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/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index 6cca166..3a38248 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -27,6 +27,7 @@
#include "talk/app/webrtc/peerconnection.h"
+#include <algorithm>
#include <vector>
#include <cctype> // for isdigit
@@ -36,6 +37,7 @@
#include "talk/app/webrtc/jsepsessiondescription.h"
#include "talk/app/webrtc/mediaconstraintsinterface.h"
#include "talk/app/webrtc/mediastream.h"
+#include "talk/app/webrtc/mediastreamobserver.h"
#include "talk/app/webrtc/mediastreamproxy.h"
#include "talk/app/webrtc/mediastreamtrackproxy.h"
#include "talk/app/webrtc/remoteaudiosource.h"
@@ -740,48 +742,22 @@
}
local_streams_->AddStream(local_stream);
+ MediaStreamObserver* observer = new MediaStreamObserver(local_stream);
+ observer->SignalAudioTrackAdded.connect(this,
+ &PeerConnection::OnAudioTrackAdded);
+ observer->SignalAudioTrackRemoved.connect(
+ this, &PeerConnection::OnAudioTrackRemoved);
+ observer->SignalVideoTrackAdded.connect(this,
+ &PeerConnection::OnVideoTrackAdded);
+ observer->SignalVideoTrackRemoved.connect(
+ this, &PeerConnection::OnVideoTrackRemoved);
+ stream_observers_.push_back(rtc::scoped_ptr<MediaStreamObserver>(observer));
for (const auto& track : local_stream->GetAudioTracks()) {
- auto sender = FindSenderForTrack(track.get());
- if (sender == senders_.end()) {
- // Normal case; we've never seen this track before.
- AudioRtpSender* new_sender = new AudioRtpSender(
- track.get(), local_stream->label(), session_.get(), stats_.get());
- senders_.push_back(new_sender);
- // If the sender has already been configured in SDP, we call SetSsrc,
- // which will connect the sender to the underlying transport. This can
- // occur if a local session description that contains the ID of the sender
- // is set before AddStream is called. It can also occur if the local
- // session description is not changed and RemoveStream is called, and
- // later AddStream is called again with the same stream.
- const TrackInfo* track_info = FindTrackInfo(
- local_audio_tracks_, local_stream->label(), track->id());
- if (track_info) {
- new_sender->SetSsrc(track_info->ssrc);
- }
- } else {
- // We already have a sender for this track, so just change the stream_id
- // so that it's correct in the next call to CreateOffer.
- (*sender)->set_stream_id(local_stream->label());
- }
+ OnAudioTrackAdded(track.get(), local_stream);
}
for (const auto& track : local_stream->GetVideoTracks()) {
- auto sender = FindSenderForTrack(track.get());
- if (sender == senders_.end()) {
- // Normal case; we've never seen this track before.
- VideoRtpSender* new_sender = new VideoRtpSender(
- track.get(), local_stream->label(), session_.get());
- senders_.push_back(new_sender);
- const TrackInfo* track_info = FindTrackInfo(
- local_video_tracks_, local_stream->label(), track->id());
- if (track_info) {
- new_sender->SetSsrc(track_info->ssrc);
- }
- } else {
- // We already have a sender for this track, so just change the stream_id
- // so that it's correct in the next call to CreateOffer.
- (*sender)->set_stream_id(local_stream->label());
- }
+ OnVideoTrackAdded(track.get(), local_stream);
}
stats_->AddStream(local_stream);
@@ -789,32 +765,24 @@
return true;
}
-// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
-// indefinitely, when we have unified plan SDP.
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
for (const auto& track : local_stream->GetAudioTracks()) {
- auto sender = FindSenderForTrack(track.get());
- if (sender == senders_.end()) {
- LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
- << " doesn't exist.";
- continue;
- }
- (*sender)->Stop();
- senders_.erase(sender);
+ OnAudioTrackRemoved(track.get(), local_stream);
}
for (const auto& track : local_stream->GetVideoTracks()) {
- auto sender = FindSenderForTrack(track.get());
- if (sender == senders_.end()) {
- LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
- << " doesn't exist.";
- continue;
- }
- (*sender)->Stop();
- senders_.erase(sender);
+ OnVideoTrackRemoved(track.get(), local_stream);
}
local_streams_->RemoveStream(local_stream);
+ stream_observers_.erase(
+ std::remove_if(
+ stream_observers_.begin(), stream_observers_.end(),
+ [local_stream](const rtc::scoped_ptr<MediaStreamObserver>& observer) {
+ return observer->stream()->label().compare(local_stream->label()) ==
+ 0;
+ }),
+ stream_observers_.end());
if (IsClosed()) {
return;
@@ -1432,6 +1400,80 @@
observer_->OnStateChange(PeerConnectionObserver::kSignalingState);
}
+void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
+ MediaStreamInterface* stream) {
+ auto sender = FindSenderForTrack(track);
+ if (sender != senders_.end()) {
+ // We already have a sender for this track, so just change the stream_id
+ // so that it's correct in the next call to CreateOffer.
+ (*sender)->set_stream_id(stream->label());
+ return;
+ }
+
+ // Normal case; we've never seen this track before.
+ AudioRtpSender* new_sender =
+ new AudioRtpSender(track, stream->label(), session_.get(), stats_.get());
+ senders_.push_back(new_sender);
+ // If the sender has already been configured in SDP, we call SetSsrc,
+ // which will connect the sender to the underlying transport. This can
+ // occur if a local session description that contains the ID of the sender
+ // is set before AddStream is called. It can also occur if the local
+ // session description is not changed and RemoveStream is called, and
+ // later AddStream is called again with the same stream.
+ const TrackInfo* track_info =
+ FindTrackInfo(local_audio_tracks_, stream->label(), track->id());
+ if (track_info) {
+ new_sender->SetSsrc(track_info->ssrc);
+ }
+}
+
+// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
+// indefinitely, when we have unified plan SDP.
+void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
+ MediaStreamInterface* stream) {
+ auto sender = FindSenderForTrack(track);
+ if (sender == senders_.end()) {
+ LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
+ << " doesn't exist.";
+ return;
+ }
+ (*sender)->Stop();
+ senders_.erase(sender);
+}
+
+void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
+ MediaStreamInterface* stream) {
+ auto sender = FindSenderForTrack(track);
+ if (sender != senders_.end()) {
+ // We already have a sender for this track, so just change the stream_id
+ // so that it's correct in the next call to CreateOffer.
+ (*sender)->set_stream_id(stream->label());
+ return;
+ }
+
+ // Normal case; we've never seen this track before.
+ VideoRtpSender* new_sender =
+ new VideoRtpSender(track, stream->label(), session_.get());
+ senders_.push_back(new_sender);
+ const TrackInfo* track_info =
+ FindTrackInfo(local_video_tracks_, stream->label(), track->id());
+ if (track_info) {
+ new_sender->SetSsrc(track_info->ssrc);
+ }
+}
+
+void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
+ MediaStreamInterface* stream) {
+ auto sender = FindSenderForTrack(track);
+ if (sender == senders_.end()) {
+ LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
+ << " doesn't exist.";
+ return;
+ }
+ (*sender)->Stop();
+ senders_.erase(sender);
+}
+
void PeerConnection::PostSetSessionDescriptionFailure(
SetSessionDescriptionObserver* observer,
const std::string& error) {
