Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.

webrtc/api/rtpreceiver.cc

Index: webrtc/api/rtpreceiver.cc
diff --git a/webrtc/api/rtpreceiver.cc b/webrtc/api/rtpreceiver.cc
index 9df336e1f042ca79afb2814905af52c2840a7753..64ad09c426296970fc471ff7c2ed6cd6c9905ca9 100644
--- a/webrtc/api/rtpreceiver.cc
+++ b/webrtc/api/rtpreceiver.cc
@@ -21,14 +21,14 @@ namespace webrtc {
 AudioRtpReceiver::AudioRtpReceiver(MediaStreamInterface* stream,
                                    const std::string& track_id,
                                    uint32_t ssrc,
-                                   AudioProviderInterface* provider)
+                                   cricket::VoiceChannel* channel)
     : id_(track_id),
       ssrc_(ssrc),
-      provider_(provider),
+      channel_(channel),
       track_(AudioTrackProxy::Create(
           rtc::Thread::Current(),
           AudioTrack::Create(track_id,
-                             RemoteAudioSource::Create(ssrc, provider)))),
+                             RemoteAudioSource::Create(ssrc, channel)))),
       cached_track_enabled_(track_->enabled()) {
   RTC_DCHECK(track_->GetSource()->remote());
   track_->RegisterObserver(this);
@@ -51,46 +51,66 @@ void AudioRtpReceiver::OnChanged() {
 }
 
 void AudioRtpReceiver::OnSetVolume(double volume) {
+  RTC_DCHECK(volume >= 0 && volume <= 10);
+  cached_volume_ = volume;
+  if (!channel_) {
+    LOG(LS_ERROR) << "AudioRtpReceiver::OnSetVolume: No audio channel exists.";
+    return;
+  }
   // When the track is disabled, the volume of the source, which is the
   // corresponding WebRtc Voice Engine channel will be 0. So we do not allow
   // setting the volume to the source when the track is disabled.
-  if (provider_ && track_->enabled())
-    provider_->SetAudioPlayoutVolume(ssrc_, volume);
+  if (!stopped_ && track_->enabled()) {
+    RTC_DCHECK(channel_->SetOutputVolume(ssrc_, cached_volume_));
+  }
 }
 
 RtpParameters AudioRtpReceiver::GetParameters() const {
-  return provider_->GetAudioRtpReceiveParameters(ssrc_);
+  if (!channel_ || stopped_) {
+    return RtpParameters();
+  }
+  return channel_->GetRtpReceiveParameters(ssrc_);
 }
 
 bool AudioRtpReceiver::SetParameters(const RtpParameters& parameters) {
   TRACE_EVENT0("webrtc", "AudioRtpReceiver::SetParameters");
-  return provider_->SetAudioRtpReceiveParameters(ssrc_, parameters);
+  if (!channel_ || stopped_) {
+    return false;
+  }
+  return channel_->SetRtpReceiveParameters(ssrc_, parameters);
 }
 
 void AudioRtpReceiver::Stop() {
   // TODO(deadbeef): Need to do more here to fully stop receiving packets.
-  if (!provider_) {
+  if (stopped_) {
     return;
   }
-  provider_->SetAudioPlayout(ssrc_, false);
-  provider_ = nullptr;
+  if (channel_) {
+    // Allow that SetOutputVolume fail. This is the normal case when the
+    // underlying media channel has already been deleted.
+    channel_->SetOutputVolume(ssrc_, 0);
+  }
+  stopped_ = true;
 }
 
 void AudioRtpReceiver::Reconfigure() {
-  if (!provider_) {
+  RTC_DCHECK(!stopped_);
+  if (!channel_) {
+    LOG(LS_ERROR) << "AudioRtpReceiver::Reconfigure: No audio channel exists.";
     return;
   }
-  provider_->SetAudioPlayout(ssrc_, track_->enabled());
+  RTC_DCHECK(
+      channel_->SetOutputVolume(ssrc_, track_->enabled() ? cached_volume_ : 0));

[Taylor Brandstetter, 2016/06/07 22:44:43]

Previously, SetAudioPlayout would always set the volume to 1. That means if you
set the track's volume to 0.5, then disable and enable it, you would get a
volume of 1. That seems like a bug.

 }
 
 VideoRtpReceiver::VideoRtpReceiver(MediaStreamInterface* stream,
                                    const std::string& track_id,
                                    rtc::Thread* worker_thread,
                                    uint32_t ssrc,
-                                   VideoProviderInterface* provider)
+                                   cricket::VideoChannel* channel)
     : id_(track_id),
       ssrc_(ssrc),
-      provider_(provider),
+      channel_(channel),
       source_(new RefCountedObject<VideoTrackSource>(&broadcaster_,
                                                      true /* remote */)),
       track_(VideoTrackProxy::Create(
@@ -102,34 +122,51 @@ VideoRtpReceiver::VideoRtpReceiver(MediaStreamInterface* stream,
                                             worker_thread,
                                             source_)))) {
   source_->SetState(MediaSourceInterface::kLive);
-  provider_->SetVideoPlayout(ssrc_, true, &broadcaster_);
+  if (!channel_) {
+    LOG(LS_ERROR)
+        << "VideoRtpReceiver::VideoRtpReceiver: No video channel exists.";
+  } else {
+    RTC_DCHECK(channel_->SetSink(ssrc_, &broadcaster_));
+  }
   stream->AddTrack(track_);
 }
 
 VideoRtpReceiver::~VideoRtpReceiver() {
   // Since cricket::VideoRenderer is not reference counted,
-  // we need to remove it from the provider before we are deleted.
+  // we need to remove it from the channel before we are deleted.
   Stop();
 }
 
 RtpParameters VideoRtpReceiver::GetParameters() const {
-  return provider_->GetVideoRtpReceiveParameters(ssrc_);
+  if (!channel_ || stopped_) {
+    return RtpParameters();
+  }
+  return channel_->GetRtpReceiveParameters(ssrc_);
 }
 
 bool VideoRtpReceiver::SetParameters(const RtpParameters& parameters) {
   TRACE_EVENT0("webrtc", "VideoRtpReceiver::SetParameters");
-  return provider_->SetVideoRtpReceiveParameters(ssrc_, parameters);
+  if (!channel_ || stopped_) {
+    return false;
+  }
+  return channel_->SetRtpReceiveParameters(ssrc_, parameters);
 }
 
 void VideoRtpReceiver::Stop() {
   // TODO(deadbeef): Need to do more here to fully stop receiving packets.
-  if (!provider_) {
+  if (stopped_) {
     return;
   }
   source_->SetState(MediaSourceInterface::kEnded);
   source_->OnSourceDestroyed();
-  provider_->SetVideoPlayout(ssrc_, false, nullptr);
-  provider_ = nullptr;
+  if (!channel_) {
+    LOG(LS_WARNING) << "VideoRtpReceiver::Stop: No video channel exists.";
+  } else {
+    // Allow that SetSink fail. This is the normal case when the underlying
+    // media channel has already been deleted.
+    channel_->SetSink(ssrc_, nullptr);
+  }
+  stopped_ = true;
 }
 
 }  // namespace webrtc