UseNetworkAnnotations") + private List<PeerConnection.IceServer> requestTurnServers(String url) + throws IOException, JSONException { + List<PeerConnection.IceServer> turnServers = new ArrayList<>(); + Log.d(TAG, "Request TURN from: " + url); + HttpURLConnection connection = (HttpURLConnection) new URL(url).openConnection(); + connection.setDoOutput(true); + connection.setRequestProperty("REFERER", "https://appr.tc"); + connection.setConnectTimeout(TURN_HTTP_TIMEOUT_MS); + connection.setReadTimeout(TURN_HTTP_TIMEOUT_MS); + int responseCode = connection.getResponseCode(); + if (responseCode != 200) { + throw new IOException("Non-200 response when requesting TURN server from " + url + " : " + + connection.getHeaderField(null)); + } + InputStream responseStream = connection.getInputStream(); + String response = drainStream(responseStream); + connection.disconnect(); + Log.d(TAG, "TURN response: " + response); + JSONObject responseJSON = new JSONObject(response); + JSONArray iceServers = responseJSON.getJSONArray("iceServers"); + for (int i = 0; i < iceServers.length(); ++i) { + JSONObject server = iceServers.getJSONObject(i); + JSONArray turnUrls = server.getJSONArray("urls"); + String username = server.has("username") ? server.getString("username") : ""; + String credential = server.has("credential") ? server.getString("credential") : ""; + for (int j = 0; j < turnUrls.length(); j++) { + String turnUrl = turnUrls.getString(j); + PeerConnection.IceServer turnServer = + PeerConnection.IceServer.builder(turnUrl) + .setUsername(username) + .setPassword(credential) + .createIceServer(); + turnServers.add(turnServer); + } + } + return turnServers; + } + + // Return the list of ICE servers described by a WebRTCPeerConnection + // configuration string. + private List<PeerConnection.IceServer> iceServersFromPCConfigJSON(String pcConfig) + throws JSONException { + JSONObject json = new JSONObject(pcConfig); + JSONArray servers = json.getJSONArray("iceServers"); + List<PeerConnection.IceServer> ret = new ArrayList<>(); + for (int i = 0; i < servers.length(); ++i) { + JSONObject server = servers.getJSONObject(i); + String url = server.getString("urls"); + String credential = server.has("credential") ? server.getString("credential") : ""; + PeerConnection.IceServer turnServer = + PeerConnection.IceServer.builder(url) + .setPassword(credential) + .createIceServer(); + ret.add(turnServer); + } + return ret; + } + + // Return the contents of an InputStream as a String. + private static String drainStream(InputStream in) { + Scanner s = new Scanner(in, "UTF-8").useDelimiter("\\A"); + return s.hasNext() ? s.next() : ""; + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RtcEventLog.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RtcEventLog.java new file mode 100644 index 0000000000..103ad10f0b --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RtcEventLog.java @@ -0,0 +1,73 @@ +/* + * Copyright 2018 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.os.ParcelFileDescriptor; +import android.util.Log; +import java.io.File; +import java.io.IOException; +import org.webrtc.PeerConnection; + +public class RtcEventLog { + private static final String TAG = "RtcEventLog"; + private static final int OUTPUT_FILE_MAX_BYTES = 10_000_000; + private final PeerConnection peerConnection; + private RtcEventLogState state = RtcEventLogState.INACTIVE; + + enum RtcEventLogState { + INACTIVE, + STARTED, + STOPPED, + } + + public RtcEventLog(PeerConnection peerConnection) { + if (peerConnection == null) { + throw new NullPointerException("The peer connection is null."); + } + this.peerConnection = peerConnection; + } + + public void start(final File outputFile) { + if (state == RtcEventLogState.STARTED) { + Log.e(TAG, "RtcEventLog has already started."); + return; + } + final ParcelFileDescriptor fileDescriptor; + try { + fileDescriptor = ParcelFileDescriptor.open(outputFile, + ParcelFileDescriptor.MODE_READ_WRITE | ParcelFileDescriptor.MODE_CREATE + | ParcelFileDescriptor.MODE_TRUNCATE); + } catch (IOException e) { + Log.e(TAG, "Failed to create a new file", e); + return; + } + + // Passes ownership of the file to WebRTC. + boolean success = + peerConnection.startRtcEventLog(fileDescriptor.detachFd(), OUTPUT_FILE_MAX_BYTES); + if (!success) { + Log.e(TAG, "Failed to start RTC event log."); + return; + } + state = RtcEventLogState.STARTED; + Log.d(TAG, "RtcEventLog started."); + } + + public void stop() { + if (state != RtcEventLogState.STARTED) { + Log.e(TAG, "RtcEventLog was not started."); + return; + } + peerConnection.stopRtcEventLog(); + state = RtcEventLogState.STOPPED; + Log.d(TAG, "RtcEventLog stopped."); + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/SettingsActivity.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/SettingsActivity.java new file mode 100644 index 0000000000..e9c6f6b798 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/SettingsActivity.java @@ -0,0 +1,317 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.app.Activity; +import android.content.SharedPreferences; +import android.content.SharedPreferences.OnSharedPreferenceChangeListener; +import android.os.Bundle; +import android.preference.ListPreference; +import android.preference.Preference; +import org.webrtc.Camera2Enumerator; +import org.webrtc.audio.JavaAudioDeviceModule; + +/** + * Settings activity for AppRTC. + */ +public class SettingsActivity extends Activity implements OnSharedPreferenceChangeListener { + private SettingsFragment settingsFragment; + private String keyprefVideoCall; + private String keyprefScreencapture; + private String keyprefCamera2; + private String keyprefResolution; + private String keyprefFps; + private String keyprefCaptureQualitySlider; + private String keyprefMaxVideoBitrateType; + private String keyprefMaxVideoBitrateValue; + private String keyPrefVideoCodec; + private String keyprefHwCodec; + private String keyprefCaptureToTexture; + private String keyprefFlexfec; + + private String keyprefStartAudioBitrateType; + private String keyprefStartAudioBitrateValue; + private String keyPrefAudioCodec; + private String keyprefNoAudioProcessing; + private String keyprefAecDump; + private String keyprefEnableSaveInputAudioToFile; + private String keyprefOpenSLES; + private String keyprefDisableBuiltInAEC; + private String keyprefDisableBuiltInAGC; + private String keyprefDisableBuiltInNS; + private String keyprefDisableWebRtcAGCAndHPF; + private String keyprefSpeakerphone; + + private String keyPrefRoomServerUrl; + private String keyPrefDisplayHud; + private String keyPrefTracing; + private String keyprefEnabledRtcEventLog; + + private String keyprefEnableDataChannel; + private String keyprefOrdered; + private String keyprefMaxRetransmitTimeMs; + private String keyprefMaxRetransmits; + private String keyprefDataProtocol; + private String keyprefNegotiated; + private String keyprefDataId; + + @Override + protected void onCreate(Bundle savedInstanceState) { + super.onCreate(savedInstanceState); + keyprefVideoCall = getString(R.string.pref_videocall_key); + keyprefScreencapture = getString(R.string.pref_screencapture_key); + keyprefCamera2 = getString(R.string.pref_camera2_key); + keyprefResolution = getString(R.string.pref_resolution_key); + keyprefFps = getString(R.string.pref_fps_key); + keyprefCaptureQualitySlider = getString(R.string.pref_capturequalityslider_key); + keyprefMaxVideoBitrateType = getString(R.string.pref_maxvideobitrate_key); + keyprefMaxVideoBitrateValue = getString(R.string.pref_maxvideobitratevalue_key); + keyPrefVideoCodec = getString(R.string.pref_videocodec_key); + keyprefHwCodec = getString(R.string.pref_hwcodec_key); + keyprefCaptureToTexture = getString(R.string.pref_capturetotexture_key); + keyprefFlexfec = getString(R.string.pref_flexfec_key); + + keyprefStartAudioBitrateType = getString(R.string.pref_startaudiobitrate_key); + keyprefStartAudioBitrateValue = getString(R.string.pref_startaudiobitratevalue_key); + keyPrefAudioCodec = getString(R.string.pref_audiocodec_key); + keyprefNoAudioProcessing = getString(R.string.pref_noaudioprocessing_key); + keyprefAecDump = getString(R.string.pref_aecdump_key); + keyprefEnableSaveInputAudioToFile = + getString(R.string.pref_enable_save_input_audio_to_file_key); + keyprefOpenSLES = getString(R.string.pref_opensles_key); + keyprefDisableBuiltInAEC = getString(R.string.pref_disable_built_in_aec_key); + keyprefDisableBuiltInAGC = getString(R.string.pref_disable_built_in_agc_key); + keyprefDisableBuiltInNS = getString(R.string.pref_disable_built_in_ns_key); + keyprefDisableWebRtcAGCAndHPF = getString(R.string.pref_disable_webrtc_agc_and_hpf_key); + keyprefSpeakerphone = getString(R.string.pref_speakerphone_key); + + keyprefEnableDataChannel = getString(R.string.pref_enable_datachannel_key); + keyprefOrdered = getString(R.string.pref_ordered_key); + keyprefMaxRetransmitTimeMs = getString(R.string.pref_max_retransmit_time_ms_key); + keyprefMaxRetransmits = getString(R.string.pref_max_retransmits_key); + keyprefDataProtocol = getString(R.string.pref_data_protocol_key); + keyprefNegotiated = getString(R.string.pref_negotiated_key); + keyprefDataId = getString(R.string.pref_data_id_key); + + keyPrefRoomServerUrl = getString(R.string.pref_room_server_url_key); + keyPrefDisplayHud = getString(R.string.pref_displayhud_key); + keyPrefTracing = getString(R.string.pref_tracing_key); + keyprefEnabledRtcEventLog = getString(R.string.pref_enable_rtceventlog_key); + + // Display the fragment as the main content. + settingsFragment = new SettingsFragment(); + getFragmentManager() + .beginTransaction() + .replace(android.R.id.content, settingsFragment) + .commit(); + } + + @Override + protected void onResume() { + super.onResume(); + // Set summary to be the user-description for the selected value + SharedPreferences sharedPreferences = + settingsFragment.getPreferenceScreen().getSharedPreferences(); + sharedPreferences.registerOnSharedPreferenceChangeListener(this); + updateSummaryB(sharedPreferences, keyprefVideoCall); + updateSummaryB(sharedPreferences, keyprefScreencapture); + updateSummaryB(sharedPreferences, keyprefCamera2); + updateSummary(sharedPreferences, keyprefResolution); + updateSummary(sharedPreferences, keyprefFps); + updateSummaryB(sharedPreferences, keyprefCaptureQualitySlider); + updateSummary(sharedPreferences, keyprefMaxVideoBitrateType); + updateSummaryBitrate(sharedPreferences, keyprefMaxVideoBitrateValue); + setVideoBitrateEnable(sharedPreferences); + updateSummary(sharedPreferences, keyPrefVideoCodec); + updateSummaryB(sharedPreferences, keyprefHwCodec); + updateSummaryB(sharedPreferences, keyprefCaptureToTexture); + updateSummaryB(sharedPreferences, keyprefFlexfec); + + updateSummary(sharedPreferences, keyprefStartAudioBitrateType); + updateSummaryBitrate(sharedPreferences, keyprefStartAudioBitrateValue); + setAudioBitrateEnable(sharedPreferences); + updateSummary(sharedPreferences, keyPrefAudioCodec); + updateSummaryB(sharedPreferences, keyprefNoAudioProcessing); + updateSummaryB(sharedPreferences, keyprefAecDump); + updateSummaryB(sharedPreferences, keyprefEnableSaveInputAudioToFile); + updateSummaryB(sharedPreferences, keyprefOpenSLES); + updateSummaryB(sharedPreferences, keyprefDisableBuiltInAEC); + updateSummaryB(sharedPreferences, keyprefDisableBuiltInAGC); + updateSummaryB(sharedPreferences, keyprefDisableBuiltInNS); + updateSummaryB(sharedPreferences, keyprefDisableWebRtcAGCAndHPF); + updateSummaryList(sharedPreferences, keyprefSpeakerphone); + + updateSummaryB(sharedPreferences, keyprefEnableDataChannel); + updateSummaryB(sharedPreferences, keyprefOrdered); + updateSummary(sharedPreferences, keyprefMaxRetransmitTimeMs); + updateSummary(sharedPreferences, keyprefMaxRetransmits); + updateSummary(sharedPreferences, keyprefDataProtocol); + updateSummaryB(sharedPreferences, keyprefNegotiated); + updateSummary(sharedPreferences, keyprefDataId); + setDataChannelEnable(sharedPreferences); + + updateSummary(sharedPreferences, keyPrefRoomServerUrl); + updateSummaryB(sharedPreferences, keyPrefDisplayHud); + updateSummaryB(sharedPreferences, keyPrefTracing); + updateSummaryB(sharedPreferences, keyprefEnabledRtcEventLog); + + if (!Camera2Enumerator.isSupported(this)) { + Preference camera2Preference = settingsFragment.findPreference(keyprefCamera2); + + camera2Preference.setSummary(getString(R.string.pref_camera2_not_supported)); + camera2Preference.setEnabled(false); + } + + if (!JavaAudioDeviceModule.isBuiltInAcousticEchoCancelerSupported()) { + Preference disableBuiltInAECPreference = + settingsFragment.findPreference(keyprefDisableBuiltInAEC); + + disableBuiltInAECPreference.setSummary(getString(R.string.pref_built_in_aec_not_available)); + disableBuiltInAECPreference.setEnabled(false); + } + + Preference disableBuiltInAGCPreference = + settingsFragment.findPreference(keyprefDisableBuiltInAGC); + + disableBuiltInAGCPreference.setSummary(getString(R.string.pref_built_in_agc_not_available)); + disableBuiltInAGCPreference.setEnabled(false); + + if (!JavaAudioDeviceModule.isBuiltInNoiseSuppressorSupported()) { + Preference disableBuiltInNSPreference = + settingsFragment.findPreference(keyprefDisableBuiltInNS); + + disableBuiltInNSPreference.setSummary(getString(R.string.pref_built_in_ns_not_available)); + disableBuiltInNSPreference.setEnabled(false); + } + } + + @Override + protected void onPause() { + super.onPause(); + SharedPreferences sharedPreferences = + settingsFragment.getPreferenceScreen().getSharedPreferences(); + sharedPreferences.unregisterOnSharedPreferenceChangeListener(this); + } + + @Override + public void onSharedPreferenceChanged(SharedPreferences sharedPreferences, String key) { + // clang-format off + if (key.equals(keyprefResolution) + || key.equals(keyprefFps) + || key.equals(keyprefMaxVideoBitrateType) + || key.equals(keyPrefVideoCodec) + || key.equals(keyprefStartAudioBitrateType) + || key.equals(keyPrefAudioCodec) + || key.equals(keyPrefRoomServerUrl) + || key.equals(keyprefMaxRetransmitTimeMs) + || key.equals(keyprefMaxRetransmits) + || key.equals(keyprefDataProtocol) + || key.equals(keyprefDataId)) { + updateSummary(sharedPreferences, key); + } else if (key.equals(keyprefMaxVideoBitrateValue) + || key.equals(keyprefStartAudioBitrateValue)) { + updateSummaryBitrate(sharedPreferences, key); + } else if (key.equals(keyprefVideoCall) + || key.equals(keyprefScreencapture) + || key.equals(keyprefCamera2) + || key.equals(keyPrefTracing) + || key.equals(keyprefCaptureQualitySlider) + || key.equals(keyprefHwCodec) + || key.equals(keyprefCaptureToTexture) + || key.equals(keyprefFlexfec) + || key.equals(keyprefNoAudioProcessing) + || key.equals(keyprefAecDump) + || key.equals(keyprefEnableSaveInputAudioToFile) + || key.equals(keyprefOpenSLES) + || key.equals(keyprefDisableBuiltInAEC) + || key.equals(keyprefDisableBuiltInAGC) + || key.equals(keyprefDisableBuiltInNS) + || key.equals(keyprefDisableWebRtcAGCAndHPF) + || key.equals(keyPrefDisplayHud) + || key.equals(keyprefEnableDataChannel) + || key.equals(keyprefOrdered) + || key.equals(keyprefNegotiated) + || key.equals(keyprefEnabledRtcEventLog)) { + updateSummaryB(sharedPreferences, key); + } else if (key.equals(keyprefSpeakerphone)) { + updateSummaryList(sharedPreferences, key); + } + // clang-format on + if (key.equals(keyprefMaxVideoBitrateType)) { + setVideoBitrateEnable(sharedPreferences); + } + if (key.equals(keyprefStartAudioBitrateType)) { + setAudioBitrateEnable(sharedPreferences); + } + if (key.equals(keyprefEnableDataChannel)) { + setDataChannelEnable(sharedPreferences); + } + } + + private void updateSummary(SharedPreferences sharedPreferences, String key) { + Preference updatedPref = settingsFragment.findPreference(key); + // Set summary to be the user-description for the selected value + updatedPref.setSummary(sharedPreferences.getString(key, "")); + } + + private void updateSummaryBitrate(SharedPreferences sharedPreferences, String key) { + Preference updatedPref = settingsFragment.findPreference(key); + updatedPref.setSummary(sharedPreferences.getString(key, "") + " kbps"); + } + + private void updateSummaryB(SharedPreferences sharedPreferences, String key) { + Preference updatedPref = settingsFragment.findPreference(key); + updatedPref.setSummary(sharedPreferences.getBoolean(key, true) + ? getString(R.string.pref_value_enabled) + : getString(R.string.pref_value_disabled)); + } + + private void updateSummaryList(SharedPreferences sharedPreferences, String key) { + ListPreference updatedPref = (ListPreference) settingsFragment.findPreference(key); + updatedPref.setSummary(updatedPref.getEntry()); + } + + private void setVideoBitrateEnable(SharedPreferences sharedPreferences) { + Preference bitratePreferenceValue = + settingsFragment.findPreference(keyprefMaxVideoBitrateValue); + String bitrateTypeDefault = getString(R.string.pref_maxvideobitrate_default); + String bitrateType = + sharedPreferences.getString(keyprefMaxVideoBitrateType, bitrateTypeDefault); + if (bitrateType.equals(bitrateTypeDefault)) { + bitratePreferenceValue.setEnabled(false); + } else { + bitratePreferenceValue.setEnabled(true); + } + } + + private void setAudioBitrateEnable(SharedPreferences sharedPreferences) { + Preference bitratePreferenceValue = + settingsFragment.findPreference(keyprefStartAudioBitrateValue); + String bitrateTypeDefault = getString(R.string.pref_startaudiobitrate_default); + String bitrateType = + sharedPreferences.getString(keyprefStartAudioBitrateType, bitrateTypeDefault); + if (bitrateType.equals(bitrateTypeDefault)) { + bitratePreferenceValue.setEnabled(false); + } else { + bitratePreferenceValue.setEnabled(true); + } + } + + private void setDataChannelEnable(SharedPreferences sharedPreferences) { + boolean enabled = sharedPreferences.getBoolean(keyprefEnableDataChannel, true); + settingsFragment.findPreference(keyprefOrdered).setEnabled(enabled); + settingsFragment.findPreference(keyprefMaxRetransmitTimeMs).setEnabled(enabled); + settingsFragment.findPreference(keyprefMaxRetransmits).setEnabled(enabled); + settingsFragment.findPreference(keyprefDataProtocol).setEnabled(enabled); + settingsFragment.findPreference(keyprefNegotiated).setEnabled(enabled); + settingsFragment.findPreference(keyprefDataId).setEnabled(enabled); + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/SettingsFragment.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/SettingsFragment.java new file mode 100644 index 0000000000..d969bd7d32 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/SettingsFragment.java @@ -0,0 +1,26 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.os.Bundle; +import android.preference.PreferenceFragment; + +/** + * Settings fragment for AppRTC. + */ +public class SettingsFragment extends PreferenceFragment { + @Override + public void onCreate(Bundle savedInstanceState) { + super.onCreate(savedInstanceState); + // Load the preferences from an XML resource + addPreferencesFromResource(R.xml.preferences); + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/TCPChannelClient.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/TCPChannelClient.java new file mode 100644 index 0000000000..d869d7ca66 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/TCPChannelClient.java @@ -0,0 +1,362 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.util.Log; +import androidx.annotation.Nullable; +import java.io.BufferedReader; +import java.io.IOException; +import java.io.InputStreamReader; +import java.io.OutputStreamWriter; +import java.io.PrintWriter; +import java.net.InetAddress; +import java.net.ServerSocket; +import java.net.Socket; +import java.net.UnknownHostException; +import java.nio.charset.Charset; +import java.util.concurrent.ExecutorService; +import org.webrtc.ThreadUtils; + +/** + * Replacement for WebSocketChannelClient for direct communication between two IP addresses. Handles + * the signaling between the two clients using a TCP connection. + * <p> + * All public methods should be called from a looper executor thread + * passed in a constructor, otherwise exception will be thrown. + * All events are dispatched on the same thread. + */ +public class TCPChannelClient { + private static final String TAG = "TCPChannelClient"; + + private final ExecutorService executor; + private final ThreadUtils.ThreadChecker executorThreadCheck; + private final TCPChannelEvents eventListener; + private TCPSocket socket; + + /** + * Callback interface for messages delivered on TCP Connection. All callbacks are invoked from the + * looper executor thread. + */ + public interface TCPChannelEvents { + void onTCPConnected(boolean server); + void onTCPMessage(String message); + void onTCPError(String description); + void onTCPClose(); + } + + /** + * Initializes the TCPChannelClient. If IP is a local IP address, starts a listening server on + * that IP. If not, instead connects to the IP. + * + * @param eventListener Listener that will receive events from the client. + * @param ip IP address to listen on or connect to. + * @param port Port to listen on or connect to. + */ + public TCPChannelClient( + ExecutorService executor, TCPChannelEvents eventListener, String ip, int port) { + this.executor = executor; + executorThreadCheck = new ThreadUtils.ThreadChecker(); + executorThreadCheck.detachThread(); + this.eventListener = eventListener; + + InetAddress address; + try { + address = InetAddress.getByName(ip); + } catch (UnknownHostException e) { + reportError("Invalid IP address."); + return; + } + + if (address.isAnyLocalAddress()) { + socket = new TCPSocketServer(address, port); + } else { + socket = new TCPSocketClient(address, port); + } + + socket.start(); + } + + /** + * Disconnects the client if not already disconnected. This will fire the onTCPClose event. + */ + public void disconnect() { + executorThreadCheck.checkIsOnValidThread(); + + socket.disconnect(); + } + + /** + * Sends a message on the socket. + * + * @param message Message to be sent. + */ + public void send(String message) { + executorThreadCheck.checkIsOnValidThread(); + + socket.send(message); + } + + /** + * Helper method for firing onTCPError events. Calls onTCPError on the executor thread. + */ + private void reportError(final String message) { + Log.e(TAG, "TCP Error: " + message); + executor.execute(new Runnable() { + @Override + public void run() { + eventListener.onTCPError(message); + } + }); + } + + /** + * Base class for server and client sockets. Contains a listening thread that will call + * eventListener.onTCPMessage on new messages. + */ + private abstract class TCPSocket extends Thread { + // Lock for editing out and rawSocket + protected final Object rawSocketLock; + @Nullable + private PrintWriter out; + @Nullable + private Socket rawSocket; + + /** + * Connect to the peer, potentially a slow operation. + * + * @return Socket connection, null if connection failed. + */ + @Nullable + public abstract Socket connect(); + + /** Returns true if sockets is a server rawSocket. */ + public abstract boolean isServer(); + + TCPSocket() { + rawSocketLock = new Object(); + } + + /** + * The listening thread. + */ + @Override + public void run() { + Log.d(TAG, "Listening thread started..."); + + // Receive connection to temporary variable first, so we don't block. + Socket tempSocket = connect(); + BufferedReader in; + + Log.d(TAG, "TCP connection established."); + + synchronized (rawSocketLock) { + if (rawSocket != null) { + Log.e(TAG, "Socket already existed and will be replaced."); + } + + rawSocket = tempSocket; + + // Connecting failed, error has already been reported, just exit. + if (rawSocket == null) { + return; + } + + try { + out = new PrintWriter( + new OutputStreamWriter(rawSocket.getOutputStream(), Charset.forName("UTF-8")), true); + in = new BufferedReader( + new InputStreamReader(rawSocket.getInputStream(), Charset.forName("UTF-8"))); + } catch (IOException e) { + reportError("Failed to open IO on rawSocket: " + e.getMessage()); + return; + } + } + + Log.v(TAG, "Execute onTCPConnected"); + executor.execute(new Runnable() { + @Override + public void run() { + Log.v(TAG, "Run onTCPConnected"); + eventListener.onTCPConnected(isServer()); + } + }); + + while (true) { + final String message; + try { + message = in.readLine(); + } catch (IOException e) { + synchronized (rawSocketLock) { + // If socket was closed, this is expected. + if (rawSocket == null) { + break; + } + } + + reportError("Failed to read from rawSocket: " + e.getMessage()); + break; + } + + // No data received, rawSocket probably closed. + if (message == null) { + break; + } + + executor.execute(new Runnable() { + @Override + public void run() { + Log.v(TAG, "Receive: " + message); + eventListener.onTCPMessage(message); + } + }); + } + + Log.d(TAG, "Receiving thread exiting..."); + + // Close the rawSocket if it is still open. + disconnect(); + } + + /** Closes the rawSocket if it is still open. Also fires the onTCPClose event. */ + public void disconnect() { + try { + synchronized (rawSocketLock) { + if (rawSocket != null) { + rawSocket.close(); + rawSocket = null; + out = null; + + executor.execute(new Runnable() { + @Override + public void run() { + eventListener.onTCPClose(); + } + }); + } + } + } catch (IOException e) { + reportError("Failed to close rawSocket: " + e.getMessage()); + } + } + + /** + * Sends a message on the socket. Should only be called on the executor thread. + */ + public void send(String message) { + Log.v(TAG, "Send: " + message); + + synchronized (rawSocketLock) { + if (out == null) { + reportError("Sending data on closed socket."); + return; + } + + out.write(message + "\n"); + out.flush(); + } + } + } + + private class TCPSocketServer extends TCPSocket { + // Server socket is also guarded by rawSocketLock. + @Nullable + private ServerSocket serverSocket; + + final private InetAddress address; + final private int port; + + public TCPSocketServer(InetAddress address, int port) { + this.address = address; + this.port = port; + } + + /** Opens a listening socket and waits for a connection. */ + @Nullable + @Override + public Socket connect() { + Log.d(TAG, "Listening on [" + address.getHostAddress() + "]:" + Integer.toString(port)); + + final ServerSocket tempSocket; + try { + tempSocket = new ServerSocket(port, 0, address); + } catch (IOException e) { + reportError("Failed to create server socket: " + e.getMessage()); + return null; + } + + synchronized (rawSocketLock) { + if (serverSocket != null) { + Log.e(TAG, "Server rawSocket was already listening and new will be opened."); + } + + serverSocket = tempSocket; + } + + try { + return tempSocket.accept(); + } catch (IOException e) { + reportError("Failed to receive connection: " + e.getMessage()); + return null; + } + } + + /** Closes the listening socket and calls super. */ + @Override + public void disconnect() { + try { + synchronized (rawSocketLock) { + if (serverSocket != null) { + serverSocket.close(); + serverSocket = null; + } + } + } catch (IOException e) { + reportError("Failed to close server socket: " + e.getMessage()); + } + + super.disconnect(); + } + + @Override + public boolean isServer() { + return true; + } + } + + private class TCPSocketClient extends TCPSocket { + final private InetAddress address; + final private int port; + + public TCPSocketClient(InetAddress address, int port) { + this.address = address; + this.port = port; + } + + /** Connects to the peer. */ + @Nullable + @Override + public Socket connect() { + Log.d(TAG, "Connecting to [" + address.getHostAddress() + "]:" + Integer.toString(port)); + + try { + return new Socket(address, port); + } catch (IOException e) { + reportError("Failed to connect: " + e.getMessage()); + return null; + } + } + + @Override + public boolean isServer() { + return false; + } + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/UnhandledExceptionHandler.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/UnhandledExceptionHandler.java new file mode 100644 index 0000000000..b256400119 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/UnhandledExceptionHandler.java @@ -0,0 +1,85 @@ +/* + * Copyright 2013 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.app.Activity; +import android.app.AlertDialog; +import android.content.DialogInterface; +import android.util.Log; +import android.util.TypedValue; +import android.widget.ScrollView; +import android.widget.TextView; + +import java.io.PrintWriter; +import java.io.StringWriter; + +/** + * Singleton helper: install a default unhandled exception handler which shows + * an informative dialog and kills the app. Useful for apps whose + * error-handling consists of throwing RuntimeExceptions. + * NOTE: almost always more useful to + * Thread.setDefaultUncaughtExceptionHandler() rather than + * Thread.setUncaughtExceptionHandler(), to apply to background threads as well. + */ +public class UnhandledExceptionHandler implements Thread.UncaughtExceptionHandler { + private static final String TAG = "AppRTCMobileActivity"; + private final Activity activity; + + public UnhandledExceptionHandler(final Activity activity) { + this.activity = activity; + } + + @Override + public void uncaughtException(Thread unusedThread, final Throwable e) { + activity.runOnUiThread(new Runnable() { + @Override + public void run() { + String title = "Fatal error: " + getTopLevelCauseMessage(e); + String msg = getRecursiveStackTrace(e); + TextView errorView = new TextView(activity); + errorView.setText(msg); + errorView.setTextSize(TypedValue.COMPLEX_UNIT_SP, 8); + ScrollView scrollingContainer = new ScrollView(activity); + scrollingContainer.addView(errorView); + Log.e(TAG, title + "\n\n" + msg); + DialogInterface.OnClickListener listener = new DialogInterface.OnClickListener() { + @Override + public void onClick(DialogInterface dialog, int which) { + dialog.dismiss(); + System.exit(1); + } + }; + AlertDialog.Builder builder = new AlertDialog.Builder(activity); + builder.setTitle(title) + .setView(scrollingContainer) + .setPositiveButton("Exit", listener) + .show(); + } + }); + } + + // Returns the Message attached to the original Cause of `t`. + private static String getTopLevelCauseMessage(Throwable t) { + Throwable topLevelCause = t; + while (topLevelCause.getCause() != null) { + topLevelCause = topLevelCause.getCause(); + } + return topLevelCause.getMessage(); + } + + // Returns a human-readable String of the stacktrace in `t`, recursively + // through all Causes that led to `t`. + private static String getRecursiveStackTrace(Throwable t) { + StringWriter writer = new StringWriter(); + t.printStackTrace(new PrintWriter(writer)); + return writer.toString(); + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketChannelClient.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketChannelClient.java new file mode 100644 index 0000000000..5fa410889a --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketChannelClient.java @@ -0,0 +1,296 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.os.Handler; +import android.util.Log; +import androidx.annotation.Nullable; +import de.tavendo.autobahn.WebSocket.WebSocketConnectionObserver; +import de.tavendo.autobahn.WebSocketConnection; +import de.tavendo.autobahn.WebSocketException; +import java.net.URI; +import java.net.URISyntaxException; +import java.util.ArrayList; +import java.util.List; +import org.appspot.apprtc.util.AsyncHttpURLConnection; +import org.appspot.apprtc.util.AsyncHttpURLConnection.AsyncHttpEvents; +import org.json.JSONException; +import org.json.JSONObject; + +/** + * WebSocket client implementation. + * + * <p>All public methods should be called from a looper executor thread + * passed in a constructor, otherwise exception will be thrown. + * All events are dispatched on the same thread. + */ +public class WebSocketChannelClient { + private static final String TAG = "WSChannelRTCClient"; + private static final int CLOSE_TIMEOUT = 1000; + private final WebSocketChannelEvents events; + private final Handler handler; + private WebSocketConnection ws; + private String wsServerUrl; + private String postServerUrl; + @Nullable + private String roomID; + @Nullable + private String clientID; + private WebSocketConnectionState state; + // Do not remove this member variable. If this is removed, the observer gets garbage collected and + // this causes test breakages. + private WebSocketObserver wsObserver; + private final Object closeEventLock = new Object(); + private boolean closeEvent; + // WebSocket send queue. Messages are added to the queue when WebSocket + // client is not registered and are consumed in register() call. + private final List<String> wsSendQueue = new ArrayList<>(); + + /** + * Possible WebSocket connection states. + */ + public enum WebSocketConnectionState { NEW, CONNECTED, REGISTERED, CLOSED, ERROR } + + /** + * Callback interface for messages delivered on WebSocket. + * All events are dispatched from a looper executor thread. + */ + public interface WebSocketChannelEvents { + void onWebSocketMessage(final String message); + void onWebSocketClose(); + void onWebSocketError(final String description); + } + + public WebSocketChannelClient(Handler handler, WebSocketChannelEvents events) { + this.handler = handler; + this.events = events; + roomID = null; + clientID = null; + state = WebSocketConnectionState.NEW; + } + + public WebSocketConnectionState getState() { + return state; + } + + public void connect(final String wsUrl, final String postUrl) { + checkIfCalledOnValidThread(); + if (state != WebSocketConnectionState.NEW) { + Log.e(TAG, "WebSocket is already connected."); + return; + } + wsServerUrl = wsUrl; + postServerUrl = postUrl; + closeEvent = false; + + Log.d(TAG, "Connecting WebSocket to: " + wsUrl + ". Post URL: " + postUrl); + ws = new WebSocketConnection(); + wsObserver = new WebSocketObserver(); + try { + ws.connect(new URI(wsServerUrl), wsObserver); + } catch (URISyntaxException e) { + reportError("URI error: " + e.getMessage()); + } catch (WebSocketException e) { + reportError("WebSocket connection error: " + e.getMessage()); + } + } + + public void register(final String roomID, final String clientID) { + checkIfCalledOnValidThread(); + this.roomID = roomID; + this.clientID = clientID; + if (state != WebSocketConnectionState.CONNECTED) { + Log.w(TAG, "WebSocket register() in state " + state); + return; + } + Log.d(TAG, "Registering WebSocket for room " + roomID + ". ClientID: " + clientID); + JSONObject json = new JSONObject(); + try { + json.put("cmd", "register"); + json.put("roomid", roomID); + json.put("clientid", clientID); + Log.d(TAG, "C->WSS: " + json.toString()); + ws.sendTextMessage(json.toString()); + state = WebSocketConnectionState.REGISTERED; + // Send any previously accumulated messages. + for (String sendMessage : wsSendQueue) { + send(sendMessage); + } + wsSendQueue.clear(); + } catch (JSONException e) { + reportError("WebSocket register JSON error: " + e.getMessage()); + } + } + + public void send(String message) { + checkIfCalledOnValidThread(); + switch (state) { + case NEW: + case CONNECTED: + // Store outgoing messages and send them after websocket client + // is registered. + Log.d(TAG, "WS ACC: " + message); + wsSendQueue.add(message); + return; + case ERROR: + case CLOSED: + Log.e(TAG, "WebSocket send() in error or closed state : " + message); + return; + case REGISTERED: + JSONObject json = new JSONObject(); + try { + json.put("cmd", "send"); + json.put("msg", message); + message = json.toString(); + Log.d(TAG, "C->WSS: " + message); + ws.sendTextMessage(message); + } catch (JSONException e) { + reportError("WebSocket send JSON error: " + e.getMessage()); + } + break; + } + } + + // This call can be used to send WebSocket messages before WebSocket + // connection is opened. + public void post(String message) { + checkIfCalledOnValidThread(); + sendWSSMessage("POST", message); + } + + public void disconnect(boolean waitForComplete) { + checkIfCalledOnValidThread(); + Log.d(TAG, "Disconnect WebSocket. State: " + state); + if (state == WebSocketConnectionState.REGISTERED) { + // Send "bye" to WebSocket server. + send("{\"type\": \"bye\"}"); + state = WebSocketConnectionState.CONNECTED; + // Send http DELETE to http WebSocket server. + sendWSSMessage("DELETE", ""); + } + // Close WebSocket in CONNECTED or ERROR states only. + if (state == WebSocketConnectionState.CONNECTED || state == WebSocketConnectionState.ERROR) { + ws.disconnect(); + state = WebSocketConnectionState.CLOSED; + + // Wait for websocket close event to prevent websocket library from + // sending any pending messages to deleted looper thread. + if (waitForComplete) { + synchronized (closeEventLock) { + while (!closeEvent) { + try { + closeEventLock.wait(CLOSE_TIMEOUT); + break; + } catch (InterruptedException e) { + Log.e(TAG, "Wait error: " + e.toString()); + } + } + } + } + } + Log.d(TAG, "Disconnecting WebSocket done."); + } + + private void reportError(final String errorMessage) { + Log.e(TAG, errorMessage); + handler.post(new Runnable() { + @Override + public void run() { + if (state != WebSocketConnectionState.ERROR) { + state = WebSocketConnectionState.ERROR; + events.onWebSocketError(errorMessage); + } + } + }); + } + + // Asynchronously send POST/DELETE to WebSocket server. + private void sendWSSMessage(final String method, final String message) { + String postUrl = postServerUrl + "/" + roomID + "/" + clientID; + Log.d(TAG, "WS " + method + " : " + postUrl + " : " + message); + AsyncHttpURLConnection httpConnection = + new AsyncHttpURLConnection(method, postUrl, message, new AsyncHttpEvents() { + @Override + public void onHttpError(String errorMessage) { + reportError("WS " + method + " error: " + errorMessage); + } + + @Override + public void onHttpComplete(String response) {} + }); + httpConnection.send(); + } + + // Helper method for debugging purposes. Ensures that WebSocket method is + // called on a looper thread. + private void checkIfCalledOnValidThread() { + if (Thread.currentThread() != handler.getLooper().getThread()) { + throw new IllegalStateException("WebSocket method is not called on valid thread"); + } + } + + private class WebSocketObserver implements WebSocketConnectionObserver { + @Override + public void onOpen() { + Log.d(TAG, "WebSocket connection opened to: " + wsServerUrl); + handler.post(new Runnable() { + @Override + public void run() { + state = WebSocketConnectionState.CONNECTED; + // Check if we have pending register request. + if (roomID != null && clientID != null) { + register(roomID, clientID); + } + } + }); + } + + @Override + public void onClose(WebSocketCloseNotification code, String reason) { + Log.d(TAG, "WebSocket connection closed. Code: " + code + ". Reason: " + reason + ". State: " + + state); + synchronized (closeEventLock) { + closeEvent = true; + closeEventLock.notify(); + } + handler.post(new Runnable() { + @Override + public void run() { + if (state != WebSocketConnectionState.CLOSED) { + state = WebSocketConnectionState.CLOSED; + events.onWebSocketClose(); + } + } + }); + } + + @Override + public void onTextMessage(String payload) { + Log.d(TAG, "WSS->C: " + payload); + final String message = payload; + handler.post(new Runnable() { + @Override + public void run() { + if (state == WebSocketConnectionState.CONNECTED + || state == WebSocketConnectionState.REGISTERED) { + events.onWebSocketMessage(message); + } + } + }); + } + + @Override + public void onRawTextMessage(byte[] payload) {} + + @Override + public void onBinaryMessage(byte[] payload) {} + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketRTCClient.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketRTCClient.java new file mode 100644 index 0000000000..cbfdb21c91 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketRTCClient.java @@ -0,0 +1,427 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import android.os.Handler; +import android.os.HandlerThread; +import android.util.Log; +import androidx.annotation.Nullable; +import org.appspot.apprtc.RoomParametersFetcher.RoomParametersFetcherEvents; +import org.appspot.apprtc.WebSocketChannelClient.WebSocketChannelEvents; +import org.appspot.apprtc.WebSocketChannelClient.WebSocketConnectionState; +import org.appspot.apprtc.util.AsyncHttpURLConnection; +import org.appspot.apprtc.util.AsyncHttpURLConnection.AsyncHttpEvents; +import org.json.JSONArray; +import org.json.JSONException; +import org.json.JSONObject; +import org.webrtc.IceCandidate; +import org.webrtc.SessionDescription; + +/** + * Negotiates signaling for chatting with https://appr.tc "rooms". + * Uses the client<->server specifics of the apprtc AppEngine webapp. + * + * <p>To use: create an instance of this object (registering a message handler) and + * call connectToRoom(). Once room connection is established + * onConnectedToRoom() callback with room parameters is invoked. + * Messages to other party (with local Ice candidates and answer SDP) can + * be sent after WebSocket connection is established. + */ +public class WebSocketRTCClient implements AppRTCClient, WebSocketChannelEvents { + private static final String TAG = "WSRTCClient"; + private static final String ROOM_JOIN = "join"; + private static final String ROOM_MESSAGE = "message"; + private static final String ROOM_LEAVE = "leave"; + + private enum ConnectionState { NEW, CONNECTED, CLOSED, ERROR } + + private enum MessageType { MESSAGE, LEAVE } + + private final Handler handler; + private boolean initiator; + private SignalingEvents events; + private WebSocketChannelClient wsClient; + private ConnectionState roomState; + private RoomConnectionParameters connectionParameters; + private String messageUrl; + private String leaveUrl; + + public WebSocketRTCClient(SignalingEvents events) { + this.events = events; + roomState = ConnectionState.NEW; + final HandlerThread handlerThread = new HandlerThread(TAG); + handlerThread.start(); + handler = new Handler(handlerThread.getLooper()); + } + + // -------------------------------------------------------------------- + // AppRTCClient interface implementation. + // Asynchronously connect to an AppRTC room URL using supplied connection + // parameters, retrieves room parameters and connect to WebSocket server. + @Override + public void connectToRoom(RoomConnectionParameters connectionParameters) { + this.connectionParameters = connectionParameters; + handler.post(new Runnable() { + @Override + public void run() { + connectToRoomInternal(); + } + }); + } + + @Override + public void disconnectFromRoom() { + handler.post(new Runnable() { + @Override + public void run() { + disconnectFromRoomInternal(); + handler.getLooper().quit(); + } + }); + } + + // Connects to room - function runs on a local looper thread. + private void connectToRoomInternal() { + String connectionUrl = getConnectionUrl(connectionParameters); + Log.d(TAG, "Connect to room: " + connectionUrl); + roomState = ConnectionState.NEW; + wsClient = new WebSocketChannelClient(handler, this); + + RoomParametersFetcherEvents callbacks = new RoomParametersFetcherEvents() { + @Override + public void onSignalingParametersReady(final SignalingParameters params) { + WebSocketRTCClient.this.handler.post(new Runnable() { + @Override + public void run() { + WebSocketRTCClient.this.signalingParametersReady(params); + } + }); + } + + @Override + public void onSignalingParametersError(String description) { + WebSocketRTCClient.this.reportError(description); + } + }; + + new RoomParametersFetcher(connectionUrl, null, callbacks).makeRequest(); + } + + // Disconnect from room and send bye messages - runs on a local looper thread. + private void disconnectFromRoomInternal() { + Log.d(TAG, "Disconnect. Room state: " + roomState); + if (roomState == ConnectionState.CONNECTED) { + Log.d(TAG, "Closing room."); + sendPostMessage(MessageType.LEAVE, leaveUrl, null); + } + roomState = ConnectionState.CLOSED; + if (wsClient != null) { + wsClient.disconnect(true); + } + } + + // Helper functions to get connection, post message and leave message URLs + private String getConnectionUrl(RoomConnectionParameters connectionParameters) { + return connectionParameters.roomUrl + "/" + ROOM_JOIN + "/" + connectionParameters.roomId + + getQueryString(connectionParameters); + } + + private String getMessageUrl( + RoomConnectionParameters connectionParameters, SignalingParameters signalingParameters) { + return connectionParameters.roomUrl + "/" + ROOM_MESSAGE + "/" + connectionParameters.roomId + + "/" + signalingParameters.clientId + getQueryString(connectionParameters); + } + + private String getLeaveUrl( + RoomConnectionParameters connectionParameters, SignalingParameters signalingParameters) { + return connectionParameters.roomUrl + "/" + ROOM_LEAVE + "/" + connectionParameters.roomId + "/" + + signalingParameters.clientId + getQueryString(connectionParameters); + } + + private String getQueryString(RoomConnectionParameters connectionParameters) { + if (connectionParameters.urlParameters != null) { + return "?" + connectionParameters.urlParameters; + } else { + return ""; + } + } + + // Callback issued when room parameters are extracted. Runs on local + // looper thread. + private void signalingParametersReady(final SignalingParameters signalingParameters) { + Log.d(TAG, "Room connection completed."); + if (connectionParameters.loopback + && (!signalingParameters.initiator || signalingParameters.offerSdp != null)) { + reportError("Loopback room is busy."); + return; + } + if (!connectionParameters.loopback && !signalingParameters.initiator + && signalingParameters.offerSdp == null) { + Log.w(TAG, "No offer SDP in room response."); + } + initiator = signalingParameters.initiator; + messageUrl = getMessageUrl(connectionParameters, signalingParameters); + leaveUrl = getLeaveUrl(connectionParameters, signalingParameters); + Log.d(TAG, "Message URL: " + messageUrl); + Log.d(TAG, "Leave URL: " + leaveUrl); + roomState = ConnectionState.CONNECTED; + + // Fire connection and signaling parameters events. + events.onConnectedToRoom(signalingParameters); + + // Connect and register WebSocket client. + wsClient.connect(signalingParameters.wssUrl, signalingParameters.wssPostUrl); + wsClient.register(connectionParameters.roomId, signalingParameters.clientId); + } + + // Send local offer SDP to the other participant. + @Override + public void sendOfferSdp(final SessionDescription sdp) { + handler.post(new Runnable() { + @Override + public void run() { + if (roomState != ConnectionState.CONNECTED) { + reportError("Sending offer SDP in non connected state."); + return; + } + JSONObject json = new JSONObject(); + jsonPut(json, "sdp", sdp.description); + jsonPut(json, "type", "offer"); + sendPostMessage(MessageType.MESSAGE, messageUrl, json.toString()); + if (connectionParameters.loopback) { + // In loopback mode rename this offer to answer and route it back. + SessionDescription sdpAnswer = new SessionDescription( + SessionDescription.Type.fromCanonicalForm("answer"), sdp.description); + events.onRemoteDescription(sdpAnswer); + } + } + }); + } + + // Send local answer SDP to the other participant. + @Override + public void sendAnswerSdp(final SessionDescription sdp) { + handler.post(new Runnable() { + @Override + public void run() { + if (connectionParameters.loopback) { + Log.e(TAG, "Sending answer in loopback mode."); + return; + } + JSONObject json = new JSONObject(); + jsonPut(json, "sdp", sdp.description); + jsonPut(json, "type", "answer"); + wsClient.send(json.toString()); + } + }); + } + + // Send Ice candidate to the other participant. + @Override + public void sendLocalIceCandidate(final IceCandidate candidate) { + handler.post(new Runnable() { + @Override + public void run() { + JSONObject json = new JSONObject(); + jsonPut(json, "type", "candidate"); + jsonPut(json, "label", candidate.sdpMLineIndex); + jsonPut(json, "id", candidate.sdpMid); + jsonPut(json, "candidate", candidate.sdp); + if (initiator) { + // Call initiator sends ice candidates to GAE server. + if (roomState != ConnectionState.CONNECTED) { + reportError("Sending ICE candidate in non connected state."); + return; + } + sendPostMessage(MessageType.MESSAGE, messageUrl, json.toString()); + if (connectionParameters.loopback) { + events.onRemoteIceCandidate(candidate); + } + } else { + // Call receiver sends ice candidates to websocket server. + wsClient.send(json.toString()); + } + } + }); + } + + // Send removed Ice candidates to the other participant. + @Override + public void sendLocalIceCandidateRemovals(final IceCandidate[] candidates) { + handler.post(new Runnable() { + @Override + public void run() { + JSONObject json = new JSONObject(); + jsonPut(json, "type", "remove-candidates"); + JSONArray jsonArray = new JSONArray(); + for (final IceCandidate candidate : candidates) { + jsonArray.put(toJsonCandidate(candidate)); + } + jsonPut(json, "candidates", jsonArray); + if (initiator) { + // Call initiator sends ice candidates to GAE server. + if (roomState != ConnectionState.CONNECTED) { + reportError("Sending ICE candidate removals in non connected state."); + return; + } + sendPostMessage(MessageType.MESSAGE, messageUrl, json.toString()); + if (connectionParameters.loopback) { + events.onRemoteIceCandidatesRemoved(candidates); + } + } else { + // Call receiver sends ice candidates to websocket server. + wsClient.send(json.toString()); + } + } + }); + } + + // -------------------------------------------------------------------- + // WebSocketChannelEvents interface implementation. + // All events are called by WebSocketChannelClient on a local looper thread + // (passed to WebSocket client constructor). + @Override + public void onWebSocketMessage(final String msg) { + if (wsClient.getState() != WebSocketConnectionState.REGISTERED) { + Log.e(TAG, "Got WebSocket message in non registered state."); + return; + } + try { + JSONObject json = new JSONObject(msg); + String msgText = json.getString("msg"); + String errorText = json.optString("error"); + if (msgText.length() > 0) { + json = new JSONObject(msgText); + String type = json.optString("type"); + if (type.equals("candidate")) { + events.onRemoteIceCandidate(toJavaCandidate(json)); + } else if (type.equals("remove-candidates")) { + JSONArray candidateArray = json.getJSONArray("candidates"); + IceCandidate[] candidates = new IceCandidate[candidateArray.length()]; + for (int i = 0; i < candidateArray.length(); ++i) { + candidates[i] = toJavaCandidate(candidateArray.getJSONObject(i)); + } + events.onRemoteIceCandidatesRemoved(candidates); + } else if (type.equals("answer")) { + if (initiator) { + SessionDescription sdp = new SessionDescription( + SessionDescription.Type.fromCanonicalForm(type), json.getString("sdp")); + events.onRemoteDescription(sdp); + } else { + reportError("Received answer for call initiator: " + msg); + } + } else if (type.equals("offer")) { + if (!initiator) { + SessionDescription sdp = new SessionDescription( + SessionDescription.Type.fromCanonicalForm(type), json.getString("sdp")); + events.onRemoteDescription(sdp); + } else { + reportError("Received offer for call receiver: " + msg); + } + } else if (type.equals("bye")) { + events.onChannelClose(); + } else { + reportError("Unexpected WebSocket message: " + msg); + } + } else { + if (errorText != null && errorText.length() > 0) { + reportError("WebSocket error message: " + errorText); + } else { + reportError("Unexpected WebSocket message: " + msg); + } + } + } catch (JSONException e) { + reportError("WebSocket message JSON parsing error: " + e.toString()); + } + } + + @Override + public void onWebSocketClose() { + events.onChannelClose(); + } + + @Override + public void onWebSocketError(String description) { + reportError("WebSocket error: " + description); + } + + // -------------------------------------------------------------------- + // Helper functions. + private void reportError(final String errorMessage) { + Log.e(TAG, errorMessage); + handler.post(new Runnable() { + @Override + public void run() { + if (roomState != ConnectionState.ERROR) { + roomState = ConnectionState.ERROR; + events.onChannelError(errorMessage); + } + } + }); + } + + // Put a `key`->`value` mapping in `json`. + private static void jsonPut(JSONObject json, String key, Object value) { + try { + json.put(key, value); + } catch (JSONException e) { + throw new RuntimeException(e); + } + } + + // Send SDP or ICE candidate to a room server. + private void sendPostMessage( + final MessageType messageType, final String url, @Nullable final String message) { + String logInfo = url; + if (message != null) { + logInfo += ". Message: " + message; + } + Log.d(TAG, "C->GAE: " + logInfo); + AsyncHttpURLConnection httpConnection = + new AsyncHttpURLConnection("POST", url, message, new AsyncHttpEvents() { + @Override + public void onHttpError(String errorMessage) { + reportError("GAE POST error: " + errorMessage); + } + + @Override + public void onHttpComplete(String response) { + if (messageType == MessageType.MESSAGE) { + try { + JSONObject roomJson = new JSONObject(response); + String result = roomJson.getString("result"); + if (!result.equals("SUCCESS")) { + reportError("GAE POST error: " + result); + } + } catch (JSONException e) { + reportError("GAE POST JSON error: " + e.toString()); + } + } + } + }); + httpConnection.send(); + } + + // Converts a Java candidate to a JSONObject. + private JSONObject toJsonCandidate(final IceCandidate candidate) { + JSONObject json = new JSONObject(); + jsonPut(json, "label", candidate.sdpMLineIndex); + jsonPut(json, "id", candidate.sdpMid); + jsonPut(json, "candidate", candidate.sdp); + return json; + } + + // Converts a JSON candidate to a Java object. + IceCandidate toJavaCandidate(JSONObject json) throws JSONException { + return new IceCandidate( + json.getString("id"), json.getInt("label"), json.getString("candidate")); + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/util/AppRTCUtils.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/util/AppRTCUtils.java new file mode 100644 index 0000000000..ee7f8c0416 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/util/AppRTCUtils.java @@ -0,0 +1,47 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc.util; + +import android.os.Build; +import android.util.Log; + +/** + * AppRTCUtils provides helper functions for managing thread safety. + */ +public final class AppRTCUtils { + private AppRTCUtils() {} + + /** Helper method which throws an exception when an assertion has failed. */ + public static void assertIsTrue(boolean condition) { + if (!condition) { + throw new AssertionError("Expected condition to be true"); + } + } + + /** Helper method for building a string of thread information.*/ + public static String getThreadInfo() { + return "@[name=" + Thread.currentThread().getName() + ", id=" + Thread.currentThread().getId() + + "]"; + } + + /** Information about the current build, taken from system properties. */ + public static void logDeviceInfo(String tag) { + Log.d(tag, "Android SDK: " + Build.VERSION.SDK_INT + ", " + + "Release: " + Build.VERSION.RELEASE + ", " + + "Brand: " + Build.BRAND + ", " + + "Device: " + Build.DEVICE + ", " + + "Id: " + Build.ID + ", " + + "Hardware: " + Build.HARDWARE + ", " + + "Manufacturer: " + Build.MANUFACTURER + ", " + + "Model: " + Build.MODEL + ", " + + "Product: " + Build.PRODUCT); + } +} diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java new file mode 100644 index 0000000000..93028ae783 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java @@ -0,0 +1,115 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc.util; + +import java.io.IOException; +import java.io.InputStream; +import java.io.OutputStream; +import java.net.HttpURLConnection; +import java.net.SocketTimeoutException; +import java.net.URL; +import java.util.Scanner; + +/** + * Asynchronous http requests implementation. + */ +public class AsyncHttpURLConnection { + private static final int HTTP_TIMEOUT_MS = 8000; + private static final String HTTP_ORIGIN = "https://appr.tc"; + private final String method; + private final String url; + private final String message; + private final AsyncHttpEvents events; + private String contentType; + + /** + * Http requests callbacks. + */ + public interface AsyncHttpEvents { + void onHttpError(String errorMessage); + void onHttpComplete(String response); + } + + public AsyncHttpURLConnection(String method, String url, String message, AsyncHttpEvents events) { + this.method = method; + this.url = url; + this.message = message; + this.events = events; + } + + public void setContentType(String contentType) { + this.contentType = contentType; + } + + public void send() { + new Thread(this ::sendHttpMessage).start(); + } + + @SuppressWarnings("UseNetworkAnnotations") + private void sendHttpMessage() { + try { + HttpURLConnection connection = (HttpURLConnection) new URL(url).openConnection(); + byte[] postData = new byte[0]; + if (message != null) { + postData = message.getBytes("UTF-8"); + } + connection.setRequestMethod(method); + connection.setUseCaches(false); + connection.setDoInput(true); + connection.setConnectTimeout(HTTP_TIMEOUT_MS); + connection.setReadTimeout(HTTP_TIMEOUT_MS); + // TODO(glaznev) - query request origin from pref_room_server_url_key preferences. + connection.addRequestProperty("origin", HTTP_ORIGIN); + boolean doOutput = false; + if (method.equals("POST")) { + doOutput = true; + connection.setDoOutput(true); + connection.setFixedLengthStreamingMode(postData.length); + } + if (contentType == null) { + connection.setRequestProperty("Content-Type", "text/plain; charset=utf-8"); + } else { + connection.setRequestProperty("Content-Type", contentType); + } + + // Send POST request. + if (doOutput && postData.length > 0) { + OutputStream outStream = connection.getOutputStream(); + outStream.write(postData); + outStream.close(); + } + + // Get response. + int responseCode = connection.getResponseCode(); + if (responseCode != 200) { + events.onHttpError("Non-200 response to " + method + " to URL: " + url + " : " + + connection.getHeaderField(null)); + connection.disconnect(); + return; + } + InputStream responseStream = connection.getInputStream(); + String response = drainStream(responseStream); + responseStream.close(); + connection.disconnect(); + events.onHttpComplete(response); + } catch (SocketTimeoutException e) { + events.onHttpError("HTTP " + method + " to " + url + " timeout"); + } catch (IOException e) { + events.onHttpError("HTTP " + method + " to " + url + " error: " + e.getMessage()); + } + } + + // Return the contents of an InputStream as a String. + private static String drainStream(InputStream in) { + Scanner s = new Scanner(in, "UTF-8").useDelimiter("\\A"); + return s.hasNext() ? s.next() : ""; + } +} diff --git a/third_party/libwebrtc/examples/androidapp/start_loopback_stubbed_camera_saved_video_out.py b/third_party/libwebrtc/examples/androidapp/start_loopback_stubbed_camera_saved_video_out.py new file mode 100644 index 0000000000..b1cf84611f --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/start_loopback_stubbed_camera_saved_video_out.py @@ -0,0 +1,127 @@ +# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. + +from optparse import OptionParser +import random +import string +import subprocess +import sys +import time + +from com.android.monkeyrunner import MonkeyRunner, MonkeyDevice + + +def main(): + parser = OptionParser() + + parser.add_option('--devname', dest='devname', help='The device id') + + parser.add_option( + '--videooutsave', + dest='videooutsave', + help='The path where to save the video out file on local computer') + + parser.add_option('--videoout', + dest='videoout', + help='The path where to put the video out file') + + parser.add_option('--videoout_width', + dest='videoout_width', + type='int', + help='The width for the video out file') + + parser.add_option('--videoout_height', + dest='videoout_height', + type='int', + help='The height for the video out file') + + parser.add_option( + '--videoin', + dest='videoin', + help='The path where to read input file instead of camera') + + parser.add_option('--call_length', + dest='call_length', + type='int', + help='The length of the call') + + (options, args) = parser.parse_args() + + print(options, args) + + devname = options.devname + + videoin = options.videoin + + videoout = options.videoout + videoout_width = options.videoout_width + videoout_height = options.videoout_height + + videooutsave = options.videooutsave + + call_length = options.call_length or 10 + + room = ''.join( + random.choice(string.ascii_letters + string.digits) for _ in range(8)) + + # Delete output video file. + if videoout: + subprocess.check_call( + ['adb', '-s', devname, 'shell', 'rm', '-f', videoout]) + + device = MonkeyRunner.waitForConnection(2, devname) + + extras = { + 'org.appspot.apprtc.USE_VALUES_FROM_INTENT': True, + 'org.appspot.apprtc.AUDIOCODEC': 'OPUS', + 'org.appspot.apprtc.LOOPBACK': True, + 'org.appspot.apprtc.VIDEOCODEC': 'VP8', + 'org.appspot.apprtc.CAPTURETOTEXTURE': False, + 'org.appspot.apprtc.CAMERA2': False, + 'org.appspot.apprtc.ROOMID': room + } + + if videoin: + extras.update({'org.appspot.apprtc.VIDEO_FILE_AS_CAMERA': videoin}) + + if videoout: + extras.update({ + 'org.appspot.apprtc.SAVE_REMOTE_VIDEO_TO_FILE': + videoout, + 'org.appspot.apprtc.SAVE_REMOTE_VIDEO_TO_FILE_WIDTH': + videoout_width, + 'org.appspot.apprtc.SAVE_REMOTE_VIDEO_TO_FILE_HEIGHT': + videoout_height + }) + + print extras + + device.startActivity(data='https://appr.tc', + action='android.intent.action.VIEW', + component='org.appspot.apprtc/.ConnectActivity', + extras=extras) + + print 'Running a call for %d seconds' % call_length + for _ in xrange(call_length): + sys.stdout.write('.') + sys.stdout.flush() + time.sleep(1) + print '\nEnding call.' + + # Press back to end the call. Will end on both sides. + device.press('KEYCODE_BACK', MonkeyDevice.DOWN_AND_UP) + + if videooutsave: + time.sleep(2) + + subprocess.check_call( + ['adb', '-s', devname, 'pull', videoout, videooutsave]) + + +if __name__ == '__main__': + main() diff --git a/third_party/libwebrtc/examples/androidapp/third_party/autobanh/BUILD.gn b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/BUILD.gn new file mode 100644 index 0000000000..b671239bae --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/BUILD.gn @@ -0,0 +1,15 @@ +# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. + +if (is_android) { + import("//build/config/android/rules.gni") + + android_java_prebuilt("autobanh_java") { + jar_path = "lib/autobanh.jar" + } +} diff --git a/third_party/libwebrtc/examples/androidapp/third_party/autobanh/LICENSE b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/LICENSE new file mode 100644 index 0000000000..f433b1a53f --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/LICENSE @@ -0,0 +1,177 @@ + + Apache License + Version 2.0, January 2004 + http://www.apache.org/licenses/ + + TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION + + 1. Definitions. + + "License" shall mean the terms and conditions for use, reproduction, + and distribution as defined by Sections 1 through 9 of this document. + + "Licensor" shall mean the copyright owner or entity authorized by + the copyright owner that is granting the License. + + "Legal Entity" shall mean the union of the acting entity and all + other entities that control, are controlled by, or are under common + control with that entity. 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For the purposes + of this License, Derivative Works shall not include works that remain + separable from, or merely link (or bind by name) to the interfaces of, + the Work and Derivative Works thereof. + + "Contribution" shall mean any work of authorship, including + the original version of the Work and any modifications or additions + to that Work or Derivative Works thereof, that is intentionally + submitted to Licensor for inclusion in the Work by the copyright owner + or by an individual or Legal Entity authorized to submit on behalf of + the copyright owner. For the purposes of this definition, "submitted" + means any form of electronic, verbal, or written communication sent + to the Licensor or its representatives, including but not limited to + communication on electronic mailing lists, source code control systems, + and issue tracking systems that are managed by, or on behalf of, the + Licensor for the purpose of discussing and improving the Work, but + excluding communication that is conspicuously marked or otherwise + designated in writing by the copyright owner as "Not a Contribution." + + "Contributor" shall mean Licensor and any individual or Legal Entity + on behalf of whom a Contribution has been received by Licensor and + subsequently incorporated within the Work. + + 2. Grant of Copyright License. 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If You + institute patent litigation against any entity (including a + cross-claim or counterclaim in a lawsuit) alleging that the Work + or a Contribution incorporated within the Work constitutes direct + or contributory patent infringement, then any patent licenses + granted to You under this License for that Work shall terminate + as of the date such litigation is filed. + + 4. Redistribution. You may reproduce and distribute copies of the + Work or Derivative Works thereof in any medium, with or without + modifications, and in Source or Object form, provided that You + meet the following conditions: + + (a) You must give any other recipients of the Work or + Derivative Works a copy of this License; and + + (b) You must cause any modified files to carry prominent notices + stating that You changed the files; and + + (c) You must retain, in the Source form of any Derivative Works + that You distribute, all copyright, patent, trademark, and + attribution notices from the Source form of the Work, + excluding those notices that do not pertain to any part of + the Derivative Works; and + + (d) If the Work includes a "NOTICE" text file as part of its + distribution, then any Derivative Works that You distribute must + include a readable copy of the attribution notices contained + within such NOTICE file, excluding those notices that do not + pertain to any part of the Derivative Works, in at least one + of the following places: within a NOTICE text file distributed + as part of the Derivative Works; within the Source form or + documentation, if provided along with the Derivative Works; or, + within a display generated by the Derivative Works, if and + wherever such third-party notices normally appear. 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In no event and under no legal theory, + whether in tort (including negligence), contract, or otherwise, + unless required by applicable law (such as deliberate and grossly + negligent acts) or agreed to in writing, shall any Contributor be + liable to You for damages, including any direct, indirect, special, + incidental, or consequential damages of any character arising as a + result of this License or out of the use or inability to use the + Work (including but not limited to damages for loss of goodwill, + work stoppage, computer failure or malfunction, or any and all + other commercial damages or losses), even if such Contributor + has been advised of the possibility of such damages. + + 9. Accepting Warranty or Additional Liability. While redistributing + the Work or Derivative Works thereof, You may choose to offer, + and charge a fee for, acceptance of support, warranty, indemnity, + or other liability obligations and/or rights consistent with this + License. However, in accepting such obligations, You may act only + on Your own behalf and on Your sole responsibility, not on behalf + of any other Contributor, and only if You agree to indemnify, + defend, and hold each Contributor harmless for any liability + incurred by, or claims asserted against, such Contributor by reason + of your accepting any such warranty or additional liability. + + END OF TERMS AND CONDITIONS diff --git a/third_party/libwebrtc/examples/androidapp/third_party/autobanh/LICENSE.md b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/LICENSE.md new file mode 100644 index 0000000000..2079e90d6b --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/LICENSE.md @@ -0,0 +1,21 @@ +The MIT License (MIT) + +Copyright (c) 2014 Cameron Lowell Palmer + +Permission is hereby granted, free of charge, to any person obtaining a copy +of this software and associated documentation files (the "Software"), to deal +in the Software without restriction, including without limitation the rights +to use, copy, modify, merge, publish, distribute, sublicense, and/or sell +copies of the Software, and to permit persons to whom the Software is +furnished to do so, subject to the following conditions: + +The above copyright notice and this permission notice shall be included in all +copies or substantial portions of the Software. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. diff --git a/third_party/libwebrtc/examples/androidapp/third_party/autobanh/NOTICE b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/NOTICE new file mode 100644 index 0000000000..91ed7dfe0e --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/NOTICE @@ -0,0 +1,3 @@ +AutobahnAndroid +Copyright 2011,2012 Tavendo GmbH. Licensed under Apache 2.0 +This product includes software developed at Tavendo GmbH http://www.tavendo.de diff --git a/third_party/libwebrtc/examples/androidapp/third_party/autobanh/lib/autobanh.jar b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/lib/autobanh.jar Binary files differnew file mode 100644 index 0000000000..5a10b7f3f1 --- /dev/null +++ b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/lib/autobanh.jar diff --git a/third_party/libwebrtc/examples/androidjunit/OWNERS b/third_party/libwebrtc/examples/androidjunit/OWNERS new file mode 100644 index 0000000000..cf092a316a --- /dev/null +++ b/third_party/libwebrtc/examples/androidjunit/OWNERS @@ -0,0 +1 @@ +xalep@webrtc.org diff --git a/third_party/libwebrtc/examples/androidjunit/README b/third_party/libwebrtc/examples/androidjunit/README new file mode 100644 index 0000000000..03902a779c --- /dev/null +++ b/third_party/libwebrtc/examples/androidjunit/README @@ -0,0 +1,8 @@ +This directory contains example JUnit tests for Android AppRTCMobile. +Many of these test utilize Robolectric to mock Android classes. + +To compile: +ninja -C out/Debug android_examples_junit_tests + +To run: +out/Debug/bin/run_android_examples_junit_tests diff --git a/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java b/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java new file mode 100644 index 0000000000..3060bd7a56 --- /dev/null +++ b/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java @@ -0,0 +1,268 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import static org.junit.Assert.assertEquals; +import static org.junit.Assert.assertNotNull; +import static org.junit.Assert.assertNull; +import static org.mockito.Mockito.mock; +import static org.mockito.Mockito.never; +import static org.mockito.Mockito.times; +import static org.mockito.Mockito.verify; +import static org.mockito.Mockito.when; + +import android.bluetooth.BluetoothAdapter; +import android.bluetooth.BluetoothDevice; +import android.bluetooth.BluetoothHeadset; +import android.bluetooth.BluetoothProfile; +import android.content.BroadcastReceiver; +import android.content.Context; +import android.content.Intent; +import android.content.IntentFilter; +import android.media.AudioManager; +import android.util.Log; +import androidx.test.core.app.ApplicationProvider; +import java.util.ArrayList; +import java.util.List; +import org.appspot.apprtc.AppRTCBluetoothManager.State; +import org.chromium.testing.local.LocalRobolectricTestRunner; +import org.junit.Before; +import org.junit.Test; +import org.junit.runner.RunWith; +import org.robolectric.annotation.Config; +import org.robolectric.shadows.ShadowLog; + +/** + * Verifies basic behavior of the AppRTCBluetoothManager class. + * Note that the test object uses an AppRTCAudioManager (injected in ctor), + * but a mocked version is used instead. Hence, the parts "driven" by the AppRTC + * audio manager are not included in this test. + */ +@RunWith(LocalRobolectricTestRunner.class) +@Config(manifest = Config.NONE) +public class BluetoothManagerTest { + private static final String TAG = "BluetoothManagerTest"; + private static final String BLUETOOTH_TEST_DEVICE_NAME = "BluetoothTestDevice"; + + private BroadcastReceiver bluetoothHeadsetStateReceiver; + private BluetoothProfile.ServiceListener bluetoothServiceListener; + private BluetoothHeadset mockedBluetoothHeadset; + private BluetoothDevice mockedBluetoothDevice; + private List<BluetoothDevice> mockedBluetoothDeviceList; + private AppRTCBluetoothManager bluetoothManager; + private AppRTCAudioManager mockedAppRtcAudioManager; + private AudioManager mockedAudioManager; + private Context context; + + @Before + public void setUp() { + ShadowLog.stream = System.out; + context = ApplicationProvider.getApplicationContext(); + mockedAppRtcAudioManager = mock(AppRTCAudioManager.class); + mockedAudioManager = mock(AudioManager.class); + mockedBluetoothHeadset = mock(BluetoothHeadset.class); + mockedBluetoothDevice = mock(BluetoothDevice.class); + mockedBluetoothDeviceList = new ArrayList<BluetoothDevice>(); + + // Simulate that bluetooth SCO audio is available by default. + when(mockedAudioManager.isBluetoothScoAvailableOffCall()).thenReturn(true); + + // Create the test object and override protected methods for this test. + bluetoothManager = new AppRTCBluetoothManager(context, mockedAppRtcAudioManager) { + @Override + protected AudioManager getAudioManager(Context context) { + Log.d(TAG, "getAudioManager"); + return mockedAudioManager; + } + + @Override + protected void registerReceiver(BroadcastReceiver receiver, IntentFilter filter) { + Log.d(TAG, "registerReceiver"); + if (filter.hasAction(BluetoothHeadset.ACTION_CONNECTION_STATE_CHANGED) + && filter.hasAction(BluetoothHeadset.ACTION_AUDIO_STATE_CHANGED)) { + // Gives access to the real broadcast receiver so the test can use it. + bluetoothHeadsetStateReceiver = receiver; + } + } + + @Override + protected void unregisterReceiver(BroadcastReceiver receiver) { + Log.d(TAG, "unregisterReceiver"); + if (receiver == bluetoothHeadsetStateReceiver) { + bluetoothHeadsetStateReceiver = null; + } + } + + @Override + protected boolean getBluetoothProfileProxy( + Context context, BluetoothProfile.ServiceListener listener, int profile) { + Log.d(TAG, "getBluetoothProfileProxy"); + if (profile == BluetoothProfile.HEADSET) { + // Allows the test to access the real Bluetooth service listener object. + bluetoothServiceListener = listener; + } + return true; + } + + @Override + protected boolean hasPermission(Context context, String permission) { + Log.d(TAG, "hasPermission(" + permission + ")"); + // Ensure that the client asks for Bluetooth permission. + return android.Manifest.permission.BLUETOOTH.equals(permission); + } + + @Override + protected void logBluetoothAdapterInfo(BluetoothAdapter localAdapter) { + // Do nothing in tests. No need to mock BluetoothAdapter. + } + }; + } + + // Verify that Bluetooth service listener for headset profile is properly initialized. + @Test + public void testBluetoothServiceListenerInitialized() { + bluetoothManager.start(); + assertNotNull(bluetoothServiceListener); + verify(mockedAppRtcAudioManager, never()).updateAudioDeviceState(); + } + + // Verify that broadcast receivers for Bluetooth SCO audio state and Bluetooth headset state + // are properly registered and unregistered. + @Test + public void testBluetoothBroadcastReceiversAreRegistered() { + bluetoothManager.start(); + assertNotNull(bluetoothHeadsetStateReceiver); + bluetoothManager.stop(); + assertNull(bluetoothHeadsetStateReceiver); + } + + // Verify that the Bluetooth manager starts and stops with correct states. + @Test + public void testBluetoothDefaultStartStopStates() { + bluetoothManager.start(); + assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE); + bluetoothManager.stop(); + assertEquals(bluetoothManager.getState(), State.UNINITIALIZED); + } + + // Verify correct state after receiving BluetoothServiceListener.onServiceConnected() + // when no BT device is enabled. + @Test + public void testBluetoothServiceListenerConnectedWithNoHeadset() { + bluetoothManager.start(); + assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE); + simulateBluetoothServiceConnectedWithNoConnectedHeadset(); + verify(mockedAppRtcAudioManager, times(1)).updateAudioDeviceState(); + assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE); + } + + // Verify correct state after receiving BluetoothServiceListener.onServiceConnected() + // when one emulated (test) BT device is enabled. Android does not support more than + // one connected BT headset. + @Test + public void testBluetoothServiceListenerConnectedWithHeadset() { + bluetoothManager.start(); + assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE); + simulateBluetoothServiceConnectedWithConnectedHeadset(); + verify(mockedAppRtcAudioManager, times(1)).updateAudioDeviceState(); + assertEquals(bluetoothManager.getState(), State.HEADSET_AVAILABLE); + } + + // Verify correct state after receiving BluetoothProfile.ServiceListener.onServiceDisconnected(). + @Test + public void testBluetoothServiceListenerDisconnected() { + bluetoothManager.start(); + assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE); + simulateBluetoothServiceDisconnected(); + verify(mockedAppRtcAudioManager, times(1)).updateAudioDeviceState(); + assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE); + } + + // Verify correct state after BluetoothServiceListener.onServiceConnected() and + // the intent indicating that the headset is actually connected. Both these callbacks + // results in calls to updateAudioDeviceState() on the AppRTC audio manager. + // No BT SCO is enabled here to keep the test limited. + @Test + public void testBluetoothHeadsetConnected() { + bluetoothManager.start(); + assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE); + simulateBluetoothServiceConnectedWithConnectedHeadset(); + simulateBluetoothHeadsetConnected(); + verify(mockedAppRtcAudioManager, times(2)).updateAudioDeviceState(); + assertEquals(bluetoothManager.getState(), State.HEADSET_AVAILABLE); + } + + // Verify correct state sequence for a case when a BT headset is available, + // followed by BT SCO audio being enabled and then stopped. + @Test + public void testBluetoothScoAudioStartAndStop() { + bluetoothManager.start(); + assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE); + simulateBluetoothServiceConnectedWithConnectedHeadset(); + assertEquals(bluetoothManager.getState(), State.HEADSET_AVAILABLE); + bluetoothManager.startScoAudio(); + assertEquals(bluetoothManager.getState(), State.SCO_CONNECTING); + simulateBluetoothScoConnectionConnected(); + assertEquals(bluetoothManager.getState(), State.SCO_CONNECTED); + bluetoothManager.stopScoAudio(); + simulateBluetoothScoConnectionDisconnected(); + assertEquals(bluetoothManager.getState(), State.SCO_DISCONNECTING); + bluetoothManager.stop(); + assertEquals(bluetoothManager.getState(), State.UNINITIALIZED); + verify(mockedAppRtcAudioManager, times(3)).updateAudioDeviceState(); + } + + /** + * Private helper methods. + */ + private void simulateBluetoothServiceConnectedWithNoConnectedHeadset() { + mockedBluetoothDeviceList.clear(); + when(mockedBluetoothHeadset.getConnectedDevices()).thenReturn(mockedBluetoothDeviceList); + bluetoothServiceListener.onServiceConnected(BluetoothProfile.HEADSET, mockedBluetoothHeadset); + // In real life, the AppRTC audio manager makes this call. + bluetoothManager.updateDevice(); + } + + private void simulateBluetoothServiceConnectedWithConnectedHeadset() { + mockedBluetoothDeviceList.clear(); + mockedBluetoothDeviceList.add(mockedBluetoothDevice); + when(mockedBluetoothHeadset.getConnectedDevices()).thenReturn(mockedBluetoothDeviceList); + when(mockedBluetoothDevice.getName()).thenReturn(BLUETOOTH_TEST_DEVICE_NAME); + bluetoothServiceListener.onServiceConnected(BluetoothProfile.HEADSET, mockedBluetoothHeadset); + // In real life, the AppRTC audio manager makes this call. + bluetoothManager.updateDevice(); + } + + private void simulateBluetoothServiceDisconnected() { + bluetoothServiceListener.onServiceDisconnected(BluetoothProfile.HEADSET); + } + + private void simulateBluetoothHeadsetConnected() { + Intent intent = new Intent(); + intent.setAction(BluetoothHeadset.ACTION_CONNECTION_STATE_CHANGED); + intent.putExtra(BluetoothHeadset.EXTRA_STATE, BluetoothHeadset.STATE_CONNECTED); + bluetoothHeadsetStateReceiver.onReceive(context, intent); + } + + private void simulateBluetoothScoConnectionConnected() { + Intent intent = new Intent(); + intent.setAction(BluetoothHeadset.ACTION_AUDIO_STATE_CHANGED); + intent.putExtra(BluetoothHeadset.EXTRA_STATE, BluetoothHeadset.STATE_AUDIO_CONNECTED); + bluetoothHeadsetStateReceiver.onReceive(context, intent); + } + + private void simulateBluetoothScoConnectionDisconnected() { + Intent intent = new Intent(); + intent.setAction(BluetoothHeadset.ACTION_AUDIO_STATE_CHANGED); + intent.putExtra(BluetoothHeadset.EXTRA_STATE, BluetoothHeadset.STATE_AUDIO_DISCONNECTED); + bluetoothHeadsetStateReceiver.onReceive(context, intent); + } +} diff --git a/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java b/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java new file mode 100644 index 0000000000..2da8164ec7 --- /dev/null +++ b/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java @@ -0,0 +1,155 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import static org.junit.Assert.assertFalse; +import static org.junit.Assert.assertTrue; +import static org.junit.Assert.fail; +import static org.mockito.ArgumentMatchers.any; +import static org.mockito.ArgumentMatchers.isNotNull; +import static org.mockito.Mockito.mock; +import static org.mockito.Mockito.timeout; +import static org.mockito.Mockito.verify; +import static org.mockito.Mockito.verifyNoMoreInteractions; + +import org.chromium.testing.local.LocalRobolectricTestRunner; +import org.junit.Before; +import org.junit.Test; +import org.junit.runner.RunWith; +import org.robolectric.annotation.Config; +import org.robolectric.shadows.ShadowLog; +import org.webrtc.IceCandidate; +import org.webrtc.SessionDescription; + +/** + * Test for DirectRTCClient. Test is very simple and only tests the overall sanity of the class + * behaviour. + */ +@RunWith(LocalRobolectricTestRunner.class) +@Config(manifest = Config.NONE) +public class DirectRTCClientTest { + private static final String ROOM_URL = ""; + private static final boolean LOOPBACK = false; + + private static final String DUMMY_SDP_MID = "sdpMid"; + private static final String DUMMY_SDP = "sdp"; + + public static final int SERVER_WAIT = 100; + public static final int NETWORK_TIMEOUT = 1000; + + private DirectRTCClient client; + private DirectRTCClient server; + + AppRTCClient.SignalingEvents clientEvents; + AppRTCClient.SignalingEvents serverEvents; + + @Before + public void setUp() { + ShadowLog.stream = System.out; + + clientEvents = mock(AppRTCClient.SignalingEvents.class); + serverEvents = mock(AppRTCClient.SignalingEvents.class); + + client = new DirectRTCClient(clientEvents); + server = new DirectRTCClient(serverEvents); + } + + @Test + public void testValidIpPattern() { + // Strings that should match the pattern. + // clang-format off + final String[] ipAddresses = new String[] { + "0.0.0.0", + "127.0.0.1", + "192.168.0.1", + "0.0.0.0:8888", + "127.0.0.1:8888", + "192.168.0.1:8888", + "::", + "::1", + "2001:0db8:85a3:0000:0000:8a2e:0370:7946", + "[::]", + "[::1]", + "[2001:0db8:85a3:0000:0000:8a2e:0370:7946]", + "[::]:8888", + "[::1]:8888", + "[2001:0db8:85a3:0000:0000:8a2e:0370:7946]:8888" + }; + // clang-format on + + for (String ip : ipAddresses) { + assertTrue(ip + " didn't match IP_PATTERN even though it should.", + DirectRTCClient.IP_PATTERN.matcher(ip).matches()); + } + } + + @Test + public void testInvalidIpPattern() { + // Strings that shouldn't match the pattern. + // clang-format off + final String[] invalidIpAddresses = new String[] { + "Hello, World!", + "aaaa", + "1111", + "[hello world]", + "hello:world" + }; + // clang-format on + + for (String invalidIp : invalidIpAddresses) { + assertFalse(invalidIp + " matched IP_PATTERN even though it shouldn't.", + DirectRTCClient.IP_PATTERN.matcher(invalidIp).matches()); + } + } + + // TODO(sakal): Replace isNotNull(class) with isNotNull() once Java 8 is used. + @SuppressWarnings("deprecation") + @Test + public void testDirectRTCClient() { + server.connectToRoom(new AppRTCClient.RoomConnectionParameters(ROOM_URL, "0.0.0.0", LOOPBACK)); + try { + Thread.sleep(SERVER_WAIT); + } catch (InterruptedException e) { + fail(e.getMessage()); + } + client.connectToRoom( + new AppRTCClient.RoomConnectionParameters(ROOM_URL, "127.0.0.1", LOOPBACK)); + verify(serverEvents, timeout(NETWORK_TIMEOUT)) + .onConnectedToRoom(any(AppRTCClient.SignalingParameters.class)); + + SessionDescription offerSdp = new SessionDescription(SessionDescription.Type.OFFER, DUMMY_SDP); + server.sendOfferSdp(offerSdp); + verify(clientEvents, timeout(NETWORK_TIMEOUT)) + .onConnectedToRoom(any(AppRTCClient.SignalingParameters.class)); + + SessionDescription answerSdp = + new SessionDescription(SessionDescription.Type.ANSWER, DUMMY_SDP); + client.sendAnswerSdp(answerSdp); + verify(serverEvents, timeout(NETWORK_TIMEOUT)) + .onRemoteDescription(isNotNull(SessionDescription.class)); + + IceCandidate candidate = new IceCandidate(DUMMY_SDP_MID, 0, DUMMY_SDP); + server.sendLocalIceCandidate(candidate); + verify(clientEvents, timeout(NETWORK_TIMEOUT)) + .onRemoteIceCandidate(isNotNull(IceCandidate.class)); + + client.sendLocalIceCandidate(candidate); + verify(serverEvents, timeout(NETWORK_TIMEOUT)) + .onRemoteIceCandidate(isNotNull(IceCandidate.class)); + + client.disconnectFromRoom(); + verify(clientEvents, timeout(NETWORK_TIMEOUT)).onChannelClose(); + verify(serverEvents, timeout(NETWORK_TIMEOUT)).onChannelClose(); + + verifyNoMoreInteractions(clientEvents); + verifyNoMoreInteractions(serverEvents); + } +} diff --git a/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java b/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java new file mode 100644 index 0000000000..b301d6317c --- /dev/null +++ b/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java @@ -0,0 +1,199 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc; + +import static org.junit.Assert.fail; +import static org.mockito.Mockito.timeout; +import static org.mockito.Mockito.verify; +import static org.mockito.Mockito.verifyNoMoreInteractions; + +import org.chromium.testing.local.LocalRobolectricTestRunner; +import org.junit.After; +import org.junit.Before; +import org.junit.Test; +import org.junit.runner.RunWith; +import org.mockito.Mock; +import org.mockito.MockitoAnnotations; +import org.robolectric.annotation.Config; +import org.robolectric.shadows.ShadowLog; + +import java.util.concurrent.ExecutorService; +import java.util.concurrent.Executors; +import java.util.concurrent.TimeUnit; + +@RunWith(LocalRobolectricTestRunner.class) +@Config(manifest = Config.NONE) +public class TCPChannelClientTest { + private static final int PORT = 8888; + /** + * How long we wait before trying to connect to the server. Note: was + * previously only 10, which was too short (tests were flaky). + */ + private static final int SERVER_WAIT = 300; + private static final int CONNECT_TIMEOUT = 1000; + private static final int SEND_TIMEOUT = 1000; + private static final int DISCONNECT_TIMEOUT = 1000; + private static final int TERMINATION_TIMEOUT = 1000; + private static final String TEST_MESSAGE_SERVER = "Hello, Server!"; + private static final String TEST_MESSAGE_CLIENT = "Hello, Client!"; + + @Mock TCPChannelClient.TCPChannelEvents serverEvents; + @Mock TCPChannelClient.TCPChannelEvents clientEvents; + + private ExecutorService executor; + private TCPChannelClient server; + private TCPChannelClient client; + + @Before + public void setUp() { + ShadowLog.stream = System.out; + + MockitoAnnotations.initMocks(this); + + executor = Executors.newSingleThreadExecutor(); + } + + @After + public void tearDown() { + verifyNoMoreEvents(); + + executeAndWait(new Runnable() { + @Override + public void run() { + client.disconnect(); + server.disconnect(); + } + }); + + // Stop the executor thread + executor.shutdown(); + try { + executor.awaitTermination(TERMINATION_TIMEOUT, TimeUnit.MILLISECONDS); + } catch (InterruptedException e) { + fail(e.getMessage()); + } + } + + @Test + public void testConnectIPv4() { + setUpIPv4Server(); + try { + Thread.sleep(SERVER_WAIT); + } catch (InterruptedException e) { + fail(e.getMessage()); + } + setUpIPv4Client(); + + verify(serverEvents, timeout(CONNECT_TIMEOUT)).onTCPConnected(true); + verify(clientEvents, timeout(CONNECT_TIMEOUT)).onTCPConnected(false); + } + + @Test + public void testConnectIPv6() { + setUpIPv6Server(); + try { + Thread.sleep(SERVER_WAIT); + } catch (InterruptedException e) { + fail(e.getMessage()); + } + setUpIPv6Client(); + + verify(serverEvents, timeout(CONNECT_TIMEOUT)).onTCPConnected(true); + verify(clientEvents, timeout(CONNECT_TIMEOUT)).onTCPConnected(false); + } + + @Test + public void testSendData() { + testConnectIPv4(); + + executeAndWait(new Runnable() { + @Override + public void run() { + client.send(TEST_MESSAGE_SERVER); + server.send(TEST_MESSAGE_CLIENT); + } + }); + + verify(serverEvents, timeout(SEND_TIMEOUT)).onTCPMessage(TEST_MESSAGE_SERVER); + verify(clientEvents, timeout(SEND_TIMEOUT)).onTCPMessage(TEST_MESSAGE_CLIENT); + } + + @Test + public void testDisconnectServer() { + testConnectIPv4(); + executeAndWait(new Runnable() { + @Override + public void run() { + server.disconnect(); + } + }); + + verify(serverEvents, timeout(DISCONNECT_TIMEOUT)).onTCPClose(); + verify(clientEvents, timeout(DISCONNECT_TIMEOUT)).onTCPClose(); + } + + @Test + public void testDisconnectClient() { + testConnectIPv4(); + executeAndWait(new Runnable() { + @Override + public void run() { + client.disconnect(); + } + }); + + verify(serverEvents, timeout(DISCONNECT_TIMEOUT)).onTCPClose(); + verify(clientEvents, timeout(DISCONNECT_TIMEOUT)).onTCPClose(); + } + + private void setUpIPv4Server() { + setUpServer("0.0.0.0", PORT); + } + + private void setUpIPv4Client() { + setUpClient("127.0.0.1", PORT); + } + + private void setUpIPv6Server() { + setUpServer("::", PORT); + } + + private void setUpIPv6Client() { + setUpClient("::1", PORT); + } + + private void setUpServer(String ip, int port) { + server = new TCPChannelClient(executor, serverEvents, ip, port); + } + + private void setUpClient(String ip, int port) { + client = new TCPChannelClient(executor, clientEvents, ip, port); + } + + /** + * Verifies no more server or client events have been issued + */ + private void verifyNoMoreEvents() { + verifyNoMoreInteractions(serverEvents); + verifyNoMoreInteractions(clientEvents); + } + + /** + * Queues runnable to be run and waits for it to be executed by the executor thread + */ + public void executeAndWait(Runnable runnable) { + try { + executor.submit(runnable).get(); + } catch (Exception e) { + fail(e.getMessage()); + } + } +} diff --git a/third_party/libwebrtc/examples/androidnativeapi/AndroidManifest.xml b/third_party/libwebrtc/examples/androidnativeapi/AndroidManifest.xml new file mode 100644 index 0000000000..27c7d9dd35 --- /dev/null +++ b/third_party/libwebrtc/examples/androidnativeapi/AndroidManifest.xml @@ -0,0 +1,23 @@ +<?xml version="1.0" encoding="utf-8"?> +<manifest xmlns:android="http://schemas.android.com/apk/res/android" + package="org.webrtc.examples.androidnativeapi"> + + + <uses-permission android:name="android.permission.INTERNET" /> + <uses-permission android:name="android.permission.CAMERA" /> + + <application + android:allowBackup="true" + android:label="@string/app_name" + android:supportsRtl="true"> + <activity android:name=".MainActivity" + android:exported="true"> + <intent-filter> + <action android:name="android.intent.action.MAIN" /> + + <category android:name="android.intent.category.LAUNCHER" /> + </intent-filter> + </activity> + </application> + +</manifest> diff --git a/third_party/libwebrtc/examples/androidnativeapi/BUILD.gn b/third_party/libwebrtc/examples/androidnativeapi/BUILD.gn new file mode 100644 index 0000000000..e4c48a2b36 --- /dev/null +++ b/third_party/libwebrtc/examples/androidnativeapi/BUILD.gn @@ -0,0 +1,80 @@ +import("//webrtc.gni") + +if (is_android) { + rtc_android_apk("androidnativeapi") { + testonly = true + apk_name = "androidnativeapi" + android_manifest = "AndroidManifest.xml" + min_sdk_version = 21 + target_sdk_version = 31 + + sources = [ + "java/org/webrtc/examples/androidnativeapi/CallClient.java", + "java/org/webrtc/examples/androidnativeapi/MainActivity.java", + ] + + deps = [ + ":resources", + "//modules/audio_device:audio_device_java", + "//rtc_base:base_java", + "//sdk/android:camera_java", + "//sdk/android:surfaceviewrenderer_java", + "//sdk/android:video_api_java", + "//sdk/android:video_java", + "//third_party/androidx:androidx_annotation_annotation_java", + ] + + shared_libraries = [ ":examples_androidnativeapi_jni" ] + } + + generate_jni("generated_jni") { + testonly = true + sources = [ "java/org/webrtc/examples/androidnativeapi/CallClient.java" ] + namespace = "webrtc_examples" + jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h" + } + + rtc_shared_library("examples_androidnativeapi_jni") { + testonly = true + sources = [ + "jni/android_call_client.cc", + "jni/android_call_client.h", + "jni/onload.cc", + ] + + suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ] + configs += [ "//build/config/android:hide_all_but_jni" ] + + deps = [ + ":generated_jni", + "../../api:scoped_refptr", + "../../api:sequence_checker", + "../../rtc_base:ssl", + "../../rtc_base/synchronization:mutex", + "//api:libjingle_peerconnection_api", + "//api/rtc_event_log:rtc_event_log_factory", + "//api/task_queue:default_task_queue_factory", + "//media:rtc_audio_video", + "//media:rtc_internal_video_codecs", + "//media:rtc_media_engine_defaults", + "//modules/utility", + "//pc:libjingle_peerconnection", + "//sdk/android:native_api_base", + "//sdk/android:native_api_jni", + "//sdk/android:native_api_video", + ] + } + + android_resources("resources") { + testonly = true + custom_package = "org.webrtc.examples.androidnativeapi" + sources = [ + "res/layout/activity_main.xml", + "res/values/strings.xml", + ] + + # Needed for Bazel converter. + resource_dirs = [ "res" ] + assert(resource_dirs != []) # Mark as used. + } +} diff --git a/third_party/libwebrtc/examples/androidnativeapi/DEPS b/third_party/libwebrtc/examples/androidnativeapi/DEPS new file mode 100644 index 0000000000..5dbfcf1bb8 --- /dev/null +++ b/third_party/libwebrtc/examples/androidnativeapi/DEPS @@ -0,0 +1,5 @@ +include_rules = [ + "+logging/rtc_event_log/rtc_event_log_factory.h", + "+modules/utility/include", + "+sdk/android/native_api", +] diff --git a/third_party/libwebrtc/examples/androidnativeapi/OWNERS b/third_party/libwebrtc/examples/androidnativeapi/OWNERS new file mode 100644 index 0000000000..cf092a316a --- /dev/null +++ b/third_party/libwebrtc/examples/androidnativeapi/OWNERS @@ -0,0 +1 @@ +xalep@webrtc.org diff --git a/third_party/libwebrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/CallClient.java b/third_party/libwebrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/CallClient.java new file mode 100644 index 0000000000..7369a1286d --- /dev/null +++ b/third_party/libwebrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/CallClient.java @@ -0,0 +1,72 @@ +/* + * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.webrtc.examples.androidnativeapi; + +import android.content.Context; +import android.os.Handler; +import android.os.HandlerThread; +import org.webrtc.CapturerObserver; +import org.webrtc.SurfaceTextureHelper; +import org.webrtc.VideoCapturer; +import org.webrtc.VideoSink; + +public class CallClient { + private static final String TAG = "CallClient"; + private static final int CAPTURE_WIDTH = 640; + private static final int CAPTURE_HEIGHT = 480; + private static final int CAPTURE_FPS = 30; + + private final Context applicationContext; + private final HandlerThread thread; + private final Handler handler; + + private long nativeClient; + private SurfaceTextureHelper surfaceTextureHelper; + private VideoCapturer videoCapturer; + + public CallClient(Context applicationContext) { + this.applicationContext = applicationContext; + thread = new HandlerThread(TAG + "Thread"); + thread.start(); + handler = new Handler(thread.getLooper()); + handler.post(() -> { nativeClient = nativeCreateClient(); }); + } + + public void call(VideoSink localSink, VideoSink remoteSink, VideoCapturer videoCapturer, + SurfaceTextureHelper videoCapturerSurfaceTextureHelper) { + handler.post(() -> { + nativeCall(nativeClient, localSink, remoteSink); + videoCapturer.initialize(videoCapturerSurfaceTextureHelper, applicationContext, + nativeGetJavaVideoCapturerObserver(nativeClient)); + videoCapturer.startCapture(CAPTURE_WIDTH, CAPTURE_HEIGHT, CAPTURE_FPS); + }); + } + + public void hangup() { + handler.post(() -> { nativeHangup(nativeClient); }); + } + + public void close() { + handler.post(() -> { + nativeDelete(nativeClient); + nativeClient = 0; + }); + thread.quitSafely(); + } + + private static native long nativeCreateClient(); + private static native void nativeCall( + long nativeAndroidCallClient, VideoSink localSink, VideoSink remoteSink); + private static native void nativeHangup(long nativeAndroidCallClient); + private static native void nativeDelete(long nativeAndroidCallClient); + private static native CapturerObserver nativeGetJavaVideoCapturerObserver( + long nativeAndroidCallClient); +} diff --git a/third_party/libwebrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/MainActivity.java b/third_party/libwebrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/MainActivity.java new file mode 100644 index 0000000000..72fc0a686d --- /dev/null +++ b/third_party/libwebrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/MainActivity.java @@ -0,0 +1,120 @@ +/* + * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.webrtc.examples.androidnativeapi; + +import android.app.Activity; +import android.content.Context; +import android.os.Bundle; +import android.widget.Button; +import androidx.annotation.Nullable; +import org.webrtc.Camera1Enumerator; +import org.webrtc.Camera2Enumerator; +import org.webrtc.CameraEnumerator; +import org.webrtc.ContextUtils; +import org.webrtc.EglBase; +import org.webrtc.GlRectDrawer; +import org.webrtc.SurfaceTextureHelper; +import org.webrtc.SurfaceViewRenderer; +import org.webrtc.VideoCapturer; + +public class MainActivity extends Activity { + private @Nullable CallClient callClient; + private @Nullable EglBase eglBase; + private @Nullable SurfaceViewRenderer localRenderer; + private @Nullable SurfaceViewRenderer remoteRenderer; + private @Nullable SurfaceTextureHelper videoCapturerSurfaceTextureHelper; + private @Nullable VideoCapturer videoCapturer; + + @Override + protected void onCreate(Bundle savedInstance) { + ContextUtils.initialize(getApplicationContext()); + + super.onCreate(savedInstance); + setContentView(R.layout.activity_main); + + System.loadLibrary("examples_androidnativeapi_jni"); + callClient = new CallClient(getApplicationContext()); + + Button callButton = (Button) findViewById(R.id.call_button); + callButton.setOnClickListener((view) -> { + if (videoCapturer == null) { + videoCapturer = createVideoCapturer(getApplicationContext()); + } + callClient.call( + localRenderer, remoteRenderer, videoCapturer, videoCapturerSurfaceTextureHelper); + }); + + Button hangupButton = (Button) findViewById(R.id.hangup_button); + hangupButton.setOnClickListener((view) -> { hangup(); }); + } + + @Override + protected void onStart() { + super.onStart(); + + eglBase = EglBase.create(null /* sharedContext */, EglBase.CONFIG_PLAIN); + localRenderer = (SurfaceViewRenderer) findViewById(R.id.local_renderer); + remoteRenderer = (SurfaceViewRenderer) findViewById(R.id.remote_renderer); + + localRenderer.init(eglBase.getEglBaseContext(), null /* rendererEvents */, EglBase.CONFIG_PLAIN, + new GlRectDrawer()); + remoteRenderer.init(eglBase.getEglBaseContext(), null /* rendererEvents */, + EglBase.CONFIG_PLAIN, new GlRectDrawer()); + + videoCapturerSurfaceTextureHelper = + SurfaceTextureHelper.create("VideoCapturerThread", eglBase.getEglBaseContext()); + } + + @Override + protected void onStop() { + hangup(); + + localRenderer.release(); + remoteRenderer.release(); + videoCapturerSurfaceTextureHelper.dispose(); + eglBase.release(); + + localRenderer = null; + remoteRenderer = null; + videoCapturerSurfaceTextureHelper = null; + eglBase = null; + + super.onStop(); + } + + @Override + protected void onDestroy() { + callClient.close(); + callClient = null; + + super.onDestroy(); + } + + private void hangup() { + if (videoCapturer != null) { + try { + videoCapturer.stopCapture(); + } catch (InterruptedException e) { + throw new RuntimeException(e); + } + videoCapturer.dispose(); + videoCapturer = null; + } + callClient.hangup(); + } + + private static VideoCapturer createVideoCapturer(Context context) { + CameraEnumerator enumerator = Camera2Enumerator.isSupported(context) + ? new Camera2Enumerator(context) + : new Camera1Enumerator(); + return enumerator.createCapturer(enumerator.getDeviceNames()[0], null /* eventsHandler */); + } +} diff --git a/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.cc b/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.cc new file mode 100644 index 0000000000..7da56e6e60 --- /dev/null +++ b/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.cc @@ -0,0 +1,294 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/androidnativeapi/jni/android_call_client.h" + +#include <utility> + +#include <memory> + +#include "api/peer_connection_interface.h" +#include "api/rtc_event_log/rtc_event_log_factory.h" +#include "api/task_queue/default_task_queue_factory.h" +#include "examples/androidnativeapi/generated_jni/CallClient_jni.h" +#include "media/engine/internal_decoder_factory.h" +#include "media/engine/internal_encoder_factory.h" +#include "media/engine/webrtc_media_engine.h" +#include "media/engine/webrtc_media_engine_defaults.h" +#include "sdk/android/native_api/jni/java_types.h" +#include "sdk/android/native_api/video/wrapper.h" + +namespace webrtc_examples { + +class AndroidCallClient::PCObserver : public webrtc::PeerConnectionObserver { + public: + explicit PCObserver(AndroidCallClient* client); + + void OnSignalingChange( + webrtc::PeerConnectionInterface::SignalingState new_state) override; + void OnDataChannel( + rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override; + void OnRenegotiationNeeded() override; + void OnIceConnectionChange( + webrtc::PeerConnectionInterface::IceConnectionState new_state) override; + void OnIceGatheringChange( + webrtc::PeerConnectionInterface::IceGatheringState new_state) override; + void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override; + + private: + AndroidCallClient* const client_; +}; + +namespace { + +class CreateOfferObserver : public webrtc::CreateSessionDescriptionObserver { + public: + explicit CreateOfferObserver( + rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc); + + void OnSuccess(webrtc::SessionDescriptionInterface* desc) override; + void OnFailure(webrtc::RTCError error) override; + + private: + const rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_; +}; + +class SetRemoteSessionDescriptionObserver + : public webrtc::SetRemoteDescriptionObserverInterface { + public: + void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override; +}; + +class SetLocalSessionDescriptionObserver + : public webrtc::SetSessionDescriptionObserver { + public: + void OnSuccess() override; + void OnFailure(webrtc::RTCError error) override; +}; + +} // namespace + +AndroidCallClient::AndroidCallClient() + : call_started_(false), pc_observer_(std::make_unique<PCObserver>(this)) { + thread_checker_.Detach(); + CreatePeerConnectionFactory(); +} + +AndroidCallClient::~AndroidCallClient() = default; + +void AndroidCallClient::Call(JNIEnv* env, + const webrtc::JavaRef<jobject>& local_sink, + const webrtc::JavaRef<jobject>& remote_sink) { + RTC_DCHECK_RUN_ON(&thread_checker_); + + webrtc::MutexLock lock(&pc_mutex_); + if (call_started_) { + RTC_LOG(LS_WARNING) << "Call already started."; + return; + } + call_started_ = true; + + local_sink_ = webrtc::JavaToNativeVideoSink(env, local_sink.obj()); + remote_sink_ = webrtc::JavaToNativeVideoSink(env, remote_sink.obj()); + + video_source_ = webrtc::CreateJavaVideoSource(env, signaling_thread_.get(), + /* is_screencast= */ false, + /* align_timestamps= */ true); + + CreatePeerConnection(); + Connect(); +} + +void AndroidCallClient::Hangup(JNIEnv* env) { + RTC_DCHECK_RUN_ON(&thread_checker_); + + call_started_ = false; + + { + webrtc::MutexLock lock(&pc_mutex_); + if (pc_ != nullptr) { + pc_->Close(); + pc_ = nullptr; + } + } + + local_sink_ = nullptr; + remote_sink_ = nullptr; + video_source_ = nullptr; +} + +void AndroidCallClient::Delete(JNIEnv* env) { + RTC_DCHECK_RUN_ON(&thread_checker_); + + delete this; +} + +webrtc::ScopedJavaLocalRef<jobject> +AndroidCallClient::GetJavaVideoCapturerObserver(JNIEnv* env) { + RTC_DCHECK_RUN_ON(&thread_checker_); + + return video_source_->GetJavaVideoCapturerObserver(env); +} + +void AndroidCallClient::CreatePeerConnectionFactory() { + network_thread_ = rtc::Thread::CreateWithSocketServer(); + network_thread_->SetName("network_thread", nullptr); + RTC_CHECK(network_thread_->Start()) << "Failed to start thread"; + + worker_thread_ = rtc::Thread::Create(); + worker_thread_->SetName("worker_thread", nullptr); + RTC_CHECK(worker_thread_->Start()) << "Failed to start thread"; + + signaling_thread_ = rtc::Thread::Create(); + signaling_thread_->SetName("signaling_thread", nullptr); + RTC_CHECK(signaling_thread_->Start()) << "Failed to start thread"; + + webrtc::PeerConnectionFactoryDependencies pcf_deps; + pcf_deps.network_thread = network_thread_.get(); + pcf_deps.worker_thread = worker_thread_.get(); + pcf_deps.signaling_thread = signaling_thread_.get(); + pcf_deps.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory(); + pcf_deps.call_factory = webrtc::CreateCallFactory(); + pcf_deps.event_log_factory = std::make_unique<webrtc::RtcEventLogFactory>( + pcf_deps.task_queue_factory.get()); + + cricket::MediaEngineDependencies media_deps; + media_deps.task_queue_factory = pcf_deps.task_queue_factory.get(); + media_deps.video_encoder_factory = + std::make_unique<webrtc::InternalEncoderFactory>(); + media_deps.video_decoder_factory = + std::make_unique<webrtc::InternalDecoderFactory>(); + webrtc::SetMediaEngineDefaults(&media_deps); + pcf_deps.media_engine = cricket::CreateMediaEngine(std::move(media_deps)); + RTC_LOG(LS_INFO) << "Media engine created: " << pcf_deps.media_engine.get(); + + pcf_ = CreateModularPeerConnectionFactory(std::move(pcf_deps)); + RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << pcf_.get(); +} + +void AndroidCallClient::CreatePeerConnection() { + webrtc::MutexLock lock(&pc_mutex_); + webrtc::PeerConnectionInterface::RTCConfiguration config; + config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; + // Encryption has to be disabled for loopback to work. + webrtc::PeerConnectionFactoryInterface::Options options; + options.disable_encryption = true; + pcf_->SetOptions(options); + webrtc::PeerConnectionDependencies deps(pc_observer_.get()); + pc_ = pcf_->CreatePeerConnectionOrError(config, std::move(deps)).MoveValue(); + + RTC_LOG(LS_INFO) << "PeerConnection created: " << pc_.get(); + + rtc::scoped_refptr<webrtc::VideoTrackInterface> local_video_track( + pcf_->CreateVideoTrack("video", video_source_.get())); + local_video_track->AddOrUpdateSink(local_sink_.get(), rtc::VideoSinkWants()); + pc_->AddTransceiver(local_video_track); + RTC_LOG(LS_INFO) << "Local video sink set up: " << local_video_track.get(); + + for (const rtc::scoped_refptr<webrtc::RtpTransceiverInterface>& tranceiver : + pc_->GetTransceivers()) { + rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track = + tranceiver->receiver()->track(); + if (track && + track->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) { + static_cast<webrtc::VideoTrackInterface*>(track.get()) + ->AddOrUpdateSink(remote_sink_.get(), rtc::VideoSinkWants()); + RTC_LOG(LS_INFO) << "Remote video sink set up: " << track.get(); + break; + } + } +} + +void AndroidCallClient::Connect() { + webrtc::MutexLock lock(&pc_mutex_); + pc_->CreateOffer(rtc::make_ref_counted<CreateOfferObserver>(pc_).get(), + webrtc::PeerConnectionInterface::RTCOfferAnswerOptions()); +} + +AndroidCallClient::PCObserver::PCObserver(AndroidCallClient* client) + : client_(client) {} + +void AndroidCallClient::PCObserver::OnSignalingChange( + webrtc::PeerConnectionInterface::SignalingState new_state) { + RTC_LOG(LS_INFO) << "OnSignalingChange: " << new_state; +} + +void AndroidCallClient::PCObserver::OnDataChannel( + rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) { + RTC_LOG(LS_INFO) << "OnDataChannel"; +} + +void AndroidCallClient::PCObserver::OnRenegotiationNeeded() { + RTC_LOG(LS_INFO) << "OnRenegotiationNeeded"; +} + +void AndroidCallClient::PCObserver::OnIceConnectionChange( + webrtc::PeerConnectionInterface::IceConnectionState new_state) { + RTC_LOG(LS_INFO) << "OnIceConnectionChange: " << new_state; +} + +void AndroidCallClient::PCObserver::OnIceGatheringChange( + webrtc::PeerConnectionInterface::IceGatheringState new_state) { + RTC_LOG(LS_INFO) << "OnIceGatheringChange: " << new_state; +} + +void AndroidCallClient::PCObserver::OnIceCandidate( + const webrtc::IceCandidateInterface* candidate) { + RTC_LOG(LS_INFO) << "OnIceCandidate: " << candidate->server_url(); + webrtc::MutexLock lock(&client_->pc_mutex_); + RTC_DCHECK(client_->pc_ != nullptr); + client_->pc_->AddIceCandidate(candidate); +} + +CreateOfferObserver::CreateOfferObserver( + rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc) + : pc_(pc) {} + +void CreateOfferObserver::OnSuccess(webrtc::SessionDescriptionInterface* desc) { + std::string sdp; + desc->ToString(&sdp); + RTC_LOG(LS_INFO) << "Created offer: " << sdp; + + // Ownership of desc was transferred to us, now we transfer it forward. + pc_->SetLocalDescription( + rtc::make_ref_counted<SetLocalSessionDescriptionObserver>().get(), desc); + + // Generate a fake answer. + std::unique_ptr<webrtc::SessionDescriptionInterface> answer( + webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp)); + pc_->SetRemoteDescription( + std::move(answer), + rtc::make_ref_counted<SetRemoteSessionDescriptionObserver>()); +} + +void CreateOfferObserver::OnFailure(webrtc::RTCError error) { + RTC_LOG(LS_INFO) << "Failed to create offer: " << ToString(error.type()) + << ": " << error.message(); +} + +void SetRemoteSessionDescriptionObserver::OnSetRemoteDescriptionComplete( + webrtc::RTCError error) { + RTC_LOG(LS_INFO) << "Set remote description: " << error.message(); +} + +void SetLocalSessionDescriptionObserver::OnSuccess() { + RTC_LOG(LS_INFO) << "Set local description success!"; +} + +void SetLocalSessionDescriptionObserver::OnFailure(webrtc::RTCError error) { + RTC_LOG(LS_INFO) << "Set local description failure: " + << ToString(error.type()) << ": " << error.message(); +} + +static jlong JNI_CallClient_CreateClient(JNIEnv* env) { + return webrtc::NativeToJavaPointer(new webrtc_examples::AndroidCallClient()); +} + +} // namespace webrtc_examples diff --git a/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.h b/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.h new file mode 100644 index 0000000000..c9153d09bd --- /dev/null +++ b/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.h @@ -0,0 +1,76 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef EXAMPLES_ANDROIDNATIVEAPI_JNI_ANDROID_CALL_CLIENT_H_ +#define EXAMPLES_ANDROIDNATIVEAPI_JNI_ANDROID_CALL_CLIENT_H_ + +#include <jni.h> + +#include <memory> +#include <string> + +#include "api/peer_connection_interface.h" +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" +#include "rtc_base/synchronization/mutex.h" +#include "sdk/android/native_api/jni/scoped_java_ref.h" +#include "sdk/android/native_api/video/video_source.h" + +namespace webrtc_examples { + +class AndroidCallClient { + public: + AndroidCallClient(); + ~AndroidCallClient(); + + void Call(JNIEnv* env, + const webrtc::JavaRef<jobject>& local_sink, + const webrtc::JavaRef<jobject>& remote_sink); + void Hangup(JNIEnv* env); + // A helper method for Java code to delete this object. Calls delete this. + void Delete(JNIEnv* env); + + webrtc::ScopedJavaLocalRef<jobject> GetJavaVideoCapturerObserver(JNIEnv* env); + + private: + class PCObserver; + + void CreatePeerConnectionFactory() RTC_RUN_ON(thread_checker_); + void CreatePeerConnection() RTC_RUN_ON(thread_checker_); + void Connect() RTC_RUN_ON(thread_checker_); + + webrtc::SequenceChecker thread_checker_; + + bool call_started_ RTC_GUARDED_BY(thread_checker_); + + const std::unique_ptr<PCObserver> pc_observer_; + + rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf_ + RTC_GUARDED_BY(thread_checker_); + std::unique_ptr<rtc::Thread> network_thread_ RTC_GUARDED_BY(thread_checker_); + std::unique_ptr<rtc::Thread> worker_thread_ RTC_GUARDED_BY(thread_checker_); + std::unique_ptr<rtc::Thread> signaling_thread_ + RTC_GUARDED_BY(thread_checker_); + + std::unique_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> local_sink_ + RTC_GUARDED_BY(thread_checker_); + std::unique_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> remote_sink_ + RTC_GUARDED_BY(thread_checker_); + rtc::scoped_refptr<webrtc::JavaVideoTrackSourceInterface> video_source_ + RTC_GUARDED_BY(thread_checker_); + + webrtc::Mutex pc_mutex_; + rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_ + RTC_GUARDED_BY(pc_mutex_); +}; + +} // namespace webrtc_examples + +#endif // EXAMPLES_ANDROIDNATIVEAPI_JNI_ANDROID_CALL_CLIENT_H_ diff --git a/third_party/libwebrtc/examples/androidnativeapi/jni/onload.cc b/third_party/libwebrtc/examples/androidnativeapi/jni/onload.cc new file mode 100644 index 0000000000..6ea5275d2a --- /dev/null +++ b/third_party/libwebrtc/examples/androidnativeapi/jni/onload.cc @@ -0,0 +1,30 @@ +/* + * Copyright 2018 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <jni.h> + +#include "modules/utility/include/jvm_android.h" +#include "rtc_base/ssl_adapter.h" +#include "sdk/android/native_api/base/init.h" + +namespace webrtc_examples { + +extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM* jvm, void* reserved) { + webrtc::InitAndroid(jvm); + webrtc::JVM::Initialize(jvm); + RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()"; + return JNI_VERSION_1_6; +} + +extern "C" void JNIEXPORT JNICALL JNI_OnUnLoad(JavaVM* jvm, void* reserved) { + RTC_CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()"; +} + +} // namespace webrtc_examples diff --git a/third_party/libwebrtc/examples/androidnativeapi/res/layout/activity_main.xml b/third_party/libwebrtc/examples/androidnativeapi/res/layout/activity_main.xml new file mode 100644 index 0000000000..ac8037320f --- /dev/null +++ b/third_party/libwebrtc/examples/androidnativeapi/res/layout/activity_main.xml @@ -0,0 +1,52 @@ +<?xml version="1.0" encoding="utf-8"?> +<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android" + xmlns:app="http://schemas.android.com/apk/res-auto" + xmlns:tools="http://schemas.android.com/tools" + android:orientation="vertical" + android:layout_width="match_parent" + android:layout_height="match_parent" + android:padding="8dp" + tools:context="org.webrtc.examples.androidnativeapi.MainActivity"> + + <org.webrtc.SurfaceViewRenderer + android:id="@+id/local_renderer" + android:layout_width="match_parent" + android:layout_height="0dp" + android:layout_weight="1" + android:layout_margin="8dp"/> + + <org.webrtc.SurfaceViewRenderer + android:id="@+id/remote_renderer" + android:layout_width="match_parent" + android:layout_height="0dp" + android:layout_weight="1" + android:layout_margin="8dp"/> + + + <LinearLayout + android:orientation="horizontal" + android:layout_width="match_parent" + android:layout_height="48dp" + style="?android:attr/buttonBarStyle"> + + <Button + android:id="@+id/call_button" + android:text="@string/call_button" + style="?android:attr/buttonBarButtonStyle" + android:layout_width="0dp" + android:layout_height="48dp" + android:layout_weight="1" + android:layout_margin="8dp"/> + + <Button + android:id="@+id/hangup_button" + android:text="@string/hangup_button" + style="?android:attr/buttonBarButtonStyle" + android:layout_width="0dp" + android:layout_height="48dp" + android:layout_weight="1" + android:layout_margin="8dp"/> + + </LinearLayout> + +</LinearLayout> diff --git a/third_party/libwebrtc/examples/androidnativeapi/res/values/strings.xml b/third_party/libwebrtc/examples/androidnativeapi/res/values/strings.xml new file mode 100644 index 0000000000..a00920c92b --- /dev/null +++ b/third_party/libwebrtc/examples/androidnativeapi/res/values/strings.xml @@ -0,0 +1,5 @@ +<resources> + <string name="app_name">androidnativeapi</string> + <string name="call_button">Call</string> + <string name="hangup_button">Hangup</string> +</resources> diff --git a/third_party/libwebrtc/examples/androidtests/AndroidManifest.xml b/third_party/libwebrtc/examples/androidtests/AndroidManifest.xml new file mode 100644 index 0000000000..38ed3e3b9a --- /dev/null +++ b/third_party/libwebrtc/examples/androidtests/AndroidManifest.xml @@ -0,0 +1,26 @@ +<!-- + * Copyright 2017 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. +--> + +<manifest + xmlns:android="http://schemas.android.com/apk/res/android" + xmlns:tools="http://schemas.android.com/tools" + package="org.appspot.apprtc.test"> + + <uses-permission android:name="android.permission.RUN_INSTRUMENTATION" /> + + <application> + <uses-library android:name="android.test.runner" /> + </application> + + <instrumentation android:name="android.support.test.runner.AndroidJUnitRunner" + tools:ignore="MissingPrefix" + android:targetPackage="org.appspot.apprtc" + android:label="Tests for AppRTCMobile"/> +</manifest> diff --git a/third_party/libwebrtc/examples/androidtests/OWNERS b/third_party/libwebrtc/examples/androidtests/OWNERS new file mode 100644 index 0000000000..cf092a316a --- /dev/null +++ b/third_party/libwebrtc/examples/androidtests/OWNERS @@ -0,0 +1 @@ +xalep@webrtc.org diff --git a/third_party/libwebrtc/examples/androidtests/README b/third_party/libwebrtc/examples/androidtests/README new file mode 100644 index 0000000000..0701b0e896 --- /dev/null +++ b/third_party/libwebrtc/examples/androidtests/README @@ -0,0 +1,14 @@ +This directory contains an example unit test for Android AppRTCMobile. + +Example of building & using the app: + +- Build Android AppRTCMobile and AppRTCMobile unit test: +cd <path/to/webrtc>/src +ninja -C out/Debug AppRTCMobile_test_apk + +- Install AppRTCMobile and AppRTCMobileTest: +adb install -r out/Debug/apks/AppRTCMobile.apk +adb install -r out/Debug/apks/AppRTCMobileTest.apk + +- Run unit tests: +adb shell am instrument -w org.appspot.apprtc.test/android.test.InstrumentationTestRunner diff --git a/third_party/libwebrtc/examples/androidtests/ant.properties b/third_party/libwebrtc/examples/androidtests/ant.properties new file mode 100644 index 0000000000..ec7d042885 --- /dev/null +++ b/third_party/libwebrtc/examples/androidtests/ant.properties @@ -0,0 +1,18 @@ +# This file is used to override default values used by the Ant build system. +# +# This file must be checked into Version Control Systems, as it is +# integral to the build system of your project. + +# This file is only used by the Ant script. + +# You can use this to override default values such as +# 'source.dir' for the location of your java source folder and +# 'out.dir' for the location of your output folder. + +# You can also use it define how the release builds are signed by declaring +# the following properties: +# 'key.store' for the location of your keystore and +# 'key.alias' for the name of the key to use. +# The password will be asked during the build when you use the 'release' target. + +tested.project.dir=../android diff --git a/third_party/libwebrtc/examples/androidtests/build.xml b/third_party/libwebrtc/examples/androidtests/build.xml new file mode 100644 index 0000000000..95847b74c2 --- /dev/null +++ b/third_party/libwebrtc/examples/androidtests/build.xml @@ -0,0 +1,92 @@ +<?xml version="1.0" encoding="UTF-8"?> +<project name="AppRTCMobileTest" default="help"> + + <!-- The local.properties file is created and updated by the 'android' tool. + It contains the path to the SDK. It should *NOT* be checked into + Version Control Systems. --> + <property file="local.properties" /> + + <!-- The ant.properties file can be created by you. It is only edited by the + 'android' tool to add properties to it. + This is the place to change some Ant specific build properties. + Here are some properties you may want to change/update: + + source.dir + The name of the source directory. Default is 'src'. + out.dir + The name of the output directory. Default is 'bin'. + + For other overridable properties, look at the beginning of the rules + files in the SDK, at tools/ant/build.xml + + Properties related to the SDK location or the project target should + be updated using the 'android' tool with the 'update' action. + + This file is an integral part of the build system for your + application and should be checked into Version Control Systems. + + --> + <property file="ant.properties" /> + + <!-- if sdk.dir was not set from one of the property file, then + get it from the ANDROID_HOME env var. + This must be done before we load project.properties since + the proguard config can use sdk.dir --> + <property environment="env" /> + <condition property="sdk.dir" value="${env.ANDROID_SDK_ROOT}"> + <isset property="env.ANDROID_SDK_ROOT" /> + </condition> + + <!-- The project.properties file is created and updated by the 'android' + tool, as well as ADT. + + This contains project specific properties such as project target, and library + dependencies. Lower level build properties are stored in ant.properties + (or in .classpath for Eclipse projects). + + This file is an integral part of the build system for your + application and should be checked into Version Control Systems. --> + <loadproperties srcFile="project.properties" /> + + <!-- quick check on sdk.dir --> + <fail + message="sdk.dir is missing. Make sure to generate local.properties using 'android update project' or to inject it through the ANDROID_HOME environment variable." + unless="sdk.dir" + /> + + <!-- + Import per project custom build rules if present at the root of the project. + This is the place to put custom intermediary targets such as: + -pre-build + -pre-compile + -post-compile (This is typically used for code obfuscation. + Compiled code location: ${out.classes.absolute.dir} + If this is not done in place, override ${out.dex.input.absolute.dir}) + -post-package + -post-build + -pre-clean + --> + <import file="custom_rules.xml" optional="true" /> + + <!-- Import the actual build file. + + To customize existing targets, there are two options: + - Customize only one target: + - copy/paste the target into this file, *before* the + <import> task. + - customize it to your needs. + - Customize the whole content of build.xml + - copy/paste the content of the rules files (minus the top node) + into this file, replacing the <import> task. + - customize to your needs. + + *********************** + ****** IMPORTANT ****** + *********************** + In all cases you must update the value of version-tag below to read 'custom' instead of an integer, + in order to avoid having your file be overridden by tools such as "android update project" + --> + <!-- version-tag: 1 --> + <import file="${sdk.dir}/tools/ant/build.xml" /> + +</project> diff --git a/third_party/libwebrtc/examples/androidtests/project.properties b/third_party/libwebrtc/examples/androidtests/project.properties new file mode 100644 index 0000000000..a6ca533fe3 --- /dev/null +++ b/third_party/libwebrtc/examples/androidtests/project.properties @@ -0,0 +1,16 @@ +# This file is automatically generated by Android Tools. +# Do not modify this file -- YOUR CHANGES WILL BE ERASED! +# +# This file must be checked in Version Control Systems. +# +# To customize properties used by the Ant build system edit +# "ant.properties", and override values to adapt the script to your +# project structure. +# +# To enable ProGuard to shrink and obfuscate your code, uncomment this (available properties: sdk.dir, user.home): +#proguard.config=${sdk.dir}/tools/proguard/proguard-android.txt:proguard-project.txt + +# Project target. +target=android-22 + +java.compilerargs=-Xlint:all -Werror diff --git a/third_party/libwebrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java b/third_party/libwebrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java new file mode 100644 index 0000000000..051d7379bd --- /dev/null +++ b/third_party/libwebrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java @@ -0,0 +1,637 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.appspot.apprtc.test; + +import static org.junit.Assert.assertTrue; +import static org.junit.Assert.fail; + +import android.os.Build; +import android.support.test.InstrumentationRegistry; +import android.support.test.runner.AndroidJUnit4; +import android.util.Log; +import androidx.test.filters.SmallTest; +import java.util.ArrayList; +import java.util.List; +import java.util.concurrent.CountDownLatch; +import java.util.concurrent.ExecutorService; +import java.util.concurrent.Executors; +import java.util.concurrent.TimeUnit; +import org.appspot.apprtc.AppRTCClient.SignalingParameters; +import org.appspot.apprtc.PeerConnectionClient; +import org.appspot.apprtc.PeerConnectionClient.PeerConnectionEvents; +import org.appspot.apprtc.PeerConnectionClient.PeerConnectionParameters; +import org.junit.After; +import org.junit.Before; +import org.junit.Test; +import org.junit.runner.RunWith; +import org.webrtc.Camera1Enumerator; +import org.webrtc.Camera2Enumerator; +import org.webrtc.CameraEnumerator; +import org.webrtc.EglBase; +import org.webrtc.IceCandidate; +import org.webrtc.PeerConnection; +import org.webrtc.PeerConnectionFactory; +import org.webrtc.RTCStatsReport; +import org.webrtc.SessionDescription; +import org.webrtc.VideoCapturer; +import org.webrtc.VideoFrame; +import org.webrtc.VideoSink; + +@RunWith(AndroidJUnit4.class) +public class PeerConnectionClientTest implements PeerConnectionEvents { + private static final String TAG = "RTCClientTest"; + private static final int ICE_CONNECTION_WAIT_TIMEOUT = 10000; + private static final int WAIT_TIMEOUT = 7000; + private static final int CAMERA_SWITCH_ATTEMPTS = 3; + private static final int VIDEO_RESTART_ATTEMPTS = 3; + private static final int CAPTURE_FORMAT_CHANGE_ATTEMPTS = 3; + private static final int VIDEO_RESTART_TIMEOUT = 500; + private static final int EXPECTED_VIDEO_FRAMES = 10; + private static final String VIDEO_CODEC_VP8 = "VP8"; + private static final String VIDEO_CODEC_VP9 = "VP9"; + private static final String VIDEO_CODEC_H264 = "H264"; + private static final int AUDIO_RUN_TIMEOUT = 1000; + private static final String LOCAL_RENDERER_NAME = "Local renderer"; + private static final String REMOTE_RENDERER_NAME = "Remote renderer"; + + private static final int MAX_VIDEO_FPS = 30; + private static final int WIDTH_VGA = 640; + private static final int HEIGHT_VGA = 480; + private static final int WIDTH_QVGA = 320; + private static final int HEIGHT_QVGA = 240; + + // The peer connection client is assumed to be thread safe in itself; the + // reference is written by the test thread and read by worker threads. + private volatile PeerConnectionClient pcClient; + private volatile boolean loopback; + + // These are protected by their respective event objects. + private ExecutorService signalingExecutor; + private boolean isClosed; + private boolean isIceConnected; + private SessionDescription localDesc; + private List<IceCandidate> iceCandidates = new ArrayList<>(); + private final Object localDescEvent = new Object(); + private final Object iceCandidateEvent = new Object(); + private final Object iceConnectedEvent = new Object(); + private final Object closeEvent = new Object(); + + // Mock VideoSink implementation. + private static class MockSink implements VideoSink { + // These are protected by 'this' since we gets called from worker threads. + private String rendererName; + private boolean renderFrameCalled; + + // Thread-safe in itself. + private CountDownLatch doneRendering; + + public MockSink(int expectedFrames, String rendererName) { + this.rendererName = rendererName; + reset(expectedFrames); + } + + // Resets render to wait for new amount of video frames. + // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression. + @SuppressWarnings("NoSynchronizedMethodCheck") + public synchronized void reset(int expectedFrames) { + renderFrameCalled = false; + doneRendering = new CountDownLatch(expectedFrames); + } + + @Override + // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression. + @SuppressWarnings("NoSynchronizedMethodCheck") + public synchronized void onFrame(VideoFrame frame) { + if (!renderFrameCalled) { + if (rendererName != null) { + Log.d(TAG, + rendererName + " render frame: " + frame.getRotatedWidth() + " x " + + frame.getRotatedHeight()); + } else { + Log.d(TAG, "Render frame: " + frame.getRotatedWidth() + " x " + frame.getRotatedHeight()); + } + } + renderFrameCalled = true; + doneRendering.countDown(); + } + + // This method shouldn't hold any locks or touch member variables since it + // blocks. + public boolean waitForFramesRendered(int timeoutMs) throws InterruptedException { + doneRendering.await(timeoutMs, TimeUnit.MILLISECONDS); + return (doneRendering.getCount() <= 0); + } + } + + // Peer connection events implementation. + @Override + public void onLocalDescription(SessionDescription desc) { + Log.d(TAG, "Local description type: " + desc.type); + synchronized (localDescEvent) { + localDesc = desc; + localDescEvent.notifyAll(); + } + } + + @Override + public void onIceCandidate(final IceCandidate candidate) { + synchronized (iceCandidateEvent) { + Log.d(TAG, "IceCandidate #" + iceCandidates.size() + " : " + candidate.toString()); + if (loopback) { + // Loopback local ICE candidate in a separate thread to avoid adding + // remote ICE candidate in a local ICE candidate callback. + signalingExecutor.execute(new Runnable() { + @Override + public void run() { + pcClient.addRemoteIceCandidate(candidate); + } + }); + } + iceCandidates.add(candidate); + iceCandidateEvent.notifyAll(); + } + } + + @Override + public void onIceCandidatesRemoved(final IceCandidate[] candidates) { + // TODO(honghaiz): Add this for tests. + } + + @Override + public void onIceConnected() { + Log.d(TAG, "ICE Connected"); + synchronized (iceConnectedEvent) { + isIceConnected = true; + iceConnectedEvent.notifyAll(); + } + } + + @Override + public void onIceDisconnected() { + Log.d(TAG, "ICE Disconnected"); + synchronized (iceConnectedEvent) { + isIceConnected = false; + iceConnectedEvent.notifyAll(); + } + } + + @Override + public void onConnected() { + Log.d(TAG, "DTLS Connected"); + } + + @Override + public void onDisconnected() { + Log.d(TAG, "DTLS Disconnected"); + } + + @Override + public void onPeerConnectionClosed() { + Log.d(TAG, "PeerConnection closed"); + synchronized (closeEvent) { + isClosed = true; + closeEvent.notifyAll(); + } + } + + @Override + public void onPeerConnectionError(String description) { + fail("PC Error: " + description); + } + + @Override + public void onPeerConnectionStatsReady(final RTCStatsReport report) {} + + // Helper wait functions. + private boolean waitForLocalDescription(int timeoutMs) throws InterruptedException { + synchronized (localDescEvent) { + final long endTimeMs = System.currentTimeMillis() + timeoutMs; + while (localDesc == null) { + final long waitTimeMs = endTimeMs - System.currentTimeMillis(); + if (waitTimeMs < 0) { + return false; + } + localDescEvent.wait(waitTimeMs); + } + return true; + } + } + + private boolean waitForIceCandidates(int timeoutMs) throws InterruptedException { + synchronized (iceCandidateEvent) { + final long endTimeMs = System.currentTimeMillis() + timeoutMs; + while (iceCandidates.size() == 0) { + final long waitTimeMs = endTimeMs - System.currentTimeMillis(); + if (waitTimeMs < 0) { + return false; + } + iceCandidateEvent.wait(timeoutMs); + } + return true; + } + } + + private boolean waitForIceConnected(int timeoutMs) throws InterruptedException { + synchronized (iceConnectedEvent) { + final long endTimeMs = System.currentTimeMillis() + timeoutMs; + while (!isIceConnected) { + final long waitTimeMs = endTimeMs - System.currentTimeMillis(); + if (waitTimeMs < 0) { + Log.e(TAG, "ICE connection failure"); + return false; + } + iceConnectedEvent.wait(timeoutMs); + } + return true; + } + } + + private boolean waitForPeerConnectionClosed(int timeoutMs) throws InterruptedException { + synchronized (closeEvent) { + final long endTimeMs = System.currentTimeMillis() + timeoutMs; + while (!isClosed) { + final long waitTimeMs = endTimeMs - System.currentTimeMillis(); + if (waitTimeMs < 0) { + return false; + } + closeEvent.wait(timeoutMs); + } + return true; + } + } + + PeerConnectionClient createPeerConnectionClient(MockSink localRenderer, MockSink remoteRenderer, + PeerConnectionParameters peerConnectionParameters, VideoCapturer videoCapturer) { + List<PeerConnection.IceServer> iceServers = new ArrayList<>(); + SignalingParameters signalingParameters = + new SignalingParameters(iceServers, true, // iceServers, initiator. + null, null, null, // clientId, wssUrl, wssPostUrl. + null, null); // offerSdp, iceCandidates. + + final EglBase eglBase = EglBase.create(); + PeerConnectionClient client = + new PeerConnectionClient(InstrumentationRegistry.getTargetContext(), eglBase, + peerConnectionParameters, this /* PeerConnectionEvents */); + PeerConnectionFactory.Options options = new PeerConnectionFactory.Options(); + options.networkIgnoreMask = 0; + options.disableNetworkMonitor = true; + client.createPeerConnectionFactory(options); + client.createPeerConnection(localRenderer, remoteRenderer, videoCapturer, signalingParameters); + client.createOffer(); + return client; + } + + private PeerConnectionParameters createParametersForAudioCall() { + return new PeerConnectionParameters(false, /* videoCallEnabled */ + true, /* loopback */ + false, /* tracing */ + // Video codec parameters. + 0, /* videoWidth */ + 0, /* videoHeight */ + 0, /* videoFps */ + 0, /* videoStartBitrate */ + "", /* videoCodec */ + true, /* videoCodecHwAcceleration */ + false, /* videoFlexfecEnabled */ + // Audio codec parameters. + 0, /* audioStartBitrate */ + "OPUS", /* audioCodec */ + false, /* noAudioProcessing */ + false, /* aecDump */ + false, /* saveInputAudioToFile */ + false /* useOpenSLES */, false /* disableBuiltInAEC */, false /* disableBuiltInAGC */, + false /* disableBuiltInNS */, false /* disableWebRtcAGC */, false /* enableRtcEventLog */, + null /* dataChannelParameters */); + } + + private VideoCapturer createCameraCapturer(boolean captureToTexture) { + final boolean useCamera2 = captureToTexture + && Camera2Enumerator.isSupported(InstrumentationRegistry.getTargetContext()); + + CameraEnumerator enumerator; + if (useCamera2) { + enumerator = new Camera2Enumerator(InstrumentationRegistry.getTargetContext()); + } else { + enumerator = new Camera1Enumerator(captureToTexture); + } + String deviceName = enumerator.getDeviceNames()[0]; + return enumerator.createCapturer(deviceName, null); + } + + private PeerConnectionParameters createParametersForVideoCall(String videoCodec) { + return new PeerConnectionParameters(true, /* videoCallEnabled */ + true, /* loopback */ + false, /* tracing */ + // Video codec parameters. + 0, /* videoWidth */ + 0, /* videoHeight */ + 0, /* videoFps */ + 0, /* videoStartBitrate */ + videoCodec, /* videoCodec */ + true, /* videoCodecHwAcceleration */ + false, /* videoFlexfecEnabled */ + // Audio codec parameters. + 0, /* audioStartBitrate */ + "OPUS", /* audioCodec */ + false, /* noAudioProcessing */ + false, /* aecDump */ + false, /* saveInputAudioToFile */ + false /* useOpenSLES */, false /* disableBuiltInAEC */, false /* disableBuiltInAGC */, + false /* disableBuiltInNS */, false /* disableWebRtcAGC */, false /* enableRtcEventLog */, + null /* dataChannelParameters */); + } + + @Before + public void setUp() { + signalingExecutor = Executors.newSingleThreadExecutor(); + } + + @After + public void tearDown() { + signalingExecutor.shutdown(); + } + + @Test + @SmallTest + public void testSetLocalOfferMakesVideoFlowLocally() throws InterruptedException { + Log.d(TAG, "testSetLocalOfferMakesVideoFlowLocally"); + MockSink localRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, LOCAL_RENDERER_NAME); + pcClient = createPeerConnectionClient(localRenderer, + new MockSink(/* expectedFrames= */ 0, /* rendererName= */ null), + createParametersForVideoCall(VIDEO_CODEC_VP8), + createCameraCapturer(false /* captureToTexture */)); + + // Wait for local description and ice candidates set events. + assertTrue("Local description was not set.", waitForLocalDescription(WAIT_TIMEOUT)); + assertTrue("ICE candidates were not generated.", waitForIceCandidates(WAIT_TIMEOUT)); + + // Check that local video frames were rendered. + assertTrue( + "Local video frames were not rendered.", localRenderer.waitForFramesRendered(WAIT_TIMEOUT)); + + pcClient.close(); + assertTrue( + "PeerConnection close event was not received.", waitForPeerConnectionClosed(WAIT_TIMEOUT)); + Log.d(TAG, "testSetLocalOfferMakesVideoFlowLocally Done."); + } + + private void doLoopbackTest(PeerConnectionParameters parameters, VideoCapturer videoCapturer, + boolean decodeToTexture) throws InterruptedException { + loopback = true; + MockSink localRenderer = null; + MockSink remoteRenderer = null; + if (parameters.videoCallEnabled) { + Log.d(TAG, "testLoopback for video " + parameters.videoCodec); + localRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, LOCAL_RENDERER_NAME); + remoteRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME); + } else { + Log.d(TAG, "testLoopback for audio."); + } + pcClient = createPeerConnectionClient(localRenderer, remoteRenderer, parameters, videoCapturer); + + // Wait for local description, change type to answer and set as remote description. + assertTrue("Local description was not set.", waitForLocalDescription(WAIT_TIMEOUT)); + SessionDescription remoteDescription = new SessionDescription( + SessionDescription.Type.fromCanonicalForm("answer"), localDesc.description); + pcClient.setRemoteDescription(remoteDescription); + + // Wait for ICE connection. + assertTrue("ICE connection failure.", waitForIceConnected(ICE_CONNECTION_WAIT_TIMEOUT)); + + if (parameters.videoCallEnabled) { + // Check that local and remote video frames were rendered. + assertTrue("Local video frames were not rendered.", + localRenderer.waitForFramesRendered(WAIT_TIMEOUT)); + assertTrue("Remote video frames were not rendered.", + remoteRenderer.waitForFramesRendered(WAIT_TIMEOUT)); + } else { + // For audio just sleep for 1 sec. + // TODO(glaznev): check how we can detect that remote audio was rendered. + Thread.sleep(AUDIO_RUN_TIMEOUT); + } + + pcClient.close(); + assertTrue(waitForPeerConnectionClosed(WAIT_TIMEOUT)); + Log.d(TAG, "testLoopback done."); + } + + @Test + @SmallTest + public void testLoopbackAudio() throws InterruptedException { + doLoopbackTest(createParametersForAudioCall(), null, false /* decodeToTexture */); + } + + @Test + @SmallTest + public void testLoopbackVp8() throws InterruptedException { + doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_VP8), + createCameraCapturer(false /* captureToTexture */), false /* decodeToTexture */); + } + + @Test + @SmallTest + public void testLoopbackVp9() throws InterruptedException { + doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_VP9), + createCameraCapturer(false /* captureToTexture */), false /* decodeToTexture */); + } + + @Test + @SmallTest + public void testLoopbackH264() throws InterruptedException { + doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_H264), + createCameraCapturer(false /* captureToTexture */), false /* decodeToTexture */); + } + + @Test + @SmallTest + public void testLoopbackVp8DecodeToTexture() throws InterruptedException { + doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_VP8), + createCameraCapturer(false /* captureToTexture */), true /* decodeToTexture */); + } + + @Test + @SmallTest + public void testLoopbackVp9DecodeToTexture() throws InterruptedException { + doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_VP9), + createCameraCapturer(false /* captureToTexture */), true /* decodeToTexture */); + } + + @Test + @SmallTest + public void testLoopbackH264DecodeToTexture() throws InterruptedException { + doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_H264), + createCameraCapturer(false /* captureToTexture */), true /* decodeToTexture */); + } + + @Test + @SmallTest + public void testLoopbackVp8CaptureToTexture() throws InterruptedException { + doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_VP8), + createCameraCapturer(true /* captureToTexture */), true /* decodeToTexture */); + } + + @Test + @SmallTest + public void testLoopbackH264CaptureToTexture() throws InterruptedException { + doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_H264), + createCameraCapturer(true /* captureToTexture */), true /* decodeToTexture */); + } + + // Checks if default front camera can be switched to back camera and then + // again to front camera. + @Test + @SmallTest + public void testCameraSwitch() throws InterruptedException { + Log.d(TAG, "testCameraSwitch"); + loopback = true; + + MockSink localRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, LOCAL_RENDERER_NAME); + MockSink remoteRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME); + + pcClient = createPeerConnectionClient(localRenderer, remoteRenderer, + createParametersForVideoCall(VIDEO_CODEC_VP8), + createCameraCapturer(false /* captureToTexture */)); + + // Wait for local description, set type to answer and set as remote description. + assertTrue("Local description was not set.", waitForLocalDescription(WAIT_TIMEOUT)); + SessionDescription remoteDescription = new SessionDescription( + SessionDescription.Type.fromCanonicalForm("answer"), localDesc.description); + pcClient.setRemoteDescription(remoteDescription); + + // Wait for ICE connection. + assertTrue("ICE connection failure.", waitForIceConnected(ICE_CONNECTION_WAIT_TIMEOUT)); + + // Check that local and remote video frames were rendered. + assertTrue("Local video frames were not rendered before camera switch.", + localRenderer.waitForFramesRendered(WAIT_TIMEOUT)); + assertTrue("Remote video frames were not rendered before camera switch.", + remoteRenderer.waitForFramesRendered(WAIT_TIMEOUT)); + + for (int i = 0; i < CAMERA_SWITCH_ATTEMPTS; i++) { + // Try to switch camera + pcClient.switchCamera(); + + // Reset video renders and check that local and remote video frames + // were rendered after camera switch. + localRenderer.reset(EXPECTED_VIDEO_FRAMES); + remoteRenderer.reset(EXPECTED_VIDEO_FRAMES); + assertTrue("Local video frames were not rendered after camera switch.", + localRenderer.waitForFramesRendered(WAIT_TIMEOUT)); + assertTrue("Remote video frames were not rendered after camera switch.", + remoteRenderer.waitForFramesRendered(WAIT_TIMEOUT)); + } + pcClient.close(); + assertTrue(waitForPeerConnectionClosed(WAIT_TIMEOUT)); + Log.d(TAG, "testCameraSwitch done."); + } + + // Checks if video source can be restarted - simulate app goes to + // background and back to foreground. + @Test + @SmallTest + public void testVideoSourceRestart() throws InterruptedException { + Log.d(TAG, "testVideoSourceRestart"); + loopback = true; + + MockSink localRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, LOCAL_RENDERER_NAME); + MockSink remoteRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME); + + pcClient = createPeerConnectionClient(localRenderer, remoteRenderer, + createParametersForVideoCall(VIDEO_CODEC_VP8), + createCameraCapturer(false /* captureToTexture */)); + + // Wait for local description, set type to answer and set as remote description. + assertTrue("Local description was not set.", waitForLocalDescription(WAIT_TIMEOUT)); + SessionDescription remoteDescription = new SessionDescription( + SessionDescription.Type.fromCanonicalForm("answer"), localDesc.description); + pcClient.setRemoteDescription(remoteDescription); + + // Wait for ICE connection. + assertTrue("ICE connection failure.", waitForIceConnected(ICE_CONNECTION_WAIT_TIMEOUT)); + + // Check that local and remote video frames were rendered. + assertTrue("Local video frames were not rendered before video restart.", + localRenderer.waitForFramesRendered(WAIT_TIMEOUT)); + assertTrue("Remote video frames were not rendered before video restart.", + remoteRenderer.waitForFramesRendered(WAIT_TIMEOUT)); + + // Stop and then start video source a few times. + for (int i = 0; i < VIDEO_RESTART_ATTEMPTS; i++) { + pcClient.stopVideoSource(); + Thread.sleep(VIDEO_RESTART_TIMEOUT); + pcClient.startVideoSource(); + + // Reset video renders and check that local and remote video frames + // were rendered after video restart. + localRenderer.reset(EXPECTED_VIDEO_FRAMES); + remoteRenderer.reset(EXPECTED_VIDEO_FRAMES); + assertTrue("Local video frames were not rendered after video restart.", + localRenderer.waitForFramesRendered(WAIT_TIMEOUT)); + assertTrue("Remote video frames were not rendered after video restart.", + remoteRenderer.waitForFramesRendered(WAIT_TIMEOUT)); + } + pcClient.close(); + assertTrue(waitForPeerConnectionClosed(WAIT_TIMEOUT)); + Log.d(TAG, "testVideoSourceRestart done."); + } + + // Checks if capture format can be changed on fly and decoder can be reset properly. + @Test + @SmallTest + public void testCaptureFormatChange() throws InterruptedException { + Log.d(TAG, "testCaptureFormatChange"); + loopback = true; + + MockSink localRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, LOCAL_RENDERER_NAME); + MockSink remoteRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME); + + pcClient = createPeerConnectionClient(localRenderer, remoteRenderer, + createParametersForVideoCall(VIDEO_CODEC_VP8), + createCameraCapturer(false /* captureToTexture */)); + + // Wait for local description, set type to answer and set as remote description. + assertTrue("Local description was not set.", waitForLocalDescription(WAIT_TIMEOUT)); + SessionDescription remoteDescription = new SessionDescription( + SessionDescription.Type.fromCanonicalForm("answer"), localDesc.description); + pcClient.setRemoteDescription(remoteDescription); + + // Wait for ICE connection. + assertTrue("ICE connection failure.", waitForIceConnected(ICE_CONNECTION_WAIT_TIMEOUT)); + + // Check that local and remote video frames were rendered. + assertTrue("Local video frames were not rendered before camera resolution change.", + localRenderer.waitForFramesRendered(WAIT_TIMEOUT)); + assertTrue("Remote video frames were not rendered before camera resolution change.", + remoteRenderer.waitForFramesRendered(WAIT_TIMEOUT)); + + // Change capture output format a few times. + for (int i = 0; i < 2 * CAPTURE_FORMAT_CHANGE_ATTEMPTS; i++) { + if (i % 2 == 0) { + pcClient.changeCaptureFormat(WIDTH_VGA, HEIGHT_VGA, MAX_VIDEO_FPS); + } else { + pcClient.changeCaptureFormat(WIDTH_QVGA, HEIGHT_QVGA, MAX_VIDEO_FPS); + } + + // Reset video renders and check that local and remote video frames + // were rendered after capture format change. + localRenderer.reset(EXPECTED_VIDEO_FRAMES); + remoteRenderer.reset(EXPECTED_VIDEO_FRAMES); + assertTrue("Local video frames were not rendered after capture format change.", + localRenderer.waitForFramesRendered(WAIT_TIMEOUT)); + assertTrue("Remote video frames were not rendered after capture format change.", + remoteRenderer.waitForFramesRendered(WAIT_TIMEOUT)); + } + + pcClient.close(); + assertTrue(waitForPeerConnectionClosed(WAIT_TIMEOUT)); + Log.d(TAG, "testCaptureFormatChange done."); + } +} diff --git a/third_party/libwebrtc/examples/androidtests/third_party/.gitignore b/third_party/libwebrtc/examples/androidtests/third_party/.gitignore new file mode 100644 index 0000000000..52acefb2ec --- /dev/null +++ b/third_party/libwebrtc/examples/androidtests/third_party/.gitignore @@ -0,0 +1,3 @@ +# This file is needed for projects that has this directory as a separate Git +# mirror in DEPS. Without it, a lot is wiped and re-downloaded for each sync. +/gradle diff --git a/third_party/libwebrtc/examples/androidtests/third_party/README.webrtc b/third_party/libwebrtc/examples/androidtests/third_party/README.webrtc new file mode 100644 index 0000000000..a6ea884923 --- /dev/null +++ b/third_party/libwebrtc/examples/androidtests/third_party/README.webrtc @@ -0,0 +1,10 @@ +The third_party directory contains sources from other projects. + +Code in third_party must document the license under which the source is being +used. If the source itself does not include a license header or file, create +an entry in this file that refers to reliable documentation of the project's +license terms on the web (and add a note pointing here in the README file in +that directory). + +<Include table of license information here, once it is available> + diff --git a/third_party/libwebrtc/examples/androidvoip/AndroidManifest.xml b/third_party/libwebrtc/examples/androidvoip/AndroidManifest.xml new file mode 100644 index 0000000000..8e096b0452 --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/AndroidManifest.xml @@ -0,0 +1,38 @@ +<?xml version="1.0" encoding="utf-8"?> + +<!-- + * Copyright 2020 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. +--> + +<manifest xmlns:android="http://schemas.android.com/apk/res/android" + package="org.webrtc.examples.androidvoip"> + + + <uses-permission android:name="android.permission.INTERNET" /> + <uses-permission android:name="android.permission.RECORD_AUDIO" /> + <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" /> + + <uses-feature android:name="android.hardware.microphone" android:required="true" /> + <uses-feature android:name="android.hardware.telephony" android:required="false" /> + + <application + android:allowBackup="true" + android:label="@string/app_name" + android:supportsRtl="true"> + <activity android:name=".MainActivity" + android:windowSoftInputMode="stateHidden" + android:exported="true"> + <intent-filter> + <action android:name="android.intent.action.MAIN" /> + <category android:name="android.intent.category.LAUNCHER" /> + </intent-filter> + </activity> + </application> + +</manifest> diff --git a/third_party/libwebrtc/examples/androidvoip/BUILD.gn b/third_party/libwebrtc/examples/androidvoip/BUILD.gn new file mode 100644 index 0000000000..3d5186f279 --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/BUILD.gn @@ -0,0 +1,95 @@ +# Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. + +import("//webrtc.gni") + +if (is_android) { + rtc_android_apk("androidvoip") { + testonly = true + apk_name = "androidvoip" + android_manifest = "AndroidManifest.xml" + min_sdk_version = 21 + target_sdk_version = 31 + + sources = [ + "java/org/webrtc/examples/androidvoip/MainActivity.java", + "java/org/webrtc/examples/androidvoip/OnVoipClientTaskCompleted.java", + "java/org/webrtc/examples/androidvoip/VoipClient.java", + ] + + deps = [ + ":resources", + "//modules/audio_device:audio_device_java", + "//rtc_base:base_java", + "//sdk/android:base_java", + "//sdk/android:java_audio_device_module_java", + "//sdk/android:video_java", + "//third_party/androidx:androidx_core_core_java", + "//third_party/androidx:androidx_legacy_legacy_support_v4_java", + ] + + shared_libraries = [ ":examples_androidvoip_jni" ] + } + + generate_jni("generated_jni") { + testonly = true + sources = [ "java/org/webrtc/examples/androidvoip/VoipClient.java" ] + namespace = "webrtc_examples" + jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h" + } + + rtc_shared_library("examples_androidvoip_jni") { + testonly = true + sources = [ + "jni/android_voip_client.cc", + "jni/android_voip_client.h", + "jni/onload.cc", + ] + + suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ] + configs += [ "//build/config/android:hide_all_but_jni" ] + + deps = [ + ":generated_jni", + "../../rtc_base:async_packet_socket", + "../../rtc_base:async_udp_socket", + "../../rtc_base:logging", + "../../rtc_base:network", + "../../rtc_base:socket_address", + "../../rtc_base:socket_server", + "../../rtc_base:ssl", + "../../rtc_base:threading", + "//api:transport_api", + "//api/audio_codecs:audio_codecs_api", + "//api/audio_codecs:builtin_audio_decoder_factory", + "//api/audio_codecs:builtin_audio_encoder_factory", + "//api/task_queue:default_task_queue_factory", + "//api/voip:voip_api", + "//api/voip:voip_engine_factory", + "//rtc_base/third_party/sigslot:sigslot", + "//sdk/android:native_api_audio_device_module", + "//sdk/android:native_api_base", + "//sdk/android:native_api_jni", + "//third_party/abseil-cpp/absl/memory:memory", + ] + } + + android_resources("resources") { + testonly = true + custom_package = "org.webrtc.examples.androidvoip" + sources = [ + "res/layout/activity_main.xml", + "res/values/colors.xml", + "res/values/strings.xml", + ] + + # Needed for Bazel converter. + resource_dirs = [ "res" ] + assert(resource_dirs != []) # Mark as used. + } +} diff --git a/third_party/libwebrtc/examples/androidvoip/DEPS b/third_party/libwebrtc/examples/androidvoip/DEPS new file mode 100644 index 0000000000..edb714dd44 --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/DEPS @@ -0,0 +1,3 @@ +include_rules = [ + "+sdk/android/native_api", +] diff --git a/third_party/libwebrtc/examples/androidvoip/OWNERS b/third_party/libwebrtc/examples/androidvoip/OWNERS new file mode 100644 index 0000000000..e7d3200562 --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/OWNERS @@ -0,0 +1,2 @@ +natim@webrtc.org +xalep@webrtc.org diff --git a/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/MainActivity.java b/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/MainActivity.java new file mode 100644 index 0000000000..d06d6adf0d --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/MainActivity.java @@ -0,0 +1,341 @@ +/* + * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.webrtc.examples.androidvoip; + +import android.Manifest.permission; +import android.app.Activity; +import android.app.AlertDialog; +import android.content.Context; +import android.content.pm.PackageManager; +import android.os.Bundle; +import android.view.Gravity; +import android.view.View; +import android.widget.AdapterView; +import android.widget.ArrayAdapter; +import android.widget.Button; +import android.widget.EditText; +import android.widget.RelativeLayout; +import android.widget.ScrollView; +import android.widget.Spinner; +import android.widget.Switch; +import android.widget.TextView; +import android.widget.Toast; +import android.widget.ToggleButton; +import androidx.core.app.ActivityCompat; +import androidx.core.content.ContextCompat; +import java.util.ArrayList; +import java.util.Arrays; +import java.util.HashSet; +import java.util.List; +import java.util.Set; +import java.util.stream.Collectors; +import org.webrtc.ContextUtils; + +public class MainActivity extends Activity implements OnVoipClientTaskCompleted { + private static final int NUM_SUPPORTED_CODECS = 6; + + private VoipClient voipClient; + private List<String> supportedCodecs; + private boolean[] isDecoderSelected; + private Set<Integer> selectedDecoders; + + private Toast toast; + private ScrollView scrollView; + private TextView localIPAddressTextView; + private EditText localPortNumberEditText; + private EditText remoteIPAddressEditText; + private EditText remotePortNumberEditText; + private Spinner encoderSpinner; + private Button decoderSelectionButton; + private TextView decodersTextView; + private ToggleButton sessionButton; + private RelativeLayout switchLayout; + private Switch sendSwitch; + private Switch playoutSwitch; + + @Override + protected void onCreate(Bundle savedInstance) { + ContextUtils.initialize(getApplicationContext()); + + super.onCreate(savedInstance); + setContentView(R.layout.activity_main); + + System.loadLibrary("examples_androidvoip_jni"); + + voipClient = new VoipClient(getApplicationContext(), this); + voipClient.getAndSetUpLocalIPAddress(); + voipClient.getAndSetUpSupportedCodecs(); + + isDecoderSelected = new boolean[NUM_SUPPORTED_CODECS]; + selectedDecoders = new HashSet<>(); + + toast = Toast.makeText(this, "", Toast.LENGTH_SHORT); + + scrollView = (ScrollView) findViewById(R.id.scroll_view); + localIPAddressTextView = (TextView) findViewById(R.id.local_ip_address_text_view); + localPortNumberEditText = (EditText) findViewById(R.id.local_port_number_edit_text); + remoteIPAddressEditText = (EditText) findViewById(R.id.remote_ip_address_edit_text); + remotePortNumberEditText = (EditText) findViewById(R.id.remote_port_number_edit_text); + encoderSpinner = (Spinner) findViewById(R.id.encoder_spinner); + decoderSelectionButton = (Button) findViewById(R.id.decoder_selection_button); + decodersTextView = (TextView) findViewById(R.id.decoders_text_view); + sessionButton = (ToggleButton) findViewById(R.id.session_button); + switchLayout = (RelativeLayout) findViewById(R.id.switch_layout); + sendSwitch = (Switch) findViewById(R.id.start_send_switch); + playoutSwitch = (Switch) findViewById(R.id.start_playout_switch); + + setUpSessionButton(); + setUpSendAndPlayoutSwitch(); + } + + private void setUpEncoderSpinner(List<String> supportedCodecs) { + ArrayAdapter<String> encoderAdapter = + new ArrayAdapter<String>(this, android.R.layout.simple_spinner_item, supportedCodecs); + encoderAdapter.setDropDownViewResource(android.R.layout.simple_spinner_dropdown_item); + encoderSpinner.setAdapter(encoderAdapter); + encoderSpinner.setOnItemSelectedListener(new AdapterView.OnItemSelectedListener() { + @Override + public void onItemSelected(AdapterView<?> parent, View view, int position, long id) { + voipClient.setEncoder((String) parent.getSelectedItem()); + } + @Override + public void onNothingSelected(AdapterView<?> parent) {} + }); + } + + private List<String> getSelectedDecoders() { + List<String> decoders = new ArrayList<>(); + for (int i = 0; i < supportedCodecs.size(); i++) { + if (selectedDecoders.contains(i)) { + decoders.add(supportedCodecs.get(i)); + } + } + return decoders; + } + + private void setUpDecoderSelectionButton(List<String> supportedCodecs) { + decoderSelectionButton.setOnClickListener((view) -> { + AlertDialog.Builder dialogBuilder = new AlertDialog.Builder(this); + dialogBuilder.setTitle(R.string.dialog_title); + + // Populate multi choice items with supported decoders. + String[] supportedCodecsArray = supportedCodecs.toArray(new String[0]); + dialogBuilder.setMultiChoiceItems( + supportedCodecsArray, isDecoderSelected, (dialog, position, isChecked) -> { + if (isChecked) { + selectedDecoders.add(position); + } else if (!isChecked) { + selectedDecoders.remove(position); + } + }); + + // "Ok" button. + dialogBuilder.setPositiveButton(R.string.ok_label, (dialog, position) -> { + List<String> decoders = getSelectedDecoders(); + String result = decoders.stream().collect(Collectors.joining(", ")); + if (result.isEmpty()) { + decodersTextView.setText(R.string.decoders_text_view_default); + } else { + decodersTextView.setText(result); + } + voipClient.setDecoders(decoders); + }); + + // "Dismiss" button. + dialogBuilder.setNegativeButton( + R.string.dismiss_label, (dialog, position) -> { dialog.dismiss(); }); + + // "Clear All" button. + dialogBuilder.setNeutralButton(R.string.clear_all_label, (dialog, position) -> { + Arrays.fill(isDecoderSelected, false); + selectedDecoders.clear(); + decodersTextView.setText(R.string.decoders_text_view_default); + }); + + AlertDialog dialog = dialogBuilder.create(); + dialog.show(); + }); + } + + private void setUpSessionButton() { + sessionButton.setOnCheckedChangeListener((button, isChecked) -> { + // Ask for permission on RECORD_AUDIO if not granted. + if (ContextCompat.checkSelfPermission(this, permission.RECORD_AUDIO) + != PackageManager.PERMISSION_GRANTED) { + String[] sList = {permission.RECORD_AUDIO}; + ActivityCompat.requestPermissions(this, sList, 1); + } + + if (isChecked) { + // Order matters here, addresses have to be set before starting session + // before setting codec. + voipClient.setLocalAddress(localIPAddressTextView.getText().toString(), + Integer.parseInt(localPortNumberEditText.getText().toString())); + voipClient.setRemoteAddress(remoteIPAddressEditText.getText().toString(), + Integer.parseInt(remotePortNumberEditText.getText().toString())); + voipClient.startSession(); + voipClient.setEncoder((String) encoderSpinner.getSelectedItem()); + voipClient.setDecoders(getSelectedDecoders()); + } else { + voipClient.stopSession(); + } + }); + } + + private void setUpSendAndPlayoutSwitch() { + sendSwitch.setOnCheckedChangeListener((button, isChecked) -> { + if (isChecked) { + voipClient.startSend(); + } else { + voipClient.stopSend(); + } + }); + + playoutSwitch.setOnCheckedChangeListener((button, isChecked) -> { + if (isChecked) { + voipClient.startPlayout(); + } else { + voipClient.stopPlayout(); + } + }); + } + + private void setUpIPAddressEditTexts(String localIPAddress) { + if (localIPAddress.isEmpty()) { + showToast("Please check your network configuration"); + } else { + localIPAddressTextView.setText(localIPAddress); + // By default remote IP address is the same as local IP address. + remoteIPAddressEditText.setText(localIPAddress); + } + } + + private void showToast(String message) { + if (toast != null) { + toast.cancel(); + toast = Toast.makeText(this, message, Toast.LENGTH_SHORT); + toast.setGravity(Gravity.TOP, 0, 200); + toast.show(); + } + } + + @Override + protected void onDestroy() { + voipClient.close(); + voipClient = null; + + super.onDestroy(); + } + + @Override + public void onGetLocalIPAddressCompleted(String localIPAddress) { + runOnUiThread(() -> { setUpIPAddressEditTexts(localIPAddress); }); + } + + @Override + public void onGetSupportedCodecsCompleted(List<String> supportedCodecs) { + runOnUiThread(() -> { + this.supportedCodecs = supportedCodecs; + setUpEncoderSpinner(supportedCodecs); + setUpDecoderSelectionButton(supportedCodecs); + }); + } + + @Override + public void onVoipClientInitializationCompleted(boolean isSuccessful) { + runOnUiThread(() -> { + if (!isSuccessful) { + showToast("Error initializing audio device"); + } + }); + } + + @Override + public void onStartSessionCompleted(boolean isSuccessful) { + runOnUiThread(() -> { + if (isSuccessful) { + showToast("Session started"); + switchLayout.setVisibility(View.VISIBLE); + scrollView.post(() -> { scrollView.fullScroll(ScrollView.FOCUS_DOWN); }); + } else { + showToast("Failed to start session"); + } + }); + } + + @Override + public void onStopSessionCompleted(boolean isSuccessful) { + runOnUiThread(() -> { + if (isSuccessful) { + showToast("Session stopped"); + // Set listeners to null so the checked state can be changed programmatically. + sendSwitch.setOnCheckedChangeListener(null); + playoutSwitch.setOnCheckedChangeListener(null); + sendSwitch.setChecked(false); + playoutSwitch.setChecked(false); + // Redo the switch listener setup. + setUpSendAndPlayoutSwitch(); + switchLayout.setVisibility(View.GONE); + } else { + showToast("Failed to stop session"); + } + }); + } + + @Override + public void onStartSendCompleted(boolean isSuccessful) { + runOnUiThread(() -> { + if (isSuccessful) { + showToast("Started sending"); + } else { + showToast("Error initializing microphone"); + } + }); + } + + @Override + public void onStopSendCompleted(boolean isSuccessful) { + runOnUiThread(() -> { + if (isSuccessful) { + showToast("Stopped sending"); + } else { + showToast("Microphone termination failed"); + } + }); + } + + @Override + public void onStartPlayoutCompleted(boolean isSuccessful) { + runOnUiThread(() -> { + if (isSuccessful) { + showToast("Started playout"); + } else { + showToast("Error initializing speaker"); + } + }); + } + + @Override + public void onStopPlayoutCompleted(boolean isSuccessful) { + runOnUiThread(() -> { + if (isSuccessful) { + showToast("Stopped playout"); + } else { + showToast("Speaker termination failed"); + } + }); + } + + @Override + public void onUninitializedVoipClient() { + runOnUiThread(() -> { showToast("Voip client is uninitialized"); }); + } +} diff --git a/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/OnVoipClientTaskCompleted.java b/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/OnVoipClientTaskCompleted.java new file mode 100644 index 0000000000..bb85e048bb --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/OnVoipClientTaskCompleted.java @@ -0,0 +1,26 @@ +/* + * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.webrtc.examples.androidvoip; + +import java.util.List; + +public interface OnVoipClientTaskCompleted { + void onGetLocalIPAddressCompleted(String localIPAddress); + void onGetSupportedCodecsCompleted(List<String> supportedCodecs); + void onVoipClientInitializationCompleted(boolean isSuccessful); + void onStartSessionCompleted(boolean isSuccessful); + void onStopSessionCompleted(boolean isSuccessful); + void onStartSendCompleted(boolean isSuccessful); + void onStopSendCompleted(boolean isSuccessful); + void onStartPlayoutCompleted(boolean isSuccessful); + void onStopPlayoutCompleted(boolean isSuccessful); + void onUninitializedVoipClient(); +} diff --git a/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/VoipClient.java b/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/VoipClient.java new file mode 100644 index 0000000000..69a993d344 --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/VoipClient.java @@ -0,0 +1,191 @@ +/* + * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.webrtc.examples.androidvoip; + +import android.content.Context; +import android.os.Handler; +import android.os.HandlerThread; +import java.util.ArrayList; +import java.util.List; +import org.webrtc.CalledByNative; + +public class VoipClient { + private long nativeClient; + private OnVoipClientTaskCompleted listener; + + public VoipClient(Context applicationContext, OnVoipClientTaskCompleted listener) { + this.listener = listener; + nativeClient = nativeCreateClient(applicationContext, this); + } + + private boolean isInitialized() { + return nativeClient != 0; + } + + public void getAndSetUpSupportedCodecs() { + if (isInitialized()) { + nativeGetSupportedCodecs(nativeClient); + } else { + listener.onUninitializedVoipClient(); + } + } + + public void getAndSetUpLocalIPAddress() { + if (isInitialized()) { + nativeGetLocalIPAddress(nativeClient); + } else { + listener.onUninitializedVoipClient(); + } + } + + public void setEncoder(String encoder) { + if (isInitialized()) { + nativeSetEncoder(nativeClient, encoder); + } else { + listener.onUninitializedVoipClient(); + } + } + + public void setDecoders(List<String> decoders) { + if (isInitialized()) { + nativeSetDecoders(nativeClient, decoders); + } else { + listener.onUninitializedVoipClient(); + } + } + + public void setLocalAddress(String ipAddress, int portNumber) { + if (isInitialized()) { + nativeSetLocalAddress(nativeClient, ipAddress, portNumber); + } else { + listener.onUninitializedVoipClient(); + } + } + + public void setRemoteAddress(String ipAddress, int portNumber) { + if (isInitialized()) { + nativeSetRemoteAddress(nativeClient, ipAddress, portNumber); + } else { + listener.onUninitializedVoipClient(); + } + } + + public void startSession() { + if (isInitialized()) { + nativeStartSession(nativeClient); + } else { + listener.onUninitializedVoipClient(); + } + } + + public void stopSession() { + if (isInitialized()) { + nativeStopSession(nativeClient); + } else { + listener.onUninitializedVoipClient(); + } + } + + public void startSend() { + if (isInitialized()) { + nativeStartSend(nativeClient); + } else { + listener.onUninitializedVoipClient(); + } + } + + public void stopSend() { + if (isInitialized()) { + nativeStopSend(nativeClient); + } else { + listener.onUninitializedVoipClient(); + } + } + + public void startPlayout() { + if (isInitialized()) { + nativeStartPlayout(nativeClient); + } else { + listener.onUninitializedVoipClient(); + } + } + + public void stopPlayout() { + if (isInitialized()) { + nativeStopPlayout(nativeClient); + } else { + listener.onUninitializedVoipClient(); + } + } + + public void close() { + nativeDelete(nativeClient); + nativeClient = 0; + } + + @CalledByNative + public void onGetLocalIPAddressCompleted(String localIPAddress) { + listener.onGetLocalIPAddressCompleted(localIPAddress); + } + + @CalledByNative + public void onGetSupportedCodecsCompleted(List<String> supportedCodecs) { + listener.onGetSupportedCodecsCompleted(supportedCodecs); + } + + @CalledByNative + public void onStartSessionCompleted(boolean isSuccessful) { + listener.onStartSessionCompleted(isSuccessful); + } + + @CalledByNative + public void onStopSessionCompleted(boolean isSuccessful) { + listener.onStopSessionCompleted(isSuccessful); + } + + @CalledByNative + public void onStartSendCompleted(boolean isSuccessful) { + listener.onStartSendCompleted(isSuccessful); + } + + @CalledByNative + public void onStopSendCompleted(boolean isSuccessful) { + listener.onStopSendCompleted(isSuccessful); + } + + @CalledByNative + public void onStartPlayoutCompleted(boolean isSuccessful) { + listener.onStartPlayoutCompleted(isSuccessful); + } + + @CalledByNative + public void onStopPlayoutCompleted(boolean isSuccessful) { + listener.onStopPlayoutCompleted(isSuccessful); + } + + private static native long nativeCreateClient( + Context applicationContext, VoipClient javaVoipClient); + private static native void nativeGetSupportedCodecs(long nativeAndroidVoipClient); + private static native void nativeGetLocalIPAddress(long nativeAndroidVoipClient); + private static native void nativeSetEncoder(long nativeAndroidVoipClient, String encoder); + private static native void nativeSetDecoders(long nativeAndroidVoipClient, List<String> decoders); + private static native void nativeSetLocalAddress( + long nativeAndroidVoipClient, String ipAddress, int portNumber); + private static native void nativeSetRemoteAddress( + long nativeAndroidVoipClient, String ipAddress, int portNumber); + private static native void nativeStartSession(long nativeAndroidVoipClient); + private static native void nativeStopSession(long nativeAndroidVoipClient); + private static native void nativeStartSend(long nativeAndroidVoipClient); + private static native void nativeStopSend(long nativeAndroidVoipClient); + private static native void nativeStartPlayout(long nativeAndroidVoipClient); + private static native void nativeStopPlayout(long nativeAndroidVoipClient); + private static native void nativeDelete(long nativeAndroidVoipClient); +} diff --git a/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.cc b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.cc new file mode 100644 index 0000000000..cf07e87e50 --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.cc @@ -0,0 +1,514 @@ +/* + * Copyright 2020 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/androidvoip/jni/android_voip_client.h" + +#include <errno.h> +#include <sys/socket.h> +#include <algorithm> +#include <map> +#include <memory> +#include <unordered_map> +#include <unordered_set> +#include <utility> +#include <vector> + +#include "absl/memory/memory.h" +#include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include "api/task_queue/default_task_queue_factory.h" +#include "api/voip/voip_codec.h" +#include "api/voip/voip_engine_factory.h" +#include "api/voip/voip_network.h" +#include "examples/androidvoip/generated_jni/VoipClient_jni.h" +#include "rtc_base/logging.h" +#include "rtc_base/network.h" +#include "rtc_base/socket_server.h" +#include "sdk/android/native_api/audio_device_module/audio_device_android.h" +#include "sdk/android/native_api/jni/java_types.h" +#include "sdk/android/native_api/jni/jvm.h" +#include "sdk/android/native_api/jni/scoped_java_ref.h" + +namespace { + +#define RUN_ON_VOIP_THREAD(method, ...) \ + if (!voip_thread_->IsCurrent()) { \ + voip_thread_->PostTask( \ + std::bind(&AndroidVoipClient::method, this, ##__VA_ARGS__)); \ + return; \ + } \ + RTC_DCHECK_RUN_ON(voip_thread_.get()); + +// Connects a UDP socket to a public address and returns the local +// address associated with it. Since it binds to the "any" address +// internally, it returns the default local address on a multi-homed +// endpoint. Implementation copied from +// BasicNetworkManager::QueryDefaultLocalAddress. +rtc::IPAddress QueryDefaultLocalAddress(int family) { + const char kPublicIPv4Host[] = "8.8.8.8"; + const char kPublicIPv6Host[] = "2001:4860:4860::8888"; + const int kPublicPort = 53; + std::unique_ptr<rtc::Thread> thread = rtc::Thread::CreateWithSocketServer(); + + RTC_DCHECK(thread->socketserver() != nullptr); + RTC_DCHECK(family == AF_INET || family == AF_INET6); + + std::unique_ptr<rtc::Socket> socket( + thread->socketserver()->CreateSocket(family, SOCK_DGRAM)); + if (!socket) { + RTC_LOG_ERR(LS_ERROR) << "Socket creation failed"; + return rtc::IPAddress(); + } + + auto host = family == AF_INET ? kPublicIPv4Host : kPublicIPv6Host; + if (socket->Connect(rtc::SocketAddress(host, kPublicPort)) < 0) { + if (socket->GetError() != ENETUNREACH && + socket->GetError() != EHOSTUNREACH) { + RTC_LOG(LS_INFO) << "Connect failed with " << socket->GetError(); + } + return rtc::IPAddress(); + } + return socket->GetLocalAddress().ipaddr(); +} + +// Assigned payload type for supported built-in codecs. PCMU, PCMA, +// and G722 have set payload types. Whereas opus, ISAC, and ILBC +// have dynamic payload types. +enum class PayloadType : int { + kPcmu = 0, + kPcma = 8, + kG722 = 9, + kOpus = 96, + kIsac = 97, + kIlbc = 98, +}; + +// Returns the payload type corresponding to codec_name. Only +// supports the built-in codecs. +int GetPayloadType(const std::string& codec_name) { + RTC_DCHECK(codec_name == "PCMU" || codec_name == "PCMA" || + codec_name == "G722" || codec_name == "opus" || + codec_name == "ISAC" || codec_name == "ILBC"); + + if (codec_name == "PCMU") { + return static_cast<int>(PayloadType::kPcmu); + } else if (codec_name == "PCMA") { + return static_cast<int>(PayloadType::kPcma); + } else if (codec_name == "G722") { + return static_cast<int>(PayloadType::kG722); + } else if (codec_name == "opus") { + return static_cast<int>(PayloadType::kOpus); + } else if (codec_name == "ISAC") { + return static_cast<int>(PayloadType::kIsac); + } else if (codec_name == "ILBC") { + return static_cast<int>(PayloadType::kIlbc); + } + + RTC_DCHECK_NOTREACHED(); + return -1; +} + +} // namespace + +namespace webrtc_examples { + +void AndroidVoipClient::Init( + JNIEnv* env, + const webrtc::JavaParamRef<jobject>& application_context) { + webrtc::VoipEngineConfig config; + config.encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory(); + config.decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory(); + config.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory(); + config.audio_device_module = + webrtc::CreateJavaAudioDeviceModule(env, application_context.obj()); + config.audio_processing = webrtc::AudioProcessingBuilder().Create(); + + voip_thread_->Start(); + + // Due to consistent thread requirement on + // modules/audio_device/android/audio_device_template.h, + // code is invoked in the context of voip_thread_. + voip_thread_->BlockingCall([this, &config] { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + supported_codecs_ = config.encoder_factory->GetSupportedEncoders(); + env_ = webrtc::AttachCurrentThreadIfNeeded(); + voip_engine_ = webrtc::CreateVoipEngine(std::move(config)); + }); +} + +AndroidVoipClient::~AndroidVoipClient() { + voip_thread_->BlockingCall([this] { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + JavaVM* jvm = nullptr; + env_->GetJavaVM(&jvm); + if (!jvm) { + RTC_LOG(LS_ERROR) << "Failed to retrieve JVM"; + return; + } + jint res = jvm->DetachCurrentThread(); + if (res != JNI_OK) { + RTC_LOG(LS_ERROR) << "DetachCurrentThread failed: " << res; + } + }); + + voip_thread_->Stop(); +} + +AndroidVoipClient* AndroidVoipClient::Create( + JNIEnv* env, + const webrtc::JavaParamRef<jobject>& application_context, + const webrtc::JavaParamRef<jobject>& j_voip_client) { + // Using `new` to access a non-public constructor. + auto voip_client = + absl::WrapUnique(new AndroidVoipClient(env, j_voip_client)); + voip_client->Init(env, application_context); + return voip_client.release(); +} + +void AndroidVoipClient::GetSupportedCodecs(JNIEnv* env) { + RUN_ON_VOIP_THREAD(GetSupportedCodecs, env); + + std::vector<std::string> names; + for (const webrtc::AudioCodecSpec& spec : supported_codecs_) { + names.push_back(spec.format.name); + } + webrtc::ScopedJavaLocalRef<jstring> (*convert_function)( + JNIEnv*, const std::string&) = &webrtc::NativeToJavaString; + Java_VoipClient_onGetSupportedCodecsCompleted( + env_, j_voip_client_, NativeToJavaList(env_, names, convert_function)); +} + +void AndroidVoipClient::GetLocalIPAddress(JNIEnv* env) { + RUN_ON_VOIP_THREAD(GetLocalIPAddress, env); + + std::string local_ip_address; + rtc::IPAddress ipv4_address = QueryDefaultLocalAddress(AF_INET); + if (!ipv4_address.IsNil()) { + local_ip_address = ipv4_address.ToString(); + } else { + rtc::IPAddress ipv6_address = QueryDefaultLocalAddress(AF_INET6); + if (!ipv6_address.IsNil()) { + local_ip_address = ipv6_address.ToString(); + } + } + Java_VoipClient_onGetLocalIPAddressCompleted( + env_, j_voip_client_, webrtc::NativeToJavaString(env_, local_ip_address)); +} + +void AndroidVoipClient::SetEncoder(const std::string& encoder) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + return; + } + for (const webrtc::AudioCodecSpec& codec : supported_codecs_) { + if (codec.format.name == encoder) { + webrtc::VoipResult result = voip_engine_->Codec().SetSendCodec( + *channel_, GetPayloadType(codec.format.name), codec.format); + RTC_CHECK(result == webrtc::VoipResult::kOk); + return; + } + } +} + +void AndroidVoipClient::SetEncoder( + JNIEnv* env, + const webrtc::JavaParamRef<jstring>& j_encoder_string) { + const std::string& chosen_encoder = + webrtc::JavaToNativeString(env, j_encoder_string); + voip_thread_->PostTask( + [this, chosen_encoder] { SetEncoder(chosen_encoder); }); +} + +void AndroidVoipClient::SetDecoders(const std::vector<std::string>& decoders) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + return; + } + std::map<int, webrtc::SdpAudioFormat> decoder_specs; + for (const webrtc::AudioCodecSpec& codec : supported_codecs_) { + if (std::find(decoders.begin(), decoders.end(), codec.format.name) != + decoders.end()) { + decoder_specs.insert({GetPayloadType(codec.format.name), codec.format}); + } + } + + webrtc::VoipResult result = + voip_engine_->Codec().SetReceiveCodecs(*channel_, decoder_specs); + RTC_CHECK(result == webrtc::VoipResult::kOk); +} + +void AndroidVoipClient::SetDecoders( + JNIEnv* env, + const webrtc::JavaParamRef<jobject>& j_decoder_strings) { + const std::vector<std::string>& chosen_decoders = + webrtc::JavaListToNativeVector<std::string, jstring>( + env, j_decoder_strings, &webrtc::JavaToNativeString); + voip_thread_->PostTask( + [this, chosen_decoders] { SetDecoders(chosen_decoders); }); +} + +void AndroidVoipClient::SetLocalAddress(const std::string& ip_address, + const int port_number) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + rtp_local_address_ = rtc::SocketAddress(ip_address, port_number); + rtcp_local_address_ = rtc::SocketAddress(ip_address, port_number + 1); +} + +void AndroidVoipClient::SetLocalAddress( + JNIEnv* env, + const webrtc::JavaParamRef<jstring>& j_ip_address_string, + jint j_port_number_int) { + const std::string& ip_address = + webrtc::JavaToNativeString(env, j_ip_address_string); + voip_thread_->PostTask([this, ip_address, j_port_number_int] { + SetLocalAddress(ip_address, j_port_number_int); + }); +} + +void AndroidVoipClient::SetRemoteAddress(const std::string& ip_address, + const int port_number) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + rtp_remote_address_ = rtc::SocketAddress(ip_address, port_number); + rtcp_remote_address_ = rtc::SocketAddress(ip_address, port_number + 1); +} + +void AndroidVoipClient::SetRemoteAddress( + JNIEnv* env, + const webrtc::JavaParamRef<jstring>& j_ip_address_string, + jint j_port_number_int) { + const std::string& ip_address = + webrtc::JavaToNativeString(env, j_ip_address_string); + voip_thread_->PostTask([this, ip_address, j_port_number_int] { + SetRemoteAddress(ip_address, j_port_number_int); + }); +} + +void AndroidVoipClient::StartSession(JNIEnv* env) { + RUN_ON_VOIP_THREAD(StartSession, env); + + // CreateChannel guarantees to return valid channel id. + channel_ = voip_engine_->Base().CreateChannel(this, absl::nullopt); + + rtp_socket_.reset(rtc::AsyncUDPSocket::Create(voip_thread_->socketserver(), + rtp_local_address_)); + if (!rtp_socket_) { + RTC_LOG_ERR(LS_ERROR) << "Socket creation failed"; + Java_VoipClient_onStartSessionCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + rtp_socket_->SignalReadPacket.connect( + this, &AndroidVoipClient::OnSignalReadRTPPacket); + + rtcp_socket_.reset(rtc::AsyncUDPSocket::Create(voip_thread_->socketserver(), + rtcp_local_address_)); + if (!rtcp_socket_) { + RTC_LOG_ERR(LS_ERROR) << "Socket creation failed"; + Java_VoipClient_onStartSessionCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + rtcp_socket_->SignalReadPacket.connect( + this, &AndroidVoipClient::OnSignalReadRTCPPacket); + Java_VoipClient_onStartSessionCompleted(env_, j_voip_client_, + /*isSuccessful=*/true); +} + +void AndroidVoipClient::StopSession(JNIEnv* env) { + RUN_ON_VOIP_THREAD(StopSession, env); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + Java_VoipClient_onStopSessionCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + if (voip_engine_->Base().StopSend(*channel_) != webrtc::VoipResult::kOk || + voip_engine_->Base().StopPlayout(*channel_) != webrtc::VoipResult::kOk) { + Java_VoipClient_onStopSessionCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + + rtp_socket_->Close(); + rtcp_socket_->Close(); + + webrtc::VoipResult result = voip_engine_->Base().ReleaseChannel(*channel_); + RTC_CHECK(result == webrtc::VoipResult::kOk); + + channel_ = absl::nullopt; + Java_VoipClient_onStopSessionCompleted(env_, j_voip_client_, + /*isSuccessful=*/true); +} + +void AndroidVoipClient::StartSend(JNIEnv* env) { + RUN_ON_VOIP_THREAD(StartSend, env); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + Java_VoipClient_onStartSendCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + bool sending_started = + (voip_engine_->Base().StartSend(*channel_) == webrtc::VoipResult::kOk); + Java_VoipClient_onStartSendCompleted(env_, j_voip_client_, sending_started); +} + +void AndroidVoipClient::StopSend(JNIEnv* env) { + RUN_ON_VOIP_THREAD(StopSend, env); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + Java_VoipClient_onStopSendCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + bool sending_stopped = + (voip_engine_->Base().StopSend(*channel_) == webrtc::VoipResult::kOk); + Java_VoipClient_onStopSendCompleted(env_, j_voip_client_, sending_stopped); +} + +void AndroidVoipClient::StartPlayout(JNIEnv* env) { + RUN_ON_VOIP_THREAD(StartPlayout, env); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + Java_VoipClient_onStartPlayoutCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + bool playout_started = + (voip_engine_->Base().StartPlayout(*channel_) == webrtc::VoipResult::kOk); + Java_VoipClient_onStartPlayoutCompleted(env_, j_voip_client_, + playout_started); +} + +void AndroidVoipClient::StopPlayout(JNIEnv* env) { + RUN_ON_VOIP_THREAD(StopPlayout, env); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + Java_VoipClient_onStopPlayoutCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + bool playout_stopped = + (voip_engine_->Base().StopPlayout(*channel_) == webrtc::VoipResult::kOk); + Java_VoipClient_onStopPlayoutCompleted(env_, j_voip_client_, playout_stopped); +} + +void AndroidVoipClient::Delete(JNIEnv* env) { + delete this; +} + +void AndroidVoipClient::SendRtpPacket(const std::vector<uint8_t>& packet_copy) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + if (!rtp_socket_->SendTo(packet_copy.data(), packet_copy.size(), + rtp_remote_address_, rtc::PacketOptions())) { + RTC_LOG(LS_ERROR) << "Failed to send RTP packet"; + } +} + +bool AndroidVoipClient::SendRtp(const uint8_t* packet, + size_t length, + const webrtc::PacketOptions& options) { + std::vector<uint8_t> packet_copy(packet, packet + length); + voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] { + SendRtpPacket(packet_copy); + }); + return true; +} + +void AndroidVoipClient::SendRtcpPacket( + const std::vector<uint8_t>& packet_copy) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + if (!rtcp_socket_->SendTo(packet_copy.data(), packet_copy.size(), + rtcp_remote_address_, rtc::PacketOptions())) { + RTC_LOG(LS_ERROR) << "Failed to send RTCP packet"; + } +} + +bool AndroidVoipClient::SendRtcp(const uint8_t* packet, size_t length) { + std::vector<uint8_t> packet_copy(packet, packet + length); + voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] { + SendRtcpPacket(packet_copy); + }); + return true; +} + +void AndroidVoipClient::ReadRTPPacket(const std::vector<uint8_t>& packet_copy) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + return; + } + webrtc::VoipResult result = voip_engine_->Network().ReceivedRTPPacket( + *channel_, + rtc::ArrayView<const uint8_t>(packet_copy.data(), packet_copy.size())); + RTC_CHECK(result == webrtc::VoipResult::kOk); +} + +void AndroidVoipClient::OnSignalReadRTPPacket(rtc::AsyncPacketSocket* socket, + const char* rtp_packet, + size_t size, + const rtc::SocketAddress& addr, + const int64_t& timestamp) { + std::vector<uint8_t> packet_copy(rtp_packet, rtp_packet + size); + voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] { + ReadRTPPacket(packet_copy); + }); +} + +void AndroidVoipClient::ReadRTCPPacket( + const std::vector<uint8_t>& packet_copy) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + return; + } + webrtc::VoipResult result = voip_engine_->Network().ReceivedRTCPPacket( + *channel_, + rtc::ArrayView<const uint8_t>(packet_copy.data(), packet_copy.size())); + RTC_CHECK(result == webrtc::VoipResult::kOk); +} + +void AndroidVoipClient::OnSignalReadRTCPPacket(rtc::AsyncPacketSocket* socket, + const char* rtcp_packet, + size_t size, + const rtc::SocketAddress& addr, + const int64_t& timestamp) { + std::vector<uint8_t> packet_copy(rtcp_packet, rtcp_packet + size); + voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] { + ReadRTCPPacket(packet_copy); + }); +} + +static jlong JNI_VoipClient_CreateClient( + JNIEnv* env, + const webrtc::JavaParamRef<jobject>& application_context, + const webrtc::JavaParamRef<jobject>& j_voip_client) { + return webrtc::NativeToJavaPointer( + AndroidVoipClient::Create(env, application_context, j_voip_client)); +} + +} // namespace webrtc_examples diff --git a/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.h b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.h new file mode 100644 index 0000000000..8e1edd5ef9 --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.h @@ -0,0 +1,189 @@ +/* + * Copyright 2020 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef EXAMPLES_ANDROIDVOIP_JNI_ANDROID_VOIP_CLIENT_H_ +#define EXAMPLES_ANDROIDVOIP_JNI_ANDROID_VOIP_CLIENT_H_ + +#include <jni.h> + +#include <memory> +#include <string> +#include <vector> + +#include "api/audio_codecs/audio_format.h" +#include "api/call/transport.h" +#include "api/voip/voip_base.h" +#include "api/voip/voip_engine.h" +#include "rtc_base/async_packet_socket.h" +#include "rtc_base/async_udp_socket.h" +#include "rtc_base/socket_address.h" +#include "rtc_base/third_party/sigslot/sigslot.h" +#include "rtc_base/thread.h" +#include "sdk/android/native_api/jni/scoped_java_ref.h" + +namespace webrtc_examples { + +// AndroidVoipClient facilitates the use of the VoIP API defined in +// api/voip/voip_engine.h. One instance of AndroidVoipClient should +// suffice for most VoIP applications. AndroidVoipClient implements +// webrtc::Transport to send RTP/RTCP packets to the remote endpoint. +// It also creates methods (slots) for sockets to connect to in +// order to receive RTP/RTCP packets. AndroidVoipClient does all +// operations with rtc::Thread (voip_thread_), this is to comply +// with consistent thread usage requirement with ProcessThread used +// within VoipEngine, as well as providing asynchronicity to the +// caller. AndroidVoipClient is meant to be used by Java through JNI. +class AndroidVoipClient : public webrtc::Transport, + public sigslot::has_slots<> { + public: + // Returns a pointer to an AndroidVoipClient object. Clients should + // use this factory method to create AndroidVoipClient objects. The + // method will return a nullptr in case of initialization errors. + // It is the client's responsibility to delete the pointer when + // they are done with it (this class provides a Delete() method). + static AndroidVoipClient* Create( + JNIEnv* env, + const webrtc::JavaParamRef<jobject>& application_context, + const webrtc::JavaParamRef<jobject>& j_voip_client); + + ~AndroidVoipClient() override; + + // Provides client with a Java List of Strings containing names of + // the built-in supported codecs through callback. + void GetSupportedCodecs(JNIEnv* env); + + // Provides client with a Java String of the default local IPv4 address + // through callback. If IPv4 address is not found, provide the default + // local IPv6 address. If IPv6 address is not found, provide an empty + // string. + void GetLocalIPAddress(JNIEnv* env); + + // Sets the encoder used by the VoIP API. + void SetEncoder(JNIEnv* env, + const webrtc::JavaParamRef<jstring>& j_encoder_string); + + // Sets the decoders used by the VoIP API. + void SetDecoders(JNIEnv* env, + const webrtc::JavaParamRef<jobject>& j_decoder_strings); + + // Sets two local/remote addresses, one for RTP packets, and another for + // RTCP packets. The RTP address will have IP address j_ip_address_string + // and port number j_port_number_int, the RTCP address will have IP address + // j_ip_address_string and port number j_port_number_int+1. + void SetLocalAddress(JNIEnv* env, + const webrtc::JavaParamRef<jstring>& j_ip_address_string, + jint j_port_number_int); + void SetRemoteAddress( + JNIEnv* env, + const webrtc::JavaParamRef<jstring>& j_ip_address_string, + jint j_port_number_int); + + // Starts a VoIP session, then calls a callback method with a boolean + // value indicating if the session has started successfully. The VoIP + // operations below can only be used after a session has already started. + void StartSession(JNIEnv* env); + + // Stops the current session, then calls a callback method with a + // boolean value indicating if the session has stopped successfully. + void StopSession(JNIEnv* env); + + // Starts sending RTP/RTCP packets to the remote endpoint, then calls + // a callback method with a boolean value indicating if sending + // has started successfully. + void StartSend(JNIEnv* env); + + // Stops sending RTP/RTCP packets to the remote endpoint, then calls + // a callback method with a boolean value indicating if sending + // has stopped successfully. + void StopSend(JNIEnv* env); + + // Starts playing out the voice data received from the remote endpoint, + // then calls a callback method with a boolean value indicating if + // playout has started successfully. + void StartPlayout(JNIEnv* env); + + // Stops playing out the voice data received from the remote endpoint, + // then calls a callback method with a boolean value indicating if + // playout has stopped successfully. + void StopPlayout(JNIEnv* env); + + // Deletes this object. Used by client when they are done. + void Delete(JNIEnv* env); + + // Implementation for Transport. + bool SendRtp(const uint8_t* packet, + size_t length, + const webrtc::PacketOptions& options) override; + bool SendRtcp(const uint8_t* packet, size_t length) override; + + // Slots for sockets to connect to. + void OnSignalReadRTPPacket(rtc::AsyncPacketSocket* socket, + const char* rtp_packet, + size_t size, + const rtc::SocketAddress& addr, + const int64_t& timestamp); + void OnSignalReadRTCPPacket(rtc::AsyncPacketSocket* socket, + const char* rtcp_packet, + size_t size, + const rtc::SocketAddress& addr, + const int64_t& timestamp); + + private: + AndroidVoipClient(JNIEnv* env, + const webrtc::JavaParamRef<jobject>& j_voip_client) + : voip_thread_(rtc::Thread::CreateWithSocketServer()), + j_voip_client_(env, j_voip_client) {} + + void Init(JNIEnv* env, + const webrtc::JavaParamRef<jobject>& application_context); + + // Overloaded methods having native C++ variables as arguments. + void SetEncoder(const std::string& encoder); + void SetDecoders(const std::vector<std::string>& decoders); + void SetLocalAddress(const std::string& ip_address, int port_number); + void SetRemoteAddress(const std::string& ip_address, int port_number); + + // Methods to send and receive RTP/RTCP packets. Takes in a + // copy of a packet as a vector to prolong the lifetime of + // the packet as these methods will be called asynchronously. + void SendRtpPacket(const std::vector<uint8_t>& packet_copy); + void SendRtcpPacket(const std::vector<uint8_t>& packet_copy); + void ReadRTPPacket(const std::vector<uint8_t>& packet_copy); + void ReadRTCPPacket(const std::vector<uint8_t>& packet_copy); + + // Used to invoke operations and send/receive RTP/RTCP packets. + std::unique_ptr<rtc::Thread> voip_thread_; + // Reference to the VoipClient java instance used to + // invoke callbacks when operations are finished. + webrtc::ScopedJavaGlobalRef<jobject> j_voip_client_ + RTC_GUARDED_BY(voip_thread_); + // A list of AudioCodecSpec supported by the built-in + // encoder/decoder factories. + std::vector<webrtc::AudioCodecSpec> supported_codecs_ + RTC_GUARDED_BY(voip_thread_); + // A JNI context used by the voip_thread_. + JNIEnv* env_ RTC_GUARDED_BY(voip_thread_); + // The entry point to all VoIP APIs. + std::unique_ptr<webrtc::VoipEngine> voip_engine_ RTC_GUARDED_BY(voip_thread_); + // Used by the VoIP API to facilitate a VoIP session. + absl::optional<webrtc::ChannelId> channel_ RTC_GUARDED_BY(voip_thread_); + // Members below are used for network related operations. + std::unique_ptr<rtc::AsyncUDPSocket> rtp_socket_ RTC_GUARDED_BY(voip_thread_); + std::unique_ptr<rtc::AsyncUDPSocket> rtcp_socket_ + RTC_GUARDED_BY(voip_thread_); + rtc::SocketAddress rtp_local_address_ RTC_GUARDED_BY(voip_thread_); + rtc::SocketAddress rtcp_local_address_ RTC_GUARDED_BY(voip_thread_); + rtc::SocketAddress rtp_remote_address_ RTC_GUARDED_BY(voip_thread_); + rtc::SocketAddress rtcp_remote_address_ RTC_GUARDED_BY(voip_thread_); +}; + +} // namespace webrtc_examples + +#endif // EXAMPLES_ANDROIDVOIP_JNI_ANDROID_VOIP_CLIENT_H_ diff --git a/third_party/libwebrtc/examples/androidvoip/jni/onload.cc b/third_party/libwebrtc/examples/androidvoip/jni/onload.cc new file mode 100644 index 0000000000..b952de348b --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/jni/onload.cc @@ -0,0 +1,28 @@ +/* + * Copyright 2020 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <jni.h> + +#include "rtc_base/ssl_adapter.h" +#include "sdk/android/native_api/base/init.h" + +namespace webrtc_examples { + +extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM* jvm, void* reserved) { + webrtc::InitAndroid(jvm); + RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()"; + return JNI_VERSION_1_6; +} + +extern "C" void JNIEXPORT JNICALL JNI_OnUnLoad(JavaVM* jvm, void* reserved) { + RTC_CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()"; +} + +} // namespace webrtc_examples diff --git a/third_party/libwebrtc/examples/androidvoip/res/layout/activity_main.xml b/third_party/libwebrtc/examples/androidvoip/res/layout/activity_main.xml new file mode 100644 index 0000000000..c7fa5a9b31 --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/res/layout/activity_main.xml @@ -0,0 +1,303 @@ +<?xml version="1.0" encoding="utf-8"?> +<ScrollView + xmlns:android="http://schemas.android.com/apk/res/android" + xmlns:app="http://schemas.android.com/apk/res-auto" + xmlns:tools="http://schemas.android.com/tools" + android:id="@+id/scroll_view" + android:layout_width="match_parent" + android:layout_height="wrap_content" + android:focusable="true" + android:focusableInTouchMode="true" + tools:context="org.webrtc.examples.androidvoip.MainActivity"> + + <LinearLayout + android:orientation="vertical" + android:layout_width="match_parent" + android:layout_height="match_parent" + android:padding="8dp"> + + <TextView + android:layout_width="match_parent" + android:layout_height="wrap_content" + android:layout_marginBottom="15dp" + android:layout_marginLeft="15dp" + android:layout_marginTop="15dp" + android:text="@string/local_endpoint_text_view" + android:textSize="19dp" + android:textStyle="bold" + android:textColor="@color/almost_black" /> + + <!--Local IP Adress--> + <LinearLayout + android:orientation="horizontal" + android:layout_width="match_parent" + android:layout_height="48dp" + android:layout_gravity="center_vertical" > + + <TextView + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_marginTop="12dp" + android:layout_marginLeft="15dp" + android:layout_marginRight="15dp" + android:text="@string/ip_address_text_view" + android:textSize="16dp" /> + + <TextView + android:id="@+id/local_ip_address_text_view" + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_marginTop="12dp" + android:layout_marginRight="15dp" + android:textSize="16dp" /> + + </LinearLayout> + + <!--Local Port Number--> + <LinearLayout + android:orientation="horizontal" + android:layout_width="match_parent" + android:layout_height="48dp" + android:layout_gravity="center_vertical"> + + <TextView + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_marginBottom="15dp" + android:layout_marginLeft="15dp" + android:layout_marginRight="15dp" + android:text="@string/port_number_text_view" + android:textSize="16dp" /> + + <EditText + android:id="@+id/local_port_number_edit_text" + android:layout_width="0dp" + android:layout_height="match_parent" + android:layout_marginRight="15dp" + android:layout_weight="1" + android:text="10000" + android:inputType="number" + android:textSize="16dp" /> + + </LinearLayout> + + <TextView + android:layout_width="match_parent" + android:layout_height="wrap_content" + android:layout_marginBottom="15dp" + android:layout_marginLeft="15dp" + android:layout_marginTop="30dp" + android:text="@string/remote_endpoint_text_view" + android:textSize="19dp" + android:textStyle="bold" + android:textColor="@color/almost_black" /> + + <!--Remote IP Adress--> + <LinearLayout + android:orientation="horizontal" + android:layout_width="match_parent" + android:layout_height="48dp" + android:layout_gravity="center_vertical"> + + <TextView + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_marginBottom="15dp" + android:layout_marginLeft="15dp" + android:layout_marginRight="15dp" + android:text="@string/ip_address_text_view" + android:textSize="16dp" /> + + <EditText + android:id="@+id/remote_ip_address_edit_text" + android:layout_width="0dp" + android:layout_height="wrap_content" + android:layout_marginRight="15dp" + android:layout_weight="1" + android:inputType="number" + android:digits="0123456789." + android:textSize="16dp" /> + + </LinearLayout> + + <!--Remote Port Number--> + <LinearLayout + android:orientation="horizontal" + android:layout_width="match_parent" + android:layout_height="48dp" + android:layout_gravity="center_vertical"> + + <TextView + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_marginBottom="15dp" + android:layout_marginLeft="15dp" + android:layout_marginRight="15dp" + android:text="@string/port_number_text_view" + android:textSize="16dp" /> + + <EditText + android:id="@+id/remote_port_number_edit_text" + android:layout_width="0dp" + android:layout_height="match_parent" + android:layout_marginRight="15dp" + android:layout_weight="1" + android:text="10000" + android:inputType="number" + android:textSize="16dp" /> + + </LinearLayout> + + <TextView + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_marginBottom="15dp" + android:layout_marginLeft="15dp" + android:layout_marginTop="30dp" + android:text="@string/encoder_text_view" + android:textSize="19dp" + android:textStyle="bold" + android:textColor="@color/almost_black" /> + + <Spinner + android:id="@+id/encoder_spinner" + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_marginBottom="15dp" + android:layout_marginLeft="15dp" + android:layout_marginTop="10dp"/> + + <LinearLayout + android:orientation="horizontal" + android:layout_width="match_parent" + android:layout_height="48dp" + android:layout_marginTop="20dp" + android:layout_gravity="center_vertical"> + + <TextView + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_marginBottom="15dp" + android:layout_marginLeft="15dp" + android:layout_marginRight="25dp" + android:text="@string/decoder_text_view" + android:textSize="19dp" + android:textStyle="bold" + android:textColor="@color/almost_black" /> + + <Button + android:id="@+id/decoder_selection_button" + android:text="@string/decoder_selection_button" + style="?android:attr/buttonBarButtonStyle" + android:layout_width="0dp" + android:layout_height="match_parent" + android:layout_marginRight="15dp" + android:layout_weight="1" /> + + </LinearLayout> + + + <TextView + android:id="@+id/decoders_text_view" + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_marginTop="15dp" + android:layout_marginBottom="30dp" + android:layout_marginLeft="15dp" + android:layout_marginRight="15dp" + android:text="@string/decoders_text_view_default" + android:textSize="16dp" /> + + + <RelativeLayout + android:id="@+id/switch_layout" + android:layout_width="fill_parent" + android:layout_height="wrap_content" + android:layout_marginTop="15dp" + android:visibility="gone" > + + <View + android:id="@+id/divider" + android:layout_width="match_parent" + android:layout_height="1dp" + android:layout_marginLeft="15dp" + android:layout_marginRight="15dp" + android:layout_marginBottom="45dp" + android:background="@color/light_gray" /> + + <LinearLayout + android:id="@+id/start_send_switch_layout" + android:orientation="horizontal" + android:layout_width="match_parent" + android:layout_height="48dp" + android:layout_gravity="center_vertical" + android:layout_below="@id/divider" > + + <TextView + android:layout_width="wrap_content" + android:layout_height="match_parent" + android:layout_marginLeft="15dp" + android:gravity="left" + android:layout_weight="1" + android:text="@string/start_send_text_view" + android:textSize="16dp" /> + + <Switch + android:id="@+id/start_send_switch" + android:layout_width="wrap_content" + android:layout_height="match_parent" + android:layout_marginRight="15dp" + android:gravity="right" + android:layout_weight="1" /> + + </LinearLayout> + + <LinearLayout + android:orientation="horizontal" + android:layout_width="match_parent" + android:layout_height="48dp" + android:layout_gravity="center_vertical" + android:layout_below="@id/start_send_switch_layout"> + + <TextView + android:id="@+id/start_playout_text_view" + android:layout_width="wrap_content" + android:layout_height="match_parent" + android:layout_marginLeft="15dp" + android:gravity="left" + android:layout_weight="1" + android:text="@string/start_playout_text_view" + android:textSize="16dp" /> + + <Switch + android:id="@+id/start_playout_switch" + android:layout_width="wrap_content" + android:layout_height="match_parent" + android:layout_marginRight="15dp" + android:gravity="right" + android:layout_weight="1" /> + + </LinearLayout> + + </RelativeLayout> + + <LinearLayout + android:layout_width="match_parent" + android:layout_height="match_parent" + android:gravity="center" + android:orientation="vertical" > + + <ToggleButton + android:id="@+id/session_button" + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_margin="8dp" + android:textOff="@string/session_button_text_off" + android:textOn="@string/session_button_text_on" + style="?android:attr/buttonStyle" /> + + </LinearLayout> + + </LinearLayout> + +</ScrollView> diff --git a/third_party/libwebrtc/examples/androidvoip/res/values/colors.xml b/third_party/libwebrtc/examples/androidvoip/res/values/colors.xml new file mode 100644 index 0000000000..4dadaa9941 --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/res/values/colors.xml @@ -0,0 +1,5 @@ +<?xml version="1.0" encoding="utf-8"?> +<resources> + <color name="almost_black">#484848</color> + <color name="light_gray">#D3D3D3</color> +</resources>
\ No newline at end of file diff --git a/third_party/libwebrtc/examples/androidvoip/res/values/strings.xml b/third_party/libwebrtc/examples/androidvoip/res/values/strings.xml new file mode 100644 index 0000000000..d519bfbbb6 --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/res/values/strings.xml @@ -0,0 +1,19 @@ +<resources> + <string name="app_name">androidvoip</string> + <string name="local_endpoint_text_view">Local Endpoint</string> + <string name="remote_endpoint_text_view">Remote Endpoint</string> + <string name="ip_address_text_view">IP Address:</string> + <string name="port_number_text_view">Port Number:</string> + <string name="encoder_text_view">Select Encoder</string> + <string name="decoder_text_view">Select Decoder</string> + <string name="decoder_selection_button">Configure Selection</string> + <string name="decoders_text_view_default">No decoders selected</string> + <string name="dialog_title">Choose Decoders</string> + <string name="ok_label">Ok</string> + <string name="dismiss_label">Dismiss</string> + <string name="clear_all_label">Clear All</string> + <string name="start_send_text_view">Start Sending</string> + <string name="start_playout_text_view">Start Playout</string> + <string name="session_button_text_off">Start Session</string> + <string name="session_button_text_on">Stop Session</string> +</resources> diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient+Internal.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient+Internal.h new file mode 100644 index 0000000000..31e0e4dd7c --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient+Internal.h @@ -0,0 +1,52 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDAppClient.h" + +#import "sdk/objc/api/peerconnection/RTCPeerConnection.h" + +#import "ARDRoomServerClient.h" +#import "ARDSignalingChannel.h" +#import "ARDTURNClient.h" + +@class RTC_OBJC_TYPE(RTCPeerConnectionFactory); + +@interface ARDAppClient () <ARDSignalingChannelDelegate, RTC_OBJC_TYPE (RTCPeerConnectionDelegate)> + +// All properties should only be mutated from the main queue. +@property(nonatomic, strong) id<ARDRoomServerClient> roomServerClient; +@property(nonatomic, strong) id<ARDSignalingChannel> channel; +@property(nonatomic, strong) id<ARDSignalingChannel> loopbackChannel; +@property(nonatomic, strong) id<ARDTURNClient> turnClient; + +@property(nonatomic, strong) RTC_OBJC_TYPE(RTCPeerConnection) * peerConnection; +@property(nonatomic, strong) RTC_OBJC_TYPE(RTCPeerConnectionFactory) * factory; +@property(nonatomic, strong) NSMutableArray *messageQueue; + +@property(nonatomic, assign) BOOL isTurnComplete; +@property(nonatomic, assign) BOOL hasReceivedSdp; +@property(nonatomic, readonly) BOOL hasJoinedRoomServerRoom; + +@property(nonatomic, strong) NSString *roomId; +@property(nonatomic, strong) NSString *clientId; +@property(nonatomic, assign) BOOL isInitiator; +@property(nonatomic, strong) NSMutableArray *iceServers; +@property(nonatomic, strong) NSURL *webSocketURL; +@property(nonatomic, strong) NSURL *webSocketRestURL; +@property(nonatomic, readonly) BOOL isLoopback; + +@property(nonatomic, strong) RTC_OBJC_TYPE(RTCMediaConstraints) * defaultPeerConnectionConstraints; + +- (instancetype)initWithRoomServerClient:(id<ARDRoomServerClient>)rsClient + signalingChannel:(id<ARDSignalingChannel>)channel + turnClient:(id<ARDTURNClient>)turnClient + delegate:(id<ARDAppClientDelegate>)delegate; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient.h new file mode 100644 index 0000000000..91d2cef1ce --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient.h @@ -0,0 +1,87 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> + +#import "sdk/objc/api/peerconnection/RTCPeerConnection.h" +#import "sdk/objc/api/peerconnection/RTCVideoTrack.h" + +typedef NS_ENUM(NSInteger, ARDAppClientState) { + // Disconnected from servers. + kARDAppClientStateDisconnected, + // Connecting to servers. + kARDAppClientStateConnecting, + // Connected to servers. + kARDAppClientStateConnected, +}; + +@class ARDAppClient; +@class ARDSettingsModel; +@class ARDExternalSampleCapturer; +@class RTC_OBJC_TYPE(RTCMediaConstraints); +@class RTC_OBJC_TYPE(RTCCameraVideoCapturer); +@class RTC_OBJC_TYPE(RTCFileVideoCapturer); + +// The delegate is informed of pertinent events and will be called on the +// main queue. +@protocol ARDAppClientDelegate <NSObject> + +- (void)appClient:(ARDAppClient *)client didChangeState:(ARDAppClientState)state; + +- (void)appClient:(ARDAppClient *)client didChangeConnectionState:(RTCIceConnectionState)state; + +- (void)appClient:(ARDAppClient *)client + didCreateLocalCapturer:(RTC_OBJC_TYPE(RTCCameraVideoCapturer) *)localCapturer; + +- (void)appClient:(ARDAppClient *)client + didReceiveLocalVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)localVideoTrack; + +- (void)appClient:(ARDAppClient *)client + didReceiveRemoteVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)remoteVideoTrack; + +- (void)appClient:(ARDAppClient *)client didError:(NSError *)error; + +- (void)appClient:(ARDAppClient *)client didGetStats:(RTC_OBJC_TYPE(RTCStatisticsReport) *)stats; + +@optional +- (void)appClient:(ARDAppClient *)client + didCreateLocalFileCapturer:(RTC_OBJC_TYPE(RTCFileVideoCapturer) *)fileCapturer; + +- (void)appClient:(ARDAppClient *)client + didCreateLocalExternalSampleCapturer:(ARDExternalSampleCapturer *)externalSampleCapturer; + +@end + +// Handles connections to the AppRTC server for a given room. Methods on this +// class should only be called from the main queue. +@interface ARDAppClient : NSObject + +// If `shouldGetStats` is true, stats will be reported in 1s intervals through +// the delegate. +@property(nonatomic, assign) BOOL shouldGetStats; +@property(nonatomic, readonly) ARDAppClientState state; +@property(nonatomic, weak) id<ARDAppClientDelegate> delegate; +@property(nonatomic, assign, getter=isBroadcast) BOOL broadcast; + +// Convenience constructor since all expected use cases will need a delegate +// in order to receive remote tracks. +- (instancetype)initWithDelegate:(id<ARDAppClientDelegate>)delegate; + +// Establishes a connection with the AppRTC servers for the given room id. +// `settings` is an object containing settings such as video codec for the call. +// If `isLoopback` is true, the call will connect to itself. +- (void)connectToRoomWithId:(NSString *)roomId + settings:(ARDSettingsModel *)settings + isLoopback:(BOOL)isLoopback; + +// Disconnects from the AppRTC servers and any connected clients. +- (void)disconnect; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient.m new file mode 100644 index 0000000000..4420972598 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient.m @@ -0,0 +1,899 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDAppClient+Internal.h" + +#import "sdk/objc/api/peerconnection/RTCAudioTrack.h" +#import "sdk/objc/api/peerconnection/RTCConfiguration.h" +#import "sdk/objc/api/peerconnection/RTCFileLogger.h" +#import "sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent.h" +#import "sdk/objc/api/peerconnection/RTCIceServer.h" +#import "sdk/objc/api/peerconnection/RTCMediaConstraints.h" +#import "sdk/objc/api/peerconnection/RTCMediaStream.h" +#import "sdk/objc/api/peerconnection/RTCPeerConnectionFactory.h" +#import "sdk/objc/api/peerconnection/RTCRtpSender.h" +#import "sdk/objc/api/peerconnection/RTCRtpTransceiver.h" +#import "sdk/objc/api/peerconnection/RTCTracing.h" +#import "sdk/objc/api/peerconnection/RTCVideoSource.h" +#import "sdk/objc/api/peerconnection/RTCVideoTrack.h" +#import "sdk/objc/base/RTCLogging.h" +#import "sdk/objc/components/capturer/RTCCameraVideoCapturer.h" +#import "sdk/objc/components/capturer/RTCFileVideoCapturer.h" +#import "sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h" +#import "sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h" + +#import "ARDAppEngineClient.h" +#import "ARDExternalSampleCapturer.h" +#import "ARDJoinResponse.h" +#import "ARDMessageResponse.h" +#import "ARDSettingsModel.h" +#import "ARDSignalingMessage.h" +#import "ARDTURNClient+Internal.h" +#import "ARDUtilities.h" +#import "ARDWebSocketChannel.h" +#import "RTCIceCandidate+JSON.h" +#import "RTCSessionDescription+JSON.h" + +static NSString * const kARDIceServerRequestUrl = @"https://appr.tc/params"; + +static NSString * const kARDAppClientErrorDomain = @"ARDAppClient"; +static NSInteger const kARDAppClientErrorUnknown = -1; +static NSInteger const kARDAppClientErrorRoomFull = -2; +static NSInteger const kARDAppClientErrorCreateSDP = -3; +static NSInteger const kARDAppClientErrorSetSDP = -4; +static NSInteger const kARDAppClientErrorInvalidClient = -5; +static NSInteger const kARDAppClientErrorInvalidRoom = -6; +static NSString * const kARDMediaStreamId = @"ARDAMS"; +static NSString * const kARDAudioTrackId = @"ARDAMSa0"; +static NSString * const kARDVideoTrackId = @"ARDAMSv0"; +static NSString * const kARDVideoTrackKind = @"video"; + +// TODO(tkchin): Add these as UI options. +#if defined(WEBRTC_IOS) +static BOOL const kARDAppClientEnableTracing = NO; +static BOOL const kARDAppClientEnableRtcEventLog = YES; +static int64_t const kARDAppClientAecDumpMaxSizeInBytes = 5e6; // 5 MB. +static int64_t const kARDAppClientRtcEventLogMaxSizeInBytes = 5e6; // 5 MB. +#endif +static int const kKbpsMultiplier = 1000; + +// We need a proxy to NSTimer because it causes a strong retain cycle. When +// using the proxy, `invalidate` must be called before it properly deallocs. +@interface ARDTimerProxy : NSObject + +- (instancetype)initWithInterval:(NSTimeInterval)interval + repeats:(BOOL)repeats + timerHandler:(void (^)(void))timerHandler; +- (void)invalidate; + +@end + +@implementation ARDTimerProxy { + NSTimer *_timer; + void (^_timerHandler)(void); +} + +- (instancetype)initWithInterval:(NSTimeInterval)interval + repeats:(BOOL)repeats + timerHandler:(void (^)(void))timerHandler { + NSParameterAssert(timerHandler); + if (self = [super init]) { + _timerHandler = timerHandler; + _timer = [NSTimer scheduledTimerWithTimeInterval:interval + target:self + selector:@selector(timerDidFire:) + userInfo:nil + repeats:repeats]; + } + return self; +} + +- (void)invalidate { + [_timer invalidate]; +} + +- (void)timerDidFire:(NSTimer *)timer { + _timerHandler(); +} + +@end + +@implementation ARDAppClient { + RTC_OBJC_TYPE(RTCFileLogger) * _fileLogger; + ARDTimerProxy *_statsTimer; + ARDSettingsModel *_settings; + RTC_OBJC_TYPE(RTCVideoTrack) * _localVideoTrack; +} + +@synthesize shouldGetStats = _shouldGetStats; +@synthesize state = _state; +@synthesize delegate = _delegate; +@synthesize roomServerClient = _roomServerClient; +@synthesize channel = _channel; +@synthesize loopbackChannel = _loopbackChannel; +@synthesize turnClient = _turnClient; +@synthesize peerConnection = _peerConnection; +@synthesize factory = _factory; +@synthesize messageQueue = _messageQueue; +@synthesize isTurnComplete = _isTurnComplete; +@synthesize hasReceivedSdp = _hasReceivedSdp; +@synthesize roomId = _roomId; +@synthesize clientId = _clientId; +@synthesize isInitiator = _isInitiator; +@synthesize iceServers = _iceServers; +@synthesize webSocketURL = _websocketURL; +@synthesize webSocketRestURL = _websocketRestURL; +@synthesize defaultPeerConnectionConstraints = + _defaultPeerConnectionConstraints; +@synthesize isLoopback = _isLoopback; +@synthesize broadcast = _broadcast; + +- (instancetype)init { + return [self initWithDelegate:nil]; +} + +- (instancetype)initWithDelegate:(id<ARDAppClientDelegate>)delegate { + if (self = [super init]) { + _roomServerClient = [[ARDAppEngineClient alloc] init]; + _delegate = delegate; + NSURL *turnRequestURL = [NSURL URLWithString:kARDIceServerRequestUrl]; + _turnClient = [[ARDTURNClient alloc] initWithURL:turnRequestURL]; + [self configure]; + } + return self; +} + +// TODO(tkchin): Provide signaling channel factory interface so we can recreate +// channel if we need to on network failure. Also, make this the default public +// constructor. +- (instancetype)initWithRoomServerClient:(id<ARDRoomServerClient>)rsClient + signalingChannel:(id<ARDSignalingChannel>)channel + turnClient:(id<ARDTURNClient>)turnClient + delegate:(id<ARDAppClientDelegate>)delegate { + NSParameterAssert(rsClient); + NSParameterAssert(channel); + NSParameterAssert(turnClient); + if (self = [super init]) { + _roomServerClient = rsClient; + _channel = channel; + _turnClient = turnClient; + _delegate = delegate; + [self configure]; + } + return self; +} + +- (void)configure { + _messageQueue = [NSMutableArray array]; + _iceServers = [NSMutableArray array]; + _fileLogger = [[RTC_OBJC_TYPE(RTCFileLogger) alloc] init]; + [_fileLogger start]; +} + +- (void)dealloc { + self.shouldGetStats = NO; + [self disconnect]; +} + +- (void)setShouldGetStats:(BOOL)shouldGetStats { + if (_shouldGetStats == shouldGetStats) { + return; + } + if (shouldGetStats) { + __weak ARDAppClient *weakSelf = self; + _statsTimer = [[ARDTimerProxy alloc] initWithInterval:1 + repeats:YES + timerHandler:^{ + ARDAppClient *strongSelf = weakSelf; + [strongSelf.peerConnection statisticsWithCompletionHandler:^( + RTC_OBJC_TYPE(RTCStatisticsReport) * stats) { + dispatch_async(dispatch_get_main_queue(), ^{ + ARDAppClient *strongSelf = weakSelf; + [strongSelf.delegate appClient:strongSelf didGetStats:stats]; + }); + }]; + }]; + } else { + [_statsTimer invalidate]; + _statsTimer = nil; + } + _shouldGetStats = shouldGetStats; +} + +- (void)setState:(ARDAppClientState)state { + if (_state == state) { + return; + } + _state = state; + [_delegate appClient:self didChangeState:_state]; +} + +- (void)connectToRoomWithId:(NSString *)roomId + settings:(ARDSettingsModel *)settings + isLoopback:(BOOL)isLoopback { + NSParameterAssert(roomId.length); + NSParameterAssert(_state == kARDAppClientStateDisconnected); + _settings = settings; + _isLoopback = isLoopback; + self.state = kARDAppClientStateConnecting; + + RTC_OBJC_TYPE(RTCDefaultVideoDecoderFactory) *decoderFactory = + [[RTC_OBJC_TYPE(RTCDefaultVideoDecoderFactory) alloc] init]; + RTC_OBJC_TYPE(RTCDefaultVideoEncoderFactory) *encoderFactory = + [[RTC_OBJC_TYPE(RTCDefaultVideoEncoderFactory) alloc] init]; + encoderFactory.preferredCodec = [settings currentVideoCodecSettingFromStore]; + _factory = + [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] initWithEncoderFactory:encoderFactory + decoderFactory:decoderFactory]; + +#if defined(WEBRTC_IOS) + if (kARDAppClientEnableTracing) { + NSString *filePath = [self documentsFilePathForFileName:@"webrtc-trace.txt"]; + RTCStartInternalCapture(filePath); + } +#endif + + // Request TURN. + __weak ARDAppClient *weakSelf = self; + [_turnClient requestServersWithCompletionHandler:^(NSArray *turnServers, + NSError *error) { + if (error) { + RTCLogError(@"Error retrieving TURN servers: %@", error.localizedDescription); + } + ARDAppClient *strongSelf = weakSelf; + [strongSelf.iceServers addObjectsFromArray:turnServers]; + strongSelf.isTurnComplete = YES; + [strongSelf startSignalingIfReady]; + }]; + + // Join room on room server. + [_roomServerClient joinRoomWithRoomId:roomId + isLoopback:isLoopback + completionHandler:^(ARDJoinResponse *response, NSError *error) { + ARDAppClient *strongSelf = weakSelf; + if (error) { + [strongSelf.delegate appClient:strongSelf didError:error]; + return; + } + NSError *joinError = + [[strongSelf class] errorForJoinResultType:response.result]; + if (joinError) { + RTCLogError(@"Failed to join room:%@ on room server.", roomId); + [strongSelf disconnect]; + [strongSelf.delegate appClient:strongSelf didError:joinError]; + return; + } + RTCLog(@"Joined room:%@ on room server.", roomId); + strongSelf.roomId = response.roomId; + strongSelf.clientId = response.clientId; + strongSelf.isInitiator = response.isInitiator; + for (ARDSignalingMessage *message in response.messages) { + if (message.type == kARDSignalingMessageTypeOffer || + message.type == kARDSignalingMessageTypeAnswer) { + strongSelf.hasReceivedSdp = YES; + [strongSelf.messageQueue insertObject:message atIndex:0]; + } else { + [strongSelf.messageQueue addObject:message]; + } + } + strongSelf.webSocketURL = response.webSocketURL; + strongSelf.webSocketRestURL = response.webSocketRestURL; + [strongSelf registerWithColliderIfReady]; + [strongSelf startSignalingIfReady]; + }]; +} + +- (void)disconnect { + if (_state == kARDAppClientStateDisconnected) { + return; + } + if (self.hasJoinedRoomServerRoom) { + [_roomServerClient leaveRoomWithRoomId:_roomId + clientId:_clientId + completionHandler:nil]; + } + if (_channel) { + if (_channel.state == kARDSignalingChannelStateRegistered) { + // Tell the other client we're hanging up. + ARDByeMessage *byeMessage = [[ARDByeMessage alloc] init]; + [_channel sendMessage:byeMessage]; + } + // Disconnect from collider. + _channel = nil; + } + _clientId = nil; + _roomId = nil; + _isInitiator = NO; + _hasReceivedSdp = NO; + _messageQueue = [NSMutableArray array]; + _localVideoTrack = nil; +#if defined(WEBRTC_IOS) + [_factory stopAecDump]; + [_peerConnection stopRtcEventLog]; +#endif + [_peerConnection close]; + _peerConnection = nil; + self.state = kARDAppClientStateDisconnected; +#if defined(WEBRTC_IOS) + if (kARDAppClientEnableTracing) { + RTCStopInternalCapture(); + } +#endif +} + +#pragma mark - ARDSignalingChannelDelegate + +- (void)channel:(id<ARDSignalingChannel>)channel + didReceiveMessage:(ARDSignalingMessage *)message { + switch (message.type) { + case kARDSignalingMessageTypeOffer: + case kARDSignalingMessageTypeAnswer: + // Offers and answers must be processed before any other message, so we + // place them at the front of the queue. + _hasReceivedSdp = YES; + [_messageQueue insertObject:message atIndex:0]; + break; + case kARDSignalingMessageTypeCandidate: + case kARDSignalingMessageTypeCandidateRemoval: + [_messageQueue addObject:message]; + break; + case kARDSignalingMessageTypeBye: + // Disconnects can be processed immediately. + [self processSignalingMessage:message]; + return; + } + [self drainMessageQueueIfReady]; +} + +- (void)channel:(id<ARDSignalingChannel>)channel + didChangeState:(ARDSignalingChannelState)state { + switch (state) { + case kARDSignalingChannelStateOpen: + break; + case kARDSignalingChannelStateRegistered: + break; + case kARDSignalingChannelStateClosed: + case kARDSignalingChannelStateError: + // TODO(tkchin): reconnection scenarios. Right now we just disconnect + // completely if the websocket connection fails. + [self disconnect]; + break; + } +} + +#pragma mark - RTC_OBJC_TYPE(RTCPeerConnectionDelegate) +// Callbacks for this delegate occur on non-main thread and need to be +// dispatched back to main queue as needed. + +- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection + didChangeSignalingState:(RTCSignalingState)stateChanged { + RTCLog(@"Signaling state changed: %ld", (long)stateChanged); +} + +- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection + didAddStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream { + RTCLog(@"Stream with %lu video tracks and %lu audio tracks was added.", + (unsigned long)stream.videoTracks.count, + (unsigned long)stream.audioTracks.count); +} + +- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection + didStartReceivingOnTransceiver:(RTC_OBJC_TYPE(RTCRtpTransceiver) *)transceiver { + RTC_OBJC_TYPE(RTCMediaStreamTrack) *track = transceiver.receiver.track; + RTCLog(@"Now receiving %@ on track %@.", track.kind, track.trackId); +} + +- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection + didRemoveStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream { + RTCLog(@"Stream was removed."); +} + +- (void)peerConnectionShouldNegotiate:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection { + RTCLog(@"WARNING: Renegotiation needed but unimplemented."); +} + +- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection + didChangeIceConnectionState:(RTCIceConnectionState)newState { + RTCLog(@"ICE state changed: %ld", (long)newState); + dispatch_async(dispatch_get_main_queue(), ^{ + [self.delegate appClient:self didChangeConnectionState:newState]; + }); +} + +- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection + didChangeConnectionState:(RTCPeerConnectionState)newState { + RTCLog(@"ICE+DTLS state changed: %ld", (long)newState); +} + +- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection + didChangeIceGatheringState:(RTCIceGatheringState)newState { + RTCLog(@"ICE gathering state changed: %ld", (long)newState); +} + +- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection + didGenerateIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate { + dispatch_async(dispatch_get_main_queue(), ^{ + ARDICECandidateMessage *message = + [[ARDICECandidateMessage alloc] initWithCandidate:candidate]; + [self sendSignalingMessage:message]; + }); +} + +- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection + didFailToGatherIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidateErrorEvent) *)event { + RTCLog(@"Failed to gather ICE candidate. address: %@, port: %d, url: %@, errorCode: %d, " + @"errorText: %@", + event.address, + event.port, + event.url, + event.errorCode, + event.errorText); +} + +- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection + didRemoveIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates { + dispatch_async(dispatch_get_main_queue(), ^{ + ARDICECandidateRemovalMessage *message = + [[ARDICECandidateRemovalMessage alloc] + initWithRemovedCandidates:candidates]; + [self sendSignalingMessage:message]; + }); +} + +- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection + didChangeLocalCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)local + didChangeRemoteCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)remote + lastReceivedMs:(int)lastDataReceivedMs + didHaveReason:(NSString *)reason { + RTCLog(@"ICE candidate pair changed because: %@", reason); +} + +- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection + didOpenDataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel { +} + +#pragma mark - RTCSessionDescriptionDelegate +// Callbacks for this delegate occur on non-main thread and need to be +// dispatched back to main queue as needed. + +- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection + didCreateSessionDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp + error:(NSError *)error { + dispatch_async(dispatch_get_main_queue(), ^{ + if (error) { + RTCLogError(@"Failed to create session description. Error: %@", error); + [self disconnect]; + NSDictionary *userInfo = @{ + NSLocalizedDescriptionKey: @"Failed to create session description.", + }; + NSError *sdpError = + [[NSError alloc] initWithDomain:kARDAppClientErrorDomain + code:kARDAppClientErrorCreateSDP + userInfo:userInfo]; + [self.delegate appClient:self didError:sdpError]; + return; + } + __weak ARDAppClient *weakSelf = self; + [self.peerConnection setLocalDescription:sdp + completionHandler:^(NSError *error) { + ARDAppClient *strongSelf = weakSelf; + [strongSelf peerConnection:strongSelf.peerConnection + didSetSessionDescriptionWithError:error]; + }]; + ARDSessionDescriptionMessage *message = + [[ARDSessionDescriptionMessage alloc] initWithDescription:sdp]; + [self sendSignalingMessage:message]; + [self setMaxBitrateForPeerConnectionVideoSender]; + }); +} + +- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection + didSetSessionDescriptionWithError:(NSError *)error { + dispatch_async(dispatch_get_main_queue(), ^{ + if (error) { + RTCLogError(@"Failed to set session description. Error: %@", error); + [self disconnect]; + NSDictionary *userInfo = @{ + NSLocalizedDescriptionKey: @"Failed to set session description.", + }; + NSError *sdpError = + [[NSError alloc] initWithDomain:kARDAppClientErrorDomain + code:kARDAppClientErrorSetSDP + userInfo:userInfo]; + [self.delegate appClient:self didError:sdpError]; + return; + } + // If we're answering and we've just set the remote offer we need to create + // an answer and set the local description. + if (!self.isInitiator && !self.peerConnection.localDescription) { + RTC_OBJC_TYPE(RTCMediaConstraints) *constraints = [self defaultAnswerConstraints]; + __weak ARDAppClient *weakSelf = self; + [self.peerConnection + answerForConstraints:constraints + completionHandler:^(RTC_OBJC_TYPE(RTCSessionDescription) * sdp, NSError * error) { + ARDAppClient *strongSelf = weakSelf; + [strongSelf peerConnection:strongSelf.peerConnection + didCreateSessionDescription:sdp + error:error]; + }]; + } + }); +} + +#pragma mark - Private + +#if defined(WEBRTC_IOS) + +- (NSString *)documentsFilePathForFileName:(NSString *)fileName { + NSParameterAssert(fileName.length); + NSArray *paths = NSSearchPathForDirectoriesInDomains( + NSDocumentDirectory, NSUserDomainMask, YES); + NSString *documentsDirPath = paths.firstObject; + NSString *filePath = + [documentsDirPath stringByAppendingPathComponent:fileName]; + return filePath; +} + +#endif + +- (BOOL)hasJoinedRoomServerRoom { + return _clientId.length; +} + +// Begins the peer connection connection process if we have both joined a room +// on the room server and tried to obtain a TURN server. Otherwise does nothing. +// A peer connection object will be created with a stream that contains local +// audio and video capture. If this client is the caller, an offer is created as +// well, otherwise the client will wait for an offer to arrive. +- (void)startSignalingIfReady { + if (!_isTurnComplete || !self.hasJoinedRoomServerRoom) { + return; + } + self.state = kARDAppClientStateConnected; + + // Create peer connection. + RTC_OBJC_TYPE(RTCMediaConstraints) *constraints = [self defaultPeerConnectionConstraints]; + RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init]; + RTC_OBJC_TYPE(RTCCertificate) *pcert = [RTC_OBJC_TYPE(RTCCertificate) + generateCertificateWithParams:@{@"expires" : @100000, @"name" : @"RSASSA-PKCS1-v1_5"}]; + config.iceServers = _iceServers; + config.sdpSemantics = RTCSdpSemanticsUnifiedPlan; + config.certificate = pcert; + + _peerConnection = [_factory peerConnectionWithConfiguration:config + constraints:constraints + delegate:self]; + // Create AV senders. + [self createMediaSenders]; + if (_isInitiator) { + // Send offer. + __weak ARDAppClient *weakSelf = self; + [_peerConnection + offerForConstraints:[self defaultOfferConstraints] + completionHandler:^(RTC_OBJC_TYPE(RTCSessionDescription) * sdp, NSError * error) { + ARDAppClient *strongSelf = weakSelf; + [strongSelf peerConnection:strongSelf.peerConnection + didCreateSessionDescription:sdp + error:error]; + }]; + } else { + // Check if we've received an offer. + [self drainMessageQueueIfReady]; + } +#if defined(WEBRTC_IOS) + // Start event log. + if (kARDAppClientEnableRtcEventLog) { + NSString *filePath = [self documentsFilePathForFileName:@"webrtc-rtceventlog"]; + if (![_peerConnection startRtcEventLogWithFilePath:filePath + maxSizeInBytes:kARDAppClientRtcEventLogMaxSizeInBytes]) { + RTCLogError(@"Failed to start event logging."); + } + } + + // Start aecdump diagnostic recording. + if ([_settings currentCreateAecDumpSettingFromStore]) { + NSString *filePath = [self documentsFilePathForFileName:@"webrtc-audio.aecdump"]; + if (![_factory startAecDumpWithFilePath:filePath + maxSizeInBytes:kARDAppClientAecDumpMaxSizeInBytes]) { + RTCLogError(@"Failed to start aec dump."); + } + } +#endif +} + +// Processes the messages that we've received from the room server and the +// signaling channel. The offer or answer message must be processed before other +// signaling messages, however they can arrive out of order. Hence, this method +// only processes pending messages if there is a peer connection object and +// if we have received either an offer or answer. +- (void)drainMessageQueueIfReady { + if (!_peerConnection || !_hasReceivedSdp) { + return; + } + for (ARDSignalingMessage *message in _messageQueue) { + [self processSignalingMessage:message]; + } + [_messageQueue removeAllObjects]; +} + +// Processes the given signaling message based on its type. +- (void)processSignalingMessage:(ARDSignalingMessage *)message { + NSParameterAssert(_peerConnection || + message.type == kARDSignalingMessageTypeBye); + switch (message.type) { + case kARDSignalingMessageTypeOffer: + case kARDSignalingMessageTypeAnswer: { + ARDSessionDescriptionMessage *sdpMessage = + (ARDSessionDescriptionMessage *)message; + RTC_OBJC_TYPE(RTCSessionDescription) *description = sdpMessage.sessionDescription; + __weak ARDAppClient *weakSelf = self; + [_peerConnection setRemoteDescription:description + completionHandler:^(NSError *error) { + ARDAppClient *strongSelf = weakSelf; + [strongSelf peerConnection:strongSelf.peerConnection + didSetSessionDescriptionWithError:error]; + }]; + break; + } + case kARDSignalingMessageTypeCandidate: { + ARDICECandidateMessage *candidateMessage = + (ARDICECandidateMessage *)message; + __weak ARDAppClient *weakSelf = self; + [_peerConnection addIceCandidate:candidateMessage.candidate + completionHandler:^(NSError *error) { + ARDAppClient *strongSelf = weakSelf; + if (error) { + [strongSelf.delegate appClient:strongSelf didError:error]; + } + }]; + break; + } + case kARDSignalingMessageTypeCandidateRemoval: { + ARDICECandidateRemovalMessage *candidateMessage = + (ARDICECandidateRemovalMessage *)message; + [_peerConnection removeIceCandidates:candidateMessage.candidates]; + break; + } + case kARDSignalingMessageTypeBye: + // Other client disconnected. + // TODO(tkchin): support waiting in room for next client. For now just + // disconnect. + [self disconnect]; + break; + } +} + +// Sends a signaling message to the other client. The caller will send messages +// through the room server, whereas the callee will send messages over the +// signaling channel. +- (void)sendSignalingMessage:(ARDSignalingMessage *)message { + if (_isInitiator) { + __weak ARDAppClient *weakSelf = self; + [_roomServerClient sendMessage:message + forRoomId:_roomId + clientId:_clientId + completionHandler:^(ARDMessageResponse *response, + NSError *error) { + ARDAppClient *strongSelf = weakSelf; + if (error) { + [strongSelf.delegate appClient:strongSelf didError:error]; + return; + } + NSError *messageError = + [[strongSelf class] errorForMessageResultType:response.result]; + if (messageError) { + [strongSelf.delegate appClient:strongSelf didError:messageError]; + return; + } + }]; + } else { + [_channel sendMessage:message]; + } +} + +- (void)setMaxBitrateForPeerConnectionVideoSender { + for (RTC_OBJC_TYPE(RTCRtpSender) * sender in _peerConnection.senders) { + if (sender.track != nil) { + if ([sender.track.kind isEqualToString:kARDVideoTrackKind]) { + [self setMaxBitrate:[_settings currentMaxBitrateSettingFromStore] forVideoSender:sender]; + } + } + } +} + +- (void)setMaxBitrate:(NSNumber *)maxBitrate forVideoSender:(RTC_OBJC_TYPE(RTCRtpSender) *)sender { + if (maxBitrate.intValue <= 0) { + return; + } + + RTC_OBJC_TYPE(RTCRtpParameters) *parametersToModify = sender.parameters; + for (RTC_OBJC_TYPE(RTCRtpEncodingParameters) * encoding in parametersToModify.encodings) { + encoding.maxBitrateBps = @(maxBitrate.intValue * kKbpsMultiplier); + } + [sender setParameters:parametersToModify]; +} + +- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)videoTransceiver { + for (RTC_OBJC_TYPE(RTCRtpTransceiver) * transceiver in _peerConnection.transceivers) { + if (transceiver.mediaType == RTCRtpMediaTypeVideo) { + return transceiver; + } + } + return nil; +} + +- (void)createMediaSenders { + RTC_OBJC_TYPE(RTCMediaConstraints) *constraints = [self defaultMediaAudioConstraints]; + RTC_OBJC_TYPE(RTCAudioSource) *source = [_factory audioSourceWithConstraints:constraints]; + RTC_OBJC_TYPE(RTCAudioTrack) *track = [_factory audioTrackWithSource:source + trackId:kARDAudioTrackId]; + [_peerConnection addTrack:track streamIds:@[ kARDMediaStreamId ]]; + _localVideoTrack = [self createLocalVideoTrack]; + if (_localVideoTrack) { + [_peerConnection addTrack:_localVideoTrack streamIds:@[ kARDMediaStreamId ]]; + [_delegate appClient:self didReceiveLocalVideoTrack:_localVideoTrack]; + // We can set up rendering for the remote track right away since the transceiver already has an + // RTC_OBJC_TYPE(RTCRtpReceiver) with a track. The track will automatically get unmuted and + // produce frames once RTP is received. + RTC_OBJC_TYPE(RTCVideoTrack) *track = + (RTC_OBJC_TYPE(RTCVideoTrack) *)([self videoTransceiver].receiver.track); + [_delegate appClient:self didReceiveRemoteVideoTrack:track]; + } +} + +- (RTC_OBJC_TYPE(RTCVideoTrack) *)createLocalVideoTrack { + if ([_settings currentAudioOnlySettingFromStore]) { + return nil; + } + + RTC_OBJC_TYPE(RTCVideoSource) *source = [_factory videoSource]; + +#if !TARGET_IPHONE_SIMULATOR + if (self.isBroadcast) { + ARDExternalSampleCapturer *capturer = + [[ARDExternalSampleCapturer alloc] initWithDelegate:source]; + [_delegate appClient:self didCreateLocalExternalSampleCapturer:capturer]; + } else { + RTC_OBJC_TYPE(RTCCameraVideoCapturer) *capturer = + [[RTC_OBJC_TYPE(RTCCameraVideoCapturer) alloc] initWithDelegate:source]; + [_delegate appClient:self didCreateLocalCapturer:capturer]; + } +#else +#if defined(__IPHONE_11_0) && (__IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_11_0) + if (@available(iOS 10, *)) { + RTC_OBJC_TYPE(RTCFileVideoCapturer) *fileCapturer = + [[RTC_OBJC_TYPE(RTCFileVideoCapturer) alloc] initWithDelegate:source]; + [_delegate appClient:self didCreateLocalFileCapturer:fileCapturer]; + } +#endif +#endif + + return [_factory videoTrackWithSource:source trackId:kARDVideoTrackId]; +} + +#pragma mark - Collider methods + +- (void)registerWithColliderIfReady { + if (!self.hasJoinedRoomServerRoom) { + return; + } + // Open WebSocket connection. + if (!_channel) { + _channel = + [[ARDWebSocketChannel alloc] initWithURL:_websocketURL + restURL:_websocketRestURL + delegate:self]; + if (_isLoopback) { + _loopbackChannel = + [[ARDLoopbackWebSocketChannel alloc] initWithURL:_websocketURL + restURL:_websocketRestURL]; + } + } + [_channel registerForRoomId:_roomId clientId:_clientId]; + if (_isLoopback) { + [_loopbackChannel registerForRoomId:_roomId clientId:@"LOOPBACK_CLIENT_ID"]; + } +} + +#pragma mark - Defaults + +- (RTC_OBJC_TYPE(RTCMediaConstraints) *)defaultMediaAudioConstraints { + NSDictionary *mandatoryConstraints = @{}; + RTC_OBJC_TYPE(RTCMediaConstraints) *constraints = + [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:mandatoryConstraints + optionalConstraints:nil]; + return constraints; +} + +- (RTC_OBJC_TYPE(RTCMediaConstraints) *)defaultAnswerConstraints { + return [self defaultOfferConstraints]; +} + +- (RTC_OBJC_TYPE(RTCMediaConstraints) *)defaultOfferConstraints { + NSDictionary *mandatoryConstraints = @{ + @"OfferToReceiveAudio" : @"true", + @"OfferToReceiveVideo" : @"true" + }; + RTC_OBJC_TYPE(RTCMediaConstraints) *constraints = + [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:mandatoryConstraints + optionalConstraints:nil]; + return constraints; +} + +- (RTC_OBJC_TYPE(RTCMediaConstraints) *)defaultPeerConnectionConstraints { + if (_defaultPeerConnectionConstraints) { + return _defaultPeerConnectionConstraints; + } + NSString *value = _isLoopback ? @"false" : @"true"; + NSDictionary *optionalConstraints = @{ @"DtlsSrtpKeyAgreement" : value }; + RTC_OBJC_TYPE(RTCMediaConstraints) *constraints = + [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:nil + optionalConstraints:optionalConstraints]; + return constraints; +} + +#pragma mark - Errors + ++ (NSError *)errorForJoinResultType:(ARDJoinResultType)resultType { + NSError *error = nil; + switch (resultType) { + case kARDJoinResultTypeSuccess: + break; + case kARDJoinResultTypeUnknown: { + error = [[NSError alloc] initWithDomain:kARDAppClientErrorDomain + code:kARDAppClientErrorUnknown + userInfo:@{ + NSLocalizedDescriptionKey: @"Unknown error.", + }]; + break; + } + case kARDJoinResultTypeFull: { + error = [[NSError alloc] initWithDomain:kARDAppClientErrorDomain + code:kARDAppClientErrorRoomFull + userInfo:@{ + NSLocalizedDescriptionKey: @"Room is full.", + }]; + break; + } + } + return error; +} + ++ (NSError *)errorForMessageResultType:(ARDMessageResultType)resultType { + NSError *error = nil; + switch (resultType) { + case kARDMessageResultTypeSuccess: + break; + case kARDMessageResultTypeUnknown: + error = [[NSError alloc] initWithDomain:kARDAppClientErrorDomain + code:kARDAppClientErrorUnknown + userInfo:@{ + NSLocalizedDescriptionKey: @"Unknown error.", + }]; + break; + case kARDMessageResultTypeInvalidClient: + error = [[NSError alloc] initWithDomain:kARDAppClientErrorDomain + code:kARDAppClientErrorInvalidClient + userInfo:@{ + NSLocalizedDescriptionKey: @"Invalid client.", + }]; + break; + case kARDMessageResultTypeInvalidRoom: + error = [[NSError alloc] initWithDomain:kARDAppClientErrorDomain + code:kARDAppClientErrorInvalidRoom + userInfo:@{ + NSLocalizedDescriptionKey: @"Invalid room.", + }]; + break; + } + return error; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.h new file mode 100644 index 0000000000..7514f3645c --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.h @@ -0,0 +1,14 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDRoomServerClient.h" + +@interface ARDAppEngineClient : NSObject <ARDRoomServerClient> +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.m new file mode 100644 index 0000000000..5139de60d6 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.m @@ -0,0 +1,175 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDAppEngineClient.h" + +#import "sdk/objc/base/RTCLogging.h" + +#import "ARDJoinResponse.h" +#import "ARDMessageResponse.h" +#import "ARDSignalingMessage.h" +#import "ARDUtilities.h" + +// TODO(tkchin): move these to a configuration object. +static NSString * const kARDRoomServerHostUrl = + @"https://appr.tc"; +static NSString * const kARDRoomServerJoinFormat = + @"https://appr.tc/join/%@"; +static NSString * const kARDRoomServerJoinFormatLoopback = + @"https://appr.tc/join/%@?debug=loopback"; +static NSString * const kARDRoomServerMessageFormat = + @"https://appr.tc/message/%@/%@"; +static NSString * const kARDRoomServerLeaveFormat = + @"https://appr.tc/leave/%@/%@"; + +static NSString * const kARDAppEngineClientErrorDomain = @"ARDAppEngineClient"; +static NSInteger const kARDAppEngineClientErrorBadResponse = -1; + +@implementation ARDAppEngineClient + +#pragma mark - ARDRoomServerClient + +- (void)joinRoomWithRoomId:(NSString *)roomId + isLoopback:(BOOL)isLoopback + completionHandler:(void (^)(ARDJoinResponse *response, + NSError *error))completionHandler { + NSParameterAssert(roomId.length); + + NSString *urlString = nil; + if (isLoopback) { + urlString = + [NSString stringWithFormat:kARDRoomServerJoinFormatLoopback, roomId]; + } else { + urlString = + [NSString stringWithFormat:kARDRoomServerJoinFormat, roomId]; + } + + NSURL *roomURL = [NSURL URLWithString:urlString]; + RTCLog(@"Joining room:%@ on room server.", roomId); + NSMutableURLRequest *request = [NSMutableURLRequest requestWithURL:roomURL]; + request.HTTPMethod = @"POST"; + [NSURLConnection sendAsyncRequest:request + completionHandler:^(NSURLResponse *response, NSData *data, NSError *error) { + if (error) { + if (completionHandler) { + completionHandler(nil, error); + } + return; + } + ARDJoinResponse *joinResponse = [ARDJoinResponse responseFromJSONData:data]; + if (!joinResponse) { + if (completionHandler) { + NSError *error = [[self class] badResponseError]; + completionHandler(nil, error); + } + return; + } + if (completionHandler) { + completionHandler(joinResponse, nil); + } + }]; +} + +- (void)sendMessage:(ARDSignalingMessage *)message + forRoomId:(NSString *)roomId + clientId:(NSString *)clientId + completionHandler:(void (^)(ARDMessageResponse *response, + NSError *error))completionHandler { + NSParameterAssert(message); + NSParameterAssert(roomId.length); + NSParameterAssert(clientId.length); + + NSData *data = [message JSONData]; + NSString *urlString = + [NSString stringWithFormat: + kARDRoomServerMessageFormat, roomId, clientId]; + NSURL *url = [NSURL URLWithString:urlString]; + RTCLog(@"C->RS POST: %@", message); + NSMutableURLRequest *request = [NSMutableURLRequest requestWithURL:url]; + request.HTTPMethod = @"POST"; + request.HTTPBody = data; + [NSURLConnection sendAsyncRequest:request + completionHandler:^(NSURLResponse *response, + NSData *data, + NSError *error) { + if (error) { + if (completionHandler) { + completionHandler(nil, error); + } + return; + } + ARDMessageResponse *messageResponse = + [ARDMessageResponse responseFromJSONData:data]; + if (!messageResponse) { + if (completionHandler) { + NSError *error = [[self class] badResponseError]; + completionHandler(nil, error); + } + return; + } + if (completionHandler) { + completionHandler(messageResponse, nil); + } + }]; +} + +- (void)leaveRoomWithRoomId:(NSString *)roomId + clientId:(NSString *)clientId + completionHandler:(void (^)(NSError *error))completionHandler { + NSParameterAssert(roomId.length); + NSParameterAssert(clientId.length); + + NSString *urlString = + [NSString stringWithFormat:kARDRoomServerLeaveFormat, roomId, clientId]; + NSURL *url = [NSURL URLWithString:urlString]; + NSMutableURLRequest *request = [NSMutableURLRequest requestWithURL:url]; + request.HTTPMethod = @"POST"; + + RTCLog(@"C->RS: BYE"); + __block NSError *error = nil; + + // We want a synchronous request so that we know that we've left the room on + // room server before we do any further work. + dispatch_semaphore_t sem = dispatch_semaphore_create(0); + [NSURLConnection sendAsyncRequest:request + completionHandler:^(NSURLResponse *response, NSData *data, NSError *e) { + if (e) { + error = e; + } + dispatch_semaphore_signal(sem); + }]; + + dispatch_semaphore_wait(sem, DISPATCH_TIME_FOREVER); + if (error) { + RTCLogError(@"Error leaving room %@ on room server: %@", roomId, error.localizedDescription); + if (completionHandler) { + completionHandler(error); + } + return; + } + RTCLog(@"Left room:%@ on room server.", roomId); + if (completionHandler) { + completionHandler(nil); + } +} + +#pragma mark - Private + ++ (NSError *)badResponseError { + NSError *error = + [[NSError alloc] initWithDomain:kARDAppEngineClientErrorDomain + code:kARDAppEngineClientErrorBadResponse + userInfo:@{ + NSLocalizedDescriptionKey: @"Error parsing response.", + }]; + return error; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDCaptureController.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDCaptureController.h new file mode 100644 index 0000000000..4febccee96 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDCaptureController.h @@ -0,0 +1,26 @@ +/* + * Copyright 2017 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "sdk/objc/components/capturer/RTCCameraVideoCapturer.h" + +@class ARDSettingsModel; + +// Controls the camera. Handles starting the capture, switching cameras etc. +@interface ARDCaptureController : NSObject + +- (instancetype)initWithCapturer:(RTC_OBJC_TYPE(RTCCameraVideoCapturer) *)capturer + settings:(ARDSettingsModel *)settings; +- (void)startCapture; +- (void)startCapture:(void (^)(NSError *))completion; +- (void)stopCapture; +- (void)switchCamera; +- (void)switchCamera:(void (^)(NSError *))completion; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDCaptureController.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDCaptureController.m new file mode 100644 index 0000000000..26cce9fdaa --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDCaptureController.m @@ -0,0 +1,116 @@ +/* + * Copyright 2017 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDCaptureController.h" + +#import "sdk/objc/base/RTCLogging.h" + +#import "ARDSettingsModel.h" + +const Float64 kFramerateLimit = 30.0; + +@implementation ARDCaptureController { + RTC_OBJC_TYPE(RTCCameraVideoCapturer) * _capturer; + ARDSettingsModel *_settings; + BOOL _usingFrontCamera; +} + +- (instancetype)initWithCapturer:(RTC_OBJC_TYPE(RTCCameraVideoCapturer) *)capturer + settings:(ARDSettingsModel *)settings { + if (self = [super init]) { + _capturer = capturer; + _settings = settings; + _usingFrontCamera = YES; + } + + return self; +} + +- (void)startCapture { + [self startCapture:nil]; +} + +- (void)startCapture:(void (^)(NSError *))completion { + AVCaptureDevicePosition position = + _usingFrontCamera ? AVCaptureDevicePositionFront : AVCaptureDevicePositionBack; + AVCaptureDevice *device = [self findDeviceForPosition:position]; + AVCaptureDeviceFormat *format = [self selectFormatForDevice:device]; + + if (format == nil) { + RTCLogError(@"No valid formats for device %@", device); + NSAssert(NO, @""); + + return; + } + + NSInteger fps = [self selectFpsForFormat:format]; + + [_capturer startCaptureWithDevice:device format:format fps:fps completionHandler:completion]; +} + +- (void)stopCapture { + [_capturer stopCapture]; +} + +- (void)switchCamera { + _usingFrontCamera = !_usingFrontCamera; + [self startCapture:nil]; +} + +- (void)switchCamera:(void (^)(NSError *))completion { + _usingFrontCamera = !_usingFrontCamera; + [self startCapture:completion]; +} + +#pragma mark - Private + +- (AVCaptureDevice *)findDeviceForPosition:(AVCaptureDevicePosition)position { + NSArray<AVCaptureDevice *> *captureDevices = + [RTC_OBJC_TYPE(RTCCameraVideoCapturer) captureDevices]; + for (AVCaptureDevice *device in captureDevices) { + if (device.position == position) { + return device; + } + } + return captureDevices[0]; +} + +- (AVCaptureDeviceFormat *)selectFormatForDevice:(AVCaptureDevice *)device { + NSArray<AVCaptureDeviceFormat *> *formats = + [RTC_OBJC_TYPE(RTCCameraVideoCapturer) supportedFormatsForDevice:device]; + int targetWidth = [_settings currentVideoResolutionWidthFromStore]; + int targetHeight = [_settings currentVideoResolutionHeightFromStore]; + AVCaptureDeviceFormat *selectedFormat = nil; + int currentDiff = INT_MAX; + + for (AVCaptureDeviceFormat *format in formats) { + CMVideoDimensions dimension = CMVideoFormatDescriptionGetDimensions(format.formatDescription); + FourCharCode pixelFormat = CMFormatDescriptionGetMediaSubType(format.formatDescription); + int diff = abs(targetWidth - dimension.width) + abs(targetHeight - dimension.height); + if (diff < currentDiff) { + selectedFormat = format; + currentDiff = diff; + } else if (diff == currentDiff && pixelFormat == [_capturer preferredOutputPixelFormat]) { + selectedFormat = format; + } + } + + return selectedFormat; +} + +- (NSInteger)selectFpsForFormat:(AVCaptureDeviceFormat *)format { + Float64 maxSupportedFramerate = 0; + for (AVFrameRateRange *fpsRange in format.videoSupportedFrameRateRanges) { + maxSupportedFramerate = fmax(maxSupportedFramerate, fpsRange.maxFrameRate); + } + return fmin(maxSupportedFramerate, kFramerateLimit); +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.h new file mode 100644 index 0000000000..7c32c4b509 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.h @@ -0,0 +1,18 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "sdk/objc/base/RTCVideoCapturer.h" + +@protocol ARDExternalSampleDelegate <NSObject> +- (void)didCaptureSampleBuffer:(CMSampleBufferRef)sampleBuffer; +@end + +@interface ARDExternalSampleCapturer : RTC_OBJC_TYPE +(RTCVideoCapturer)<ARDExternalSampleDelegate> @end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.m new file mode 100644 index 0000000000..8bf6716ddb --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.m @@ -0,0 +1,52 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDExternalSampleCapturer.h" + +#import "sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.h" +#import "sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h" +#import "sdk/objc/base/RTCI420Buffer.h" +#import "sdk/objc/base/RTCMutableI420Buffer.h" +#import "sdk/objc/base/RTCMutableYUVPlanarBuffer.h" +#import "sdk/objc/base/RTCVideoFrameBuffer.h" +#import "sdk/objc/base/RTCYUVPlanarBuffer.h" +#import "sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.h" + +@implementation ARDExternalSampleCapturer + +- (instancetype)initWithDelegate:(__weak id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>)delegate { + return [super initWithDelegate:delegate]; +} + +#pragma mark - ARDExternalSampleDelegate + +- (void)didCaptureSampleBuffer:(CMSampleBufferRef)sampleBuffer { + if (CMSampleBufferGetNumSamples(sampleBuffer) != 1 || !CMSampleBufferIsValid(sampleBuffer) || + !CMSampleBufferDataIsReady(sampleBuffer)) { + return; + } + + CVPixelBufferRef pixelBuffer = CMSampleBufferGetImageBuffer(sampleBuffer); + if (pixelBuffer == nil) { + return; + } + + RTC_OBJC_TYPE(RTCCVPixelBuffer) *rtcPixelBuffer = + [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBuffer]; + int64_t timeStampNs = + CMTimeGetSeconds(CMSampleBufferGetPresentationTimeStamp(sampleBuffer)) * NSEC_PER_SEC; + RTC_OBJC_TYPE(RTCVideoFrame) *videoFrame = + [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:rtcPixelBuffer + rotation:RTCVideoRotation_0 + timeStampNs:timeStampNs]; + [self.delegate capturer:self didCaptureVideoFrame:videoFrame]; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse+Internal.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse+Internal.h new file mode 100644 index 0000000000..0edf7083c0 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse+Internal.h @@ -0,0 +1,23 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDJoinResponse.h" + +@interface ARDJoinResponse () + +@property(nonatomic, assign) ARDJoinResultType result; +@property(nonatomic, assign) BOOL isInitiator; +@property(nonatomic, strong) NSString* roomId; +@property(nonatomic, strong) NSString* clientId; +@property(nonatomic, strong) NSArray* messages; +@property(nonatomic, strong) NSURL* webSocketURL; +@property(nonatomic, strong) NSURL* webSocketRestURL; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse.h new file mode 100644 index 0000000000..2911202af1 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse.h @@ -0,0 +1,32 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> + +typedef NS_ENUM(NSInteger, ARDJoinResultType) { + kARDJoinResultTypeUnknown, + kARDJoinResultTypeSuccess, + kARDJoinResultTypeFull +}; + +// Result of joining a room on the room server. +@interface ARDJoinResponse : NSObject + +@property(nonatomic, readonly) ARDJoinResultType result; +@property(nonatomic, readonly) BOOL isInitiator; +@property(nonatomic, readonly) NSString *roomId; +@property(nonatomic, readonly) NSString *clientId; +@property(nonatomic, readonly) NSArray *messages; +@property(nonatomic, readonly) NSURL *webSocketURL; +@property(nonatomic, readonly) NSURL *webSocketRestURL; + ++ (ARDJoinResponse *)responseFromJSONData:(NSData *)data; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse.m new file mode 100644 index 0000000000..87d58e0db1 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse.m @@ -0,0 +1,82 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDJoinResponse+Internal.h" + +#import "ARDSignalingMessage.h" +#import "ARDUtilities.h" +#import "RTCIceServer+JSON.h" + +static NSString const *kARDJoinResultKey = @"result"; +static NSString const *kARDJoinResultParamsKey = @"params"; +static NSString const *kARDJoinInitiatorKey = @"is_initiator"; +static NSString const *kARDJoinRoomIdKey = @"room_id"; +static NSString const *kARDJoinClientIdKey = @"client_id"; +static NSString const *kARDJoinMessagesKey = @"messages"; +static NSString const *kARDJoinWebSocketURLKey = @"wss_url"; +static NSString const *kARDJoinWebSocketRestURLKey = @"wss_post_url"; + +@implementation ARDJoinResponse + +@synthesize result = _result; +@synthesize isInitiator = _isInitiator; +@synthesize roomId = _roomId; +@synthesize clientId = _clientId; +@synthesize messages = _messages; +@synthesize webSocketURL = _webSocketURL; +@synthesize webSocketRestURL = _webSocketRestURL; + ++ (ARDJoinResponse *)responseFromJSONData:(NSData *)data { + NSDictionary *responseJSON = [NSDictionary dictionaryWithJSONData:data]; + if (!responseJSON) { + return nil; + } + ARDJoinResponse *response = [[ARDJoinResponse alloc] init]; + NSString *resultString = responseJSON[kARDJoinResultKey]; + response.result = [[self class] resultTypeFromString:resultString]; + NSDictionary *params = responseJSON[kARDJoinResultParamsKey]; + + response.isInitiator = [params[kARDJoinInitiatorKey] boolValue]; + response.roomId = params[kARDJoinRoomIdKey]; + response.clientId = params[kARDJoinClientIdKey]; + + // Parse messages. + NSArray *messages = params[kARDJoinMessagesKey]; + NSMutableArray *signalingMessages = + [NSMutableArray arrayWithCapacity:messages.count]; + for (NSString *message in messages) { + ARDSignalingMessage *signalingMessage = + [ARDSignalingMessage messageFromJSONString:message]; + [signalingMessages addObject:signalingMessage]; + } + response.messages = signalingMessages; + + // Parse websocket urls. + NSString *webSocketURLString = params[kARDJoinWebSocketURLKey]; + response.webSocketURL = [NSURL URLWithString:webSocketURLString]; + NSString *webSocketRestURLString = params[kARDJoinWebSocketRestURLKey]; + response.webSocketRestURL = [NSURL URLWithString:webSocketRestURLString]; + + return response; +} + +#pragma mark - Private + ++ (ARDJoinResultType)resultTypeFromString:(NSString *)resultString { + ARDJoinResultType result = kARDJoinResultTypeUnknown; + if ([resultString isEqualToString:@"SUCCESS"]) { + result = kARDJoinResultTypeSuccess; + } else if ([resultString isEqualToString:@"FULL"]) { + result = kARDJoinResultTypeFull; + } + return result; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse+Internal.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse+Internal.h new file mode 100644 index 0000000000..66ee76172f --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse+Internal.h @@ -0,0 +1,17 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDMessageResponse.h" + +@interface ARDMessageResponse () + +@property(nonatomic, assign) ARDMessageResultType result; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse.h new file mode 100644 index 0000000000..65468cdf78 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse.h @@ -0,0 +1,26 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> + +typedef NS_ENUM(NSInteger, ARDMessageResultType) { + kARDMessageResultTypeUnknown, + kARDMessageResultTypeSuccess, + kARDMessageResultTypeInvalidRoom, + kARDMessageResultTypeInvalidClient +}; + +@interface ARDMessageResponse : NSObject + +@property(nonatomic, readonly) ARDMessageResultType result; + ++ (ARDMessageResponse *)responseFromJSONData:(NSData *)data; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse.m new file mode 100644 index 0000000000..0f5383f6d6 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse.m @@ -0,0 +1,46 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDMessageResponse+Internal.h" + +#import "ARDUtilities.h" + +static NSString const *kARDMessageResultKey = @"result"; + +@implementation ARDMessageResponse + +@synthesize result = _result; + ++ (ARDMessageResponse *)responseFromJSONData:(NSData *)data { + NSDictionary *responseJSON = [NSDictionary dictionaryWithJSONData:data]; + if (!responseJSON) { + return nil; + } + ARDMessageResponse *response = [[ARDMessageResponse alloc] init]; + response.result = + [[self class] resultTypeFromString:responseJSON[kARDMessageResultKey]]; + return response; +} + +#pragma mark - Private + ++ (ARDMessageResultType)resultTypeFromString:(NSString *)resultString { + ARDMessageResultType result = kARDMessageResultTypeUnknown; + if ([resultString isEqualToString:@"SUCCESS"]) { + result = kARDMessageResultTypeSuccess; + } else if ([resultString isEqualToString:@"INVALID_CLIENT"]) { + result = kARDMessageResultTypeInvalidClient; + } else if ([resultString isEqualToString:@"INVALID_ROOM"]) { + result = kARDMessageResultTypeInvalidRoom; + } + return result; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDRoomServerClient.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDRoomServerClient.h new file mode 100644 index 0000000000..3a5818d6d6 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDRoomServerClient.h @@ -0,0 +1,32 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> + +@class ARDJoinResponse; +@class ARDMessageResponse; +@class ARDSignalingMessage; + +@protocol ARDRoomServerClient <NSObject> + +- (void)joinRoomWithRoomId:(NSString *)roomId + isLoopback:(BOOL)isLoopback + completionHandler:(void (^)(ARDJoinResponse *response, NSError *error))completionHandler; + +- (void)sendMessage:(ARDSignalingMessage *)message + forRoomId:(NSString *)roomId + clientId:(NSString *)clientId + completionHandler:(void (^)(ARDMessageResponse *response, NSError *error))completionHandler; + +- (void)leaveRoomWithRoomId:(NSString *)roomId + clientId:(NSString *)clientId + completionHandler:(void (^)(NSError *error))completionHandler; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel+Private.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel+Private.h new file mode 100644 index 0000000000..dc3f24ced8 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel+Private.h @@ -0,0 +1,21 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> + +#import "ARDSettingsModel.h" + +@class ARDSettingsStore; + +NS_ASSUME_NONNULL_BEGIN +@interface ARDSettingsModel () +- (ARDSettingsStore *)settingsStore; +@end +NS_ASSUME_NONNULL_END diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel.h new file mode 100644 index 0000000000..47c7defacd --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel.h @@ -0,0 +1,123 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> + +#import "sdk/objc/base/RTCVideoCodecInfo.h" + +NS_ASSUME_NONNULL_BEGIN + +/** + * Model class for user defined settings. + * + * Handles storing the settings and provides default values if setting is not + * set. Also provides list of available options for different settings. Stores + * for example video codec, video resolution and maximum bitrate. + */ +@interface ARDSettingsModel : NSObject + +/** + * Returns array of available capture resoultions. + * + * The capture resolutions are represented as strings in the following format + * [width]x[height] + */ +- (NSArray<NSString *> *)availableVideoResolutions; + +/** + * Returns current video resolution string. + * If no resolution is in store, default value of 640x480 is returned. + * When defaulting to value, the default is saved in store for consistency reasons. + */ +- (NSString *)currentVideoResolutionSettingFromStore; +- (int)currentVideoResolutionWidthFromStore; +- (int)currentVideoResolutionHeightFromStore; + +/** + * Stores the provided video resolution string into the store. + * + * If the provided resolution is no part of the available video resolutions + * the store operation will not be executed and NO will be returned. + * @param resolution the string to be stored. + * @return YES/NO depending on success. + */ +- (BOOL)storeVideoResolutionSetting:(NSString *)resolution; + +/** + * Returns array of available video codecs. + */ +- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)availableVideoCodecs; + +/** + * Returns current video codec setting from store if present or default (H264) otherwise. + */ +- (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)currentVideoCodecSettingFromStore; + +/** + * Stores the provided video codec setting into the store. + * + * If the provided video codec is not part of the available video codecs + * the store operation will not be executed and NO will be returned. + * @param video codec settings the string to be stored. + * @return YES/NO depending on success. + */ +- (BOOL)storeVideoCodecSetting:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)videoCodec; + +/** + * Returns current max bitrate setting from store if present. + */ +- (nullable NSNumber *)currentMaxBitrateSettingFromStore; + +/** + * Stores the provided bitrate value into the store. + * + * @param bitrate NSNumber representation of the max bitrate value. + */ +- (void)storeMaxBitrateSetting:(nullable NSNumber *)bitrate; + +/** + * Returns current audio only setting from store if present or default (NO) otherwise. + */ +- (BOOL)currentAudioOnlySettingFromStore; + +/** + * Stores the provided audio only setting into the store. + * + * @param setting the boolean value to be stored. + */ +- (void)storeAudioOnlySetting:(BOOL)audioOnly; + +/** + * Returns current create AecDump setting from store if present or default (NO) otherwise. + */ +- (BOOL)currentCreateAecDumpSettingFromStore; + +/** + * Stores the provided create AecDump setting into the store. + * + * @param setting the boolean value to be stored. + */ +- (void)storeCreateAecDumpSetting:(BOOL)createAecDump; + +/** + * Returns current setting whether to use manual audio config from store if present or default (YES) + * otherwise. + */ +- (BOOL)currentUseManualAudioConfigSettingFromStore; + +/** + * Stores the provided use manual audio config setting into the store. + * + * @param setting the boolean value to be stored. + */ +- (void)storeUseManualAudioConfigSetting:(BOOL)useManualAudioConfig; + +@end +NS_ASSUME_NONNULL_END diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel.m new file mode 100644 index 0000000000..9e709b0553 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel.m @@ -0,0 +1,211 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDSettingsModel+Private.h" +#import "ARDSettingsStore.h" + +#import "sdk/objc/api/peerconnection/RTCMediaConstraints.h" +#import "sdk/objc/components/capturer/RTCCameraVideoCapturer.h" +#import "sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h" + +NS_ASSUME_NONNULL_BEGIN + +@interface ARDSettingsModel () { + ARDSettingsStore *_settingsStore; +} +@end + +@implementation ARDSettingsModel + +- (NSArray<NSString *> *)availableVideoResolutions { + NSMutableSet<NSArray<NSNumber *> *> *resolutions = + [[NSMutableSet<NSArray<NSNumber *> *> alloc] init]; + for (AVCaptureDevice *device in [RTC_OBJC_TYPE(RTCCameraVideoCapturer) captureDevices]) { + for (AVCaptureDeviceFormat *format in + [RTC_OBJC_TYPE(RTCCameraVideoCapturer) supportedFormatsForDevice:device]) { + CMVideoDimensions resolution = + CMVideoFormatDescriptionGetDimensions(format.formatDescription); + NSArray<NSNumber *> *resolutionObject = @[ @(resolution.width), @(resolution.height) ]; + [resolutions addObject:resolutionObject]; + } + } + + NSArray<NSArray<NSNumber *> *> *sortedResolutions = + [[resolutions allObjects] sortedArrayUsingComparator:^NSComparisonResult( + NSArray<NSNumber *> *obj1, NSArray<NSNumber *> *obj2) { + NSComparisonResult cmp = [obj1.firstObject compare:obj2.firstObject]; + if (cmp != NSOrderedSame) { + return cmp; + } + return [obj1.lastObject compare:obj2.lastObject]; + }]; + + NSMutableArray<NSString *> *resolutionStrings = [[NSMutableArray<NSString *> alloc] init]; + for (NSArray<NSNumber *> *resolution in sortedResolutions) { + NSString *resolutionString = + [NSString stringWithFormat:@"%@x%@", resolution.firstObject, resolution.lastObject]; + [resolutionStrings addObject:resolutionString]; + } + + return [resolutionStrings copy]; +} + +- (NSString *)currentVideoResolutionSettingFromStore { + [self registerStoreDefaults]; + return [[self settingsStore] videoResolution]; +} + +- (BOOL)storeVideoResolutionSetting:(NSString *)resolution { + if (![[self availableVideoResolutions] containsObject:resolution]) { + return NO; + } + [[self settingsStore] setVideoResolution:resolution]; + return YES; +} + +- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)availableVideoCodecs { + return [RTC_OBJC_TYPE(RTCDefaultVideoEncoderFactory) supportedCodecs]; +} + +- (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)currentVideoCodecSettingFromStore { + [self registerStoreDefaults]; + NSData *codecData = [[self settingsStore] videoCodec]; +#if defined(WEBRTC_IOS) || __MAC_OS_X_VERSION_MIN_REQUIRED >= __MAC_10_13 + Class expectedClass = [RTC_OBJC_TYPE(RTCVideoCodecInfo) class]; + NSError *error; + RTC_OBJC_TYPE(RTCVideoCodecInfo) *videoCodecSetting = + [NSKeyedUnarchiver unarchivedObjectOfClass:expectedClass fromData:codecData error:&error]; + if (!error) { + return videoCodecSetting; + } + return nil; +#else + return [NSKeyedUnarchiver unarchiveObjectWithData:codecData]; +#endif +} + +- (BOOL)storeVideoCodecSetting:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)videoCodec { + if (![[self availableVideoCodecs] containsObject:videoCodec]) { + return NO; + } + +#if defined(WEBRTC_IOS) || __MAC_OS_X_VERSION_MIN_REQUIRED >= __MAC_10_13 + NSError *error; + NSData *codecData = [NSKeyedArchiver archivedDataWithRootObject:videoCodec + requiringSecureCoding:NO + error:&error]; + if (error) { + return NO; + } +#else + NSData *codecData = [NSKeyedArchiver archivedDataWithRootObject:videoCodec]; +#endif + + [[self settingsStore] setVideoCodec:codecData]; + return YES; +} + +- (nullable NSNumber *)currentMaxBitrateSettingFromStore { + [self registerStoreDefaults]; + return [[self settingsStore] maxBitrate]; +} + +- (void)storeMaxBitrateSetting:(nullable NSNumber *)bitrate { + [[self settingsStore] setMaxBitrate:bitrate]; +} + +- (BOOL)currentAudioOnlySettingFromStore { + return [[self settingsStore] audioOnly]; +} + +- (void)storeAudioOnlySetting:(BOOL)audioOnly { + [[self settingsStore] setAudioOnly:audioOnly]; +} + +- (BOOL)currentCreateAecDumpSettingFromStore { + return [[self settingsStore] createAecDump]; +} + +- (void)storeCreateAecDumpSetting:(BOOL)createAecDump { + [[self settingsStore] setCreateAecDump:createAecDump]; +} + +- (BOOL)currentUseManualAudioConfigSettingFromStore { + return [[self settingsStore] useManualAudioConfig]; +} + +- (void)storeUseManualAudioConfigSetting:(BOOL)useManualAudioConfig { + [[self settingsStore] setUseManualAudioConfig:useManualAudioConfig]; +} + +#pragma mark - Testable + +- (ARDSettingsStore *)settingsStore { + if (!_settingsStore) { + _settingsStore = [[ARDSettingsStore alloc] init]; + [self registerStoreDefaults]; + } + return _settingsStore; +} + +- (int)currentVideoResolutionWidthFromStore { + NSString *resolution = [self currentVideoResolutionSettingFromStore]; + + return [self videoResolutionComponentAtIndex:0 inString:resolution]; +} + +- (int)currentVideoResolutionHeightFromStore { + NSString *resolution = [self currentVideoResolutionSettingFromStore]; + return [self videoResolutionComponentAtIndex:1 inString:resolution]; +} + +#pragma mark - + +- (NSString *)defaultVideoResolutionSetting { + return [self availableVideoResolutions].firstObject; +} + +- (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)defaultVideoCodecSetting { + return [self availableVideoCodecs].firstObject; +} + +- (int)videoResolutionComponentAtIndex:(int)index inString:(NSString *)resolution { + if (index != 0 && index != 1) { + return 0; + } + NSArray<NSString *> *components = [resolution componentsSeparatedByString:@"x"]; + if (components.count != 2) { + return 0; + } + return components[index].intValue; +} + +- (void)registerStoreDefaults { +#if defined(WEBRTC_IOS) || __MAC_OS_X_VERSION_MIN_REQUIRED >= __MAC_10_13 + NSError *error; + NSData *codecData = [NSKeyedArchiver archivedDataWithRootObject:[self defaultVideoCodecSetting] + requiringSecureCoding:NO + error:&error]; + if (error) { + return; + } +#else + NSData *codecData = [NSKeyedArchiver archivedDataWithRootObject:[self defaultVideoCodecSetting]]; +#endif + + [ARDSettingsStore setDefaultsForVideoResolution:[self defaultVideoResolutionSetting] + videoCodec:codecData + bitrate:nil + audioOnly:NO + createAecDump:NO + useManualAudioConfig:YES]; +} +@end +NS_ASSUME_NONNULL_END diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsStore.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsStore.h new file mode 100644 index 0000000000..bb051dbb26 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsStore.h @@ -0,0 +1,52 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> + +NS_ASSUME_NONNULL_BEGIN + +/** + * Light-weight persistent store for user settings. + * + * It will persist between application launches and application updates. + */ +@interface ARDSettingsStore : NSObject + +/** + * Set fallback values in case the setting has not been written by the user. + * @param dictionary of values to store + */ ++ (void)setDefaultsForVideoResolution:(NSString *)videoResolution + videoCodec:(NSData *)videoCodec + bitrate:(nullable NSNumber *)bitrate + audioOnly:(BOOL)audioOnly + createAecDump:(BOOL)createAecDump + useManualAudioConfig:(BOOL)useManualAudioConfig; + +@property(nonatomic) NSString *videoResolution; +@property(nonatomic) NSData *videoCodec; + +/** + * Returns current max bitrate number stored in the store. + */ +- (nullable NSNumber *)maxBitrate; + +/** + * Stores the provided value as maximum bitrate setting. + * @param value the number to be stored + */ +- (void)setMaxBitrate:(nullable NSNumber *)value; + +@property(nonatomic) BOOL audioOnly; +@property(nonatomic) BOOL createAecDump; +@property(nonatomic) BOOL useManualAudioConfig; + +@end +NS_ASSUME_NONNULL_END diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsStore.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsStore.m new file mode 100644 index 0000000000..a3713e2f0e --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsStore.m @@ -0,0 +1,115 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDSettingsStore.h" + +static NSString *const kVideoResolutionKey = @"rtc_video_resolution_key"; +static NSString *const kVideoCodecKey = @"rtc_video_codec_info_key"; +static NSString *const kBitrateKey = @"rtc_max_bitrate_key"; +static NSString *const kAudioOnlyKey = @"rtc_audio_only_key"; +static NSString *const kCreateAecDumpKey = @"rtc_create_aec_dump_key"; +static NSString *const kUseManualAudioConfigKey = @"rtc_use_manual_audio_config_key"; + +NS_ASSUME_NONNULL_BEGIN +@interface ARDSettingsStore () { + NSUserDefaults *_storage; +} +@property(nonatomic, strong, readonly) NSUserDefaults *storage; +@end + +@implementation ARDSettingsStore + ++ (void)setDefaultsForVideoResolution:(NSString *)videoResolution + videoCodec:(NSData *)videoCodec + bitrate:(nullable NSNumber *)bitrate + audioOnly:(BOOL)audioOnly + createAecDump:(BOOL)createAecDump + useManualAudioConfig:(BOOL)useManualAudioConfig { + NSMutableDictionary<NSString *, id> *defaultsDictionary = [@{ + kAudioOnlyKey : @(audioOnly), + kCreateAecDumpKey : @(createAecDump), + kUseManualAudioConfigKey : @(useManualAudioConfig) + } mutableCopy]; + + if (videoResolution) { + defaultsDictionary[kVideoResolutionKey] = videoResolution; + } + if (videoCodec) { + defaultsDictionary[kVideoCodecKey] = videoCodec; + } + if (bitrate) { + defaultsDictionary[kBitrateKey] = bitrate; + } + [[NSUserDefaults standardUserDefaults] registerDefaults:defaultsDictionary]; +} + +- (NSUserDefaults *)storage { + if (!_storage) { + _storage = [NSUserDefaults standardUserDefaults]; + } + return _storage; +} + +- (NSString *)videoResolution { + return [self.storage objectForKey:kVideoResolutionKey]; +} + +- (void)setVideoResolution:(NSString *)resolution { + [self.storage setObject:resolution forKey:kVideoResolutionKey]; + [self.storage synchronize]; +} + +- (NSData *)videoCodec { + return [self.storage objectForKey:kVideoCodecKey]; +} + +- (void)setVideoCodec:(NSData *)videoCodec { + [self.storage setObject:videoCodec forKey:kVideoCodecKey]; + [self.storage synchronize]; +} + +- (nullable NSNumber *)maxBitrate { + return [self.storage objectForKey:kBitrateKey]; +} + +- (void)setMaxBitrate:(nullable NSNumber *)value { + [self.storage setObject:value forKey:kBitrateKey]; + [self.storage synchronize]; +} + +- (BOOL)audioOnly { + return [self.storage boolForKey:kAudioOnlyKey]; +} + +- (void)setAudioOnly:(BOOL)audioOnly { + [self.storage setBool:audioOnly forKey:kAudioOnlyKey]; + [self.storage synchronize]; +} + +- (BOOL)createAecDump { + return [self.storage boolForKey:kCreateAecDumpKey]; +} + +- (void)setCreateAecDump:(BOOL)createAecDump { + [self.storage setBool:createAecDump forKey:kCreateAecDumpKey]; + [self.storage synchronize]; +} + +- (BOOL)useManualAudioConfig { + return [self.storage boolForKey:kUseManualAudioConfigKey]; +} + +- (void)setUseManualAudioConfig:(BOOL)useManualAudioConfig { + [self.storage setBool:useManualAudioConfig forKey:kUseManualAudioConfigKey]; + [self.storage synchronize]; +} + +@end +NS_ASSUME_NONNULL_END diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingChannel.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingChannel.h new file mode 100644 index 0000000000..396b117b17 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingChannel.h @@ -0,0 +1,48 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> + +#import "ARDSignalingMessage.h" + +typedef NS_ENUM(NSInteger, ARDSignalingChannelState) { + // State when disconnected. + kARDSignalingChannelStateClosed, + // State when connection is established but not ready for use. + kARDSignalingChannelStateOpen, + // State when connection is established and registered. + kARDSignalingChannelStateRegistered, + // State when connection encounters a fatal error. + kARDSignalingChannelStateError +}; + +@protocol ARDSignalingChannel; +@protocol ARDSignalingChannelDelegate <NSObject> + +- (void)channel:(id<ARDSignalingChannel>)channel didChangeState:(ARDSignalingChannelState)state; + +- (void)channel:(id<ARDSignalingChannel>)channel didReceiveMessage:(ARDSignalingMessage *)message; + +@end + +@protocol ARDSignalingChannel <NSObject> + +@property(nonatomic, readonly) NSString *roomId; +@property(nonatomic, readonly) NSString *clientId; +@property(nonatomic, readonly) ARDSignalingChannelState state; +@property(nonatomic, weak) id<ARDSignalingChannelDelegate> delegate; + +// Registers the channel for the given room and client id. +- (void)registerForRoomId:(NSString *)roomId clientId:(NSString *)clientId; + +// Sends signaling message over the channel. +- (void)sendMessage:(ARDSignalingMessage *)message; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.h new file mode 100644 index 0000000000..ac19e8fba7 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.h @@ -0,0 +1,58 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> + +#import "sdk/objc/api/peerconnection/RTCIceCandidate.h" +#import "sdk/objc/api/peerconnection/RTCSessionDescription.h" + +typedef enum { + kARDSignalingMessageTypeCandidate, + kARDSignalingMessageTypeCandidateRemoval, + kARDSignalingMessageTypeOffer, + kARDSignalingMessageTypeAnswer, + kARDSignalingMessageTypeBye, +} ARDSignalingMessageType; + +@interface ARDSignalingMessage : NSObject + +@property(nonatomic, readonly) ARDSignalingMessageType type; + ++ (ARDSignalingMessage *)messageFromJSONString:(NSString *)jsonString; +- (NSData *)JSONData; + +@end + +@interface ARDICECandidateMessage : ARDSignalingMessage + +@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCIceCandidate) * candidate; + +- (instancetype)initWithCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate; + +@end + +@interface ARDICECandidateRemovalMessage : ARDSignalingMessage + +@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *candidates; + +- (instancetype)initWithRemovedCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates; + +@end + +@interface ARDSessionDescriptionMessage : ARDSignalingMessage + +@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCSessionDescription) * sessionDescription; + +- (instancetype)initWithDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)description; + +@end + +@interface ARDByeMessage : ARDSignalingMessage +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.m new file mode 100644 index 0000000000..049c0f5b0a --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.m @@ -0,0 +1,160 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDSignalingMessage.h" + +#import "sdk/objc/base/RTCLogging.h" + +#import "ARDUtilities.h" +#import "RTCIceCandidate+JSON.h" +#import "RTCSessionDescription+JSON.h" + +static NSString * const kARDSignalingMessageTypeKey = @"type"; +static NSString * const kARDTypeValueRemoveCandidates = @"remove-candidates"; + +@implementation ARDSignalingMessage + +@synthesize type = _type; + +- (instancetype)initWithType:(ARDSignalingMessageType)type { + if (self = [super init]) { + _type = type; + } + return self; +} + +- (NSString *)description { + return [[NSString alloc] initWithData:[self JSONData] + encoding:NSUTF8StringEncoding]; +} + ++ (ARDSignalingMessage *)messageFromJSONString:(NSString *)jsonString { + NSDictionary *values = [NSDictionary dictionaryWithJSONString:jsonString]; + if (!values) { + RTCLogError(@"Error parsing signaling message JSON."); + return nil; + } + + NSString *typeString = values[kARDSignalingMessageTypeKey]; + ARDSignalingMessage *message = nil; + if ([typeString isEqualToString:@"candidate"]) { + RTC_OBJC_TYPE(RTCIceCandidate) *candidate = + [RTC_OBJC_TYPE(RTCIceCandidate) candidateFromJSONDictionary:values]; + message = [[ARDICECandidateMessage alloc] initWithCandidate:candidate]; + } else if ([typeString isEqualToString:kARDTypeValueRemoveCandidates]) { + RTCLogInfo(@"Received remove-candidates message"); + NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *candidates = + [RTC_OBJC_TYPE(RTCIceCandidate) candidatesFromJSONDictionary:values]; + message = [[ARDICECandidateRemovalMessage alloc] + initWithRemovedCandidates:candidates]; + } else if ([typeString isEqualToString:@"offer"] || + [typeString isEqualToString:@"answer"]) { + RTC_OBJC_TYPE(RTCSessionDescription) *description = + [RTC_OBJC_TYPE(RTCSessionDescription) descriptionFromJSONDictionary:values]; + message = + [[ARDSessionDescriptionMessage alloc] initWithDescription:description]; + } else if ([typeString isEqualToString:@"bye"]) { + message = [[ARDByeMessage alloc] init]; + } else { + RTCLogError(@"Unexpected type: %@", typeString); + } + return message; +} + +- (NSData *)JSONData { + return nil; +} + +@end + +@implementation ARDICECandidateMessage + +@synthesize candidate = _candidate; + +- (instancetype)initWithCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate { + if (self = [super initWithType:kARDSignalingMessageTypeCandidate]) { + _candidate = candidate; + } + return self; +} + +- (NSData *)JSONData { + return [_candidate JSONData]; +} + +@end + +@implementation ARDICECandidateRemovalMessage + +@synthesize candidates = _candidates; + +- (instancetype)initWithRemovedCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates { + NSParameterAssert(candidates.count); + if (self = [super initWithType:kARDSignalingMessageTypeCandidateRemoval]) { + _candidates = candidates; + } + return self; +} + +- (NSData *)JSONData { + return [RTC_OBJC_TYPE(RTCIceCandidate) JSONDataForIceCandidates:_candidates + withType:kARDTypeValueRemoveCandidates]; +} + +@end + +@implementation ARDSessionDescriptionMessage + +@synthesize sessionDescription = _sessionDescription; + +- (instancetype)initWithDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)description { + ARDSignalingMessageType messageType = kARDSignalingMessageTypeOffer; + RTCSdpType sdpType = description.type; + switch (sdpType) { + case RTCSdpTypeOffer: + messageType = kARDSignalingMessageTypeOffer; + break; + case RTCSdpTypeAnswer: + messageType = kARDSignalingMessageTypeAnswer; + break; + case RTCSdpTypePrAnswer: + case RTCSdpTypeRollback: + NSAssert( + NO, @"Unexpected type: %@", [RTC_OBJC_TYPE(RTCSessionDescription) stringForType:sdpType]); + break; + } + if (self = [super initWithType:messageType]) { + _sessionDescription = description; + } + return self; +} + +- (NSData *)JSONData { + return [_sessionDescription JSONData]; +} + +@end + +@implementation ARDByeMessage + +- (instancetype)init { + return [super initWithType:kARDSignalingMessageTypeBye]; +} + +- (NSData *)JSONData { + NSDictionary *message = @{ + @"type": @"bye" + }; + return [NSJSONSerialization dataWithJSONObject:message + options:NSJSONWritingPrettyPrinted + error:NULL]; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.h new file mode 100644 index 0000000000..eaffa67049 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.h @@ -0,0 +1,26 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> + +#import "sdk/objc/api/peerconnection/RTCStatisticsReport.h" +#import "sdk/objc/base/RTCMacros.h" + +/** Class used to accumulate stats information into a single displayable string. + */ +@interface ARDStatsBuilder : NSObject + +/** String that represents the accumulated stats reports passed into this + * class. + */ +@property(nonatomic, readonly) NSString *statsString; +@property(nonatomic) RTC_OBJC_TYPE(RTCStatisticsReport) * stats; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.m new file mode 100644 index 0000000000..7ebf9fb1c7 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.m @@ -0,0 +1,36 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDStatsBuilder.h" + +#import "sdk/objc/api/peerconnection/RTCLegacyStatsReport.h" +#import "sdk/objc/base/RTCMacros.h" + +#import "ARDUtilities.h" + +@implementation ARDStatsBuilder + +@synthesize stats = _stats; + +- (NSString *)statsString { + NSMutableString *result = [NSMutableString string]; + + [result appendFormat:@"(cpu)%ld%%\n", (long)ARDGetCpuUsagePercentage()]; + + for (NSString *key in _stats.statistics) { + RTC_OBJC_TYPE(RTCStatistics) *stat = _stats.statistics[key]; + [result appendFormat:@"%@\n", stat.description]; + } + + return result; +} + +@end + diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient+Internal.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient+Internal.h new file mode 100644 index 0000000000..3a579f8f7f --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient+Internal.h @@ -0,0 +1,17 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDTURNClient.h" + +@interface ARDTURNClient : NSObject <ARDTURNClient> + +- (instancetype)initWithURL:(NSURL *)url; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient.h new file mode 100644 index 0000000000..0399736f03 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient.h @@ -0,0 +1,23 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> + +#import "sdk/objc/base/RTCMacros.h" + +@class RTC_OBJC_TYPE(RTCIceServer); + +@protocol ARDTURNClient <NSObject> + +// Returns TURN server urls if successful. +- (void)requestServersWithCompletionHandler:(void (^)(NSArray *turnServers, + NSError *error))completionHandler; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient.m new file mode 100644 index 0000000000..069231cd7e --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient.m @@ -0,0 +1,86 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDTURNClient+Internal.h" + +#import "ARDUtilities.h" +#import "RTCIceServer+JSON.h" + +// TODO(tkchin): move this to a configuration object. +static NSString *kTURNRefererURLString = @"https://appr.tc"; +static NSString *kARDTURNClientErrorDomain = @"ARDTURNClient"; +static NSInteger kARDTURNClientErrorBadResponse = -1; + +@implementation ARDTURNClient { + NSURL *_url; +} + +- (instancetype)initWithURL:(NSURL *)url { + NSParameterAssert([url absoluteString].length); + if (self = [super init]) { + _url = url; + } + return self; +} + +- (void)requestServersWithCompletionHandler: + (void (^)(NSArray *turnServers, NSError *error))completionHandler { + + NSMutableURLRequest *request = [NSMutableURLRequest requestWithURL:_url]; + [NSURLConnection sendAsyncRequest:request + completionHandler:^(NSURLResponse *response, NSData *data, NSError *error) { + if (error) { + completionHandler(nil, error); + return; + } + NSDictionary *responseDict = [NSDictionary dictionaryWithJSONData:data]; + NSString *iceServerUrl = responseDict[@"ice_server_url"]; + [self makeTurnServerRequestToURL:[NSURL URLWithString:iceServerUrl] + WithCompletionHandler:completionHandler]; + }]; +} + +#pragma mark - Private + +- (void)makeTurnServerRequestToURL:(NSURL *)url + WithCompletionHandler:(void (^)(NSArray *turnServers, + NSError *error))completionHandler { + NSMutableURLRequest *iceServerRequest = [NSMutableURLRequest requestWithURL:url]; + iceServerRequest.HTTPMethod = @"POST"; + [iceServerRequest addValue:kTURNRefererURLString forHTTPHeaderField:@"referer"]; + [NSURLConnection sendAsyncRequest:iceServerRequest + completionHandler:^(NSURLResponse *response, + NSData *data, + NSError *error) { + if (error) { + completionHandler(nil, error); + return; + } + NSDictionary *turnResponseDict = [NSDictionary dictionaryWithJSONData:data]; + NSMutableArray *turnServers = [NSMutableArray array]; + [turnResponseDict[@"iceServers"] + enumerateObjectsUsingBlock:^(NSDictionary *obj, NSUInteger idx, BOOL *stop) { + [turnServers addObject:[RTC_OBJC_TYPE(RTCIceServer) serverFromJSONDictionary:obj]]; + }]; + if (!turnServers) { + NSError *responseError = + [[NSError alloc] initWithDomain:kARDTURNClientErrorDomain + code:kARDTURNClientErrorBadResponse + userInfo:@{ + NSLocalizedDescriptionKey: @"Bad TURN response.", + }]; + completionHandler(nil, responseError); + return; + } + completionHandler(turnServers, nil); + }]; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.h new file mode 100644 index 0000000000..81888e6e83 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.h @@ -0,0 +1,40 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> + +#import "ARDSignalingChannel.h" + +// Wraps a WebSocket connection to the AppRTC WebSocket server. +@interface ARDWebSocketChannel : NSObject <ARDSignalingChannel> + +- (instancetype)initWithURL:(NSURL *)url + restURL:(NSURL *)restURL + delegate:(id<ARDSignalingChannelDelegate>)delegate; + +// Registers with the WebSocket server for the given room and client id once +// the web socket connection is open. +- (void)registerForRoomId:(NSString *)roomId clientId:(NSString *)clientId; + +// Sends message over the WebSocket connection if registered, otherwise POSTs to +// the web socket server instead. +- (void)sendMessage:(ARDSignalingMessage *)message; + +@end + +// Loopback mode is used to cause the client to connect to itself for testing. +// A second web socket connection is established simulating the other client. +// Any messages received are sent back to the WebSocket server after modifying +// them as appropriate. +@interface ARDLoopbackWebSocketChannel : ARDWebSocketChannel + +- (instancetype)initWithURL:(NSURL *)url restURL:(NSURL *)restURL; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.m new file mode 100644 index 0000000000..bbb0bf87f8 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.m @@ -0,0 +1,252 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDWebSocketChannel.h" + +#import "sdk/objc/base/RTCLogging.h" + +#import "SRWebSocket.h" + +#import "ARDSignalingMessage.h" +#import "ARDUtilities.h" + +// TODO(tkchin): move these to a configuration object. +static NSString const *kARDWSSMessageErrorKey = @"error"; +static NSString const *kARDWSSMessagePayloadKey = @"msg"; + +@interface ARDWebSocketChannel () <SRWebSocketDelegate> +@end + +@implementation ARDWebSocketChannel { + NSURL *_url; + NSURL *_restURL; + SRWebSocket *_socket; +} + +@synthesize delegate = _delegate; +@synthesize state = _state; +@synthesize roomId = _roomId; +@synthesize clientId = _clientId; + +- (instancetype)initWithURL:(NSURL *)url + restURL:(NSURL *)restURL + delegate:(id<ARDSignalingChannelDelegate>)delegate { + if (self = [super init]) { + _url = url; + _restURL = restURL; + _delegate = delegate; + _socket = [[SRWebSocket alloc] initWithURL:url]; + _socket.delegate = self; + RTCLog(@"Opening WebSocket."); + [_socket open]; + } + return self; +} + +- (void)dealloc { + [self disconnect]; +} + +- (void)setState:(ARDSignalingChannelState)state { + if (_state == state) { + return; + } + _state = state; + [_delegate channel:self didChangeState:_state]; +} + +- (void)registerForRoomId:(NSString *)roomId + clientId:(NSString *)clientId { + NSParameterAssert(roomId.length); + NSParameterAssert(clientId.length); + _roomId = roomId; + _clientId = clientId; + if (_state == kARDSignalingChannelStateOpen) { + [self registerWithCollider]; + } +} + +- (void)sendMessage:(ARDSignalingMessage *)message { + NSParameterAssert(_clientId.length); + NSParameterAssert(_roomId.length); + NSData *data = [message JSONData]; + if (_state == kARDSignalingChannelStateRegistered) { + NSString *payload = + [[NSString alloc] initWithData:data encoding:NSUTF8StringEncoding]; + NSDictionary *message = @{ + @"cmd": @"send", + @"msg": payload, + }; + NSData *messageJSONObject = + [NSJSONSerialization dataWithJSONObject:message + options:NSJSONWritingPrettyPrinted + error:nil]; + NSString *messageString = + [[NSString alloc] initWithData:messageJSONObject + encoding:NSUTF8StringEncoding]; + RTCLog(@"C->WSS: %@", messageString); + [_socket send:messageString]; + } else { + NSString *dataString = + [[NSString alloc] initWithData:data encoding:NSUTF8StringEncoding]; + RTCLog(@"C->WSS POST: %@", dataString); + NSString *urlString = + [NSString stringWithFormat:@"%@/%@/%@", + [_restURL absoluteString], _roomId, _clientId]; + NSURL *url = [NSURL URLWithString:urlString]; + [NSURLConnection sendAsyncPostToURL:url + withData:data + completionHandler:nil]; + } +} + +- (void)disconnect { + if (_state == kARDSignalingChannelStateClosed || + _state == kARDSignalingChannelStateError) { + return; + } + [_socket close]; + RTCLog(@"C->WSS DELETE rid:%@ cid:%@", _roomId, _clientId); + NSString *urlString = + [NSString stringWithFormat:@"%@/%@/%@", + [_restURL absoluteString], _roomId, _clientId]; + NSURL *url = [NSURL URLWithString:urlString]; + NSMutableURLRequest *request = [NSMutableURLRequest requestWithURL:url]; + request.HTTPMethod = @"DELETE"; + request.HTTPBody = nil; + [NSURLConnection sendAsyncRequest:request completionHandler:nil]; +} + +#pragma mark - SRWebSocketDelegate + +- (void)webSocketDidOpen:(SRWebSocket *)webSocket { + RTCLog(@"WebSocket connection opened."); + self.state = kARDSignalingChannelStateOpen; + if (_roomId.length && _clientId.length) { + [self registerWithCollider]; + } +} + +- (void)webSocket:(SRWebSocket *)webSocket didReceiveMessage:(id)message { + NSString *messageString = message; + NSData *messageData = [messageString dataUsingEncoding:NSUTF8StringEncoding]; + id jsonObject = [NSJSONSerialization JSONObjectWithData:messageData + options:0 + error:nil]; + if (![jsonObject isKindOfClass:[NSDictionary class]]) { + RTCLogError(@"Unexpected message: %@", jsonObject); + return; + } + NSDictionary *wssMessage = jsonObject; + NSString *errorString = wssMessage[kARDWSSMessageErrorKey]; + if (errorString.length) { + RTCLogError(@"WSS error: %@", errorString); + return; + } + NSString *payload = wssMessage[kARDWSSMessagePayloadKey]; + ARDSignalingMessage *signalingMessage = + [ARDSignalingMessage messageFromJSONString:payload]; + RTCLog(@"WSS->C: %@", payload); + [_delegate channel:self didReceiveMessage:signalingMessage]; +} + +- (void)webSocket:(SRWebSocket *)webSocket didFailWithError:(NSError *)error { + RTCLogError(@"WebSocket error: %@", error); + self.state = kARDSignalingChannelStateError; +} + +- (void)webSocket:(SRWebSocket *)webSocket + didCloseWithCode:(NSInteger)code + reason:(NSString *)reason + wasClean:(BOOL)wasClean { + RTCLog(@"WebSocket closed with code: %ld reason:%@ wasClean:%d", + (long)code, reason, wasClean); + NSParameterAssert(_state != kARDSignalingChannelStateError); + self.state = kARDSignalingChannelStateClosed; +} + +#pragma mark - Private + +- (void)registerWithCollider { + if (_state == kARDSignalingChannelStateRegistered) { + return; + } + NSParameterAssert(_roomId.length); + NSParameterAssert(_clientId.length); + NSDictionary *registerMessage = @{ + @"cmd": @"register", + @"roomid" : _roomId, + @"clientid" : _clientId, + }; + NSData *message = + [NSJSONSerialization dataWithJSONObject:registerMessage + options:NSJSONWritingPrettyPrinted + error:nil]; + NSString *messageString = + [[NSString alloc] initWithData:message encoding:NSUTF8StringEncoding]; + RTCLog(@"Registering on WSS for rid:%@ cid:%@", _roomId, _clientId); + // Registration can fail if server rejects it. For example, if the room is + // full. + [_socket send:messageString]; + self.state = kARDSignalingChannelStateRegistered; +} + +@end + +@interface ARDLoopbackWebSocketChannel () <ARDSignalingChannelDelegate> +@end + +@implementation ARDLoopbackWebSocketChannel + +- (instancetype)initWithURL:(NSURL *)url restURL:(NSURL *)restURL { + return [super initWithURL:url restURL:restURL delegate:self]; +} + +#pragma mark - ARDSignalingChannelDelegate + +- (void)channel:(id<ARDSignalingChannel>)channel + didReceiveMessage:(ARDSignalingMessage *)message { + switch (message.type) { + case kARDSignalingMessageTypeOffer: { + // Change message to answer, send back to server. + ARDSessionDescriptionMessage *sdpMessage = + (ARDSessionDescriptionMessage *)message; + RTC_OBJC_TYPE(RTCSessionDescription) *description = sdpMessage.sessionDescription; + NSString *dsc = description.sdp; + dsc = [dsc stringByReplacingOccurrencesOfString:@"offer" + withString:@"answer"]; + RTC_OBJC_TYPE(RTCSessionDescription) *answerDescription = + [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithType:RTCSdpTypeAnswer sdp:dsc]; + ARDSignalingMessage *answer = + [[ARDSessionDescriptionMessage alloc] + initWithDescription:answerDescription]; + [self sendMessage:answer]; + break; + } + case kARDSignalingMessageTypeAnswer: + // Should not receive answer in loopback scenario. + break; + case kARDSignalingMessageTypeCandidate: + case kARDSignalingMessageTypeCandidateRemoval: + // Send back to server. + [self sendMessage:message]; + break; + case kARDSignalingMessageTypeBye: + // Nothing to do. + return; + } +} + +- (void)channel:(id<ARDSignalingChannel>)channel + didChangeState:(ARDSignalingChannelState)state { +} + +@end + diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.h new file mode 100644 index 0000000000..5fd823f2de --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.h @@ -0,0 +1,23 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "sdk/objc/api/peerconnection/RTCIceCandidate.h" + +@interface RTC_OBJC_TYPE (RTCIceCandidate) +(JSON) + + + (RTC_OBJC_TYPE(RTCIceCandidate) *)candidateFromJSONDictionary : (NSDictionary *)dictionary; ++ (NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidatesFromJSONDictionary: + (NSDictionary *)dictionary; ++ (NSData *)JSONDataForIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates + withType:(NSString *)typeValue; +- (NSData *)JSONData; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.m new file mode 100644 index 0000000000..99cefbff0b --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.m @@ -0,0 +1,100 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "RTCIceCandidate+JSON.h" + +#import "sdk/objc/base/RTCLogging.h" + +static NSString const *kRTCICECandidateTypeKey = @"type"; +static NSString const *kRTCICECandidateTypeValue = @"candidate"; +static NSString const *kRTCICECandidateMidKey = @"id"; +static NSString const *kRTCICECandidateMLineIndexKey = @"label"; +static NSString const *kRTCICECandidateSdpKey = @"candidate"; +static NSString const *kRTCICECandidatesTypeKey = @"candidates"; + +@implementation RTC_OBJC_TYPE (RTCIceCandidate) +(JSON) + + + (RTC_OBJC_TYPE(RTCIceCandidate) *)candidateFromJSONDictionary : (NSDictionary *)dictionary { + NSString *mid = dictionary[kRTCICECandidateMidKey]; + NSString *sdp = dictionary[kRTCICECandidateSdpKey]; + NSNumber *num = dictionary[kRTCICECandidateMLineIndexKey]; + NSInteger mLineIndex = [num integerValue]; + return [[RTC_OBJC_TYPE(RTCIceCandidate) alloc] initWithSdp:sdp + sdpMLineIndex:mLineIndex + sdpMid:mid]; +} + ++ (NSData *)JSONDataForIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates + withType:(NSString *)typeValue { + NSMutableArray *jsonCandidates = + [NSMutableArray arrayWithCapacity:candidates.count]; + for (RTC_OBJC_TYPE(RTCIceCandidate) * candidate in candidates) { + NSDictionary *jsonCandidate = [candidate JSONDictionary]; + [jsonCandidates addObject:jsonCandidate]; + } + NSDictionary *json = @{ + kRTCICECandidateTypeKey : typeValue, + kRTCICECandidatesTypeKey : jsonCandidates + }; + NSError *error = nil; + NSData *data = + [NSJSONSerialization dataWithJSONObject:json + options:NSJSONWritingPrettyPrinted + error:&error]; + if (error) { + RTCLogError(@"Error serializing JSON: %@", error); + return nil; + } + return data; +} + ++ (NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidatesFromJSONDictionary: + (NSDictionary *)dictionary { + NSArray *jsonCandidates = dictionary[kRTCICECandidatesTypeKey]; + NSMutableArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *candidates = + [NSMutableArray arrayWithCapacity:jsonCandidates.count]; + for (NSDictionary *jsonCandidate in jsonCandidates) { + RTC_OBJC_TYPE(RTCIceCandidate) *candidate = + [RTC_OBJC_TYPE(RTCIceCandidate) candidateFromJSONDictionary:jsonCandidate]; + [candidates addObject:candidate]; + } + return candidates; +} + +- (NSData *)JSONData { + NSDictionary *json = @{ + kRTCICECandidateTypeKey : kRTCICECandidateTypeValue, + kRTCICECandidateMLineIndexKey : @(self.sdpMLineIndex), + kRTCICECandidateMidKey : self.sdpMid, + kRTCICECandidateSdpKey : self.sdp + }; + NSError *error = nil; + NSData *data = + [NSJSONSerialization dataWithJSONObject:json + options:NSJSONWritingPrettyPrinted + error:&error]; + if (error) { + RTCLogError(@"Error serializing JSON: %@", error); + return nil; + } + return data; +} + +- (NSDictionary *)JSONDictionary{ + NSDictionary *json = @{ + kRTCICECandidateMLineIndexKey : @(self.sdpMLineIndex), + kRTCICECandidateMidKey : self.sdpMid, + kRTCICECandidateSdpKey : self.sdp + }; + return json; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceServer+JSON.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceServer+JSON.h new file mode 100644 index 0000000000..35f6af7583 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceServer+JSON.h @@ -0,0 +1,18 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "sdk/objc/api/peerconnection/RTCIceServer.h" + +@interface RTC_OBJC_TYPE (RTCIceServer) +(JSON) + + + (RTC_OBJC_TYPE(RTCIceServer) *)serverFromJSONDictionary : (NSDictionary *)dictionary; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceServer+JSON.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceServer+JSON.m new file mode 100644 index 0000000000..b5272a2f64 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceServer+JSON.m @@ -0,0 +1,25 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "RTCIceServer+JSON.h" + +@implementation RTC_OBJC_TYPE (RTCIceServer) +(JSON) + + + (RTC_OBJC_TYPE(RTCIceServer) *)serverFromJSONDictionary : (NSDictionary *)dictionary { + NSArray *turnUrls = dictionary[@"urls"]; + NSString *username = dictionary[@"username"] ?: @""; + NSString *credential = dictionary[@"credential"] ?: @""; + return [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:turnUrls + username:username + credential:credential]; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.h new file mode 100644 index 0000000000..74d03d11b9 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.h @@ -0,0 +1,20 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "sdk/objc/api/peerconnection/RTCSessionDescription.h" + +@interface RTC_OBJC_TYPE (RTCSessionDescription) +(JSON) + + + (RTC_OBJC_TYPE(RTCSessionDescription) *)descriptionFromJSONDictionary + : (NSDictionary *)dictionary; +- (NSData *)JSONData; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.m new file mode 100644 index 0000000000..28268faa84 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.m @@ -0,0 +1,36 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "RTCSessionDescription+JSON.h" + +static NSString const *kRTCSessionDescriptionTypeKey = @"type"; +static NSString const *kRTCSessionDescriptionSdpKey = @"sdp"; + +@implementation RTC_OBJC_TYPE (RTCSessionDescription) +(JSON) + + + (RTC_OBJC_TYPE(RTCSessionDescription) *)descriptionFromJSONDictionary + : (NSDictionary *)dictionary { + NSString *typeString = dictionary[kRTCSessionDescriptionTypeKey]; + RTCSdpType type = [[self class] typeForString:typeString]; + NSString *sdp = dictionary[kRTCSessionDescriptionSdpKey]; + return [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithType:type sdp:sdp]; +} + +- (NSData *)JSONData { + NSString *type = [[self class] stringForType:self.type]; + NSDictionary *json = @{ + kRTCSessionDescriptionTypeKey : type, + kRTCSessionDescriptionSdpKey : self.sdp + }; + return [NSJSONSerialization dataWithJSONObject:json options:0 error:nil]; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/common/ARDUtilities.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/common/ARDUtilities.h new file mode 100644 index 0000000000..5f0d7dbef7 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/common/ARDUtilities.h @@ -0,0 +1,35 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> + +@interface NSDictionary (ARDUtilites) + +// Creates a dictionary with the keys and values in the JSON object. ++ (NSDictionary *)dictionaryWithJSONString:(NSString *)jsonString; ++ (NSDictionary *)dictionaryWithJSONData:(NSData *)jsonData; + +@end + +@interface NSURLConnection (ARDUtilities) + +// Issues an asynchronous request that calls back on main queue. ++ (void)sendAsyncRequest:(NSURLRequest *)request + completionHandler: + (void (^)(NSURLResponse *response, NSData *data, NSError *error))completionHandler; + +// Posts data to the specified URL. ++ (void)sendAsyncPostToURL:(NSURL *)url + withData:(NSData *)data + completionHandler:(void (^)(BOOL succeeded, NSData *data))completionHandler; + +@end + +NSInteger ARDGetCpuUsagePercentage(void); diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/common/ARDUtilities.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/common/ARDUtilities.m new file mode 100644 index 0000000000..e0674f5210 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/common/ARDUtilities.m @@ -0,0 +1,126 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDUtilities.h" + +#import <mach/mach.h> + +#import "sdk/objc/base/RTCLogging.h" + +@implementation NSDictionary (ARDUtilites) + ++ (NSDictionary *)dictionaryWithJSONString:(NSString *)jsonString { + NSParameterAssert(jsonString.length > 0); + NSData *data = [jsonString dataUsingEncoding:NSUTF8StringEncoding]; + NSError *error = nil; + NSDictionary *dict = + [NSJSONSerialization JSONObjectWithData:data options:0 error:&error]; + if (error) { + RTCLogError(@"Error parsing JSON: %@", error.localizedDescription); + } + return dict; +} + ++ (NSDictionary *)dictionaryWithJSONData:(NSData *)jsonData { + NSError *error = nil; + NSDictionary *dict = + [NSJSONSerialization JSONObjectWithData:jsonData options:0 error:&error]; + if (error) { + RTCLogError(@"Error parsing JSON: %@", error.localizedDescription); + } + return dict; +} + +@end + +@implementation NSURLConnection (ARDUtilities) + ++ (void)sendAsyncRequest:(NSURLRequest *)request + completionHandler:(void (^)(NSURLResponse *response, + NSData *data, + NSError *error))completionHandler { + // Kick off an async request which will call back on main thread. + NSURLSession *session = [NSURLSession sharedSession]; + [[session dataTaskWithRequest:request + completionHandler:^(NSData *data, NSURLResponse *response, NSError *error) { + if (completionHandler) { + completionHandler(response, data, error); + } + }] resume]; +} + +// Posts data to the specified URL. ++ (void)sendAsyncPostToURL:(NSURL *)url + withData:(NSData *)data + completionHandler:(void (^)(BOOL succeeded, + NSData *data))completionHandler { + NSMutableURLRequest *request = [NSMutableURLRequest requestWithURL:url]; + request.HTTPMethod = @"POST"; + request.HTTPBody = data; + [[self class] sendAsyncRequest:request + completionHandler:^(NSURLResponse *response, + NSData *data, + NSError *error) { + if (error) { + RTCLogError(@"Error posting data: %@", error.localizedDescription); + if (completionHandler) { + completionHandler(NO, data); + } + return; + } + NSHTTPURLResponse *httpResponse = (NSHTTPURLResponse *)response; + if (httpResponse.statusCode != 200) { + NSString *serverResponse = data.length > 0 ? + [[NSString alloc] initWithData:data encoding:NSUTF8StringEncoding] : + nil; + RTCLogError(@"Received bad response: %@", serverResponse); + if (completionHandler) { + completionHandler(NO, data); + } + return; + } + if (completionHandler) { + completionHandler(YES, data); + } + }]; +} + +@end + +NSInteger ARDGetCpuUsagePercentage(void) { + // Create an array of thread ports for the current task. + const task_t task = mach_task_self(); + thread_act_array_t thread_array; + mach_msg_type_number_t thread_count; + if (task_threads(task, &thread_array, &thread_count) != KERN_SUCCESS) { + return -1; + } + + // Sum cpu usage from all threads. + float cpu_usage_percentage = 0; + thread_basic_info_data_t thread_info_data = {}; + mach_msg_type_number_t thread_info_count; + for (size_t i = 0; i < thread_count; ++i) { + thread_info_count = THREAD_BASIC_INFO_COUNT; + kern_return_t ret = thread_info(thread_array[i], + THREAD_BASIC_INFO, + (thread_info_t)&thread_info_data, + &thread_info_count); + if (ret == KERN_SUCCESS) { + cpu_usage_percentage += + 100.f * (float)thread_info_data.cpu_usage / TH_USAGE_SCALE; + } + } + + // Dealloc the created array. + vm_deallocate(task, (vm_address_t)thread_array, + sizeof(thread_act_t) * thread_count); + return lroundf(cpu_usage_percentage); +} diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.h new file mode 100644 index 0000000000..7eafff8ebc --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.h @@ -0,0 +1,17 @@ +/* + * Copyright 2013 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <UIKit/UIKit.h> + +// The main application class of the AppRTCMobile iOS app demonstrating +// interoperability between the Objective C implementation of PeerConnection +// and the appr.tc demo webapp. +@interface ARDAppDelegate : NSObject <UIApplicationDelegate> +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.m new file mode 100644 index 0000000000..51e9910b87 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.m @@ -0,0 +1,56 @@ +/* + * Copyright 2013 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDAppDelegate.h" + +#import "sdk/objc/api/peerconnection/RTCFieldTrials.h" +#import "sdk/objc/api/peerconnection/RTCSSLAdapter.h" +#import "sdk/objc/api/peerconnection/RTCTracing.h" +#import "sdk/objc/base/RTCLogging.h" + +#import "ARDMainViewController.h" + +@implementation ARDAppDelegate { + UIWindow *_window; +} + +#pragma mark - UIApplicationDelegate methods + +- (BOOL)application:(UIApplication *)application + didFinishLaunchingWithOptions:(NSDictionary *)launchOptions { + NSDictionary *fieldTrials = @{}; + RTCInitFieldTrialDictionary(fieldTrials); + RTCInitializeSSL(); + RTCSetupInternalTracer(); + _window = [[UIWindow alloc] initWithFrame:[[UIScreen mainScreen] bounds]]; + [_window makeKeyAndVisible]; + ARDMainViewController *viewController = [[ARDMainViewController alloc] init]; + + UINavigationController *root = + [[UINavigationController alloc] initWithRootViewController:viewController]; + root.navigationBar.translucent = NO; + _window.rootViewController = root; + +#if defined(NDEBUG) + // In debug builds the default level is LS_INFO and in non-debug builds it is + // disabled. Continue to log to console in non-debug builds, but only + // warnings and errors. + RTCSetMinDebugLogLevel(RTCLoggingSeverityWarning); +#endif + + return YES; +} + +- (void)applicationWillTerminate:(UIApplication *)application { + RTCShutdownInternalTracer(); + RTCCleanupSSL(); +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDFileCaptureController.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDFileCaptureController.h new file mode 100644 index 0000000000..82f8fcdd1b --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDFileCaptureController.h @@ -0,0 +1,42 @@ +/* + * Copyright 2017 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> + +#import "sdk/objc/base/RTCMacros.h" + +@class RTC_OBJC_TYPE(RTCFileVideoCapturer); + +/** + * Controls a file capturer. + */ +NS_CLASS_AVAILABLE_IOS(10) +@interface ARDFileCaptureController : NSObject + +/** + * Creates instance of the controller. + * + * @param capturer The capturer to be controlled. + */ +- (instancetype)initWithCapturer:(RTC_OBJC_TYPE(RTCFileVideoCapturer) *)capturer; + +/** + * Starts the file capturer. + * + * Possible errors produced by the capturer will be logged. + */ +- (void)startCapture; + +/** + * Immediately stops capturer. + */ +- (void)stopCapture; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDFileCaptureController.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDFileCaptureController.m new file mode 100644 index 0000000000..2ddde6dd59 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDFileCaptureController.m @@ -0,0 +1,45 @@ +/* + * Copyright 2017 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDFileCaptureController.h" + +#import "sdk/objc/components/capturer/RTCFileVideoCapturer.h" + +@interface ARDFileCaptureController () + +@property(nonatomic, strong) RTC_OBJC_TYPE(RTCFileVideoCapturer) * fileCapturer; + +@end + +@implementation ARDFileCaptureController +@synthesize fileCapturer = _fileCapturer; + +- (instancetype)initWithCapturer:(RTC_OBJC_TYPE(RTCFileVideoCapturer) *)capturer { + if (self = [super init]) { + _fileCapturer = capturer; + } + return self; +} + +- (void)startCapture { + [self startFileCapture]; +} + +- (void)startFileCapture { + [self.fileCapturer startCapturingFromFileNamed:@"foreman.mp4" + onError:^(NSError *_Nonnull error) { + NSLog(@"Error %@", error.userInfo); + }]; +} + +- (void)stopCapture { + [self.fileCapturer stopCapture]; +} +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainView.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainView.h new file mode 100644 index 0000000000..c6691c2d84 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainView.h @@ -0,0 +1,30 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <UIKit/UIKit.h> + +@class ARDMainView; + +@protocol ARDMainViewDelegate <NSObject> + +- (void)mainView:(ARDMainView *)mainView didInputRoom:(NSString *)room isLoopback:(BOOL)isLoopback; +- (void)mainViewDidToggleAudioLoop:(ARDMainView *)mainView; + +@end + +// The main view of AppRTCMobile. It contains an input field for entering a room +// name on apprtc to connect to. +@interface ARDMainView : UIView + +@property(nonatomic, weak) id<ARDMainViewDelegate> delegate; +// Updates the audio loop button as needed. +@property(nonatomic, assign) BOOL isAudioLoopPlaying; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainView.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainView.m new file mode 100644 index 0000000000..d9521060eb --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainView.m @@ -0,0 +1,196 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDMainView.h" + +#import "UIImage+ARDUtilities.h" + +static CGFloat const kRoomTextFieldHeight = 40; +static CGFloat const kRoomTextFieldMargin = 8; +static CGFloat const kCallControlMargin = 8; + +// Helper view that contains a text field and a clear button. +@interface ARDRoomTextField : UIView <UITextFieldDelegate> +@property(nonatomic, readonly) NSString *roomText; +@end + +@implementation ARDRoomTextField { + UITextField *_roomText; +} + +- (instancetype)initWithFrame:(CGRect)frame { + if (self = [super initWithFrame:frame]) { + _roomText = [[UITextField alloc] initWithFrame:CGRectZero]; + _roomText.borderStyle = UITextBorderStyleNone; + _roomText.font = [UIFont systemFontOfSize:12]; + _roomText.placeholder = @"Room name"; + _roomText.autocorrectionType = UITextAutocorrectionTypeNo; + _roomText.autocapitalizationType = UITextAutocapitalizationTypeNone; + _roomText.clearButtonMode = UITextFieldViewModeAlways; + _roomText.delegate = self; + [self addSubview:_roomText]; + + // Give rounded corners and a light gray border. + self.layer.borderWidth = 1; + self.layer.borderColor = [[UIColor lightGrayColor] CGColor]; + self.layer.cornerRadius = 2; + } + return self; +} + +- (void)layoutSubviews { + _roomText.frame = + CGRectMake(kRoomTextFieldMargin, 0, CGRectGetWidth(self.bounds) - kRoomTextFieldMargin, + kRoomTextFieldHeight); +} + +- (CGSize)sizeThatFits:(CGSize)size { + size.height = kRoomTextFieldHeight; + return size; +} + +- (NSString *)roomText { + return _roomText.text; +} + +#pragma mark - UITextFieldDelegate + +- (BOOL)textFieldShouldReturn:(UITextField *)textField { + // There is no other control that can take focus, so manually resign focus + // when return (Join) is pressed to trigger `textFieldDidEndEditing`. + [textField resignFirstResponder]; + return YES; +} + +@end + +@implementation ARDMainView { + ARDRoomTextField *_roomText; + UIButton *_startRegularCallButton; + UIButton *_startLoopbackCallButton; + UIButton *_audioLoopButton; +} + +@synthesize delegate = _delegate; +@synthesize isAudioLoopPlaying = _isAudioLoopPlaying; + +- (instancetype)initWithFrame:(CGRect)frame { + if (self = [super initWithFrame:frame]) { + _roomText = [[ARDRoomTextField alloc] initWithFrame:CGRectZero]; + [self addSubview:_roomText]; + + UIFont *controlFont = [UIFont boldSystemFontOfSize:18.0]; + UIColor *controlFontColor = [UIColor whiteColor]; + + _startRegularCallButton = [UIButton buttonWithType:UIButtonTypeSystem]; + _startRegularCallButton.titleLabel.font = controlFont; + [_startRegularCallButton setTitleColor:controlFontColor forState:UIControlStateNormal]; + _startRegularCallButton.backgroundColor + = [UIColor colorWithRed:66.0/255.0 green:200.0/255.0 blue:90.0/255.0 alpha:1.0]; + [_startRegularCallButton setTitle:@"Call room" forState:UIControlStateNormal]; + [_startRegularCallButton addTarget:self + action:@selector(onStartRegularCall:) + forControlEvents:UIControlEventTouchUpInside]; + [self addSubview:_startRegularCallButton]; + + _startLoopbackCallButton = [UIButton buttonWithType:UIButtonTypeSystem]; + _startLoopbackCallButton.titleLabel.font = controlFont; + [_startLoopbackCallButton setTitleColor:controlFontColor forState:UIControlStateNormal]; + _startLoopbackCallButton.backgroundColor = + [UIColor colorWithRed:0.0 green:122.0/255.0 blue:1.0 alpha:1.0]; + [_startLoopbackCallButton setTitle:@"Loopback call" forState:UIControlStateNormal]; + [_startLoopbackCallButton addTarget:self + action:@selector(onStartLoopbackCall:) + forControlEvents:UIControlEventTouchUpInside]; + [self addSubview:_startLoopbackCallButton]; + + + // Used to test what happens to sounds when calls are in progress. + _audioLoopButton = [UIButton buttonWithType:UIButtonTypeSystem]; + _audioLoopButton.titleLabel.font = controlFont; + [_audioLoopButton setTitleColor:controlFontColor forState:UIControlStateNormal]; + _audioLoopButton.backgroundColor = + [UIColor colorWithRed:1.0 green:149.0/255.0 blue:0.0 alpha:1.0]; + [self updateAudioLoopButton]; + [_audioLoopButton addTarget:self + action:@selector(onToggleAudioLoop:) + forControlEvents:UIControlEventTouchUpInside]; + [self addSubview:_audioLoopButton]; + + self.backgroundColor = [UIColor whiteColor]; + } + return self; +} + +- (void)setIsAudioLoopPlaying:(BOOL)isAudioLoopPlaying { + if (_isAudioLoopPlaying == isAudioLoopPlaying) { + return; + } + _isAudioLoopPlaying = isAudioLoopPlaying; + [self updateAudioLoopButton]; +} + +- (void)layoutSubviews { + CGRect bounds = self.bounds; + CGFloat roomTextWidth = bounds.size.width - 2 * kRoomTextFieldMargin; + CGFloat roomTextHeight = [_roomText sizeThatFits:bounds.size].height; + _roomText.frame = + CGRectMake(kRoomTextFieldMargin, kRoomTextFieldMargin, roomTextWidth, + roomTextHeight); + + CGFloat buttonHeight = + (CGRectGetMaxY(self.bounds) - CGRectGetMaxY(_roomText.frame) - kCallControlMargin * 4) / 3; + + CGFloat regularCallFrameTop = CGRectGetMaxY(_roomText.frame) + kCallControlMargin; + CGRect regularCallFrame = CGRectMake(kCallControlMargin, + regularCallFrameTop, + bounds.size.width - 2*kCallControlMargin, + buttonHeight); + + CGFloat loopbackCallFrameTop = CGRectGetMaxY(regularCallFrame) + kCallControlMargin; + CGRect loopbackCallFrame = CGRectMake(kCallControlMargin, + loopbackCallFrameTop, + bounds.size.width - 2*kCallControlMargin, + buttonHeight); + + CGFloat audioLoopTop = CGRectGetMaxY(loopbackCallFrame) + kCallControlMargin; + CGRect audioLoopFrame = CGRectMake(kCallControlMargin, + audioLoopTop, + bounds.size.width - 2*kCallControlMargin, + buttonHeight); + + _startRegularCallButton.frame = regularCallFrame; + _startLoopbackCallButton.frame = loopbackCallFrame; + _audioLoopButton.frame = audioLoopFrame; +} + +#pragma mark - Private + +- (void)updateAudioLoopButton { + if (_isAudioLoopPlaying) { + [_audioLoopButton setTitle:@"Stop sound" forState:UIControlStateNormal]; + } else { + [_audioLoopButton setTitle:@"Play sound" forState:UIControlStateNormal]; + } +} + +- (void)onToggleAudioLoop:(id)sender { + [_delegate mainViewDidToggleAudioLoop:self]; +} + +- (void)onStartRegularCall:(id)sender { + [_delegate mainView:self didInputRoom:_roomText.roomText isLoopback:NO]; +} + +- (void)onStartLoopbackCall:(id)sender { + [_delegate mainView:self didInputRoom:_roomText.roomText isLoopback:YES]; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.h new file mode 100644 index 0000000000..e5c92dd304 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.h @@ -0,0 +1,14 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <UIKit/UIKit.h> + +@interface ARDMainViewController : UIViewController +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.m new file mode 100644 index 0000000000..e8b8112e41 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.m @@ -0,0 +1,263 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDMainViewController.h" + +#import <AVFoundation/AVFoundation.h> + +#import "sdk/objc/base/RTCLogging.h" +#import "sdk/objc/components/audio/RTCAudioSession.h" +#import "sdk/objc/components/audio/RTCAudioSessionConfiguration.h" +#import "sdk/objc/helpers/RTCDispatcher.h" + +#import "ARDAppClient.h" +#import "ARDMainView.h" +#import "ARDSettingsModel.h" +#import "ARDSettingsViewController.h" +#import "ARDVideoCallViewController.h" + +static NSString *const barButtonImageString = @"ic_settings_black_24dp.png"; + +// Launch argument to be passed to indicate that the app should start loopback immediatly +static NSString *const loopbackLaunchProcessArgument = @"loopback"; + +@interface ARDMainViewController () <ARDMainViewDelegate, + ARDVideoCallViewControllerDelegate, + RTC_OBJC_TYPE (RTCAudioSessionDelegate)> +@property(nonatomic, strong) ARDMainView *mainView; +@property(nonatomic, strong) AVAudioPlayer *audioPlayer; +@end + +@implementation ARDMainViewController { + BOOL _useManualAudio; +} + +@synthesize mainView = _mainView; +@synthesize audioPlayer = _audioPlayer; + +- (void)viewDidLoad { + [super viewDidLoad]; + if ([[[NSProcessInfo processInfo] arguments] containsObject:loopbackLaunchProcessArgument]) { + [self mainView:nil didInputRoom:@"" isLoopback:YES]; + } +} + +- (void)loadView { + self.title = @"AppRTC Mobile"; + _mainView = [[ARDMainView alloc] initWithFrame:CGRectZero]; + _mainView.delegate = self; + self.view = _mainView; + [self addSettingsBarButton]; + + RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *webRTCConfig = + [RTC_OBJC_TYPE(RTCAudioSessionConfiguration) webRTCConfiguration]; + webRTCConfig.categoryOptions = webRTCConfig.categoryOptions | + AVAudioSessionCategoryOptionDefaultToSpeaker; + [RTC_OBJC_TYPE(RTCAudioSessionConfiguration) setWebRTCConfiguration:webRTCConfig]; + + RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance]; + [session addDelegate:self]; + + [self configureAudioSession]; + [self setupAudioPlayer]; +} + +- (void)addSettingsBarButton { + UIBarButtonItem *settingsButton = + [[UIBarButtonItem alloc] initWithImage:[UIImage imageNamed:barButtonImageString] + style:UIBarButtonItemStylePlain + target:self + action:@selector(showSettings:)]; + self.navigationItem.rightBarButtonItem = settingsButton; +} + ++ (NSString *)loopbackRoomString { + NSString *loopbackRoomString = + [[NSUUID UUID].UUIDString stringByReplacingOccurrencesOfString:@"-" withString:@""]; + return loopbackRoomString; +} + +#pragma mark - ARDMainViewDelegate + +- (void)mainView:(ARDMainView *)mainView didInputRoom:(NSString *)room isLoopback:(BOOL)isLoopback { + if (!room.length) { + if (isLoopback) { + // If this is a loopback call, allow a generated room name. + room = [[self class] loopbackRoomString]; + } else { + [self showAlertWithMessage:@"Missing room name."]; + return; + } + } + // Trim whitespaces. + NSCharacterSet *whitespaceSet = [NSCharacterSet whitespaceCharacterSet]; + NSString *trimmedRoom = [room stringByTrimmingCharactersInSet:whitespaceSet]; + + // Check that room name is valid. + NSError *error = nil; + NSRegularExpressionOptions options = NSRegularExpressionCaseInsensitive; + NSRegularExpression *regex = + [NSRegularExpression regularExpressionWithPattern:@"\\w+" + options:options + error:&error]; + if (error) { + [self showAlertWithMessage:error.localizedDescription]; + return; + } + NSRange matchRange = + [regex rangeOfFirstMatchInString:trimmedRoom + options:0 + range:NSMakeRange(0, trimmedRoom.length)]; + if (matchRange.location == NSNotFound || + matchRange.length != trimmedRoom.length) { + [self showAlertWithMessage:@"Invalid room name."]; + return; + } + + ARDSettingsModel *settingsModel = [[ARDSettingsModel alloc] init]; + + RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance]; + session.useManualAudio = [settingsModel currentUseManualAudioConfigSettingFromStore]; + session.isAudioEnabled = NO; + + // Kick off the video call. + ARDVideoCallViewController *videoCallViewController = + [[ARDVideoCallViewController alloc] initForRoom:trimmedRoom + isLoopback:isLoopback + delegate:self]; + videoCallViewController.modalTransitionStyle = + UIModalTransitionStyleCrossDissolve; + videoCallViewController.modalPresentationStyle = UIModalPresentationFullScreen; + [self presentViewController:videoCallViewController + animated:YES + completion:nil]; +} + +- (void)mainViewDidToggleAudioLoop:(ARDMainView *)mainView { + if (mainView.isAudioLoopPlaying) { + [_audioPlayer stop]; + } else { + [_audioPlayer play]; + } + mainView.isAudioLoopPlaying = _audioPlayer.playing; +} + +#pragma mark - ARDVideoCallViewControllerDelegate + +- (void)viewControllerDidFinish:(ARDVideoCallViewController *)viewController { + if (![viewController isBeingDismissed]) { + RTCLog(@"Dismissing VC"); + [self dismissViewControllerAnimated:YES completion:^{ + [self restartAudioPlayerIfNeeded]; + }]; + } + RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance]; + session.isAudioEnabled = NO; +} + +#pragma mark - RTC_OBJC_TYPE(RTCAudioSessionDelegate) + +- (void)audioSessionDidStartPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session { + // Stop playback on main queue and then configure WebRTC. + [RTC_OBJC_TYPE(RTCDispatcher) + dispatchAsyncOnType:RTCDispatcherTypeMain + block:^{ + if (self.mainView.isAudioLoopPlaying) { + RTCLog(@"Stopping audio loop due to WebRTC start."); + [self.audioPlayer stop]; + } + RTCLog(@"Setting isAudioEnabled to YES."); + session.isAudioEnabled = YES; + }]; +} + +- (void)audioSessionDidStopPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session { + // WebRTC is done with the audio session. Restart playback. + [RTC_OBJC_TYPE(RTCDispatcher) dispatchAsyncOnType:RTCDispatcherTypeMain + block:^{ + RTCLog(@"audioSessionDidStopPlayOrRecord"); + [self restartAudioPlayerIfNeeded]; + }]; +} + +#pragma mark - Private +- (void)showSettings:(id)sender { + ARDSettingsViewController *settingsController = + [[ARDSettingsViewController alloc] initWithStyle:UITableViewStyleGrouped + settingsModel:[[ARDSettingsModel alloc] init]]; + + UINavigationController *navigationController = + [[UINavigationController alloc] initWithRootViewController:settingsController]; + [self presentViewControllerAsModal:navigationController]; +} + +- (void)presentViewControllerAsModal:(UIViewController *)viewController { + [self presentViewController:viewController animated:YES completion:nil]; +} + +- (void)configureAudioSession { + RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *configuration = + [[RTC_OBJC_TYPE(RTCAudioSessionConfiguration) alloc] init]; + configuration.category = AVAudioSessionCategoryAmbient; + configuration.categoryOptions = AVAudioSessionCategoryOptionDuckOthers; + configuration.mode = AVAudioSessionModeDefault; + + RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance]; + [session lockForConfiguration]; + BOOL hasSucceeded = NO; + NSError *error = nil; + if (session.isActive) { + hasSucceeded = [session setConfiguration:configuration error:&error]; + } else { + hasSucceeded = [session setConfiguration:configuration + active:YES + error:&error]; + } + if (!hasSucceeded) { + RTCLogError(@"Error setting configuration: %@", error.localizedDescription); + } + [session unlockForConfiguration]; +} + +- (void)setupAudioPlayer { + NSString *audioFilePath = + [[NSBundle mainBundle] pathForResource:@"mozart" ofType:@"mp3"]; + NSURL *audioFileURL = [NSURL URLWithString:audioFilePath]; + _audioPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:audioFileURL + error:nil]; + _audioPlayer.numberOfLoops = -1; + _audioPlayer.volume = 1.0; + [_audioPlayer prepareToPlay]; +} + +- (void)restartAudioPlayerIfNeeded { + [self configureAudioSession]; + if (_mainView.isAudioLoopPlaying && !self.presentedViewController) { + RTCLog(@"Starting audio loop due to WebRTC end."); + [_audioPlayer play]; + } +} + +- (void)showAlertWithMessage:(NSString*)message { + UIAlertController *alert = + [UIAlertController alertControllerWithTitle:nil + message:message + preferredStyle:UIAlertControllerStyleAlert]; + + UIAlertAction *defaultAction = [UIAlertAction actionWithTitle:@"OK" + style:UIAlertActionStyleDefault + handler:^(UIAlertAction *action){ + }]; + + [alert addAction:defaultAction]; + [self presentViewController:alert animated:YES completion:nil]; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.h new file mode 100644 index 0000000000..759af5416f --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.h @@ -0,0 +1,37 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <UIKit/UIKit.h> + +@class ARDSettingsModel; + +NS_ASSUME_NONNULL_BEGIN +/** + * Displays settings options. + */ +@interface ARDSettingsViewController : UITableViewController + +/** + * Creates new instance. + * + * @param style the table view style that should be used + * @param settingsModel model class for the user settings. + */ +- (instancetype)initWithStyle:(UITableViewStyle)style + settingsModel:(ARDSettingsModel *)settingsModel; + +#pragma mark - Unavailable + +- (instancetype)initWithStyle:(UITableViewStyle)style NS_UNAVAILABLE; +- (instancetype)init NS_UNAVAILABLE; ++ (instancetype)new NS_UNAVAILABLE; + +@end +NS_ASSUME_NONNULL_END diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.m new file mode 100644 index 0000000000..9bcbd3aa5c --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.m @@ -0,0 +1,361 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDSettingsViewController.h" +#import "ARDSettingsModel.h" +#import "RTCVideoCodecInfo+HumanReadable.h" + +NS_ASSUME_NONNULL_BEGIN + +typedef NS_ENUM(int, ARDSettingsSections) { + ARDSettingsSectionAudioSettings = 0, + ARDSettingsSectionVideoResolution, + ARDSettingsSectionVideoCodec, + ARDSettingsSectionBitRate, +}; + +typedef NS_ENUM(int, ARDAudioSettingsOptions) { + ARDAudioSettingsAudioOnly = 0, + ARDAudioSettingsCreateAecDump, + ARDAudioSettingsUseManualAudioConfig, +}; + +@interface ARDSettingsViewController () <UITextFieldDelegate> { + ARDSettingsModel *_settingsModel; +} + +@end + +@implementation ARDSettingsViewController + +- (instancetype)initWithStyle:(UITableViewStyle)style + settingsModel:(ARDSettingsModel *)settingsModel { + self = [super initWithStyle:style]; + if (self) { + _settingsModel = settingsModel; + } + return self; +} + +#pragma mark - View lifecycle + +- (void)viewDidLoad { + [super viewDidLoad]; + self.title = @"Settings"; + [self addDoneBarButton]; +} + +- (void)viewWillAppear:(BOOL)animated { + [super viewWillAppear:animated]; +} + +#pragma mark - Data source + +- (NSArray<NSString *> *)videoResolutionArray { + return [_settingsModel availableVideoResolutions]; +} + +- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)videoCodecArray { + return [_settingsModel availableVideoCodecs]; +} + +#pragma mark - + +- (void)addDoneBarButton { + UIBarButtonItem *barItem = + [[UIBarButtonItem alloc] initWithBarButtonSystemItem:UIBarButtonSystemItemDone + target:self + action:@selector(dismissModally:)]; + self.navigationItem.leftBarButtonItem = barItem; +} + +#pragma mark - Dismissal of view controller + +- (void)dismissModally:(id)sender { + [self dismissViewControllerAnimated:YES completion:nil]; +} + +#pragma mark - Table view data source + +- (NSInteger)numberOfSectionsInTableView:(UITableView *)tableView { + return 4; +} + +- (NSInteger)tableView:(UITableView *)tableView numberOfRowsInSection:(NSInteger)section { + switch (section) { + case ARDSettingsSectionAudioSettings: + return 3; + case ARDSettingsSectionVideoResolution: + return self.videoResolutionArray.count; + case ARDSettingsSectionVideoCodec: + return self.videoCodecArray.count; + default: + return 1; + } +} + +#pragma mark - Table view delegate helpers + +- (void)removeAllAccessories:(UITableView *)tableView + inSection:(int)section +{ + for (int i = 0; i < [tableView numberOfRowsInSection:section]; i++) { + NSIndexPath *rowPath = [NSIndexPath indexPathForRow:i inSection:section]; + UITableViewCell *cell = [tableView cellForRowAtIndexPath:rowPath]; + cell.accessoryType = UITableViewCellAccessoryNone; + } +} + +- (void)tableView:(UITableView *)tableView +updateListSelectionAtIndexPath:(NSIndexPath *)indexPath + inSection:(int)section { + [self removeAllAccessories:tableView inSection:section]; + UITableViewCell *cell = [tableView cellForRowAtIndexPath:indexPath]; + cell.accessoryType = UITableViewCellAccessoryCheckmark; + [tableView deselectRowAtIndexPath:indexPath animated:YES]; +} + +#pragma mark - Table view delegate + +- (nullable NSString *)tableView:(UITableView *)tableView + titleForHeaderInSection:(NSInteger)section { + switch (section) { + case ARDSettingsSectionAudioSettings: + return @"Audio"; + case ARDSettingsSectionVideoResolution: + return @"Video resolution"; + case ARDSettingsSectionVideoCodec: + return @"Video codec"; + case ARDSettingsSectionBitRate: + return @"Maximum bitrate"; + default: + return @""; + } +} + +- (UITableViewCell *)tableView:(UITableView *)tableView + cellForRowAtIndexPath:(NSIndexPath *)indexPath { + switch (indexPath.section) { + case ARDSettingsSectionAudioSettings: + return [self audioSettingsTableViewCellForTableView:tableView atIndexPath:indexPath]; + + case ARDSettingsSectionVideoResolution: + return [self videoResolutionTableViewCellForTableView:tableView atIndexPath:indexPath]; + + case ARDSettingsSectionVideoCodec: + return [self videoCodecTableViewCellForTableView:tableView atIndexPath:indexPath]; + + case ARDSettingsSectionBitRate: + return [self bitrateTableViewCellForTableView:tableView atIndexPath:indexPath]; + + default: + return [[UITableViewCell alloc] initWithStyle:UITableViewCellStyleDefault + reuseIdentifier:@"identifier"]; + } +} + +- (void)tableView:(UITableView *)tableView didSelectRowAtIndexPath:(NSIndexPath *)indexPath { + switch (indexPath.section) { + case ARDSettingsSectionVideoResolution: + [self tableView:tableView disSelectVideoResolutionAtIndex:indexPath]; + break; + + case ARDSettingsSectionVideoCodec: + [self tableView:tableView didSelectVideoCodecCellAtIndexPath:indexPath]; + break; + } +} + +#pragma mark - Table view delegate(Video Resolution) + +- (UITableViewCell *)videoResolutionTableViewCellForTableView:(UITableView *)tableView + atIndexPath:(NSIndexPath *)indexPath { + NSString *dequeueIdentifier = @"ARDSettingsVideoResolutionViewCellIdentifier"; + UITableViewCell *cell = [tableView dequeueReusableCellWithIdentifier:dequeueIdentifier]; + if (!cell) { + cell = [[UITableViewCell alloc] initWithStyle:UITableViewCellStyleDefault + reuseIdentifier:dequeueIdentifier]; + } + NSString *resolution = self.videoResolutionArray[indexPath.row]; + cell.textLabel.text = resolution; + if ([resolution isEqualToString:[_settingsModel currentVideoResolutionSettingFromStore]]) { + cell.accessoryType = UITableViewCellAccessoryCheckmark; + } else { + cell.accessoryType = UITableViewCellAccessoryNone; + } + + return cell; +} + +- (void)tableView:(UITableView *)tableView + disSelectVideoResolutionAtIndex:(NSIndexPath *)indexPath { + [self tableView:tableView + updateListSelectionAtIndexPath:indexPath + inSection:ARDSettingsSectionVideoResolution]; + + NSString *videoResolution = self.videoResolutionArray[indexPath.row]; + [_settingsModel storeVideoResolutionSetting:videoResolution]; +} + +#pragma mark - Table view delegate(Video Codec) + +- (UITableViewCell *)videoCodecTableViewCellForTableView:(UITableView *)tableView + atIndexPath:(NSIndexPath *)indexPath { + NSString *dequeueIdentifier = @"ARDSettingsVideoCodecCellIdentifier"; + UITableViewCell *cell = [tableView dequeueReusableCellWithIdentifier:dequeueIdentifier]; + if (!cell) { + cell = [[UITableViewCell alloc] initWithStyle:UITableViewCellStyleDefault + reuseIdentifier:dequeueIdentifier]; + } + RTC_OBJC_TYPE(RTCVideoCodecInfo) *codec = self.videoCodecArray[indexPath.row]; + cell.textLabel.text = [codec humanReadableDescription]; + if ([codec isEqualToCodecInfo:[_settingsModel currentVideoCodecSettingFromStore]]) { + cell.accessoryType = UITableViewCellAccessoryCheckmark; + } else { + cell.accessoryType = UITableViewCellAccessoryNone; + } + + return cell; +} + +- (void)tableView:(UITableView *)tableView + didSelectVideoCodecCellAtIndexPath:(NSIndexPath *)indexPath { + [self tableView:tableView + updateListSelectionAtIndexPath:indexPath + inSection:ARDSettingsSectionVideoCodec]; + + RTC_OBJC_TYPE(RTCVideoCodecInfo) *videoCodec = self.videoCodecArray[indexPath.row]; + [_settingsModel storeVideoCodecSetting:videoCodec]; +} + +#pragma mark - Table view delegate(Bitrate) + +- (UITableViewCell *)bitrateTableViewCellForTableView:(UITableView *)tableView + atIndexPath:(NSIndexPath *)indexPath { + NSString *dequeueIdentifier = @"ARDSettingsBitrateCellIdentifier"; + UITableViewCell *cell = [tableView dequeueReusableCellWithIdentifier:dequeueIdentifier]; + if (!cell) { + cell = [[UITableViewCell alloc] initWithStyle:UITableViewCellStyleDefault + reuseIdentifier:dequeueIdentifier]; + + UITextField *textField = [[UITextField alloc] + initWithFrame:CGRectMake(10, 0, cell.bounds.size.width - 20, cell.bounds.size.height)]; + NSString *currentMaxBitrate = [_settingsModel currentMaxBitrateSettingFromStore].stringValue; + textField.text = currentMaxBitrate; + textField.placeholder = @"Enter max bit rate (kbps)"; + textField.keyboardType = UIKeyboardTypeNumberPad; + textField.delegate = self; + + // Numerical keyboards have no return button, we need to add one manually. + UIToolbar *numberToolbar = + [[UIToolbar alloc] initWithFrame:CGRectMake(0, 0, self.view.bounds.size.width, 50)]; + numberToolbar.items = @[ + [[UIBarButtonItem alloc] initWithBarButtonSystemItem:UIBarButtonSystemItemFlexibleSpace + target:nil + action:nil], + [[UIBarButtonItem alloc] initWithTitle:@"Apply" + style:UIBarButtonItemStyleDone + target:self + action:@selector(numberTextFieldDidEndEditing:)] + ]; + [numberToolbar sizeToFit]; + + textField.inputAccessoryView = numberToolbar; + [cell addSubview:textField]; + } + return cell; +} + +- (void)numberTextFieldDidEndEditing:(id)sender { + [self.view endEditing:YES]; +} + +- (void)textFieldDidEndEditing:(UITextField *)textField { + NSNumber *bitrateNumber = nil; + + if (textField.text.length != 0) { + bitrateNumber = [NSNumber numberWithInteger:textField.text.intValue]; + } + + [_settingsModel storeMaxBitrateSetting:bitrateNumber]; +} + +#pragma mark - Table view delegate(Audio settings) + +- (UITableViewCell *)audioSettingsTableViewCellForTableView:(UITableView *)tableView + atIndexPath:(NSIndexPath *)indexPath { + NSString *dequeueIdentifier = @"ARDSettingsAudioSettingsCellIdentifier"; + UITableViewCell *cell = [tableView dequeueReusableCellWithIdentifier:dequeueIdentifier]; + if (!cell) { + cell = [[UITableViewCell alloc] initWithStyle:UITableViewCellStyleDefault + reuseIdentifier:dequeueIdentifier]; + cell.selectionStyle = UITableViewCellSelectionStyleNone; + UISwitch *switchView = [[UISwitch alloc] initWithFrame:CGRectZero]; + switchView.tag = indexPath.row; + [switchView addTarget:self + action:@selector(audioSettingSwitchChanged:) + forControlEvents:UIControlEventValueChanged]; + cell.accessoryView = switchView; + } + + cell.textLabel.text = [self labelForAudioSettingAtIndexPathRow:indexPath.row]; + UISwitch *switchView = (UISwitch *)cell.accessoryView; + switchView.on = [self valueForAudioSettingAtIndexPathRow:indexPath.row]; + + return cell; +} + +- (NSString *)labelForAudioSettingAtIndexPathRow:(NSInteger)setting { + switch (setting) { + case ARDAudioSettingsAudioOnly: + return @"Audio only"; + case ARDAudioSettingsCreateAecDump: + return @"Create AecDump"; + case ARDAudioSettingsUseManualAudioConfig: + return @"Use manual audio config"; + default: + return @""; + } +} + +- (BOOL)valueForAudioSettingAtIndexPathRow:(NSInteger)setting { + switch (setting) { + case ARDAudioSettingsAudioOnly: + return [_settingsModel currentAudioOnlySettingFromStore]; + case ARDAudioSettingsCreateAecDump: + return [_settingsModel currentCreateAecDumpSettingFromStore]; + case ARDAudioSettingsUseManualAudioConfig: + return [_settingsModel currentUseManualAudioConfigSettingFromStore]; + default: + return NO; + } +} + +- (void)audioSettingSwitchChanged:(UISwitch *)sender { + switch (sender.tag) { + case ARDAudioSettingsAudioOnly: { + [_settingsModel storeAudioOnlySetting:sender.isOn]; + break; + } + case ARDAudioSettingsCreateAecDump: { + [_settingsModel storeCreateAecDumpSetting:sender.isOn]; + break; + } + case ARDAudioSettingsUseManualAudioConfig: { + [_settingsModel storeUseManualAudioConfigSetting:sender.isOn]; + break; + } + default: + break; + } +} + +@end +NS_ASSUME_NONNULL_END diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDStatsView.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDStatsView.h new file mode 100644 index 0000000000..72207de64e --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDStatsView.h @@ -0,0 +1,21 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <UIKit/UIKit.h> + +#import "sdk/objc/base/RTCMacros.h" + +@class RTC_OBJC_TYPE(RTCStatisticsReport); + +@interface ARDStatsView : UIView + +- (void)setStats:(RTC_OBJC_TYPE(RTCStatisticsReport) *)stats; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDStatsView.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDStatsView.m new file mode 100644 index 0000000000..867ba5b09e --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDStatsView.m @@ -0,0 +1,50 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDStatsView.h" + +#import "sdk/objc/api/peerconnection/RTCLegacyStatsReport.h" + +#import "ARDStatsBuilder.h" + +@implementation ARDStatsView { + UILabel *_statsLabel; + ARDStatsBuilder *_statsBuilder; +} + +- (instancetype)initWithFrame:(CGRect)frame { + if (self = [super initWithFrame:frame]) { + _statsLabel = [[UILabel alloc] initWithFrame:CGRectZero]; + _statsLabel.numberOfLines = 0; + _statsLabel.font = [UIFont fontWithName:@"Roboto" size:12]; + _statsLabel.adjustsFontSizeToFitWidth = YES; + _statsLabel.minimumScaleFactor = 0.6; + _statsLabel.textColor = [UIColor greenColor]; + [self addSubview:_statsLabel]; + self.backgroundColor = [UIColor colorWithWhite:0 alpha:.6]; + _statsBuilder = [[ARDStatsBuilder alloc] init]; + } + return self; +} + +- (void)setStats:(RTC_OBJC_TYPE(RTCStatisticsReport) *)stats { + _statsBuilder.stats = stats; + _statsLabel.text = _statsBuilder.statsString; +} + +- (void)layoutSubviews { + _statsLabel.frame = self.bounds; +} + +- (CGSize)sizeThatFits:(CGSize)size { + return [_statsLabel sizeThatFits:size]; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallView.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallView.h new file mode 100644 index 0000000000..a31c7fe742 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallView.h @@ -0,0 +1,47 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <UIKit/UIKit.h> + +#import "sdk/objc/base/RTCVideoRenderer.h" +#import "sdk/objc/helpers/RTCCameraPreviewView.h" + +#import "ARDStatsView.h" + +@class ARDVideoCallView; +@protocol ARDVideoCallViewDelegate <NSObject> + +// Called when the camera switch button is pressed. +- (void)videoCallView:(ARDVideoCallView *)view + shouldSwitchCameraWithCompletion:(void (^)(NSError *))completion; + +// Called when the route change button is pressed. +- (void)videoCallView:(ARDVideoCallView *)view + shouldChangeRouteWithCompletion:(void (^)(void))completion; + +// Called when the hangup button is pressed. +- (void)videoCallViewDidHangup:(ARDVideoCallView *)view; + +// Called when stats are enabled by triple tapping. +- (void)videoCallViewDidEnableStats:(ARDVideoCallView *)view; + +@end + +// Video call view that shows local and remote video, provides a label to +// display status, and also a hangup button. +@interface ARDVideoCallView : UIView + +@property(nonatomic, readonly) UILabel *statusLabel; +@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCCameraPreviewView) * localVideoView; +@property(nonatomic, readonly) __kindof UIView<RTC_OBJC_TYPE(RTCVideoRenderer)> *remoteVideoView; +@property(nonatomic, readonly) ARDStatsView *statsView; +@property(nonatomic, weak) id<ARDVideoCallViewDelegate> delegate; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallView.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallView.m new file mode 100644 index 0000000000..437aea8d56 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallView.m @@ -0,0 +1,213 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDVideoCallView.h" + +#import <AVFoundation/AVFoundation.h> + +#import "sdk/objc/components/renderer/metal/RTCMTLVideoView.h" + +#import "UIImage+ARDUtilities.h" + +static CGFloat const kButtonPadding = 16; +static CGFloat const kButtonSize = 48; +static CGFloat const kLocalVideoViewSize = 120; +static CGFloat const kLocalVideoViewPadding = 8; +static CGFloat const kStatusBarHeight = 20; + +@interface ARDVideoCallView () <RTC_OBJC_TYPE (RTCVideoViewDelegate)> +@end + +@implementation ARDVideoCallView { + UIButton *_routeChangeButton; + UIButton *_cameraSwitchButton; + UIButton *_hangupButton; + CGSize _remoteVideoSize; +} + +@synthesize statusLabel = _statusLabel; +@synthesize localVideoView = _localVideoView; +@synthesize remoteVideoView = _remoteVideoView; +@synthesize statsView = _statsView; +@synthesize delegate = _delegate; + +- (instancetype)initWithFrame:(CGRect)frame { + if (self = [super initWithFrame:frame]) { + + _remoteVideoView = [[RTC_OBJC_TYPE(RTCMTLVideoView) alloc] initWithFrame:CGRectZero]; + + [self addSubview:_remoteVideoView]; + + _localVideoView = [[RTC_OBJC_TYPE(RTCCameraPreviewView) alloc] initWithFrame:CGRectZero]; + [self addSubview:_localVideoView]; + + _statsView = [[ARDStatsView alloc] initWithFrame:CGRectZero]; + _statsView.hidden = YES; + [self addSubview:_statsView]; + + _routeChangeButton = [UIButton buttonWithType:UIButtonTypeCustom]; + _routeChangeButton.backgroundColor = [UIColor grayColor]; + _routeChangeButton.layer.cornerRadius = kButtonSize / 2; + _routeChangeButton.layer.masksToBounds = YES; + UIImage *image = [UIImage imageForName:@"ic_surround_sound_black_24dp.png" + color:[UIColor whiteColor]]; + [_routeChangeButton setImage:image forState:UIControlStateNormal]; + [_routeChangeButton addTarget:self + action:@selector(onRouteChange:) + forControlEvents:UIControlEventTouchUpInside]; + [self addSubview:_routeChangeButton]; + + // TODO(tkchin): don't display this if we can't actually do camera switch. + _cameraSwitchButton = [UIButton buttonWithType:UIButtonTypeCustom]; + _cameraSwitchButton.backgroundColor = [UIColor grayColor]; + _cameraSwitchButton.layer.cornerRadius = kButtonSize / 2; + _cameraSwitchButton.layer.masksToBounds = YES; + image = [UIImage imageForName:@"ic_switch_video_black_24dp.png" color:[UIColor whiteColor]]; + [_cameraSwitchButton setImage:image forState:UIControlStateNormal]; + [_cameraSwitchButton addTarget:self + action:@selector(onCameraSwitch:) + forControlEvents:UIControlEventTouchUpInside]; + [self addSubview:_cameraSwitchButton]; + + _hangupButton = [UIButton buttonWithType:UIButtonTypeCustom]; + _hangupButton.backgroundColor = [UIColor redColor]; + _hangupButton.layer.cornerRadius = kButtonSize / 2; + _hangupButton.layer.masksToBounds = YES; + image = [UIImage imageForName:@"ic_call_end_black_24dp.png" + color:[UIColor whiteColor]]; + [_hangupButton setImage:image forState:UIControlStateNormal]; + [_hangupButton addTarget:self + action:@selector(onHangup:) + forControlEvents:UIControlEventTouchUpInside]; + [self addSubview:_hangupButton]; + + _statusLabel = [[UILabel alloc] initWithFrame:CGRectZero]; + _statusLabel.font = [UIFont fontWithName:@"Roboto" size:16]; + _statusLabel.textColor = [UIColor whiteColor]; + [self addSubview:_statusLabel]; + + UITapGestureRecognizer *tapRecognizer = + [[UITapGestureRecognizer alloc] + initWithTarget:self + action:@selector(didTripleTap:)]; + tapRecognizer.numberOfTapsRequired = 3; + [self addGestureRecognizer:tapRecognizer]; + } + return self; +} + +- (void)layoutSubviews { + CGRect bounds = self.bounds; + if (_remoteVideoSize.width > 0 && _remoteVideoSize.height > 0) { + // Aspect fill remote video into bounds. + CGRect remoteVideoFrame = + AVMakeRectWithAspectRatioInsideRect(_remoteVideoSize, bounds); + CGFloat scale = 1; + if (remoteVideoFrame.size.width > remoteVideoFrame.size.height) { + // Scale by height. + scale = bounds.size.height / remoteVideoFrame.size.height; + } else { + // Scale by width. + scale = bounds.size.width / remoteVideoFrame.size.width; + } + remoteVideoFrame.size.height *= scale; + remoteVideoFrame.size.width *= scale; + _remoteVideoView.frame = remoteVideoFrame; + _remoteVideoView.center = + CGPointMake(CGRectGetMidX(bounds), CGRectGetMidY(bounds)); + } else { + _remoteVideoView.frame = bounds; + } + + // Aspect fit local video view into a square box. + CGRect localVideoFrame = + CGRectMake(0, 0, kLocalVideoViewSize, kLocalVideoViewSize); + // Place the view in the bottom right. + localVideoFrame.origin.x = CGRectGetMaxX(bounds) + - localVideoFrame.size.width - kLocalVideoViewPadding; + localVideoFrame.origin.y = CGRectGetMaxY(bounds) + - localVideoFrame.size.height - kLocalVideoViewPadding; + _localVideoView.frame = localVideoFrame; + + // Place stats at the top. + CGSize statsSize = [_statsView sizeThatFits:bounds.size]; + _statsView.frame = CGRectMake(CGRectGetMinX(bounds), + CGRectGetMinY(bounds) + kStatusBarHeight, + statsSize.width, statsSize.height); + + // Place hangup button in the bottom left. + _hangupButton.frame = + CGRectMake(CGRectGetMinX(bounds) + kButtonPadding, + CGRectGetMaxY(bounds) - kButtonPadding - + kButtonSize, + kButtonSize, + kButtonSize); + + // Place button to the right of hangup button. + CGRect cameraSwitchFrame = _hangupButton.frame; + cameraSwitchFrame.origin.x = + CGRectGetMaxX(cameraSwitchFrame) + kButtonPadding; + _cameraSwitchButton.frame = cameraSwitchFrame; + + // Place route button to the right of camera button. + CGRect routeChangeFrame = _cameraSwitchButton.frame; + routeChangeFrame.origin.x = + CGRectGetMaxX(routeChangeFrame) + kButtonPadding; + _routeChangeButton.frame = routeChangeFrame; + + [_statusLabel sizeToFit]; + _statusLabel.center = + CGPointMake(CGRectGetMidX(bounds), CGRectGetMidY(bounds)); +} + +#pragma mark - RTC_OBJC_TYPE(RTCVideoViewDelegate) + +- (void)videoView:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)videoView didChangeVideoSize:(CGSize)size { + if (videoView == _remoteVideoView) { + _remoteVideoSize = size; + } + [self setNeedsLayout]; +} + +#pragma mark - Private + +- (void)onCameraSwitch:(UIButton *)sender { + sender.enabled = false; + [_delegate videoCallView:self + shouldSwitchCameraWithCompletion:^(NSError *error) { + dispatch_async(dispatch_get_main_queue(), ^(void) { + sender.enabled = true; + }); + }]; +} + +- (void)onRouteChange:(UIButton *)sender { + sender.enabled = false; + __weak ARDVideoCallView *weakSelf = self; + [_delegate videoCallView:self + shouldChangeRouteWithCompletion:^(void) { + ARDVideoCallView *strongSelf = weakSelf; + if (strongSelf) { + dispatch_async(dispatch_get_main_queue(), ^(void) { + sender.enabled = true; + }); + } + }]; +} + +- (void)onHangup:(id)sender { + [_delegate videoCallViewDidHangup:self]; +} + +- (void)didTripleTap:(UITapGestureRecognizer *)recognizer { + [_delegate videoCallViewDidEnableStats:self]; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.h new file mode 100644 index 0000000000..bdb8747524 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.h @@ -0,0 +1,28 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <UIKit/UIKit.h> + +@class ARDVideoCallViewController; +@protocol ARDVideoCallViewControllerDelegate <NSObject> + +- (void)viewControllerDidFinish:(ARDVideoCallViewController *)viewController; + +@end + +@interface ARDVideoCallViewController : UIViewController + +@property(nonatomic, weak) id<ARDVideoCallViewControllerDelegate> delegate; + +- (instancetype)initForRoom:(NSString *)room + isLoopback:(BOOL)isLoopback + delegate:(id<ARDVideoCallViewControllerDelegate>)delegate; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m new file mode 100644 index 0000000000..a82d90b290 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m @@ -0,0 +1,250 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDVideoCallViewController.h" + +#import "sdk/objc/api/peerconnection/RTCMediaConstraints.h" +#import "sdk/objc/base/RTCLogging.h" +#import "sdk/objc/components/audio/RTCAudioSession.h" +#import "sdk/objc/components/capturer/RTCCameraVideoCapturer.h" +#import "sdk/objc/helpers/RTCDispatcher.h" + +#import "ARDAppClient.h" +#import "ARDCaptureController.h" +#import "ARDFileCaptureController.h" +#import "ARDSettingsModel.h" +#import "ARDVideoCallView.h" + +@interface ARDVideoCallViewController () <ARDAppClientDelegate, + ARDVideoCallViewDelegate, + RTC_OBJC_TYPE (RTCAudioSessionDelegate)> +@property(nonatomic, strong) RTC_OBJC_TYPE(RTCVideoTrack) * remoteVideoTrack; +@property(nonatomic, readonly) ARDVideoCallView *videoCallView; +@property(nonatomic, assign) AVAudioSessionPortOverride portOverride; +@end + +@implementation ARDVideoCallViewController { + ARDAppClient *_client; + RTC_OBJC_TYPE(RTCVideoTrack) * _remoteVideoTrack; + ARDCaptureController *_captureController; + ARDFileCaptureController *_fileCaptureController NS_AVAILABLE_IOS(10); +} + +@synthesize videoCallView = _videoCallView; +@synthesize remoteVideoTrack = _remoteVideoTrack; +@synthesize delegate = _delegate; +@synthesize portOverride = _portOverride; + +- (instancetype)initForRoom:(NSString *)room + isLoopback:(BOOL)isLoopback + delegate:(id<ARDVideoCallViewControllerDelegate>)delegate { + if (self = [super init]) { + ARDSettingsModel *settingsModel = [[ARDSettingsModel alloc] init]; + _delegate = delegate; + + _client = [[ARDAppClient alloc] initWithDelegate:self]; + [_client connectToRoomWithId:room settings:settingsModel isLoopback:isLoopback]; + } + return self; +} + +- (void)loadView { + _videoCallView = [[ARDVideoCallView alloc] initWithFrame:CGRectZero]; + _videoCallView.delegate = self; + _videoCallView.statusLabel.text = + [self statusTextForState:RTCIceConnectionStateNew]; + self.view = _videoCallView; + + RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance]; + [session addDelegate:self]; +} + +- (UIInterfaceOrientationMask)supportedInterfaceOrientations { + return UIInterfaceOrientationMaskAll; +} + +#pragma mark - ARDAppClientDelegate + +- (void)appClient:(ARDAppClient *)client + didChangeState:(ARDAppClientState)state { + switch (state) { + case kARDAppClientStateConnected: + RTCLog(@"Client connected."); + break; + case kARDAppClientStateConnecting: + RTCLog(@"Client connecting."); + break; + case kARDAppClientStateDisconnected: + RTCLog(@"Client disconnected."); + [self hangup]; + break; + } +} + +- (void)appClient:(ARDAppClient *)client + didChangeConnectionState:(RTCIceConnectionState)state { + RTCLog(@"ICE state changed: %ld", (long)state); + __weak ARDVideoCallViewController *weakSelf = self; + dispatch_async(dispatch_get_main_queue(), ^{ + ARDVideoCallViewController *strongSelf = weakSelf; + strongSelf.videoCallView.statusLabel.text = + [strongSelf statusTextForState:state]; + }); +} + +- (void)appClient:(ARDAppClient *)client + didCreateLocalCapturer:(RTC_OBJC_TYPE(RTCCameraVideoCapturer) *)localCapturer { + _videoCallView.localVideoView.captureSession = localCapturer.captureSession; + ARDSettingsModel *settingsModel = [[ARDSettingsModel alloc] init]; + _captureController = + [[ARDCaptureController alloc] initWithCapturer:localCapturer settings:settingsModel]; + [_captureController startCapture]; +} + +- (void)appClient:(ARDAppClient *)client + didCreateLocalFileCapturer:(RTC_OBJC_TYPE(RTCFileVideoCapturer) *)fileCapturer { +#if defined(__IPHONE_11_0) && (__IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_11_0) + if (@available(iOS 10, *)) { + _fileCaptureController = [[ARDFileCaptureController alloc] initWithCapturer:fileCapturer]; + [_fileCaptureController startCapture]; + } +#endif +} + +- (void)appClient:(ARDAppClient *)client + didReceiveLocalVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)localVideoTrack { +} + +- (void)appClient:(ARDAppClient *)client + didReceiveRemoteVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)remoteVideoTrack { + self.remoteVideoTrack = remoteVideoTrack; + __weak ARDVideoCallViewController *weakSelf = self; + dispatch_async(dispatch_get_main_queue(), ^{ + ARDVideoCallViewController *strongSelf = weakSelf; + strongSelf.videoCallView.statusLabel.hidden = YES; + }); +} + +- (void)appClient:(ARDAppClient *)client didGetStats:(RTC_OBJC_TYPE(RTCStatisticsReport) *)stats { + _videoCallView.statsView.stats = stats; + [_videoCallView setNeedsLayout]; +} + +- (void)appClient:(ARDAppClient *)client + didError:(NSError *)error { + NSString *message = + [NSString stringWithFormat:@"%@", error.localizedDescription]; + [self hangup]; + [self showAlertWithMessage:message]; +} + +#pragma mark - ARDVideoCallViewDelegate + +- (void)videoCallViewDidHangup:(ARDVideoCallView *)view { + [self hangup]; +} + +- (void)videoCallView:(ARDVideoCallView *)view + shouldSwitchCameraWithCompletion:(void (^)(NSError *))completion { + [_captureController switchCamera:completion]; +} + +- (void)videoCallView:(ARDVideoCallView *)view + shouldChangeRouteWithCompletion:(void (^)(void))completion { + NSParameterAssert(completion); + AVAudioSessionPortOverride override = AVAudioSessionPortOverrideNone; + if (_portOverride == AVAudioSessionPortOverrideNone) { + override = AVAudioSessionPortOverrideSpeaker; + } + [RTC_OBJC_TYPE(RTCDispatcher) dispatchAsyncOnType:RTCDispatcherTypeAudioSession + block:^{ + RTC_OBJC_TYPE(RTCAudioSession) *session = + [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance]; + [session lockForConfiguration]; + NSError *error = nil; + if ([session overrideOutputAudioPort:override + error:&error]) { + self.portOverride = override; + } else { + RTCLogError(@"Error overriding output port: %@", + error.localizedDescription); + } + [session unlockForConfiguration]; + completion(); + }]; +} + +- (void)videoCallViewDidEnableStats:(ARDVideoCallView *)view { + _client.shouldGetStats = YES; + _videoCallView.statsView.hidden = NO; +} + +#pragma mark - RTC_OBJC_TYPE(RTCAudioSessionDelegate) + +- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession + didDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches { + RTCLog(@"Audio session detected glitch, total: %lld", totalNumberOfGlitches); +} + +#pragma mark - Private + +- (void)setRemoteVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)remoteVideoTrack { + if (_remoteVideoTrack == remoteVideoTrack) { + return; + } + [_remoteVideoTrack removeRenderer:_videoCallView.remoteVideoView]; + _remoteVideoTrack = nil; + [_videoCallView.remoteVideoView renderFrame:nil]; + _remoteVideoTrack = remoteVideoTrack; + [_remoteVideoTrack addRenderer:_videoCallView.remoteVideoView]; +} + +- (void)hangup { + self.remoteVideoTrack = nil; + _videoCallView.localVideoView.captureSession = nil; + [_captureController stopCapture]; + _captureController = nil; + [_fileCaptureController stopCapture]; + _fileCaptureController = nil; + [_client disconnect]; + [_delegate viewControllerDidFinish:self]; +} + +- (NSString *)statusTextForState:(RTCIceConnectionState)state { + switch (state) { + case RTCIceConnectionStateNew: + case RTCIceConnectionStateChecking: + return @"Connecting..."; + case RTCIceConnectionStateConnected: + case RTCIceConnectionStateCompleted: + case RTCIceConnectionStateFailed: + case RTCIceConnectionStateDisconnected: + case RTCIceConnectionStateClosed: + case RTCIceConnectionStateCount: + return nil; + } +} + +- (void)showAlertWithMessage:(NSString*)message { + UIAlertController *alert = + [UIAlertController alertControllerWithTitle:nil + message:message + preferredStyle:UIAlertControllerStyleAlert]; + + UIAlertAction *defaultAction = [UIAlertAction actionWithTitle:@"OK" + style:UIAlertActionStyleDefault + handler:^(UIAlertAction *action){ + }]; + + [alert addAction:defaultAction]; + [self presentViewController:alert animated:YES completion:nil]; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/Info.plist b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/Info.plist new file mode 100644 index 0000000000..a2f0a683ed --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/Info.plist @@ -0,0 +1,109 @@ +<?xml version="1.0" encoding="UTF-8"?> +<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd"> +<plist version="1.0"> +<dict> + <key>BuildMachineOSBuild</key> + <string>12E55</string> + <key>CFBundleDevelopmentRegion</key> + <string>en</string> + <key>CFBundleDisplayName</key> + <string>AppRTCMobile</string> + <key>CFBundleExecutable</key> + <string>${EXECUTABLE_NAME}</string> + <key>CFBundleIcons</key> + <dict> + <key>CFBundlePrimaryIcon</key> + <dict> + <key>CFBundleIconFiles</key> + <array> + <string>Icon.png</string> + <string>Icon-120.png</string> + <string>Icon-180.png</string> + </array> + </dict> + </dict> + <key>CFBundleIdentifier</key> + <string>com.google.AppRTCMobile</string> + <key>CFBundleInfoDictionaryVersion</key> + <string>6.0</string> + <key>CFBundleName</key> + <string>${EXECUTABLE_NAME}</string> + <key>CFBundlePackageType</key> + <string>APPL</string> + <key>CFBundleShortVersionString</key> + <string>1.0</string> + <key>CFBundleSignature</key> + <string>????</string> + <key>CFBundleSupportedPlatforms</key> + <array> + <string>iPhoneOS</string> + </array> + <key>CFBundleVersion</key> + <string>1.0</string> + <key>UIStatusBarTintParameters</key> + <dict> + <key>UINavigationBar</key> + <dict> + <key>Style</key> + <string>UIBarStyleDefault</string> + <key>Translucent</key> + <false/> + </dict> + </dict> + <key>UISupportedInterfaceOrientations</key> + <array> + <string>UIInterfaceOrientationPortrait</string> + <string>UIInterfaceOrientationLandscapeLeft</string> + <string>UIInterfaceOrientationLandscapeRight</string> + <string>UIInterfaceOrientationPortraitUpsideDown</string> + </array> + <key>UIAppFonts</key> + <array> + <string>Roboto-Regular.ttf</string> + </array> + <key>UIBackgroundModes</key> + <array> + <string>audio</string> + <string>voip</string> + </array> + <key>NSCameraUsageDescription</key> + <string>Camera access needed for video calling</string> + <key>NSMicrophoneUsageDescription</key> + <string>Microphone access needed for video calling</string> + <key>UIFileSharingEnabled</key> + <true/> + <key>UILaunchImages</key> + <array> + <dict> + <key>UILaunchImageMinimumOSVersion</key> + <string>7.0</string> + <key>UILaunchImageName</key> + <string>iPhone5</string> + <key>UILaunchImageOrientation</key> + <string>Portrait</string> + <key>UILaunchImageSize</key> + <string>{320, 568}</string> + </dict> + <dict> + <key>UILaunchImageMinimumOSVersion</key> + <string>8.0</string> + <key>UILaunchImageName</key> + <string>iPhone6</string> + <key>UILaunchImageOrientation</key> + <string>Portrait</string> + <key>UILaunchImageSize</key> + <string>{375, 667}</string> + </dict> + <dict> + <key>UILaunchImageMinimumOSVersion</key> + <string>8.0</string> + <key>UILaunchImageName</key> + <string>iPhone6p</string> + <key>UILaunchImageOrientation</key> + <string>Portrait</string> + <key>UILaunchImageSize</key> + <string>{414, 736}</string> + </dict> + </array> +</dict> +</plist> diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.h new file mode 100644 index 0000000000..3a93c253b2 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.h @@ -0,0 +1,18 @@ +/* + * Copyright 2017 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "sdk/objc/base/RTCVideoCodecInfo.h" + +@interface RTC_OBJC_TYPE (RTCVideoCodecInfo) +(HumanReadable) + + - (NSString *)humanReadableDescription; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.m new file mode 100644 index 0000000000..5e0c52c5c4 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.m @@ -0,0 +1,37 @@ +/* + * Copyright 2017 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "RTCVideoCodecInfo+HumanReadable.h" + +#import "sdk/objc/components/video_codec/RTCH264ProfileLevelId.h" + +@implementation RTC_OBJC_TYPE (RTCVideoCodecInfo) +(HumanReadable) + + - (NSString *)humanReadableDescription { + if ([self.name isEqualToString:@"H264"]) { + NSString *profileId = self.parameters[@"profile-level-id"]; + RTC_OBJC_TYPE(RTCH264ProfileLevelId) *profileLevelId = + [[RTC_OBJC_TYPE(RTCH264ProfileLevelId) alloc] initWithHexString:profileId]; + if (profileLevelId.profile == RTCH264ProfileConstrainedHigh || + profileLevelId.profile == RTCH264ProfileHigh) { + return @"H264 (High)"; + } else if (profileLevelId.profile == RTCH264ProfileConstrainedBaseline || + profileLevelId.profile == RTCH264ProfileBaseline) { + return @"H264 (Baseline)"; + } else { + return [NSString stringWithFormat:@"H264 (%@)", profileId]; + } + } else { + return self.name; + } +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/UIImage+ARDUtilities.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/UIImage+ARDUtilities.h new file mode 100644 index 0000000000..d56ba02c2e --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/UIImage+ARDUtilities.h @@ -0,0 +1,18 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <UIKit/UIKit.h> + +@interface UIImage (ARDUtilities) + +// Returns an color tinted version for the given image resource. ++ (UIImage *)imageForName:(NSString *)name color:(UIColor *)color; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/UIImage+ARDUtilities.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/UIImage+ARDUtilities.m new file mode 100644 index 0000000000..1bbe8c342f --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/UIImage+ARDUtilities.m @@ -0,0 +1,31 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "UIImage+ARDUtilities.h" + +@implementation UIImage (ARDUtilities) + ++ (UIImage *)imageForName:(NSString *)name color:(UIColor *)color { + UIImage *image = [UIImage imageNamed:name]; + if (!image) { + return nil; + } + UIGraphicsBeginImageContextWithOptions(image.size, NO, 0.0f); + [color setFill]; + CGRect bounds = CGRectMake(0, 0, image.size.width, image.size.height); + UIRectFill(bounds); + [image drawInRect:bounds blendMode:kCGBlendModeDestinationIn alpha:1.0f]; + UIImage *coloredImage = UIGraphicsGetImageFromCurrentImageContext(); + UIGraphicsEndImageContext(); + + return coloredImage; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.h new file mode 100644 index 0000000000..2c4a56368a --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.h @@ -0,0 +1,24 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <ReplayKit/ReplayKit.h> + +#import "sdk/objc/base/RTCLogging.h" + +#import "ARDAppClient.h" + +@protocol ARDExternalSampleDelegate; + +API_AVAILABLE(ios(10.0)) +@interface ARDBroadcastSampleHandler : RPBroadcastSampleHandler <ARDAppClientDelegate> + +@property(nonatomic, strong) id<ARDExternalSampleDelegate> capturer; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.m new file mode 100644 index 0000000000..1c276d965f --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.m @@ -0,0 +1,130 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDBroadcastSampleHandler.h" + +#import <os/log.h> + +#import "ARDExternalSampleCapturer.h" +#import "ARDSettingsModel.h" + +#import "sdk/objc/api/logging/RTCCallbackLogger.h" +#import "sdk/objc/base/RTCLogging.h" + +@implementation ARDBroadcastSampleHandler { + ARDAppClient *_client; + RTC_OBJC_TYPE(RTCCallbackLogger) * _callbackLogger; +} + +@synthesize capturer = _capturer; + +- (instancetype)init { + if (self = [super init]) { + _callbackLogger = [[RTC_OBJC_TYPE(RTCCallbackLogger) alloc] init]; + os_log_t rtc_os_log = os_log_create("com.google.AppRTCMobile", "RTCLog"); + [_callbackLogger start:^(NSString *logMessage) { + os_log(rtc_os_log, "%{public}s", [logMessage cStringUsingEncoding:NSUTF8StringEncoding]); + }]; + } + return self; +} + +- (void)broadcastStartedWithSetupInfo:(NSDictionary<NSString *, NSObject *> *)setupInfo { + // User has requested to start the broadcast. Setup info from the UI extension can be supplied but + // optional. + ARDSettingsModel *settingsModel = [[ARDSettingsModel alloc] init]; + + _client = [[ARDAppClient alloc] initWithDelegate:self]; + _client.broadcast = YES; + + NSString *roomName = nil; + if (setupInfo[@"roomName"]) { + roomName = (NSString *)setupInfo[@"roomName"]; + } else { + u_int32_t randomRoomSuffix = arc4random_uniform(1000); + roomName = [NSString stringWithFormat:@"broadcast_%d", randomRoomSuffix]; + } + [_client connectToRoomWithId:roomName settings:settingsModel isLoopback:NO]; + RTCLog(@"Broadcast started."); +} + +- (void)broadcastPaused { + // User has requested to pause the broadcast. Samples will stop being delivered. +} + +- (void)broadcastResumed { + // User has requested to resume the broadcast. Samples delivery will resume. +} + +- (void)broadcastFinished { + // User has requested to finish the broadcast. + [_client disconnect]; +} + +- (void)processSampleBuffer:(CMSampleBufferRef)sampleBuffer + withType:(RPSampleBufferType)sampleBufferType { + switch (sampleBufferType) { + case RPSampleBufferTypeVideo: + [self.capturer didCaptureSampleBuffer:sampleBuffer]; + break; + case RPSampleBufferTypeAudioApp: + break; + case RPSampleBufferTypeAudioMic: + break; + default: + break; + } +} + +#pragma mark - ARDAppClientDelegate + +- (void)appClient:(ARDAppClient *)client didChangeState:(ARDAppClientState)state { + switch (state) { + case kARDAppClientStateConnected: + RTCLog(@"Client connected."); + break; + case kARDAppClientStateConnecting: + RTCLog("Client connecting."); + break; + case kARDAppClientStateDisconnected: + RTCLog(@"Client disconnected."); + break; + } +} + +- (void)appClient:(ARDAppClient *)client didChangeConnectionState:(RTCIceConnectionState)state { + RTCLog(@"ICE state changed: %ld", (long)state); +} + +- (void)appClient:(ARDAppClient *)client + didCreateLocalCapturer:(RTC_OBJC_TYPE(RTCCameraVideoCapturer) *)localCapturer { +} + +- (void)appClient:(ARDAppClient *)client + didCreateLocalExternalSampleCapturer:(ARDExternalSampleCapturer *)externalSampleCapturer { + self.capturer = externalSampleCapturer; +} + +- (void)appClient:(ARDAppClient *)client + didReceiveLocalVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)localVideoTrack { +} + +- (void)appClient:(ARDAppClient *)client + didReceiveRemoteVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)remoteVideoTrack { +} + +- (void)appClient:(ARDAppClient *)client didGetStats:(RTC_OBJC_TYPE(RTCStatisticsReport) *)stats { +} + +- (void)appClient:(ARDAppClient *)client didError:(NSError *)error { + RTCLog(@"Error: %@", error); +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.h new file mode 100644 index 0000000000..bbf397d8a9 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.h @@ -0,0 +1,17 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <ReplayKit/ReplayKit.h> +#import <UIKit/UIKit.h> + +API_AVAILABLE(ios(11.0)) +@interface ARDBroadcastSetupViewController : UIViewController <UITextFieldDelegate> + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.m new file mode 100644 index 0000000000..55438f17d8 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.m @@ -0,0 +1,107 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "ARDBroadcastSetupViewController.h" + +@implementation ARDBroadcastSetupViewController { + UITextField *_roomNameField; +} + +- (void)loadView { + UIView *view = [[UIView alloc] initWithFrame:CGRectZero]; + view.backgroundColor = [UIColor colorWithWhite:1.0 alpha:0.7]; + + UIImageView *imageView = [[UIImageView alloc] initWithImage:[UIImage imageNamed:@"Icon-180"]]; + imageView.translatesAutoresizingMaskIntoConstraints = NO; + [view addSubview:imageView]; + + _roomNameField = [[UITextField alloc] initWithFrame:CGRectZero]; + _roomNameField.borderStyle = UITextBorderStyleRoundedRect; + _roomNameField.font = [UIFont systemFontOfSize:14.0]; + _roomNameField.translatesAutoresizingMaskIntoConstraints = NO; + _roomNameField.placeholder = @"Room name"; + _roomNameField.returnKeyType = UIReturnKeyDone; + _roomNameField.delegate = self; + [view addSubview:_roomNameField]; + + UIButton *doneButton = [UIButton buttonWithType:UIButtonTypeSystem]; + doneButton.translatesAutoresizingMaskIntoConstraints = NO; + doneButton.titleLabel.font = [UIFont systemFontOfSize:20.0]; + [doneButton setTitle:@"Done" forState:UIControlStateNormal]; + [doneButton addTarget:self + action:@selector(userDidFinishSetup) + forControlEvents:UIControlEventTouchUpInside]; + [view addSubview:doneButton]; + + UIButton *cancelButton = [UIButton buttonWithType:UIButtonTypeSystem]; + cancelButton.translatesAutoresizingMaskIntoConstraints = NO; + cancelButton.titleLabel.font = [UIFont systemFontOfSize:20.0]; + [cancelButton setTitle:@"Cancel" forState:UIControlStateNormal]; + [cancelButton addTarget:self + action:@selector(userDidCancelSetup) + forControlEvents:UIControlEventTouchUpInside]; + [view addSubview:cancelButton]; + + UILayoutGuide *margin = view.layoutMarginsGuide; + [imageView.widthAnchor constraintEqualToConstant:60.0].active = YES; + [imageView.heightAnchor constraintEqualToConstant:60.0].active = YES; + [imageView.topAnchor constraintEqualToAnchor:margin.topAnchor constant:20].active = YES; + [imageView.centerXAnchor constraintEqualToAnchor:view.centerXAnchor].active = YES; + + [_roomNameField.leadingAnchor constraintEqualToAnchor:margin.leadingAnchor].active = YES; + [_roomNameField.topAnchor constraintEqualToAnchor:imageView.bottomAnchor constant:20].active = + YES; + [_roomNameField.trailingAnchor constraintEqualToAnchor:margin.trailingAnchor].active = YES; + + [doneButton.leadingAnchor constraintEqualToAnchor:margin.leadingAnchor].active = YES; + [doneButton.bottomAnchor constraintEqualToAnchor:margin.bottomAnchor constant:-20].active = YES; + + [cancelButton.trailingAnchor constraintEqualToAnchor:margin.trailingAnchor].active = YES; + [cancelButton.bottomAnchor constraintEqualToAnchor:margin.bottomAnchor constant:-20].active = YES; + + UITapGestureRecognizer *tgr = + [[UITapGestureRecognizer alloc] initWithTarget:self action:@selector(didTap:)]; + [view addGestureRecognizer:tgr]; + + self.view = view; +} + +- (IBAction)didTap:(id)sender { + [self.view endEditing:YES]; +} + +- (void)userDidFinishSetup { + // URL of the resource where broadcast can be viewed that will be returned to the application + NSURL *broadcastURL = [NSURL + URLWithString:[NSString stringWithFormat:@"https://appr.tc/r/%@", _roomNameField.text]]; + + // Dictionary with setup information that will be provided to broadcast extension when broadcast + // is started + NSDictionary *setupInfo = @{@"roomName" : _roomNameField.text}; + + // Tell ReplayKit that the extension is finished setting up and can begin broadcasting + [self.extensionContext completeRequestWithBroadcastURL:broadcastURL setupInfo:setupInfo]; +} + +- (void)userDidCancelSetup { + // Tell ReplayKit that the extension was cancelled by the user + [self.extensionContext cancelRequestWithError:[NSError errorWithDomain:@"com.google.AppRTCMobile" + code:-1 + userInfo:nil]]; +} + +#pragma mark - UITextFieldDelegate + +- (BOOL)textFieldShouldReturn:(UITextField *)textField { + [self userDidFinishSetup]; + return YES; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastSetupUIInfo.plist b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastSetupUIInfo.plist new file mode 100644 index 0000000000..a123c111e5 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastSetupUIInfo.plist @@ -0,0 +1,39 @@ +<?xml version="1.0" encoding="UTF-8"?> +<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd"> +<plist version="1.0"> +<dict> + <key>CFBundleDevelopmentRegion</key> + <string>en</string> + <key>CFBundleDisplayName</key> + <string>AppRTCMobile</string> + <key>CFBundleExecutable</key> + <string>$(EXECUTABLE_NAME)</string> + <key>CFBundleIdentifier</key> + <string>com.google.AppRTCMobile.BroadcastSetupUI</string> + <key>CFBundleInfoDictionaryVersion</key> + <string>6.0</string> + <key>CFBundleName</key> + <string>$(PRODUCT_NAME)</string> + <key>CFBundlePackageType</key> + <string>XPC!</string> + <key>CFBundleShortVersionString</key> + <string>1.0</string> + <key>CFBundleVersion</key> + <string>1</string> + <key>NSExtension</key> + <dict> + <key>NSExtensionAttributes</key> + <dict> + <key>NSExtensionActivationRule</key> + <dict> + <key>NSExtensionActivationSupportsReplayKitStreaming</key> + <true/> + </dict> + </dict> + <key>NSExtensionPointIdentifier</key> + <string>com.apple.broadcast-services-setupui</string> + <key>NSExtensionPrincipalClass</key> + <string>ARDBroadcastSetupViewController</string> + </dict> +</dict> +</plist> diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastUploadInfo.plist b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastUploadInfo.plist new file mode 100644 index 0000000000..2bab60ea8f --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastUploadInfo.plist @@ -0,0 +1,33 @@ +<?xml version="1.0" encoding="UTF-8"?> +<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd"> +<plist version="1.0"> +<dict> + <key>CFBundleDevelopmentRegion</key> + <string>en</string> + <key>CFBundleDisplayName</key> + <string>AppRTCMobile</string> + <key>CFBundleExecutable</key> + <string>$(EXECUTABLE_NAME)</string> + <key>CFBundleIdentifier</key> + <string>com.google.AppRTCMobile.BroadcastUpload</string> + <key>CFBundleInfoDictionaryVersion</key> + <string>6.0</string> + <key>CFBundleName</key> + <string>$(PRODUCT_NAME)</string> + <key>CFBundlePackageType</key> + <string>XPC!</string> + <key>CFBundleShortVersionString</key> + <string>1.0</string> + <key>CFBundleVersion</key> + <string>1</string> + <key>NSExtension</key> + <dict> + <key>NSExtensionPointIdentifier</key> + <string>com.apple.broadcast-services-upload</string> + <key>NSExtensionPrincipalClass</key> + <string>ARDBroadcastSampleHandler</string> + <key>RPBroadcastProcessMode</key> + <string>RPBroadcastProcessModeSampleBuffer</string> + </dict> +</dict> +</plist> diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/main.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/main.m new file mode 100644 index 0000000000..00b83f7fd2 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/main.m @@ -0,0 +1,20 @@ +/* + * Copyright 2013 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <UIKit/UIKit.h> + +#import "ARDAppDelegate.h" + +int main(int argc, char* argv[]) { + @autoreleasepool { + return UIApplicationMain( + argc, argv, nil, NSStringFromClass([ARDAppDelegate class])); + } +} diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/Roboto-Regular.ttf b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/Roboto-Regular.ttf Binary files differnew file mode 100644 index 0000000000..0e58508a64 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/Roboto-Regular.ttf diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/foreman.mp4 b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/foreman.mp4 Binary files differnew file mode 100644 index 0000000000..ccffbf4722 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/foreman.mp4 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b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_switch_video_black_24dp@2x.png diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/mozart.mp3 b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/mozart.mp3 Binary files differnew file mode 100644 index 0000000000..5981ba3a91 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/mozart.mp3 diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCAppDelegate.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCAppDelegate.h new file mode 100644 index 0000000000..2b3ce094a2 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCAppDelegate.h @@ -0,0 +1,14 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Cocoa/Cocoa.h> + +@interface APPRTCAppDelegate : NSObject <NSApplicationDelegate> +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCAppDelegate.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCAppDelegate.m new file mode 100644 index 0000000000..36a470021d --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCAppDelegate.m @@ -0,0 +1,55 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "APPRTCAppDelegate.h" +#import "APPRTCViewController.h" +#import "sdk/objc/api/peerconnection/RTCSSLAdapter.h" + +@interface APPRTCAppDelegate () <NSWindowDelegate> +@end + +@implementation APPRTCAppDelegate { + APPRTCViewController* _viewController; + NSWindow* _window; +} + +#pragma mark - NSApplicationDelegate + +- (void)applicationDidFinishLaunching:(NSNotification*)notification { + RTCInitializeSSL(); + NSScreen* screen = [NSScreen mainScreen]; + NSRect visibleRect = [screen visibleFrame]; + NSRect windowRect = NSMakeRect(NSMidX(visibleRect), + NSMidY(visibleRect), + 1320, + 1140); + NSUInteger styleMask = NSWindowStyleMaskTitled | NSWindowStyleMaskClosable; + _window = [[NSWindow alloc] initWithContentRect:windowRect + styleMask:styleMask + backing:NSBackingStoreBuffered + defer:NO]; + _window.delegate = self; + [_window makeKeyAndOrderFront:self]; + [_window makeMainWindow]; + _viewController = [[APPRTCViewController alloc] initWithNibName:nil + bundle:nil]; + [_window setContentView:[_viewController view]]; +} + +#pragma mark - NSWindow + +- (void)windowWillClose:(NSNotification*)notification { + [_viewController windowWillClose:notification]; + RTCCleanupSSL(); + [NSApp terminate:self]; +} + +@end + diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.h new file mode 100644 index 0000000000..306ecd9c7f --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.h @@ -0,0 +1,17 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <AppKit/AppKit.h> + +@interface APPRTCViewController : NSViewController + +- (void)windowWillClose:(NSNotification*)notification; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.m new file mode 100644 index 0000000000..8904187215 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.m @@ -0,0 +1,438 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "APPRTCViewController.h" + +#import <AVFoundation/AVFoundation.h> + +#import "sdk/objc/api/peerconnection/RTCVideoTrack.h" +#import "sdk/objc/components/renderer/metal/RTCMTLNSVideoView.h" +#import "sdk/objc/components/renderer/opengl/RTCNSGLVideoView.h" + +#import "ARDAppClient.h" +#import "ARDCaptureController.h" +#import "ARDSettingsModel.h" + +static NSUInteger const kContentWidth = 900; +static NSUInteger const kRoomFieldWidth = 200; +static NSUInteger const kActionItemHeight = 30; +static NSUInteger const kBottomViewHeight = 200; + +@class APPRTCMainView; +@protocol APPRTCMainViewDelegate + +- (void)appRTCMainView:(APPRTCMainView*)mainView + didEnterRoomId:(NSString*)roomId + loopback:(BOOL)isLoopback; + +@end + +@interface APPRTCMainView : NSView + +@property(nonatomic, weak) id<APPRTCMainViewDelegate> delegate; +@property(nonatomic, readonly) NSView<RTC_OBJC_TYPE(RTCVideoRenderer)>* localVideoView; +@property(nonatomic, readonly) NSView<RTC_OBJC_TYPE(RTCVideoRenderer)>* remoteVideoView; +@property(nonatomic, readonly) NSTextView* logView; + +- (void)displayLogMessage:(NSString*)message; + +@end + +@interface APPRTCMainView () <NSTextFieldDelegate, RTC_OBJC_TYPE (RTCNSGLVideoViewDelegate)> +@end +@implementation APPRTCMainView { + NSScrollView* _scrollView; + NSView* _actionItemsView; + NSButton* _connectButton; + NSButton* _loopbackButton; + NSTextField* _roomField; + CGSize _localVideoSize; + CGSize _remoteVideoSize; +} + +@synthesize delegate = _delegate; +@synthesize localVideoView = _localVideoView; +@synthesize remoteVideoView = _remoteVideoView; +@synthesize logView = _logView; + +- (void)displayLogMessage:(NSString *)message { + dispatch_async(dispatch_get_main_queue(), ^{ + self.logView.string = [NSString stringWithFormat:@"%@%@\n", self.logView.string, message]; + NSRange range = NSMakeRange(self.logView.string.length, 0); + [self.logView scrollRangeToVisible:range]; + }); +} + +#pragma mark - Private + +- (instancetype)initWithFrame:(NSRect)frame { + if (self = [super initWithFrame:frame]) { + [self setupViews]; + } + return self; +} + ++ (BOOL)requiresConstraintBasedLayout { + return YES; +} + +- (void)updateConstraints { + NSParameterAssert( + _roomField != nil && + _scrollView != nil && + _remoteVideoView != nil && + _localVideoView != nil && + _actionItemsView!= nil && + _connectButton != nil && + _loopbackButton != nil); + + [self removeConstraints:[self constraints]]; + NSDictionary* viewsDictionary = + NSDictionaryOfVariableBindings(_roomField, + _scrollView, + _remoteVideoView, + _localVideoView, + _actionItemsView, + _connectButton, + _loopbackButton); + + NSSize remoteViewSize = [self remoteVideoViewSize]; + NSDictionary* metrics = @{ + @"remoteViewWidth" : @(remoteViewSize.width), + @"remoteViewHeight" : @(remoteViewSize.height), + @"kBottomViewHeight" : @(kBottomViewHeight), + @"localViewHeight" : @(remoteViewSize.height / 3), + @"localViewWidth" : @(remoteViewSize.width / 3), + @"kRoomFieldWidth" : @(kRoomFieldWidth), + @"kActionItemHeight" : @(kActionItemHeight) + }; + // Declare this separately to avoid compiler warning about splitting string + // within an NSArray expression. + NSString* verticalConstraintLeft = + @"V:|-[_remoteVideoView(remoteViewHeight)]-[_scrollView(kBottomViewHeight)]-|"; + NSString* verticalConstraintRight = + @"V:|-[_remoteVideoView(remoteViewHeight)]-[_actionItemsView(kBottomViewHeight)]-|"; + NSArray* constraintFormats = @[ + verticalConstraintLeft, + verticalConstraintRight, + @"H:|-[_remoteVideoView(remoteViewWidth)]-|", + @"V:|-[_localVideoView(localViewHeight)]", + @"H:|-[_localVideoView(localViewWidth)]", + @"H:|-[_scrollView(==_actionItemsView)]-[_actionItemsView]-|" + ]; + + NSArray* actionItemsConstraints = @[ + @"H:|-[_roomField(kRoomFieldWidth)]-[_loopbackButton(kRoomFieldWidth)]", + @"H:|-[_connectButton(kRoomFieldWidth)]", + @"V:|-[_roomField(kActionItemHeight)]-[_connectButton(kActionItemHeight)]", + @"V:|-[_loopbackButton(kActionItemHeight)]", + ]; + + [APPRTCMainView addConstraints:constraintFormats + toView:self + viewsDictionary:viewsDictionary + metrics:metrics]; + [APPRTCMainView addConstraints:actionItemsConstraints + toView:_actionItemsView + viewsDictionary:viewsDictionary + metrics:metrics]; + [super updateConstraints]; +} + +#pragma mark - Constraints helper + ++ (void)addConstraints:(NSArray*)constraints toView:(NSView*)view + viewsDictionary:(NSDictionary*)viewsDictionary + metrics:(NSDictionary*)metrics { + for (NSString* constraintFormat in constraints) { + NSArray* constraints = + [NSLayoutConstraint constraintsWithVisualFormat:constraintFormat + options:0 + metrics:metrics + views:viewsDictionary]; + for (NSLayoutConstraint* constraint in constraints) { + [view addConstraint:constraint]; + } + } +} + +#pragma mark - Control actions + +- (void)startCall:(id)sender { + NSString* roomString = _roomField.stringValue; + // Generate room id for loopback options. + if (_loopbackButton.intValue && [roomString isEqualToString:@""]) { + roomString = [NSUUID UUID].UUIDString; + roomString = [roomString stringByReplacingOccurrencesOfString:@"-" withString:@""]; + } + [self.delegate appRTCMainView:self + didEnterRoomId:roomString + loopback:_loopbackButton.intValue]; + [self setNeedsUpdateConstraints:YES]; +} + +#pragma mark - RTC_OBJC_TYPE(RTCNSGLVideoViewDelegate) + +- (void)videoView:(RTC_OBJC_TYPE(RTCNSGLVideoView) *)videoView didChangeVideoSize:(NSSize)size { + if (videoView == _remoteVideoView) { + _remoteVideoSize = size; + } else if (videoView == _localVideoView) { + _localVideoSize = size; + } else { + return; + } + + [self setNeedsUpdateConstraints:YES]; +} + +#pragma mark - Private + +- (void)setupViews { + NSParameterAssert([[self subviews] count] == 0); + + _logView = [[NSTextView alloc] initWithFrame:NSZeroRect]; + [_logView setMinSize:NSMakeSize(0, kBottomViewHeight)]; + [_logView setMaxSize:NSMakeSize(FLT_MAX, FLT_MAX)]; + [_logView setVerticallyResizable:YES]; + [_logView setAutoresizingMask:NSViewWidthSizable]; + NSTextContainer* textContainer = [_logView textContainer]; + NSSize containerSize = NSMakeSize(kContentWidth, FLT_MAX); + [textContainer setContainerSize:containerSize]; + [textContainer setWidthTracksTextView:YES]; + [_logView setEditable:NO]; + + [self setupActionItemsView]; + + _scrollView = [[NSScrollView alloc] initWithFrame:NSZeroRect]; + [_scrollView setTranslatesAutoresizingMaskIntoConstraints:NO]; + [_scrollView setHasVerticalScroller:YES]; + [_scrollView setDocumentView:_logView]; + [self addSubview:_scrollView]; + +// NOTE (daniela): Ignoring Clang diagonstic here. +// We're performing run time check to make sure class is available on runtime. +// If not we're providing sensible default. +#pragma clang diagnostic push +#pragma clang diagnostic ignored "-Wpartial-availability" + if ([RTC_OBJC_TYPE(RTCMTLNSVideoView) class] && + [RTC_OBJC_TYPE(RTCMTLNSVideoView) isMetalAvailable]) { + _remoteVideoView = [[RTC_OBJC_TYPE(RTCMTLNSVideoView) alloc] initWithFrame:NSZeroRect]; + _localVideoView = [[RTC_OBJC_TYPE(RTCMTLNSVideoView) alloc] initWithFrame:NSZeroRect]; + } +#pragma clang diagnostic pop + if (_remoteVideoView == nil) { + NSOpenGLPixelFormatAttribute attributes[] = { + NSOpenGLPFADoubleBuffer, + NSOpenGLPFADepthSize, 24, + NSOpenGLPFAOpenGLProfile, + NSOpenGLProfileVersion3_2Core, + 0 + }; + NSOpenGLPixelFormat* pixelFormat = + [[NSOpenGLPixelFormat alloc] initWithAttributes:attributes]; + + RTC_OBJC_TYPE(RTCNSGLVideoView)* remote = + [[RTC_OBJC_TYPE(RTCNSGLVideoView) alloc] initWithFrame:NSZeroRect pixelFormat:pixelFormat]; + remote.delegate = self; + _remoteVideoView = remote; + + RTC_OBJC_TYPE(RTCNSGLVideoView)* local = + [[RTC_OBJC_TYPE(RTCNSGLVideoView) alloc] initWithFrame:NSZeroRect pixelFormat:pixelFormat]; + local.delegate = self; + _localVideoView = local; + } + + [_remoteVideoView setTranslatesAutoresizingMaskIntoConstraints:NO]; + [self addSubview:_remoteVideoView]; + [_localVideoView setTranslatesAutoresizingMaskIntoConstraints:NO]; + [self addSubview:_localVideoView]; +} + +- (void)setupActionItemsView { + _actionItemsView = [[NSView alloc] initWithFrame:NSZeroRect]; + [_actionItemsView setTranslatesAutoresizingMaskIntoConstraints:NO]; + [self addSubview:_actionItemsView]; + + _roomField = [[NSTextField alloc] initWithFrame:NSZeroRect]; + [_roomField setTranslatesAutoresizingMaskIntoConstraints:NO]; + [[_roomField cell] setPlaceholderString: @"Enter AppRTC room id"]; + [_actionItemsView addSubview:_roomField]; + [_roomField setEditable:YES]; + + _connectButton = [[NSButton alloc] initWithFrame:NSZeroRect]; + [_connectButton setTranslatesAutoresizingMaskIntoConstraints:NO]; + _connectButton.title = @"Start call"; + _connectButton.bezelStyle = NSRoundedBezelStyle; + _connectButton.target = self; + _connectButton.action = @selector(startCall:); + [_actionItemsView addSubview:_connectButton]; + + _loopbackButton = [[NSButton alloc] initWithFrame:NSZeroRect]; + [_loopbackButton setTranslatesAutoresizingMaskIntoConstraints:NO]; + _loopbackButton.title = @"Loopback"; + [_loopbackButton setButtonType:NSSwitchButton]; + [_actionItemsView addSubview:_loopbackButton]; +} + +- (NSSize)remoteVideoViewSize { + if (!_remoteVideoView.bounds.size.width) { + return NSMakeSize(kContentWidth, 0); + } + NSInteger width = MAX(_remoteVideoView.bounds.size.width, kContentWidth); + NSInteger height = (width/16) * 9; + return NSMakeSize(width, height); +} + +@end + +@interface APPRTCViewController () + <ARDAppClientDelegate, APPRTCMainViewDelegate> +@property(nonatomic, readonly) APPRTCMainView* mainView; +@end + +@implementation APPRTCViewController { + ARDAppClient* _client; + RTC_OBJC_TYPE(RTCVideoTrack) * _localVideoTrack; + RTC_OBJC_TYPE(RTCVideoTrack) * _remoteVideoTrack; + ARDCaptureController* _captureController; +} + +- (void)dealloc { + [self disconnect]; +} + +- (void)viewDidAppear { + [super viewDidAppear]; + [self displayUsageInstructions]; +} + +- (void)loadView { + APPRTCMainView* view = [[APPRTCMainView alloc] initWithFrame:NSZeroRect]; + [view setTranslatesAutoresizingMaskIntoConstraints:NO]; + view.delegate = self; + self.view = view; +} + +- (void)windowWillClose:(NSNotification*)notification { + [self disconnect]; +} + +#pragma mark - Usage + +- (void)displayUsageInstructions { + [self.mainView displayLogMessage: + @"To start call:\n" + @"• Enter AppRTC room id (not neccessary for loopback)\n" + @"• Start call"]; +} + +#pragma mark - ARDAppClientDelegate + +- (void)appClient:(ARDAppClient *)client + didChangeState:(ARDAppClientState)state { + switch (state) { + case kARDAppClientStateConnected: + [self.mainView displayLogMessage:@"Client connected."]; + break; + case kARDAppClientStateConnecting: + [self.mainView displayLogMessage:@"Client connecting."]; + break; + case kARDAppClientStateDisconnected: + [self.mainView displayLogMessage:@"Client disconnected."]; + [self resetUI]; + _client = nil; + break; + } +} + +- (void)appClient:(ARDAppClient *)client + didChangeConnectionState:(RTCIceConnectionState)state { +} + +- (void)appClient:(ARDAppClient*)client + didCreateLocalCapturer:(RTC_OBJC_TYPE(RTCCameraVideoCapturer) *)localCapturer { + _captureController = + [[ARDCaptureController alloc] initWithCapturer:localCapturer + settings:[[ARDSettingsModel alloc] init]]; + [_captureController startCapture]; +} + +- (void)appClient:(ARDAppClient*)client + didReceiveLocalVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)localVideoTrack { + _localVideoTrack = localVideoTrack; + [_localVideoTrack addRenderer:self.mainView.localVideoView]; +} + +- (void)appClient:(ARDAppClient*)client + didReceiveRemoteVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)remoteVideoTrack { + _remoteVideoTrack = remoteVideoTrack; + [_remoteVideoTrack addRenderer:self.mainView.remoteVideoView]; +} + +- (void)appClient:(ARDAppClient *)client + didError:(NSError *)error { + [self showAlertWithMessage:[NSString stringWithFormat:@"%@", error]]; + [self disconnect]; +} + +- (void)appClient:(ARDAppClient *)client + didGetStats:(NSArray *)stats { +} + +#pragma mark - APPRTCMainViewDelegate + +- (void)appRTCMainView:(APPRTCMainView*)mainView + didEnterRoomId:(NSString*)roomId + loopback:(BOOL)isLoopback { + + if ([roomId isEqualToString:@""]) { + [self.mainView displayLogMessage:@"Missing room id"]; + return; + } + + [self disconnect]; + ARDAppClient* client = [[ARDAppClient alloc] initWithDelegate:self]; + [client connectToRoomWithId:roomId + settings:[[ARDSettingsModel alloc] init] // Use default settings. + isLoopback:isLoopback]; + _client = client; +} + +#pragma mark - Private + +- (APPRTCMainView*)mainView { + return (APPRTCMainView*)self.view; +} + +- (void)showAlertWithMessage:(NSString*)message { + dispatch_async(dispatch_get_main_queue(), ^{ + NSAlert* alert = [[NSAlert alloc] init]; + [alert setMessageText:message]; + [alert runModal]; + }); +} + +- (void)resetUI { + [_remoteVideoTrack removeRenderer:self.mainView.remoteVideoView]; + [_localVideoTrack removeRenderer:self.mainView.localVideoView]; + _remoteVideoTrack = nil; + _localVideoTrack = nil; + [self.mainView.remoteVideoView renderFrame:nil]; + [self.mainView.localVideoView renderFrame:nil]; +} + +- (void)disconnect { + [self resetUI]; + [_captureController stopCapture]; + _captureController = nil; + [_client disconnect]; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/Info.plist b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/Info.plist new file mode 100644 index 0000000000..d2970eba74 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/Info.plist @@ -0,0 +1,33 @@ +<?xml version="1.0" encoding="UTF-8"?> + +<!DOCTYPE plist PUBLIC "-//Apple/DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd"> +<plist version="1.0"> +<dict> + <key>CFBundleDevelopmentRegion</key> + <string>en</string> + <key>CFBundleDisplayName</key> + <string>${PRODUCT_NAME}</string> + <key>CFBundleExecutable</key> + <string>${EXECUTABLE_NAME}</string> + <key>CFBundleIdentifier</key> + <string>com.Google.${PRODUCT_NAME:rfc1034identifier}</string> + <key>CFBundleInfoDictionaryVersion</key> + <string>6.0</string> + <key>CFBundleName</key> + <string>${PRODUCT_NAME}</string> + <key>CFBundlePackageType</key> + <string>APPL</string> + <key>CFBundleShortVersionString</key> + <string>1.0</string> + <key>CFBundleVersion</key> + <string>1.0</string> + <key>LSMinimumSystemVersion</key> + <string>${MACOSX_DEPLOYMENT_TARGET}</string> + <key>NSPrincipalClass</key> + <string>NSApplication</string> + <key>NSCameraUsageDescription</key> + <string>Camera access needed for video calling</string> + <key>NSMicrophoneUsageDescription</key> + <string>Microphone access needed for video calling</string> +</dict> +</plist>
\ No newline at end of file diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/main.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/main.m new file mode 100644 index 0000000000..79b17f5492 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/main.m @@ -0,0 +1,22 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <AppKit/AppKit.h> + +#import "APPRTCAppDelegate.h" + +int main(int argc, char* argv[]) { + @autoreleasepool { + [NSApplication sharedApplication]; + APPRTCAppDelegate* delegate = [[APPRTCAppDelegate alloc] init]; + [NSApp setDelegate:delegate]; + [NSApp run]; + } +} diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDAppClient_xctest.mm b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDAppClient_xctest.mm new file mode 100644 index 0000000000..2694e49914 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDAppClient_xctest.mm @@ -0,0 +1,266 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> +#import <OCMock/OCMock.h> +#import <QuartzCore/CoreAnimation.h> +#import <XCTest/XCTest.h> + +#include "rtc_base/ssl_adapter.h" + +#import "sdk/objc/api/peerconnection/RTCMediaConstraints.h" +#import "sdk/objc/api/peerconnection/RTCPeerConnectionFactory.h" + +#import "ARDAppClient+Internal.h" +#import "ARDJoinResponse+Internal.h" +#import "ARDMessageResponse+Internal.h" +#import "ARDSettingsModel.h" + +@interface ARDAppClientTest : XCTestCase +@end + +@implementation ARDAppClientTest + +#pragma mark - Mock helpers + +- (id)mockRoomServerClientForRoomId:(NSString *)roomId + clientId:(NSString *)clientId + isInitiator:(BOOL)isInitiator + messages:(NSArray *)messages + messageHandler: + (void (^)(ARDSignalingMessage *))messageHandler { + id mockRoomServerClient = + [OCMockObject mockForProtocol:@protocol(ARDRoomServerClient)]; + + // Successful join response. + ARDJoinResponse *joinResponse = [[ARDJoinResponse alloc] init]; + joinResponse.result = kARDJoinResultTypeSuccess; + joinResponse.roomId = roomId; + joinResponse.clientId = clientId; + joinResponse.isInitiator = isInitiator; + joinResponse.messages = messages; + + // Successful message response. + ARDMessageResponse *messageResponse = [[ARDMessageResponse alloc] init]; + messageResponse.result = kARDMessageResultTypeSuccess; + + // Return join response from above on join. + [[[mockRoomServerClient stub] andDo:^(NSInvocation *invocation) { + __unsafe_unretained void (^completionHandler)(ARDJoinResponse *response, + NSError *error); + [invocation getArgument:&completionHandler atIndex:4]; + completionHandler(joinResponse, nil); + }] joinRoomWithRoomId:roomId isLoopback:NO completionHandler:[OCMArg any]]; + + // Return message response from above on join. + [[[mockRoomServerClient stub] andDo:^(NSInvocation *invocation) { + __unsafe_unretained ARDSignalingMessage *message; + __unsafe_unretained void (^completionHandler)(ARDMessageResponse *response, + NSError *error); + [invocation getArgument:&message atIndex:2]; + [invocation getArgument:&completionHandler atIndex:5]; + messageHandler(message); + completionHandler(messageResponse, nil); + }] sendMessage:[OCMArg any] + forRoomId:roomId + clientId:clientId + completionHandler:[OCMArg any]]; + + // Do nothing on leave. + [[[mockRoomServerClient stub] andDo:^(NSInvocation *invocation) { + __unsafe_unretained void (^completionHandler)(NSError *error); + [invocation getArgument:&completionHandler atIndex:4]; + if (completionHandler) { + completionHandler(nil); + } + }] leaveRoomWithRoomId:roomId + clientId:clientId + completionHandler:[OCMArg any]]; + + return mockRoomServerClient; +} + +- (id)mockSignalingChannelForRoomId:(NSString *)roomId + clientId:(NSString *)clientId + messageHandler: + (void (^)(ARDSignalingMessage *message))messageHandler { + id mockSignalingChannel = + [OCMockObject niceMockForProtocol:@protocol(ARDSignalingChannel)]; + [[mockSignalingChannel stub] registerForRoomId:roomId clientId:clientId]; + [[[mockSignalingChannel stub] andDo:^(NSInvocation *invocation) { + __unsafe_unretained ARDSignalingMessage *message; + [invocation getArgument:&message atIndex:2]; + messageHandler(message); + }] sendMessage:[OCMArg any]]; + return mockSignalingChannel; +} + +- (id)mockTURNClient { + id mockTURNClient = + [OCMockObject mockForProtocol:@protocol(ARDTURNClient)]; + [[[mockTURNClient stub] andDo:^(NSInvocation *invocation) { + // Don't return anything in TURN response. + __unsafe_unretained void (^completionHandler)(NSArray *turnServers, + NSError *error); + [invocation getArgument:&completionHandler atIndex:2]; + completionHandler([NSArray array], nil); + }] requestServersWithCompletionHandler:[OCMArg any]]; + return mockTURNClient; +} + +- (id)mockSettingsModel { + ARDSettingsModel *model = [[ARDSettingsModel alloc] init]; + id partialMock = [OCMockObject partialMockForObject:model]; + [[[partialMock stub] andReturn:@[ @"640x480", @"960x540", @"1280x720" ]] + availableVideoResolutions]; + + return model; +} + +- (ARDAppClient *)createAppClientForRoomId:(NSString *)roomId + clientId:(NSString *)clientId + isInitiator:(BOOL)isInitiator + messages:(NSArray *)messages + messageHandler: + (void (^)(ARDSignalingMessage *message))messageHandler + connectedHandler:(void (^)(void))connectedHandler + localVideoTrackHandler:(void (^)(void))localVideoTrackHandler { + id turnClient = [self mockTURNClient]; + id signalingChannel = [self mockSignalingChannelForRoomId:roomId + clientId:clientId + messageHandler:messageHandler]; + id roomServerClient = + [self mockRoomServerClientForRoomId:roomId + clientId:clientId + isInitiator:isInitiator + messages:messages + messageHandler:messageHandler]; + id delegate = + [OCMockObject niceMockForProtocol:@protocol(ARDAppClientDelegate)]; + [[[delegate stub] andDo:^(NSInvocation *invocation) { + connectedHandler(); + }] appClient:[OCMArg any] + didChangeConnectionState:RTCIceConnectionStateConnected]; + [[[delegate stub] andDo:^(NSInvocation *invocation) { + localVideoTrackHandler(); + }] appClient:[OCMArg any] + didReceiveLocalVideoTrack:[OCMArg any]]; + + return [[ARDAppClient alloc] initWithRoomServerClient:roomServerClient + signalingChannel:signalingChannel + turnClient:turnClient + delegate:delegate]; +} + +#pragma mark - Cases + +// Tests that an ICE connection is established between two ARDAppClient objects +// where one is set up as a caller and the other the answerer. Network +// components are mocked out and messages are relayed directly from object to +// object. It's expected that both clients reach the +// RTCIceConnectionStateConnected state within a reasonable amount of time. +- (void)testSession { + // Need block arguments here because we're setting up a callbacks before we + // create the clients. + ARDAppClient *caller = nil; + ARDAppClient *answerer = nil; + __block __weak ARDAppClient *weakCaller = nil; + __block __weak ARDAppClient *weakAnswerer = nil; + NSString *roomId = @"testRoom"; + NSString *callerId = @"testCallerId"; + NSString *answererId = @"testAnswererId"; + + XCTestExpectation *callerConnectionExpectation = + [self expectationWithDescription:@"Caller PC connected"]; + XCTestExpectation *answererConnectionExpectation = + [self expectationWithDescription:@"Answerer PC connected"]; + + caller = [self createAppClientForRoomId:roomId + clientId:callerId + isInitiator:YES + messages:[NSArray array] + messageHandler:^(ARDSignalingMessage *message) { + ARDAppClient *strongAnswerer = weakAnswerer; + [strongAnswerer channel:strongAnswerer.channel didReceiveMessage:message]; + } connectedHandler:^{ + [callerConnectionExpectation fulfill]; + } localVideoTrackHandler:^{ + }]; + // TODO(tkchin): Figure out why DTLS-SRTP constraint causes thread assertion + // crash in Debug. + caller.defaultPeerConnectionConstraints = + [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:nil + optionalConstraints:nil]; + weakCaller = caller; + + answerer = [self createAppClientForRoomId:roomId + clientId:answererId + isInitiator:NO + messages:[NSArray array] + messageHandler:^(ARDSignalingMessage *message) { + ARDAppClient *strongCaller = weakCaller; + [strongCaller channel:strongCaller.channel didReceiveMessage:message]; + } connectedHandler:^{ + [answererConnectionExpectation fulfill]; + } localVideoTrackHandler:^{ + }]; + // TODO(tkchin): Figure out why DTLS-SRTP constraint causes thread assertion + // crash in Debug. + answerer.defaultPeerConnectionConstraints = + [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:nil + optionalConstraints:nil]; + weakAnswerer = answerer; + + // Kick off connection. + [caller connectToRoomWithId:roomId settings:[self mockSettingsModel] isLoopback:NO]; + [answerer connectToRoomWithId:roomId settings:[self mockSettingsModel] isLoopback:NO]; + [self waitForExpectationsWithTimeout:20 handler:^(NSError *error) { + if (error) { + XCTFail(@"Expectation failed with error %@.", error); + } + }]; +} + +// Test to see that we get a local video connection +// Note this will currently pass even when no camera is connected as a local +// video track is created regardless (Perhaps there should be a test for that...) +#if !TARGET_IPHONE_SIMULATOR // Expect to fail on simulator due to no camera support +- (void)testSessionShouldGetLocalVideoTrackCallback { + ARDAppClient *caller = nil; + NSString *roomId = @"testRoom"; + NSString *callerId = @"testCallerId"; + + XCTestExpectation *localVideoTrackExpectation = + [self expectationWithDescription:@"Caller got local video."]; + + caller = [self createAppClientForRoomId:roomId + clientId:callerId + isInitiator:YES + messages:[NSArray array] + messageHandler:^(ARDSignalingMessage *message) {} + connectedHandler:^{} + localVideoTrackHandler:^{ [localVideoTrackExpectation fulfill]; }]; + caller.defaultPeerConnectionConstraints = + [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:nil + optionalConstraints:nil]; + + // Kick off connection. + [caller connectToRoomWithId:roomId + settings:[self mockSettingsModel] + isLoopback:NO]; + [self waitForExpectationsWithTimeout:20 handler:^(NSError *error) { + if (error) { + XCTFail("Expectation timed out with error: %@.", error); + } + }]; +} +#endif + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDFileCaptureController_xctest.mm b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDFileCaptureController_xctest.mm new file mode 100644 index 0000000000..2e39834190 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDFileCaptureController_xctest.mm @@ -0,0 +1,62 @@ +/* + * Copyright 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> +#import <OCMock/OCMock.h> +#import <XCTest/XCTest.h> + +#import "ARDFileCaptureController.h" + +#import "sdk/objc/components/capturer/RTCFileVideoCapturer.h" + +NS_CLASS_AVAILABLE_IOS(10) +@interface ARDFileCaptureControllerTests : XCTestCase + +@property(nonatomic, strong) ARDFileCaptureController *fileCaptureController; +@property(nonatomic, strong) id fileCapturerMock; + +@end + +@implementation ARDFileCaptureControllerTests + +@synthesize fileCaptureController = _fileCaptureController; +@synthesize fileCapturerMock = _fileCapturerMock; + +- (void)setUp { + [super setUp]; + self.fileCapturerMock = OCMClassMock([RTC_OBJC_TYPE(RTCFileVideoCapturer) class]); + self.fileCaptureController = + [[ARDFileCaptureController alloc] initWithCapturer:self.fileCapturerMock]; +} + +- (void)tearDown { + self.fileCaptureController = nil; + [self.fileCapturerMock stopMocking]; + self.fileCapturerMock = nil; + [super tearDown]; +} + +- (void)testCaptureIsStarted { + [[self.fileCapturerMock expect] startCapturingFromFileNamed:[OCMArg any] onError:[OCMArg any]]; + + [self.fileCaptureController startCapture]; + + [self.fileCapturerMock verify]; +} + +- (void)testCaptureIsStoped { + [[self.fileCapturerMock expect] stopCapture]; + + [self.fileCaptureController stopCapture]; + + [self.fileCapturerMock verify]; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm new file mode 100644 index 0000000000..dc62798963 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm @@ -0,0 +1,96 @@ +/* + * Copyright 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> +#import <OCMock/OCMock.h> +#import <XCTest/XCTest.h> + +#import "sdk/objc/api/peerconnection/RTCMediaConstraints.h" + +#import "ARDSettingsModel+Private.h" +#import "ARDSettingsStore.h" + + +@interface ARDSettingsModelTests : XCTestCase { + ARDSettingsModel *_model; +} +@end + +@implementation ARDSettingsModelTests + +- (id)setupMockStore { + id storeMock = [OCMockObject mockForClass:[ARDSettingsStore class]]; + + id partialMock = [OCMockObject partialMockForObject:_model]; + [[[partialMock stub] andReturn:storeMock] settingsStore]; + [[[partialMock stub] andReturn:@[ @"640x480", @"960x540", @"1280x720" ]] + availableVideoResolutions]; + + return storeMock; +} + +- (void)setUp { + _model = [[ARDSettingsModel alloc] init]; +} + +- (void)testRetrievingSetting { + id storeMock = [self setupMockStore]; + [[[storeMock expect] andReturn:@"640x480"] videoResolution]; + NSString *string = [_model currentVideoResolutionSettingFromStore]; + + XCTAssertEqualObjects(string, @"640x480"); +} + +- (void)testStoringInvalidConstraintReturnsNo { + id storeMock = [self setupMockStore]; + [([[storeMock stub] andReturn:@"960x480"])videoResolution]; + XCTAssertFalse([_model storeVideoResolutionSetting:@"960x480"]); +} + +- (void)testWidthConstraintFromStore { + id storeMock = [self setupMockStore]; + [([[storeMock stub] andReturn:@"1270x480"])videoResolution]; + int width = [_model currentVideoResolutionWidthFromStore]; + + XCTAssertEqual(width, 1270); +} + +- (void)testHeightConstraintFromStore { + id storeMock = [self setupMockStore]; + [([[storeMock stub] andReturn:@"960x540"])videoResolution]; + int height = [_model currentVideoResolutionHeightFromStore]; + + XCTAssertEqual(height, 540); +} + +- (void)testConstraintComponentIsNilWhenInvalidConstraintString { + id storeMock = [self setupMockStore]; + [([[storeMock stub] andReturn:@"invalid"])videoResolution]; + int width = [_model currentVideoResolutionWidthFromStore]; + + XCTAssertEqual(width, 0); +} + +- (void)testStoringAudioSetting { + id storeMock = [self setupMockStore]; + [[storeMock expect] setAudioOnly:YES]; + + [_model storeAudioOnlySetting:YES]; + [storeMock verify]; +} + +- (void)testReturningDefaultCallOption { + id storeMock = [self setupMockStore]; + [[[storeMock stub] andReturnValue:@YES] useManualAudioConfig]; + + XCTAssertTrue([_model currentUseManualAudioConfigSettingFromStore]); +} + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/main.mm b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/main.mm new file mode 100644 index 0000000000..3625ffd7bf --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/main.mm @@ -0,0 +1,21 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <UIKit/UIKit.h> + +#include "test/ios/coverage_util_ios.h" + +int main(int argc, char* argv[]) { + rtc::test::ConfigureCoverageReportPath(); + + @autoreleasepool { + return UIApplicationMain(argc, argv, nil, nil); + } +} diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/LICENSE b/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/LICENSE new file mode 100644 index 0000000000..c01a79c3bd --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/LICENSE @@ -0,0 +1,15 @@ + + Copyright 2012 Square Inc. + + Licensed under the Apache License, Version 2.0 (the "License"); + you may not use this file except in compliance with the License. + You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + + Unless required by applicable law or agreed to in writing, software + distributed under the License is distributed on an "AS IS" BASIS, + WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + See the License for the specific language governing permissions and + limitations under the License. + diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.h new file mode 100644 index 0000000000..a230646073 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.h @@ -0,0 +1,135 @@ +// +// Copyright 2012 Square Inc. +// +// Licensed under the Apache License, Version 2.0 (the "License"); +// you may not use this file except in compliance with the License. +// You may obtain a copy of the License at +// +// http://www.apache.org/licenses/LICENSE-2.0 +// +// Unless required by applicable law or agreed to in writing, software +// distributed under the License is distributed on an "AS IS" BASIS, +// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +// See the License for the specific language governing permissions and +// limitations under the License. +// + +#import <Foundation/Foundation.h> +#import <Security/SecCertificate.h> + +typedef enum { + SR_CONNECTING = 0, + SR_OPEN = 1, + SR_CLOSING = 2, + SR_CLOSED = 3, +} SRReadyState; + +typedef enum SRStatusCode : NSInteger { + SRStatusCodeNormal = 1000, + SRStatusCodeGoingAway = 1001, + SRStatusCodeProtocolError = 1002, + SRStatusCodeUnhandledType = 1003, + // 1004 reserved. + SRStatusNoStatusReceived = 1005, + // 1004-1006 reserved. + SRStatusCodeInvalidUTF8 = 1007, + SRStatusCodePolicyViolated = 1008, + SRStatusCodeMessageTooBig = 1009, +} SRStatusCode; + +@class SRWebSocket; + +extern NSString *const SRWebSocketErrorDomain; +extern NSString *const SRHTTPResponseErrorKey; + +#pragma mark - SRWebSocketDelegate + +@protocol SRWebSocketDelegate; + +#pragma mark - SRWebSocket + +@interface SRWebSocket : NSObject <NSStreamDelegate> + +@property(nonatomic, weak) id<SRWebSocketDelegate> delegate; + +@property(nonatomic, readonly) SRReadyState readyState; +@property(nonatomic, readonly, retain) NSURL *url; + +// This returns the negotiated protocol. +// It will be nil until after the handshake completes. +@property(nonatomic, readonly, copy) NSString *protocol; + +// Protocols should be an array of strings that turn into Sec-WebSocket-Protocol. +- (id)initWithURLRequest:(NSURLRequest *)request protocols:(NSArray *)protocols; +- (id)initWithURLRequest:(NSURLRequest *)request; + +// Some helper constructors. +- (id)initWithURL:(NSURL *)url protocols:(NSArray *)protocols; +- (id)initWithURL:(NSURL *)url; + +// Delegate queue will be dispatch_main_queue by default. +// You cannot set both OperationQueue and dispatch_queue. +- (void)setDelegateOperationQueue:(NSOperationQueue *)queue; +- (void)setDelegateDispatchQueue:(dispatch_queue_t)queue; + +// By default, it will schedule itself on +[NSRunLoop SR_networkRunLoop] using defaultModes. +- (void)scheduleInRunLoop:(NSRunLoop *)aRunLoop forMode:(NSString *)mode; +- (void)unscheduleFromRunLoop:(NSRunLoop *)aRunLoop forMode:(NSString *)mode; + +// SRWebSockets are intended for one-time-use only. Open should be called once and only once. +- (void)open; + +- (void)close; +- (void)closeWithCode:(NSInteger)code reason:(NSString *)reason; + +// Send a UTF8 String or Data. +- (void)send:(id)data; + +// Send Data (can be nil) in a ping message. +- (void)sendPing:(NSData *)data; + +@end + +#pragma mark - SRWebSocketDelegate + +@protocol SRWebSocketDelegate <NSObject> + +// message will either be an NSString if the server is using text +// or NSData if the server is using binary. +- (void)webSocket:(SRWebSocket *)webSocket didReceiveMessage:(id)message; + +@optional + +- (void)webSocketDidOpen:(SRWebSocket *)webSocket; +- (void)webSocket:(SRWebSocket *)webSocket didFailWithError:(NSError *)error; +- (void)webSocket:(SRWebSocket *)webSocket + didCloseWithCode:(NSInteger)code + reason:(NSString *)reason + wasClean:(BOOL)wasClean; +- (void)webSocket:(SRWebSocket *)webSocket didReceivePong:(NSData *)pongPayload; + +@end + +#pragma mark - NSURLRequest (CertificateAdditions) + +@interface NSURLRequest (CertificateAdditions) + +@property(nonatomic, retain, readonly) NSArray *SR_SSLPinnedCertificates; + +@end + +#pragma mark - NSMutableURLRequest (CertificateAdditions) + +@interface NSMutableURLRequest (CertificateAdditions) + +@property(nonatomic, retain) NSArray *SR_SSLPinnedCertificates; + +@end + +#pragma mark - NSRunLoop (SRWebSocket) + +@interface NSRunLoop (SRWebSocket) + ++ (NSRunLoop *)SR_networkRunLoop; + +@end diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m new file mode 100644 index 0000000000..ab0d1b89bc --- /dev/null +++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m @@ -0,0 +1,1774 @@ +// +// Copyright 2012 Square Inc. +// +// Licensed under the Apache License, Version 2.0 (the "License"); +// you may not use this file except in compliance with the License. +// You may obtain a copy of the License at +// +// http://www.apache.org/licenses/LICENSE-2.0 +// +// Unless required by applicable law or agreed to in writing, software +// distributed under the License is distributed on an "AS IS" BASIS, +// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +// See the License for the specific language governing permissions and +// limitations under the License. +// + + +#import "SRWebSocket.h" + +#if TARGET_OS_IPHONE +#define HAS_ICU +#endif + +#ifdef HAS_ICU +#import <unicode/utf8.h> +#endif + +#if TARGET_OS_IPHONE +#import <Endian.h> +#else +#import <CoreServices/CoreServices.h> +#endif + +#import <CommonCrypto/CommonDigest.h> +#import <Security/SecRandom.h> + +#if OS_OBJECT_USE_OBJC_RETAIN_RELEASE +#define sr_dispatch_retain(x) +#define sr_dispatch_release(x) +#define maybe_bridge(x) ((__bridge void *) x) +#else +#define sr_dispatch_retain(x) dispatch_retain(x) +#define sr_dispatch_release(x) dispatch_release(x) +#define maybe_bridge(x) (x) +#endif + +#if !__has_feature(objc_arc) +#error SocketRocket must be compiled with ARC enabled +#endif + + +typedef enum { + SROpCodeTextFrame = 0x1, + SROpCodeBinaryFrame = 0x2, + // 3-7 reserved. + SROpCodeConnectionClose = 0x8, + SROpCodePing = 0x9, + SROpCodePong = 0xA, + // B-F reserved. +} SROpCode; + +typedef struct { + BOOL fin; +// BOOL rsv1; +// BOOL rsv2; +// BOOL rsv3; + uint8_t opcode; + BOOL masked; + uint64_t payload_length; +} frame_header; + +static NSString *const SRWebSocketAppendToSecKeyString = @"258EAFA5-E914-47DA-95CA-C5AB0DC85B11"; + +static inline int32_t validate_dispatch_data_partial_string(NSData *data); +static inline void SRFastLog(NSString *format, ...); + +@interface NSData (SRWebSocket) + +- (NSString *)stringBySHA1ThenBase64Encoding; + +@end + + +@interface NSString (SRWebSocket) + +- (NSString *)stringBySHA1ThenBase64Encoding; + +@end + + +@interface NSURL (SRWebSocket) + +// The origin isn't really applicable for a native application. +// So instead, just map ws -> http and wss -> https. +- (NSString *)SR_origin; + +@end + + +@interface _SRRunLoopThread : NSThread + +@property (nonatomic, readonly) NSRunLoop *runLoop; + +@end + + +static NSString *newSHA1String(const char *bytes, size_t length) { + uint8_t md[CC_SHA1_DIGEST_LENGTH]; + + assert(length >= 0); + assert(length <= UINT32_MAX); + CC_SHA1(bytes, (CC_LONG)length, md); + + NSData *data = [NSData dataWithBytes:md length:CC_SHA1_DIGEST_LENGTH]; + + if ([data respondsToSelector:@selector(base64EncodedStringWithOptions:)]) { + return [data base64EncodedStringWithOptions:0]; + } + + return [data base64Encoding]; +} + +@implementation NSData (SRWebSocket) + +- (NSString *)stringBySHA1ThenBase64Encoding; +{ + return newSHA1String(self.bytes, self.length); +} + +@end + + +@implementation NSString (SRWebSocket) + +- (NSString *)stringBySHA1ThenBase64Encoding; +{ + return newSHA1String(self.UTF8String, self.length); +} + +@end + +NSString *const SRWebSocketErrorDomain = @"SRWebSocketErrorDomain"; +NSString *const SRHTTPResponseErrorKey = @"HTTPResponseStatusCode"; + +// Returns number of bytes consumed. Returning 0 means you didn't match. +// Sends bytes to callback handler; +typedef size_t (^stream_scanner)(NSData *collected_data); + +typedef void (^data_callback)(SRWebSocket *webSocket, NSData *data); + +@interface SRIOConsumer : NSObject { + stream_scanner _scanner; + data_callback _handler; + size_t _bytesNeeded; + BOOL _readToCurrentFrame; + BOOL _unmaskBytes; +} +@property (nonatomic, copy, readonly) stream_scanner consumer; +@property (nonatomic, copy, readonly) data_callback handler; +@property (nonatomic, assign) size_t bytesNeeded; +@property (nonatomic, assign, readonly) BOOL readToCurrentFrame; +@property (nonatomic, assign, readonly) BOOL unmaskBytes; + +@end + +// This class is not thread-safe, and is expected to always be run on the same queue. +@interface SRIOConsumerPool : NSObject + +- (id)initWithBufferCapacity:(NSUInteger)poolSize; + +- (SRIOConsumer *)consumerWithScanner:(stream_scanner)scanner handler:(data_callback)handler bytesNeeded:(size_t)bytesNeeded readToCurrentFrame:(BOOL)readToCurrentFrame unmaskBytes:(BOOL)unmaskBytes; +- (void)returnConsumer:(SRIOConsumer *)consumer; + +@end + +@interface SRWebSocket () <NSStreamDelegate> + +- (void)_writeData:(NSData *)data; +- (void)_closeWithProtocolError:(NSString *)message; +- (void)_failWithError:(NSError *)error; + +- (void)_disconnect; + +- (void)_readFrameNew; +- (void)_readFrameContinue; + +- (void)_pumpScanner; + +- (void)_pumpWriting; + +- (void)_addConsumerWithScanner:(stream_scanner)consumer callback:(data_callback)callback; +- (void)_addConsumerWithDataLength:(size_t)dataLength callback:(data_callback)callback readToCurrentFrame:(BOOL)readToCurrentFrame unmaskBytes:(BOOL)unmaskBytes; +- (void)_addConsumerWithScanner:(stream_scanner)consumer callback:(data_callback)callback dataLength:(size_t)dataLength; +- (void)_readUntilBytes:(const void *)bytes length:(size_t)length callback:(data_callback)dataHandler; +- (void)_readUntilHeaderCompleteWithCallback:(data_callback)dataHandler; + +- (void)_sendFrameWithOpcode:(SROpCode)opcode data:(id)data; + +- (BOOL)_checkHandshake:(CFHTTPMessageRef)httpMessage; +- (void)_SR_commonInit; + +- (void)_initializeStreams; +- (void)_connect; + +@property (nonatomic) SRReadyState readyState; + +@property (nonatomic) NSOperationQueue *delegateOperationQueue; +@property (nonatomic) dispatch_queue_t delegateDispatchQueue; + +@end + + +@implementation SRWebSocket { + NSInteger _webSocketVersion; + + NSOperationQueue *_delegateOperationQueue; + dispatch_queue_t _delegateDispatchQueue; + + dispatch_queue_t _workQueue; + NSMutableArray *_consumers; + + NSInputStream *_inputStream; + NSOutputStream *_outputStream; + + NSMutableData *_readBuffer; + NSUInteger _readBufferOffset; + + NSMutableData *_outputBuffer; + NSUInteger _outputBufferOffset; + + uint8_t _currentFrameOpcode; + size_t _currentFrameCount; + size_t _readOpCount; + uint32_t _currentStringScanPosition; + NSMutableData *_currentFrameData; + + NSString *_closeReason; + + NSString *_secKey; + + BOOL _pinnedCertFound; + + uint8_t _currentReadMaskKey[4]; + size_t _currentReadMaskOffset; + + BOOL _consumerStopped; + + BOOL _closeWhenFinishedWriting; + BOOL _failed; + + BOOL _secure; + NSURLRequest *_urlRequest; + + CFHTTPMessageRef _receivedHTTPHeaders; + + BOOL _sentClose; + BOOL _didFail; + int _closeCode; + + BOOL _isPumping; + + NSMutableSet *_scheduledRunloops; + + // We use this to retain ourselves. + __strong SRWebSocket *_selfRetain; + + NSArray *_requestedProtocols; + SRIOConsumerPool *_consumerPool; +} + +@synthesize delegate = _delegate; +@synthesize url = _url; +@synthesize readyState = _readyState; +@synthesize protocol = _protocol; + +static __strong NSData *CRLFCRLF; + ++ (void)initialize; +{ + CRLFCRLF = [[NSData alloc] initWithBytes:"\r\n\r\n" length:4]; +} + +- (id)initWithURLRequest:(NSURLRequest *)request protocols:(NSArray *)protocols; +{ + self = [super init]; + if (self) { + assert(request.URL); + _url = request.URL; + _urlRequest = request; + + _requestedProtocols = [protocols copy]; + + [self _SR_commonInit]; + } + + return self; +} + +- (id)initWithURLRequest:(NSURLRequest *)request; +{ + return [self initWithURLRequest:request protocols:nil]; +} + +- (id)initWithURL:(NSURL *)url; +{ + return [self initWithURL:url protocols:nil]; +} + +- (id)initWithURL:(NSURL *)url protocols:(NSArray *)protocols; +{ + NSMutableURLRequest *request = [[NSMutableURLRequest alloc] initWithURL:url]; + return [self initWithURLRequest:request protocols:protocols]; +} + +- (void)_SR_commonInit; +{ + + NSString *scheme = _url.scheme.lowercaseString; + assert([scheme isEqualToString:@"ws"] || [scheme isEqualToString:@"http"] || [scheme isEqualToString:@"wss"] || [scheme isEqualToString:@"https"]); + + if ([scheme isEqualToString:@"wss"] || [scheme isEqualToString:@"https"]) { + _secure = YES; + } + + _readyState = SR_CONNECTING; + _consumerStopped = YES; + _webSocketVersion = 13; + + _workQueue = dispatch_queue_create(NULL, DISPATCH_QUEUE_SERIAL); + + // Going to set a specific on the queue so we can validate we're on the work queue + dispatch_queue_set_specific(_workQueue, (__bridge void *)self, maybe_bridge(_workQueue), NULL); + + _delegateDispatchQueue = dispatch_get_main_queue(); + sr_dispatch_retain(_delegateDispatchQueue); + + _readBuffer = [[NSMutableData alloc] init]; + _outputBuffer = [[NSMutableData alloc] init]; + + _currentFrameData = [[NSMutableData alloc] init]; + + _consumers = [[NSMutableArray alloc] init]; + + _consumerPool = [[SRIOConsumerPool alloc] init]; + + _scheduledRunloops = [[NSMutableSet alloc] init]; + + [self _initializeStreams]; + + // default handlers +} + +- (void)assertOnWorkQueue; +{ + assert(dispatch_get_specific((__bridge void *)self) == maybe_bridge(_workQueue)); +} + +- (void)dealloc +{ + _inputStream.delegate = nil; + _outputStream.delegate = nil; + + [_inputStream close]; + [_outputStream close]; + + sr_dispatch_release(_workQueue); + _workQueue = NULL; + + if (_receivedHTTPHeaders) { + CFRelease(_receivedHTTPHeaders); + _receivedHTTPHeaders = NULL; + } + + if (_delegateDispatchQueue) { + sr_dispatch_release(_delegateDispatchQueue); + _delegateDispatchQueue = NULL; + } +} + +#ifndef NDEBUG + +- (void)setReadyState:(SRReadyState)aReadyState; +{ + [self willChangeValueForKey:@"readyState"]; + assert(aReadyState > _readyState); + _readyState = aReadyState; + [self didChangeValueForKey:@"readyState"]; +} + +#endif + +- (void)open; +{ + assert(_url); + NSAssert(_readyState == SR_CONNECTING, @"Cannot call -(void)open on SRWebSocket more than once"); + + _selfRetain = self; + + [self _connect]; +} + +// Calls block on delegate queue +- (void)_performDelegateBlock:(dispatch_block_t)block; +{ + if (_delegateOperationQueue) { + [_delegateOperationQueue addOperationWithBlock:block]; + } else { + assert(_delegateDispatchQueue); + dispatch_async(_delegateDispatchQueue, block); + } +} + +- (void)setDelegateDispatchQueue:(dispatch_queue_t)queue; +{ + if (queue) { + sr_dispatch_retain(queue); + } + + if (_delegateDispatchQueue) { + sr_dispatch_release(_delegateDispatchQueue); + } + + _delegateDispatchQueue = queue; +} + +- (BOOL)_checkHandshake:(CFHTTPMessageRef)httpMessage; +{ + NSString *acceptHeader = CFBridgingRelease(CFHTTPMessageCopyHeaderFieldValue(httpMessage, CFSTR("Sec-WebSocket-Accept"))); + + if (acceptHeader == nil) { + return NO; + } + + NSString *concattedString = [_secKey stringByAppendingString:SRWebSocketAppendToSecKeyString]; + NSString *expectedAccept = [concattedString stringBySHA1ThenBase64Encoding]; + + return [acceptHeader isEqualToString:expectedAccept]; +} + +- (void)_HTTPHeadersDidFinish; +{ + NSInteger responseCode = CFHTTPMessageGetResponseStatusCode(_receivedHTTPHeaders); + + if (responseCode >= 400) { + SRFastLog(@"Request failed with response code %d", responseCode); + [self _failWithError:[NSError errorWithDomain:SRWebSocketErrorDomain code:2132 userInfo:@{NSLocalizedDescriptionKey:[NSString stringWithFormat:@"received bad response code from server %ld", (long)responseCode], SRHTTPResponseErrorKey:@(responseCode)}]]; + return; + } + + if(![self _checkHandshake:_receivedHTTPHeaders]) { + [self _failWithError:[NSError errorWithDomain:SRWebSocketErrorDomain code:2133 userInfo:[NSDictionary dictionaryWithObject:[NSString stringWithFormat:@"Invalid Sec-WebSocket-Accept response"] forKey:NSLocalizedDescriptionKey]]]; + return; + } + + NSString *negotiatedProtocol = CFBridgingRelease(CFHTTPMessageCopyHeaderFieldValue(_receivedHTTPHeaders, CFSTR("Sec-WebSocket-Protocol"))); + if (negotiatedProtocol) { + // Make sure we requested the protocol + if ([_requestedProtocols indexOfObject:negotiatedProtocol] == NSNotFound) { + [self _failWithError:[NSError errorWithDomain:SRWebSocketErrorDomain code:2133 userInfo:[NSDictionary dictionaryWithObject:[NSString stringWithFormat:@"Server specified Sec-WebSocket-Protocol that wasn't requested"] forKey:NSLocalizedDescriptionKey]]]; + return; + } + + _protocol = negotiatedProtocol; + } + + self.readyState = SR_OPEN; + + if (!_didFail) { + [self _readFrameNew]; + } + + [self _performDelegateBlock:^{ + if ([self.delegate respondsToSelector:@selector(webSocketDidOpen:)]) { + [self.delegate webSocketDidOpen:self]; + }; + }]; +} + + +- (void)_readHTTPHeader; +{ + if (_receivedHTTPHeaders == NULL) { + _receivedHTTPHeaders = CFHTTPMessageCreateEmpty(NULL, NO); + } + + [self _readUntilHeaderCompleteWithCallback:^(SRWebSocket *self, NSData *data) { + CFHTTPMessageAppendBytes(self->_receivedHTTPHeaders, (const UInt8 *)data.bytes, data.length); + + if (CFHTTPMessageIsHeaderComplete(self->_receivedHTTPHeaders)) { + SRFastLog(@"Finished reading headers %@", + CFBridgingRelease(CFHTTPMessageCopyAllHeaderFields(self->_receivedHTTPHeaders))); + [self _HTTPHeadersDidFinish]; + } else { + [self _readHTTPHeader]; + } + }]; +} + +- (void)didConnect +{ + SRFastLog(@"Connected"); + CFHTTPMessageRef request = CFHTTPMessageCreateRequest(NULL, CFSTR("GET"), (__bridge CFURLRef)_url, kCFHTTPVersion1_1); + + // Set host first so it defaults + CFHTTPMessageSetHeaderFieldValue(request, CFSTR("Host"), (__bridge CFStringRef)(_url.port ? [NSString stringWithFormat:@"%@:%@", _url.host, _url.port] : _url.host)); + + NSMutableData *keyBytes = [[NSMutableData alloc] initWithLength:16]; + BOOL success = !SecRandomCopyBytes(kSecRandomDefault, keyBytes.length, keyBytes.mutableBytes); + assert(success); + + if ([keyBytes respondsToSelector:@selector(base64EncodedStringWithOptions:)]) { + _secKey = [keyBytes base64EncodedStringWithOptions:0]; + } else { + _secKey = [keyBytes base64Encoding]; + } + + assert([_secKey length] == 24); + + CFHTTPMessageSetHeaderFieldValue(request, CFSTR("Upgrade"), CFSTR("websocket")); + CFHTTPMessageSetHeaderFieldValue(request, CFSTR("Connection"), CFSTR("Upgrade")); + CFHTTPMessageSetHeaderFieldValue(request, CFSTR("Sec-WebSocket-Key"), (__bridge CFStringRef)_secKey); + CFHTTPMessageSetHeaderFieldValue(request, CFSTR("Sec-WebSocket-Version"), (__bridge CFStringRef)[NSString stringWithFormat:@"%ld", (long)_webSocketVersion]); + + CFHTTPMessageSetHeaderFieldValue(request, CFSTR("Origin"), (__bridge CFStringRef)_url.SR_origin); + + if (_requestedProtocols) { + CFHTTPMessageSetHeaderFieldValue(request, CFSTR("Sec-WebSocket-Protocol"), (__bridge CFStringRef)[_requestedProtocols componentsJoinedByString:@", "]); + } + + [_urlRequest.allHTTPHeaderFields enumerateKeysAndObjectsUsingBlock:^(id key, id obj, BOOL *stop) { + CFHTTPMessageSetHeaderFieldValue(request, (__bridge CFStringRef)key, (__bridge CFStringRef)obj); + }]; + + NSData *message = CFBridgingRelease(CFHTTPMessageCopySerializedMessage(request)); + + CFRelease(request); + + [self _writeData:message]; + [self _readHTTPHeader]; +} + +- (void)_initializeStreams; +{ + assert(_url.port.unsignedIntValue <= UINT32_MAX); + uint32_t port = _url.port.unsignedIntValue; + if (port == 0) { + if (!_secure) { + port = 80; + } else { + port = 443; + } + } + NSString *host = _url.host; + + CFReadStreamRef readStream = NULL; + CFWriteStreamRef writeStream = NULL; + + CFStreamCreatePairWithSocketToHost(NULL, (__bridge CFStringRef)host, port, &readStream, &writeStream); + + _outputStream = CFBridgingRelease(writeStream); + _inputStream = CFBridgingRelease(readStream); + + + if (_secure) { + NSMutableDictionary *SSLOptions = [[NSMutableDictionary alloc] init]; + + [_outputStream setProperty:(__bridge id)kCFStreamSocketSecurityLevelNegotiatedSSL forKey:(__bridge id)kCFStreamPropertySocketSecurityLevel]; + + // If we're using pinned certs, don't validate the certificate chain + if ([_urlRequest SR_SSLPinnedCertificates].count) { + [SSLOptions setValue:[NSNumber numberWithBool:NO] forKey:(__bridge id)kCFStreamSSLValidatesCertificateChain]; + } + +#ifdef DEBUG + [SSLOptions setValue:[NSNumber numberWithBool:NO] forKey:(__bridge id)kCFStreamSSLValidatesCertificateChain]; + NSLog(@"SocketRocket: In debug mode. Allowing connection to any root cert"); +#endif + + [_outputStream setProperty:SSLOptions + forKey:(__bridge id)kCFStreamPropertySSLSettings]; + } + + _inputStream.delegate = self; + _outputStream.delegate = self; +} + +- (void)_connect; +{ + if (!_scheduledRunloops.count) { + [self scheduleInRunLoop:[NSRunLoop SR_networkRunLoop] forMode:NSDefaultRunLoopMode]; + } + + + [_outputStream open]; + [_inputStream open]; +} + +- (void)scheduleInRunLoop:(NSRunLoop *)aRunLoop forMode:(NSString *)mode; +{ + [_outputStream scheduleInRunLoop:aRunLoop forMode:mode]; + [_inputStream scheduleInRunLoop:aRunLoop forMode:mode]; + + [_scheduledRunloops addObject:@[aRunLoop, mode]]; +} + +- (void)unscheduleFromRunLoop:(NSRunLoop *)aRunLoop forMode:(NSString *)mode; +{ + [_outputStream removeFromRunLoop:aRunLoop forMode:mode]; + [_inputStream removeFromRunLoop:aRunLoop forMode:mode]; + + [_scheduledRunloops removeObject:@[aRunLoop, mode]]; +} + +- (void)close; +{ + [self closeWithCode:SRStatusCodeNormal reason:nil]; +} + +- (void)closeWithCode:(NSInteger)code reason:(NSString *)reason; +{ + assert(code); + dispatch_async(_workQueue, ^{ + if (self.readyState == SR_CLOSING || self.readyState == SR_CLOSED) { + return; + } + + BOOL wasConnecting = self.readyState == SR_CONNECTING; + + self.readyState = SR_CLOSING; + + SRFastLog(@"Closing with code %d reason %@", code, reason); + + if (wasConnecting) { + [self _disconnect]; + return; + } + + size_t maxMsgSize = [reason maximumLengthOfBytesUsingEncoding:NSUTF8StringEncoding]; + NSMutableData *mutablePayload = [[NSMutableData alloc] initWithLength:sizeof(uint16_t) + maxMsgSize]; + NSData *payload = mutablePayload; + + ((uint16_t *)mutablePayload.mutableBytes)[0] = EndianU16_BtoN(code); + + if (reason) { + NSRange remainingRange = {0}; + + NSUInteger usedLength = 0; + + BOOL success = [reason getBytes:(char *)mutablePayload.mutableBytes + sizeof(uint16_t) maxLength:payload.length - sizeof(uint16_t) usedLength:&usedLength encoding:NSUTF8StringEncoding options:NSStringEncodingConversionExternalRepresentation range:NSMakeRange(0, reason.length) remainingRange:&remainingRange]; + + assert(success); + assert(remainingRange.length == 0); + + if (usedLength != maxMsgSize) { + payload = [payload subdataWithRange:NSMakeRange(0, usedLength + sizeof(uint16_t))]; + } + } + + + [self _sendFrameWithOpcode:SROpCodeConnectionClose data:payload]; + }); +} + +- (void)_closeWithProtocolError:(NSString *)message; +{ + // Need to shunt this on the _callbackQueue first to see if they received any messages + [self _performDelegateBlock:^{ + [self closeWithCode:SRStatusCodeProtocolError reason:message]; + dispatch_async(self->_workQueue, ^{ + [self _disconnect]; + }); + }]; +} + +- (void)_failWithError:(NSError *)error; +{ + dispatch_async(_workQueue, ^{ + if (self.readyState != SR_CLOSED) { + self->_failed = YES; + [self _performDelegateBlock:^{ + if ([self.delegate respondsToSelector:@selector(webSocket:didFailWithError:)]) { + [self.delegate webSocket:self didFailWithError:error]; + } + }]; + + self.readyState = SR_CLOSED; + self->_selfRetain = nil; + + SRFastLog(@"Failing with error %@", error.localizedDescription); + + [self _disconnect]; + } + }); +} + +- (void)_writeData:(NSData *)data; +{ + [self assertOnWorkQueue]; + + if (_closeWhenFinishedWriting) { + return; + } + [_outputBuffer appendData:data]; + [self _pumpWriting]; +} + +- (void)send:(id)data; +{ + NSAssert(self.readyState != SR_CONNECTING, @"Invalid State: Cannot call send: until connection is open"); + // TODO: maybe not copy this for performance + data = [data copy]; + dispatch_async(_workQueue, ^{ + if ([data isKindOfClass:[NSString class]]) { + [self _sendFrameWithOpcode:SROpCodeTextFrame data:[(NSString *)data dataUsingEncoding:NSUTF8StringEncoding]]; + } else if ([data isKindOfClass:[NSData class]]) { + [self _sendFrameWithOpcode:SROpCodeBinaryFrame data:data]; + } else if (data == nil) { + [self _sendFrameWithOpcode:SROpCodeTextFrame data:data]; + } else { + assert(NO); + } + }); +} + +- (void)sendPing:(NSData *)data; +{ + NSAssert(self.readyState == SR_OPEN, @"Invalid State: Cannot call send: until connection is open"); + // TODO: maybe not copy this for performance + data = [data copy] ?: [NSData data]; // It's okay for a ping to be empty + dispatch_async(_workQueue, ^{ + [self _sendFrameWithOpcode:SROpCodePing data:data]; + }); +} + +- (void)handlePing:(NSData *)pingData; +{ + // Need to pingpong this off _callbackQueue first to make sure messages happen in order + [self _performDelegateBlock:^{ + dispatch_async(self->_workQueue, ^{ + [self _sendFrameWithOpcode:SROpCodePong data:pingData]; + }); + }]; +} + +- (void)handlePong:(NSData *)pongData; +{ + SRFastLog(@"Received pong"); + [self _performDelegateBlock:^{ + if ([self.delegate respondsToSelector:@selector(webSocket:didReceivePong:)]) { + [self.delegate webSocket:self didReceivePong:pongData]; + } + }]; +} + +- (void)_handleMessage:(id)message +{ + SRFastLog(@"Received message"); + [self _performDelegateBlock:^{ + [self.delegate webSocket:self didReceiveMessage:message]; + }]; +} + + +static inline BOOL closeCodeIsValid(int closeCode) { + if (closeCode < 1000) { + return NO; + } + + if (closeCode >= 1000 && closeCode <= 1011) { + if (closeCode == 1004 || + closeCode == 1005 || + closeCode == 1006) { + return NO; + } + return YES; + } + + if (closeCode >= 3000 && closeCode <= 3999) { + return YES; + } + + if (closeCode >= 4000 && closeCode <= 4999) { + return YES; + } + + return NO; +} + +// Note from RFC: +// +// If there is a body, the first two +// bytes of the body MUST be a 2-byte unsigned integer (in network byte +// order) representing a status code with value /code/ defined in +// Section 7.4. Following the 2-byte integer the body MAY contain UTF-8 +// encoded data with value /reason/, the interpretation of which is not +// defined by this specification. + +- (void)handleCloseWithData:(NSData *)data; +{ + size_t dataSize = data.length; + __block uint16_t closeCode = 0; + + SRFastLog(@"Received close frame"); + + if (dataSize == 1) { + // TODO handle error + [self _closeWithProtocolError:@"Payload for close must be larger than 2 bytes"]; + return; + } else if (dataSize >= 2) { + [data getBytes:&closeCode length:sizeof(closeCode)]; + _closeCode = EndianU16_BtoN(closeCode); + if (!closeCodeIsValid(_closeCode)) { + [self _closeWithProtocolError:[NSString stringWithFormat:@"Cannot have close code of %d", _closeCode]]; + return; + } + if (dataSize > 2) { + _closeReason = [[NSString alloc] initWithData:[data subdataWithRange:NSMakeRange(2, dataSize - 2)] encoding:NSUTF8StringEncoding]; + if (!_closeReason) { + [self _closeWithProtocolError:@"Close reason MUST be valid UTF-8"]; + return; + } + } + } else { + _closeCode = SRStatusNoStatusReceived; + } + + [self assertOnWorkQueue]; + + if (self.readyState == SR_OPEN) { + [self closeWithCode:1000 reason:nil]; + } + dispatch_async(_workQueue, ^{ + [self _disconnect]; + }); +} + +- (void)_disconnect; +{ + [self assertOnWorkQueue]; + SRFastLog(@"Trying to disconnect"); + _closeWhenFinishedWriting = YES; + [self _pumpWriting]; +} + +- (void)_handleFrameWithData:(NSData *)frameData opCode:(NSInteger)opcode; +{ + // Check that the current data is valid UTF8 + + BOOL isControlFrame = (opcode == SROpCodePing || opcode == SROpCodePong || opcode == SROpCodeConnectionClose); + if (!isControlFrame) { + [self _readFrameNew]; + } else { + dispatch_async(_workQueue, ^{ + [self _readFrameContinue]; + }); + } + + switch (opcode) { + case SROpCodeTextFrame: { + NSString *str = [[NSString alloc] initWithData:frameData encoding:NSUTF8StringEncoding]; + if (str == nil && frameData) { + [self closeWithCode:SRStatusCodeInvalidUTF8 reason:@"Text frames must be valid UTF-8"]; + dispatch_async(_workQueue, ^{ + [self _disconnect]; + }); + + return; + } + [self _handleMessage:str]; + break; + } + case SROpCodeBinaryFrame: + [self _handleMessage:[frameData copy]]; + break; + case SROpCodeConnectionClose: + [self handleCloseWithData:frameData]; + break; + case SROpCodePing: + [self handlePing:frameData]; + break; + case SROpCodePong: + [self handlePong:frameData]; + break; + default: + [self _closeWithProtocolError:[NSString stringWithFormat:@"Unknown opcode %ld", (long)opcode]]; + // TODO: Handle invalid opcode + break; + } +} + +- (void)_handleFrameHeader:(frame_header)frame_header curData:(NSData *)curData; +{ + assert(frame_header.opcode != 0); + + if (self.readyState != SR_OPEN) { + return; + } + + + BOOL isControlFrame = (frame_header.opcode == SROpCodePing || frame_header.opcode == SROpCodePong || frame_header.opcode == SROpCodeConnectionClose); + + if (isControlFrame && !frame_header.fin) { + [self _closeWithProtocolError:@"Fragmented control frames not allowed"]; + return; + } + + if (isControlFrame && frame_header.payload_length >= 126) { + [self _closeWithProtocolError:@"Control frames cannot have payloads larger than 126 bytes"]; + return; + } + + if (!isControlFrame) { + _currentFrameOpcode = frame_header.opcode; + _currentFrameCount += 1; + } + + if (frame_header.payload_length == 0) { + if (isControlFrame) { + [self _handleFrameWithData:curData opCode:frame_header.opcode]; + } else { + if (frame_header.fin) { + [self _handleFrameWithData:_currentFrameData opCode:frame_header.opcode]; + } else { + // TODO add assert that opcode is not a control; + [self _readFrameContinue]; + } + } + } else { + assert(frame_header.payload_length <= SIZE_T_MAX); + [self _addConsumerWithDataLength:(size_t)frame_header.payload_length callback:^(SRWebSocket *self, NSData *newData) { + if (isControlFrame) { + [self _handleFrameWithData:newData opCode:frame_header.opcode]; + } else { + if (frame_header.fin) { + [self _handleFrameWithData:self->_currentFrameData opCode:frame_header.opcode]; + } else { + // TODO add assert that opcode is not a control; + [self _readFrameContinue]; + } + + } + } readToCurrentFrame:!isControlFrame unmaskBytes:frame_header.masked]; + } +} + +/* From RFC: + + 0 1 2 3 + 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + +-+-+-+-+-------+-+-------------+-------------------------------+ + |F|R|R|R| opcode|M| Payload len | Extended payload length | + |I|S|S|S| (4) |A| (7) | (16/64) | + |N|V|V|V| |S| | (if payload len==126/127) | + | |1|2|3| |K| | | + +-+-+-+-+-------+-+-------------+ - - - - - - - - - - - - - - - + + | Extended payload length continued, if payload len == 127 | + + - - - - - - - - - - - - - - - +-------------------------------+ + | |Masking-key, if MASK set to 1 | + +-------------------------------+-------------------------------+ + | Masking-key (continued) | Payload Data | + +-------------------------------- - - - - - - - - - - - - - - - + + : Payload Data continued ... : + + - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - + + | Payload Data continued ... | + +---------------------------------------------------------------+ + */ + +static const uint8_t SRFinMask = 0x80; +static const uint8_t SROpCodeMask = 0x0F; +static const uint8_t SRRsvMask = 0x70; +static const uint8_t SRMaskMask = 0x80; +static const uint8_t SRPayloadLenMask = 0x7F; + + +- (void)_readFrameContinue; +{ + assert((_currentFrameCount == 0 && _currentFrameOpcode == 0) || (_currentFrameCount > 0 && _currentFrameOpcode > 0)); + + [self _addConsumerWithDataLength:2 callback:^(SRWebSocket *self, NSData *data) { + __block frame_header header = {0}; + + const uint8_t *headerBuffer = data.bytes; + assert(data.length >= 2); + + if (headerBuffer[0] & SRRsvMask) { + [self _closeWithProtocolError:@"Server used RSV bits"]; + return; + } + + uint8_t receivedOpcode = (SROpCodeMask & headerBuffer[0]); + + BOOL isControlFrame = (receivedOpcode == SROpCodePing || receivedOpcode == SROpCodePong || receivedOpcode == SROpCodeConnectionClose); + + if (!isControlFrame && receivedOpcode != 0 && self->_currentFrameCount > 0) { + [self _closeWithProtocolError:@"all data frames after the initial data frame must have opcode 0"]; + return; + } + + if (receivedOpcode == 0 && self->_currentFrameCount == 0) { + [self _closeWithProtocolError:@"cannot continue a message"]; + return; + } + + header.opcode = receivedOpcode == 0 ? self->_currentFrameOpcode : receivedOpcode; + + header.fin = !!(SRFinMask & headerBuffer[0]); + + + header.masked = !!(SRMaskMask & headerBuffer[1]); + header.payload_length = SRPayloadLenMask & headerBuffer[1]; + + headerBuffer = NULL; + + if (header.masked) { + [self _closeWithProtocolError:@"Client must receive unmasked data"]; + } + + size_t extra_bytes_needed = header.masked ? sizeof(self->_currentReadMaskKey) : 0; + + if (header.payload_length == 126) { + extra_bytes_needed += sizeof(uint16_t); + } else if (header.payload_length == 127) { + extra_bytes_needed += sizeof(uint64_t); + } + + if (extra_bytes_needed == 0) { + [self _handleFrameHeader:header curData:self->_currentFrameData]; + } else { + [self _addConsumerWithDataLength:extra_bytes_needed callback:^(SRWebSocket *self, NSData *data) { + size_t mapped_size = data.length; + const void *mapped_buffer = data.bytes; + size_t offset = 0; + + if (header.payload_length == 126) { + assert(mapped_size >= sizeof(uint16_t)); + uint16_t newLen = EndianU16_BtoN(*(uint16_t *)(mapped_buffer)); + header.payload_length = newLen; + offset += sizeof(uint16_t); + } else if (header.payload_length == 127) { + assert(mapped_size >= sizeof(uint64_t)); + header.payload_length = EndianU64_BtoN(*(uint64_t *)(mapped_buffer)); + offset += sizeof(uint64_t); + } else { + assert(header.payload_length < 126 && header.payload_length >= 0); + } + + + if (header.masked) { + assert(mapped_size >= sizeof(self->_currentReadMaskOffset) + offset); + memcpy(self->_currentReadMaskKey, + ((uint8_t *)mapped_buffer) + offset, + sizeof(self->_currentReadMaskKey)); + } + + [self _handleFrameHeader:header curData:self->_currentFrameData]; + } readToCurrentFrame:NO unmaskBytes:NO]; + } + } readToCurrentFrame:NO unmaskBytes:NO]; +} + +- (void)_readFrameNew; +{ + dispatch_async(_workQueue, ^{ + [self->_currentFrameData setLength:0]; + + self->_currentFrameOpcode = 0; + self->_currentFrameCount = 0; + self->_readOpCount = 0; + self->_currentStringScanPosition = 0; + + [self _readFrameContinue]; + }); +} + +- (void)_pumpWriting; +{ + [self assertOnWorkQueue]; + + NSUInteger dataLength = _outputBuffer.length; + if (dataLength - _outputBufferOffset > 0 && _outputStream.hasSpaceAvailable) { + NSInteger bytesWritten = [_outputStream write:_outputBuffer.bytes + _outputBufferOffset maxLength:dataLength - _outputBufferOffset]; + if (bytesWritten == -1) { + [self _failWithError:[NSError errorWithDomain:SRWebSocketErrorDomain code:2145 userInfo:[NSDictionary dictionaryWithObject:@"Error writing to stream" forKey:NSLocalizedDescriptionKey]]]; + return; + } + + _outputBufferOffset += bytesWritten; + + if (_outputBufferOffset > 4096 && _outputBufferOffset > (_outputBuffer.length >> 1)) { + _outputBuffer = [[NSMutableData alloc] initWithBytes:(char *)_outputBuffer.bytes + _outputBufferOffset length:_outputBuffer.length - _outputBufferOffset]; + _outputBufferOffset = 0; + } + } + + if (_closeWhenFinishedWriting && + _outputBuffer.length - _outputBufferOffset == 0 && + (_inputStream.streamStatus != NSStreamStatusNotOpen && + _inputStream.streamStatus != NSStreamStatusClosed) && + !_sentClose) { + _sentClose = YES; + + [_outputStream close]; + [_inputStream close]; + + + for (NSArray *runLoop in [_scheduledRunloops copy]) { + [self unscheduleFromRunLoop:[runLoop objectAtIndex:0] forMode:[runLoop objectAtIndex:1]]; + } + + if (!_failed) { + [self _performDelegateBlock:^{ + if ([self.delegate respondsToSelector:@selector(webSocket:didCloseWithCode:reason:wasClean:)]) { + [self.delegate webSocket:self + didCloseWithCode:self->_closeCode + reason:self->_closeReason + wasClean:YES]; + } + }]; + } + + _selfRetain = nil; + } +} + +- (void)_addConsumerWithScanner:(stream_scanner)consumer callback:(data_callback)callback; +{ + [self assertOnWorkQueue]; + [self _addConsumerWithScanner:consumer callback:callback dataLength:0]; +} + +- (void)_addConsumerWithDataLength:(size_t)dataLength callback:(data_callback)callback readToCurrentFrame:(BOOL)readToCurrentFrame unmaskBytes:(BOOL)unmaskBytes; +{ + [self assertOnWorkQueue]; + assert(dataLength); + + [_consumers addObject:[_consumerPool consumerWithScanner:nil handler:callback bytesNeeded:dataLength readToCurrentFrame:readToCurrentFrame unmaskBytes:unmaskBytes]]; + [self _pumpScanner]; +} + +- (void)_addConsumerWithScanner:(stream_scanner)consumer callback:(data_callback)callback dataLength:(size_t)dataLength; +{ + [self assertOnWorkQueue]; + [_consumers addObject:[_consumerPool consumerWithScanner:consumer handler:callback bytesNeeded:dataLength readToCurrentFrame:NO unmaskBytes:NO]]; + [self _pumpScanner]; +} + + +static const char CRLFCRLFBytes[] = {'\r', '\n', '\r', '\n'}; + +- (void)_readUntilHeaderCompleteWithCallback:(data_callback)dataHandler; +{ + [self _readUntilBytes:CRLFCRLFBytes length:sizeof(CRLFCRLFBytes) callback:dataHandler]; +} + +- (void)_readUntilBytes:(const void *)bytes length:(size_t)length callback:(data_callback)dataHandler; +{ + // TODO optimize so this can continue from where we last searched + stream_scanner consumer = ^size_t(NSData *data) { + __block size_t found_size = 0; + __block size_t match_count = 0; + + size_t size = data.length; + const unsigned char *buffer = data.bytes; + for (size_t i = 0; i < size; i++ ) { + if (((const unsigned char *)buffer)[i] == ((const unsigned char *)bytes)[match_count]) { + match_count += 1; + if (match_count == length) { + found_size = i + 1; + break; + } + } else { + match_count = 0; + } + } + return found_size; + }; + [self _addConsumerWithScanner:consumer callback:dataHandler]; +} + + +// Returns true if did work +- (BOOL)_innerPumpScanner { + + BOOL didWork = NO; + + if (self.readyState >= SR_CLOSING) { + return didWork; + } + + if (!_consumers.count) { + return didWork; + } + + size_t curSize = _readBuffer.length - _readBufferOffset; + if (!curSize) { + return didWork; + } + + SRIOConsumer *consumer = [_consumers objectAtIndex:0]; + + size_t bytesNeeded = consumer.bytesNeeded; + + size_t foundSize = 0; + if (consumer.consumer) { + NSData *tempView = [NSData dataWithBytesNoCopy:(char *)_readBuffer.bytes + _readBufferOffset length:_readBuffer.length - _readBufferOffset freeWhenDone:NO]; + foundSize = consumer.consumer(tempView); + } else { + assert(consumer.bytesNeeded); + if (curSize >= bytesNeeded) { + foundSize = bytesNeeded; + } else if (consumer.readToCurrentFrame) { + foundSize = curSize; + } + } + + NSData *slice = nil; + if (consumer.readToCurrentFrame || foundSize) { + NSRange sliceRange = NSMakeRange(_readBufferOffset, foundSize); + slice = [_readBuffer subdataWithRange:sliceRange]; + + _readBufferOffset += foundSize; + + if (_readBufferOffset > 4096 && _readBufferOffset > (_readBuffer.length >> 1)) { + _readBuffer = [[NSMutableData alloc] initWithBytes:(char *)_readBuffer.bytes + _readBufferOffset length:_readBuffer.length - _readBufferOffset]; _readBufferOffset = 0; + } + + if (consumer.unmaskBytes) { + NSMutableData *mutableSlice = [slice mutableCopy]; + + NSUInteger len = mutableSlice.length; + uint8_t *bytes = mutableSlice.mutableBytes; + + for (NSUInteger i = 0; i < len; i++) { + bytes[i] = bytes[i] ^ _currentReadMaskKey[_currentReadMaskOffset % sizeof(_currentReadMaskKey)]; + _currentReadMaskOffset += 1; + } + + slice = mutableSlice; + } + + if (consumer.readToCurrentFrame) { + [_currentFrameData appendData:slice]; + + _readOpCount += 1; + + if (_currentFrameOpcode == SROpCodeTextFrame) { + // Validate UTF8 stuff. + size_t currentDataSize = _currentFrameData.length; + if (_currentFrameOpcode == SROpCodeTextFrame && currentDataSize > 0) { + // TODO: Optimize the crap out of this. Don't really have to copy all the data each time + + size_t scanSize = currentDataSize - _currentStringScanPosition; + + NSData *scan_data = [_currentFrameData subdataWithRange:NSMakeRange(_currentStringScanPosition, scanSize)]; + int32_t valid_utf8_size = validate_dispatch_data_partial_string(scan_data); + + if (valid_utf8_size == -1) { + [self closeWithCode:SRStatusCodeInvalidUTF8 reason:@"Text frames must be valid UTF-8"]; + dispatch_async(_workQueue, ^{ + [self _disconnect]; + }); + return didWork; + } else { + _currentStringScanPosition += valid_utf8_size; + } + } + + } + + consumer.bytesNeeded -= foundSize; + + if (consumer.bytesNeeded == 0) { + [_consumers removeObjectAtIndex:0]; + consumer.handler(self, nil); + [_consumerPool returnConsumer:consumer]; + didWork = YES; + } + } else if (foundSize) { + [_consumers removeObjectAtIndex:0]; + consumer.handler(self, slice); + [_consumerPool returnConsumer:consumer]; + didWork = YES; + } + } + return didWork; +} + +-(void)_pumpScanner; +{ + [self assertOnWorkQueue]; + + if (!_isPumping) { + _isPumping = YES; + } else { + return; + } + + while ([self _innerPumpScanner]) { + + } + + _isPumping = NO; +} + +//#define NOMASK + +static const size_t SRFrameHeaderOverhead = 32; + +- (void)_sendFrameWithOpcode:(SROpCode)opcode data:(id)data; +{ + [self assertOnWorkQueue]; + + if (nil == data) { + return; + } + + NSAssert([data isKindOfClass:[NSData class]] || [data isKindOfClass:[NSString class]], @"NSString or NSData"); + + size_t payloadLength = [data isKindOfClass:[NSString class]] ? [(NSString *)data lengthOfBytesUsingEncoding:NSUTF8StringEncoding] : [data length]; + + NSMutableData *frame = [[NSMutableData alloc] initWithLength:payloadLength + SRFrameHeaderOverhead]; + if (!frame) { + [self closeWithCode:SRStatusCodeMessageTooBig reason:@"Message too big"]; + return; + } + uint8_t *frame_buffer = (uint8_t *)[frame mutableBytes]; + + // set fin + frame_buffer[0] = SRFinMask | opcode; + + BOOL useMask = YES; +#ifdef NOMASK + useMask = NO; +#endif + + if (useMask) { + // set the mask and header + frame_buffer[1] |= SRMaskMask; + } + + size_t frame_buffer_size = 2; + + const uint8_t *unmasked_payload = NULL; + if ([data isKindOfClass:[NSData class]]) { + unmasked_payload = (uint8_t *)[data bytes]; + } else if ([data isKindOfClass:[NSString class]]) { + unmasked_payload = (const uint8_t *)[data UTF8String]; + } else { + return; + } + + if (payloadLength < 126) { + frame_buffer[1] |= payloadLength; + } else if (payloadLength <= UINT16_MAX) { + frame_buffer[1] |= 126; + *((uint16_t *)(frame_buffer + frame_buffer_size)) = EndianU16_BtoN((uint16_t)payloadLength); + frame_buffer_size += sizeof(uint16_t); + } else { + frame_buffer[1] |= 127; + *((uint64_t *)(frame_buffer + frame_buffer_size)) = EndianU64_BtoN((uint64_t)payloadLength); + frame_buffer_size += sizeof(uint64_t); + } + + if (!useMask) { + for (size_t i = 0; i < payloadLength; i++) { + frame_buffer[frame_buffer_size] = unmasked_payload[i]; + frame_buffer_size += 1; + } + } else { + uint8_t *mask_key = frame_buffer + frame_buffer_size; + BOOL success = !SecRandomCopyBytes(kSecRandomDefault, sizeof(uint32_t), (uint8_t *)mask_key); + assert(success); + frame_buffer_size += sizeof(uint32_t); + + // TODO: could probably optimize this with SIMD + for (size_t i = 0; i < payloadLength; i++) { + frame_buffer[frame_buffer_size] = unmasked_payload[i] ^ mask_key[i % sizeof(uint32_t)]; + frame_buffer_size += 1; + } + } + + assert(frame_buffer_size <= [frame length]); + frame.length = frame_buffer_size; + + [self _writeData:frame]; +} + +- (void)stream:(NSStream *)aStream handleEvent:(NSStreamEvent)eventCode; +{ + if (_secure && !_pinnedCertFound && (eventCode == NSStreamEventHasBytesAvailable || eventCode == NSStreamEventHasSpaceAvailable)) { + + NSArray *sslCerts = [_urlRequest SR_SSLPinnedCertificates]; + if (sslCerts) { + SecTrustRef secTrust = (__bridge SecTrustRef)[aStream propertyForKey:(__bridge id)kCFStreamPropertySSLPeerTrust]; + if (secTrust) { + NSInteger numCerts = SecTrustGetCertificateCount(secTrust); + for (NSInteger i = 0; i < numCerts && !_pinnedCertFound; i++) { + SecCertificateRef cert = SecTrustGetCertificateAtIndex(secTrust, i); + NSData *certData = CFBridgingRelease(SecCertificateCopyData(cert)); + + for (id ref in sslCerts) { + SecCertificateRef trustedCert = (__bridge SecCertificateRef)ref; + NSData *trustedCertData = CFBridgingRelease(SecCertificateCopyData(trustedCert)); + + if ([trustedCertData isEqualToData:certData]) { + _pinnedCertFound = YES; + break; + } + } + } + } + + if (!_pinnedCertFound) { + dispatch_async(_workQueue, ^{ + [self _failWithError:[NSError errorWithDomain:SRWebSocketErrorDomain code:23556 userInfo:[NSDictionary dictionaryWithObject:[NSString stringWithFormat:@"Invalid server cert"] forKey:NSLocalizedDescriptionKey]]]; + }); + return; + } + } + } + + dispatch_async(_workQueue, ^{ + switch (eventCode) { + case NSStreamEventOpenCompleted: { + SRFastLog(@"NSStreamEventOpenCompleted %@", aStream); + if (self.readyState >= SR_CLOSING) { + return; + } + assert(self->_readBuffer); + + if (self.readyState == SR_CONNECTING && aStream == self->_inputStream) { + [self didConnect]; + } + [self _pumpWriting]; + [self _pumpScanner]; + break; + } + + case NSStreamEventErrorOccurred: { + SRFastLog(@"NSStreamEventErrorOccurred %@ %@", aStream, [[aStream streamError] copy]); + /// TODO specify error better! + [self _failWithError:aStream.streamError]; + self->_readBufferOffset = 0; + [self->_readBuffer setLength:0]; + break; + + } + + case NSStreamEventEndEncountered: { + [self _pumpScanner]; + SRFastLog(@"NSStreamEventEndEncountered %@", aStream); + if (aStream.streamError) { + [self _failWithError:aStream.streamError]; + } else { + if (self.readyState != SR_CLOSED) { + self.readyState = SR_CLOSED; + self->_selfRetain = nil; + } + + if (!self->_sentClose && !self->_failed) { + self->_sentClose = YES; + // If we get closed in this state it's probably not clean because we should be + // sending this when we send messages + [self + _performDelegateBlock:^{ + if ([self.delegate respondsToSelector:@selector(webSocket:didCloseWithCode:reason:wasClean:)]) { + [self.delegate webSocket:self + didCloseWithCode:SRStatusCodeGoingAway + reason:@"Stream end encountered" + wasClean:NO]; + } + }]; + } + } + + break; + } + + case NSStreamEventHasBytesAvailable: { + SRFastLog(@"NSStreamEventHasBytesAvailable %@", aStream); + enum EnumType : int { bufferSize = 2048 }; + uint8_t buffer[bufferSize]; + + while (self->_inputStream.hasBytesAvailable) { + NSInteger bytes_read = [self->_inputStream read:buffer maxLength:bufferSize]; + + if (bytes_read > 0) { + [self->_readBuffer appendBytes:buffer length:bytes_read]; + } else if (bytes_read < 0) { + [self _failWithError:self->_inputStream.streamError]; + } + + if (bytes_read != bufferSize) { + break; + } + }; + [self _pumpScanner]; + break; + } + + case NSStreamEventHasSpaceAvailable: { + SRFastLog(@"NSStreamEventHasSpaceAvailable %@", aStream); + [self _pumpWriting]; + break; + } + + default: + SRFastLog(@"(default) %@", aStream); + break; + } + }); +} + +@end + + +@implementation SRIOConsumer + +@synthesize bytesNeeded = _bytesNeeded; +@synthesize consumer = _scanner; +@synthesize handler = _handler; +@synthesize readToCurrentFrame = _readToCurrentFrame; +@synthesize unmaskBytes = _unmaskBytes; + +- (void)setupWithScanner:(stream_scanner)scanner handler:(data_callback)handler bytesNeeded:(size_t)bytesNeeded readToCurrentFrame:(BOOL)readToCurrentFrame unmaskBytes:(BOOL)unmaskBytes; +{ + _scanner = [scanner copy]; + _handler = [handler copy]; + _bytesNeeded = bytesNeeded; + _readToCurrentFrame = readToCurrentFrame; + _unmaskBytes = unmaskBytes; + assert(_scanner || _bytesNeeded); +} + + +@end + + +@implementation SRIOConsumerPool { + NSUInteger _poolSize; + NSMutableArray *_bufferedConsumers; +} + +- (id)initWithBufferCapacity:(NSUInteger)poolSize; +{ + self = [super init]; + if (self) { + _poolSize = poolSize; + _bufferedConsumers = [[NSMutableArray alloc] initWithCapacity:poolSize]; + } + return self; +} + +- (id)init +{ + return [self initWithBufferCapacity:8]; +} + +- (SRIOConsumer *)consumerWithScanner:(stream_scanner)scanner handler:(data_callback)handler bytesNeeded:(size_t)bytesNeeded readToCurrentFrame:(BOOL)readToCurrentFrame unmaskBytes:(BOOL)unmaskBytes; +{ + SRIOConsumer *consumer = nil; + if (_bufferedConsumers.count) { + consumer = [_bufferedConsumers lastObject]; + [_bufferedConsumers removeLastObject]; + } else { + consumer = [[SRIOConsumer alloc] init]; + } + + [consumer setupWithScanner:scanner handler:handler bytesNeeded:bytesNeeded readToCurrentFrame:readToCurrentFrame unmaskBytes:unmaskBytes]; + + return consumer; +} + +- (void)returnConsumer:(SRIOConsumer *)consumer; +{ + if (_bufferedConsumers.count < _poolSize) { + [_bufferedConsumers addObject:consumer]; + } +} + +@end + + +@implementation NSURLRequest (CertificateAdditions) + +- (NSArray *)SR_SSLPinnedCertificates; +{ + return [NSURLProtocol propertyForKey:@"SR_SSLPinnedCertificates" inRequest:self]; +} + +@end + +@implementation NSMutableURLRequest (CertificateAdditions) + +- (NSArray *)SR_SSLPinnedCertificates; +{ + return [NSURLProtocol propertyForKey:@"SR_SSLPinnedCertificates" inRequest:self]; +} + +- (void)setSR_SSLPinnedCertificates:(NSArray *)SR_SSLPinnedCertificates; +{ + [NSURLProtocol setProperty:SR_SSLPinnedCertificates forKey:@"SR_SSLPinnedCertificates" inRequest:self]; +} + +@end + +@implementation NSURL (SRWebSocket) + +- (NSString *)SR_origin; +{ + NSString *scheme = [self.scheme lowercaseString]; + + if ([scheme isEqualToString:@"wss"]) { + scheme = @"https"; + } else if ([scheme isEqualToString:@"ws"]) { + scheme = @"http"; + } + + if (self.port) { + return [NSString stringWithFormat:@"%@://%@:%@/", scheme, self.host, self.port]; + } else { + return [NSString stringWithFormat:@"%@://%@/", scheme, self.host]; + } +} + +@end + +//#define SR_ENABLE_LOG + +static inline void SRFastLog(NSString *format, ...) { +#ifdef SR_ENABLE_LOG + __block va_list arg_list; + va_start (arg_list, format); + + NSString *formattedString = [[NSString alloc] initWithFormat:format arguments:arg_list]; + + va_end(arg_list); + + NSLog(@"[SR] %@", formattedString); +#endif +} + + +#ifdef HAS_ICU + +static inline int32_t validate_dispatch_data_partial_string(NSData *data) { + if ([data length] > INT32_MAX) { + // INT32_MAX is the limit so long as this Framework is using 32 bit ints everywhere. + return -1; + } + + int32_t size = (int32_t)[data length]; + + const void * contents = [data bytes]; + const uint8_t *str = (const uint8_t *)contents; + + UChar32 codepoint = 1; + int32_t offset = 0; + int32_t lastOffset = 0; + while(offset < size && codepoint > 0) { + lastOffset = offset; + U8_NEXT(str, offset, size, codepoint); + } + + if (codepoint == -1) { + // Check to see if the last byte is valid or whether it was just continuing + if (!U8_IS_LEAD(str[lastOffset]) || U8_COUNT_TRAIL_BYTES(str[lastOffset]) + lastOffset < (int32_t)size) { + + size = -1; + } else { + uint8_t leadByte = str[lastOffset]; + U8_MASK_LEAD_BYTE(leadByte, U8_COUNT_TRAIL_BYTES(leadByte)); + + for (int i = lastOffset + 1; i < offset; i++) { + if (U8_IS_SINGLE(str[i]) || U8_IS_LEAD(str[i]) || !U8_IS_TRAIL(str[i])) { + size = -1; + } + } + + if (size != -1) { + size = lastOffset; + } + } + } + + if (size != -1 && ![[NSString alloc] initWithBytesNoCopy:(char *)[data bytes] length:size encoding:NSUTF8StringEncoding freeWhenDone:NO]) { + size = -1; + } + + return size; +} + +#else + +// This is a hack, and probably not optimal +static inline int32_t validate_dispatch_data_partial_string(NSData *data) { + static const int maxCodepointSize = 3; + + for (int i = 0; i < maxCodepointSize; i++) { + NSString *str = [[NSString alloc] initWithBytesNoCopy:(char *)data.bytes length:data.length - i encoding:NSUTF8StringEncoding freeWhenDone:NO]; + if (str) { + return data.length - i; + } + } + + return -1; +} + +#endif + +static _SRRunLoopThread *networkThread = nil; +static NSRunLoop *networkRunLoop = nil; + +@implementation NSRunLoop (SRWebSocket) + ++ (NSRunLoop *)SR_networkRunLoop { + static dispatch_once_t onceToken; + dispatch_once(&onceToken, ^{ + networkThread = [[_SRRunLoopThread alloc] init]; + networkThread.name = @"com.squareup.SocketRocket.NetworkThread"; + [networkThread start]; + networkRunLoop = networkThread.runLoop; + }); + + return networkRunLoop; +} + +@end + + +@implementation _SRRunLoopThread { + dispatch_group_t _waitGroup; +} + +@synthesize runLoop = _runLoop; + +- (void)dealloc +{ + sr_dispatch_release(_waitGroup); +} + +- (id)init +{ + self = [super init]; + if (self) { + _waitGroup = dispatch_group_create(); + dispatch_group_enter(_waitGroup); + } + return self; +} + +- (void)main; +{ + @autoreleasepool { + _runLoop = [NSRunLoop currentRunLoop]; + dispatch_group_leave(_waitGroup); + + NSTimer *timer = [[NSTimer alloc] initWithFireDate:[NSDate distantFuture] interval:0.0 target:nil selector:nil userInfo:nil repeats:NO]; + [_runLoop addTimer:timer forMode:NSDefaultRunLoopMode]; + + while ([_runLoop runMode:NSDefaultRunLoopMode beforeDate:[NSDate distantFuture]]) { + + } + assert(NO); + } +} + +- (NSRunLoop *)runLoop; +{ + dispatch_group_wait(_waitGroup, DISPATCH_TIME_FOREVER); + return _runLoop; +} + +@end diff --git a/third_party/libwebrtc/examples/objc/Icon-120.png b/third_party/libwebrtc/examples/objc/Icon-120.png Binary files differnew file mode 100644 index 0000000000..938fef477b --- /dev/null +++ b/third_party/libwebrtc/examples/objc/Icon-120.png diff --git a/third_party/libwebrtc/examples/objc/Icon-180.png b/third_party/libwebrtc/examples/objc/Icon-180.png Binary files differnew file mode 100644 index 0000000000..a5b7609680 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/Icon-180.png diff --git a/third_party/libwebrtc/examples/objc/Icon.png b/third_party/libwebrtc/examples/objc/Icon.png Binary files differnew file mode 100644 index 0000000000..55773ca9d9 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/Icon.png diff --git a/third_party/libwebrtc/examples/objc/README b/third_party/libwebrtc/examples/objc/README new file mode 100644 index 0000000000..bfe18b37c5 --- /dev/null +++ b/third_party/libwebrtc/examples/objc/README @@ -0,0 +1,3 @@ +This directory contains sample iOS and mac clients for http://apprtc.appspot.com + +See ../../app/webrtc/objc/README for information on how to use it. diff --git a/third_party/libwebrtc/examples/objcnativeapi/Info.plist b/third_party/libwebrtc/examples/objcnativeapi/Info.plist new file mode 100644 index 0000000000..cbc9e5f9f3 --- /dev/null +++ b/third_party/libwebrtc/examples/objcnativeapi/Info.plist @@ -0,0 +1,45 @@ +<?xml version="1.0" encoding="UTF-8"?> +<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd"> +<plist version="1.0"> +<dict> + <key>CFBundleDevelopmentRegion</key> + <string>en</string> + <key>CFBundleExecutable</key> + <string>$(EXECUTABLE_NAME)</string> + <key>CFBundleIdentifier</key> + <string>com.google.ObjCNativeAPIDemo</string> + <key>CFBundleInfoDictionaryVersion</key> + <string>6.0</string> + <key>CFBundleName</key> + <string>ObjCNativeAPIDemo</string> + <key>CFBundlePackageType</key> + <string>APPL</string> + <key>CFBundleShortVersionString</key> + <string>1.0</string> + <key>CFBundleVersion</key> + <string>1</string> + <key>LSRequiresIPhoneOS</key> + <true/> + <key>UIRequiredDeviceCapabilities</key> + <array> + <string>armv7</string> + </array> + <key>UISupportedInterfaceOrientations</key> + <array> + <string>UIInterfaceOrientationPortrait</string> + <string>UIInterfaceOrientationLandscapeLeft</string> + <string>UIInterfaceOrientationLandscapeRight</string> + </array> + <key>UISupportedInterfaceOrientations~ipad</key> + <array> + <string>UIInterfaceOrientationPortrait</string> + <string>UIInterfaceOrientationPortraitUpsideDown</string> + <string>UIInterfaceOrientationLandscapeLeft</string> + <string>UIInterfaceOrientationLandscapeRight</string> + </array> + <key>NSCameraUsageDescription</key> + <string>Camera access needed for video calling</string> + <key>NSMicrophoneUsageDescription</key> + <string>Microphone access needed for video calling</string> +</dict> +</plist> diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/NADAppDelegate.h b/third_party/libwebrtc/examples/objcnativeapi/objc/NADAppDelegate.h new file mode 100644 index 0000000000..02372dbfd2 --- /dev/null +++ b/third_party/libwebrtc/examples/objcnativeapi/objc/NADAppDelegate.h @@ -0,0 +1,17 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <UIKit/UIKit.h> + +@interface NADAppDelegate : UIResponder <UIApplicationDelegate> + +@property(strong, nonatomic) UIWindow* window; + +@end diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/NADAppDelegate.m b/third_party/libwebrtc/examples/objcnativeapi/objc/NADAppDelegate.m new file mode 100644 index 0000000000..254dd3be76 --- /dev/null +++ b/third_party/libwebrtc/examples/objcnativeapi/objc/NADAppDelegate.m @@ -0,0 +1,63 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "NADAppDelegate.h" + +#import "NADViewController.h" + +@interface NADAppDelegate () +@end + +@implementation NADAppDelegate + +@synthesize window = _window; + +- (BOOL)application:(UIApplication *)application + didFinishLaunchingWithOptions:(NSDictionary *)launchOptions { + _window = [[UIWindow alloc] initWithFrame:[[UIScreen mainScreen] bounds]]; + [_window makeKeyAndVisible]; + + NADViewController *viewController = [[NADViewController alloc] init]; + _window.rootViewController = viewController; + + return YES; +} + +- (void)applicationWillResignActive:(UIApplication *)application { + // Sent when the application is about to move from active to inactive state. This can occur for + // certain types of temporary interruptions (such as an incoming phone call or SMS message) or + // when the user quits the application and it begins the transition to the background state. Use + // this method to pause ongoing tasks, disable timers, and invalidate graphics rendering + // callbacks. Games should use this method to pause the game. +} + +- (void)applicationDidEnterBackground:(UIApplication *)application { + // Use this method to release shared resources, save user data, invalidate timers, and store + // enough application state information to restore your application to its current state in case + // it is terminated later. If your application supports background execution, this method is + // called instead of applicationWillTerminate: when the user quits. +} + +- (void)applicationWillEnterForeground:(UIApplication *)application { + // Called as part of the transition from the background to the active state; here you can undo + // many of the changes made on entering the background. +} + +- (void)applicationDidBecomeActive:(UIApplication *)application { + // Restart any tasks that were paused (or not yet started) while the application was inactive. If + // the application was previously in the background, optionally refresh the user interface. +} + +- (void)applicationWillTerminate:(UIApplication *)application { + // Called when the application is about to terminate. Save data if appropriate. See also + // applicationDidEnterBackground:. +} + +@end diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/NADViewController.h b/third_party/libwebrtc/examples/objcnativeapi/objc/NADViewController.h new file mode 100644 index 0000000000..c43bebb52d --- /dev/null +++ b/third_party/libwebrtc/examples/objcnativeapi/objc/NADViewController.h @@ -0,0 +1,15 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <UIKit/UIKit.h> + +@interface NADViewController : UIViewController + +@end diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/NADViewController.mm b/third_party/libwebrtc/examples/objcnativeapi/objc/NADViewController.mm new file mode 100644 index 0000000000..fd244799f8 --- /dev/null +++ b/third_party/libwebrtc/examples/objcnativeapi/objc/NADViewController.mm @@ -0,0 +1,154 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import "NADViewController.h" + +#import "sdk/objc/base/RTCVideoRenderer.h" +#import "sdk/objc/components/capturer/RTCCameraVideoCapturer.h" +#import "sdk/objc/components/renderer/metal/RTCMTLVideoView.h" +#import "sdk/objc/helpers/RTCCameraPreviewView.h" + +#include <memory> + +#include "examples/objcnativeapi/objc/objc_call_client.h" + +@interface NADViewController () + +@property(nonatomic) RTC_OBJC_TYPE(RTCCameraVideoCapturer) * capturer; +@property(nonatomic) RTC_OBJC_TYPE(RTCCameraPreviewView) * localVideoView; +@property(nonatomic) __kindof UIView<RTC_OBJC_TYPE(RTCVideoRenderer)> *remoteVideoView; +@property(nonatomic) UIButton *callButton; +@property(nonatomic) UIButton *hangUpButton; + +@end + +@implementation NADViewController { + std::unique_ptr<webrtc_examples::ObjCCallClient> _call_client; + + UIView *_view; +} + +@synthesize capturer = _capturer; +@synthesize localVideoView = _localVideoView; +@synthesize remoteVideoView = _remoteVideoView; +@synthesize callButton = _callButton; +@synthesize hangUpButton = _hangUpButton; + +#pragma mark - View controller lifecycle + +- (void)loadView { + _view = [[UIView alloc] initWithFrame:CGRectZero]; + + _remoteVideoView = [[RTC_OBJC_TYPE(RTCMTLVideoView) alloc] initWithFrame:CGRectZero]; + _remoteVideoView.translatesAutoresizingMaskIntoConstraints = NO; + [_view addSubview:_remoteVideoView]; + + _localVideoView = [[RTC_OBJC_TYPE(RTCCameraPreviewView) alloc] initWithFrame:CGRectZero]; + _localVideoView.translatesAutoresizingMaskIntoConstraints = NO; + [_view addSubview:_localVideoView]; + + _callButton = [UIButton buttonWithType:UIButtonTypeSystem]; + _callButton.translatesAutoresizingMaskIntoConstraints = NO; + [_callButton setTitle:@"Call" forState:UIControlStateNormal]; + [_callButton addTarget:self action:@selector(call:) forControlEvents:UIControlEventTouchUpInside]; + [_view addSubview:_callButton]; + + _hangUpButton = [UIButton buttonWithType:UIButtonTypeSystem]; + _hangUpButton.translatesAutoresizingMaskIntoConstraints = NO; + [_hangUpButton setTitle:@"Hang up" forState:UIControlStateNormal]; + [_hangUpButton addTarget:self + action:@selector(hangUp:) + forControlEvents:UIControlEventTouchUpInside]; + [_view addSubview:_hangUpButton]; + + UILayoutGuide *margin = _view.layoutMarginsGuide; + [_remoteVideoView.leadingAnchor constraintEqualToAnchor:margin.leadingAnchor].active = YES; + [_remoteVideoView.topAnchor constraintEqualToAnchor:margin.topAnchor].active = YES; + [_remoteVideoView.trailingAnchor constraintEqualToAnchor:margin.trailingAnchor].active = YES; + [_remoteVideoView.bottomAnchor constraintEqualToAnchor:margin.bottomAnchor].active = YES; + + [_localVideoView.leadingAnchor constraintEqualToAnchor:margin.leadingAnchor constant:8.0].active = + YES; + [_localVideoView.topAnchor constraintEqualToAnchor:margin.topAnchor constant:8.0].active = YES; + [_localVideoView.widthAnchor constraintEqualToConstant:60].active = YES; + [_localVideoView.heightAnchor constraintEqualToConstant:60].active = YES; + + [_callButton.leadingAnchor constraintEqualToAnchor:margin.leadingAnchor constant:8.0].active = + YES; + [_callButton.bottomAnchor constraintEqualToAnchor:margin.bottomAnchor constant:8.0].active = YES; + [_callButton.widthAnchor constraintEqualToConstant:100].active = YES; + [_callButton.heightAnchor constraintEqualToConstant:40].active = YES; + + [_hangUpButton.trailingAnchor constraintEqualToAnchor:margin.trailingAnchor constant:8.0].active = + YES; + [_hangUpButton.bottomAnchor constraintEqualToAnchor:margin.bottomAnchor constant:8.0].active = + YES; + [_hangUpButton.widthAnchor constraintEqualToConstant:100].active = YES; + [_hangUpButton.heightAnchor constraintEqualToConstant:40].active = YES; + + self.view = _view; +} + +- (void)viewDidLoad { + [super viewDidLoad]; + + self.capturer = [[RTC_OBJC_TYPE(RTCCameraVideoCapturer) alloc] init]; + self.localVideoView.captureSession = self.capturer.captureSession; + + _call_client.reset(new webrtc_examples::ObjCCallClient()); + + // Start capturer. + AVCaptureDevice *selectedDevice = nil; + NSArray<AVCaptureDevice *> *captureDevices = + [RTC_OBJC_TYPE(RTCCameraVideoCapturer) captureDevices]; + for (AVCaptureDevice *device in captureDevices) { + if (device.position == AVCaptureDevicePositionFront) { + selectedDevice = device; + break; + } + } + + AVCaptureDeviceFormat *selectedFormat = nil; + int targetWidth = 640; + int targetHeight = 480; + int currentDiff = INT_MAX; + NSArray<AVCaptureDeviceFormat *> *formats = + [RTC_OBJC_TYPE(RTCCameraVideoCapturer) supportedFormatsForDevice:selectedDevice]; + for (AVCaptureDeviceFormat *format in formats) { + CMVideoDimensions dimension = CMVideoFormatDescriptionGetDimensions(format.formatDescription); + FourCharCode pixelFormat = CMFormatDescriptionGetMediaSubType(format.formatDescription); + int diff = abs(targetWidth - dimension.width) + abs(targetHeight - dimension.height); + if (diff < currentDiff) { + selectedFormat = format; + currentDiff = diff; + } else if (diff == currentDiff && pixelFormat == [_capturer preferredOutputPixelFormat]) { + selectedFormat = format; + } + } + + [self.capturer startCaptureWithDevice:selectedDevice format:selectedFormat fps:30]; +} + +- (void)didReceiveMemoryWarning { + [super didReceiveMemoryWarning]; + // Dispose of any resources that can be recreated. +} + +#pragma mark - Actions + +- (IBAction)call:(id)sender { + _call_client->Call(self.capturer, self.remoteVideoView); +} + +- (IBAction)hangUp:(id)sender { + _call_client->Hangup(); +} + +@end diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/main.m b/third_party/libwebrtc/examples/objcnativeapi/objc/main.m new file mode 100644 index 0000000000..2c3b5fbbfb --- /dev/null +++ b/third_party/libwebrtc/examples/objcnativeapi/objc/main.m @@ -0,0 +1,18 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <UIKit/UIKit.h> +#import "NADAppDelegate.h" + +int main(int argc, char* argv[]) { + @autoreleasepool { + return UIApplicationMain(argc, argv, nil, NSStringFromClass([NADAppDelegate class])); + } +} diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.h b/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.h new file mode 100644 index 0000000000..cb8501d9ce --- /dev/null +++ b/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.h @@ -0,0 +1,82 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef EXAMPLES_OBJCNATIVEAPI_OBJCCALLCLIENT_H_ +#define EXAMPLES_OBJCNATIVEAPI_OBJCCALLCLIENT_H_ + +#include <memory> +#include <string> + +#import "sdk/objc/base/RTCMacros.h" + +#include "api/peer_connection_interface.h" +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" +#include "rtc_base/synchronization/mutex.h" + +@class RTC_OBJC_TYPE(RTCVideoCapturer); +@protocol RTC_OBJC_TYPE +(RTCVideoRenderer); + +namespace webrtc_examples { + +class ObjCCallClient { + public: + ObjCCallClient(); + + void Call(RTC_OBJC_TYPE(RTCVideoCapturer) * capturer, + id<RTC_OBJC_TYPE(RTCVideoRenderer)> remote_renderer); + void Hangup(); + + private: + class PCObserver : public webrtc::PeerConnectionObserver { + public: + explicit PCObserver(ObjCCallClient* client); + + void OnSignalingChange(webrtc::PeerConnectionInterface::SignalingState new_state) override; + void OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override; + void OnRenegotiationNeeded() override; + void OnIceConnectionChange( + webrtc::PeerConnectionInterface::IceConnectionState new_state) override; + void OnIceGatheringChange( + webrtc::PeerConnectionInterface::IceGatheringState new_state) override; + void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override; + + private: + ObjCCallClient* const client_; + }; + + void CreatePeerConnectionFactory() RTC_RUN_ON(thread_checker_); + void CreatePeerConnection() RTC_RUN_ON(thread_checker_); + void Connect() RTC_RUN_ON(thread_checker_); + + webrtc::SequenceChecker thread_checker_; + + bool call_started_ RTC_GUARDED_BY(thread_checker_); + + const std::unique_ptr<PCObserver> pc_observer_; + + rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf_ RTC_GUARDED_BY(thread_checker_); + std::unique_ptr<rtc::Thread> network_thread_ RTC_GUARDED_BY(thread_checker_); + std::unique_ptr<rtc::Thread> worker_thread_ RTC_GUARDED_BY(thread_checker_); + std::unique_ptr<rtc::Thread> signaling_thread_ RTC_GUARDED_BY(thread_checker_); + + std::unique_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> remote_sink_ + RTC_GUARDED_BY(thread_checker_); + rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> video_source_ + RTC_GUARDED_BY(thread_checker_); + + webrtc::Mutex pc_mutex_; + rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_ RTC_GUARDED_BY(pc_mutex_); +}; + +} // namespace webrtc_examples + +#endif // EXAMPLES_OBJCNATIVEAPI_OBJCCALLCLIENT_H_ diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.mm b/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.mm new file mode 100644 index 0000000000..7dd57b499b --- /dev/null +++ b/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.mm @@ -0,0 +1,238 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/objcnativeapi/objc/objc_call_client.h" + +#include <memory> +#include <utility> + +#import "sdk/objc/base/RTCVideoRenderer.h" +#import "sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h" +#import "sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h" +#import "sdk/objc/helpers/RTCCameraPreviewView.h" + +#include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include "api/peer_connection_interface.h" +#include "api/rtc_event_log/rtc_event_log_factory.h" +#include "api/task_queue/default_task_queue_factory.h" +#include "media/engine/webrtc_media_engine.h" +#include "modules/audio_processing/include/audio_processing.h" +#include "sdk/objc/native/api/video_capturer.h" +#include "sdk/objc/native/api/video_decoder_factory.h" +#include "sdk/objc/native/api/video_encoder_factory.h" +#include "sdk/objc/native/api/video_renderer.h" + +namespace webrtc_examples { + +namespace { + +class CreateOfferObserver : public webrtc::CreateSessionDescriptionObserver { + public: + explicit CreateOfferObserver(rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc); + + void OnSuccess(webrtc::SessionDescriptionInterface* desc) override; + void OnFailure(webrtc::RTCError error) override; + + private: + const rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_; +}; + +class SetRemoteSessionDescriptionObserver : public webrtc::SetRemoteDescriptionObserverInterface { + public: + void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override; +}; + +class SetLocalSessionDescriptionObserver : public webrtc::SetLocalDescriptionObserverInterface { + public: + void OnSetLocalDescriptionComplete(webrtc::RTCError error) override; +}; + +} // namespace + +ObjCCallClient::ObjCCallClient() + : call_started_(false), pc_observer_(std::make_unique<PCObserver>(this)) { + thread_checker_.Detach(); + CreatePeerConnectionFactory(); +} + +void ObjCCallClient::Call(RTC_OBJC_TYPE(RTCVideoCapturer) * capturer, + id<RTC_OBJC_TYPE(RTCVideoRenderer)> remote_renderer) { + RTC_DCHECK_RUN_ON(&thread_checker_); + + webrtc::MutexLock lock(&pc_mutex_); + if (call_started_) { + RTC_LOG(LS_WARNING) << "Call already started."; + return; + } + call_started_ = true; + + remote_sink_ = webrtc::ObjCToNativeVideoRenderer(remote_renderer); + + video_source_ = + webrtc::ObjCToNativeVideoCapturer(capturer, signaling_thread_.get(), worker_thread_.get()); + + CreatePeerConnection(); + Connect(); +} + +void ObjCCallClient::Hangup() { + RTC_DCHECK_RUN_ON(&thread_checker_); + + call_started_ = false; + + { + webrtc::MutexLock lock(&pc_mutex_); + if (pc_ != nullptr) { + pc_->Close(); + pc_ = nullptr; + } + } + + remote_sink_ = nullptr; + video_source_ = nullptr; +} + +void ObjCCallClient::CreatePeerConnectionFactory() { + network_thread_ = rtc::Thread::CreateWithSocketServer(); + network_thread_->SetName("network_thread", nullptr); + RTC_CHECK(network_thread_->Start()) << "Failed to start thread"; + + worker_thread_ = rtc::Thread::Create(); + worker_thread_->SetName("worker_thread", nullptr); + RTC_CHECK(worker_thread_->Start()) << "Failed to start thread"; + + signaling_thread_ = rtc::Thread::Create(); + signaling_thread_->SetName("signaling_thread", nullptr); + RTC_CHECK(signaling_thread_->Start()) << "Failed to start thread"; + + webrtc::PeerConnectionFactoryDependencies dependencies; + dependencies.network_thread = network_thread_.get(); + dependencies.worker_thread = worker_thread_.get(); + dependencies.signaling_thread = signaling_thread_.get(); + dependencies.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory(); + cricket::MediaEngineDependencies media_deps; + media_deps.task_queue_factory = dependencies.task_queue_factory.get(); + media_deps.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory(); + media_deps.audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory(); + media_deps.video_encoder_factory = webrtc::ObjCToNativeVideoEncoderFactory( + [[RTC_OBJC_TYPE(RTCDefaultVideoEncoderFactory) alloc] init]); + media_deps.video_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory( + [[RTC_OBJC_TYPE(RTCDefaultVideoDecoderFactory) alloc] init]); + media_deps.audio_processing = webrtc::AudioProcessingBuilder().Create(); + dependencies.media_engine = cricket::CreateMediaEngine(std::move(media_deps)); + RTC_LOG(LS_INFO) << "Media engine created: " << dependencies.media_engine.get(); + dependencies.call_factory = webrtc::CreateCallFactory(); + dependencies.event_log_factory = + std::make_unique<webrtc::RtcEventLogFactory>(dependencies.task_queue_factory.get()); + pcf_ = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies)); + RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << pcf_.get(); +} + +void ObjCCallClient::CreatePeerConnection() { + webrtc::MutexLock lock(&pc_mutex_); + webrtc::PeerConnectionInterface::RTCConfiguration config; + config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; + // Encryption has to be disabled for loopback to work. + webrtc::PeerConnectionFactoryInterface::Options options; + options.disable_encryption = true; + pcf_->SetOptions(options); + webrtc::PeerConnectionDependencies pc_dependencies(pc_observer_.get()); + pc_ = pcf_->CreatePeerConnectionOrError(config, std::move(pc_dependencies)).MoveValue(); + RTC_LOG(LS_INFO) << "PeerConnection created: " << pc_.get(); + + rtc::scoped_refptr<webrtc::VideoTrackInterface> local_video_track = + pcf_->CreateVideoTrack("video", video_source_.get()); + pc_->AddTransceiver(local_video_track); + RTC_LOG(LS_INFO) << "Local video sink set up: " << local_video_track.get(); + + for (const rtc::scoped_refptr<webrtc::RtpTransceiverInterface>& tranceiver : + pc_->GetTransceivers()) { + rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track = tranceiver->receiver()->track(); + if (track && track->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) { + static_cast<webrtc::VideoTrackInterface*>(track.get()) + ->AddOrUpdateSink(remote_sink_.get(), rtc::VideoSinkWants()); + RTC_LOG(LS_INFO) << "Remote video sink set up: " << track.get(); + break; + } + } +} + +void ObjCCallClient::Connect() { + webrtc::MutexLock lock(&pc_mutex_); + pc_->CreateOffer(rtc::make_ref_counted<CreateOfferObserver>(pc_).get(), + webrtc::PeerConnectionInterface::RTCOfferAnswerOptions()); +} + +ObjCCallClient::PCObserver::PCObserver(ObjCCallClient* client) : client_(client) {} + +void ObjCCallClient::PCObserver::OnSignalingChange( + webrtc::PeerConnectionInterface::SignalingState new_state) { + RTC_LOG(LS_INFO) << "OnSignalingChange: " << new_state; +} + +void ObjCCallClient::PCObserver::OnDataChannel( + rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) { + RTC_LOG(LS_INFO) << "OnDataChannel"; +} + +void ObjCCallClient::PCObserver::OnRenegotiationNeeded() { + RTC_LOG(LS_INFO) << "OnRenegotiationNeeded"; +} + +void ObjCCallClient::PCObserver::OnIceConnectionChange( + webrtc::PeerConnectionInterface::IceConnectionState new_state) { + RTC_LOG(LS_INFO) << "OnIceConnectionChange: " << new_state; +} + +void ObjCCallClient::PCObserver::OnIceGatheringChange( + webrtc::PeerConnectionInterface::IceGatheringState new_state) { + RTC_LOG(LS_INFO) << "OnIceGatheringChange: " << new_state; +} + +void ObjCCallClient::PCObserver::OnIceCandidate(const webrtc::IceCandidateInterface* candidate) { + RTC_LOG(LS_INFO) << "OnIceCandidate: " << candidate->server_url(); + webrtc::MutexLock lock(&client_->pc_mutex_); + RTC_DCHECK(client_->pc_ != nullptr); + client_->pc_->AddIceCandidate(candidate); +} + +CreateOfferObserver::CreateOfferObserver(rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc) + : pc_(pc) {} + +void CreateOfferObserver::OnSuccess(webrtc::SessionDescriptionInterface* desc) { + std::string sdp; + desc->ToString(&sdp); + RTC_LOG(LS_INFO) << "Created offer: " << sdp; + + // Ownership of desc was transferred to us, now we transfer it forward. + pc_->SetLocalDescription(absl::WrapUnique(desc), + rtc::make_ref_counted<SetLocalSessionDescriptionObserver>()); + + // Generate a fake answer. + std::unique_ptr<webrtc::SessionDescriptionInterface> answer( + webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp)); + pc_->SetRemoteDescription(std::move(answer), + rtc::make_ref_counted<SetRemoteSessionDescriptionObserver>()); +} + +void CreateOfferObserver::OnFailure(webrtc::RTCError error) { + RTC_LOG(LS_INFO) << "Failed to create offer: " << error.message(); +} + +void SetRemoteSessionDescriptionObserver::OnSetRemoteDescriptionComplete(webrtc::RTCError error) { + RTC_LOG(LS_INFO) << "Set remote description: " << error.message(); +} + +void SetLocalSessionDescriptionObserver::OnSetLocalDescriptionComplete(webrtc::RTCError error) { + RTC_LOG(LS_INFO) << "Set local description: " << error.message(); +} + +} // namespace webrtc_examples diff --git a/third_party/libwebrtc/examples/peerconnection/OWNERS b/third_party/libwebrtc/examples/peerconnection/OWNERS new file mode 100644 index 0000000000..0fba125734 --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/OWNERS @@ -0,0 +1 @@ +tommi@webrtc.org diff --git a/third_party/libwebrtc/examples/peerconnection/client/conductor.cc b/third_party/libwebrtc/examples/peerconnection/client/conductor.cc new file mode 100644 index 0000000000..965525abff --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/client/conductor.cc @@ -0,0 +1,599 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/peerconnection/client/conductor.h" + +#include <stddef.h> +#include <stdint.h> + +#include <memory> +#include <utility> +#include <vector> + +#include "absl/memory/memory.h" +#include "absl/types/optional.h" +#include "api/audio/audio_mixer.h" +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_encoder_factory.h" +#include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include "api/audio_options.h" +#include "api/create_peerconnection_factory.h" +#include "api/rtp_sender_interface.h" +#include "api/video_codecs/builtin_video_decoder_factory.h" +#include "api/video_codecs/builtin_video_encoder_factory.h" +#include "api/video_codecs/video_decoder_factory.h" +#include "api/video_codecs/video_encoder_factory.h" +#include "examples/peerconnection/client/defaults.h" +#include "modules/audio_device/include/audio_device.h" +#include "modules/audio_processing/include/audio_processing.h" +#include "modules/video_capture/video_capture.h" +#include "modules/video_capture/video_capture_factory.h" +#include "p2p/base/port_allocator.h" +#include "pc/video_track_source.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" +#include "rtc_base/rtc_certificate_generator.h" +#include "rtc_base/strings/json.h" +#include "test/vcm_capturer.h" + +namespace { +// Names used for a IceCandidate JSON object. +const char kCandidateSdpMidName[] = "sdpMid"; +const char kCandidateSdpMlineIndexName[] = "sdpMLineIndex"; +const char kCandidateSdpName[] = "candidate"; + +// Names used for a SessionDescription JSON object. +const char kSessionDescriptionTypeName[] = "type"; +const char kSessionDescriptionSdpName[] = "sdp"; + +class DummySetSessionDescriptionObserver + : public webrtc::SetSessionDescriptionObserver { + public: + static rtc::scoped_refptr<DummySetSessionDescriptionObserver> Create() { + return rtc::make_ref_counted<DummySetSessionDescriptionObserver>(); + } + virtual void OnSuccess() { RTC_LOG(LS_INFO) << __FUNCTION__; } + virtual void OnFailure(webrtc::RTCError error) { + RTC_LOG(LS_INFO) << __FUNCTION__ << " " << ToString(error.type()) << ": " + << error.message(); + } +}; + +class CapturerTrackSource : public webrtc::VideoTrackSource { + public: + static rtc::scoped_refptr<CapturerTrackSource> Create() { + const size_t kWidth = 640; + const size_t kHeight = 480; + const size_t kFps = 30; + std::unique_ptr<webrtc::test::VcmCapturer> capturer; + std::unique_ptr<webrtc::VideoCaptureModule::DeviceInfo> info( + webrtc::VideoCaptureFactory::CreateDeviceInfo()); + if (!info) { + return nullptr; + } + int num_devices = info->NumberOfDevices(); + for (int i = 0; i < num_devices; ++i) { + capturer = absl::WrapUnique( + webrtc::test::VcmCapturer::Create(kWidth, kHeight, kFps, i)); + if (capturer) { + return rtc::make_ref_counted<CapturerTrackSource>(std::move(capturer)); + } + } + + return nullptr; + } + + protected: + explicit CapturerTrackSource( + std::unique_ptr<webrtc::test::VcmCapturer> capturer) + : VideoTrackSource(/*remote=*/false), capturer_(std::move(capturer)) {} + + private: + rtc::VideoSourceInterface<webrtc::VideoFrame>* source() override { + return capturer_.get(); + } + std::unique_ptr<webrtc::test::VcmCapturer> capturer_; +}; + +} // namespace + +Conductor::Conductor(PeerConnectionClient* client, MainWindow* main_wnd) + : peer_id_(-1), loopback_(false), client_(client), main_wnd_(main_wnd) { + client_->RegisterObserver(this); + main_wnd->RegisterObserver(this); +} + +Conductor::~Conductor() { + RTC_DCHECK(!peer_connection_); +} + +bool Conductor::connection_active() const { + return peer_connection_ != nullptr; +} + +void Conductor::Close() { + client_->SignOut(); + DeletePeerConnection(); +} + +bool Conductor::InitializePeerConnection() { + RTC_DCHECK(!peer_connection_factory_); + RTC_DCHECK(!peer_connection_); + + if (!signaling_thread_.get()) { + signaling_thread_ = rtc::Thread::CreateWithSocketServer(); + signaling_thread_->Start(); + } + peer_connection_factory_ = webrtc::CreatePeerConnectionFactory( + nullptr /* network_thread */, nullptr /* worker_thread */, + signaling_thread_.get(), nullptr /* default_adm */, + webrtc::CreateBuiltinAudioEncoderFactory(), + webrtc::CreateBuiltinAudioDecoderFactory(), + webrtc::CreateBuiltinVideoEncoderFactory(), + webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */, + nullptr /* audio_processing */); + + if (!peer_connection_factory_) { + main_wnd_->MessageBox("Error", "Failed to initialize PeerConnectionFactory", + true); + DeletePeerConnection(); + return false; + } + + if (!CreatePeerConnection()) { + main_wnd_->MessageBox("Error", "CreatePeerConnection failed", true); + DeletePeerConnection(); + } + + AddTracks(); + + return peer_connection_ != nullptr; +} + +bool Conductor::ReinitializePeerConnectionForLoopback() { + loopback_ = true; + std::vector<rtc::scoped_refptr<webrtc::RtpSenderInterface>> senders = + peer_connection_->GetSenders(); + peer_connection_ = nullptr; + // Loopback is only possible if encryption is disabled. + webrtc::PeerConnectionFactoryInterface::Options options; + options.disable_encryption = true; + peer_connection_factory_->SetOptions(options); + if (CreatePeerConnection()) { + for (const auto& sender : senders) { + peer_connection_->AddTrack(sender->track(), sender->stream_ids()); + } + peer_connection_->CreateOffer( + this, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions()); + } + options.disable_encryption = false; + peer_connection_factory_->SetOptions(options); + return peer_connection_ != nullptr; +} + +bool Conductor::CreatePeerConnection() { + RTC_DCHECK(peer_connection_factory_); + RTC_DCHECK(!peer_connection_); + + webrtc::PeerConnectionInterface::RTCConfiguration config; + config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; + webrtc::PeerConnectionInterface::IceServer server; + server.uri = GetPeerConnectionString(); + config.servers.push_back(server); + + webrtc::PeerConnectionDependencies pc_dependencies(this); + auto error_or_peer_connection = + peer_connection_factory_->CreatePeerConnectionOrError( + config, std::move(pc_dependencies)); + if (error_or_peer_connection.ok()) { + peer_connection_ = std::move(error_or_peer_connection.value()); + } + return peer_connection_ != nullptr; +} + +void Conductor::DeletePeerConnection() { + main_wnd_->StopLocalRenderer(); + main_wnd_->StopRemoteRenderer(); + peer_connection_ = nullptr; + peer_connection_factory_ = nullptr; + peer_id_ = -1; + loopback_ = false; +} + +void Conductor::EnsureStreamingUI() { + RTC_DCHECK(peer_connection_); + if (main_wnd_->IsWindow()) { + if (main_wnd_->current_ui() != MainWindow::STREAMING) + main_wnd_->SwitchToStreamingUI(); + } +} + +// +// PeerConnectionObserver implementation. +// + +void Conductor::OnAddTrack( + rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver, + const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>& + streams) { + RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id(); + main_wnd_->QueueUIThreadCallback(NEW_TRACK_ADDED, + receiver->track().release()); +} + +void Conductor::OnRemoveTrack( + rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver) { + RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id(); + main_wnd_->QueueUIThreadCallback(TRACK_REMOVED, receiver->track().release()); +} + +void Conductor::OnIceCandidate(const webrtc::IceCandidateInterface* candidate) { + RTC_LOG(LS_INFO) << __FUNCTION__ << " " << candidate->sdp_mline_index(); + // For loopback test. To save some connecting delay. + if (loopback_) { + if (!peer_connection_->AddIceCandidate(candidate)) { + RTC_LOG(LS_WARNING) << "Failed to apply the received candidate"; + } + return; + } + + Json::Value jmessage; + jmessage[kCandidateSdpMidName] = candidate->sdp_mid(); + jmessage[kCandidateSdpMlineIndexName] = candidate->sdp_mline_index(); + std::string sdp; + if (!candidate->ToString(&sdp)) { + RTC_LOG(LS_ERROR) << "Failed to serialize candidate"; + return; + } + jmessage[kCandidateSdpName] = sdp; + + Json::StreamWriterBuilder factory; + SendMessage(Json::writeString(factory, jmessage)); +} + +// +// PeerConnectionClientObserver implementation. +// + +void Conductor::OnSignedIn() { + RTC_LOG(LS_INFO) << __FUNCTION__; + main_wnd_->SwitchToPeerList(client_->peers()); +} + +void Conductor::OnDisconnected() { + RTC_LOG(LS_INFO) << __FUNCTION__; + + DeletePeerConnection(); + + if (main_wnd_->IsWindow()) + main_wnd_->SwitchToConnectUI(); +} + +void Conductor::OnPeerConnected(int id, const std::string& name) { + RTC_LOG(LS_INFO) << __FUNCTION__; + // Refresh the list if we're showing it. + if (main_wnd_->current_ui() == MainWindow::LIST_PEERS) + main_wnd_->SwitchToPeerList(client_->peers()); +} + +void Conductor::OnPeerDisconnected(int id) { + RTC_LOG(LS_INFO) << __FUNCTION__; + if (id == peer_id_) { + RTC_LOG(LS_INFO) << "Our peer disconnected"; + main_wnd_->QueueUIThreadCallback(PEER_CONNECTION_CLOSED, NULL); + } else { + // Refresh the list if we're showing it. + if (main_wnd_->current_ui() == MainWindow::LIST_PEERS) + main_wnd_->SwitchToPeerList(client_->peers()); + } +} + +void Conductor::OnMessageFromPeer(int peer_id, const std::string& message) { + RTC_DCHECK(peer_id_ == peer_id || peer_id_ == -1); + RTC_DCHECK(!message.empty()); + + if (!peer_connection_.get()) { + RTC_DCHECK(peer_id_ == -1); + peer_id_ = peer_id; + + if (!InitializePeerConnection()) { + RTC_LOG(LS_ERROR) << "Failed to initialize our PeerConnection instance"; + client_->SignOut(); + return; + } + } else if (peer_id != peer_id_) { + RTC_DCHECK(peer_id_ != -1); + RTC_LOG(LS_WARNING) + << "Received a message from unknown peer while already in a " + "conversation with a different peer."; + return; + } + + Json::CharReaderBuilder factory; + std::unique_ptr<Json::CharReader> reader = + absl::WrapUnique(factory.newCharReader()); + Json::Value jmessage; + if (!reader->parse(message.data(), message.data() + message.length(), + &jmessage, nullptr)) { + RTC_LOG(LS_WARNING) << "Received unknown message. " << message; + return; + } + std::string type_str; + std::string json_object; + + rtc::GetStringFromJsonObject(jmessage, kSessionDescriptionTypeName, + &type_str); + if (!type_str.empty()) { + if (type_str == "offer-loopback") { + // This is a loopback call. + // Recreate the peerconnection with DTLS disabled. + if (!ReinitializePeerConnectionForLoopback()) { + RTC_LOG(LS_ERROR) << "Failed to initialize our PeerConnection instance"; + DeletePeerConnection(); + client_->SignOut(); + } + return; + } + absl::optional<webrtc::SdpType> type_maybe = + webrtc::SdpTypeFromString(type_str); + if (!type_maybe) { + RTC_LOG(LS_ERROR) << "Unknown SDP type: " << type_str; + return; + } + webrtc::SdpType type = *type_maybe; + std::string sdp; + if (!rtc::GetStringFromJsonObject(jmessage, kSessionDescriptionSdpName, + &sdp)) { + RTC_LOG(LS_WARNING) + << "Can't parse received session description message."; + return; + } + webrtc::SdpParseError error; + std::unique_ptr<webrtc::SessionDescriptionInterface> session_description = + webrtc::CreateSessionDescription(type, sdp, &error); + if (!session_description) { + RTC_LOG(LS_WARNING) + << "Can't parse received session description message. " + "SdpParseError was: " + << error.description; + return; + } + RTC_LOG(LS_INFO) << " Received session description :" << message; + peer_connection_->SetRemoteDescription( + DummySetSessionDescriptionObserver::Create().get(), + session_description.release()); + if (type == webrtc::SdpType::kOffer) { + peer_connection_->CreateAnswer( + this, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions()); + } + } else { + std::string sdp_mid; + int sdp_mlineindex = 0; + std::string sdp; + if (!rtc::GetStringFromJsonObject(jmessage, kCandidateSdpMidName, + &sdp_mid) || + !rtc::GetIntFromJsonObject(jmessage, kCandidateSdpMlineIndexName, + &sdp_mlineindex) || + !rtc::GetStringFromJsonObject(jmessage, kCandidateSdpName, &sdp)) { + RTC_LOG(LS_WARNING) << "Can't parse received message."; + return; + } + webrtc::SdpParseError error; + std::unique_ptr<webrtc::IceCandidateInterface> candidate( + webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, sdp, &error)); + if (!candidate.get()) { + RTC_LOG(LS_WARNING) << "Can't parse received candidate message. " + "SdpParseError was: " + << error.description; + return; + } + if (!peer_connection_->AddIceCandidate(candidate.get())) { + RTC_LOG(LS_WARNING) << "Failed to apply the received candidate"; + return; + } + RTC_LOG(LS_INFO) << " Received candidate :" << message; + } +} + +void Conductor::OnMessageSent(int err) { + // Process the next pending message if any. + main_wnd_->QueueUIThreadCallback(SEND_MESSAGE_TO_PEER, NULL); +} + +void Conductor::OnServerConnectionFailure() { + main_wnd_->MessageBox("Error", ("Failed to connect to " + server_).c_str(), + true); +} + +// +// MainWndCallback implementation. +// + +void Conductor::StartLogin(const std::string& server, int port) { + if (client_->is_connected()) + return; + server_ = server; + client_->Connect(server, port, GetPeerName()); +} + +void Conductor::DisconnectFromServer() { + if (client_->is_connected()) + client_->SignOut(); +} + +void Conductor::ConnectToPeer(int peer_id) { + RTC_DCHECK(peer_id_ == -1); + RTC_DCHECK(peer_id != -1); + + if (peer_connection_.get()) { + main_wnd_->MessageBox( + "Error", "We only support connecting to one peer at a time", true); + return; + } + + if (InitializePeerConnection()) { + peer_id_ = peer_id; + peer_connection_->CreateOffer( + this, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions()); + } else { + main_wnd_->MessageBox("Error", "Failed to initialize PeerConnection", true); + } +} + +void Conductor::AddTracks() { + if (!peer_connection_->GetSenders().empty()) { + return; // Already added tracks. + } + + rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( + peer_connection_factory_->CreateAudioTrack( + kAudioLabel, + peer_connection_factory_->CreateAudioSource(cricket::AudioOptions()) + .get())); + auto result_or_error = peer_connection_->AddTrack(audio_track, {kStreamId}); + if (!result_or_error.ok()) { + RTC_LOG(LS_ERROR) << "Failed to add audio track to PeerConnection: " + << result_or_error.error().message(); + } + + rtc::scoped_refptr<CapturerTrackSource> video_device = + CapturerTrackSource::Create(); + if (video_device) { + rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track_( + peer_connection_factory_->CreateVideoTrack(kVideoLabel, + video_device.get())); + main_wnd_->StartLocalRenderer(video_track_.get()); + + result_or_error = peer_connection_->AddTrack(video_track_, {kStreamId}); + if (!result_or_error.ok()) { + RTC_LOG(LS_ERROR) << "Failed to add video track to PeerConnection: " + << result_or_error.error().message(); + } + } else { + RTC_LOG(LS_ERROR) << "OpenVideoCaptureDevice failed"; + } + + main_wnd_->SwitchToStreamingUI(); +} + +void Conductor::DisconnectFromCurrentPeer() { + RTC_LOG(LS_INFO) << __FUNCTION__; + if (peer_connection_.get()) { + client_->SendHangUp(peer_id_); + DeletePeerConnection(); + } + + if (main_wnd_->IsWindow()) + main_wnd_->SwitchToPeerList(client_->peers()); +} + +void Conductor::UIThreadCallback(int msg_id, void* data) { + switch (msg_id) { + case PEER_CONNECTION_CLOSED: + RTC_LOG(LS_INFO) << "PEER_CONNECTION_CLOSED"; + DeletePeerConnection(); + + if (main_wnd_->IsWindow()) { + if (client_->is_connected()) { + main_wnd_->SwitchToPeerList(client_->peers()); + } else { + main_wnd_->SwitchToConnectUI(); + } + } else { + DisconnectFromServer(); + } + break; + + case SEND_MESSAGE_TO_PEER: { + RTC_LOG(LS_INFO) << "SEND_MESSAGE_TO_PEER"; + std::string* msg = reinterpret_cast<std::string*>(data); + if (msg) { + // For convenience, we always run the message through the queue. + // This way we can be sure that messages are sent to the server + // in the same order they were signaled without much hassle. + pending_messages_.push_back(msg); + } + + if (!pending_messages_.empty() && !client_->IsSendingMessage()) { + msg = pending_messages_.front(); + pending_messages_.pop_front(); + + if (!client_->SendToPeer(peer_id_, *msg) && peer_id_ != -1) { + RTC_LOG(LS_ERROR) << "SendToPeer failed"; + DisconnectFromServer(); + } + delete msg; + } + + if (!peer_connection_.get()) + peer_id_ = -1; + + break; + } + + case NEW_TRACK_ADDED: { + auto* track = reinterpret_cast<webrtc::MediaStreamTrackInterface*>(data); + if (track->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) { + auto* video_track = static_cast<webrtc::VideoTrackInterface*>(track); + main_wnd_->StartRemoteRenderer(video_track); + } + track->Release(); + break; + } + + case TRACK_REMOVED: { + // Remote peer stopped sending a track. + auto* track = reinterpret_cast<webrtc::MediaStreamTrackInterface*>(data); + track->Release(); + break; + } + + default: + RTC_DCHECK_NOTREACHED(); + break; + } +} + +void Conductor::OnSuccess(webrtc::SessionDescriptionInterface* desc) { + peer_connection_->SetLocalDescription( + DummySetSessionDescriptionObserver::Create().get(), desc); + + std::string sdp; + desc->ToString(&sdp); + + // For loopback test. To save some connecting delay. + if (loopback_) { + // Replace message type from "offer" to "answer" + std::unique_ptr<webrtc::SessionDescriptionInterface> session_description = + webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp); + peer_connection_->SetRemoteDescription( + DummySetSessionDescriptionObserver::Create().get(), + session_description.release()); + return; + } + + Json::Value jmessage; + jmessage[kSessionDescriptionTypeName] = + webrtc::SdpTypeToString(desc->GetType()); + jmessage[kSessionDescriptionSdpName] = sdp; + + Json::StreamWriterBuilder factory; + SendMessage(Json::writeString(factory, jmessage)); +} + +void Conductor::OnFailure(webrtc::RTCError error) { + RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message(); +} + +void Conductor::SendMessage(const std::string& json_object) { + std::string* msg = new std::string(json_object); + main_wnd_->QueueUIThreadCallback(SEND_MESSAGE_TO_PEER, msg); +} diff --git a/third_party/libwebrtc/examples/peerconnection/client/conductor.h b/third_party/libwebrtc/examples/peerconnection/client/conductor.h new file mode 100644 index 0000000000..80617d3cf4 --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/client/conductor.h @@ -0,0 +1,136 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef EXAMPLES_PEERCONNECTION_CLIENT_CONDUCTOR_H_ +#define EXAMPLES_PEERCONNECTION_CLIENT_CONDUCTOR_H_ + +#include <deque> +#include <map> +#include <memory> +#include <string> +#include <vector> + +#include "api/media_stream_interface.h" +#include "api/peer_connection_interface.h" +#include "examples/peerconnection/client/main_wnd.h" +#include "examples/peerconnection/client/peer_connection_client.h" +#include "rtc_base/thread.h" + +namespace webrtc { +class VideoCaptureModule; +} // namespace webrtc + +namespace cricket { +class VideoRenderer; +} // namespace cricket + +class Conductor : public webrtc::PeerConnectionObserver, + public webrtc::CreateSessionDescriptionObserver, + public PeerConnectionClientObserver, + public MainWndCallback { + public: + enum CallbackID { + MEDIA_CHANNELS_INITIALIZED = 1, + PEER_CONNECTION_CLOSED, + SEND_MESSAGE_TO_PEER, + NEW_TRACK_ADDED, + TRACK_REMOVED, + }; + + Conductor(PeerConnectionClient* client, MainWindow* main_wnd); + + bool connection_active() const; + + void Close() override; + + protected: + ~Conductor(); + bool InitializePeerConnection(); + bool ReinitializePeerConnectionForLoopback(); + bool CreatePeerConnection(); + void DeletePeerConnection(); + void EnsureStreamingUI(); + void AddTracks(); + + // + // PeerConnectionObserver implementation. + // + + void OnSignalingChange( + webrtc::PeerConnectionInterface::SignalingState new_state) override {} + void OnAddTrack( + rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver, + const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>& + streams) override; + void OnRemoveTrack( + rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver) override; + void OnDataChannel( + rtc::scoped_refptr<webrtc::DataChannelInterface> channel) override {} + void OnRenegotiationNeeded() override {} + void OnIceConnectionChange( + webrtc::PeerConnectionInterface::IceConnectionState new_state) override {} + void OnIceGatheringChange( + webrtc::PeerConnectionInterface::IceGatheringState new_state) override {} + void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override; + void OnIceConnectionReceivingChange(bool receiving) override {} + + // + // PeerConnectionClientObserver implementation. + // + + void OnSignedIn() override; + + void OnDisconnected() override; + + void OnPeerConnected(int id, const std::string& name) override; + + void OnPeerDisconnected(int id) override; + + void OnMessageFromPeer(int peer_id, const std::string& message) override; + + void OnMessageSent(int err) override; + + void OnServerConnectionFailure() override; + + // + // MainWndCallback implementation. + // + + void StartLogin(const std::string& server, int port) override; + + void DisconnectFromServer() override; + + void ConnectToPeer(int peer_id) override; + + void DisconnectFromCurrentPeer() override; + + void UIThreadCallback(int msg_id, void* data) override; + + // CreateSessionDescriptionObserver implementation. + void OnSuccess(webrtc::SessionDescriptionInterface* desc) override; + void OnFailure(webrtc::RTCError error) override; + + protected: + // Send a message to the remote peer. + void SendMessage(const std::string& json_object); + + int peer_id_; + bool loopback_; + std::unique_ptr<rtc::Thread> signaling_thread_; + rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; + rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> + peer_connection_factory_; + PeerConnectionClient* client_; + MainWindow* main_wnd_; + std::deque<std::string*> pending_messages_; + std::string server_; +}; + +#endif // EXAMPLES_PEERCONNECTION_CLIENT_CONDUCTOR_H_ diff --git a/third_party/libwebrtc/examples/peerconnection/client/defaults.cc b/third_party/libwebrtc/examples/peerconnection/client/defaults.cc new file mode 100644 index 0000000000..ee3a9e1f0a --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/client/defaults.cc @@ -0,0 +1,59 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/peerconnection/client/defaults.h" + +#include <stdlib.h> + +#ifdef WIN32 +#include <winsock2.h> +#else +#include <unistd.h> +#endif + +#include "rtc_base/arraysize.h" + +const char kAudioLabel[] = "audio_label"; +const char kVideoLabel[] = "video_label"; +const char kStreamId[] = "stream_id"; +const uint16_t kDefaultServerPort = 8888; + +std::string GetEnvVarOrDefault(const char* env_var_name, + const char* default_value) { + std::string value; + const char* env_var = getenv(env_var_name); + if (env_var) + value = env_var; + + if (value.empty()) + value = default_value; + + return value; +} + +std::string GetPeerConnectionString() { + return GetEnvVarOrDefault("WEBRTC_CONNECT", "stun:stun.l.google.com:19302"); +} + +std::string GetDefaultServerName() { + return GetEnvVarOrDefault("WEBRTC_SERVER", "localhost"); +} + +std::string GetPeerName() { + char computer_name[256]; + std::string ret(GetEnvVarOrDefault("USERNAME", "user")); + ret += '@'; + if (gethostname(computer_name, arraysize(computer_name)) == 0) { + ret += computer_name; + } else { + ret += "host"; + } + return ret; +} diff --git a/third_party/libwebrtc/examples/peerconnection/client/defaults.h b/third_party/libwebrtc/examples/peerconnection/client/defaults.h new file mode 100644 index 0000000000..30936fd9d4 --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/client/defaults.h @@ -0,0 +1,29 @@ +/* + * Copyright 2011 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef EXAMPLES_PEERCONNECTION_CLIENT_DEFAULTS_H_ +#define EXAMPLES_PEERCONNECTION_CLIENT_DEFAULTS_H_ + +#include <stdint.h> + +#include <string> + +extern const char kAudioLabel[]; +extern const char kVideoLabel[]; +extern const char kStreamId[]; +extern const uint16_t kDefaultServerPort; + +std::string GetEnvVarOrDefault(const char* env_var_name, + const char* default_value); +std::string GetPeerConnectionString(); +std::string GetDefaultServerName(); +std::string GetPeerName(); + +#endif // EXAMPLES_PEERCONNECTION_CLIENT_DEFAULTS_H_ diff --git a/third_party/libwebrtc/examples/peerconnection/client/flag_defs.h b/third_party/libwebrtc/examples/peerconnection/client/flag_defs.h new file mode 100644 index 0000000000..986daf64ce --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/client/flag_defs.h @@ -0,0 +1,52 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef EXAMPLES_PEERCONNECTION_CLIENT_FLAG_DEFS_H_ +#define EXAMPLES_PEERCONNECTION_CLIENT_FLAG_DEFS_H_ + +#include <string> + +#include "absl/flags/flag.h" + +extern const uint16_t kDefaultServerPort; // From defaults.[h|cc] + +// Define flags for the peerconnect_client testing tool, in a separate +// header file so that they can be shared across the different main.cc's +// for each platform. + +ABSL_FLAG(bool, + autoconnect, + false, + "Connect to the server without user " + "intervention."); +ABSL_FLAG(std::string, server, "localhost", "The server to connect to."); +ABSL_FLAG(int, + port, + kDefaultServerPort, + "The port on which the server is listening."); +ABSL_FLAG( + bool, + autocall, + false, + "Call the first available other client on " + "the server without user intervention. Note: this flag should only be set " + "to true on one of the two clients."); + +ABSL_FLAG( + std::string, + force_fieldtrials, + "", + "Field trials control experimental features. This flag specifies the field " + "trials in effect. E.g. running with " + "--force_fieldtrials=WebRTC-FooFeature/Enabled/ " + "will assign the group Enabled to field trial WebRTC-FooFeature. Multiple " + "trials are separated by \"/\""); + +#endif // EXAMPLES_PEERCONNECTION_CLIENT_FLAG_DEFS_H_ diff --git a/third_party/libwebrtc/examples/peerconnection/client/linux/main.cc b/third_party/libwebrtc/examples/peerconnection/client/linux/main.cc new file mode 100644 index 0000000000..ad3d671073 --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/client/linux/main.cc @@ -0,0 +1,121 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <glib.h> +#include <gtk/gtk.h> +#include <stdio.h> + +#include "absl/flags/parse.h" +#include "api/scoped_refptr.h" +#include "examples/peerconnection/client/conductor.h" +#include "examples/peerconnection/client/flag_defs.h" +#include "examples/peerconnection/client/linux/main_wnd.h" +#include "examples/peerconnection/client/peer_connection_client.h" +#include "rtc_base/physical_socket_server.h" +#include "rtc_base/ssl_adapter.h" +#include "rtc_base/thread.h" +#include "system_wrappers/include/field_trial.h" +#include "test/field_trial.h" + +class CustomSocketServer : public rtc::PhysicalSocketServer { + public: + explicit CustomSocketServer(GtkMainWnd* wnd) + : wnd_(wnd), conductor_(NULL), client_(NULL) {} + virtual ~CustomSocketServer() {} + + void SetMessageQueue(rtc::Thread* queue) override { message_queue_ = queue; } + + void set_client(PeerConnectionClient* client) { client_ = client; } + void set_conductor(Conductor* conductor) { conductor_ = conductor; } + + // Override so that we can also pump the GTK message loop. + // This function never waits. + bool Wait(webrtc::TimeDelta max_wait_duration, bool process_io) override { + // Pump GTK events. + // TODO(henrike): We really should move either the socket server or UI to a + // different thread. Alternatively we could look at merging the two loops + // by implementing a dispatcher for the socket server and/or use + // g_main_context_set_poll_func. + while (gtk_events_pending()) + gtk_main_iteration(); + + if (!wnd_->IsWindow() && !conductor_->connection_active() && + client_ != NULL && !client_->is_connected()) { + message_queue_->Quit(); + } + return rtc::PhysicalSocketServer::Wait(webrtc::TimeDelta::Zero(), + process_io); + } + + protected: + rtc::Thread* message_queue_; + GtkMainWnd* wnd_; + Conductor* conductor_; + PeerConnectionClient* client_; +}; + +int main(int argc, char* argv[]) { + gtk_init(&argc, &argv); +// g_type_init API is deprecated (and does nothing) since glib 2.35.0, see: +// https://mail.gnome.org/archives/commits-list/2012-November/msg07809.html +#if !GLIB_CHECK_VERSION(2, 35, 0) + g_type_init(); +#endif +// g_thread_init API is deprecated since glib 2.31.0, see release note: +// http://mail.gnome.org/archives/gnome-announce-list/2011-October/msg00041.html +#if !GLIB_CHECK_VERSION(2, 31, 0) + g_thread_init(NULL); +#endif + + absl::ParseCommandLine(argc, argv); + + // InitFieldTrialsFromString stores the char*, so the char array must outlive + // the application. + const std::string forced_field_trials = + absl::GetFlag(FLAGS_force_fieldtrials); + webrtc::field_trial::InitFieldTrialsFromString(forced_field_trials.c_str()); + + // Abort if the user specifies a port that is outside the allowed + // range [1, 65535]. + if ((absl::GetFlag(FLAGS_port) < 1) || (absl::GetFlag(FLAGS_port) > 65535)) { + printf("Error: %i is not a valid port.\n", absl::GetFlag(FLAGS_port)); + return -1; + } + + const std::string server = absl::GetFlag(FLAGS_server); + GtkMainWnd wnd(server.c_str(), absl::GetFlag(FLAGS_port), + absl::GetFlag(FLAGS_autoconnect), + absl::GetFlag(FLAGS_autocall)); + wnd.Create(); + + CustomSocketServer socket_server(&wnd); + rtc::AutoSocketServerThread thread(&socket_server); + + rtc::InitializeSSL(); + // Must be constructed after we set the socketserver. + PeerConnectionClient client; + auto conductor = rtc::make_ref_counted<Conductor>(&client, &wnd); + socket_server.set_client(&client); + socket_server.set_conductor(conductor.get()); + + thread.Run(); + + // gtk_main(); + wnd.Destroy(); + + // TODO(henrike): Run the Gtk main loop to tear down the connection. + /* + while (gtk_events_pending()) { + gtk_main_iteration(); + } + */ + rtc::CleanupSSL(); + return 0; +} diff --git a/third_party/libwebrtc/examples/peerconnection/client/linux/main_wnd.cc b/third_party/libwebrtc/examples/peerconnection/client/linux/main_wnd.cc new file mode 100644 index 0000000000..2be75d8f8d --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/client/linux/main_wnd.cc @@ -0,0 +1,545 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/peerconnection/client/linux/main_wnd.h" + +#include <cairo.h> +#include <gdk/gdk.h> +#include <gdk/gdkkeysyms.h> +#include <glib-object.h> +#include <glib.h> +#include <gobject/gclosure.h> +#include <gtk/gtk.h> +#include <stddef.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> + +#include <cstdint> +#include <map> +#include <utility> + +#include "api/video/i420_buffer.h" +#include "api/video/video_frame_buffer.h" +#include "api/video/video_rotation.h" +#include "api/video/video_source_interface.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" +#include "third_party/libyuv/include/libyuv/convert.h" +#include "third_party/libyuv/include/libyuv/convert_from.h" + +namespace { + +// +// Simple static functions that simply forward the callback to the +// GtkMainWnd instance. +// + +gboolean OnDestroyedCallback(GtkWidget* widget, + GdkEvent* event, + gpointer data) { + reinterpret_cast<GtkMainWnd*>(data)->OnDestroyed(widget, event); + return FALSE; +} + +void OnClickedCallback(GtkWidget* widget, gpointer data) { + reinterpret_cast<GtkMainWnd*>(data)->OnClicked(widget); +} + +gboolean SimulateButtonClick(gpointer button) { + g_signal_emit_by_name(button, "clicked"); + return false; +} + +gboolean OnKeyPressCallback(GtkWidget* widget, + GdkEventKey* key, + gpointer data) { + reinterpret_cast<GtkMainWnd*>(data)->OnKeyPress(widget, key); + return false; +} + +void OnRowActivatedCallback(GtkTreeView* tree_view, + GtkTreePath* path, + GtkTreeViewColumn* column, + gpointer data) { + reinterpret_cast<GtkMainWnd*>(data)->OnRowActivated(tree_view, path, column); +} + +gboolean SimulateLastRowActivated(gpointer data) { + GtkTreeView* tree_view = reinterpret_cast<GtkTreeView*>(data); + GtkTreeModel* model = gtk_tree_view_get_model(tree_view); + + // "if iter is NULL, then the number of toplevel nodes is returned." + int rows = gtk_tree_model_iter_n_children(model, NULL); + GtkTreePath* lastpath = gtk_tree_path_new_from_indices(rows - 1, -1); + + // Select the last item in the list + GtkTreeSelection* selection = gtk_tree_view_get_selection(tree_view); + gtk_tree_selection_select_path(selection, lastpath); + + // Our TreeView only has one column, so it is column 0. + GtkTreeViewColumn* column = gtk_tree_view_get_column(tree_view, 0); + + gtk_tree_view_row_activated(tree_view, lastpath, column); + + gtk_tree_path_free(lastpath); + return false; +} + +// Creates a tree view, that we use to display the list of peers. +void InitializeList(GtkWidget* list) { + GtkCellRenderer* renderer = gtk_cell_renderer_text_new(); + GtkTreeViewColumn* column = gtk_tree_view_column_new_with_attributes( + "List Items", renderer, "text", 0, NULL); + gtk_tree_view_append_column(GTK_TREE_VIEW(list), column); + GtkListStore* store = gtk_list_store_new(2, G_TYPE_STRING, G_TYPE_INT); + gtk_tree_view_set_model(GTK_TREE_VIEW(list), GTK_TREE_MODEL(store)); + g_object_unref(store); +} + +// Adds an entry to a tree view. +void AddToList(GtkWidget* list, const gchar* str, int value) { + GtkListStore* store = + GTK_LIST_STORE(gtk_tree_view_get_model(GTK_TREE_VIEW(list))); + + GtkTreeIter iter; + gtk_list_store_append(store, &iter); + gtk_list_store_set(store, &iter, 0, str, 1, value, -1); +} + +struct UIThreadCallbackData { + explicit UIThreadCallbackData(MainWndCallback* cb, int id, void* d) + : callback(cb), msg_id(id), data(d) {} + MainWndCallback* callback; + int msg_id; + void* data; +}; + +gboolean HandleUIThreadCallback(gpointer data) { + UIThreadCallbackData* cb_data = reinterpret_cast<UIThreadCallbackData*>(data); + cb_data->callback->UIThreadCallback(cb_data->msg_id, cb_data->data); + delete cb_data; + return false; +} + +gboolean Redraw(gpointer data) { + GtkMainWnd* wnd = reinterpret_cast<GtkMainWnd*>(data); + wnd->OnRedraw(); + return false; +} + +gboolean Draw(GtkWidget* widget, cairo_t* cr, gpointer data) { + GtkMainWnd* wnd = reinterpret_cast<GtkMainWnd*>(data); + wnd->Draw(widget, cr); + return false; +} + +} // namespace + +// +// GtkMainWnd implementation. +// + +GtkMainWnd::GtkMainWnd(const char* server, + int port, + bool autoconnect, + bool autocall) + : window_(NULL), + draw_area_(NULL), + vbox_(NULL), + server_edit_(NULL), + port_edit_(NULL), + peer_list_(NULL), + callback_(NULL), + server_(server), + autoconnect_(autoconnect), + autocall_(autocall) { + char buffer[10]; + snprintf(buffer, sizeof(buffer), "%i", port); + port_ = buffer; +} + +GtkMainWnd::~GtkMainWnd() { + RTC_DCHECK(!IsWindow()); +} + +void GtkMainWnd::RegisterObserver(MainWndCallback* callback) { + callback_ = callback; +} + +bool GtkMainWnd::IsWindow() { + return window_ != NULL && GTK_IS_WINDOW(window_); +} + +void GtkMainWnd::MessageBox(const char* caption, + const char* text, + bool is_error) { + GtkWidget* dialog = gtk_message_dialog_new( + GTK_WINDOW(window_), GTK_DIALOG_DESTROY_WITH_PARENT, + is_error ? GTK_MESSAGE_ERROR : GTK_MESSAGE_INFO, GTK_BUTTONS_CLOSE, "%s", + text); + gtk_window_set_title(GTK_WINDOW(dialog), caption); + gtk_dialog_run(GTK_DIALOG(dialog)); + gtk_widget_destroy(dialog); +} + +MainWindow::UI GtkMainWnd::current_ui() { + if (vbox_) + return CONNECT_TO_SERVER; + + if (peer_list_) + return LIST_PEERS; + + return STREAMING; +} + +void GtkMainWnd::StartLocalRenderer(webrtc::VideoTrackInterface* local_video) { + local_renderer_.reset(new VideoRenderer(this, local_video)); +} + +void GtkMainWnd::StopLocalRenderer() { + local_renderer_.reset(); +} + +void GtkMainWnd::StartRemoteRenderer( + webrtc::VideoTrackInterface* remote_video) { + remote_renderer_.reset(new VideoRenderer(this, remote_video)); +} + +void GtkMainWnd::StopRemoteRenderer() { + remote_renderer_.reset(); +} + +void GtkMainWnd::QueueUIThreadCallback(int msg_id, void* data) { + g_idle_add(HandleUIThreadCallback, + new UIThreadCallbackData(callback_, msg_id, data)); +} + +bool GtkMainWnd::Create() { + RTC_DCHECK(window_ == NULL); + + window_ = gtk_window_new(GTK_WINDOW_TOPLEVEL); + if (window_) { + gtk_window_set_position(GTK_WINDOW(window_), GTK_WIN_POS_CENTER); + gtk_window_set_default_size(GTK_WINDOW(window_), 640, 480); + gtk_window_set_title(GTK_WINDOW(window_), "PeerConnection client"); + g_signal_connect(G_OBJECT(window_), "delete-event", + G_CALLBACK(&OnDestroyedCallback), this); + g_signal_connect(window_, "key-press-event", G_CALLBACK(OnKeyPressCallback), + this); + + SwitchToConnectUI(); + } + + return window_ != NULL; +} + +bool GtkMainWnd::Destroy() { + if (!IsWindow()) + return false; + + gtk_widget_destroy(window_); + window_ = NULL; + + return true; +} + +void GtkMainWnd::SwitchToConnectUI() { + RTC_LOG(LS_INFO) << __FUNCTION__; + + RTC_DCHECK(IsWindow()); + RTC_DCHECK(vbox_ == NULL); + + gtk_container_set_border_width(GTK_CONTAINER(window_), 10); + + if (peer_list_) { + gtk_widget_destroy(peer_list_); + peer_list_ = NULL; + } + + vbox_ = gtk_box_new(GTK_ORIENTATION_VERTICAL, 5); + GtkWidget* valign = gtk_alignment_new(0, 1, 0, 0); + gtk_container_add(GTK_CONTAINER(vbox_), valign); + gtk_container_add(GTK_CONTAINER(window_), vbox_); + + GtkWidget* hbox = gtk_box_new(GTK_ORIENTATION_HORIZONTAL, 5); + + GtkWidget* label = gtk_label_new("Server"); + gtk_container_add(GTK_CONTAINER(hbox), label); + + server_edit_ = gtk_entry_new(); + gtk_entry_set_text(GTK_ENTRY(server_edit_), server_.c_str()); + gtk_widget_set_size_request(server_edit_, 400, 30); + gtk_container_add(GTK_CONTAINER(hbox), server_edit_); + + port_edit_ = gtk_entry_new(); + gtk_entry_set_text(GTK_ENTRY(port_edit_), port_.c_str()); + gtk_widget_set_size_request(port_edit_, 70, 30); + gtk_container_add(GTK_CONTAINER(hbox), port_edit_); + + GtkWidget* button = gtk_button_new_with_label("Connect"); + gtk_widget_set_size_request(button, 70, 30); + g_signal_connect(button, "clicked", G_CALLBACK(OnClickedCallback), this); + gtk_container_add(GTK_CONTAINER(hbox), button); + + GtkWidget* halign = gtk_alignment_new(1, 0, 0, 0); + gtk_container_add(GTK_CONTAINER(halign), hbox); + gtk_box_pack_start(GTK_BOX(vbox_), halign, FALSE, FALSE, 0); + + gtk_widget_show_all(window_); + + if (autoconnect_) + g_idle_add(SimulateButtonClick, button); +} + +void GtkMainWnd::SwitchToPeerList(const Peers& peers) { + RTC_LOG(LS_INFO) << __FUNCTION__; + + if (!peer_list_) { + gtk_container_set_border_width(GTK_CONTAINER(window_), 0); + if (vbox_) { + gtk_widget_destroy(vbox_); + vbox_ = NULL; + server_edit_ = NULL; + port_edit_ = NULL; + } else if (draw_area_) { + gtk_widget_destroy(draw_area_); + draw_area_ = NULL; + draw_buffer_.reset(); + } + + peer_list_ = gtk_tree_view_new(); + g_signal_connect(peer_list_, "row-activated", + G_CALLBACK(OnRowActivatedCallback), this); + gtk_tree_view_set_headers_visible(GTK_TREE_VIEW(peer_list_), FALSE); + InitializeList(peer_list_); + gtk_container_add(GTK_CONTAINER(window_), peer_list_); + gtk_widget_show_all(window_); + } else { + GtkListStore* store = + GTK_LIST_STORE(gtk_tree_view_get_model(GTK_TREE_VIEW(peer_list_))); + gtk_list_store_clear(store); + } + + AddToList(peer_list_, "List of currently connected peers:", -1); + for (Peers::const_iterator i = peers.begin(); i != peers.end(); ++i) + AddToList(peer_list_, i->second.c_str(), i->first); + + if (autocall_ && peers.begin() != peers.end()) + g_idle_add(SimulateLastRowActivated, peer_list_); +} + +void GtkMainWnd::SwitchToStreamingUI() { + RTC_LOG(LS_INFO) << __FUNCTION__; + + RTC_DCHECK(draw_area_ == NULL); + + gtk_container_set_border_width(GTK_CONTAINER(window_), 0); + if (peer_list_) { + gtk_widget_destroy(peer_list_); + peer_list_ = NULL; + } + + draw_area_ = gtk_drawing_area_new(); + gtk_container_add(GTK_CONTAINER(window_), draw_area_); + g_signal_connect(G_OBJECT(draw_area_), "draw", G_CALLBACK(&::Draw), this); + + gtk_widget_show_all(window_); +} + +void GtkMainWnd::OnDestroyed(GtkWidget* widget, GdkEvent* event) { + callback_->Close(); + window_ = NULL; + draw_area_ = NULL; + vbox_ = NULL; + server_edit_ = NULL; + port_edit_ = NULL; + peer_list_ = NULL; +} + +void GtkMainWnd::OnClicked(GtkWidget* widget) { + // Make the connect button insensitive, so that it cannot be clicked more than + // once. Now that the connection includes auto-retry, it should not be + // necessary to click it more than once. + gtk_widget_set_sensitive(widget, false); + server_ = gtk_entry_get_text(GTK_ENTRY(server_edit_)); + port_ = gtk_entry_get_text(GTK_ENTRY(port_edit_)); + int port = port_.length() ? atoi(port_.c_str()) : 0; + callback_->StartLogin(server_, port); +} + +void GtkMainWnd::OnKeyPress(GtkWidget* widget, GdkEventKey* key) { + if (key->type == GDK_KEY_PRESS) { + switch (key->keyval) { + case GDK_KEY_Escape: + if (draw_area_) { + callback_->DisconnectFromCurrentPeer(); + } else if (peer_list_) { + callback_->DisconnectFromServer(); + } + break; + + case GDK_KEY_KP_Enter: + case GDK_KEY_Return: + if (vbox_) { + OnClicked(NULL); + } else if (peer_list_) { + // OnRowActivated will be called automatically when the user + // presses enter. + } + break; + + default: + break; + } + } +} + +void GtkMainWnd::OnRowActivated(GtkTreeView* tree_view, + GtkTreePath* path, + GtkTreeViewColumn* column) { + RTC_DCHECK(peer_list_ != NULL); + GtkTreeIter iter; + GtkTreeModel* model; + GtkTreeSelection* selection = + gtk_tree_view_get_selection(GTK_TREE_VIEW(tree_view)); + if (gtk_tree_selection_get_selected(selection, &model, &iter)) { + char* text; + int id = -1; + gtk_tree_model_get(model, &iter, 0, &text, 1, &id, -1); + if (id != -1) + callback_->ConnectToPeer(id); + g_free(text); + } +} + +void GtkMainWnd::OnRedraw() { + gdk_threads_enter(); + + VideoRenderer* remote_renderer = remote_renderer_.get(); + if (remote_renderer && remote_renderer->image() != NULL && + draw_area_ != NULL) { + width_ = remote_renderer->width(); + height_ = remote_renderer->height(); + + if (!draw_buffer_.get()) { + draw_buffer_size_ = (width_ * height_ * 4) * 4; + draw_buffer_.reset(new uint8_t[draw_buffer_size_]); + gtk_widget_set_size_request(draw_area_, width_ * 2, height_ * 2); + } + + const uint32_t* image = + reinterpret_cast<const uint32_t*>(remote_renderer->image()); + uint32_t* scaled = reinterpret_cast<uint32_t*>(draw_buffer_.get()); + for (int r = 0; r < height_; ++r) { + for (int c = 0; c < width_; ++c) { + int x = c * 2; + scaled[x] = scaled[x + 1] = image[c]; + } + + uint32_t* prev_line = scaled; + scaled += width_ * 2; + memcpy(scaled, prev_line, (width_ * 2) * 4); + + image += width_; + scaled += width_ * 2; + } + + VideoRenderer* local_renderer = local_renderer_.get(); + if (local_renderer && local_renderer->image()) { + image = reinterpret_cast<const uint32_t*>(local_renderer->image()); + scaled = reinterpret_cast<uint32_t*>(draw_buffer_.get()); + // Position the local preview on the right side. + scaled += (width_ * 2) - (local_renderer->width() / 2); + // right margin... + scaled -= 10; + // ... towards the bottom. + scaled += (height_ * width_ * 4) - ((local_renderer->height() / 2) * + (local_renderer->width() / 2) * 4); + // bottom margin... + scaled -= (width_ * 2) * 5; + for (int r = 0; r < local_renderer->height(); r += 2) { + for (int c = 0; c < local_renderer->width(); c += 2) { + scaled[c / 2] = image[c + r * local_renderer->width()]; + } + scaled += width_ * 2; + } + } + + gtk_widget_queue_draw(draw_area_); + } + + gdk_threads_leave(); +} + +void GtkMainWnd::Draw(GtkWidget* widget, cairo_t* cr) { + cairo_format_t format = CAIRO_FORMAT_ARGB32; + cairo_surface_t* surface = cairo_image_surface_create_for_data( + draw_buffer_.get(), format, width_ * 2, height_ * 2, + cairo_format_stride_for_width(format, width_ * 2)); + cairo_set_source_surface(cr, surface, 0, 0); + cairo_rectangle(cr, 0, 0, width_ * 2, height_ * 2); + cairo_fill(cr); + cairo_surface_destroy(surface); +} + +GtkMainWnd::VideoRenderer::VideoRenderer( + GtkMainWnd* main_wnd, + webrtc::VideoTrackInterface* track_to_render) + : width_(0), + height_(0), + main_wnd_(main_wnd), + rendered_track_(track_to_render) { + rendered_track_->AddOrUpdateSink(this, rtc::VideoSinkWants()); +} + +GtkMainWnd::VideoRenderer::~VideoRenderer() { + rendered_track_->RemoveSink(this); +} + +void GtkMainWnd::VideoRenderer::SetSize(int width, int height) { + gdk_threads_enter(); + + if (width_ == width && height_ == height) { + return; + } + + width_ = width; + height_ = height; + image_.reset(new uint8_t[width * height * 4]); + gdk_threads_leave(); +} + +void GtkMainWnd::VideoRenderer::OnFrame(const webrtc::VideoFrame& video_frame) { + gdk_threads_enter(); + + rtc::scoped_refptr<webrtc::I420BufferInterface> buffer( + video_frame.video_frame_buffer()->ToI420()); + if (video_frame.rotation() != webrtc::kVideoRotation_0) { + buffer = webrtc::I420Buffer::Rotate(*buffer, video_frame.rotation()); + } + SetSize(buffer->width(), buffer->height()); + + // TODO(bugs.webrtc.org/6857): This conversion is correct for little-endian + // only. Cairo ARGB32 treats pixels as 32-bit values in *native* byte order, + // with B in the least significant byte of the 32-bit value. Which on + // little-endian means that memory layout is BGRA, with the B byte stored at + // lowest address. Libyuv's ARGB format (surprisingly?) uses the same + // little-endian format, with B in the first byte in memory, regardless of + // native endianness. + libyuv::I420ToARGB(buffer->DataY(), buffer->StrideY(), buffer->DataU(), + buffer->StrideU(), buffer->DataV(), buffer->StrideV(), + image_.get(), width_ * 4, buffer->width(), + buffer->height()); + + gdk_threads_leave(); + + g_idle_add(Redraw, main_wnd_); +} diff --git a/third_party/libwebrtc/examples/peerconnection/client/linux/main_wnd.h b/third_party/libwebrtc/examples/peerconnection/client/linux/main_wnd.h new file mode 100644 index 0000000000..3b31e1be3b --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/client/linux/main_wnd.h @@ -0,0 +1,128 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef EXAMPLES_PEERCONNECTION_CLIENT_LINUX_MAIN_WND_H_ +#define EXAMPLES_PEERCONNECTION_CLIENT_LINUX_MAIN_WND_H_ + +#include <stdint.h> + +#include <memory> +#include <string> + +#include "api/media_stream_interface.h" +#include "api/scoped_refptr.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" +#include "examples/peerconnection/client/main_wnd.h" +#include "examples/peerconnection/client/peer_connection_client.h" + +// Forward declarations. +typedef struct _GtkWidget GtkWidget; +typedef union _GdkEvent GdkEvent; +typedef struct _GdkEventKey GdkEventKey; +typedef struct _GtkTreeView GtkTreeView; +typedef struct _GtkTreePath GtkTreePath; +typedef struct _GtkTreeViewColumn GtkTreeViewColumn; +typedef struct _cairo cairo_t; + +// Implements the main UI of the peer connection client. +// This is functionally equivalent to the MainWnd class in the Windows +// implementation. +class GtkMainWnd : public MainWindow { + public: + GtkMainWnd(const char* server, int port, bool autoconnect, bool autocall); + ~GtkMainWnd(); + + virtual void RegisterObserver(MainWndCallback* callback); + virtual bool IsWindow(); + virtual void SwitchToConnectUI(); + virtual void SwitchToPeerList(const Peers& peers); + virtual void SwitchToStreamingUI(); + virtual void MessageBox(const char* caption, const char* text, bool is_error); + virtual MainWindow::UI current_ui(); + virtual void StartLocalRenderer(webrtc::VideoTrackInterface* local_video); + virtual void StopLocalRenderer(); + virtual void StartRemoteRenderer(webrtc::VideoTrackInterface* remote_video); + virtual void StopRemoteRenderer(); + + virtual void QueueUIThreadCallback(int msg_id, void* data); + + // Creates and shows the main window with the |Connect UI| enabled. + bool Create(); + + // Destroys the window. When the window is destroyed, it ends the + // main message loop. + bool Destroy(); + + // Callback for when the main window is destroyed. + void OnDestroyed(GtkWidget* widget, GdkEvent* event); + + // Callback for when the user clicks the "Connect" button. + void OnClicked(GtkWidget* widget); + + // Callback for keystrokes. Used to capture Esc and Return. + void OnKeyPress(GtkWidget* widget, GdkEventKey* key); + + // Callback when the user double clicks a peer in order to initiate a + // connection. + void OnRowActivated(GtkTreeView* tree_view, + GtkTreePath* path, + GtkTreeViewColumn* column); + + void OnRedraw(); + + void Draw(GtkWidget* widget, cairo_t* cr); + + protected: + class VideoRenderer : public rtc::VideoSinkInterface<webrtc::VideoFrame> { + public: + VideoRenderer(GtkMainWnd* main_wnd, + webrtc::VideoTrackInterface* track_to_render); + virtual ~VideoRenderer(); + + // VideoSinkInterface implementation + void OnFrame(const webrtc::VideoFrame& frame) override; + + const uint8_t* image() const { return image_.get(); } + + int width() const { return width_; } + + int height() const { return height_; } + + protected: + void SetSize(int width, int height); + std::unique_ptr<uint8_t[]> image_; + int width_; + int height_; + GtkMainWnd* main_wnd_; + rtc::scoped_refptr<webrtc::VideoTrackInterface> rendered_track_; + }; + + protected: + GtkWidget* window_; // Our main window. + GtkWidget* draw_area_; // The drawing surface for rendering video streams. + GtkWidget* vbox_; // Container for the Connect UI. + GtkWidget* server_edit_; + GtkWidget* port_edit_; + GtkWidget* peer_list_; // The list of peers. + MainWndCallback* callback_; + std::string server_; + std::string port_; + bool autoconnect_; + bool autocall_; + std::unique_ptr<VideoRenderer> local_renderer_; + std::unique_ptr<VideoRenderer> remote_renderer_; + int width_; + int height_; + std::unique_ptr<uint8_t[]> draw_buffer_; + int draw_buffer_size_; +}; + +#endif // EXAMPLES_PEERCONNECTION_CLIENT_LINUX_MAIN_WND_H_ diff --git a/third_party/libwebrtc/examples/peerconnection/client/main.cc b/third_party/libwebrtc/examples/peerconnection/client/main.cc new file mode 100644 index 0000000000..32bc52bda4 --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/client/main.cc @@ -0,0 +1,133 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// clang-format off +// clang formating would change include order. +#include <windows.h> +#include <shellapi.h> // must come after windows.h +// clang-format on + +#include <string> +#include <vector> + +#include "absl/flags/parse.h" +#include "examples/peerconnection/client/conductor.h" +#include "examples/peerconnection/client/flag_defs.h" +#include "examples/peerconnection/client/main_wnd.h" +#include "examples/peerconnection/client/peer_connection_client.h" +#include "rtc_base/checks.h" +#include "rtc_base/ssl_adapter.h" +#include "rtc_base/string_utils.h" // For ToUtf8 +#include "rtc_base/win32_socket_init.h" +#include "system_wrappers/include/field_trial.h" +#include "test/field_trial.h" + +namespace { +// A helper class to translate Windows command line arguments into UTF8, +// which then allows us to just pass them to the flags system. +// This encapsulates all the work of getting the command line and translating +// it to an array of 8-bit strings; all you have to do is create one of these, +// and then call argc() and argv(). +class WindowsCommandLineArguments { + public: + WindowsCommandLineArguments(); + + WindowsCommandLineArguments(const WindowsCommandLineArguments&) = delete; + WindowsCommandLineArguments& operator=(WindowsCommandLineArguments&) = delete; + + int argc() { return argv_.size(); } + char** argv() { return argv_.data(); } + + private: + // Owned argument strings. + std::vector<std::string> args_; + // Pointers, to get layout compatible with char** argv. + std::vector<char*> argv_; +}; + +WindowsCommandLineArguments::WindowsCommandLineArguments() { + // start by getting the command line. + LPCWSTR command_line = ::GetCommandLineW(); + // now, convert it to a list of wide char strings. + int argc; + LPWSTR* wide_argv = ::CommandLineToArgvW(command_line, &argc); + + // iterate over the returned wide strings; + for (int i = 0; i < argc; ++i) { + args_.push_back(rtc::ToUtf8(wide_argv[i], wcslen(wide_argv[i]))); + // make sure the argv array points to the string data. + argv_.push_back(const_cast<char*>(args_.back().c_str())); + } + LocalFree(wide_argv); +} + +} // namespace +int PASCAL wWinMain(HINSTANCE instance, + HINSTANCE prev_instance, + wchar_t* cmd_line, + int cmd_show) { + rtc::WinsockInitializer winsock_init; + rtc::PhysicalSocketServer ss; + rtc::AutoSocketServerThread main_thread(&ss); + + WindowsCommandLineArguments win_args; + int argc = win_args.argc(); + char** argv = win_args.argv(); + + absl::ParseCommandLine(argc, argv); + + // InitFieldTrialsFromString stores the char*, so the char array must outlive + // the application. + const std::string forced_field_trials = + absl::GetFlag(FLAGS_force_fieldtrials); + webrtc::field_trial::InitFieldTrialsFromString(forced_field_trials.c_str()); + + // Abort if the user specifies a port that is outside the allowed + // range [1, 65535]. + if ((absl::GetFlag(FLAGS_port) < 1) || (absl::GetFlag(FLAGS_port) > 65535)) { + printf("Error: %i is not a valid port.\n", absl::GetFlag(FLAGS_port)); + return -1; + } + + const std::string server = absl::GetFlag(FLAGS_server); + MainWnd wnd(server.c_str(), absl::GetFlag(FLAGS_port), + absl::GetFlag(FLAGS_autoconnect), absl::GetFlag(FLAGS_autocall)); + if (!wnd.Create()) { + RTC_DCHECK_NOTREACHED(); + return -1; + } + + rtc::InitializeSSL(); + PeerConnectionClient client; + auto conductor = rtc::make_ref_counted<Conductor>(&client, &wnd); + + // Main loop. + MSG msg; + BOOL gm; + while ((gm = ::GetMessage(&msg, NULL, 0, 0)) != 0 && gm != -1) { + if (!wnd.PreTranslateMessage(&msg)) { + ::TranslateMessage(&msg); + ::DispatchMessage(&msg); + } + } + + if (conductor->connection_active() || client.is_connected()) { + while ((conductor->connection_active() || client.is_connected()) && + (gm = ::GetMessage(&msg, NULL, 0, 0)) != 0 && gm != -1) { + if (!wnd.PreTranslateMessage(&msg)) { + ::TranslateMessage(&msg); + ::DispatchMessage(&msg); + } + } + } + + rtc::CleanupSSL(); + return 0; +} diff --git a/third_party/libwebrtc/examples/peerconnection/client/main_wnd.cc b/third_party/libwebrtc/examples/peerconnection/client/main_wnd.cc new file mode 100644 index 0000000000..afafa621b3 --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/client/main_wnd.cc @@ -0,0 +1,633 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/peerconnection/client/main_wnd.h" + +#include <math.h> + +#include "api/video/i420_buffer.h" +#include "examples/peerconnection/client/defaults.h" +#include "rtc_base/arraysize.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" +#include "third_party/libyuv/include/libyuv/convert_argb.h" + +ATOM MainWnd::wnd_class_ = 0; +const wchar_t MainWnd::kClassName[] = L"WebRTC_MainWnd"; + +namespace { + +const char kConnecting[] = "Connecting... "; +const char kNoVideoStreams[] = "(no video streams either way)"; +const char kNoIncomingStream[] = "(no incoming video)"; + +void CalculateWindowSizeForText(HWND wnd, + const wchar_t* text, + size_t* width, + size_t* height) { + HDC dc = ::GetDC(wnd); + RECT text_rc = {0}; + ::DrawTextW(dc, text, -1, &text_rc, DT_CALCRECT | DT_SINGLELINE); + ::ReleaseDC(wnd, dc); + RECT client, window; + ::GetClientRect(wnd, &client); + ::GetWindowRect(wnd, &window); + + *width = text_rc.right - text_rc.left; + *width += (window.right - window.left) - (client.right - client.left); + *height = text_rc.bottom - text_rc.top; + *height += (window.bottom - window.top) - (client.bottom - client.top); +} + +HFONT GetDefaultFont() { + static HFONT font = reinterpret_cast<HFONT>(GetStockObject(DEFAULT_GUI_FONT)); + return font; +} + +std::string GetWindowText(HWND wnd) { + char text[MAX_PATH] = {0}; + ::GetWindowTextA(wnd, &text[0], ARRAYSIZE(text)); + return text; +} + +void AddListBoxItem(HWND listbox, const std::string& str, LPARAM item_data) { + LRESULT index = ::SendMessageA(listbox, LB_ADDSTRING, 0, + reinterpret_cast<LPARAM>(str.c_str())); + ::SendMessageA(listbox, LB_SETITEMDATA, index, item_data); +} + +} // namespace + +MainWnd::MainWnd(const char* server, + int port, + bool auto_connect, + bool auto_call) + : ui_(CONNECT_TO_SERVER), + wnd_(NULL), + edit1_(NULL), + edit2_(NULL), + label1_(NULL), + label2_(NULL), + button_(NULL), + listbox_(NULL), + destroyed_(false), + nested_msg_(NULL), + callback_(NULL), + server_(server), + auto_connect_(auto_connect), + auto_call_(auto_call) { + char buffer[10]; + snprintf(buffer, sizeof(buffer), "%i", port); + port_ = buffer; +} + +MainWnd::~MainWnd() { + RTC_DCHECK(!IsWindow()); +} + +bool MainWnd::Create() { + RTC_DCHECK(wnd_ == NULL); + if (!RegisterWindowClass()) + return false; + + ui_thread_id_ = ::GetCurrentThreadId(); + wnd_ = + ::CreateWindowExW(WS_EX_OVERLAPPEDWINDOW, kClassName, L"WebRTC", + WS_OVERLAPPEDWINDOW | WS_VISIBLE | WS_CLIPCHILDREN, + CW_USEDEFAULT, CW_USEDEFAULT, CW_USEDEFAULT, + CW_USEDEFAULT, NULL, NULL, GetModuleHandle(NULL), this); + + ::SendMessage(wnd_, WM_SETFONT, reinterpret_cast<WPARAM>(GetDefaultFont()), + TRUE); + + CreateChildWindows(); + SwitchToConnectUI(); + + return wnd_ != NULL; +} + +bool MainWnd::Destroy() { + BOOL ret = FALSE; + if (IsWindow()) { + ret = ::DestroyWindow(wnd_); + } + + return ret != FALSE; +} + +void MainWnd::RegisterObserver(MainWndCallback* callback) { + callback_ = callback; +} + +bool MainWnd::IsWindow() { + return wnd_ && ::IsWindow(wnd_) != FALSE; +} + +bool MainWnd::PreTranslateMessage(MSG* msg) { + bool ret = false; + if (msg->message == WM_CHAR) { + if (msg->wParam == VK_TAB) { + HandleTabbing(); + ret = true; + } else if (msg->wParam == VK_RETURN) { + OnDefaultAction(); + ret = true; + } else if (msg->wParam == VK_ESCAPE) { + if (callback_) { + if (ui_ == STREAMING) { + callback_->DisconnectFromCurrentPeer(); + } else { + callback_->DisconnectFromServer(); + } + } + } + } else if (msg->hwnd == NULL && msg->message == UI_THREAD_CALLBACK) { + callback_->UIThreadCallback(static_cast<int>(msg->wParam), + reinterpret_cast<void*>(msg->lParam)); + ret = true; + } + return ret; +} + +void MainWnd::SwitchToConnectUI() { + RTC_DCHECK(IsWindow()); + LayoutPeerListUI(false); + ui_ = CONNECT_TO_SERVER; + LayoutConnectUI(true); + ::SetFocus(edit1_); + + if (auto_connect_) + ::PostMessage(button_, BM_CLICK, 0, 0); +} + +void MainWnd::SwitchToPeerList(const Peers& peers) { + LayoutConnectUI(false); + + ::SendMessage(listbox_, LB_RESETCONTENT, 0, 0); + + AddListBoxItem(listbox_, "List of currently connected peers:", -1); + Peers::const_iterator i = peers.begin(); + for (; i != peers.end(); ++i) + AddListBoxItem(listbox_, i->second.c_str(), i->first); + + ui_ = LIST_PEERS; + LayoutPeerListUI(true); + ::SetFocus(listbox_); + + if (auto_call_ && peers.begin() != peers.end()) { + // Get the number of items in the list + LRESULT count = ::SendMessage(listbox_, LB_GETCOUNT, 0, 0); + if (count != LB_ERR) { + // Select the last item in the list + LRESULT selection = ::SendMessage(listbox_, LB_SETCURSEL, count - 1, 0); + if (selection != LB_ERR) + ::PostMessage(wnd_, WM_COMMAND, + MAKEWPARAM(GetDlgCtrlID(listbox_), LBN_DBLCLK), + reinterpret_cast<LPARAM>(listbox_)); + } + } +} + +void MainWnd::SwitchToStreamingUI() { + LayoutConnectUI(false); + LayoutPeerListUI(false); + ui_ = STREAMING; +} + +void MainWnd::MessageBox(const char* caption, const char* text, bool is_error) { + DWORD flags = MB_OK; + if (is_error) + flags |= MB_ICONERROR; + + ::MessageBoxA(handle(), text, caption, flags); +} + +void MainWnd::StartLocalRenderer(webrtc::VideoTrackInterface* local_video) { + local_renderer_.reset(new VideoRenderer(handle(), 1, 1, local_video)); +} + +void MainWnd::StopLocalRenderer() { + local_renderer_.reset(); +} + +void MainWnd::StartRemoteRenderer(webrtc::VideoTrackInterface* remote_video) { + remote_renderer_.reset(new VideoRenderer(handle(), 1, 1, remote_video)); +} + +void MainWnd::StopRemoteRenderer() { + remote_renderer_.reset(); +} + +void MainWnd::QueueUIThreadCallback(int msg_id, void* data) { + ::PostThreadMessage(ui_thread_id_, UI_THREAD_CALLBACK, + static_cast<WPARAM>(msg_id), + reinterpret_cast<LPARAM>(data)); +} + +void MainWnd::OnPaint() { + PAINTSTRUCT ps; + ::BeginPaint(handle(), &ps); + + RECT rc; + ::GetClientRect(handle(), &rc); + + VideoRenderer* local_renderer = local_renderer_.get(); + VideoRenderer* remote_renderer = remote_renderer_.get(); + if (ui_ == STREAMING && remote_renderer && local_renderer) { + AutoLock<VideoRenderer> local_lock(local_renderer); + AutoLock<VideoRenderer> remote_lock(remote_renderer); + + const BITMAPINFO& bmi = remote_renderer->bmi(); + int height = abs(bmi.bmiHeader.biHeight); + int width = bmi.bmiHeader.biWidth; + + const uint8_t* image = remote_renderer->image(); + if (image != NULL) { + HDC dc_mem = ::CreateCompatibleDC(ps.hdc); + ::SetStretchBltMode(dc_mem, HALFTONE); + + // Set the map mode so that the ratio will be maintained for us. + HDC all_dc[] = {ps.hdc, dc_mem}; + for (size_t i = 0; i < arraysize(all_dc); ++i) { + SetMapMode(all_dc[i], MM_ISOTROPIC); + SetWindowExtEx(all_dc[i], width, height, NULL); + SetViewportExtEx(all_dc[i], rc.right, rc.bottom, NULL); + } + + HBITMAP bmp_mem = ::CreateCompatibleBitmap(ps.hdc, rc.right, rc.bottom); + HGDIOBJ bmp_old = ::SelectObject(dc_mem, bmp_mem); + + POINT logical_area = {rc.right, rc.bottom}; + DPtoLP(ps.hdc, &logical_area, 1); + + HBRUSH brush = ::CreateSolidBrush(RGB(0, 0, 0)); + RECT logical_rect = {0, 0, logical_area.x, logical_area.y}; + ::FillRect(dc_mem, &logical_rect, brush); + ::DeleteObject(brush); + + int x = (logical_area.x / 2) - (width / 2); + int y = (logical_area.y / 2) - (height / 2); + + StretchDIBits(dc_mem, x, y, width, height, 0, 0, width, height, image, + &bmi, DIB_RGB_COLORS, SRCCOPY); + + if ((rc.right - rc.left) > 200 && (rc.bottom - rc.top) > 200) { + const BITMAPINFO& bmi = local_renderer->bmi(); + image = local_renderer->image(); + int thumb_width = bmi.bmiHeader.biWidth / 4; + int thumb_height = abs(bmi.bmiHeader.biHeight) / 4; + StretchDIBits(dc_mem, logical_area.x - thumb_width - 10, + logical_area.y - thumb_height - 10, thumb_width, + thumb_height, 0, 0, bmi.bmiHeader.biWidth, + -bmi.bmiHeader.biHeight, image, &bmi, DIB_RGB_COLORS, + SRCCOPY); + } + + BitBlt(ps.hdc, 0, 0, logical_area.x, logical_area.y, dc_mem, 0, 0, + SRCCOPY); + + // Cleanup. + ::SelectObject(dc_mem, bmp_old); + ::DeleteObject(bmp_mem); + ::DeleteDC(dc_mem); + } else { + // We're still waiting for the video stream to be initialized. + HBRUSH brush = ::CreateSolidBrush(RGB(0, 0, 0)); + ::FillRect(ps.hdc, &rc, brush); + ::DeleteObject(brush); + + HGDIOBJ old_font = ::SelectObject(ps.hdc, GetDefaultFont()); + ::SetTextColor(ps.hdc, RGB(0xff, 0xff, 0xff)); + ::SetBkMode(ps.hdc, TRANSPARENT); + + std::string text(kConnecting); + if (!local_renderer->image()) { + text += kNoVideoStreams; + } else { + text += kNoIncomingStream; + } + ::DrawTextA(ps.hdc, text.c_str(), -1, &rc, + DT_SINGLELINE | DT_CENTER | DT_VCENTER); + ::SelectObject(ps.hdc, old_font); + } + } else { + HBRUSH brush = ::CreateSolidBrush(::GetSysColor(COLOR_WINDOW)); + ::FillRect(ps.hdc, &rc, brush); + ::DeleteObject(brush); + } + + ::EndPaint(handle(), &ps); +} + +void MainWnd::OnDestroyed() { + PostQuitMessage(0); +} + +void MainWnd::OnDefaultAction() { + if (!callback_) + return; + if (ui_ == CONNECT_TO_SERVER) { + std::string server(GetWindowText(edit1_)); + std::string port_str(GetWindowText(edit2_)); + int port = port_str.length() ? atoi(port_str.c_str()) : 0; + callback_->StartLogin(server, port); + } else if (ui_ == LIST_PEERS) { + LRESULT sel = ::SendMessage(listbox_, LB_GETCURSEL, 0, 0); + if (sel != LB_ERR) { + LRESULT peer_id = ::SendMessage(listbox_, LB_GETITEMDATA, sel, 0); + if (peer_id != -1 && callback_) { + callback_->ConnectToPeer(peer_id); + } + } + } else { + ::MessageBoxA(wnd_, "OK!", "Yeah", MB_OK); + } +} + +bool MainWnd::OnMessage(UINT msg, WPARAM wp, LPARAM lp, LRESULT* result) { + switch (msg) { + case WM_ERASEBKGND: + *result = TRUE; + return true; + + case WM_PAINT: + OnPaint(); + return true; + + case WM_SETFOCUS: + if (ui_ == CONNECT_TO_SERVER) { + SetFocus(edit1_); + } else if (ui_ == LIST_PEERS) { + SetFocus(listbox_); + } + return true; + + case WM_SIZE: + if (ui_ == CONNECT_TO_SERVER) { + LayoutConnectUI(true); + } else if (ui_ == LIST_PEERS) { + LayoutPeerListUI(true); + } + break; + + case WM_CTLCOLORSTATIC: + *result = reinterpret_cast<LRESULT>(GetSysColorBrush(COLOR_WINDOW)); + return true; + + case WM_COMMAND: + if (button_ == reinterpret_cast<HWND>(lp)) { + if (BN_CLICKED == HIWORD(wp)) + OnDefaultAction(); + } else if (listbox_ == reinterpret_cast<HWND>(lp)) { + if (LBN_DBLCLK == HIWORD(wp)) { + OnDefaultAction(); + } + } + return true; + + case WM_CLOSE: + if (callback_) + callback_->Close(); + break; + } + return false; +} + +// static +LRESULT CALLBACK MainWnd::WndProc(HWND hwnd, UINT msg, WPARAM wp, LPARAM lp) { + MainWnd* me = + reinterpret_cast<MainWnd*>(::GetWindowLongPtr(hwnd, GWLP_USERDATA)); + if (!me && WM_CREATE == msg) { + CREATESTRUCT* cs = reinterpret_cast<CREATESTRUCT*>(lp); + me = reinterpret_cast<MainWnd*>(cs->lpCreateParams); + me->wnd_ = hwnd; + ::SetWindowLongPtr(hwnd, GWLP_USERDATA, reinterpret_cast<LONG_PTR>(me)); + } + + LRESULT result = 0; + if (me) { + void* prev_nested_msg = me->nested_msg_; + me->nested_msg_ = &msg; + + bool handled = me->OnMessage(msg, wp, lp, &result); + if (WM_NCDESTROY == msg) { + me->destroyed_ = true; + } else if (!handled) { + result = ::DefWindowProc(hwnd, msg, wp, lp); + } + + if (me->destroyed_ && prev_nested_msg == NULL) { + me->OnDestroyed(); + me->wnd_ = NULL; + me->destroyed_ = false; + } + + me->nested_msg_ = prev_nested_msg; + } else { + result = ::DefWindowProc(hwnd, msg, wp, lp); + } + + return result; +} + +// static +bool MainWnd::RegisterWindowClass() { + if (wnd_class_) + return true; + + WNDCLASSEXW wcex = {sizeof(WNDCLASSEX)}; + wcex.style = CS_DBLCLKS; + wcex.hInstance = GetModuleHandle(NULL); + wcex.hbrBackground = reinterpret_cast<HBRUSH>(COLOR_WINDOW + 1); + wcex.hCursor = ::LoadCursor(NULL, IDC_ARROW); + wcex.lpfnWndProc = &WndProc; + wcex.lpszClassName = kClassName; + wnd_class_ = ::RegisterClassExW(&wcex); + RTC_DCHECK(wnd_class_ != 0); + return wnd_class_ != 0; +} + +void MainWnd::CreateChildWindow(HWND* wnd, + MainWnd::ChildWindowID id, + const wchar_t* class_name, + DWORD control_style, + DWORD ex_style) { + if (::IsWindow(*wnd)) + return; + + // Child windows are invisible at first, and shown after being resized. + DWORD style = WS_CHILD | control_style; + *wnd = ::CreateWindowExW(ex_style, class_name, L"", style, 100, 100, 100, 100, + wnd_, reinterpret_cast<HMENU>(id), + GetModuleHandle(NULL), NULL); + RTC_DCHECK(::IsWindow(*wnd) != FALSE); + ::SendMessage(*wnd, WM_SETFONT, reinterpret_cast<WPARAM>(GetDefaultFont()), + TRUE); +} + +void MainWnd::CreateChildWindows() { + // Create the child windows in tab order. + CreateChildWindow(&label1_, LABEL1_ID, L"Static", ES_CENTER | ES_READONLY, 0); + CreateChildWindow(&edit1_, EDIT_ID, L"Edit", + ES_LEFT | ES_NOHIDESEL | WS_TABSTOP, WS_EX_CLIENTEDGE); + CreateChildWindow(&label2_, LABEL2_ID, L"Static", ES_CENTER | ES_READONLY, 0); + CreateChildWindow(&edit2_, EDIT_ID, L"Edit", + ES_LEFT | ES_NOHIDESEL | WS_TABSTOP, WS_EX_CLIENTEDGE); + CreateChildWindow(&button_, BUTTON_ID, L"Button", BS_CENTER | WS_TABSTOP, 0); + + CreateChildWindow(&listbox_, LISTBOX_ID, L"ListBox", + LBS_HASSTRINGS | LBS_NOTIFY, WS_EX_CLIENTEDGE); + + ::SetWindowTextA(edit1_, server_.c_str()); + ::SetWindowTextA(edit2_, port_.c_str()); +} + +void MainWnd::LayoutConnectUI(bool show) { + struct Windows { + HWND wnd; + const wchar_t* text; + size_t width; + size_t height; + } windows[] = { + {label1_, L"Server"}, {edit1_, L"XXXyyyYYYgggXXXyyyYYYggg"}, + {label2_, L":"}, {edit2_, L"XyXyX"}, + {button_, L"Connect"}, + }; + + if (show) { + const size_t kSeparator = 5; + size_t total_width = (ARRAYSIZE(windows) - 1) * kSeparator; + + for (size_t i = 0; i < ARRAYSIZE(windows); ++i) { + CalculateWindowSizeForText(windows[i].wnd, windows[i].text, + &windows[i].width, &windows[i].height); + total_width += windows[i].width; + } + + RECT rc; + ::GetClientRect(wnd_, &rc); + size_t x = (rc.right / 2) - (total_width / 2); + size_t y = rc.bottom / 2; + for (size_t i = 0; i < ARRAYSIZE(windows); ++i) { + size_t top = y - (windows[i].height / 2); + ::MoveWindow(windows[i].wnd, static_cast<int>(x), static_cast<int>(top), + static_cast<int>(windows[i].width), + static_cast<int>(windows[i].height), TRUE); + x += kSeparator + windows[i].width; + if (windows[i].text[0] != 'X') + ::SetWindowTextW(windows[i].wnd, windows[i].text); + ::ShowWindow(windows[i].wnd, SW_SHOWNA); + } + } else { + for (size_t i = 0; i < ARRAYSIZE(windows); ++i) { + ::ShowWindow(windows[i].wnd, SW_HIDE); + } + } +} + +void MainWnd::LayoutPeerListUI(bool show) { + if (show) { + RECT rc; + ::GetClientRect(wnd_, &rc); + ::MoveWindow(listbox_, 0, 0, rc.right, rc.bottom, TRUE); + ::ShowWindow(listbox_, SW_SHOWNA); + } else { + ::ShowWindow(listbox_, SW_HIDE); + InvalidateRect(wnd_, NULL, TRUE); + } +} + +void MainWnd::HandleTabbing() { + bool shift = ((::GetAsyncKeyState(VK_SHIFT) & 0x8000) != 0); + UINT next_cmd = shift ? GW_HWNDPREV : GW_HWNDNEXT; + UINT loop_around_cmd = shift ? GW_HWNDLAST : GW_HWNDFIRST; + HWND focus = GetFocus(), next; + do { + next = ::GetWindow(focus, next_cmd); + if (IsWindowVisible(next) && + (GetWindowLong(next, GWL_STYLE) & WS_TABSTOP)) { + break; + } + + if (!next) { + next = ::GetWindow(focus, loop_around_cmd); + if (IsWindowVisible(next) && + (GetWindowLong(next, GWL_STYLE) & WS_TABSTOP)) { + break; + } + } + focus = next; + } while (true); + ::SetFocus(next); +} + +// +// MainWnd::VideoRenderer +// + +MainWnd::VideoRenderer::VideoRenderer( + HWND wnd, + int width, + int height, + webrtc::VideoTrackInterface* track_to_render) + : wnd_(wnd), rendered_track_(track_to_render) { + ::InitializeCriticalSection(&buffer_lock_); + ZeroMemory(&bmi_, sizeof(bmi_)); + bmi_.bmiHeader.biSize = sizeof(BITMAPINFOHEADER); + bmi_.bmiHeader.biPlanes = 1; + bmi_.bmiHeader.biBitCount = 32; + bmi_.bmiHeader.biCompression = BI_RGB; + bmi_.bmiHeader.biWidth = width; + bmi_.bmiHeader.biHeight = -height; + bmi_.bmiHeader.biSizeImage = + width * height * (bmi_.bmiHeader.biBitCount >> 3); + rendered_track_->AddOrUpdateSink(this, rtc::VideoSinkWants()); +} + +MainWnd::VideoRenderer::~VideoRenderer() { + rendered_track_->RemoveSink(this); + ::DeleteCriticalSection(&buffer_lock_); +} + +void MainWnd::VideoRenderer::SetSize(int width, int height) { + AutoLock<VideoRenderer> lock(this); + + if (width == bmi_.bmiHeader.biWidth && height == bmi_.bmiHeader.biHeight) { + return; + } + + bmi_.bmiHeader.biWidth = width; + bmi_.bmiHeader.biHeight = -height; + bmi_.bmiHeader.biSizeImage = + width * height * (bmi_.bmiHeader.biBitCount >> 3); + image_.reset(new uint8_t[bmi_.bmiHeader.biSizeImage]); +} + +void MainWnd::VideoRenderer::OnFrame(const webrtc::VideoFrame& video_frame) { + { + AutoLock<VideoRenderer> lock(this); + + rtc::scoped_refptr<webrtc::I420BufferInterface> buffer( + video_frame.video_frame_buffer()->ToI420()); + if (video_frame.rotation() != webrtc::kVideoRotation_0) { + buffer = webrtc::I420Buffer::Rotate(*buffer, video_frame.rotation()); + } + + SetSize(buffer->width(), buffer->height()); + + RTC_DCHECK(image_.get() != NULL); + libyuv::I420ToARGB(buffer->DataY(), buffer->StrideY(), buffer->DataU(), + buffer->StrideU(), buffer->DataV(), buffer->StrideV(), + image_.get(), + bmi_.bmiHeader.biWidth * bmi_.bmiHeader.biBitCount / 8, + buffer->width(), buffer->height()); + } + InvalidateRect(wnd_, NULL, TRUE); +} diff --git a/third_party/libwebrtc/examples/peerconnection/client/main_wnd.h b/third_party/libwebrtc/examples/peerconnection/client/main_wnd.h new file mode 100644 index 0000000000..898fea9d92 --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/client/main_wnd.h @@ -0,0 +1,206 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef EXAMPLES_PEERCONNECTION_CLIENT_MAIN_WND_H_ +#define EXAMPLES_PEERCONNECTION_CLIENT_MAIN_WND_H_ + +#include <map> +#include <memory> +#include <string> + +#include "api/media_stream_interface.h" +#include "api/video/video_frame.h" +#include "examples/peerconnection/client/peer_connection_client.h" +#include "media/base/media_channel.h" +#include "media/base/video_common.h" +#if defined(WEBRTC_WIN) +#include "rtc_base/win32.h" +#endif // WEBRTC_WIN + +class MainWndCallback { + public: + virtual void StartLogin(const std::string& server, int port) = 0; + virtual void DisconnectFromServer() = 0; + virtual void ConnectToPeer(int peer_id) = 0; + virtual void DisconnectFromCurrentPeer() = 0; + virtual void UIThreadCallback(int msg_id, void* data) = 0; + virtual void Close() = 0; + + protected: + virtual ~MainWndCallback() {} +}; + +// Pure virtual interface for the main window. +class MainWindow { + public: + virtual ~MainWindow() {} + + enum UI { + CONNECT_TO_SERVER, + LIST_PEERS, + STREAMING, + }; + + virtual void RegisterObserver(MainWndCallback* callback) = 0; + + virtual bool IsWindow() = 0; + virtual void MessageBox(const char* caption, + const char* text, + bool is_error) = 0; + + virtual UI current_ui() = 0; + + virtual void SwitchToConnectUI() = 0; + virtual void SwitchToPeerList(const Peers& peers) = 0; + virtual void SwitchToStreamingUI() = 0; + + virtual void StartLocalRenderer(webrtc::VideoTrackInterface* local_video) = 0; + virtual void StopLocalRenderer() = 0; + virtual void StartRemoteRenderer( + webrtc::VideoTrackInterface* remote_video) = 0; + virtual void StopRemoteRenderer() = 0; + + virtual void QueueUIThreadCallback(int msg_id, void* data) = 0; +}; + +#ifdef WIN32 + +class MainWnd : public MainWindow { + public: + static const wchar_t kClassName[]; + + enum WindowMessages { + UI_THREAD_CALLBACK = WM_APP + 1, + }; + + MainWnd(const char* server, int port, bool auto_connect, bool auto_call); + ~MainWnd(); + + bool Create(); + bool Destroy(); + bool PreTranslateMessage(MSG* msg); + + virtual void RegisterObserver(MainWndCallback* callback); + virtual bool IsWindow(); + virtual void SwitchToConnectUI(); + virtual void SwitchToPeerList(const Peers& peers); + virtual void SwitchToStreamingUI(); + virtual void MessageBox(const char* caption, const char* text, bool is_error); + virtual UI current_ui() { return ui_; } + + virtual void StartLocalRenderer(webrtc::VideoTrackInterface* local_video); + virtual void StopLocalRenderer(); + virtual void StartRemoteRenderer(webrtc::VideoTrackInterface* remote_video); + virtual void StopRemoteRenderer(); + + virtual void QueueUIThreadCallback(int msg_id, void* data); + + HWND handle() const { return wnd_; } + + class VideoRenderer : public rtc::VideoSinkInterface<webrtc::VideoFrame> { + public: + VideoRenderer(HWND wnd, + int width, + int height, + webrtc::VideoTrackInterface* track_to_render); + virtual ~VideoRenderer(); + + void Lock() { ::EnterCriticalSection(&buffer_lock_); } + + void Unlock() { ::LeaveCriticalSection(&buffer_lock_); } + + // VideoSinkInterface implementation + void OnFrame(const webrtc::VideoFrame& frame) override; + + const BITMAPINFO& bmi() const { return bmi_; } + const uint8_t* image() const { return image_.get(); } + + protected: + void SetSize(int width, int height); + + enum { + SET_SIZE, + RENDER_FRAME, + }; + + HWND wnd_; + BITMAPINFO bmi_; + std::unique_ptr<uint8_t[]> image_; + CRITICAL_SECTION buffer_lock_; + rtc::scoped_refptr<webrtc::VideoTrackInterface> rendered_track_; + }; + + // A little helper class to make sure we always to proper locking and + // unlocking when working with VideoRenderer buffers. + template <typename T> + class AutoLock { + public: + explicit AutoLock(T* obj) : obj_(obj) { obj_->Lock(); } + ~AutoLock() { obj_->Unlock(); } + + protected: + T* obj_; + }; + + protected: + enum ChildWindowID { + EDIT_ID = 1, + BUTTON_ID, + LABEL1_ID, + LABEL2_ID, + LISTBOX_ID, + }; + + void OnPaint(); + void OnDestroyed(); + + void OnDefaultAction(); + + bool OnMessage(UINT msg, WPARAM wp, LPARAM lp, LRESULT* result); + + static LRESULT CALLBACK WndProc(HWND hwnd, UINT msg, WPARAM wp, LPARAM lp); + static bool RegisterWindowClass(); + + void CreateChildWindow(HWND* wnd, + ChildWindowID id, + const wchar_t* class_name, + DWORD control_style, + DWORD ex_style); + void CreateChildWindows(); + + void LayoutConnectUI(bool show); + void LayoutPeerListUI(bool show); + + void HandleTabbing(); + + private: + std::unique_ptr<VideoRenderer> local_renderer_; + std::unique_ptr<VideoRenderer> remote_renderer_; + UI ui_; + HWND wnd_; + DWORD ui_thread_id_; + HWND edit1_; + HWND edit2_; + HWND label1_; + HWND label2_; + HWND button_; + HWND listbox_; + bool destroyed_; + void* nested_msg_; + MainWndCallback* callback_; + static ATOM wnd_class_; + std::string server_; + std::string port_; + bool auto_connect_; + bool auto_call_; +}; +#endif // WIN32 + +#endif // EXAMPLES_PEERCONNECTION_CLIENT_MAIN_WND_H_ diff --git a/third_party/libwebrtc/examples/peerconnection/client/peer_connection_client.cc b/third_party/libwebrtc/examples/peerconnection/client/peer_connection_client.cc new file mode 100644 index 0000000000..2746752d80 --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/client/peer_connection_client.cc @@ -0,0 +1,489 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/peerconnection/client/peer_connection_client.h" + +#include "api/units/time_delta.h" +#include "examples/peerconnection/client/defaults.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" +#include "rtc_base/net_helpers.h" + +namespace { + +// This is our magical hangup signal. +constexpr char kByeMessage[] = "BYE"; +// Delay between server connection retries, in milliseconds +constexpr webrtc::TimeDelta kReconnectDelay = webrtc::TimeDelta::Seconds(2); + +rtc::Socket* CreateClientSocket(int family) { + rtc::Thread* thread = rtc::Thread::Current(); + RTC_DCHECK(thread != NULL); + return thread->socketserver()->CreateSocket(family, SOCK_STREAM); +} + +} // namespace + +PeerConnectionClient::PeerConnectionClient() + : callback_(NULL), resolver_(NULL), state_(NOT_CONNECTED), my_id_(-1) {} + +PeerConnectionClient::~PeerConnectionClient() = default; + +void PeerConnectionClient::InitSocketSignals() { + RTC_DCHECK(control_socket_.get() != NULL); + RTC_DCHECK(hanging_get_.get() != NULL); + control_socket_->SignalCloseEvent.connect(this, + &PeerConnectionClient::OnClose); + hanging_get_->SignalCloseEvent.connect(this, &PeerConnectionClient::OnClose); + control_socket_->SignalConnectEvent.connect(this, + &PeerConnectionClient::OnConnect); + hanging_get_->SignalConnectEvent.connect( + this, &PeerConnectionClient::OnHangingGetConnect); + control_socket_->SignalReadEvent.connect(this, &PeerConnectionClient::OnRead); + hanging_get_->SignalReadEvent.connect( + this, &PeerConnectionClient::OnHangingGetRead); +} + +int PeerConnectionClient::id() const { + return my_id_; +} + +bool PeerConnectionClient::is_connected() const { + return my_id_ != -1; +} + +const Peers& PeerConnectionClient::peers() const { + return peers_; +} + +void PeerConnectionClient::RegisterObserver( + PeerConnectionClientObserver* callback) { + RTC_DCHECK(!callback_); + callback_ = callback; +} + +void PeerConnectionClient::Connect(const std::string& server, + int port, + const std::string& client_name) { + RTC_DCHECK(!server.empty()); + RTC_DCHECK(!client_name.empty()); + + if (state_ != NOT_CONNECTED) { + RTC_LOG(LS_WARNING) + << "The client must not be connected before you can call Connect()"; + callback_->OnServerConnectionFailure(); + return; + } + + if (server.empty() || client_name.empty()) { + callback_->OnServerConnectionFailure(); + return; + } + + if (port <= 0) + port = kDefaultServerPort; + + server_address_.SetIP(server); + server_address_.SetPort(port); + client_name_ = client_name; + + if (server_address_.IsUnresolvedIP()) { + state_ = RESOLVING; + resolver_ = new rtc::AsyncResolver(); + resolver_->SignalDone.connect(this, &PeerConnectionClient::OnResolveResult); + resolver_->Start(server_address_); + } else { + DoConnect(); + } +} + +void PeerConnectionClient::OnResolveResult( + rtc::AsyncResolverInterface* resolver) { + if (resolver_->GetError() != 0) { + callback_->OnServerConnectionFailure(); + resolver_->Destroy(false); + resolver_ = NULL; + state_ = NOT_CONNECTED; + } else { + server_address_ = resolver_->address(); + DoConnect(); + } +} + +void PeerConnectionClient::DoConnect() { + control_socket_.reset(CreateClientSocket(server_address_.ipaddr().family())); + hanging_get_.reset(CreateClientSocket(server_address_.ipaddr().family())); + InitSocketSignals(); + char buffer[1024]; + snprintf(buffer, sizeof(buffer), "GET /sign_in?%s HTTP/1.0\r\n\r\n", + client_name_.c_str()); + onconnect_data_ = buffer; + + bool ret = ConnectControlSocket(); + if (ret) + state_ = SIGNING_IN; + if (!ret) { + callback_->OnServerConnectionFailure(); + } +} + +bool PeerConnectionClient::SendToPeer(int peer_id, const std::string& message) { + if (state_ != CONNECTED) + return false; + + RTC_DCHECK(is_connected()); + RTC_DCHECK(control_socket_->GetState() == rtc::Socket::CS_CLOSED); + if (!is_connected() || peer_id == -1) + return false; + + char headers[1024]; + snprintf(headers, sizeof(headers), + "POST /message?peer_id=%i&to=%i HTTP/1.0\r\n" + "Content-Length: %zu\r\n" + "Content-Type: text/plain\r\n" + "\r\n", + my_id_, peer_id, message.length()); + onconnect_data_ = headers; + onconnect_data_ += message; + return ConnectControlSocket(); +} + +bool PeerConnectionClient::SendHangUp(int peer_id) { + return SendToPeer(peer_id, kByeMessage); +} + +bool PeerConnectionClient::IsSendingMessage() { + return state_ == CONNECTED && + control_socket_->GetState() != rtc::Socket::CS_CLOSED; +} + +bool PeerConnectionClient::SignOut() { + if (state_ == NOT_CONNECTED || state_ == SIGNING_OUT) + return true; + + if (hanging_get_->GetState() != rtc::Socket::CS_CLOSED) + hanging_get_->Close(); + + if (control_socket_->GetState() == rtc::Socket::CS_CLOSED) { + state_ = SIGNING_OUT; + + if (my_id_ != -1) { + char buffer[1024]; + snprintf(buffer, sizeof(buffer), + "GET /sign_out?peer_id=%i HTTP/1.0\r\n\r\n", my_id_); + onconnect_data_ = buffer; + return ConnectControlSocket(); + } else { + // Can occur if the app is closed before we finish connecting. + return true; + } + } else { + state_ = SIGNING_OUT_WAITING; + } + + return true; +} + +void PeerConnectionClient::Close() { + control_socket_->Close(); + hanging_get_->Close(); + onconnect_data_.clear(); + peers_.clear(); + if (resolver_ != NULL) { + resolver_->Destroy(false); + resolver_ = NULL; + } + my_id_ = -1; + state_ = NOT_CONNECTED; +} + +bool PeerConnectionClient::ConnectControlSocket() { + RTC_DCHECK(control_socket_->GetState() == rtc::Socket::CS_CLOSED); + int err = control_socket_->Connect(server_address_); + if (err == SOCKET_ERROR) { + Close(); + return false; + } + return true; +} + +void PeerConnectionClient::OnConnect(rtc::Socket* socket) { + RTC_DCHECK(!onconnect_data_.empty()); + size_t sent = socket->Send(onconnect_data_.c_str(), onconnect_data_.length()); + RTC_DCHECK(sent == onconnect_data_.length()); + onconnect_data_.clear(); +} + +void PeerConnectionClient::OnHangingGetConnect(rtc::Socket* socket) { + char buffer[1024]; + snprintf(buffer, sizeof(buffer), "GET /wait?peer_id=%i HTTP/1.0\r\n\r\n", + my_id_); + int len = static_cast<int>(strlen(buffer)); + int sent = socket->Send(buffer, len); + RTC_DCHECK(sent == len); +} + +void PeerConnectionClient::OnMessageFromPeer(int peer_id, + const std::string& message) { + if (message.length() == (sizeof(kByeMessage) - 1) && + message.compare(kByeMessage) == 0) { + callback_->OnPeerDisconnected(peer_id); + } else { + callback_->OnMessageFromPeer(peer_id, message); + } +} + +bool PeerConnectionClient::GetHeaderValue(const std::string& data, + size_t eoh, + const char* header_pattern, + size_t* value) { + RTC_DCHECK(value != NULL); + size_t found = data.find(header_pattern); + if (found != std::string::npos && found < eoh) { + *value = atoi(&data[found + strlen(header_pattern)]); + return true; + } + return false; +} + +bool PeerConnectionClient::GetHeaderValue(const std::string& data, + size_t eoh, + const char* header_pattern, + std::string* value) { + RTC_DCHECK(value != NULL); + size_t found = data.find(header_pattern); + if (found != std::string::npos && found < eoh) { + size_t begin = found + strlen(header_pattern); + size_t end = data.find("\r\n", begin); + if (end == std::string::npos) + end = eoh; + value->assign(data.substr(begin, end - begin)); + return true; + } + return false; +} + +bool PeerConnectionClient::ReadIntoBuffer(rtc::Socket* socket, + std::string* data, + size_t* content_length) { + char buffer[0xffff]; + do { + int bytes = socket->Recv(buffer, sizeof(buffer), nullptr); + if (bytes <= 0) + break; + data->append(buffer, bytes); + } while (true); + + bool ret = false; + size_t i = data->find("\r\n\r\n"); + if (i != std::string::npos) { + RTC_LOG(LS_INFO) << "Headers received"; + if (GetHeaderValue(*data, i, "\r\nContent-Length: ", content_length)) { + size_t total_response_size = (i + 4) + *content_length; + if (data->length() >= total_response_size) { + ret = true; + std::string should_close; + const char kConnection[] = "\r\nConnection: "; + if (GetHeaderValue(*data, i, kConnection, &should_close) && + should_close.compare("close") == 0) { + socket->Close(); + // Since we closed the socket, there was no notification delivered + // to us. Compensate by letting ourselves know. + OnClose(socket, 0); + } + } else { + // We haven't received everything. Just continue to accept data. + } + } else { + RTC_LOG(LS_ERROR) << "No content length field specified by the server."; + } + } + return ret; +} + +void PeerConnectionClient::OnRead(rtc::Socket* socket) { + size_t content_length = 0; + if (ReadIntoBuffer(socket, &control_data_, &content_length)) { + size_t peer_id = 0, eoh = 0; + bool ok = + ParseServerResponse(control_data_, content_length, &peer_id, &eoh); + if (ok) { + if (my_id_ == -1) { + // First response. Let's store our server assigned ID. + RTC_DCHECK(state_ == SIGNING_IN); + my_id_ = static_cast<int>(peer_id); + RTC_DCHECK(my_id_ != -1); + + // The body of the response will be a list of already connected peers. + if (content_length) { + size_t pos = eoh + 4; + while (pos < control_data_.size()) { + size_t eol = control_data_.find('\n', pos); + if (eol == std::string::npos) + break; + int id = 0; + std::string name; + bool connected; + if (ParseEntry(control_data_.substr(pos, eol - pos), &name, &id, + &connected) && + id != my_id_) { + peers_[id] = name; + callback_->OnPeerConnected(id, name); + } + pos = eol + 1; + } + } + RTC_DCHECK(is_connected()); + callback_->OnSignedIn(); + } else if (state_ == SIGNING_OUT) { + Close(); + callback_->OnDisconnected(); + } else if (state_ == SIGNING_OUT_WAITING) { + SignOut(); + } + } + + control_data_.clear(); + + if (state_ == SIGNING_IN) { + RTC_DCHECK(hanging_get_->GetState() == rtc::Socket::CS_CLOSED); + state_ = CONNECTED; + hanging_get_->Connect(server_address_); + } + } +} + +void PeerConnectionClient::OnHangingGetRead(rtc::Socket* socket) { + RTC_LOG(LS_INFO) << __FUNCTION__; + size_t content_length = 0; + if (ReadIntoBuffer(socket, ¬ification_data_, &content_length)) { + size_t peer_id = 0, eoh = 0; + bool ok = + ParseServerResponse(notification_data_, content_length, &peer_id, &eoh); + + if (ok) { + // Store the position where the body begins. + size_t pos = eoh + 4; + + if (my_id_ == static_cast<int>(peer_id)) { + // A notification about a new member or a member that just + // disconnected. + int id = 0; + std::string name; + bool connected = false; + if (ParseEntry(notification_data_.substr(pos), &name, &id, + &connected)) { + if (connected) { + peers_[id] = name; + callback_->OnPeerConnected(id, name); + } else { + peers_.erase(id); + callback_->OnPeerDisconnected(id); + } + } + } else { + OnMessageFromPeer(static_cast<int>(peer_id), + notification_data_.substr(pos)); + } + } + + notification_data_.clear(); + } + + if (hanging_get_->GetState() == rtc::Socket::CS_CLOSED && + state_ == CONNECTED) { + hanging_get_->Connect(server_address_); + } +} + +bool PeerConnectionClient::ParseEntry(const std::string& entry, + std::string* name, + int* id, + bool* connected) { + RTC_DCHECK(name != NULL); + RTC_DCHECK(id != NULL); + RTC_DCHECK(connected != NULL); + RTC_DCHECK(!entry.empty()); + + *connected = false; + size_t separator = entry.find(','); + if (separator != std::string::npos) { + *id = atoi(&entry[separator + 1]); + name->assign(entry.substr(0, separator)); + separator = entry.find(',', separator + 1); + if (separator != std::string::npos) { + *connected = atoi(&entry[separator + 1]) ? true : false; + } + } + return !name->empty(); +} + +int PeerConnectionClient::GetResponseStatus(const std::string& response) { + int status = -1; + size_t pos = response.find(' '); + if (pos != std::string::npos) + status = atoi(&response[pos + 1]); + return status; +} + +bool PeerConnectionClient::ParseServerResponse(const std::string& response, + size_t content_length, + size_t* peer_id, + size_t* eoh) { + int status = GetResponseStatus(response.c_str()); + if (status != 200) { + RTC_LOG(LS_ERROR) << "Received error from server"; + Close(); + callback_->OnDisconnected(); + return false; + } + + *eoh = response.find("\r\n\r\n"); + RTC_DCHECK(*eoh != std::string::npos); + if (*eoh == std::string::npos) + return false; + + *peer_id = -1; + + // See comment in peer_channel.cc for why we use the Pragma header. + GetHeaderValue(response, *eoh, "\r\nPragma: ", peer_id); + + return true; +} + +void PeerConnectionClient::OnClose(rtc::Socket* socket, int err) { + RTC_LOG(LS_INFO) << __FUNCTION__; + + socket->Close(); + +#ifdef WIN32 + if (err != WSAECONNREFUSED) { +#else + if (err != ECONNREFUSED) { +#endif + if (socket == hanging_get_.get()) { + if (state_ == CONNECTED) { + hanging_get_->Close(); + hanging_get_->Connect(server_address_); + } + } else { + callback_->OnMessageSent(err); + } + } else { + if (socket == control_socket_.get()) { + RTC_LOG(LS_WARNING) << "Connection refused; retrying in 2 seconds"; + rtc::Thread::Current()->PostDelayedTask( + SafeTask(safety_.flag(), [this] { DoConnect(); }), kReconnectDelay); + } else { + Close(); + callback_->OnDisconnected(); + } + } +} diff --git a/third_party/libwebrtc/examples/peerconnection/client/peer_connection_client.h b/third_party/libwebrtc/examples/peerconnection/client/peer_connection_client.h new file mode 100644 index 0000000000..8f9c5b6a75 --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/client/peer_connection_client.h @@ -0,0 +1,129 @@ +/* + * Copyright 2011 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef EXAMPLES_PEERCONNECTION_CLIENT_PEER_CONNECTION_CLIENT_H_ +#define EXAMPLES_PEERCONNECTION_CLIENT_PEER_CONNECTION_CLIENT_H_ + +#include <map> +#include <memory> +#include <string> + +#include "api/task_queue/pending_task_safety_flag.h" +#include "rtc_base/net_helpers.h" +#include "rtc_base/physical_socket_server.h" +#include "rtc_base/third_party/sigslot/sigslot.h" + +typedef std::map<int, std::string> Peers; + +struct PeerConnectionClientObserver { + virtual void OnSignedIn() = 0; // Called when we're logged on. + virtual void OnDisconnected() = 0; + virtual void OnPeerConnected(int id, const std::string& name) = 0; + virtual void OnPeerDisconnected(int peer_id) = 0; + virtual void OnMessageFromPeer(int peer_id, const std::string& message) = 0; + virtual void OnMessageSent(int err) = 0; + virtual void OnServerConnectionFailure() = 0; + + protected: + virtual ~PeerConnectionClientObserver() {} +}; + +class PeerConnectionClient : public sigslot::has_slots<> { + public: + enum State { + NOT_CONNECTED, + RESOLVING, + SIGNING_IN, + CONNECTED, + SIGNING_OUT_WAITING, + SIGNING_OUT, + }; + + PeerConnectionClient(); + ~PeerConnectionClient(); + + int id() const; + bool is_connected() const; + const Peers& peers() const; + + void RegisterObserver(PeerConnectionClientObserver* callback); + + void Connect(const std::string& server, + int port, + const std::string& client_name); + + bool SendToPeer(int peer_id, const std::string& message); + bool SendHangUp(int peer_id); + bool IsSendingMessage(); + + bool SignOut(); + + protected: + void DoConnect(); + void Close(); + void InitSocketSignals(); + bool ConnectControlSocket(); + void OnConnect(rtc::Socket* socket); + void OnHangingGetConnect(rtc::Socket* socket); + void OnMessageFromPeer(int peer_id, const std::string& message); + + // Quick and dirty support for parsing HTTP header values. + bool GetHeaderValue(const std::string& data, + size_t eoh, + const char* header_pattern, + size_t* value); + + bool GetHeaderValue(const std::string& data, + size_t eoh, + const char* header_pattern, + std::string* value); + + // Returns true if the whole response has been read. + bool ReadIntoBuffer(rtc::Socket* socket, + std::string* data, + size_t* content_length); + + void OnRead(rtc::Socket* socket); + + void OnHangingGetRead(rtc::Socket* socket); + + // Parses a single line entry in the form "<name>,<id>,<connected>" + bool ParseEntry(const std::string& entry, + std::string* name, + int* id, + bool* connected); + + int GetResponseStatus(const std::string& response); + + bool ParseServerResponse(const std::string& response, + size_t content_length, + size_t* peer_id, + size_t* eoh); + + void OnClose(rtc::Socket* socket, int err); + + void OnResolveResult(rtc::AsyncResolverInterface* resolver); + + PeerConnectionClientObserver* callback_; + rtc::SocketAddress server_address_; + rtc::AsyncResolver* resolver_; + std::unique_ptr<rtc::Socket> control_socket_; + std::unique_ptr<rtc::Socket> hanging_get_; + std::string onconnect_data_; + std::string control_data_; + std::string notification_data_; + std::string client_name_; + Peers peers_; + State state_; + int my_id_; + webrtc::ScopedTaskSafety safety_; +}; + +#endif // EXAMPLES_PEERCONNECTION_CLIENT_PEER_CONNECTION_CLIENT_H_ diff --git a/third_party/libwebrtc/examples/peerconnection/server/data_socket.cc b/third_party/libwebrtc/examples/peerconnection/server/data_socket.cc new file mode 100644 index 0000000000..855ebd8c0c --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/server/data_socket.cc @@ -0,0 +1,299 @@ +/* + * Copyright 2011 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/peerconnection/server/data_socket.h" + +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#if defined(WEBRTC_POSIX) +#include <unistd.h> +#endif + +#include "examples/peerconnection/server/utils.h" +#include "rtc_base/checks.h" + +static const char kHeaderTerminator[] = "\r\n\r\n"; +static const int kHeaderTerminatorLength = sizeof(kHeaderTerminator) - 1; + +// static +const char DataSocket::kCrossOriginAllowHeaders[] = + "Access-Control-Allow-Origin: *\r\n" + "Access-Control-Allow-Credentials: true\r\n" + "Access-Control-Allow-Methods: POST, GET, OPTIONS\r\n" + "Access-Control-Allow-Headers: Content-Type, " + "Content-Length, Connection, Cache-Control\r\n" + "Access-Control-Expose-Headers: Content-Length\r\n"; + +#if defined(WIN32) +class WinsockInitializer { + static WinsockInitializer singleton; + + WinsockInitializer() { + WSADATA data; + WSAStartup(MAKEWORD(1, 0), &data); + } + + public: + ~WinsockInitializer() { WSACleanup(); } +}; +WinsockInitializer WinsockInitializer::singleton; +#endif + +// +// SocketBase +// + +bool SocketBase::Create() { + RTC_DCHECK(!valid()); + socket_ = ::socket(AF_INET, SOCK_STREAM, 0); + return valid(); +} + +void SocketBase::Close() { + if (socket_ != INVALID_SOCKET) { + closesocket(socket_); + socket_ = INVALID_SOCKET; + } +} + +// +// DataSocket +// + +std::string DataSocket::request_arguments() const { + size_t args = request_path_.find('?'); + if (args != std::string::npos) + return request_path_.substr(args + 1); + return ""; +} + +bool DataSocket::PathEquals(const char* path) const { + RTC_DCHECK(path); + size_t args = request_path_.find('?'); + if (args != std::string::npos) + return request_path_.substr(0, args).compare(path) == 0; + return request_path_.compare(path) == 0; +} + +bool DataSocket::OnDataAvailable(bool* close_socket) { + RTC_DCHECK(valid()); + char buffer[0xfff] = {0}; + int bytes = recv(socket_, buffer, sizeof(buffer), 0); + if (bytes == SOCKET_ERROR || bytes == 0) { + *close_socket = true; + return false; + } + + *close_socket = false; + + bool ret = true; + if (headers_received()) { + if (method_ != POST) { + // unexpectedly received data. + ret = false; + } else { + data_.append(buffer, bytes); + } + } else { + request_headers_.append(buffer, bytes); + size_t found = request_headers_.find(kHeaderTerminator); + if (found != std::string::npos) { + data_ = request_headers_.substr(found + kHeaderTerminatorLength); + request_headers_.resize(found + kHeaderTerminatorLength); + ret = ParseHeaders(); + } + } + return ret; +} + +bool DataSocket::Send(const std::string& data) const { + return send(socket_, data.data(), static_cast<int>(data.length()), 0) != + SOCKET_ERROR; +} + +bool DataSocket::Send(const std::string& status, + bool connection_close, + const std::string& content_type, + const std::string& extra_headers, + const std::string& data) const { + RTC_DCHECK(valid()); + RTC_DCHECK(!status.empty()); + std::string buffer("HTTP/1.1 " + status + "\r\n"); + + buffer += + "Server: PeerConnectionTestServer/0.1\r\n" + "Cache-Control: no-cache\r\n"; + + if (connection_close) + buffer += "Connection: close\r\n"; + + if (!content_type.empty()) + buffer += "Content-Type: " + content_type + "\r\n"; + + buffer += + "Content-Length: " + int2str(static_cast<int>(data.size())) + "\r\n"; + + if (!extra_headers.empty()) { + buffer += extra_headers; + // Extra headers are assumed to have a separator per header. + } + + buffer += kCrossOriginAllowHeaders; + + buffer += "\r\n"; + buffer += data; + + return Send(buffer); +} + +void DataSocket::Clear() { + method_ = INVALID; + content_length_ = 0; + content_type_.clear(); + request_path_.clear(); + request_headers_.clear(); + data_.clear(); +} + +bool DataSocket::ParseHeaders() { + RTC_DCHECK(!request_headers_.empty()); + RTC_DCHECK_EQ(method_, INVALID); + size_t i = request_headers_.find("\r\n"); + if (i == std::string::npos) + return false; + + if (!ParseMethodAndPath(request_headers_.data(), i)) + return false; + + RTC_DCHECK_NE(method_, INVALID); + RTC_DCHECK(!request_path_.empty()); + + if (method_ == POST) { + const char* headers = request_headers_.data() + i + 2; + size_t len = request_headers_.length() - i - 2; + if (!ParseContentLengthAndType(headers, len)) + return false; + } + + return true; +} + +bool DataSocket::ParseMethodAndPath(const char* begin, size_t len) { + struct { + const char* method_name; + size_t method_name_len; + RequestMethod id; + } supported_methods[] = { + {"GET", 3, GET}, + {"POST", 4, POST}, + {"OPTIONS", 7, OPTIONS}, + }; + + const char* path = NULL; + for (size_t i = 0; i < ARRAYSIZE(supported_methods); ++i) { + if (len > supported_methods[i].method_name_len && + isspace(begin[supported_methods[i].method_name_len]) && + strncmp(begin, supported_methods[i].method_name, + supported_methods[i].method_name_len) == 0) { + method_ = supported_methods[i].id; + path = begin + supported_methods[i].method_name_len; + break; + } + } + + const char* end = begin + len; + if (!path || path >= end) + return false; + + ++path; + begin = path; + while (!isspace(*path) && path < end) + ++path; + + request_path_.assign(begin, path - begin); + + return true; +} + +bool DataSocket::ParseContentLengthAndType(const char* headers, size_t length) { + RTC_DCHECK_EQ(content_length_, 0); + RTC_DCHECK(content_type_.empty()); + + const char* end = headers + length; + while (headers && headers < end) { + if (!isspace(headers[0])) { + static const char kContentLength[] = "Content-Length:"; + static const char kContentType[] = "Content-Type:"; + if ((headers + ARRAYSIZE(kContentLength)) < end && + strncmp(headers, kContentLength, ARRAYSIZE(kContentLength) - 1) == + 0) { + headers += ARRAYSIZE(kContentLength) - 1; + while (headers[0] == ' ') + ++headers; + content_length_ = atoi(headers); + } else if ((headers + ARRAYSIZE(kContentType)) < end && + strncmp(headers, kContentType, ARRAYSIZE(kContentType) - 1) == + 0) { + headers += ARRAYSIZE(kContentType) - 1; + while (headers[0] == ' ') + ++headers; + const char* type_end = strstr(headers, "\r\n"); + if (type_end == NULL) + type_end = end; + content_type_.assign(headers, type_end); + } + } else { + ++headers; + } + headers = strstr(headers, "\r\n"); + if (headers) + headers += 2; + } + + return !content_type_.empty() && content_length_ != 0; +} + +// +// ListeningSocket +// + +bool ListeningSocket::Listen(unsigned short port) { + RTC_DCHECK(valid()); + int enabled = 1; + if (setsockopt(socket_, SOL_SOCKET, SO_REUSEADDR, + reinterpret_cast<const char*>(&enabled), + sizeof(enabled)) != 0) { + printf("setsockopt failed\n"); + return false; + } + struct sockaddr_in addr = {0}; + addr.sin_family = AF_INET; + addr.sin_addr.s_addr = htonl(INADDR_ANY); + addr.sin_port = htons(port); + if (bind(socket_, reinterpret_cast<const sockaddr*>(&addr), sizeof(addr)) == + SOCKET_ERROR) { + printf("bind failed\n"); + return false; + } + return listen(socket_, 5) != SOCKET_ERROR; +} + +DataSocket* ListeningSocket::Accept() const { + RTC_DCHECK(valid()); + struct sockaddr_in addr = {0}; + socklen_t size = sizeof(addr); + NativeSocket client = + accept(socket_, reinterpret_cast<sockaddr*>(&addr), &size); + if (client == INVALID_SOCKET) + return NULL; + + return new DataSocket(client); +} diff --git a/third_party/libwebrtc/examples/peerconnection/server/data_socket.h b/third_party/libwebrtc/examples/peerconnection/server/data_socket.h new file mode 100644 index 0000000000..57ad5b9aee --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/server/data_socket.h @@ -0,0 +1,152 @@ +/* + * Copyright 2011 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef EXAMPLES_PEERCONNECTION_SERVER_DATA_SOCKET_H_ +#define EXAMPLES_PEERCONNECTION_SERVER_DATA_SOCKET_H_ + +#ifdef WIN32 +#include <winsock2.h> +typedef int socklen_t; +typedef SOCKET NativeSocket; +#else +#include <netinet/in.h> +#include <sys/select.h> +#include <sys/socket.h> +#define closesocket close +typedef int NativeSocket; + +#ifndef SOCKET_ERROR +#define SOCKET_ERROR (-1) +#endif + +#ifndef INVALID_SOCKET +#define INVALID_SOCKET static_cast<NativeSocket>(-1) +#endif +#endif + +#include <string> + +class SocketBase { + public: + SocketBase() : socket_(INVALID_SOCKET) {} + explicit SocketBase(NativeSocket socket) : socket_(socket) {} + SocketBase(SocketBase& other) = delete; + SocketBase& operator=(const SocketBase& other) = delete; + ~SocketBase() { Close(); } + + NativeSocket socket() const { return socket_; } + bool valid() const { return socket_ != INVALID_SOCKET; } + + bool Create(); + void Close(); + + protected: + NativeSocket socket_; +}; + +// Represents an HTTP server socket. +class DataSocket : public SocketBase { + public: + enum RequestMethod { + INVALID, + GET, + POST, + OPTIONS, + }; + + explicit DataSocket(NativeSocket socket) + : SocketBase(socket), method_(INVALID), content_length_(0) {} + + ~DataSocket() {} + + static const char kCrossOriginAllowHeaders[]; + + bool headers_received() const { return method_ != INVALID; } + + RequestMethod method() const { return method_; } + + const std::string& request_path() const { return request_path_; } + std::string request_arguments() const; + + const std::string& data() const { return data_; } + + const std::string& content_type() const { return content_type_; } + + size_t content_length() const { return content_length_; } + + bool request_received() const { + return headers_received() && (method_ != POST || data_received()); + } + + bool data_received() const { + return method_ != POST || data_.length() >= content_length_; + } + + // Checks if the request path (minus arguments) matches a given path. + bool PathEquals(const char* path) const; + + // Called when we have received some data from clients. + // Returns false if an error occurred. + bool OnDataAvailable(bool* close_socket); + + // Send a raw buffer of bytes. + bool Send(const std::string& data) const; + + // Send an HTTP response. The `status` should start with a valid HTTP + // response code, followed by a string. E.g. "200 OK". + // If `connection_close` is set to true, an extra "Connection: close" HTTP + // header will be included. `content_type` is the mime content type, not + // including the "Content-Type: " string. + // `extra_headers` should be either empty or a list of headers where each + // header terminates with "\r\n". + // `data` is the body of the message. It's length will be specified via + // a "Content-Length" header. + bool Send(const std::string& status, + bool connection_close, + const std::string& content_type, + const std::string& extra_headers, + const std::string& data) const; + + // Clears all held state and prepares the socket for receiving a new request. + void Clear(); + + protected: + // A fairly relaxed HTTP header parser. Parses the method, path and + // content length (POST only) of a request. + // Returns true if a valid request was received and no errors occurred. + bool ParseHeaders(); + + // Figures out whether the request is a GET or POST and what path is + // being requested. + bool ParseMethodAndPath(const char* begin, size_t len); + + // Determines the length of the body and it's mime type. + bool ParseContentLengthAndType(const char* headers, size_t length); + + protected: + RequestMethod method_; + size_t content_length_; + std::string content_type_; + std::string request_path_; + std::string request_headers_; + std::string data_; +}; + +// The server socket. Accepts connections and generates DataSocket instances +// for each new connection. +class ListeningSocket : public SocketBase { + public: + ListeningSocket() {} + + bool Listen(unsigned short port); + DataSocket* Accept() const; +}; + +#endif // EXAMPLES_PEERCONNECTION_SERVER_DATA_SOCKET_H_ diff --git a/third_party/libwebrtc/examples/peerconnection/server/main.cc b/third_party/libwebrtc/examples/peerconnection/server/main.cc new file mode 100644 index 0000000000..50b8c23401 --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/server/main.cc @@ -0,0 +1,193 @@ +/* + * Copyright 2011 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <stdio.h> +#include <stdlib.h> +#if defined(WEBRTC_POSIX) +#include <sys/select.h> +#endif +#include <time.h> + +#include <string> +#include <vector> + +#include "absl/flags/flag.h" +#include "absl/flags/parse.h" +#include "absl/flags/usage.h" +#include "examples/peerconnection/server/data_socket.h" +#include "examples/peerconnection/server/peer_channel.h" +#include "rtc_base/checks.h" +#include "system_wrappers/include/field_trial.h" +#include "test/field_trial.h" + +ABSL_FLAG( + std::string, + force_fieldtrials, + "", + "Field trials control experimental features. This flag specifies the field " + "trials in effect. E.g. running with " + "--force_fieldtrials=WebRTC-FooFeature/Enabled/ " + "will assign the group Enabled to field trial WebRTC-FooFeature. Multiple " + "trials are separated by \"/\""); +ABSL_FLAG(int, port, 8888, "default: 8888"); + +static const size_t kMaxConnections = (FD_SETSIZE - 2); + +void HandleBrowserRequest(DataSocket* ds, bool* quit) { + RTC_DCHECK(ds && ds->valid()); + RTC_DCHECK(quit); + + const std::string& path = ds->request_path(); + + *quit = (path.compare("/quit") == 0); + + if (*quit) { + ds->Send("200 OK", true, "text/html", "", + "<html><body>Quitting...</body></html>"); + } else if (ds->method() == DataSocket::OPTIONS) { + // We'll get this when a browsers do cross-resource-sharing requests. + // The headers to allow cross-origin script support will be set inside + // Send. + ds->Send("200 OK", true, "", "", ""); + } else { + // Here we could write some useful output back to the browser depending on + // the path. + printf("Received an invalid request: %s\n", ds->request_path().c_str()); + ds->Send("500 Sorry", true, "text/html", "", + "<html><body>Sorry, not yet implemented</body></html>"); + } +} + +int main(int argc, char* argv[]) { + absl::SetProgramUsageMessage( + "Example usage: ./peerconnection_server --port=8888\n"); + absl::ParseCommandLine(argc, argv); + + // InitFieldTrialsFromString stores the char*, so the char array must outlive + // the application. + const std::string force_field_trials = absl::GetFlag(FLAGS_force_fieldtrials); + webrtc::field_trial::InitFieldTrialsFromString(force_field_trials.c_str()); + + int port = absl::GetFlag(FLAGS_port); + + // Abort if the user specifies a port that is outside the allowed + // range [1, 65535]. + if ((port < 1) || (port > 65535)) { + printf("Error: %i is not a valid port.\n", port); + return -1; + } + + ListeningSocket listener; + if (!listener.Create()) { + printf("Failed to create server socket\n"); + return -1; + } else if (!listener.Listen(port)) { + printf("Failed to listen on server socket\n"); + return -1; + } + + printf("Server listening on port %i\n", port); + + PeerChannel clients; + typedef std::vector<DataSocket*> SocketArray; + SocketArray sockets; + bool quit = false; + while (!quit) { + fd_set socket_set; + FD_ZERO(&socket_set); + if (listener.valid()) + FD_SET(listener.socket(), &socket_set); + + for (SocketArray::iterator i = sockets.begin(); i != sockets.end(); ++i) + FD_SET((*i)->socket(), &socket_set); + + struct timeval timeout = {10, 0}; + if (select(FD_SETSIZE, &socket_set, NULL, NULL, &timeout) == SOCKET_ERROR) { + printf("select failed\n"); + break; + } + + for (SocketArray::iterator i = sockets.begin(); i != sockets.end(); ++i) { + DataSocket* s = *i; + bool socket_done = true; + if (FD_ISSET(s->socket(), &socket_set)) { + if (s->OnDataAvailable(&socket_done) && s->request_received()) { + ChannelMember* member = clients.Lookup(s); + if (member || PeerChannel::IsPeerConnection(s)) { + if (!member) { + if (s->PathEquals("/sign_in")) { + clients.AddMember(s); + } else { + printf("No member found for: %s\n", s->request_path().c_str()); + s->Send("500 Error", true, "text/plain", "", + "Peer most likely gone."); + } + } else if (member->is_wait_request(s)) { + // no need to do anything. + socket_done = false; + } else { + ChannelMember* target = clients.IsTargetedRequest(s); + if (target) { + member->ForwardRequestToPeer(s, target); + } else if (s->PathEquals("/sign_out")) { + s->Send("200 OK", true, "text/plain", "", ""); + } else { + printf("Couldn't find target for request: %s\n", + s->request_path().c_str()); + s->Send("500 Error", true, "text/plain", "", + "Peer most likely gone."); + } + } + } else { + HandleBrowserRequest(s, &quit); + if (quit) { + printf("Quitting...\n"); + FD_CLR(listener.socket(), &socket_set); + listener.Close(); + clients.CloseAll(); + } + } + } + } else { + socket_done = false; + } + + if (socket_done) { + printf("Disconnecting socket\n"); + clients.OnClosing(s); + RTC_DCHECK(s->valid()); // Close must not have been called yet. + FD_CLR(s->socket(), &socket_set); + delete (*i); + i = sockets.erase(i); + if (i == sockets.end()) + break; + } + } + + clients.CheckForTimeout(); + + if (FD_ISSET(listener.socket(), &socket_set)) { + DataSocket* s = listener.Accept(); + if (sockets.size() >= kMaxConnections) { + delete s; // sorry, that's all we can take. + printf("Connection limit reached\n"); + } else { + sockets.push_back(s); + printf("New connection...\n"); + } + } + } + + for (SocketArray::iterator i = sockets.begin(); i != sockets.end(); ++i) + delete (*i); + sockets.clear(); + + return 0; +} diff --git a/third_party/libwebrtc/examples/peerconnection/server/peer_channel.cc b/third_party/libwebrtc/examples/peerconnection/server/peer_channel.cc new file mode 100644 index 0000000000..f53820cc60 --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/server/peer_channel.cc @@ -0,0 +1,360 @@ +/* + * Copyright 2011 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/peerconnection/server/peer_channel.h" + +#include <stdio.h> +#include <stdlib.h> + +#include <algorithm> + +#include "examples/peerconnection/server/data_socket.h" +#include "examples/peerconnection/server/utils.h" +#include "rtc_base/checks.h" + +// Set to the peer id of the originator when messages are being +// exchanged between peers, but set to the id of the receiving peer +// itself when notifications are sent from the server about the state +// of other peers. +// +// WORKAROUND: Since support for CORS varies greatly from one browser to the +// next, we don't use a custom name for our peer-id header (originally it was +// "X-Peer-Id: "). Instead, we use a "simple header", "Pragma" which should +// always be exposed to CORS requests. There is a special CORS header devoted +// to exposing proprietary headers (Access-Control-Expose-Headers), however +// at this point it is not working correctly in some popular browsers. +static const char kPeerIdHeader[] = "Pragma: "; + +static const char* kRequestPaths[] = { + "/wait", + "/sign_out", + "/message", +}; + +enum RequestPathIndex { + kWait, + kSignOut, + kMessage, +}; + +const size_t kMaxNameLength = 512; + +// +// ChannelMember +// + +int ChannelMember::s_member_id_ = 0; + +ChannelMember::ChannelMember(DataSocket* socket) + : waiting_socket_(NULL), + id_(++s_member_id_), + connected_(true), + timestamp_(time(NULL)) { + RTC_DCHECK(socket); + RTC_DCHECK_EQ(socket->method(), DataSocket::GET); + RTC_DCHECK(socket->PathEquals("/sign_in")); + name_ = socket->request_arguments(); + if (name_.empty()) + name_ = "peer_" + int2str(id_); + else if (name_.length() > kMaxNameLength) + name_.resize(kMaxNameLength); + + std::replace(name_.begin(), name_.end(), ',', '_'); +} + +ChannelMember::~ChannelMember() {} + +bool ChannelMember::is_wait_request(DataSocket* ds) const { + return ds && ds->PathEquals(kRequestPaths[kWait]); +} + +bool ChannelMember::TimedOut() { + return waiting_socket_ == NULL && (time(NULL) - timestamp_) > 30; +} + +std::string ChannelMember::GetPeerIdHeader() const { + std::string ret(kPeerIdHeader + int2str(id_) + "\r\n"); + return ret; +} + +bool ChannelMember::NotifyOfOtherMember(const ChannelMember& other) { + RTC_DCHECK_NE(&other, this); + QueueResponse("200 OK", "text/plain", GetPeerIdHeader(), other.GetEntry()); + return true; +} + +// Returns a string in the form "name,id,connected\n". +std::string ChannelMember::GetEntry() const { + RTC_DCHECK(name_.length() <= kMaxNameLength); + + // name, 11-digit int, 1-digit bool, newline, null + char entry[kMaxNameLength + 15]; + snprintf(entry, sizeof(entry), "%s,%d,%d\n", + name_.substr(0, kMaxNameLength).c_str(), id_, connected_); + return entry; +} + +void ChannelMember::ForwardRequestToPeer(DataSocket* ds, ChannelMember* peer) { + RTC_DCHECK(peer); + RTC_DCHECK(ds); + + std::string extra_headers(GetPeerIdHeader()); + + if (peer == this) { + ds->Send("200 OK", true, ds->content_type(), extra_headers, ds->data()); + } else { + printf("Client %s sending to %s\n", name_.c_str(), peer->name().c_str()); + peer->QueueResponse("200 OK", ds->content_type(), extra_headers, + ds->data()); + ds->Send("200 OK", true, "text/plain", "", ""); + } +} + +void ChannelMember::OnClosing(DataSocket* ds) { + if (ds == waiting_socket_) { + waiting_socket_ = NULL; + timestamp_ = time(NULL); + } +} + +void ChannelMember::QueueResponse(const std::string& status, + const std::string& content_type, + const std::string& extra_headers, + const std::string& data) { + if (waiting_socket_) { + RTC_DCHECK(queue_.empty()); + RTC_DCHECK_EQ(waiting_socket_->method(), DataSocket::GET); + bool ok = + waiting_socket_->Send(status, true, content_type, extra_headers, data); + if (!ok) { + printf("Failed to deliver data to waiting socket\n"); + } + waiting_socket_ = NULL; + timestamp_ = time(NULL); + } else { + QueuedResponse qr; + qr.status = status; + qr.content_type = content_type; + qr.extra_headers = extra_headers; + qr.data = data; + queue_.push(qr); + } +} + +void ChannelMember::SetWaitingSocket(DataSocket* ds) { + RTC_DCHECK_EQ(ds->method(), DataSocket::GET); + if (ds && !queue_.empty()) { + RTC_DCHECK(!waiting_socket_); + const QueuedResponse& response = queue_.front(); + ds->Send(response.status, true, response.content_type, + response.extra_headers, response.data); + queue_.pop(); + } else { + waiting_socket_ = ds; + } +} + +// +// PeerChannel +// + +// static +bool PeerChannel::IsPeerConnection(const DataSocket* ds) { + RTC_DCHECK(ds); + return (ds->method() == DataSocket::POST && ds->content_length() > 0) || + (ds->method() == DataSocket::GET && ds->PathEquals("/sign_in")); +} + +ChannelMember* PeerChannel::Lookup(DataSocket* ds) const { + RTC_DCHECK(ds); + + if (ds->method() != DataSocket::GET && ds->method() != DataSocket::POST) + return NULL; + + size_t i = 0; + for (; i < ARRAYSIZE(kRequestPaths); ++i) { + if (ds->PathEquals(kRequestPaths[i])) + break; + } + + if (i == ARRAYSIZE(kRequestPaths)) + return NULL; + + std::string args(ds->request_arguments()); + static const char kPeerId[] = "peer_id="; + size_t found = args.find(kPeerId); + if (found == std::string::npos) + return NULL; + + int id = atoi(&args[found + ARRAYSIZE(kPeerId) - 1]); + Members::const_iterator iter = members_.begin(); + for (; iter != members_.end(); ++iter) { + if (id == (*iter)->id()) { + if (i == kWait) + (*iter)->SetWaitingSocket(ds); + if (i == kSignOut) + (*iter)->set_disconnected(); + return *iter; + } + } + + return NULL; +} + +ChannelMember* PeerChannel::IsTargetedRequest(const DataSocket* ds) const { + RTC_DCHECK(ds); + // Regardless of GET or POST, we look for the peer_id parameter + // only in the request_path. + const std::string& path = ds->request_path(); + size_t args = path.find('?'); + if (args == std::string::npos) + return NULL; + size_t found; + const char kTargetPeerIdParam[] = "to="; + do { + found = path.find(kTargetPeerIdParam, args); + if (found == std::string::npos) + return NULL; + if (found == (args + 1) || path[found - 1] == '&') { + found += ARRAYSIZE(kTargetPeerIdParam) - 1; + break; + } + args = found + ARRAYSIZE(kTargetPeerIdParam) - 1; + } while (true); + int id = atoi(&path[found]); + Members::const_iterator i = members_.begin(); + for (; i != members_.end(); ++i) { + if ((*i)->id() == id) { + return *i; + } + } + return NULL; +} + +bool PeerChannel::AddMember(DataSocket* ds) { + RTC_DCHECK(IsPeerConnection(ds)); + ChannelMember* new_guy = new ChannelMember(ds); + Members failures; + BroadcastChangedState(*new_guy, &failures); + HandleDeliveryFailures(&failures); + members_.push_back(new_guy); + + printf("New member added (total=%s): %s\n", + size_t2str(members_.size()).c_str(), new_guy->name().c_str()); + + // Let the newly connected peer know about other members of the channel. + std::string content_type; + std::string response = BuildResponseForNewMember(*new_guy, &content_type); + ds->Send("200 Added", true, content_type, new_guy->GetPeerIdHeader(), + response); + return true; +} + +void PeerChannel::CloseAll() { + Members::const_iterator i = members_.begin(); + for (; i != members_.end(); ++i) { + (*i)->QueueResponse("200 OK", "text/plain", "", "Server shutting down"); + } + DeleteAll(); +} + +void PeerChannel::OnClosing(DataSocket* ds) { + for (Members::iterator i = members_.begin(); i != members_.end(); ++i) { + ChannelMember* m = (*i); + m->OnClosing(ds); + if (!m->connected()) { + i = members_.erase(i); + Members failures; + BroadcastChangedState(*m, &failures); + HandleDeliveryFailures(&failures); + delete m; + if (i == members_.end()) + break; + } + } + printf("Total connected: %s\n", size_t2str(members_.size()).c_str()); +} + +void PeerChannel::CheckForTimeout() { + for (Members::iterator i = members_.begin(); i != members_.end(); ++i) { + ChannelMember* m = (*i); + if (m->TimedOut()) { + printf("Timeout: %s\n", m->name().c_str()); + m->set_disconnected(); + i = members_.erase(i); + Members failures; + BroadcastChangedState(*m, &failures); + HandleDeliveryFailures(&failures); + delete m; + if (i == members_.end()) + break; + } + } +} + +void PeerChannel::DeleteAll() { + for (Members::iterator i = members_.begin(); i != members_.end(); ++i) + delete (*i); + members_.clear(); +} + +void PeerChannel::BroadcastChangedState(const ChannelMember& member, + Members* delivery_failures) { + // This function should be called prior to DataSocket::Close(). + RTC_DCHECK(delivery_failures); + + if (!member.connected()) { + printf("Member disconnected: %s\n", member.name().c_str()); + } + + Members::iterator i = members_.begin(); + for (; i != members_.end(); ++i) { + if (&member != (*i)) { + if (!(*i)->NotifyOfOtherMember(member)) { + (*i)->set_disconnected(); + delivery_failures->push_back(*i); + i = members_.erase(i); + if (i == members_.end()) + break; + } + } + } +} + +void PeerChannel::HandleDeliveryFailures(Members* failures) { + RTC_DCHECK(failures); + + while (!failures->empty()) { + Members::iterator i = failures->begin(); + ChannelMember* member = *i; + RTC_DCHECK(!member->connected()); + failures->erase(i); + BroadcastChangedState(*member, failures); + delete member; + } +} + +// Builds a simple list of "name,id\n" entries for each member. +std::string PeerChannel::BuildResponseForNewMember(const ChannelMember& member, + std::string* content_type) { + RTC_DCHECK(content_type); + + *content_type = "text/plain"; + // The peer itself will always be the first entry. + std::string response(member.GetEntry()); + for (Members::iterator i = members_.begin(); i != members_.end(); ++i) { + if (member.id() != (*i)->id()) { + RTC_DCHECK((*i)->connected()); + response += (*i)->GetEntry(); + } + } + + return response; +} diff --git a/third_party/libwebrtc/examples/peerconnection/server/peer_channel.h b/third_party/libwebrtc/examples/peerconnection/server/peer_channel.h new file mode 100644 index 0000000000..c3624908ac --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/server/peer_channel.h @@ -0,0 +1,118 @@ +/* + * Copyright 2011 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef EXAMPLES_PEERCONNECTION_SERVER_PEER_CHANNEL_H_ +#define EXAMPLES_PEERCONNECTION_SERVER_PEER_CHANNEL_H_ + +#include <time.h> + +#include <queue> +#include <string> +#include <vector> + +class DataSocket; + +// Represents a single peer connected to the server. +class ChannelMember { + public: + explicit ChannelMember(DataSocket* socket); + ~ChannelMember(); + + bool connected() const { return connected_; } + int id() const { return id_; } + void set_disconnected() { connected_ = false; } + bool is_wait_request(DataSocket* ds) const; + const std::string& name() const { return name_; } + + bool TimedOut(); + + std::string GetPeerIdHeader() const; + + bool NotifyOfOtherMember(const ChannelMember& other); + + // Returns a string in the form "name,id\n". + std::string GetEntry() const; + + void ForwardRequestToPeer(DataSocket* ds, ChannelMember* peer); + + void OnClosing(DataSocket* ds); + + void QueueResponse(const std::string& status, + const std::string& content_type, + const std::string& extra_headers, + const std::string& data); + + void SetWaitingSocket(DataSocket* ds); + + protected: + struct QueuedResponse { + std::string status, content_type, extra_headers, data; + }; + + DataSocket* waiting_socket_; + int id_; + bool connected_; + time_t timestamp_; + std::string name_; + std::queue<QueuedResponse> queue_; + static int s_member_id_; +}; + +// Manages all currently connected peers. +class PeerChannel { + public: + typedef std::vector<ChannelMember*> Members; + + PeerChannel() {} + + ~PeerChannel() { DeleteAll(); } + + const Members& members() const { return members_; } + + // Returns true if the request should be treated as a new ChannelMember + // request. Otherwise the request is not peerconnection related. + static bool IsPeerConnection(const DataSocket* ds); + + // Finds a connected peer that's associated with the `ds` socket. + ChannelMember* Lookup(DataSocket* ds) const; + + // Checks if the request has a "peer_id" parameter and if so, looks up the + // peer for which the request is targeted at. + ChannelMember* IsTargetedRequest(const DataSocket* ds) const; + + // Adds a new ChannelMember instance to the list of connected peers and + // associates it with the socket. + bool AddMember(DataSocket* ds); + + // Closes all connections and sends a "shutting down" message to all + // connected peers. + void CloseAll(); + + // Called when a socket was determined to be closing by the peer (or if the + // connection went dead). + void OnClosing(DataSocket* ds); + + void CheckForTimeout(); + + protected: + void DeleteAll(); + void BroadcastChangedState(const ChannelMember& member, + Members* delivery_failures); + void HandleDeliveryFailures(Members* failures); + + // Builds a simple list of "name,id\n" entries for each member. + std::string BuildResponseForNewMember(const ChannelMember& member, + std::string* content_type); + + protected: + Members members_; +}; + +#endif // EXAMPLES_PEERCONNECTION_SERVER_PEER_CHANNEL_H_ diff --git a/third_party/libwebrtc/examples/peerconnection/server/server_test.html b/third_party/libwebrtc/examples/peerconnection/server/server_test.html new file mode 100644 index 0000000000..0a165f19d5 --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/server/server_test.html @@ -0,0 +1,237 @@ +<html> +<head> +<title>PeerConnection server test page</title> + +<script> +var request = null; +var hangingGet = null; +var localName; +var server; +var my_id = -1; +var other_peers = {}; +var message_counter = 0; + +function trace(txt) { + var elem = document.getElementById("debug"); + elem.innerHTML += txt + "<br>"; +} + +function handleServerNotification(data) { + trace("Server notification: " + data); + var parsed = data.split(','); + if (parseInt(parsed[2]) != 0) + other_peers[parseInt(parsed[1])] = parsed[0]; +} + +function handlePeerMessage(peer_id, data) { + ++message_counter; + var str = "Message from '" + other_peers[peer_id] + "' "; + str += "<span id='toggle_" + message_counter + "' onclick='toggleMe(this);' "; + str += "style='cursor: pointer'>+</span><br>"; + str += "<blockquote id='msg_" + message_counter + "' style='display:none'>"; + str += data + "</blockquote>"; + trace(str); + if (document.getElementById("loopback").checked) { + if (data.search("offer") != -1) { + // In loopback mode, if DTLS is enabled, notify the client to disable it. + // Otherwise replace the offer with an answer. + if (data.search("fingerprint") != -1) + data = data.replace("offer", "offer-loopback"); + else + data = data.replace("offer", "answer"); + } + sendToPeer(peer_id, data); + } +} + +function GetIntHeader(r, name) { + var val = r.getResponseHeader(name); + return val != null && val.length ? parseInt(val) : -1; +} + +function hangingGetCallback() { + try { + if (hangingGet.readyState != 4) + return; + if (hangingGet.status != 200) { + trace("server error: " + hangingGet.statusText); + disconnect(); + } else { + var peer_id = GetIntHeader(hangingGet, "Pragma"); + if (peer_id == my_id) { + handleServerNotification(hangingGet.responseText); + } else { + handlePeerMessage(peer_id, hangingGet.responseText); + } + } + + if (hangingGet) { + hangingGet.abort(); + hangingGet = null; + } + + if (my_id != -1) + window.setTimeout(startHangingGet, 0); + } catch (e) { + trace("Hanging get error: " + e.description); + } +} + +function startHangingGet() { + try { + hangingGet = new XMLHttpRequest(); + hangingGet.onreadystatechange = hangingGetCallback; + hangingGet.ontimeout = onHangingGetTimeout; + hangingGet.open("GET", server + "/wait?peer_id=" + my_id, true); + hangingGet.send(); + } catch (e) { + trace("error" + e.description); + } +} + +function onHangingGetTimeout() { + trace("hanging get timeout. issuing again."); + hangingGet.abort(); + hangingGet = null; + if (my_id != -1) + window.setTimeout(startHangingGet, 0); +} + +function signInCallback() { + try { + if (request.readyState == 4) { + if (request.status == 200) { + var peers = request.responseText.split("\n"); + my_id = parseInt(peers[0].split(',')[1]); + trace("My id: " + my_id); + for (var i = 1; i < peers.length; ++i) { + if (peers[i].length > 0) { + trace("Peer " + i + ": " + peers[i]); + var parsed = peers[i].split(','); + other_peers[parseInt(parsed[1])] = parsed[0]; + } + } + startHangingGet(); + request = null; + } + } + } catch (e) { + trace("error: " + e.description); + } +} + +function signIn() { + try { + request = new XMLHttpRequest(); + request.onreadystatechange = signInCallback; + request.open("GET", server + "/sign_in?" + localName, true); + request.send(); + } catch (e) { + trace("error: " + e.description); + } +} + +function sendToPeer(peer_id, data) { + if (my_id == -1) { + alert("Not connected"); + return; + } + if (peer_id == my_id) { + alert("Can't send a message to oneself :)"); + return; + } + var r = new XMLHttpRequest(); + r.open("POST", server + "/message?peer_id=" + my_id + "&to=" + peer_id, + false); + r.setRequestHeader("Content-Type", "text/plain"); + r.send(data); + r = null; +} + +function connect() { + localName = document.getElementById("local").value.toLowerCase(); + server = document.getElementById("server").value.toLowerCase(); + if (localName.length == 0) { + alert("I need a name please."); + document.getElementById("local").focus(); + } else { + document.getElementById("connect").disabled = true; + document.getElementById("disconnect").disabled = false; + document.getElementById("send").disabled = false; + signIn(); + } +} + +function disconnect() { + if (request) { + request.abort(); + request = null; + } + + if (hangingGet) { + hangingGet.abort(); + hangingGet = null; + } + + if (my_id != -1) { + request = new XMLHttpRequest(); + request.open("GET", server + "/sign_out?peer_id=" + my_id, false); + request.send(); + request = null; + my_id = -1; + } + + document.getElementById("connect").disabled = false; + document.getElementById("disconnect").disabled = true; + document.getElementById("send").disabled = true; +} + +window.onbeforeunload = disconnect; + +function send() { + var text = document.getElementById("message").value; + var peer_id = parseInt(document.getElementById("peer_id").value); + if (!text.length || peer_id == 0) { + alert("No text supplied or invalid peer id"); + } else { + sendToPeer(peer_id, text); + } +} + +function toggleMe(obj) { + var id = obj.id.replace("toggle", "msg"); + var t = document.getElementById(id); + if (obj.innerText == "+") { + obj.innerText = "-"; + t.style.display = "block"; + } else { + obj.innerText = "+"; + t.style.display = "none"; + } +} + +</script> + +</head> +<body> +Server: <input type="text" id="server" value="http://localhost:8888" /><br> +<input type="checkbox" id="loopback" checked="checked"/> Loopback (just send +received messages right back)<br> +Your name: <input type="text" id="local" value="my_name"/> +<button id="connect" onclick="connect();">Connect</button> +<button disabled="true" id="disconnect" + onclick="disconnect();">Disconnect</button> +<br> +<table><tr><td> +Target peer id: <input type="text" id="peer_id" size="3"/></td><td> +Message: <input type="text" id="message"/></td><td> +<button disabled="true" id="send" onclick="send();">Send</button> +</td></tr></table> +<button onclick="document.getElementById('debug').innerHTML='';"> +Clear log</button> + +<pre id="debug"> +</pre> +<br><hr> +</body> +</html> diff --git a/third_party/libwebrtc/examples/peerconnection/server/utils.cc b/third_party/libwebrtc/examples/peerconnection/server/utils.cc new file mode 100644 index 0000000000..5e61e601d9 --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/server/utils.cc @@ -0,0 +1,25 @@ +/* + * Copyright 2011 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/peerconnection/server/utils.h" + +#include <stdio.h> + +#include "rtc_base/string_encode.h" + +using rtc::ToString; + +std::string int2str(int i) { + return ToString(i); +} + +std::string size_t2str(size_t i) { + return ToString(i); +} diff --git a/third_party/libwebrtc/examples/peerconnection/server/utils.h b/third_party/libwebrtc/examples/peerconnection/server/utils.h new file mode 100644 index 0000000000..85c04a40e9 --- /dev/null +++ b/third_party/libwebrtc/examples/peerconnection/server/utils.h @@ -0,0 +1,25 @@ +/* + * Copyright 2011 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef EXAMPLES_PEERCONNECTION_SERVER_UTILS_H_ +#define EXAMPLES_PEERCONNECTION_SERVER_UTILS_H_ + +#include <stddef.h> + +#include <string> + +#ifndef ARRAYSIZE +#define ARRAYSIZE(x) (sizeof(x) / sizeof(x[0])) +#endif + +std::string int2str(int i); +std::string size_t2str(size_t i); + +#endif // EXAMPLES_PEERCONNECTION_SERVER_UTILS_H_ diff --git a/third_party/libwebrtc/examples/stunprober/main.cc b/third_party/libwebrtc/examples/stunprober/main.cc new file mode 100644 index 0000000000..3b3c06be8f --- /dev/null +++ b/third_party/libwebrtc/examples/stunprober/main.cc @@ -0,0 +1,146 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <memory> +#include <set> +#include <sstream> +#include <string> +#include <vector> + +#include "absl/flags/flag.h" +#include "absl/flags/parse.h" +#include "p2p/base/basic_packet_socket_factory.h" +#include "p2p/stunprober/stun_prober.h" +#include "rtc_base/helpers.h" +#include "rtc_base/logging.h" +#include "rtc_base/network.h" +#include "rtc_base/physical_socket_server.h" +#include "rtc_base/socket_address.h" +#include "rtc_base/ssl_adapter.h" +#include "rtc_base/thread.h" +#include "rtc_base/time_utils.h" +#include "test/scoped_key_value_config.h" + +using stunprober::AsyncCallback; +using stunprober::StunProber; + +ABSL_FLAG(int, + interval, + 10, + "Interval of consecutive stun pings in milliseconds"); +ABSL_FLAG(bool, + shared_socket, + false, + "Share socket mode for different remote IPs"); +ABSL_FLAG(int, + pings_per_ip, + 10, + "Number of consecutive stun pings to send for each IP"); +ABSL_FLAG(int, + timeout, + 1000, + "Milliseconds of wait after the last ping sent before exiting"); +ABSL_FLAG( + std::string, + servers, + "stun.l.google.com:19302,stun1.l.google.com:19302,stun2.l.google.com:19302", + "Comma separated STUN server addresses with ports"); + +namespace { + +const char* PrintNatType(stunprober::NatType type) { + switch (type) { + case stunprober::NATTYPE_NONE: + return "Not behind a NAT"; + case stunprober::NATTYPE_UNKNOWN: + return "Unknown NAT type"; + case stunprober::NATTYPE_SYMMETRIC: + return "Symmetric NAT"; + case stunprober::NATTYPE_NON_SYMMETRIC: + return "Non-Symmetric NAT"; + default: + return "Invalid"; + } +} + +void PrintStats(StunProber* prober) { + StunProber::Stats stats; + if (!prober->GetStats(&stats)) { + RTC_LOG(LS_WARNING) << "Results are inconclusive."; + return; + } + + RTC_LOG(LS_INFO) << "Shared Socket Mode: " << stats.shared_socket_mode; + RTC_LOG(LS_INFO) << "Requests sent: " << stats.num_request_sent; + RTC_LOG(LS_INFO) << "Responses received: " << stats.num_response_received; + RTC_LOG(LS_INFO) << "Target interval (ns): " + << stats.target_request_interval_ns; + RTC_LOG(LS_INFO) << "Actual interval (ns): " + << stats.actual_request_interval_ns; + RTC_LOG(LS_INFO) << "NAT Type: " << PrintNatType(stats.nat_type); + RTC_LOG(LS_INFO) << "Host IP: " << stats.host_ip; + RTC_LOG(LS_INFO) << "Server-reflexive ips: "; + for (auto& ip : stats.srflx_addrs) { + RTC_LOG(LS_INFO) << "\t" << ip; + } + + RTC_LOG(LS_INFO) << "Success Precent: " << stats.success_percent; + RTC_LOG(LS_INFO) << "Response Latency:" << stats.average_rtt_ms; +} + +void StopTrial(rtc::Thread* thread, StunProber* prober, int result) { + thread->Quit(); + if (prober) { + RTC_LOG(LS_INFO) << "Result: " << result; + if (result == StunProber::SUCCESS) { + PrintStats(prober); + } + } +} + +} // namespace + +int main(int argc, char* argv[]) { + absl::ParseCommandLine(argc, argv); + + std::vector<rtc::SocketAddress> server_addresses; + std::istringstream servers(absl::GetFlag(FLAGS_servers)); + std::string server; + while (getline(servers, server, ',')) { + rtc::SocketAddress addr; + if (!addr.FromString(server)) { + RTC_LOG(LS_ERROR) << "Parsing " << server << " failed."; + return -1; + } + server_addresses.push_back(addr); + } + + rtc::InitializeSSL(); + rtc::InitRandom(rtc::Time32()); + webrtc::test::ScopedKeyValueConfig field_trials; + rtc::PhysicalSocketServer socket_server; + rtc::AutoSocketServerThread thread(&socket_server); + auto socket_factory = + std::make_unique<rtc::BasicPacketSocketFactory>(&socket_server); + std::unique_ptr<rtc::BasicNetworkManager> network_manager( + new rtc::BasicNetworkManager(&socket_server, &field_trials)); + std::vector<const rtc::Network*> networks = network_manager->GetNetworks(); + auto prober = std::make_unique<StunProber>(socket_factory.get(), + rtc::Thread::Current(), networks); + auto finish_callback = [&thread](StunProber* prober, int result) { + StopTrial(&thread, prober, result); + }; + prober->Start(server_addresses, absl::GetFlag(FLAGS_shared_socket), + absl::GetFlag(FLAGS_interval), + absl::GetFlag(FLAGS_pings_per_ip), absl::GetFlag(FLAGS_timeout), + AsyncCallback(finish_callback)); + thread.Run(); + return 0; +} diff --git a/third_party/libwebrtc/examples/stunserver/stunserver_main.cc b/third_party/libwebrtc/examples/stunserver/stunserver_main.cc new file mode 100644 index 0000000000..8180069bf0 --- /dev/null +++ b/third_party/libwebrtc/examples/stunserver/stunserver_main.cc @@ -0,0 +1,49 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#include <iostream> + +#include "p2p/base/stun_server.h" +#include "rtc_base/async_udp_socket.h" +#include "rtc_base/socket_address.h" +#include "rtc_base/socket_server.h" +#include "rtc_base/thread.h" + +using cricket::StunServer; + +int main(int argc, char* argv[]) { + if (argc != 2) { + std::cerr << "usage: stunserver address" << std::endl; + return 1; + } + + rtc::SocketAddress server_addr; + if (!server_addr.FromString(argv[1])) { + std::cerr << "Unable to parse IP address: " << argv[1]; + return 1; + } + + rtc::Thread* pthMain = rtc::Thread::Current(); + + rtc::AsyncUDPSocket* server_socket = + rtc::AsyncUDPSocket::Create(pthMain->socketserver(), server_addr); + if (!server_socket) { + std::cerr << "Failed to create a UDP socket" << std::endl; + return 1; + } + + StunServer* server = new StunServer(server_socket); + + std::cout << "Listening at " << server_addr.ToString() << std::endl; + + pthMain->Run(); + + delete server; + return 0; +} diff --git a/third_party/libwebrtc/examples/turnserver/read_auth_file.cc b/third_party/libwebrtc/examples/turnserver/read_auth_file.cc new file mode 100644 index 0000000000..4b0b21b8ae --- /dev/null +++ b/third_party/libwebrtc/examples/turnserver/read_auth_file.cc @@ -0,0 +1,37 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/turnserver/read_auth_file.h" + +#include <stddef.h> + +#include "absl/strings/string_view.h" +#include "api/array_view.h" +#include "rtc_base/string_encode.h" + +namespace webrtc_examples { + +std::map<std::string, std::string> ReadAuthFile(std::istream* s) { + std::map<std::string, std::string> name_to_key; + for (std::string line; std::getline(*s, line);) { + const size_t sep = line.find('='); + if (sep == std::string::npos) + continue; + char buf[32]; + size_t len = rtc::hex_decode(rtc::ArrayView<char>(buf), + absl::string_view(line).substr(sep + 1)); + if (len > 0) { + name_to_key.emplace(line.substr(0, sep), std::string(buf, len)); + } + } + return name_to_key; +} + +} // namespace webrtc_examples diff --git a/third_party/libwebrtc/examples/turnserver/read_auth_file.h b/third_party/libwebrtc/examples/turnserver/read_auth_file.h new file mode 100644 index 0000000000..1c139c9924 --- /dev/null +++ b/third_party/libwebrtc/examples/turnserver/read_auth_file.h @@ -0,0 +1,24 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef EXAMPLES_TURNSERVER_READ_AUTH_FILE_H_ +#define EXAMPLES_TURNSERVER_READ_AUTH_FILE_H_ + +#include <istream> +#include <map> +#include <string> + +namespace webrtc_examples { + +std::map<std::string, std::string> ReadAuthFile(std::istream* s); + +} // namespace webrtc_examples + +#endif // EXAMPLES_TURNSERVER_READ_AUTH_FILE_H_ diff --git a/third_party/libwebrtc/examples/turnserver/read_auth_file_unittest.cc b/third_party/libwebrtc/examples/turnserver/read_auth_file_unittest.cc new file mode 100644 index 0000000000..23b026429b --- /dev/null +++ b/third_party/libwebrtc/examples/turnserver/read_auth_file_unittest.cc @@ -0,0 +1,45 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/turnserver/read_auth_file.h" + +#include <sstream> + +#include "test/gtest.h" + +namespace webrtc_examples { + +TEST(ReadAuthFile, HandlesEmptyFile) { + std::istringstream empty; + auto map = ReadAuthFile(&empty); + EXPECT_TRUE(map.empty()); +} + +TEST(ReadAuthFile, RecognizesValidUser) { + std::istringstream file("foo=deadbeaf\n"); + auto map = ReadAuthFile(&file); + ASSERT_NE(map.find("foo"), map.end()); + EXPECT_EQ(map["foo"], "\xde\xad\xbe\xaf"); +} + +TEST(ReadAuthFile, EmptyValueForInvalidHex) { + std::istringstream file( + "foo=deadbeaf\n" + "bar=xxxxinvalidhex\n" + "baz=cafe\n"); + auto map = ReadAuthFile(&file); + ASSERT_NE(map.find("foo"), map.end()); + EXPECT_EQ(map["foo"], "\xde\xad\xbe\xaf"); + EXPECT_EQ(map.find("bar"), map.end()); + ASSERT_NE(map.find("baz"), map.end()); + EXPECT_EQ(map["baz"], "\xca\xfe"); +} + +} // namespace webrtc_examples diff --git a/third_party/libwebrtc/examples/turnserver/turnserver_main.cc b/third_party/libwebrtc/examples/turnserver/turnserver_main.cc new file mode 100644 index 0000000000..8db6162306 --- /dev/null +++ b/third_party/libwebrtc/examples/turnserver/turnserver_main.cc @@ -0,0 +1,101 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <fstream> +#include <iostream> +#include <map> +#include <string> +#include <utility> + +#include "absl/strings/string_view.h" +#include "examples/turnserver/read_auth_file.h" +#include "p2p/base/basic_packet_socket_factory.h" +#include "p2p/base/port_interface.h" +#include "p2p/base/turn_server.h" +#include "rtc_base/async_udp_socket.h" +#include "rtc_base/ip_address.h" +#include "rtc_base/physical_socket_server.h" +#include "rtc_base/socket_address.h" +#include "rtc_base/thread.h" + +namespace { +const char kSoftware[] = "libjingle TurnServer"; + +class TurnFileAuth : public cricket::TurnAuthInterface { + public: + explicit TurnFileAuth(std::map<std::string, std::string> name_to_key) + : name_to_key_(std::move(name_to_key)) {} + + virtual bool GetKey(absl::string_view username, + absl::string_view realm, + std::string* key) { + // File is stored as lines of <username>=<HA1>. + // Generate HA1 via "echo -n "<username>:<realm>:<password>" | md5sum" + auto it = name_to_key_.find(std::string(username)); + if (it == name_to_key_.end()) + return false; + *key = it->second; + return true; + } + + private: + const std::map<std::string, std::string> name_to_key_; +}; + +} // namespace + +int main(int argc, char* argv[]) { + if (argc != 5) { + std::cerr << "usage: turnserver int-addr ext-ip realm auth-file" + << std::endl; + return 1; + } + + rtc::SocketAddress int_addr; + if (!int_addr.FromString(argv[1])) { + std::cerr << "Unable to parse IP address: " << argv[1] << std::endl; + return 1; + } + + rtc::IPAddress ext_addr; + if (!IPFromString(argv[2], &ext_addr)) { + std::cerr << "Unable to parse IP address: " << argv[2] << std::endl; + return 1; + } + + rtc::PhysicalSocketServer socket_server; + rtc::AutoSocketServerThread main(&socket_server); + rtc::AsyncUDPSocket* int_socket = + rtc::AsyncUDPSocket::Create(&socket_server, int_addr); + if (!int_socket) { + std::cerr << "Failed to create a UDP socket bound at" << int_addr.ToString() + << std::endl; + return 1; + } + + cricket::TurnServer server(&main); + std::fstream auth_file(argv[4], std::fstream::in); + + TurnFileAuth auth(auth_file.is_open() + ? webrtc_examples::ReadAuthFile(&auth_file) + : std::map<std::string, std::string>()); + server.set_realm(argv[3]); + server.set_software(kSoftware); + server.set_auth_hook(&auth); + server.AddInternalSocket(int_socket, cricket::PROTO_UDP); + server.SetExternalSocketFactory( + new rtc::BasicPacketSocketFactory(&socket_server), + rtc::SocketAddress(ext_addr, 0)); + + std::cout << "Listening internally at " << int_addr.ToString() << std::endl; + + main.Run(); + return 0; +} diff --git a/third_party/libwebrtc/examples/unityplugin/ANDROID_INSTRUCTION b/third_party/libwebrtc/examples/unityplugin/ANDROID_INSTRUCTION new file mode 100644 index 0000000000..d5f7399bca --- /dev/null +++ b/third_party/libwebrtc/examples/unityplugin/ANDROID_INSTRUCTION @@ -0,0 +1,33 @@ +Instruction of running webrtc_unity_plugin on Android Unity + +1. On Linux machine, compile target webrtc_unity_plugin. + Checkout WebRTC codebase: fetch --nohooks webrtc_android + If you already have a checkout for linux, add target_os=”android” into .gclient file. + Run gclient sync + Run gn args out/Android, and again set target_os=”android” in the args.gn + Run ninja -C out/Android webrtc_unity_plugin + +2. On Linux machine, build target libwebrtc_unity under webrtc checkout. This is the java code for webrtc to work on Android. + +3. Copy libwebrtc_unity.jar and libwebrtc_unity_plugin.so into Unity project folder, under Assets/Plugins/Android folder. + +4. Rename libwebrtc_unity_plugin.so to libjingle_peerconnection_so.so. This is hacky, and the purpose is to let the java code in libwebrtc_unity.jar to find their JNI implementations. Simultaneously, in your C# wrapper script for the native plugin libjingle_peerconnection_so.so, the dll_path should be set to “jingle_peerconnection_so”. + +5. In the Unity Main Scene’s Start method, write the following code to initialize the Java environment for webrtc (otherwise, webrtc will not be able to access audio device or camera from C++ code): + +#if UNITY_ANDROID + AndroidJavaClass playerClass = new AndroidJavaClass("com.unity3d.player.UnityPlayer"); + AndroidJavaObject activity = playerClass.GetStatic<AndroidJavaObject>("currentActivity"); + AndroidJavaClass utilityClass = new AndroidJavaClass("org.webrtc.UnityUtility"); + utilityClass.CallStatic("InitializePeerConncectionFactory", new object[1] { activity }); +#endif + +6. Compile the unity project into an APK, and decompile the apk using apktool that you can download from https://ibotpeaches.github.io/Apktool/ + Run apktool d apkname.apk. +Then copy the AndroidManifest.xml in the decompiled folder to the Assets/Plugins/Android folder, and add two lines: + <uses-permission android:name="android.permission.RECORD_AUDIO" /> + <uses-permission android:name="android.permission.CAMERA" /> + +The purpose of using apktool is to get a well-written android manifest xml file. If you know how to write manifest file from scratch, you can skip using apktool. + +7. Compile the unity project into an APK again and deploy it to an android device. diff --git a/third_party/libwebrtc/examples/unityplugin/DEPS b/third_party/libwebrtc/examples/unityplugin/DEPS new file mode 100644 index 0000000000..604005ac73 --- /dev/null +++ b/third_party/libwebrtc/examples/unityplugin/DEPS @@ -0,0 +1,4 @@ +include_rules = [ + "+modules/utility", + "+sdk", +] diff --git a/third_party/libwebrtc/examples/unityplugin/README b/third_party/libwebrtc/examples/unityplugin/README new file mode 100644 index 0000000000..da8f07aa11 --- /dev/null +++ b/third_party/libwebrtc/examples/unityplugin/README @@ -0,0 +1,309 @@ +This directory contains an example Unity native plugin for Windows OS and Android. + +The APIs use Platform Invoke (P/Invoke) technology as required by Unity native plugin. +This plugin dll can also be used by Windows C# applications other than Unity. + +For detailed build instruction on Android, see ANDROID_INSTRUCTION + +An example of wrapping native plugin into a C# managed class in Unity is given as following: + +using System; +using System.Collections.Generic; +using System.Runtime.InteropServices; + +namespace SimplePeerConnectionM { + // A class for ice candidate. + public class IceCandidate { + public IceCandidate(string candidate, int sdpMlineIndex, string sdpMid) { + mCandidate = candidate; + mSdpMlineIndex = sdpMlineIndex; + mSdpMid = sdpMid; + } + string mCandidate; + int mSdpMlineIndex; + string mSdpMid; + + public string Candidate { + get { return mCandidate; } + set { mCandidate = value; } + } + + public int SdpMlineIndex { + get { return mSdpMlineIndex; } + set { mSdpMlineIndex = value; } + } + + public string SdpMid { + get { return mSdpMid; } + set { mSdpMid = value; } + } + } + + // A managed wrapper up class for the native c style peer connection APIs. + public class PeerConnectionM { + private const string dllPath = "webrtc_unity_plugin"; + + //create a peerconnection with turn servers + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern int CreatePeerConnection(string[] turnUrls, int noOfUrls, + string username, string credential); + + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool ClosePeerConnection(int peerConnectionId); + + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool AddStream(int peerConnectionId, bool audioOnly); + + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool AddDataChannel(int peerConnectionId); + + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool CreateOffer(int peerConnectionId); + + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool CreateAnswer(int peerConnectionId); + + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool SendDataViaDataChannel(int peerConnectionId, string data); + + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool SetAudioControl(int peerConnectionId, bool isMute, bool isRecord); + + [UnmanagedFunctionPointer(CallingConvention.Cdecl)] + private delegate void LocalDataChannelReadyInternalDelegate(); + public delegate void LocalDataChannelReadyDelegate(int id); + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool RegisterOnLocalDataChannelReady( + int peerConnectionId, LocalDataChannelReadyInternalDelegate callback); + + [UnmanagedFunctionPointer(CallingConvention.Cdecl)] + private delegate void DataFromDataChannelReadyInternalDelegate(string s); + public delegate void DataFromDataChannelReadyDelegate(int id, string s); + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool RegisterOnDataFromDataChannelReady( + int peerConnectionId, DataFromDataChannelReadyInternalDelegate callback); + + [UnmanagedFunctionPointer(CallingConvention.Cdecl)] + private delegate void FailureMessageInternalDelegate(string msg); + public delegate void FailureMessageDelegate(int id, string msg); + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool RegisterOnFailure(int peerConnectionId, + FailureMessageInternalDelegate callback); + + [UnmanagedFunctionPointer(CallingConvention.Cdecl)] + private delegate void AudioBusReadyInternalDelegate(IntPtr data, int bitsPerSample, + int sampleRate, int numberOfChannels, int numberOfFrames); + public delegate void AudioBusReadyDelegate(int id, IntPtr data, int bitsPerSample, + int sampleRate, int numberOfChannels, int numberOfFrames); + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool RegisterOnAudioBusReady(int peerConnectionId, + AudioBusReadyInternalDelegate callback); + + // Video callbacks. + [UnmanagedFunctionPointer(CallingConvention.Cdecl)] + private delegate void I420FrameReadyInternalDelegate( + IntPtr dataY, IntPtr dataU, IntPtr dataV, + int strideY, int strideU, int strideV, + uint width, uint height); + public delegate void I420FrameReadyDelegate(int id, + IntPtr dataY, IntPtr dataU, IntPtr dataV, + int strideY, int strideU, int strideV, + uint width, uint height); + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool RegisterOnLocalI420FrameReady(int peerConnectionId, + I420FrameReadyInternalDelegate callback); + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool RegisterOnRemoteI420FrameReady(int peerConnectionId, + I420FrameReadyInternalDelegate callback); + + [UnmanagedFunctionPointer(CallingConvention.Cdecl)] + private delegate void LocalSdpReadytoSendInternalDelegate(string type, string sdp); + public delegate void LocalSdpReadytoSendDelegate(int id, string type, string sdp); + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool RegisterOnLocalSdpReadytoSend(int peerConnectionId, + LocalSdpReadytoSendInternalDelegate callback); + + [UnmanagedFunctionPointer(CallingConvention.Cdecl)] + private delegate void IceCandidateReadytoSendInternalDelegate( + string candidate, int sdpMlineIndex, string sdpMid); + public delegate void IceCandidateReadytoSendDelegate( + int id, string candidate, int sdpMlineIndex, string sdpMid); + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool RegisterOnIceCandidateReadytoSend( + int peerConnectionId, IceCandidateReadytoSendInternalDelegate callback); + + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool SetRemoteDescription(int peerConnectionId, string type, string sdp); + + [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)] + private static extern bool AddIceCandidate(int peerConnectionId, string sdp, + int sdpMlineindex, string sdpMid); + + public PeerConnectionM(List<string> turnUrls, string username, string credential) { + string[] urls = turnUrls != null ? turnUrls.ToArray() : null; + int length = turnUrls != null ? turnUrls.Count : 0; + mPeerConnectionId = CreatePeerConnection(urls, length, username, credential); + RegisterCallbacks(); + } + + public void ClosePeerConnection() { + ClosePeerConnection(mPeerConnectionId); + mPeerConnectionId = -1; + } + + // Return -1 if Peerconnection is not available. + public int GetUniqueId() { + return mPeerConnectionId; + } + + public void AddStream(bool audioOnly) { + AddStream(mPeerConnectionId, audioOnly); + } + + public void AddDataChannel() { + AddDataChannel(mPeerConnectionId); + } + + public void CreateOffer() { + CreateOffer(mPeerConnectionId); + } + + public void CreateAnswer() { + CreateAnswer(mPeerConnectionId); + } + + public void SendDataViaDataChannel(string data) { + SendDataViaDataChannel(mPeerConnectionId, data); + } + + public void SetAudioControl(bool isMute, bool isRecord) { + SetAudioControl(mPeerConnectionId, isMute, isRecord); + } + + public void SetRemoteDescription(string type, string sdp) { + SetRemoteDescription(mPeerConnectionId, type, sdp); + } + + public void AddIceCandidate(string candidate, int sdpMlineindex, string sdpMid) { + AddIceCandidate(mPeerConnectionId, candidate, sdpMlineindex, sdpMid); + } + + private void RegisterCallbacks() { + localDataChannelReadyDelegate = new LocalDataChannelReadyInternalDelegate( + RaiseLocalDataChannelReady); + RegisterOnLocalDataChannelReady(mPeerConnectionId, localDataChannelReadyDelegate); + + dataFromDataChannelReadyDelegate = new DataFromDataChannelReadyInternalDelegate( + RaiseDataFromDataChannelReady); + RegisterOnDataFromDataChannelReady(mPeerConnectionId, dataFromDataChannelReadyDelegate); + + failureMessageDelegate = new FailureMessageInternalDelegate(RaiseFailureMessage); + RegisterOnFailure(mPeerConnectionId, failureMessageDelegate); + + audioBusReadyDelegate = new AudioBusReadyInternalDelegate(RaiseAudioBusReady); + RegisterOnAudioBusReady(mPeerConnectionId, audioBusReadyDelegate); + + localI420FrameReadyDelegate = new I420FrameReadyInternalDelegate( + RaiseLocalVideoFrameReady); + RegisterOnLocalI420FrameReady(mPeerConnectionId, localI420FrameReadyDelegate); + + remoteI420FrameReadyDelegate = new I420FrameReadyInternalDelegate( + RaiseRemoteVideoFrameReady); + RegisterOnRemoteI420FrameReady(mPeerConnectionId, remoteI420FrameReadyDelegate); + + localSdpReadytoSendDelegate = new LocalSdpReadytoSendInternalDelegate( + RaiseLocalSdpReadytoSend); + RegisterOnLocalSdpReadytoSend(mPeerConnectionId, localSdpReadytoSendDelegate); + + iceCandidateReadytoSendDelegate = + new IceCandidateReadytoSendInternalDelegate(RaiseIceCandidateReadytoSend); + RegisterOnIceCandidateReadytoSend( + mPeerConnectionId, iceCandidateReadytoSendDelegate); + } + + private void RaiseLocalDataChannelReady() { + if (OnLocalDataChannelReady != null) + OnLocalDataChannelReady(mPeerConnectionId); + } + + private void RaiseDataFromDataChannelReady(string data) { + if (OnDataFromDataChannelReady != null) + OnDataFromDataChannelReady(mPeerConnectionId, data); + } + + private void RaiseFailureMessage(string msg) { + if (OnFailureMessage != null) + OnFailureMessage(mPeerConnectionId, msg); + } + + private void RaiseAudioBusReady(IntPtr data, int bitsPerSample, + int sampleRate, int numberOfChannels, int numberOfFrames) { + if (OnAudioBusReady != null) + OnAudioBusReady(mPeerConnectionId, data, bitsPerSample, sampleRate, + numberOfChannels, numberOfFrames); + } + + private void RaiseLocalVideoFrameReady( + IntPtr dataY, IntPtr dataU, IntPtr dataV, + int strideY, int strideU, int strideV, + uint width, uint height) { + if (OnLocalVideoFrameReady != null) + OnLocalVideoFrameReady(mPeerConnectionId, dataY, dataU, dataV, strideY, strideU, strideV, + width, height); + } + + private void RaiseRemoteVideoFrameReady( + IntPtr dataY, IntPtr dataU, IntPtr dataV, + int strideY, int strideU, int strideV, + uint width, uint height) { + if (OnRemoteVideoFrameReady != null) + OnRemoteVideoFrameReady(mPeerConnectionId, dataY, dataU, dataV, strideY, strideU, strideV, + width, height); + } + + + private void RaiseLocalSdpReadytoSend(string type, string sdp) { + if (OnLocalSdpReadytoSend != null) + OnLocalSdpReadytoSend(mPeerConnectionId, type, sdp); + } + + private void RaiseIceCandidateReadytoSend(string candidate, int sdpMlineIndex, string sdpMid) { + if (OnIceCandidateReadytoSend != null) + OnIceCandidateReadytoSend(mPeerConnectionId, candidate, sdpMlineIndex, sdpMid); + } + + public void AddQueuedIceCandidate(List<IceCandidate> iceCandidateQueue) { + if (iceCandidateQueue != null) { + foreach (IceCandidate ic in iceCandidateQueue) { + AddIceCandidate(mPeerConnectionId, ic.Candidate, ic.SdpMlineIndex, ic.SdpMid); + } + } + } + + private LocalDataChannelReadyInternalDelegate localDataChannelReadyDelegate = null; + public event LocalDataChannelReadyDelegate OnLocalDataChannelReady; + + private DataFromDataChannelReadyInternalDelegate dataFromDataChannelReadyDelegate = null; + public event DataFromDataChannelReadyDelegate OnDataFromDataChannelReady; + + private FailureMessageInternalDelegate failureMessageDelegate = null; + public event FailureMessageDelegate OnFailureMessage; + + private AudioBusReadyInternalDelegate audioBusReadyDelegate = null; + public event AudioBusReadyDelegate OnAudioBusReady; + + private I420FrameReadyInternalDelegate localI420FrameReadyDelegate = null; + public event I420FrameReadyDelegate OnLocalVideoFrameReady; + + private I420FrameReadyInternalDelegate remoteI420FrameReadyDelegate = null; + public event I420FrameReadyDelegate OnRemoteVideoFrameReady; + + private LocalSdpReadytoSendInternalDelegate localSdpReadytoSendDelegate = null; + public event LocalSdpReadytoSendDelegate OnLocalSdpReadytoSend; + + private IceCandidateReadytoSendInternalDelegate iceCandidateReadytoSendDelegate = null; + public event IceCandidateReadytoSendDelegate OnIceCandidateReadytoSend; + + private int mPeerConnectionId = -1; + } +} diff --git a/third_party/libwebrtc/examples/unityplugin/class_reference_holder.cc b/third_party/libwebrtc/examples/unityplugin/class_reference_holder.cc new file mode 100644 index 0000000000..00ca772e76 --- /dev/null +++ b/third_party/libwebrtc/examples/unityplugin/class_reference_holder.cc @@ -0,0 +1,88 @@ +/* + * Copyright 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#include "examples/unityplugin/class_reference_holder.h" + +#include <utility> + +#include "sdk/android/src/jni/jni_helpers.h" + +namespace unity_plugin { + +// ClassReferenceHolder holds global reference to Java classes in app/webrtc. +class ClassReferenceHolder { + public: + explicit ClassReferenceHolder(JNIEnv* jni); + ~ClassReferenceHolder(); + + void FreeReferences(JNIEnv* jni); + jclass GetClass(const std::string& name); + + void LoadClass(JNIEnv* jni, const std::string& name); + + private: + std::map<std::string, jclass> classes_; +}; + +// Allocated in LoadGlobalClassReferenceHolder(), +// freed in FreeGlobalClassReferenceHolder(). +static ClassReferenceHolder* g_class_reference_holder = nullptr; + +void LoadGlobalClassReferenceHolder() { + RTC_CHECK(g_class_reference_holder == nullptr); + g_class_reference_holder = new ClassReferenceHolder(webrtc::jni::GetEnv()); +} + +void FreeGlobalClassReferenceHolder() { + g_class_reference_holder->FreeReferences( + webrtc::jni::AttachCurrentThreadIfNeeded()); + delete g_class_reference_holder; + g_class_reference_holder = nullptr; +} + +ClassReferenceHolder::ClassReferenceHolder(JNIEnv* jni) { + LoadClass(jni, "org/webrtc/UnityUtility"); +} + +ClassReferenceHolder::~ClassReferenceHolder() { + RTC_CHECK(classes_.empty()) << "Must call FreeReferences() before dtor!"; +} + +void ClassReferenceHolder::FreeReferences(JNIEnv* jni) { + for (std::map<std::string, jclass>::const_iterator it = classes_.begin(); + it != classes_.end(); ++it) { + jni->DeleteGlobalRef(it->second); + } + classes_.clear(); +} + +jclass ClassReferenceHolder::GetClass(const std::string& name) { + std::map<std::string, jclass>::iterator it = classes_.find(name); + RTC_CHECK(it != classes_.end()) << "Unexpected GetClass() call for: " << name; + return it->second; +} + +void ClassReferenceHolder::LoadClass(JNIEnv* jni, const std::string& name) { + jclass localRef = jni->FindClass(name.c_str()); + CHECK_EXCEPTION(jni) << "error during FindClass: " << name; + RTC_CHECK(localRef) << name; + jclass globalRef = reinterpret_cast<jclass>(jni->NewGlobalRef(localRef)); + CHECK_EXCEPTION(jni) << "error during NewGlobalRef: " << name; + RTC_CHECK(globalRef) << name; + bool inserted = classes_.insert(std::make_pair(name, globalRef)).second; + RTC_CHECK(inserted) << "Duplicate class name: " << name; +} + +// Returns a global reference guaranteed to be valid for the lifetime of the +// process. +jclass FindClass(JNIEnv* jni, const char* name) { + return g_class_reference_holder->GetClass(name); +} + +} // namespace unity_plugin diff --git a/third_party/libwebrtc/examples/unityplugin/class_reference_holder.h b/third_party/libwebrtc/examples/unityplugin/class_reference_holder.h new file mode 100644 index 0000000000..884d471ceb --- /dev/null +++ b/third_party/libwebrtc/examples/unityplugin/class_reference_holder.h @@ -0,0 +1,38 @@ +/* + * Copyright 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// This is a supplement of webrtc::jni::ClassReferenceHolder. +// The purpose of this ClassReferenceHolder is to load the example +// specific java class into JNI c++ side, so that our c++ code can +// call those java functions. + +#ifndef EXAMPLES_UNITYPLUGIN_CLASS_REFERENCE_HOLDER_H_ +#define EXAMPLES_UNITYPLUGIN_CLASS_REFERENCE_HOLDER_H_ + +#include <jni.h> + +#include <map> +#include <string> +#include <vector> + +namespace unity_plugin { + +// LoadGlobalClassReferenceHolder must be called in JNI_OnLoad. +void LoadGlobalClassReferenceHolder(); +// FreeGlobalClassReferenceHolder must be called in JNI_UnLoad. +void FreeGlobalClassReferenceHolder(); + +// Returns a global reference guaranteed to be valid for the lifetime of the +// process. +jclass FindClass(JNIEnv* jni, const char* name); + +} // namespace unity_plugin + +#endif // EXAMPLES_UNITYPLUGIN_CLASS_REFERENCE_HOLDER_H_ diff --git a/third_party/libwebrtc/examples/unityplugin/java/src/org/webrtc/UnityUtility.java b/third_party/libwebrtc/examples/unityplugin/java/src/org/webrtc/UnityUtility.java new file mode 100644 index 0000000000..bd8bbfa449 --- /dev/null +++ b/third_party/libwebrtc/examples/unityplugin/java/src/org/webrtc/UnityUtility.java @@ -0,0 +1,68 @@ +/* + * Copyright 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.webrtc; + +import android.content.Context; +import androidx.annotation.Nullable; + +public class UnityUtility { + private static final String VIDEO_CAPTURER_THREAD_NAME = "VideoCapturerThread"; + + public static SurfaceTextureHelper LoadSurfaceTextureHelper() { + final SurfaceTextureHelper surfaceTextureHelper = + SurfaceTextureHelper.create(VIDEO_CAPTURER_THREAD_NAME, null); + return surfaceTextureHelper; + } + + private static boolean useCamera2() { + return Camera2Enumerator.isSupported(ContextUtils.getApplicationContext()); + } + + private static @Nullable VideoCapturer createCameraCapturer(CameraEnumerator enumerator) { + final String[] deviceNames = enumerator.getDeviceNames(); + + for (String deviceName : deviceNames) { + if (enumerator.isFrontFacing(deviceName)) { + VideoCapturer videoCapturer = enumerator.createCapturer(deviceName, null); + + if (videoCapturer != null) { + return videoCapturer; + } + } + } + + return null; + } + + public static VideoCapturer LinkCamera( + long nativeTrackSource, SurfaceTextureHelper surfaceTextureHelper) { + VideoCapturer capturer = + createCameraCapturer(new Camera2Enumerator(ContextUtils.getApplicationContext())); + + VideoSource videoSource = new VideoSource(nativeTrackSource); + + capturer.initialize(surfaceTextureHelper, ContextUtils.getApplicationContext(), + videoSource.getCapturerObserver()); + + capturer.startCapture(720, 480, 30); + return capturer; + } + + public static void StopCamera(VideoCapturer camera) throws InterruptedException { + camera.stopCapture(); + camera.dispose(); + } + + public static void InitializePeerConncectionFactory(Context context) throws InterruptedException { + PeerConnectionFactory.initialize( + PeerConnectionFactory.InitializationOptions.builder(context).createInitializationOptions()); + } +} diff --git a/third_party/libwebrtc/examples/unityplugin/jni_onload.cc b/third_party/libwebrtc/examples/unityplugin/jni_onload.cc new file mode 100644 index 0000000000..b9c92d5ef4 --- /dev/null +++ b/third_party/libwebrtc/examples/unityplugin/jni_onload.cc @@ -0,0 +1,42 @@ +/* + * Copyright 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <jni.h> +#undef JNIEXPORT +#define JNIEXPORT __attribute__((visibility("default"))) + +#include "examples/unityplugin/class_reference_holder.h" +#include "rtc_base/ssl_adapter.h" +#include "sdk/android/native_api/jni/class_loader.h" +#include "sdk/android/src/jni/jni_helpers.h" + +namespace webrtc { +namespace jni { + +extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM* jvm, void* reserved) { + jint ret = InitGlobalJniVariables(jvm); + RTC_DCHECK_GE(ret, 0); + if (ret < 0) + return -1; + + RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()"; + webrtc::InitClassLoader(GetEnv()); + unity_plugin::LoadGlobalClassReferenceHolder(); + + return ret; +} + +extern "C" void JNIEXPORT JNICALL JNI_OnUnLoad(JavaVM* jvm, void* reserved) { + unity_plugin::FreeGlobalClassReferenceHolder(); + RTC_CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()"; +} + +} // namespace jni +} // namespace webrtc diff --git a/third_party/libwebrtc/examples/unityplugin/simple_peer_connection.cc b/third_party/libwebrtc/examples/unityplugin/simple_peer_connection.cc new file mode 100644 index 0000000000..861b22f29c --- /dev/null +++ b/third_party/libwebrtc/examples/unityplugin/simple_peer_connection.cc @@ -0,0 +1,587 @@ +/* + * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/unityplugin/simple_peer_connection.h" + +#include <utility> + +#include "absl/memory/memory.h" +#include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include "api/create_peerconnection_factory.h" +#include "media/engine/internal_decoder_factory.h" +#include "media/engine/internal_encoder_factory.h" +#include "media/engine/multiplex_codec_factory.h" +#include "modules/audio_device/include/audio_device.h" +#include "modules/audio_processing/include/audio_processing.h" +#include "modules/video_capture/video_capture_factory.h" +#include "pc/video_track_source.h" +#include "test/vcm_capturer.h" + +#if defined(WEBRTC_ANDROID) +#include "examples/unityplugin/class_reference_holder.h" +#include "modules/utility/include/helpers_android.h" +#include "sdk/android/src/jni/android_video_track_source.h" +#include "sdk/android/src/jni/jni_helpers.h" +#endif + +// Names used for media stream ids. +const char kAudioLabel[] = "audio_label"; +const char kVideoLabel[] = "video_label"; +const char kStreamId[] = "stream_id"; + +namespace { +static int g_peer_count = 0; +static std::unique_ptr<rtc::Thread> g_worker_thread; +static std::unique_ptr<rtc::Thread> g_signaling_thread; +static rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> + g_peer_connection_factory; +#if defined(WEBRTC_ANDROID) +// Android case: the video track does not own the capturer, and it +// relies on the app to dispose the capturer when the peerconnection +// shuts down. +static jobject g_camera = nullptr; +#else +class CapturerTrackSource : public webrtc::VideoTrackSource { + public: + static rtc::scoped_refptr<CapturerTrackSource> Create() { + const size_t kWidth = 640; + const size_t kHeight = 480; + const size_t kFps = 30; + const size_t kDeviceIndex = 0; + std::unique_ptr<webrtc::test::VcmCapturer> capturer = absl::WrapUnique( + webrtc::test::VcmCapturer::Create(kWidth, kHeight, kFps, kDeviceIndex)); + if (!capturer) { + return nullptr; + } + return rtc::make_ref_counted<CapturerTrackSource>(std::move(capturer)); + } + + protected: + explicit CapturerTrackSource( + std::unique_ptr<webrtc::test::VcmCapturer> capturer) + : VideoTrackSource(/*remote=*/false), capturer_(std::move(capturer)) {} + + private: + rtc::VideoSourceInterface<webrtc::VideoFrame>* source() override { + return capturer_.get(); + } + std::unique_ptr<webrtc::test::VcmCapturer> capturer_; +}; + +#endif + +std::string GetEnvVarOrDefault(const char* env_var_name, + const char* default_value) { + std::string value; + const char* env_var = getenv(env_var_name); + if (env_var) + value = env_var; + + if (value.empty()) + value = default_value; + + return value; +} + +std::string GetPeerConnectionString() { + return GetEnvVarOrDefault("WEBRTC_CONNECT", "stun:stun.l.google.com:19302"); +} + +class DummySetSessionDescriptionObserver + : public webrtc::SetSessionDescriptionObserver { + public: + static rtc::scoped_refptr<DummySetSessionDescriptionObserver> Create() { + return rtc::make_ref_counted<DummySetSessionDescriptionObserver>(); + } + virtual void OnSuccess() { RTC_LOG(LS_INFO) << __FUNCTION__; } + virtual void OnFailure(webrtc::RTCError error) { + RTC_LOG(LS_INFO) << __FUNCTION__ << " " << ToString(error.type()) << ": " + << error.message(); + } + + protected: + DummySetSessionDescriptionObserver() {} + ~DummySetSessionDescriptionObserver() {} +}; + +} // namespace + +bool SimplePeerConnection::InitializePeerConnection(const char** turn_urls, + const int no_of_urls, + const char* username, + const char* credential, + bool is_receiver) { + RTC_DCHECK(peer_connection_.get() == nullptr); + + if (g_peer_connection_factory == nullptr) { + g_worker_thread = rtc::Thread::Create(); + g_worker_thread->Start(); + g_signaling_thread = rtc::Thread::Create(); + g_signaling_thread->Start(); + + g_peer_connection_factory = webrtc::CreatePeerConnectionFactory( + g_worker_thread.get(), g_worker_thread.get(), g_signaling_thread.get(), + nullptr, webrtc::CreateBuiltinAudioEncoderFactory(), + webrtc::CreateBuiltinAudioDecoderFactory(), + std::unique_ptr<webrtc::VideoEncoderFactory>( + new webrtc::MultiplexEncoderFactory( + std::make_unique<webrtc::InternalEncoderFactory>())), + std::unique_ptr<webrtc::VideoDecoderFactory>( + new webrtc::MultiplexDecoderFactory( + std::make_unique<webrtc::InternalDecoderFactory>())), + nullptr, nullptr); + } + if (!g_peer_connection_factory.get()) { + DeletePeerConnection(); + return false; + } + + g_peer_count++; + if (!CreatePeerConnection(turn_urls, no_of_urls, username, credential)) { + DeletePeerConnection(); + return false; + } + + mandatory_receive_ = is_receiver; + return peer_connection_.get() != nullptr; +} + +bool SimplePeerConnection::CreatePeerConnection(const char** turn_urls, + const int no_of_urls, + const char* username, + const char* credential) { + RTC_DCHECK(g_peer_connection_factory.get() != nullptr); + RTC_DCHECK(peer_connection_.get() == nullptr); + + local_video_observer_.reset(new VideoObserver()); + remote_video_observer_.reset(new VideoObserver()); + + // Add the turn server. + if (turn_urls != nullptr) { + if (no_of_urls > 0) { + webrtc::PeerConnectionInterface::IceServer turn_server; + for (int i = 0; i < no_of_urls; i++) { + std::string url(turn_urls[i]); + if (url.length() > 0) + turn_server.urls.push_back(turn_urls[i]); + } + + std::string user_name(username); + if (user_name.length() > 0) + turn_server.username = username; + + std::string password(credential); + if (password.length() > 0) + turn_server.password = credential; + + config_.servers.push_back(turn_server); + } + } + + // Add the stun server. + webrtc::PeerConnectionInterface::IceServer stun_server; + stun_server.uri = GetPeerConnectionString(); + config_.servers.push_back(stun_server); + + auto result = g_peer_connection_factory->CreatePeerConnectionOrError( + config_, webrtc::PeerConnectionDependencies(this)); + if (!result.ok()) { + peer_connection_ = nullptr; + return false; + } + peer_connection_ = result.MoveValue(); + return true; +} + +void SimplePeerConnection::DeletePeerConnection() { + g_peer_count--; + +#if defined(WEBRTC_ANDROID) + if (g_camera) { + JNIEnv* env = webrtc::jni::GetEnv(); + jclass pc_factory_class = + unity_plugin::FindClass(env, "org/webrtc/UnityUtility"); + jmethodID stop_camera_method = webrtc::GetStaticMethodID( + env, pc_factory_class, "StopCamera", "(Lorg/webrtc/VideoCapturer;)V"); + + env->CallStaticVoidMethod(pc_factory_class, stop_camera_method, g_camera); + CHECK_EXCEPTION(env); + + g_camera = nullptr; + } +#endif + + CloseDataChannel(); + peer_connection_ = nullptr; + active_streams_.clear(); + + if (g_peer_count == 0) { + g_peer_connection_factory = nullptr; + g_signaling_thread.reset(); + g_worker_thread.reset(); + } +} + +bool SimplePeerConnection::CreateOffer() { + if (!peer_connection_.get()) + return false; + + webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options; + if (mandatory_receive_) { + options.offer_to_receive_audio = true; + options.offer_to_receive_video = true; + } + peer_connection_->CreateOffer(this, options); + return true; +} + +bool SimplePeerConnection::CreateAnswer() { + if (!peer_connection_.get()) + return false; + + webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options; + if (mandatory_receive_) { + options.offer_to_receive_audio = true; + options.offer_to_receive_video = true; + } + peer_connection_->CreateAnswer(this, options); + return true; +} + +void SimplePeerConnection::OnSuccess( + webrtc::SessionDescriptionInterface* desc) { + peer_connection_->SetLocalDescription( + DummySetSessionDescriptionObserver::Create().get(), desc); + + std::string sdp; + desc->ToString(&sdp); + + if (OnLocalSdpReady) + OnLocalSdpReady(desc->type().c_str(), sdp.c_str()); +} + +void SimplePeerConnection::OnFailure(webrtc::RTCError error) { + RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message(); + + // TODO(hta): include error.type in the message + if (OnFailureMessage) + OnFailureMessage(error.message()); +} + +void SimplePeerConnection::OnIceCandidate( + const webrtc::IceCandidateInterface* candidate) { + RTC_LOG(LS_INFO) << __FUNCTION__ << " " << candidate->sdp_mline_index(); + + std::string sdp; + if (!candidate->ToString(&sdp)) { + RTC_LOG(LS_ERROR) << "Failed to serialize candidate"; + return; + } + + if (OnIceCandidateReady) + OnIceCandidateReady(sdp.c_str(), candidate->sdp_mline_index(), + candidate->sdp_mid().c_str()); +} + +void SimplePeerConnection::RegisterOnLocalI420FrameReady( + I420FRAMEREADY_CALLBACK callback) { + if (local_video_observer_) + local_video_observer_->SetVideoCallback(callback); +} + +void SimplePeerConnection::RegisterOnRemoteI420FrameReady( + I420FRAMEREADY_CALLBACK callback) { + if (remote_video_observer_) + remote_video_observer_->SetVideoCallback(callback); +} + +void SimplePeerConnection::RegisterOnLocalDataChannelReady( + LOCALDATACHANNELREADY_CALLBACK callback) { + OnLocalDataChannelReady = callback; +} + +void SimplePeerConnection::RegisterOnDataFromDataChannelReady( + DATAFROMEDATECHANNELREADY_CALLBACK callback) { + OnDataFromDataChannelReady = callback; +} + +void SimplePeerConnection::RegisterOnFailure(FAILURE_CALLBACK callback) { + OnFailureMessage = callback; +} + +void SimplePeerConnection::RegisterOnAudioBusReady( + AUDIOBUSREADY_CALLBACK callback) { + OnAudioReady = callback; +} + +void SimplePeerConnection::RegisterOnLocalSdpReadytoSend( + LOCALSDPREADYTOSEND_CALLBACK callback) { + OnLocalSdpReady = callback; +} + +void SimplePeerConnection::RegisterOnIceCandidateReadytoSend( + ICECANDIDATEREADYTOSEND_CALLBACK callback) { + OnIceCandidateReady = callback; +} + +bool SimplePeerConnection::SetRemoteDescription(const char* type, + const char* sdp) { + if (!peer_connection_) + return false; + + std::string remote_desc(sdp); + std::string desc_type(type); + webrtc::SdpParseError error; + webrtc::SessionDescriptionInterface* session_description( + webrtc::CreateSessionDescription(desc_type, remote_desc, &error)); + if (!session_description) { + RTC_LOG(LS_WARNING) << "Can't parse received session description message. " + "SdpParseError was: " + << error.description; + return false; + } + RTC_LOG(LS_INFO) << " Received session description :" << remote_desc; + peer_connection_->SetRemoteDescription( + DummySetSessionDescriptionObserver::Create().get(), session_description); + + return true; +} + +bool SimplePeerConnection::AddIceCandidate(const char* candidate, + const int sdp_mlineindex, + const char* sdp_mid) { + if (!peer_connection_) + return false; + + webrtc::SdpParseError error; + std::unique_ptr<webrtc::IceCandidateInterface> ice_candidate( + webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, candidate, &error)); + if (!ice_candidate.get()) { + RTC_LOG(LS_WARNING) << "Can't parse received candidate message. " + "SdpParseError was: " + << error.description; + return false; + } + if (!peer_connection_->AddIceCandidate(ice_candidate.get())) { + RTC_LOG(LS_WARNING) << "Failed to apply the received candidate"; + return false; + } + RTC_LOG(LS_INFO) << " Received candidate :" << candidate; + return true; +} + +void SimplePeerConnection::SetAudioControl(bool is_mute, bool is_record) { + is_mute_audio_ = is_mute; + is_record_audio_ = is_record; + + SetAudioControl(); +} + +void SimplePeerConnection::SetAudioControl() { + if (!remote_stream_) + return; + webrtc::AudioTrackVector tracks = remote_stream_->GetAudioTracks(); + if (tracks.empty()) + return; + + rtc::scoped_refptr<webrtc::AudioTrackInterface>& audio_track = tracks[0]; + if (is_record_audio_) + audio_track->AddSink(this); + else + audio_track->RemoveSink(this); + + for (auto& track : tracks) { + if (is_mute_audio_) + track->set_enabled(false); + else + track->set_enabled(true); + } +} + +void SimplePeerConnection::OnAddStream( + rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) { + RTC_LOG(LS_INFO) << __FUNCTION__ << " " << stream->id(); + remote_stream_ = stream; + if (remote_video_observer_ && !remote_stream_->GetVideoTracks().empty()) { + remote_stream_->GetVideoTracks()[0]->AddOrUpdateSink( + remote_video_observer_.get(), rtc::VideoSinkWants()); + } + SetAudioControl(); +} + +void SimplePeerConnection::AddStreams(bool audio_only) { + if (active_streams_.find(kStreamId) != active_streams_.end()) + return; // Already added. + + rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = + g_peer_connection_factory->CreateLocalMediaStream(kStreamId); + + rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( + g_peer_connection_factory->CreateAudioTrack( + kAudioLabel, + g_peer_connection_factory->CreateAudioSource(cricket::AudioOptions()) + .get())); + stream->AddTrack(audio_track); + + if (!audio_only) { +#if defined(WEBRTC_ANDROID) + JNIEnv* env = webrtc::jni::GetEnv(); + jclass pc_factory_class = + unity_plugin::FindClass(env, "org/webrtc/UnityUtility"); + jmethodID load_texture_helper_method = webrtc::GetStaticMethodID( + env, pc_factory_class, "LoadSurfaceTextureHelper", + "()Lorg/webrtc/SurfaceTextureHelper;"); + jobject texture_helper = env->CallStaticObjectMethod( + pc_factory_class, load_texture_helper_method); + CHECK_EXCEPTION(env); + RTC_DCHECK(texture_helper != nullptr) + << "Cannot get the Surface Texture Helper."; + + auto source = rtc::make_ref_counted<webrtc::jni::AndroidVideoTrackSource>( + g_signaling_thread.get(), env, /*is_screencast=*/false, + /*align_timestamps=*/true); + + // link with VideoCapturer (Camera); + jmethodID link_camera_method = webrtc::GetStaticMethodID( + env, pc_factory_class, "LinkCamera", + "(JLorg/webrtc/SurfaceTextureHelper;)Lorg/webrtc/VideoCapturer;"); + jobject camera_tmp = + env->CallStaticObjectMethod(pc_factory_class, link_camera_method, + (jlong)source.get(), texture_helper); + CHECK_EXCEPTION(env); + g_camera = (jobject)env->NewGlobalRef(camera_tmp); + + rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track( + g_peer_connection_factory->CreateVideoTrack(kVideoLabel, + source.release())); + stream->AddTrack(video_track); +#else + rtc::scoped_refptr<CapturerTrackSource> video_device = + CapturerTrackSource::Create(); + if (video_device) { + rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track( + g_peer_connection_factory->CreateVideoTrack(kVideoLabel, + video_device.get())); + + stream->AddTrack(video_track); + } +#endif + if (local_video_observer_ && !stream->GetVideoTracks().empty()) { + stream->GetVideoTracks()[0]->AddOrUpdateSink(local_video_observer_.get(), + rtc::VideoSinkWants()); + } + } + + if (!peer_connection_->AddStream(stream.get())) { + RTC_LOG(LS_ERROR) << "Adding stream to PeerConnection failed"; + } + + typedef std::pair<std::string, + rtc::scoped_refptr<webrtc::MediaStreamInterface>> + MediaStreamPair; + active_streams_.insert(MediaStreamPair(stream->id(), stream)); +} + +bool SimplePeerConnection::CreateDataChannel() { + struct webrtc::DataChannelInit init; + init.ordered = true; + init.reliable = true; + auto result = peer_connection_->CreateDataChannelOrError("Hello", &init); + if (result.ok()) { + data_channel_ = result.MoveValue(); + data_channel_->RegisterObserver(this); + RTC_LOG(LS_INFO) << "Succeeds to create data channel"; + return true; + } else { + RTC_LOG(LS_INFO) << "Fails to create data channel"; + return false; + } +} + +void SimplePeerConnection::CloseDataChannel() { + if (data_channel_.get()) { + data_channel_->UnregisterObserver(); + data_channel_->Close(); + } + data_channel_ = nullptr; +} + +bool SimplePeerConnection::SendDataViaDataChannel(const std::string& data) { + if (!data_channel_.get()) { + RTC_LOG(LS_INFO) << "Data channel is not established"; + return false; + } + webrtc::DataBuffer buffer(data); + data_channel_->Send(buffer); + return true; +} + +// Peerconnection observer +void SimplePeerConnection::OnDataChannel( + rtc::scoped_refptr<webrtc::DataChannelInterface> channel) { + channel->RegisterObserver(this); +} + +void SimplePeerConnection::OnStateChange() { + if (data_channel_) { + webrtc::DataChannelInterface::DataState state = data_channel_->state(); + if (state == webrtc::DataChannelInterface::kOpen) { + if (OnLocalDataChannelReady) + OnLocalDataChannelReady(); + RTC_LOG(LS_INFO) << "Data channel is open"; + } + } +} + +// A data buffer was successfully received. +void SimplePeerConnection::OnMessage(const webrtc::DataBuffer& buffer) { + size_t size = buffer.data.size(); + char* msg = new char[size + 1]; + memcpy(msg, buffer.data.data(), size); + msg[size] = 0; + if (OnDataFromDataChannelReady) + OnDataFromDataChannelReady(msg); + delete[] msg; +} + +// AudioTrackSinkInterface implementation. +void SimplePeerConnection::OnData(const void* audio_data, + int bits_per_sample, + int sample_rate, + size_t number_of_channels, + size_t number_of_frames) { + if (OnAudioReady) + OnAudioReady(audio_data, bits_per_sample, sample_rate, + static_cast<int>(number_of_channels), + static_cast<int>(number_of_frames)); +} + +std::vector<uint32_t> SimplePeerConnection::GetRemoteAudioTrackSsrcs() { + std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>> receivers = + peer_connection_->GetReceivers(); + + std::vector<uint32_t> ssrcs; + for (const auto& receiver : receivers) { + if (receiver->media_type() != cricket::MEDIA_TYPE_AUDIO) + continue; + + std::vector<webrtc::RtpEncodingParameters> params = + receiver->GetParameters().encodings; + + for (const auto& param : params) { + uint32_t ssrc = param.ssrc.value_or(0); + if (ssrc > 0) + ssrcs.push_back(ssrc); + } + } + + return ssrcs; +} diff --git a/third_party/libwebrtc/examples/unityplugin/simple_peer_connection.h b/third_party/libwebrtc/examples/unityplugin/simple_peer_connection.h new file mode 100644 index 0000000000..de652ef118 --- /dev/null +++ b/third_party/libwebrtc/examples/unityplugin/simple_peer_connection.h @@ -0,0 +1,135 @@ +/* + * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_ +#define EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_ + +#include <map> +#include <memory> +#include <string> +#include <vector> + +#include "api/data_channel_interface.h" +#include "api/media_stream_interface.h" +#include "api/peer_connection_interface.h" +#include "examples/unityplugin/unity_plugin_apis.h" +#include "examples/unityplugin/video_observer.h" + +class SimplePeerConnection : public webrtc::PeerConnectionObserver, + public webrtc::CreateSessionDescriptionObserver, + public webrtc::DataChannelObserver, + public webrtc::AudioTrackSinkInterface { + public: + SimplePeerConnection() {} + ~SimplePeerConnection() {} + + bool InitializePeerConnection(const char** turn_urls, + int no_of_urls, + const char* username, + const char* credential, + bool is_receiver); + void DeletePeerConnection(); + void AddStreams(bool audio_only); + bool CreateDataChannel(); + bool CreateOffer(); + bool CreateAnswer(); + bool SendDataViaDataChannel(const std::string& data); + void SetAudioControl(bool is_mute, bool is_record); + + // Register callback functions. + void RegisterOnLocalI420FrameReady(I420FRAMEREADY_CALLBACK callback); + void RegisterOnRemoteI420FrameReady(I420FRAMEREADY_CALLBACK callback); + void RegisterOnLocalDataChannelReady(LOCALDATACHANNELREADY_CALLBACK callback); + void RegisterOnDataFromDataChannelReady( + DATAFROMEDATECHANNELREADY_CALLBACK callback); + void RegisterOnFailure(FAILURE_CALLBACK callback); + void RegisterOnAudioBusReady(AUDIOBUSREADY_CALLBACK callback); + void RegisterOnLocalSdpReadytoSend(LOCALSDPREADYTOSEND_CALLBACK callback); + void RegisterOnIceCandidateReadytoSend( + ICECANDIDATEREADYTOSEND_CALLBACK callback); + bool SetRemoteDescription(const char* type, const char* sdp); + bool AddIceCandidate(const char* sdp, + int sdp_mlineindex, + const char* sdp_mid); + + protected: + // create a peerconneciton and add the turn servers info to the configuration. + bool CreatePeerConnection(const char** turn_urls, + int no_of_urls, + const char* username, + const char* credential); + void CloseDataChannel(); + void SetAudioControl(); + + // PeerConnectionObserver implementation. + void OnSignalingChange( + webrtc::PeerConnectionInterface::SignalingState new_state) override {} + void OnAddStream( + rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override; + void OnRemoveStream( + rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override {} + void OnDataChannel( + rtc::scoped_refptr<webrtc::DataChannelInterface> channel) override; + void OnRenegotiationNeeded() override {} + void OnIceConnectionChange( + webrtc::PeerConnectionInterface::IceConnectionState new_state) override {} + void OnIceGatheringChange( + webrtc::PeerConnectionInterface::IceGatheringState new_state) override {} + void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override; + void OnIceConnectionReceivingChange(bool receiving) override {} + + // CreateSessionDescriptionObserver implementation. + void OnSuccess(webrtc::SessionDescriptionInterface* desc) override; + void OnFailure(webrtc::RTCError error) override; + + // DataChannelObserver implementation. + void OnStateChange() override; + void OnMessage(const webrtc::DataBuffer& buffer) override; + + // AudioTrackSinkInterface implementation. + void OnData(const void* audio_data, + int bits_per_sample, + int sample_rate, + size_t number_of_channels, + size_t number_of_frames) override; + + // Get remote audio tracks ssrcs. + std::vector<uint32_t> GetRemoteAudioTrackSsrcs(); + + private: + rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; + rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel_; + std::map<std::string, rtc::scoped_refptr<webrtc::MediaStreamInterface> > + active_streams_; + + std::unique_ptr<VideoObserver> local_video_observer_; + std::unique_ptr<VideoObserver> remote_video_observer_; + + rtc::scoped_refptr<webrtc::MediaStreamInterface> remote_stream_ = nullptr; + webrtc::PeerConnectionInterface::RTCConfiguration config_; + + LOCALDATACHANNELREADY_CALLBACK OnLocalDataChannelReady = nullptr; + DATAFROMEDATECHANNELREADY_CALLBACK OnDataFromDataChannelReady = nullptr; + FAILURE_CALLBACK OnFailureMessage = nullptr; + AUDIOBUSREADY_CALLBACK OnAudioReady = nullptr; + + LOCALSDPREADYTOSEND_CALLBACK OnLocalSdpReady = nullptr; + ICECANDIDATEREADYTOSEND_CALLBACK OnIceCandidateReady = nullptr; + + bool is_mute_audio_ = false; + bool is_record_audio_ = false; + bool mandatory_receive_ = false; + + // disallow copy-and-assign + SimplePeerConnection(const SimplePeerConnection&) = delete; + SimplePeerConnection& operator=(const SimplePeerConnection&) = delete; +}; + +#endif // EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_ diff --git a/third_party/libwebrtc/examples/unityplugin/unity_plugin_apis.cc b/third_party/libwebrtc/examples/unityplugin/unity_plugin_apis.cc new file mode 100644 index 0000000000..6e34d7e1e0 --- /dev/null +++ b/third_party/libwebrtc/examples/unityplugin/unity_plugin_apis.cc @@ -0,0 +1,196 @@ +/* + * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/unityplugin/unity_plugin_apis.h" + +#include <map> +#include <string> + +#include "examples/unityplugin/simple_peer_connection.h" + +namespace { +static int g_peer_connection_id = 1; +static std::map<int, rtc::scoped_refptr<SimplePeerConnection>> + g_peer_connection_map; +} // namespace + +int CreatePeerConnection(const char** turn_urls, + const int no_of_urls, + const char* username, + const char* credential, + bool mandatory_receive_video) { + g_peer_connection_map[g_peer_connection_id] = + rtc::make_ref_counted<SimplePeerConnection>(); + + if (!g_peer_connection_map[g_peer_connection_id]->InitializePeerConnection( + turn_urls, no_of_urls, username, credential, mandatory_receive_video)) + return -1; + + return g_peer_connection_id++; +} + +bool ClosePeerConnection(int peer_connection_id) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + g_peer_connection_map[peer_connection_id]->DeletePeerConnection(); + g_peer_connection_map.erase(peer_connection_id); + return true; +} + +bool AddStream(int peer_connection_id, bool audio_only) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + g_peer_connection_map[peer_connection_id]->AddStreams(audio_only); + return true; +} + +bool AddDataChannel(int peer_connection_id) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + return g_peer_connection_map[peer_connection_id]->CreateDataChannel(); +} + +bool CreateOffer(int peer_connection_id) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + return g_peer_connection_map[peer_connection_id]->CreateOffer(); +} + +bool CreateAnswer(int peer_connection_id) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + return g_peer_connection_map[peer_connection_id]->CreateAnswer(); +} + +bool SendDataViaDataChannel(int peer_connection_id, const char* data) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + std::string s(data); + g_peer_connection_map[peer_connection_id]->SendDataViaDataChannel(s); + + return true; +} + +bool SetAudioControl(int peer_connection_id, bool is_mute, bool is_record) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + g_peer_connection_map[peer_connection_id]->SetAudioControl(is_mute, + is_record); + return true; +} + +bool SetRemoteDescription(int peer_connection_id, + const char* type, + const char* sdp) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + return g_peer_connection_map[peer_connection_id]->SetRemoteDescription(type, + sdp); +} + +bool AddIceCandidate(const int peer_connection_id, + const char* candidate, + const int sdp_mlineindex, + const char* sdp_mid) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + return g_peer_connection_map[peer_connection_id]->AddIceCandidate( + candidate, sdp_mlineindex, sdp_mid); +} + +// Register callback functions. +bool RegisterOnLocalI420FrameReady(int peer_connection_id, + I420FRAMEREADY_CALLBACK callback) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + g_peer_connection_map[peer_connection_id]->RegisterOnLocalI420FrameReady( + callback); + return true; +} + +bool RegisterOnRemoteI420FrameReady(int peer_connection_id, + I420FRAMEREADY_CALLBACK callback) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + g_peer_connection_map[peer_connection_id]->RegisterOnRemoteI420FrameReady( + callback); + return true; +} + +bool RegisterOnLocalDataChannelReady(int peer_connection_id, + LOCALDATACHANNELREADY_CALLBACK callback) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + g_peer_connection_map[peer_connection_id]->RegisterOnLocalDataChannelReady( + callback); + return true; +} + +bool RegisterOnDataFromDataChannelReady( + int peer_connection_id, + DATAFROMEDATECHANNELREADY_CALLBACK callback) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + g_peer_connection_map[peer_connection_id]->RegisterOnDataFromDataChannelReady( + callback); + return true; +} + +bool RegisterOnFailure(int peer_connection_id, FAILURE_CALLBACK callback) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + g_peer_connection_map[peer_connection_id]->RegisterOnFailure(callback); + return true; +} + +bool RegisterOnAudioBusReady(int peer_connection_id, + AUDIOBUSREADY_CALLBACK callback) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + g_peer_connection_map[peer_connection_id]->RegisterOnAudioBusReady(callback); + return true; +} + +// Singnaling channel related functions. +bool RegisterOnLocalSdpReadytoSend(int peer_connection_id, + LOCALSDPREADYTOSEND_CALLBACK callback) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + g_peer_connection_map[peer_connection_id]->RegisterOnLocalSdpReadytoSend( + callback); + return true; +} + +bool RegisterOnIceCandidateReadytoSend( + int peer_connection_id, + ICECANDIDATEREADYTOSEND_CALLBACK callback) { + if (!g_peer_connection_map.count(peer_connection_id)) + return false; + + g_peer_connection_map[peer_connection_id]->RegisterOnIceCandidateReadytoSend( + callback); + return true; +} diff --git a/third_party/libwebrtc/examples/unityplugin/unity_plugin_apis.h b/third_party/libwebrtc/examples/unityplugin/unity_plugin_apis.h new file mode 100644 index 0000000000..9790dc57b9 --- /dev/null +++ b/third_party/libwebrtc/examples/unityplugin/unity_plugin_apis.h @@ -0,0 +1,108 @@ +/* + * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// This file provides an example of unity native plugin APIs. + +#ifndef EXAMPLES_UNITYPLUGIN_UNITY_PLUGIN_APIS_H_ +#define EXAMPLES_UNITYPLUGIN_UNITY_PLUGIN_APIS_H_ + +#include <stdint.h> + +// Definitions of callback functions. +typedef void (*I420FRAMEREADY_CALLBACK)(const uint8_t* data_y, + const uint8_t* data_u, + const uint8_t* data_v, + const uint8_t* data_a, + int stride_y, + int stride_u, + int stride_v, + int stride_a, + uint32_t width, + uint32_t height); +typedef void (*LOCALDATACHANNELREADY_CALLBACK)(); +typedef void (*DATAFROMEDATECHANNELREADY_CALLBACK)(const char* msg); +typedef void (*FAILURE_CALLBACK)(const char* msg); +typedef void (*LOCALSDPREADYTOSEND_CALLBACK)(const char* type, const char* sdp); +typedef void (*ICECANDIDATEREADYTOSEND_CALLBACK)(const char* candidate, + int sdp_mline_index, + const char* sdp_mid); +typedef void (*AUDIOBUSREADY_CALLBACK)(const void* audio_data, + int bits_per_sample, + int sample_rate, + int number_of_channels, + int number_of_frames); + +#if defined(WEBRTC_WIN) +#define WEBRTC_PLUGIN_API __declspec(dllexport) +#elif defined(WEBRTC_ANDROID) +#define WEBRTC_PLUGIN_API __attribute__((visibility("default"))) +#endif +extern "C" { +// Create a peerconnection and return a unique peer connection id. +WEBRTC_PLUGIN_API int CreatePeerConnection(const char** turn_urls, + int no_of_urls, + const char* username, + const char* credential, + bool mandatory_receive_video); +// Close a peerconnection. +WEBRTC_PLUGIN_API bool ClosePeerConnection(int peer_connection_id); +// Add a audio stream. If audio_only is true, the stream only has an audio +// track and no video track. +WEBRTC_PLUGIN_API bool AddStream(int peer_connection_id, bool audio_only); +// Add a data channel to peer connection. +WEBRTC_PLUGIN_API bool AddDataChannel(int peer_connection_id); +// Create a peer connection offer. +WEBRTC_PLUGIN_API bool CreateOffer(int peer_connection_id); +// Create a peer connection answer. +WEBRTC_PLUGIN_API bool CreateAnswer(int peer_connection_id); +// Send data through data channel. +WEBRTC_PLUGIN_API bool SendDataViaDataChannel(int peer_connection_id, + const char* data); +// Set audio control. If is_mute=true, no audio will playout. If is_record=true, +// AUDIOBUSREADY_CALLBACK will be called every 10 ms. +WEBRTC_PLUGIN_API bool SetAudioControl(int peer_connection_id, + bool is_mute, + bool is_record); +// Set remote sdp. +WEBRTC_PLUGIN_API bool SetRemoteDescription(int peer_connection_id, + const char* type, + const char* sdp); +// Add ice candidate. +WEBRTC_PLUGIN_API bool AddIceCandidate(int peer_connection_id, + const char* candidate, + int sdp_mlineindex, + const char* sdp_mid); + +// Register callback functions. +WEBRTC_PLUGIN_API bool RegisterOnLocalI420FrameReady( + int peer_connection_id, + I420FRAMEREADY_CALLBACK callback); +WEBRTC_PLUGIN_API bool RegisterOnRemoteI420FrameReady( + int peer_connection_id, + I420FRAMEREADY_CALLBACK callback); +WEBRTC_PLUGIN_API bool RegisterOnLocalDataChannelReady( + int peer_connection_id, + LOCALDATACHANNELREADY_CALLBACK callback); +WEBRTC_PLUGIN_API bool RegisterOnDataFromDataChannelReady( + int peer_connection_id, + DATAFROMEDATECHANNELREADY_CALLBACK callback); +WEBRTC_PLUGIN_API bool RegisterOnFailure(int peer_connection_id, + FAILURE_CALLBACK callback); +WEBRTC_PLUGIN_API bool RegisterOnAudioBusReady(int peer_connection_id, + AUDIOBUSREADY_CALLBACK callback); +WEBRTC_PLUGIN_API bool RegisterOnLocalSdpReadytoSend( + int peer_connection_id, + LOCALSDPREADYTOSEND_CALLBACK callback); +WEBRTC_PLUGIN_API bool RegisterOnIceCandidateReadytoSend( + int peer_connection_id, + ICECANDIDATEREADYTOSEND_CALLBACK callback); +} + +#endif // EXAMPLES_UNITYPLUGIN_UNITY_PLUGIN_APIS_H_ diff --git a/third_party/libwebrtc/examples/unityplugin/video_observer.cc b/third_party/libwebrtc/examples/unityplugin/video_observer.cc new file mode 100644 index 0000000000..7e33b08e27 --- /dev/null +++ b/third_party/libwebrtc/examples/unityplugin/video_observer.cc @@ -0,0 +1,44 @@ +/* + * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "examples/unityplugin/video_observer.h" + +void VideoObserver::SetVideoCallback(I420FRAMEREADY_CALLBACK callback) { + std::lock_guard<std::mutex> lock(mutex); + OnI420FrameReady = callback; +} + +void VideoObserver::OnFrame(const webrtc::VideoFrame& frame) { + std::unique_lock<std::mutex> lock(mutex); + if (!OnI420FrameReady) + return; + + rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer( + frame.video_frame_buffer()); + + if (buffer->type() != webrtc::VideoFrameBuffer::Type::kI420A) { + rtc::scoped_refptr<webrtc::I420BufferInterface> i420_buffer = + buffer->ToI420(); + OnI420FrameReady(i420_buffer->DataY(), i420_buffer->DataU(), + i420_buffer->DataV(), nullptr, i420_buffer->StrideY(), + i420_buffer->StrideU(), i420_buffer->StrideV(), 0, + frame.width(), frame.height()); + + } else { + // The buffer has alpha channel. + const webrtc::I420ABufferInterface* i420a_buffer = buffer->GetI420A(); + + OnI420FrameReady(i420a_buffer->DataY(), i420a_buffer->DataU(), + i420a_buffer->DataV(), i420a_buffer->DataA(), + i420a_buffer->StrideY(), i420a_buffer->StrideU(), + i420a_buffer->StrideV(), i420a_buffer->StrideA(), + frame.width(), frame.height()); + } +} diff --git a/third_party/libwebrtc/examples/unityplugin/video_observer.h b/third_party/libwebrtc/examples/unityplugin/video_observer.h new file mode 100644 index 0000000000..01ccd2191a --- /dev/null +++ b/third_party/libwebrtc/examples/unityplugin/video_observer.h @@ -0,0 +1,35 @@ +/* + * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef EXAMPLES_UNITYPLUGIN_VIDEO_OBSERVER_H_ +#define EXAMPLES_UNITYPLUGIN_VIDEO_OBSERVER_H_ + +#include <mutex> + +#include "api/media_stream_interface.h" +#include "api/video/video_sink_interface.h" +#include "examples/unityplugin/unity_plugin_apis.h" + +class VideoObserver : public rtc::VideoSinkInterface<webrtc::VideoFrame> { + public: + VideoObserver() {} + ~VideoObserver() {} + void SetVideoCallback(I420FRAMEREADY_CALLBACK callback); + + protected: + // VideoSinkInterface implementation + void OnFrame(const webrtc::VideoFrame& frame) override; + + private: + I420FRAMEREADY_CALLBACK OnI420FrameReady = nullptr; + std::mutex mutex; +}; + +#endif // EXAMPLES_UNITYPLUGIN_VIDEO_OBSERVER_H_ |
