Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/rtpsender.cc

Index: talk/app/webrtc/rtpsender.cc
diff --git a/talk/app/webrtc/rtpsender.cc b/talk/app/webrtc/rtpsender.cc
index 3049222f2a4c9d615c03c132dd34d7a69b398012..28ba073fc5ab342a538cd6d435211fa2a6fab509 100644
--- a/talk/app/webrtc/rtpsender.cc
+++ b/talk/app/webrtc/rtpsender.cc
@@ -27,4 +27,181 @@
 
 #include "talk/app/webrtc/rtpsender.h"
 
-// This file is currently stubbed so that Chromium's build files can be updated.
+#include "talk/app/webrtc/localaudiosource.h"
+#include "talk/app/webrtc/videosourceinterface.h"
+
+namespace webrtc {
+
+LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {}
+
+LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
+  rtc::CritScope lock(&lock_);
+  if (sink_)
+    sink_->OnClose();
+}
+
+void LocalAudioSinkAdapter::OnData(const void* audio_data,
+                                   int bits_per_sample,
+                                   int sample_rate,
+                                   int number_of_channels,
+                                   size_t number_of_frames) {
+  rtc::CritScope lock(&lock_);
+  if (sink_) {
+    sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
+                  number_of_frames);
+  }
+}
+
+void LocalAudioSinkAdapter::SetSink(cricket::AudioRenderer::Sink* sink) {
+  rtc::CritScope lock(&lock_);
+  ASSERT(!sink || !sink_);
+  sink_ = sink;
+}
+
+AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
+                               uint32 ssrc,
+                               AudioProviderInterface* provider)
+    : id_(track->id()),
+      track_(track),
+      ssrc_(ssrc),
+      provider_(provider),
+      cached_track_enabled_(track->enabled()),
+      sink_adapter_(new LocalAudioSinkAdapter()) {
+  track_->RegisterObserver(this);
+  track_->AddSink(sink_adapter_.get());
+  Reconfigure();
+}
+
+AudioRtpSender::~AudioRtpSender() {
+  track_->RemoveSink(sink_adapter_.get());
+  track_->UnregisterObserver(this);
+  Stop();
+}
+
+void AudioRtpSender::OnChanged() {
+  if (cached_track_enabled_ != track_->enabled()) {
+    cached_track_enabled_ = track_->enabled();
+    Reconfigure();
+  }
+}
+
+bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
+  if (track->kind() != "audio") {
+    LOG(LS_ERROR) << "SetTrack called on audio RtpSender with " << track->kind()
+                  << " track.";
+    return false;
+  }
+  AudioTrackInterface* audio_track = static_cast<AudioTrackInterface*>(track);
+
+  // Detach from old track.
+  track_->RemoveSink(sink_adapter_.get());
+  track_->UnregisterObserver(this);
+
+  // Attach to new track.
+  track_ = audio_track;
+  cached_track_enabled_ = track_->enabled();
+  track_->RegisterObserver(this);
+  track_->AddSink(sink_adapter_.get());
+  Reconfigure();
+  return true;
+}
+
+void AudioRtpSender::Stop() {
+  // TODO(deadbeef): Need to do more here to fully stop sending packets.
+  if (!provider_) {
+    return;
+  }
+  cricket::AudioOptions options;
+  provider_->SetAudioSend(ssrc_, false, options, nullptr);
+  provider_ = nullptr;
+}
+
+void AudioRtpSender::Reconfigure() {
+  if (!provider_) {
+    return;
+  }
+  cricket::AudioOptions options;
+  if (track_->enabled() && track_->GetSource()) {
+    // TODO(xians): Remove this static_cast since we should be able to connect
+    // a remote audio track to peer connection.
+    options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
+  }
+
+  // Use the renderer if the audio track has one, otherwise use the sink
+  // adapter owned by this class.
+  cricket::AudioRenderer* renderer =
+      track_->GetRenderer() ? track_->GetRenderer() : sink_adapter_.get();
+  ASSERT(renderer != nullptr);
+  provider_->SetAudioSend(ssrc_, track_->enabled(), options, renderer);
+}
+
+VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
+                               uint32 ssrc,
+                               VideoProviderInterface* provider)
+    : id_(track->id()),
+      track_(track),
+      ssrc_(ssrc),
+      provider_(provider),
+      cached_track_enabled_(track->enabled()) {
+  track_->RegisterObserver(this);
+  VideoSourceInterface* source = track_->GetSource();
+  if (source) {
+    provider_->SetCaptureDevice(ssrc_, source->GetVideoCapturer());
+  }
+  Reconfigure();
+}
+
+VideoRtpSender::~VideoRtpSender() {
+  track_->UnregisterObserver(this);
+  Stop();
+}
+
+void VideoRtpSender::OnChanged() {
+  if (cached_track_enabled_ != track_->enabled()) {
+    cached_track_enabled_ = track_->enabled();
+    Reconfigure();
+  }
+}
+
+bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
+  if (track->kind() != "video") {
+    LOG(LS_ERROR) << "SetTrack called on video RtpSender with " << track->kind()
+                  << " track.";
+    return false;
+  }
+  VideoTrackInterface* video_track = static_cast<VideoTrackInterface*>(track);
+
+  // Detach from old track.
+  track_->UnregisterObserver(this);
+
+  // Attach to new track.
+  track_ = video_track;
+  cached_track_enabled_ = track_->enabled();
+  track_->RegisterObserver(this);
+  Reconfigure();
+  return true;
+}
+
+void VideoRtpSender::Stop() {
+  // TODO(deadbeef): Need to do more here to fully stop sending packets.
+  if (!provider_) {
+    return;
+  }
+  provider_->SetCaptureDevice(ssrc_, nullptr);
+  provider_->SetVideoSend(ssrc_, false, nullptr);
+  provider_ = nullptr;
+}
+
+void VideoRtpSender::Reconfigure() {
+  if (!provider_) {
+    return;
+  }
+  const cricket::VideoOptions* options = nullptr;
+  VideoSourceInterface* source = track_->GetSource();
+  if (track_->enabled() && source) {
+    options = source->options();
+  }
+  provider_->SetVideoSend(ssrc_, track_->enabled(), options);
+}
+
+}  // namespace webrtc