Chromium Code Reviews| 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 |