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Restoring behavior where PeerConnection tracks changes to MediaStreams.

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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}
This commit is contained in:
deadbeef
2015-12-15 19:24:43 -08:00
committed by Commit bot
parent 44f0819978
commit eb45981165
6 changed files with 261 additions and 59 deletions
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73
talk/app/webrtc/mediastreamobserver.cc
View File

@ -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

36
talk/app/webrtc/mediastreamobserver.h
View File

@ -25,9 +25,41 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
// This file is currently stubbed so that Chromium's build files can be updated.
#ifndef TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_
#define TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_
#include "talk/app/webrtc/mediastreaminterface.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/sigslot.h"
namespace webrtc {
// Helper class which will listen for changes to a stream and emit the
// corresponding signals.
class MediaStreamObserver : public ObserverInterface {
public:
explicit MediaStreamObserver(MediaStreamInterface* stream);
~MediaStreamObserver();
const MediaStreamInterface* stream() const { return stream_; }
void OnChanged() override;
sigslot::signal2<AudioTrackInterface*, MediaStreamInterface*>
SignalAudioTrackAdded;
sigslot::signal2<AudioTrackInterface*, MediaStreamInterface*>
SignalAudioTrackRemoved;
sigslot::signal2<VideoTrackInterface*, MediaStreamInterface*>
SignalVideoTrackAdded;
sigslot::signal2<VideoTrackInterface*, MediaStreamInterface*>
SignalVideoTrackRemoved;
private:
rtc::scoped_refptr<MediaStreamInterface> stream_;
AudioTrackVector cached_audio_tracks_;
VideoTrackVector cached_video_tracks_;
};
} // namespace webrtc
#endif // TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_

154
talk/app/webrtc/peerconnection.cc
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@ -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 @@ bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
}
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 @@ bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
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 @@ void PeerConnection::ChangeSignalingState(
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) {

13
talk/app/webrtc/peerconnection.h
View File

@ -42,6 +42,7 @@
namespace webrtc {
class MediaStreamObserver;
class RemoteMediaStreamFactory;
typedef std::vector<PortAllocatorFactoryInterface::StunConfiguration>
@ -201,6 +202,16 @@ class PeerConnection : public PeerConnectionInterface,
void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state);
void ChangeSignalingState(SignalingState signaling_state);
// Signals from MediaStreamObserver.
void OnAudioTrackAdded(AudioTrackInterface* track,
MediaStreamInterface* stream);
void OnAudioTrackRemoved(AudioTrackInterface* track,
MediaStreamInterface* stream);
void OnVideoTrackAdded(VideoTrackInterface* track,
MediaStreamInterface* stream);
void OnVideoTrackRemoved(VideoTrackInterface* track,
MediaStreamInterface* stream);
rtc::Thread* signaling_thread() const {
return factory_->signaling_thread();
}
@ -364,6 +375,8 @@ class PeerConnection : public PeerConnectionInterface,
// Streams created as a result of SetRemoteDescription.
rtc::scoped_refptr<StreamCollection> remote_streams_;
std::vector<rtc::scoped_ptr<MediaStreamObserver>> stream_observers_;
// These lists store track info seen in local/remote descriptions.
TrackInfos remote_audio_tracks_;
TrackInfos remote_video_tracks_;

42
talk/app/webrtc/peerconnectioninterface_unittest.cc
View File

@ -1156,6 +1156,48 @@ TEST_F(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
EXPECT_NE(audio_ssrc, video_ssrc);
}
// Test that it's possible to call AddTrack on a MediaStream after adding
// the stream to a PeerConnection.
// TODO(deadbeef): Remove this test once this behavior is no longer supported.
TEST_F(PeerConnectionInterfaceTest, AddTrackAfterAddStream) {
CreatePeerConnection();
// Create audio stream and add to PeerConnection.
AddVoiceStream(kStreamLabel1);
MediaStreamInterface* stream = pc_->local_streams()->at(0);
// Add video track to the audio-only stream.
scoped_refptr<VideoTrackInterface> video_track(
pc_factory_->CreateVideoTrack("video_label", nullptr));
stream->AddTrack(video_track.get());
scoped_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(offer.use(), nullptr));
const cricket::MediaContentDescription* video_desc =
cricket::GetFirstVideoContentDescription(offer->description());
EXPECT_TRUE(video_desc != nullptr);
}
// Test that it's possible to call RemoveTrack on a MediaStream after adding
// the stream to a PeerConnection.
// TODO(deadbeef): Remove this test once this behavior is no longer supported.
TEST_F(PeerConnectionInterfaceTest, RemoveTrackAfterAddStream) {
CreatePeerConnection();
// Create audio/video stream and add to PeerConnection.
AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
MediaStreamInterface* stream = pc_->local_streams()->at(0);
// Remove the video track.
stream->RemoveTrack(stream->GetVideoTracks()[0]);
scoped_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(offer.use(), nullptr));
const cricket::MediaContentDescription* video_desc =
cricket::GetFirstVideoContentDescription(offer->description());
EXPECT_TRUE(video_desc == nullptr);
}
// Test that we can specify a certain track that we want statistics about.
TEST_F(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) {
InitiateCall();

2
talk/libjingle.gyp
View File

@ -746,6 +746,8 @@
'app/webrtc/mediastream.cc',
'app/webrtc/mediastream.h',
'app/webrtc/mediastreaminterface.h',
'app/webrtc/mediastreamobserver.cc',
'app/webrtc/mediastreamobserver.h',
'app/webrtc/mediastreamprovider.h',
'app/webrtc/mediastreamproxy.h',
'app/webrtc/mediastreamtrack.h',