Restoring behavior where PeerConnection tracks changes to MediaStreams.
If a MediaStream is added to a PeerConnection, and later a track
is added to the MediaStream, a new RtpSender will now be created for
that track, and it will appear in subsequent offers.
Similarly, removed tracks will remove RtpSenders.
BUG=webrtc:5265
Review URL: https://codereview.webrtc.org/1507973003
Cr-Commit-Position: refs/heads/master@{#11040}
diff --git a/talk/app/webrtc/mediastreamobserver.cc b/talk/app/webrtc/mediastreamobserver.cc
index a856164..2650b9a 100644
--- a/talk/app/webrtc/mediastreamobserver.cc
+++ b/talk/app/webrtc/mediastreamobserver.cc
@@ -27,4 +27,75 @@
#include "talk/app/webrtc/mediastreamobserver.h"
-// This file is currently stubbed so that Chromium's build files can be updated.
+#include <algorithm>
+
+namespace webrtc {
+
+MediaStreamObserver::MediaStreamObserver(MediaStreamInterface* stream)
+ : stream_(stream),
+ cached_audio_tracks_(stream->GetAudioTracks()),
+ cached_video_tracks_(stream->GetVideoTracks()) {
+ stream_->RegisterObserver(this);
+}
+
+MediaStreamObserver::~MediaStreamObserver() {
+ stream_->UnregisterObserver(this);
+}
+
+void MediaStreamObserver::OnChanged() {
+ AudioTrackVector new_audio_tracks = stream_->GetAudioTracks();
+ VideoTrackVector new_video_tracks = stream_->GetVideoTracks();
+
+ // Find removed audio tracks.
+ for (const auto& cached_track : cached_audio_tracks_) {
+ auto it = std::find_if(
+ new_audio_tracks.begin(), new_audio_tracks.end(),
+ [cached_track](const AudioTrackVector::value_type& new_track) {
+ return new_track->id().compare(cached_track->id()) == 0;
+ });
+ if (it == new_audio_tracks.end()) {
+ SignalAudioTrackRemoved(cached_track.get(), stream_);
+ }
+ }
+
+ // Find added audio tracks.
+ for (const auto& new_track : new_audio_tracks) {
+ auto it = std::find_if(
+ cached_audio_tracks_.begin(), cached_audio_tracks_.end(),
+ [new_track](const AudioTrackVector::value_type& cached_track) {
+ return new_track->id().compare(cached_track->id()) == 0;
+ });
+ if (it == cached_audio_tracks_.end()) {
+ SignalAudioTrackAdded(new_track.get(), stream_);
+ }
+ }
+
+ // Find removed video tracks.
+ for (const auto& cached_track : cached_video_tracks_) {
+ auto it = std::find_if(
+ new_video_tracks.begin(), new_video_tracks.end(),
+ [cached_track](const VideoTrackVector::value_type& new_track) {
+ return new_track->id().compare(cached_track->id()) == 0;
+ });
+ if (it == new_video_tracks.end()) {
+ SignalVideoTrackRemoved(cached_track.get(), stream_);
+ }
+ }
+
+ // Find added video tracks.
+ for (const auto& new_track : new_video_tracks) {
+ auto it = std::find_if(
+ cached_video_tracks_.begin(), cached_video_tracks_.end(),
+ [new_track](const VideoTrackVector::value_type& cached_track) {
+ return new_track->id().compare(cached_track->id()) == 0;
+ });
+ if (it == cached_video_tracks_.end()) {
+ SignalVideoTrackAdded(new_track.get(), stream_);
+ }
+ }
+
+ cached_audio_tracks_ = new_audio_tracks;
+ cached_video_tracks_ = new_video_tracks;
+}
+
+} // namespace webrtc
