chore(webrtc-sys): bump to m125_release (#505)

.nanpa/bump-webrtc.kdl

@@ -0,0 +1,2 @@
+patch package="webrtc-sys/build" type="added" "bump libwebrtc to m125"
+patch package="webrtc-sys" type="added" "bump libwebrtc to m125"

webrtc-sys/include/livekit/audio_device.h

@@ -19,9 +19,9 @@
 #include <atomic>
 
 #include "api/task_queue/task_queue_factory.h"
+#include "api/task_queue/task_queue_base.h"
 #include "modules/audio_device/include/audio_device.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/task_queue.h"
 #include "rtc_base/task_utils/repeating_task.h"
 
 namespace livekit {
@@ -119,7 +119,7 @@ class AudioDevice : public webrtc::AudioDeviceModule {
  private:
   mutable webrtc::Mutex mutex_;
   std::vector<int16_t> data_;
-  std::unique_ptr<rtc::TaskQueue> audio_queue_;
+  std::unique_ptr<webrtc::TaskQueueBase, webrtc::TaskQueueDeleter> audio_queue_;
   webrtc::RepeatingTaskHandle audio_task_;
   webrtc::AudioTransport* audio_transport_;
   webrtc::TaskQueueFactory* task_queue_factory_;

webrtc-sys/include/livekit/audio_track.h

@@ -27,7 +27,7 @@
 #include "livekit/webrtc.h"
 #include "pc/local_audio_source.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/task_queue.h"
+#include "api/task_queue/task_queue_base.h"
 #include "rtc_base/task_utils/repeating_task.h"
 #include "rtc_base/thread_annotations.h"
 #include "rust/cxx.h"
@@ -127,7 +127,7 @@ class AudioTrackSource {
 
    private:
     mutable webrtc::Mutex mutex_;
-    std::unique_ptr<rtc::TaskQueue> audio_queue_;
+    std::unique_ptr<webrtc::TaskQueueBase, webrtc::TaskQueueDeleter> audio_queue_;
     webrtc::RepeatingTaskHandle audio_task_;
 
     std::vector<webrtc::AudioTrackSinkInterface*> sinks_ RTC_GUARDED_BY(mutex_);

webrtc-sys/include/livekit/video_decoder_factory.h

@@ -18,6 +18,7 @@
 
 #include "api/video_codecs/video_decoder.h"
 #include "api/video_codecs/video_decoder_factory.h"
+#include "absl/strings/match.h"
 
 namespace livekit {
 class VideoDecoderFactory : public webrtc::VideoDecoderFactory {
@@ -29,8 +30,8 @@ class VideoDecoderFactory : public webrtc::VideoDecoderFactory {
   CodecSupport QueryCodecSupport(const webrtc::SdpVideoFormat& format,
                                  bool reference_scaling) const override;
 
-  std::unique_ptr<webrtc::VideoDecoder> CreateVideoDecoder(
-      const webrtc::SdpVideoFormat& format) override;
+  std::unique_ptr<webrtc::VideoDecoder> Create(
+      const webrtc::Environment& env, const webrtc::SdpVideoFormat& format) override;
 
  private:
   std::vector<std::unique_ptr<webrtc::VideoDecoderFactory>> factories_;

webrtc-sys/include/livekit/video_encoder_factory.h

@@ -31,8 +31,8 @@ class VideoEncoderFactory : public webrtc::VideoEncoderFactory {
         const webrtc::SdpVideoFormat& format,
         absl::optional<std::string> scalability_mode) const override;
 
-    std::unique_ptr<webrtc::VideoEncoder> CreateVideoEncoder(
-        const webrtc::SdpVideoFormat& format) override;
+    std::unique_ptr<webrtc::VideoEncoder> Create(
+        const webrtc::Environment& env, const webrtc::SdpVideoFormat& format) override;
 
    private:
     std::vector<std::unique_ptr<webrtc::VideoEncoderFactory>> factories_;
@@ -47,8 +47,8 @@ class VideoEncoderFactory : public webrtc::VideoEncoderFactory {
       const webrtc::SdpVideoFormat& format,
       absl::optional<std::string> scalability_mode) const override;
 
-  std::unique_ptr<webrtc::VideoEncoder> CreateVideoEncoder(
-      const webrtc::SdpVideoFormat& format) override;
+  std::unique_ptr<webrtc::VideoEncoder> Create(
+      const webrtc::Environment& env, const webrtc::SdpVideoFormat& format) override;
 
  private:
   std::unique_ptr<InternalFactory> internal_factory_;

webrtc-sys/libwebrtc/.gclient

@@ -1,7 +1,7 @@
 solutions = [
   {
     "name": 'src',
-    "url": 'https://github.com/webrtc-sdk/webrtc.git@m114_release',
+    "url": 'https://github.com/webrtc-sdk/webrtc.git@m125_release',
     "custom_deps": {},
     "deps_file": "DEPS",
     "managed": False,

webrtc-sys/libwebrtc/build_android.sh

@@ -71,6 +71,11 @@ cd src
 git apply "$COMMAND_DIR/patches/ssl_verify_callback_with_native_handle.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
 git apply "$COMMAND_DIR/patches/add_deps.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
 git apply "$COMMAND_DIR/patches/android_use_libunwind.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
+
+cd third_party
+git apply "$COMMAND_DIR/patches/abseil_use_optional.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
+cd ..
+
 cd ..
 
 mkdir -p "$ARTIFACTS_DIR/lib"

webrtc-sys/libwebrtc/build_ios.sh

@@ -81,6 +81,11 @@ cd src
 # git apply "$COMMAND_DIR/patches/add_licenses.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
 git apply "$COMMAND_DIR/patches/ssl_verify_callback_with_native_handle.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
 git apply "$COMMAND_DIR/patches/add_deps.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
+
+cd third_party
+git apply "$COMMAND_DIR/patches/abseil_use_optional.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
+cd ..
+
 cd ..
 
 mkdir -p "$ARTIFACTS_DIR/lib"

webrtc-sys/libwebrtc/build_linux.sh

@@ -70,6 +70,11 @@ cd src
 git apply "$COMMAND_DIR/patches/add_licenses.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
 git apply "$COMMAND_DIR/patches/ssl_verify_callback_with_native_handle.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
 git apply "$COMMAND_DIR/patches/add_deps.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
+
+cd third_party
+git apply "$COMMAND_DIR/patches/abseil_use_optional.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
+cd ..
+
 cd ..
 
 mkdir -p "$ARTIFACTS_DIR/lib"

webrtc-sys/libwebrtc/build_macos.sh

@@ -70,6 +70,11 @@ cd src
 git apply "$COMMAND_DIR/patches/add_licenses.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
 git apply "$COMMAND_DIR/patches/ssl_verify_callback_with_native_handle.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
 git apply "$COMMAND_DIR/patches/add_deps.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
+
+cd third_party
+git apply "$COMMAND_DIR/patches/abseil_use_optional.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
+cd ..
+
 cd ..
 
 mkdir -p "$ARTIFACTS_DIR/lib"

webrtc-sys/libwebrtc/build_windows.cmd

@@ -54,6 +54,11 @@ call git apply "%COMMAND_DIR%/patches/add_licenses.patch" -v --ignore-space-chan
 call git apply "%COMMAND_DIR%/patches/add_deps.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
 call git apply "%COMMAND_DIR%/patches/windows_silence_warnings.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
 call git apply "%COMMAND_DIR%/patches/ssl_verify_callback_with_native_handle.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
+
+cd third_party
+call git apply "%COMMAND_DIR%/patches/abseil_use_optional.patch" -v --ignore-space-change --ignore-whitespace --whitespace=nowarn
+cd ..
+
 cd ..
 
 mkdir "%ARTIFACTS_DIR%\lib"

webrtc-sys/libwebrtc/patches/abseil_use_optional.patch

@@ -0,0 +1,13 @@
+diff --git a/abseil-cpp/absl/base/options.h b/abseil-cpp/absl/base/options.h
+index bd43b6ef0..ab5917e75 100644
+--- a/abseil-cpp/absl/base/options.h
++++ b/abseil-cpp/absl/base/options.h
+@@ -121,7 +121,7 @@
+ // absl::optional is a typedef of std::optional, use the feature macro
+ // ABSL_USES_STD_OPTIONAL.
+
+-#define ABSL_OPTION_USE_STD_OPTIONAL 2
++#define ABSL_OPTION_USE_STD_OPTIONAL 0
+
+
+ // ABSL_OPTION_USE_STD_STRING_VIEW

webrtc-sys/src/audio_device.cpp

@@ -48,11 +48,11 @@ int32_t AudioDevice::Init() {
     return 0;
 
   audio_queue_ =
-      std::make_unique<rtc::TaskQueue>(task_queue_factory_->CreateTaskQueue(
-          "AudioDevice", webrtc::TaskQueueFactory::Priority::NORMAL));
+      task_queue_factory_->CreateTaskQueue(
+          "AudioDevice", webrtc::TaskQueueFactory::Priority::NORMAL);
 
   audio_task_ =
-      webrtc::RepeatingTaskHandle::Start(audio_queue_->Get(), [this]() {
+      webrtc::RepeatingTaskHandle::Start(audio_queue_.get(), [this]() {
         webrtc::MutexLock lock(&mutex_);
 
         if (playing_) {

webrtc-sys/src/audio_track.cpp

@@ -153,11 +153,11 @@ AudioTrackSource::InternalSource::InternalSource(
   buffer_.reserve(queue_size_samples_ + notify_threshold_samples_);
 
   audio_queue_ =
-      std::make_unique<rtc::TaskQueue>(task_queue_factory->CreateTaskQueue(
-          "AudioSourceCapture", webrtc::TaskQueueFactory::Priority::NORMAL));
+      task_queue_factory->CreateTaskQueue(
+          "AudioSourceCapture", webrtc::TaskQueueFactory::Priority::NORMAL);
 
   audio_task_ = webrtc::RepeatingTaskHandle::Start(
-      audio_queue_->Get(),
+      audio_queue_.get(),
       [this, samples10ms]() {
         webrtc::MutexLock lock(&mutex_);
 

webrtc-sys/src/peer_connection_factory.cpp

@@ -23,6 +23,7 @@
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "api/peer_connection_interface.h"
 #include "api/rtc_error.h"
+#include "api/enable_media.h"
 #include "api/rtc_event_log/rtc_event_log_factory.h"
 #include "api/task_queue/default_task_queue_factory.h"
 #include "api/video_codecs/builtin_video_decoder_factory.h"
@@ -35,7 +36,6 @@
 #include "livekit/video_decoder_factory.h"
 #include "livekit/video_encoder_factory.h"
 #include "livekit/webrtc.h"
-#include "media/engine/webrtc_media_engine.h"
 #include "rtc_base/thread.h"
 #include "webrtc-sys/src/peer_connection.rs.h"
 #include "webrtc-sys/src/peer_connection_factory.rs.h"
@@ -55,32 +55,25 @@ PeerConnectionFactory::PeerConnectionFactory(
   dependencies.signaling_thread = rtc_runtime_->signaling_thread();
   dependencies.socket_factory = rtc_runtime_->network_thread()->socketserver();
   dependencies.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
-  dependencies.event_log_factory = std::make_unique<webrtc::RtcEventLogFactory>(
-      dependencies.task_queue_factory.get());
-  dependencies.call_factory = webrtc::CreateCallFactory();
+  dependencies.event_log_factory = std::make_unique<webrtc::RtcEventLogFactory>();
   dependencies.trials = std::make_unique<webrtc::FieldTrialBasedConfig>();
 
-  cricket::MediaEngineDependencies media_deps;
-  media_deps.task_queue_factory = dependencies.task_queue_factory.get();
-
   audio_device_ = rtc_runtime_->worker_thread()->BlockingCall([&] {
     return rtc::make_ref_counted<livekit::AudioDevice>(
-        media_deps.task_queue_factory);
+        dependencies.task_queue_factory.get());
   });
 
-  media_deps.adm = audio_device_;
+  dependencies.adm = audio_device_;
 
-  media_deps.video_encoder_factory =
+  dependencies.video_encoder_factory =
       std::move(std::make_unique<livekit::VideoEncoderFactory>());
-  media_deps.video_decoder_factory =
+  dependencies.video_decoder_factory =
       std::move(std::make_unique<livekit::VideoDecoderFactory>());
-  media_deps.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
-  media_deps.audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
-  media_deps.audio_processing = webrtc::AudioProcessingBuilder().Create();
-  media_deps.trials = dependencies.trials.get();
-
-  dependencies.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
+  dependencies.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
+  dependencies.audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
+  dependencies.audio_processing = webrtc::AudioProcessingBuilder().Create();
 
+  webrtc::EnableMedia(dependencies);
   peer_factory_ =
       webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
 

webrtc-sys/src/video_decoder_factory.cpp

@@ -16,6 +16,7 @@
 
 #include "livekit/video_decoder_factory.h"
 
+#include "api/environment/environment.h"
 #include "api/video_codecs/av1_profile.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "livekit/objc_video_factory.h"
@@ -69,7 +70,7 @@ std::vector<webrtc::SdpVideoFormat> VideoDecoderFactory::GetSupportedFormats()
   formats.push_back(webrtc::SdpVideoFormat(cricket::kAv1CodecName));
   formats.push_back(webrtc::SdpVideoFormat(
       cricket::kAv1CodecName,
-      {{webrtc::kAV1FmtpProfile,
+      {{cricket::kAv1FmtpProfile,
         AV1ProfileToString(webrtc::AV1Profile::kProfile1).data()}}));
 #endif
 
@@ -92,17 +93,17 @@ VideoDecoderFactory::CodecSupport VideoDecoderFactory::QueryCodecSupport(
   return codec_support;
 }
 
-std::unique_ptr<webrtc::VideoDecoder> VideoDecoderFactory::CreateVideoDecoder(
-    const webrtc::SdpVideoFormat& format) {
+std::unique_ptr<webrtc::VideoDecoder> VideoDecoderFactory::Create(
+    const webrtc::Environment& env, const webrtc::SdpVideoFormat& format) {
   for (const auto& factory : factories_) {
     for (const auto& supported_format : factory->GetSupportedFormats()) {
       if (supported_format.IsSameCodec(format))
-        return factory->CreateVideoDecoder(format);
+        return factory->Create(env, format);
     }
   }
 
   if (absl::EqualsIgnoreCase(format.name, cricket::kVp8CodecName))
-    return webrtc::VP8Decoder::Create();
+    return webrtc::CreateVp8Decoder(env);
   if (absl::EqualsIgnoreCase(format.name, cricket::kVp9CodecName))
     return webrtc::VP9Decoder::Create();
   if (absl::EqualsIgnoreCase(format.name, cricket::kH264CodecName))

webrtc-sys/src/video_encoder_factory.cpp

@@ -16,6 +16,7 @@
 
 #include "livekit/video_encoder_factory.h"
 
+#include "api/environment/environment_factory.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "api/video_codecs/video_encoder.h"
 #include "api/video_codecs/video_encoder_factory_template.h"
@@ -84,20 +85,20 @@ VideoEncoderFactory::InternalFactory::QueryCodecSupport(
 }
 
 std::unique_ptr<webrtc::VideoEncoder>
-VideoEncoderFactory::InternalFactory::CreateVideoEncoder(
-    const webrtc::SdpVideoFormat& format) {
+VideoEncoderFactory::InternalFactory::Create(
+    const webrtc::Environment& env, const webrtc::SdpVideoFormat& format) {
   for (const auto& factory : factories_) {
     for (const auto& supported_format : factory->GetSupportedFormats()) {
       if (supported_format.IsSameCodec(format))
-        return factory->CreateVideoEncoder(format);
+        return factory->Create(env, format);
     }
   }
 
   auto original_format =
       webrtc::FuzzyMatchSdpVideoFormat(Factory().GetSupportedFormats(), format);
 
   if (original_format) {
-    return Factory().CreateVideoEncoder(*original_format);
+    return Factory().Create(env, *original_format);
   }
 
   RTC_LOG(LS_ERROR) << "No VideoEncoder found for " << format.name;
@@ -119,12 +120,12 @@ VideoEncoderFactory::CodecSupport VideoEncoderFactory::QueryCodecSupport(
   return internal_factory_->QueryCodecSupport(format, scalability_mode);
 }
 
-std::unique_ptr<webrtc::VideoEncoder> VideoEncoderFactory::CreateVideoEncoder(
-    const webrtc::SdpVideoFormat& format) {
+std::unique_ptr<webrtc::VideoEncoder> VideoEncoderFactory::Create(
+    const webrtc::Environment& env, const webrtc::SdpVideoFormat& format) {
   std::unique_ptr<webrtc::VideoEncoder> encoder;
   if (format.IsCodecInList(internal_factory_->GetSupportedFormats())) {
     encoder = std::make_unique<webrtc::SimulcastEncoderAdapter>(
-        internal_factory_.get(), format);
+        env, internal_factory_.get(), nullptr, format);
   }
 
   return encoder;