Exposing RtpSenders and RtpReceivers from PeerConnection.
This CL essentially converts [Local|Remote]TrackHandler to
Rtp[Sender|Receiver], and adds a "SetTrack" method for RtpSender.
It also gets rid of MediaStreamHandler and MediaStreamHandlerContainer,
since these classes weren't really anything more than containers.
PeerConnection now manages the RtpSenders and RtpReceivers directly.
Review URL: https://codereview.webrtc.org/1351803002
Cr-Commit-Position: refs/heads/master@{#10100}
diff --git a/talk/app/webrtc/rtpreceiver.cc b/talk/app/webrtc/rtpreceiver.cc
index 39b669f..b8eca30 100644
--- a/talk/app/webrtc/rtpreceiver.cc
+++ b/talk/app/webrtc/rtpreceiver.cc
@@ -27,4 +27,80 @@
#include "talk/app/webrtc/rtpreceiver.h"
-// This file is currently stubbed so that Chromium's build files can be updated.
+#include "talk/app/webrtc/videosourceinterface.h"
+
+namespace webrtc {
+
+AudioRtpReceiver::AudioRtpReceiver(AudioTrackInterface* track,
+ uint32 ssrc,
+ AudioProviderInterface* provider)
+ : id_(track->id()),
+ track_(track),
+ ssrc_(ssrc),
+ provider_(provider),
+ cached_track_enabled_(track->enabled()) {
+ track_->RegisterObserver(this);
+ track_->GetSource()->RegisterAudioObserver(this);
+ Reconfigure();
+}
+
+AudioRtpReceiver::~AudioRtpReceiver() {
+ track_->GetSource()->UnregisterAudioObserver(this);
+ track_->UnregisterObserver(this);
+ Stop();
+}
+
+void AudioRtpReceiver::OnChanged() {
+ if (cached_track_enabled_ != track_->enabled()) {
+ cached_track_enabled_ = track_->enabled();
+ Reconfigure();
+ }
+}
+
+void AudioRtpReceiver::OnSetVolume(double volume) {
+ // 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);
+}
+
+void AudioRtpReceiver::Stop() {
+ // TODO(deadbeef): Need to do more here to fully stop receiving packets.
+ if (!provider_) {
+ return;
+ }
+ provider_->SetAudioPlayout(ssrc_, false, nullptr);
+ provider_ = nullptr;
+}
+
+void AudioRtpReceiver::Reconfigure() {
+ if (!provider_) {
+ return;
+ }
+ provider_->SetAudioPlayout(ssrc_, track_->enabled(), track_->GetRenderer());
+}
+
+VideoRtpReceiver::VideoRtpReceiver(VideoTrackInterface* track,
+ uint32 ssrc,
+ VideoProviderInterface* provider)
+ : id_(track->id()), track_(track), ssrc_(ssrc), provider_(provider) {
+ provider_->SetVideoPlayout(ssrc_, true, track_->GetSource()->FrameInput());
+}
+
+VideoRtpReceiver::~VideoRtpReceiver() {
+ // Since cricket::VideoRenderer is not reference counted,
+ // we need to remove it from the provider before we are deleted.
+ Stop();
+}
+
+void VideoRtpReceiver::Stop() {
+ // TODO(deadbeef): Need to do more here to fully stop receiving packets.
+ if (!provider_) {
+ return;
+ }
+ provider_->SetVideoPlayout(ssrc_, false, nullptr);
+ provider_ = nullptr;
+}
+
+} // namespace webrtc