From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/moz-patch-stack/0035.patch | 778 +++++++++++++++++++++++ 1 file changed, 778 insertions(+) create mode 100644 third_party/libwebrtc/moz-patch-stack/0035.patch (limited to 'third_party/libwebrtc/moz-patch-stack/0035.patch') diff --git a/third_party/libwebrtc/moz-patch-stack/0035.patch b/third_party/libwebrtc/moz-patch-stack/0035.patch new file mode 100644 index 0000000000..c215ed8b8a --- /dev/null +++ b/third_party/libwebrtc/moz-patch-stack/0035.patch @@ -0,0 +1,778 @@ +From: stransky +Date: Thu, 5 Nov 2020 07:47:00 +0000 +Subject: Bug 1654112 - Tweak upstream gn files for Firefox build. r=ng + +Differential Revision: https://phabricator.services.mozilla.com/D130075 +Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/127ace4d8887f11abb201d300a849772a2b519f8 +--- + .gn | 2 ++ + BUILD.gn | 26 +++++++++++++- + api/BUILD.gn | 8 ++++- + api/task_queue/BUILD.gn | 2 ++ + api/transport/BUILD.gn | 2 ++ + common_audio/BUILD.gn | 4 --- + common_audio/fir_filter_avx2.cc | 2 ++ + common_audio/intrin.h | 8 +++++ + media/BUILD.gn | 30 ++++++++++++++++ + modules/audio_coding/BUILD.gn | 2 +- + modules/audio_device/BUILD.gn | 8 ++--- + modules/audio_processing/aec3/BUILD.gn | 13 +++---- + .../aec3/adaptive_fir_filter_avx2.cc | 3 +- + .../audio_processing/agc2/rnn_vad/BUILD.gn | 2 +- + modules/desktop_capture/BUILD.gn | 29 +--------------- + modules/portal/BUILD.gn | 24 +++++++++++++ + modules/video_capture/BUILD.gn | 11 +----- + modules/video_coding/BUILD.gn | 6 ++++ + rtc_base/BUILD.gn | 4 ++- + rtc_base/system/BUILD.gn | 2 +- + test/BUILD.gn | 14 ++++++++ + webrtc.gni | 34 ++++++++++++------- + 22 files changed, 162 insertions(+), 74 deletions(-) + create mode 100644 common_audio/intrin.h + +diff --git a/.gn b/.gn +index c9824916ad..d35ea79ced 100644 +--- a/.gn ++++ b/.gn +@@ -69,6 +69,8 @@ default_args = { + # Prevent jsoncpp to pass -Wno-deprecated-declarations to users + jsoncpp_no_deprecated_declarations = false + ++ use_custom_libcxx = false ++ + # Fixes the abi-revision issue. + # TODO(https://bugs.webrtc.org/14437): Remove this section if general + # Chromium fix resolves the problem. +diff --git a/BUILD.gn b/BUILD.gn +index 5817d22227..8b2648a306 100644 +--- a/BUILD.gn ++++ b/BUILD.gn +@@ -33,7 +33,7 @@ if (is_android) { + import("//build/config/android/rules.gni") + } + +-if (!build_with_chromium) { ++if (!build_with_chromium && !build_with_mozilla) { + # This target should (transitively) cause everything to be built; if you run + # 'ninja default' and then 'ninja all', the second build should do no work. + group("default") { +@@ -152,6 +152,10 @@ config("common_inherited_config") { + defines += [ "WEBRTC_ENABLE_OBJC_SYMBOL_EXPORT" ] + } + ++ if (build_with_mozilla) { ++ defines += [ "WEBRTC_MOZILLA_BUILD" ] ++ } ++ + if (!rtc_builtin_ssl_root_certificates) { + defines += [ "WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS" ] + } +@@ -460,9 +464,11 @@ config("common_config") { + } + } + ++if (is_mac) { + config("common_objc") { + frameworks = [ "Foundation.framework" ] + } ++} + + if (!build_with_chromium) { + # Target to build all the WebRTC production code. +@@ -502,6 +508,23 @@ if (!build_with_chromium) { + "sdk", + "video", + ] ++ if (build_with_mozilla) { ++ deps -= [ ++ "api:create_peerconnection_factory", ++ "api:rtc_error", ++ "api:transport_api", ++ "api/crypto", ++ "api/rtc_event_log:rtc_event_log_factory", ++ "api/task_queue", ++ "api/task_queue:default_task_queue_factory", ++ "api/test/metrics", ++ "logging:rtc_event_log_api", ++ "p2p:rtc_p2p", ++ "pc:libjingle_peerconnection", ++ "pc:rtc_pc", ++ "sdk", ++ ] ++ } + + if (rtc_include_builtin_audio_codecs) { + deps += [ +@@ -521,6 +544,7 @@ if (!build_with_chromium) { + deps += [ + "api/video:video_frame", + "api/video:video_rtp_headers", ++ "test:rtp_test_utils", + ] + } else { + deps += [ +diff --git a/api/BUILD.gn b/api/BUILD.gn +index 33a6b0aaa6..ab5d6c91ce 100644 +--- a/api/BUILD.gn ++++ b/api/BUILD.gn +@@ -35,7 +35,7 @@ rtc_source_set("callfactory_api") { + ] + } + +-if (!build_with_chromium) { ++if (!build_with_chromium && !build_with_mozilla) { + rtc_library("create_peerconnection_factory") { + visibility = [ "*" ] + allow_poison = [ "default_task_queue" ] +@@ -227,6 +227,7 @@ rtc_library("rtp_sender_interface") { + } + + rtc_library("libjingle_peerconnection_api") { ++if (!build_with_mozilla) { + visibility = [ "*" ] + cflags = [] + sources = [ +@@ -343,6 +344,7 @@ rtc_library("libjingle_peerconnection_api") { + "//third_party/abseil-cpp/absl/types:optional", + ] + } ++} + + rtc_source_set("frame_transformer_interface") { + visibility = [ "*" ] +@@ -550,6 +552,7 @@ rtc_source_set("peer_network_dependencies") { + } + + rtc_source_set("peer_connection_quality_test_fixture_api") { ++if (!build_with_mozilla) { + visibility = [ "*" ] + testonly = true + sources = [ "test/peerconnection_quality_test_fixture.h" ] +@@ -600,6 +603,7 @@ rtc_source_set("peer_connection_quality_test_fixture_api") { + "//third_party/abseil-cpp/absl/types:optional", + ] + } ++} + + rtc_source_set("frame_generator_api") { + visibility = [ "*" ] +@@ -885,6 +889,7 @@ rtc_source_set("refcountedbase") { + ] + } + ++if (!build_with_mozilla) { + rtc_library("ice_transport_factory") { + visibility = [ "*" ] + sources = [ +@@ -903,6 +908,7 @@ rtc_library("ice_transport_factory") { + "rtc_event_log:rtc_event_log", + ] + } ++} + + rtc_library("neteq_simulator_api") { + visibility = [ "*" ] +diff --git a/api/task_queue/BUILD.gn b/api/task_queue/BUILD.gn +index 69393b80ff..c9b4a5d0ec 100644 +--- a/api/task_queue/BUILD.gn ++++ b/api/task_queue/BUILD.gn +@@ -30,6 +30,7 @@ rtc_library("task_queue") { + ] + } + ++if (rtc_include_tests) { + rtc_library("task_queue_test") { + visibility = [ "*" ] + testonly = true +@@ -78,6 +79,7 @@ rtc_library("task_queue_test") { + ] + } + } ++} + + rtc_library("default_task_queue_factory") { + visibility = [ "*" ] +diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn +index 08f3d6d1d0..e0b31122b2 100644 +--- a/api/transport/BUILD.gn ++++ b/api/transport/BUILD.gn +@@ -90,6 +90,7 @@ rtc_source_set("sctp_transport_factory_interface") { + } + + rtc_source_set("stun_types") { ++if (!build_with_mozilla) { + visibility = [ "*" ] + sources = [ + "stun.cc", +@@ -110,6 +111,7 @@ rtc_source_set("stun_types") { + ] + absl_deps = [ "//third_party/abseil-cpp/absl/strings" ] + } ++} + + if (rtc_include_tests) { + rtc_source_set("test_feedback_generator_interface") { +diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn +index 2ae6d32710..a45214f754 100644 +--- a/common_audio/BUILD.gn ++++ b/common_audio/BUILD.gn +@@ -267,14 +267,10 @@ if (current_cpu == "x86" || current_cpu == "x64") { + "resampler/sinc_resampler_avx2.cc", + ] + +- if (is_win) { +- cflags = [ "/arch:AVX2" ] +- } else { + cflags = [ + "-mavx2", + "-mfma", + ] +- } + + deps = [ + ":fir_filter", +diff --git a/common_audio/fir_filter_avx2.cc b/common_audio/fir_filter_avx2.cc +index 9cb0f770ca..0031392f8a 100644 +--- a/common_audio/fir_filter_avx2.cc ++++ b/common_audio/fir_filter_avx2.cc +@@ -15,6 +15,8 @@ + #include + #include + ++#include "common_audio/intrin.h" ++ + #include "rtc_base/checks.h" + #include "rtc_base/memory/aligned_malloc.h" + +diff --git a/common_audio/intrin.h b/common_audio/intrin.h +new file mode 100644 +index 0000000000..f6ff7f218f +--- /dev/null ++++ b/common_audio/intrin.h +@@ -0,0 +1,8 @@ ++#if defined (__SSE__) ++ #include ++ #if defined (__clang__) ++ #include ++ #include ++ #include ++ #endif ++#endif +diff --git a/media/BUILD.gn b/media/BUILD.gn +index 2e5c0bb872..4ddc8349a8 100644 +--- a/media/BUILD.gn ++++ b/media/BUILD.gn +@@ -147,6 +147,27 @@ rtc_library("rtc_media_base") { + "base/video_source_base.cc", + "base/video_source_base.h", + ] ++ if (build_with_mozilla) { ++ sources -= [ ++ "base/adapted_video_track_source.cc", ++ "base/adapted_video_track_source.h", ++ "base/audio_source.h", ++ "base/delayable.h", ++ "base/media_channel.h", ++ "base/media_channel_impl.cc", ++ "base/media_channel_impl.h", ++ "base/media_engine.cc", ++ "base/media_engine.h", ++ "base/rid_description.cc", ++ "base/rid_description.h", ++ "base/rtp_utils.cc", ++ "base/rtp_utils.h", ++ "base/stream_params.cc", ++ "base/stream_params.h", ++ "base/turn_utils.cc", ++ "base/turn_utils.h", ++ ] ++ } + } + + rtc_library("media_channel_impl") { +@@ -420,6 +441,9 @@ rtc_library("rtc_internal_video_codecs") { + "../system_wrappers:field_trial", + "../test:fake_video_codecs", + ] ++ if (build_with_mozilla) { ++ deps -= [ "../test:fake_video_codecs" ] ++ } + + if (enable_libaom) { + defines += [ "RTC_USE_LIBAOM_AV1_ENCODER" ] +@@ -449,6 +473,12 @@ rtc_library("rtc_internal_video_codecs") { + # targets depend on :rtc_encoder_simulcast_proxy directly. + "engine/encoder_simulcast_proxy.h", + ] ++ if (build_with_mozilla) { ++ sources -= [ ++ "engine/fake_video_codec_factory.cc", ++ "engine/fake_video_codec_factory.h", ++ ] ++ } + } + + rtc_library("rtc_audio_video") { +diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn +index dddc3edd83..eac0650a26 100644 +--- a/modules/audio_coding/BUILD.gn ++++ b/modules/audio_coding/BUILD.gn +@@ -553,7 +553,7 @@ rtc_library("webrtc_opus_wrapper") { + deps += [ rtc_opus_dir ] + public_configs = [ "//third_party/opus:opus_config" ] + } else if (build_with_mozilla) { +- include_dirs = [ getenv("DIST") + "/include/opus" ] ++ public_configs = [ "//third_party/opus:opus_config" ] + } + } + +diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn +index 4a6a0ab41c..e35a442025 100644 +--- a/modules/audio_device/BUILD.gn ++++ b/modules/audio_device/BUILD.gn +@@ -233,9 +233,9 @@ rtc_library("audio_device_impl") { + ] + + if (build_with_mozilla) { +- sources += [ +- "opensl/single_rw_fifo.cc", +- "opensl/single_rw_fifo.h", ++ sources -= [ ++ "include/test_audio_device.cc", ++ "include/test_audio_device.h", + ] + } + +@@ -477,7 +477,7 @@ if (rtc_include_tests && !build_with_chromium) { + } + } + +-if (!build_with_chromium && is_android) { ++if ((!build_with_chromium && !build_with_mozilla) && is_android) { + rtc_android_library("audio_device_java") { + sources = [ + "android/java/src/org/webrtc/voiceengine/BuildInfo.java", +diff --git a/modules/audio_processing/aec3/BUILD.gn b/modules/audio_processing/aec3/BUILD.gn +index f5eb5d5951..3e11a245a1 100644 +--- a/modules/audio_processing/aec3/BUILD.gn ++++ b/modules/audio_processing/aec3/BUILD.gn +@@ -264,14 +264,11 @@ if (current_cpu == "x86" || current_cpu == "x64") { + "vector_math_avx2.cc", + ] + +- if (is_win) { +- cflags = [ "/arch:AVX2" ] +- } else { +- cflags = [ +- "-mavx2", +- "-mfma", +- ] +- } ++ cflags = [ ++ "-mavx", ++ "-mavx2", ++ "-mfma", ++ ] + + deps = [ + ":adaptive_fir_filter", +diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc b/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc +index 6c8c948026..44d4514275 100644 +--- a/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc ++++ b/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc +@@ -10,8 +10,7 @@ + + #include "modules/audio_processing/aec3/adaptive_fir_filter.h" + +-#include +- ++#include "common_audio/intrin.h" + #include "rtc_base/checks.h" + + namespace webrtc { +diff --git a/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/modules/audio_processing/agc2/rnn_vad/BUILD.gn +index 9093a68cf3..3003a585bd 100644 +--- a/modules/audio_processing/agc2/rnn_vad/BUILD.gn ++++ b/modules/audio_processing/agc2/rnn_vad/BUILD.gn +@@ -122,7 +122,7 @@ rtc_source_set("vector_math") { + if (current_cpu == "x86" || current_cpu == "x64") { + rtc_library("vector_math_avx2") { + sources = [ "vector_math_avx2.cc" ] +- if (is_win) { ++ if (is_win && !build_with_mozilla) { + cflags = [ "/arch:AVX2" ] + } else { + cflags = [ +diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn +index c9993dd47b..4aa9186527 100644 +--- a/modules/desktop_capture/BUILD.gn ++++ b/modules/desktop_capture/BUILD.gn +@@ -342,37 +342,12 @@ rtc_library("desktop_capture") { + ] + deps += [ ":desktop_capture_objc" ] + } +- +- if (build_with_mozilla) { +- sources += [ +- "desktop_device_info.cc", +- "desktop_device_info.h", +- ] +- if (is_win) { +- sources += [ +- "app_capturer_win.cc", +- "win/desktop_device_info_win.cc", +- "win/win_shared.cc", +- ] +- } +- } + if (rtc_use_x11_extensions || rtc_use_pipewire) { + sources += [ + "mouse_cursor_monitor_linux.cc", + "screen_capturer_linux.cc", + "window_capturer_linux.cc", + ] +- +- if (build_with_mozilla && (is_linux || is_chromeos)) { +- sources += [ +- "app_capturer_linux.cc", +- "linux/x11/app_capturer_x11.cc", +- "linux/x11/desktop_device_info_linux.cc", +- "linux/x11/desktop_device_info_linux.h", +- "linux/x11/shared_x_util.cc", +- "linux/x11/shared_x_util.h", +- ] +- } + } + + if (rtc_use_x11_extensions) { +@@ -536,9 +511,7 @@ rtc_library("desktop_capture") { + deps += [ "../../rtc_base:sanitizer" ] + } + +- if (!build_with_mozilla) { +- deps += [ "//third_party/libyuv" ] +- } ++ deps += [ "//third_party/libyuv" ] + + if (use_desktop_capture_differ_sse2) { + deps += [ ":desktop_capture_differ_sse2" ] +diff --git a/modules/portal/BUILD.gn b/modules/portal/BUILD.gn +index d0756f269b..d7768b2323 100644 +--- a/modules/portal/BUILD.gn ++++ b/modules/portal/BUILD.gn +@@ -11,6 +11,7 @@ import("//tools/generate_stubs/rules.gni") + import("../../webrtc.gni") + + if ((is_linux || is_chromeos) && rtc_use_pipewire) { ++if (!build_with_mozilla) { + pkg_config("gio") { + packages = [ + "gio-2.0", +@@ -88,6 +89,12 @@ if ((is_linux || is_chromeos) && rtc_use_pipewire) { + defines += [ "WEBRTC_USE_GIO" ] + } + } ++} else { ++ config("pipewire_all") { ++ } ++ config("pipewire_config") { ++ } ++} + + rtc_library("portal") { + sources = [ +@@ -120,5 +127,22 @@ if ((is_linux || is_chromeos) && rtc_use_pipewire) { + + deps += [ ":pipewire_stubs" ] + } ++ ++ if (build_with_mozilla) { ++ configs -= [ ++ ":gio", ++ ":pipewire", ++ ":pipewire_config", ++ ] ++ deps -= [ ":pipewire_stubs" ] ++ defines -= [ "WEBRTC_DLOPEN_PIPEWIRE" ] ++ public_deps = [ ++ "//third_party/pipewire", ++ "//third_party/drm", ++ "//third_party/gbm", ++ "//third_party/libepoxy" ++ ] ++ } + } + } ++ +diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn +index cfa5184d71..95548906c4 100644 +--- a/modules/video_capture/BUILD.gn ++++ b/modules/video_capture/BUILD.gn +@@ -93,21 +93,12 @@ if (!build_with_chromium) { + "strmiids.lib", + "user32.lib", + ] +- +- if (build_with_mozilla) { +- sources += [ +- "windows/BaseFilter.cpp", +- "windows/BaseInputPin.cpp", +- "windows/BasePin.cpp", +- "windows/MediaType.cpp", +- ] +- } + } + if (is_fuchsia) { + sources = [ "video_capture_factory_null.cc" ] + } + +- if (build_with_mozilla && is_android) { ++ if (!build_with_mozilla && is_android) { + include_dirs = [ + "/config/external/nspr", + "/nsprpub/lib/ds", +diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn +index fe63804b19..e1b5e4ba84 100644 +--- a/modules/video_coding/BUILD.gn ++++ b/modules/video_coding/BUILD.gn +@@ -237,6 +237,12 @@ rtc_library("video_coding") { + "video_receiver2.cc", + "video_receiver2.h", + ] ++ if (build_with_mozilla) { ++ sources += [ ++ "event_wrapper.cc", ++ "event_wrapper.h", ++ ] ++ } + + deps = [ + ":codec_globals_headers", +diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn +index 0d8bd4f759..7e162cecbb 100644 +--- a/rtc_base/BUILD.gn ++++ b/rtc_base/BUILD.gn +@@ -820,7 +820,9 @@ rtc_library("rtc_json") { + "strings/json.h", + ] + deps = [ ":stringutils" ] ++if (!build_with_mozilla) { + all_dependent_configs = [ "//third_party/jsoncpp:jsoncpp_config" ] ++} + if (rtc_build_json) { + deps += [ "//third_party/jsoncpp" ] + } else { +@@ -2073,7 +2075,7 @@ if (rtc_include_tests) { + } + } + +-if (is_android) { ++if (is_android && !build_with_mozilla) { + rtc_android_library("base_java") { + visibility = [ "*" ] + sources = [ +diff --git a/rtc_base/system/BUILD.gn b/rtc_base/system/BUILD.gn +index 77f5139a2f..486b37590c 100644 +--- a/rtc_base/system/BUILD.gn ++++ b/rtc_base/system/BUILD.gn +@@ -101,7 +101,7 @@ if (is_mac || is_ios) { + rtc_source_set("warn_current_thread_is_deadlocked") { + sources = [ "warn_current_thread_is_deadlocked.h" ] + deps = [] +- if (is_android && !build_with_chromium) { ++ if (is_android && (!build_with_chromium && !build_with_mozilla)) { + sources += [ "warn_current_thread_is_deadlocked.cc" ] + deps += [ + "..:logging", +diff --git a/test/BUILD.gn b/test/BUILD.gn +index 0c71115f0d..04a718c411 100644 +--- a/test/BUILD.gn ++++ b/test/BUILD.gn +@@ -156,6 +156,7 @@ rtc_library("audio_test_common") { + absl_deps = [ "//third_party/abseil-cpp/absl/memory" ] + } + ++if (!build_with_mozilla) { + if (!build_with_chromium) { + if (is_mac || is_ios) { + rtc_library("video_test_mac") { +@@ -204,8 +205,12 @@ if (!build_with_chromium) { + } + } + } ++} + + rtc_library("rtp_test_utils") { ++ if (build_with_mozilla) { ++ sources = [] ++ } else { + testonly = true + sources = [ + "rtcp_packet_parser.cc", +@@ -215,6 +220,7 @@ rtc_library("rtp_test_utils") { + "rtp_file_writer.cc", + "rtp_file_writer.h", + ] ++ } + + deps = [ + "../api:array_view", +@@ -467,7 +473,9 @@ rtc_library("video_test_support") { + absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] + + if (!is_ios) { ++ if (!build_with_mozilla) { + deps += [ "//third_party:jpeg" ] ++ } + sources += [ "testsupport/jpeg_frame_writer.cc" ] + } else { + sources += [ "testsupport/jpeg_frame_writer_ios.cc" ] +@@ -1036,6 +1044,10 @@ rtc_library("test_common") { + if (!is_android && !build_with_chromium) { + deps += [ "../modules/video_capture:video_capture_internal_impl" ] + } ++ # This, or some form of it should be upstreamed. ++ if (!rtc_include_tests) { ++ deps -= [ "../rtc_base:task_queue_for_test" ] ++ } + } + + rtc_library("mock_transport") { +@@ -1203,6 +1215,7 @@ rtc_library("copy_to_file_audio_capturer_unittest") { + ] + } + ++if (!build_with_mozilla) { + if (!build_with_chromium && is_android) { + rtc_android_library("native_test_java") { + testonly = true +@@ -1216,6 +1229,7 @@ if (!build_with_chromium && is_android) { + ] + } + } ++} + + rtc_library("call_config_utils") { + # TODO(bugs.webrtc.org/10814): Remove rtc_json_suppressions as soon as it +diff --git a/webrtc.gni b/webrtc.gni +index 8dfcc9d244..6ae1b2329c 100644 +--- a/webrtc.gni ++++ b/webrtc.gni +@@ -35,6 +35,11 @@ if (is_mac) { + import("//build/config/mac/rules.gni") + } + ++if (is_android) { ++ import("//build/config/android/config.gni") ++ import("//build/config/android/rules.gni") ++} ++ + if (is_fuchsia) { + import("//build/config/fuchsia/config.gni") + } +@@ -42,6 +47,11 @@ if (is_fuchsia) { + # This declare_args is separated from the next one because args declared + # in this one, can be read from the next one (args defined in the same + # declare_args cannot be referenced in that scope). ++declare_args() { ++ # Enable to use the Mozilla internal settings. ++ build_with_mozilla = true ++} ++ + declare_args() { + # Setting this to true will make RTC_EXPORT (see rtc_base/system/rtc_export.h) + # expand to code that will manage symbols visibility. +@@ -84,7 +94,7 @@ declare_args() { + # will tell the pre-processor to remove the default definition of the + # SystemTimeNanos() which is defined in rtc_base/system_time.cc. In + # that case a new implementation needs to be provided. +- rtc_exclude_system_time = build_with_chromium ++ rtc_exclude_system_time = build_with_chromium || build_with_mozilla + + # Setting this to false will require the API user to pass in their own + # SSLCertificateVerifier to verify the certificates presented from a +@@ -110,7 +120,7 @@ declare_args() { + + # Used to specify an external OpenSSL include path when not compiling the + # library that comes with WebRTC (i.e. rtc_build_ssl == 0). +- rtc_ssl_root = "" ++ rtc_ssl_root = "unused" + + # Enable when an external authentication mechanism is used for performing + # packet authentication for RTP packets instead of libsrtp. +@@ -127,13 +137,13 @@ declare_args() { + rtc_enable_bwe_test_logging = false + + # Set this to false to skip building examples. +- rtc_build_examples = true ++ rtc_build_examples = false + + # Set this to false to skip building tools. +- rtc_build_tools = true ++ rtc_build_tools = false + + # Set this to false to skip building code that requires X11. +- rtc_use_x11 = ozone_platform_x11 ++ rtc_use_x11 = use_x11 + + # Set this to use PipeWire on the Wayland display server. + # By default it's only enabled on desktop Linux (excludes ChromeOS) and +@@ -144,9 +154,6 @@ declare_args() { + # Set this to link PipeWire and required libraries directly instead of using the dlopen. + rtc_link_pipewire = false + +- # Enable to use the Mozilla internal settings. +- build_with_mozilla = false +- + # Experimental: enable use of Android AAudio which requires Android SDK 26 or above + # and NDK r16 or above. + rtc_enable_android_aaudio = false +@@ -227,7 +234,7 @@ declare_args() { + # When set to true, a capturer implementation that uses the + # Windows.Graphics.Capture APIs will be available for use. This introduces a + # dependency on the Win 10 SDK v10.0.17763.0. +- rtc_enable_win_wgc = is_win ++ rtc_enable_win_wgc = false + + # Includes the dav1d decoder in the internal decoder factory when set to true. + rtc_include_dav1d_in_internal_decoder_factory = true +@@ -256,7 +263,7 @@ declare_args() { + rtc_build_json = !build_with_mozilla + rtc_build_libsrtp = !build_with_mozilla + rtc_build_libvpx = !build_with_mozilla +- rtc_libvpx_build_vp9 = !build_with_mozilla ++ rtc_libvpx_build_vp9 = true + rtc_build_opus = !build_with_mozilla + rtc_build_ssl = !build_with_mozilla + +@@ -275,7 +282,7 @@ declare_args() { + + # Chromium uses its own IO handling, so the internal ADM is only built for + # standalone WebRTC. +- rtc_include_internal_audio_device = !build_with_chromium ++ rtc_include_internal_audio_device = !build_with_chromium && !build_with_mozilla + + # Set this to true to enable the avx2 support in webrtc. + # TODO: Make sure that AVX2 works also for non-clang compilers. +@@ -319,6 +326,9 @@ declare_args() { + rtc_enable_grpc = rtc_enable_protobuf && (is_linux || is_mac) + } + ++# Enable liboam only on non-mozilla builds. ++enable_libaom = !build_with_mozilla ++ + # Make it possible to provide custom locations for some libraries (move these + # up into declare_args should we need to actually use them for the GN build). + rtc_libvpx_dir = "//third_party/libvpx" +@@ -1114,7 +1124,7 @@ if (is_mac || is_ios) { + } + } + +-if (is_android) { ++if (is_android && !build_with_mozilla) { + template("rtc_android_library") { + android_library(target_name) { + forward_variables_from(invoker, +-- +2.34.1 + -- cgit v1.2.3