Android improvements.

Start/Stop receiving stream method for VideoTrack (#25) Properly remove observer upon deconstruction (#26) feat: Expose setCodecPreferences/getCapabilities for android. (#61) Co-authored-by: davidliu <[email protected]>
webrtc-sdk · Jun 12, 2023 · c70cf2e · c70cf2e
1 parent 03dc44e
commit c70cf2e
Show file tree

Hide file tree

Showing 22 changed files with 491 additions and 2 deletions.
diff --git a/api/media_stream_interface.cc b/api/media_stream_interface.cc
@@ -18,6 +18,10 @@ const char* const MediaStreamTrackInterface::kVideoKind =
 const char* const MediaStreamTrackInterface::kAudioKind =
     cricket::kMediaTypeAudio;
 
+bool VideoTrackInterface::should_receive() const {
+  return true;
+}
+
 VideoTrackInterface::ContentHint VideoTrackInterface::content_hint() const {
   return ContentHint::kNone;
 }

diff --git a/api/media_stream_interface.h b/api/media_stream_interface.h
@@ -188,6 +188,8 @@ class RTC_EXPORT VideoTrackInterface
 
   virtual VideoTrackSourceInterface* GetSource() const = 0;
 
+  virtual void set_should_receive(bool should_receive) {}
+  virtual bool should_receive() const;
   virtual ContentHint content_hint() const;
   virtual void set_content_hint(ContentHint hint) {}
 

diff --git a/media/base/media_channel.h b/media/base/media_channel.h
@@ -997,6 +997,8 @@ class VideoMediaReceiveChannelInterface : public MediaReceiveChannelInterface {
                                              bool nack_enabled,
                                              webrtc::RtcpMode rtcp_mode,
                                              absl::optional<int> rtx_time) = 0;
+  virtual void StartReceive(uint32_t ssrc) {}
+  virtual void StopReceive(uint32_t ssrc) {}
 };
 
 }  // namespace cricket

diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
@@ -928,6 +928,24 @@ void WebRtcVideoChannel::RequestEncoderSwitch(
   }
 }
 
+void WebRtcVideoChannel::StartReceive(uint32_t ssrc) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  WebRtcVideoReceiveStream* stream = FindReceiveStream(ssrc);
+  if(!stream) {
+    return;
+  }
+  stream->StartStream();
+}
+
+void WebRtcVideoChannel::StopReceive(uint32_t ssrc) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  WebRtcVideoReceiveStream* stream = FindReceiveStream(ssrc);
+  if(!stream) {
+    return;
+  }
+  stream->StopStream();
+}
+
 bool WebRtcVideoChannel::ApplyChangedParams(
     const ChangedSendParameters& changed_params) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
@@ -3182,6 +3200,17 @@ void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetRecvParameters(
   }
 }
 
+void WebRtcVideoChannel::WebRtcVideoReceiveStream::StartStream(){
+  if (stream_) {
+    stream_->Start();
+  }
+}
+void WebRtcVideoChannel::WebRtcVideoReceiveStream::StopStream(){
+  if (stream_) {
+    stream_->Stop();
+  }
+}
+
 void WebRtcVideoChannel::WebRtcVideoReceiveStream::RecreateReceiveStream() {
   RTC_DCHECK(stream_);
   absl::optional<int> base_minimum_playout_delay_ms;

diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
@@ -270,6 +270,8 @@ class WebRtcVideoChannel : public VideoMediaChannel,
                                      webrtc::RtcpMode rtcp_mode,
                                      absl::optional<int> rtx_time) override;
 
+  void StartReceive(uint32_t ssrc) override;
+  void StopReceive(uint32_t ssrc) override;
  private:
   class WebRtcVideoReceiveStream;
 
@@ -537,6 +539,9 @@ class WebRtcVideoChannel : public VideoMediaChannel,
     void SetDepacketizerToDecoderFrameTransformer(
         rtc::scoped_refptr<webrtc::FrameTransformerInterface>
             frame_transformer);
+
+    void StartStream();
+    void StopStream();
 
     void SetLocalSsrc(uint32_t local_ssrc);
     void UpdateRtxSsrc(uint32_t ssrc);

diff --git a/pc/media_stream_track_proxy.h b/pc/media_stream_track_proxy.h
@@ -55,6 +55,8 @@ PROXY_SECONDARY_METHOD2(void,
 PROXY_SECONDARY_METHOD1(void, RemoveSink, rtc::VideoSinkInterface<VideoFrame>*)
 PROXY_SECONDARY_METHOD0(void, RequestRefreshFrame)
 BYPASS_PROXY_CONSTMETHOD0(VideoTrackSourceInterface*, GetSource)
+PROXY_CONSTMETHOD0(bool, should_receive)
+PROXY_METHOD1(void, set_should_receive, bool)
 
 PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
 PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)

diff --git a/pc/video_rtp_receiver.cc b/pc/video_rtp_receiver.cc
@@ -41,15 +41,20 @@ VideoRtpReceiver::VideoRtpReceiver(
           rtc::Thread::Current(),
           worker_thread,
           VideoTrack::Create(receiver_id, source_, worker_thread))),
-      attachment_id_(GenerateUniqueId()) {
+      cached_track_should_receive_(track_->should_receive()),
+      attachment_id_(GenerateUniqueId()),
+      worker_thread_safety_(PendingTaskSafetyFlag::CreateDetachedInactive()) {
   RTC_DCHECK(worker_thread_);
   SetStreams(streams);
+  track_->RegisterObserver(this);
   RTC_DCHECK_EQ(source_->state(), MediaSourceInterface::kInitializing);
 }
 
 VideoRtpReceiver::~VideoRtpReceiver() {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
   RTC_DCHECK(!media_channel_);
+
+  track_->UnregisterObserver(this);
 }
 
 std::vector<std::string> VideoRtpReceiver::stream_ids() const {
@@ -114,6 +119,39 @@ void VideoRtpReceiver::Stop() {
   track_->internal()->set_ended();
 }
 
+void VideoRtpReceiver::OnChanged() {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  if (cached_track_should_receive_ != track_->should_receive()) {
+    cached_track_should_receive_ = track_->should_receive();
+    worker_thread_->PostTask(
+        [this, receive = cached_track_should_receive_]() {
+          RTC_DCHECK_RUN_ON(worker_thread_);
+          if(receive) {
+            StartMediaChannel();
+          } else {
+            StopMediaChannel();
+          }
+        });
+  }
+}
+
+void VideoRtpReceiver::StartMediaChannel() {  
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  if (!media_channel_) {
+    return;
+  }
+  media_channel_->StartReceive(signaled_ssrc_.value_or(0));
+  OnGenerateKeyFrame();
+}
+
+void VideoRtpReceiver::StopMediaChannel() {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  if (!media_channel_) {
+    return;
+  }
+  media_channel_->StopReceive(signaled_ssrc_.value_or(0));
+}
+
 void VideoRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
   MediaSourceInterface::SourceState state = source_->state();
@@ -209,6 +247,7 @@ void VideoRtpReceiver::set_transport(
 void VideoRtpReceiver::SetStreams(
     const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+
   // Remove remote track from any streams that are going away.
   for (const auto& existing_stream : streams_) {
     bool removed = true;

diff --git a/pc/video_rtp_receiver.h b/pc/video_rtp_receiver.h
@@ -42,7 +42,8 @@
 
 namespace webrtc {
 
-class VideoRtpReceiver : public RtpReceiverInternal {
+class VideoRtpReceiver : public RtpReceiverInternal,
+                         public ObserverInterface {
  public:
   // An SSRC of 0 will create a receiver that will match the first SSRC it
   // sees. Must be called on signaling thread.
@@ -60,6 +61,9 @@ class VideoRtpReceiver : public RtpReceiverInternal {
 
   rtc::scoped_refptr<VideoTrackInterface> video_track() const { return track_; }
 
+  // ObserverInterface implementation
+  void OnChanged() override;
+
   // RtpReceiverInterface implementation
   rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
     return track_;
@@ -115,6 +119,8 @@ class VideoRtpReceiver : public RtpReceiverInternal {
                          cricket::MediaReceiveChannelInterface* media_channel);
 
  private:
+  void StartMediaChannel();
+  void StopMediaChannel();
   void RestartMediaChannel(absl::optional<uint32_t> ssrc)
       RTC_RUN_ON(&signaling_thread_checker_);
   void RestartMediaChannel_w(absl::optional<uint32_t> ssrc,
@@ -162,6 +168,8 @@ class VideoRtpReceiver : public RtpReceiverInternal {
       RTC_GUARDED_BY(&signaling_thread_checker_) = nullptr;
   bool received_first_packet_ RTC_GUARDED_BY(&signaling_thread_checker_) =
       false;
+
+  bool cached_track_should_receive_ RTC_GUARDED_BY(&signaling_thread_checker_);
   const int attachment_id_;
   rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_
       RTC_GUARDED_BY(worker_thread_);
@@ -177,6 +185,7 @@ class VideoRtpReceiver : public RtpReceiverInternal {
   // or switched.
   bool saved_generate_keyframe_ RTC_GUARDED_BY(worker_thread_) = false;
   bool saved_encoded_sink_enabled_ RTC_GUARDED_BY(worker_thread_) = false;
+  const rtc::scoped_refptr<PendingTaskSafetyFlag> worker_thread_safety_;
 };
 
 }  // namespace webrtc

diff --git a/pc/video_track.cc b/pc/video_track.cc
@@ -76,6 +76,19 @@ VideoTrackSourceInterface* VideoTrack::GetSourceInternal() const {
   return video_source_->internal();
 }
 
+void VideoTrack::set_should_receive(bool receive) {
+  RTC_DCHECK_RUN_ON(&signaling_thread_);
+  if (should_receive_ == receive)
+    return;
+  should_receive_ = receive;
+  Notifier<VideoTrackInterface>::FireOnChanged();
+}
+
+bool VideoTrack::should_receive() const {
+  RTC_DCHECK_RUN_ON(&signaling_thread_);
+  return should_receive_;
+}
+
 VideoTrackInterface::ContentHint VideoTrack::content_hint() const {
   RTC_DCHECK_RUN_ON(&signaling_thread_);
   return content_hint_;

diff --git a/pc/video_track.h b/pc/video_track.h
@@ -48,6 +48,9 @@ class VideoTrack : public MediaStreamTrack<VideoTrackInterface>,
   void RequestRefreshFrame() override;
   VideoTrackSourceInterface* GetSource() const override;
 
+  void set_should_receive(bool should_receive) override;
+  bool should_receive() const override;
+
   ContentHint content_hint() const override;
   void set_content_hint(ContentHint hint) override;
   bool set_enabled(bool enable) override;
@@ -81,6 +84,7 @@ class VideoTrack : public MediaStreamTrack<VideoTrackInterface>,
   // be queried without blocking on the worker thread by callers that don't
   // use an api proxy to call the `enabled()` method.
   bool enabled_w_ RTC_GUARDED_BY(worker_thread_) = true;
+  bool should_receive_ RTC_GUARDED_BY(signaling_thread_) = true;
 };
 
 }  // namespace webrtc

diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
@@ -287,6 +287,7 @@ if (is_android) {
       "api/org/webrtc/RTCStatsCollectorCallback.java",
       "api/org/webrtc/RTCStatsReport.java",
       "api/org/webrtc/RtcCertificatePem.java",
+      "api/org/webrtc/RtpCapabilities.java",
       "api/org/webrtc/RtpParameters.java",
       "api/org/webrtc/RtpReceiver.java",
       "api/org/webrtc/RtpSender.java",
@@ -736,6 +737,8 @@ if (current_os == "linux" || is_android) {
       "src/jni/pc/rtc_certificate.h",
       "src/jni/pc/rtc_stats_collector_callback_wrapper.cc",
       "src/jni/pc/rtc_stats_collector_callback_wrapper.h",
+      "src/jni/pc/rtp_capabilities.cc",
+      "src/jni/pc/rtp_capabilities.h",
       "src/jni/pc/rtp_parameters.cc",
       "src/jni/pc/rtp_parameters.h",
       "src/jni/pc/rtp_receiver.cc",
@@ -1409,6 +1412,7 @@ if (current_os == "linux" || is_android) {
       "api/org/webrtc/RTCStatsCollectorCallback.java",
       "api/org/webrtc/RTCStatsReport.java",
       "api/org/webrtc/RtcCertificatePem.java",
+      "api/org/webrtc/RtpCapabilities.java",
       "api/org/webrtc/RtpParameters.java",
       "api/org/webrtc/RtpReceiver.java",
       "api/org/webrtc/RtpSender.java",

diff --git a/sdk/android/api/org/webrtc/PeerConnectionFactory.java b/sdk/android/api/org/webrtc/PeerConnectionFactory.java
@@ -18,6 +18,7 @@
 import org.webrtc.PeerConnection;
 import org.webrtc.audio.AudioDeviceModule;
 import org.webrtc.audio.JavaAudioDeviceModule;
+import org.webrtc.RtpCapabilities;
 
 /**
  * Java wrapper for a C++ PeerConnectionFactoryInterface.  Main entry point to
@@ -471,6 +472,16 @@ public AudioTrack createAudioTrack(String id, AudioSource source) {
     return new AudioTrack(nativeCreateAudioTrack(nativeFactory, id, source.getNativeAudioSource()));
   }
 
+  public RtpCapabilities getRtpReceiverCapabilities(MediaStreamTrack.MediaType mediaType) {
+    checkPeerConnectionFactoryExists();
+    return nativeGetRtpReceiverCapabilities(nativeFactory, mediaType);
+  }
+
+  public RtpCapabilities getRtpSenderCapabilities(MediaStreamTrack.MediaType mediaType) {
+    checkPeerConnectionFactoryExists();
+    return nativeGetRtpSenderCapabilities(nativeFactory, mediaType);
+  }
+
   // Starts recording an AEC dump. Ownership of the file is transfered to the
   // native code. If an AEC dump is already in progress, it will be stopped and
   // a new one will start using the provided file.
@@ -615,4 +626,6 @@ private static native boolean nativeStartAecDump(
   private static native void nativeInjectLoggable(JNILogging jniLogging, int severity);
   private static native void nativeDeleteLoggable();
   private static native void nativePrintStackTrace(int tid);
+  private static native RtpCapabilities nativeGetRtpSenderCapabilities(long factory, MediaStreamTrack.MediaType mediaType);
+  private static native RtpCapabilities nativeGetRtpReceiverCapabilities(long factory, MediaStreamTrack.MediaType mediaType);
 }