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Unified Diff: talk/app/webrtc/peerconnection.cc

Issue 1351803002: Exposing RtpSenders and RtpReceivers from PeerConnection. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 3 months ago
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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

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