Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

3 files changed, 399 insertions, 0 deletions

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..2713a563cd
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.cc
@@ -0,0 +1,293 @@
+/*
+ *  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 <memory>
+#include <utility>
+
+#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_source_, "video");
+  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