Index: talk/app/webrtc/peerconnection.cc |
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc |
index 9e2cc731c07cdab9af64a69c103f30ccb28878cc..7666fd35621222a2bc91fa325db09e485ee4836c 100644 |
--- a/talk/app/webrtc/peerconnection.cc |
+++ b/talk/app/webrtc/peerconnection.cc |
@@ -33,7 +33,8 @@ |
#include "talk/app/webrtc/jsepicecandidate.h" |
#include "talk/app/webrtc/jsepsessiondescription.h" |
#include "talk/app/webrtc/mediaconstraintsinterface.h" |
-#include "talk/app/webrtc/mediastreamhandler.h" |
+#include "talk/app/webrtc/rtpreceiver.h" |
+#include "talk/app/webrtc/rtpsender.h" |
#include "talk/app/webrtc/streamcollection.h" |
#include "webrtc/p2p/client/basicportallocator.h" |
#include "talk/session/media/channelmanager.h" |
@@ -339,10 +340,17 @@ PeerConnection::PeerConnection(PeerConnectionFactory* factory) |
PeerConnection::~PeerConnection() { |
ASSERT(signaling_thread()->IsCurrent()); |
- if (mediastream_signaling_) |
+ if (mediastream_signaling_) { |
mediastream_signaling_->TearDown(); |
- if (stream_handler_container_) |
- stream_handler_container_->TearDown(); |
+ } |
+ // Need to detach RTP senders/receivers from WebRtcSession, |
+ // since it's about to be destroyed. |
+ for (const auto& sender : rtp_senders_) { |
+ sender->Stop(); |
+ } |
+ for (const auto& receiver : rtp_receivers_) { |
+ receiver->Stop(); |
+ } |
} |
bool PeerConnection::Initialize( |
@@ -399,8 +407,6 @@ bool PeerConnection::Initialize( |
factory_->worker_thread(), |
port_allocator_.get(), |
mediastream_signaling_.get())); |
- stream_handler_container_.reset(new MediaStreamHandlerContainer( |
- session_.get(), session_.get())); |
stats_.reset(new StatsCollector(session_.get())); |
// Initialize the WebRtcSession. It creates transport channels etc. |
@@ -806,7 +812,6 @@ void PeerConnection::OnAddRemoteStream(MediaStreamInterface* stream) { |
} |
void PeerConnection::OnRemoveRemoteStream(MediaStreamInterface* stream) { |
- stream_handler_container_->RemoveRemoteStream(stream); |
observer_->OnRemoveStream(stream); |
} |
@@ -817,53 +822,96 @@ void PeerConnection::OnAddDataChannel(DataChannelInterface* data_channel) { |
void PeerConnection::OnAddRemoteAudioTrack(MediaStreamInterface* stream, |
AudioTrackInterface* audio_track, |
- uint32 ssrc) { |
- stream_handler_container_->AddRemoteAudioTrack(stream, audio_track, ssrc); |
+ uint32 ssrc, |
+ const std::string& mid) { |
+ rtp_receivers_.push_back(rtc::scoped_refptr<RtpReceiverInterface>( |
+ new rtc::RefCountedObject<AudioRtpReceiver>(audio_track, ssrc, mid, |
+ session_.get()))); |
} |
void PeerConnection::OnAddRemoteVideoTrack(MediaStreamInterface* stream, |
VideoTrackInterface* video_track, |
- uint32 ssrc) { |
- stream_handler_container_->AddRemoteVideoTrack(stream, video_track, ssrc); |
+ uint32 ssrc, |
+ const std::string& mid) { |
+ rtp_receivers_.push_back(rtc::scoped_refptr<RtpReceiverInterface>( |
+ new rtc::RefCountedObject<VideoRtpReceiver>(video_track, ssrc, mid, |
+ session_.get()))); |
} |
void PeerConnection::OnRemoveRemoteAudioTrack( |
MediaStreamInterface* stream, |
AudioTrackInterface* audio_track) { |
- stream_handler_container_->RemoveRemoteTrack(stream, audio_track); |
+ auto it = FindRtpReceiverForTrack(audio_track); |
+ if (it == rtp_receivers_.end()) { |
+ LOG(LS_WARNING) << "RtpReceiver for track with id " << audio_track->id() |
+ << " doesn't exist."; |
+ } else { |
+ (*it)->Stop(); |
+ rtp_receivers_.erase(it); |
pthatcher1
2015/09/17 04:25:46
Long-term, I think we want to keep senders and rec
Taylor Brandstetter
2015/09/23 00:10:45
Can't an m-line go away if someone calls "stop" on
pthatcher1
2015/09/23 14:47:39
No, an m-line can never go away.
|
+ } |
} |
void PeerConnection::OnRemoveRemoteVideoTrack( |
MediaStreamInterface* stream, |
VideoTrackInterface* video_track) { |
- stream_handler_container_->RemoveRemoteTrack(stream, video_track); |
+ auto it = FindRtpReceiverForTrack(video_track); |
+ if (it == rtp_receivers_.end()) { |
+ LOG(LS_WARNING) << "RtpReceiver for track with id " << video_track->id() |
+ << " doesn't exist."; |
+ } else { |
+ (*it)->Stop(); |
+ rtp_receivers_.erase(it); |
+ } |
} |
+ |
void PeerConnection::OnAddLocalAudioTrack(MediaStreamInterface* stream, |
AudioTrackInterface* audio_track, |
- uint32 ssrc) { |
- stream_handler_container_->AddLocalAudioTrack(stream, audio_track, ssrc); |
+ uint32 ssrc, |
+ const std::string& mid) { |
+ rtp_senders_.push_back(rtc::scoped_refptr<RtpSenderInterface>( |
+ new rtc::RefCountedObject<AudioRtpSender>(audio_track, ssrc, mid, |
+ session_.get()))); |
stats_->AddLocalAudioTrack(audio_track, ssrc); |
} |
+ |
void PeerConnection::OnAddLocalVideoTrack(MediaStreamInterface* stream, |
VideoTrackInterface* video_track, |
- uint32 ssrc) { |
- stream_handler_container_->AddLocalVideoTrack(stream, video_track, ssrc); |
+ uint32 ssrc, |
+ const std::string& mid) { |
+ rtp_senders_.push_back(rtc::scoped_refptr<RtpSenderInterface>( |
+ new rtc::RefCountedObject<VideoRtpSender>(video_track, ssrc, mid, |
+ session_.get()))); |
} |
void PeerConnection::OnRemoveLocalAudioTrack(MediaStreamInterface* stream, |
AudioTrackInterface* audio_track, |
uint32 ssrc) { |
- stream_handler_container_->RemoveLocalTrack(stream, audio_track); |
+ auto it = FindRtpSenderForTrack(audio_track); |
+ if (it == rtp_senders_.end()) { |
+ LOG(LS_WARNING) << "RtpSender for track with id " << audio_track->id() |
+ << " doesn't exist."; |
+ return; |
+ } else { |
+ (*it)->Stop(); |
+ rtp_senders_.erase(it); |
+ } |
stats_->RemoveLocalAudioTrack(audio_track, ssrc); |
} |
void PeerConnection::OnRemoveLocalVideoTrack(MediaStreamInterface* stream, |
VideoTrackInterface* video_track) { |
- stream_handler_container_->RemoveLocalTrack(stream, video_track); |
+ auto it = FindRtpSenderForTrack(video_track); |
+ if (it == rtp_senders_.end()) { |
+ LOG(LS_WARNING) << "RtpSender for track with id " << video_track->id() |
+ << " doesn't exist."; |
+ return; |
+ } else { |
+ (*it)->Stop(); |
+ rtp_senders_.erase(it); |
+ } |
} |
void PeerConnection::OnRemoveLocalStream(MediaStreamInterface* stream) { |
- stream_handler_container_->RemoveLocalStream(stream); |
} |
void PeerConnection::OnIceConnectionChange( |
@@ -913,4 +961,19 @@ void PeerConnection::ChangeSignalingState( |
observer_->OnStateChange(PeerConnectionObserver::kSignalingState); |
} |
+PeerConnectionInterface::RtpSenders::iterator |
+PeerConnection::FindRtpSenderForTrack(MediaStreamTrackInterface* track) { |
+ return std::find_if( |
+ rtp_senders_.begin(), rtp_senders_.end(), |
+ [track](const rtc::scoped_refptr<RtpSenderInterface>& sender) |
+ -> bool { return sender->track() == track; }); |
pthatcher1
2015/09/17 04:25:46
Is the "-> bool" necessary? I thought it could in
Taylor Brandstetter
2015/09/23 00:10:45
Oh, I guess so. Removed.
|
+} |
+PeerConnectionInterface::RtpReceivers::iterator |
+PeerConnection::FindRtpReceiverForTrack(MediaStreamTrackInterface* track) { |
+ return std::find_if( |
+ rtp_receivers_.begin(), rtp_receivers_.end(), |
+ [track](const rtc::scoped_refptr<RtpReceiverInterface>& receiver) |
+ -> bool { return receiver->track() == track; }); |
+} |
+ |
} // namespace webrtc |