Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/peerconnection.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 15 matching lines @@
  */
 
 #include "talk/app/webrtc/peerconnection.h"
 
 #include <vector>
 
 #include "talk/app/webrtc/dtmfsender.h"
 #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/streamcollection.h"
 #include "webrtc/p2p/client/basicportallocator.h"
 #include "talk/session/media/channelmanager.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/system_wrappers/interface/field_trial.h"
 
 namespace {
 
 using webrtc::PeerConnectionInterface;
@@ skipping 285 matching lines @@
       observer_(NULL),
       uma_observer_(NULL),
       signaling_state_(kStable),
       ice_state_(kIceNew),
       ice_connection_state_(kIceConnectionNew),
       ice_gathering_state_(kIceGatheringNew) {
 }
 
 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& pair : rtp_sender_pairs_) {
+    pair.first->DetachFromProvider();
+  }
+  for (const auto& pair : rtp_receiver_pairs_) {
+    pair.first->DetachFromProvider();
+  }
 }
 
 bool PeerConnection::Initialize(
     const PeerConnectionInterface::RTCConfiguration& configuration,
     const MediaConstraintsInterface* constraints,
     PortAllocatorFactoryInterface* allocator_factory,
     rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
     PeerConnectionObserver* observer) {
   ASSERT(observer != NULL);
   if (!observer)
@@ skipping 36 matching lines @@
   port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
 
   mediastream_signaling_.reset(new MediaStreamSignaling(
       factory_->signaling_thread(), this, factory_->channel_manager()));
 
   session_.reset(new WebRtcSession(factory_->channel_manager(),
                                    factory_->signaling_thread(),
                                    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.
   if (!session_->Initialize(factory_->options(), constraints,
                             dtls_identity_store.Pass(), configuration))
     return false;
 
   // Register PeerConnection as receiver of local ice candidates.
   // All the callbacks will be posted to the application from PeerConnection.
   session_->RegisterIceObserver(this);
   session_->SignalState.connect(this, &PeerConnection::OnSessionStateChange);
   return true;
 }
 
 rtc::scoped_refptr<StreamCollectionInterface>
 PeerConnection::local_streams() {
   return mediastream_signaling_->local_streams();
 }
 
 rtc::scoped_refptr<StreamCollectionInterface>
 PeerConnection::remote_streams() {
   return mediastream_signaling_->remote_streams();
 }
 
+// TODO(deadbeef): Create RtpSenders immediately here, even if local
+// description hasn't yet been set.

[pthatcher1, 2015/09/23 14:47:39]

More like "Replace with addTrack, which may create RtpSenders and RtpReceivers
immediately.".

[pthatcher1, 2015/09/24 06:32:29]

Can you update this comment?

 bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
   if (IsClosed()) {
     return false;
   }
   if (!CanAddLocalMediaStream(mediastream_signaling_->local_streams(),
                               local_stream))
     return false;
 
   if (!mediastream_signaling_->AddLocalStream(local_stream)) {
     return false;
@@ skipping 24 matching lines @@
 
   rtc::scoped_refptr<DtmfSenderInterface> sender(
       DtmfSender::Create(track, signaling_thread(), session_.get()));
   if (!sender.get()) {
     LOG(LS_ERROR) << "CreateDtmfSender failed on DtmfSender::Create.";
     return NULL;
   }
   return DtmfSenderProxy::Create(signaling_thread(), sender.get());
 }
 
+RtpSenderRefptrs PeerConnection::GetSenders() const {
+  RtpSenderRefptrs senders;
+  for (const auto& pair : rtp_sender_pairs_) {
+    senders.push_back(pair.second);
+  }
+  return senders;
+}
+
+RtpReceiverRefptrs PeerConnection::GetReceivers() const {
+  RtpReceiverRefptrs receivers;
+  for (const auto& pair : rtp_receiver_pairs_) {
+    receivers.push_back(pair.second);
+  }
+  return receivers;
+}
+
 bool PeerConnection::GetStats(StatsObserver* observer,
                               MediaStreamTrackInterface* track,
                               StatsOutputLevel level) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!VERIFY(observer != NULL)) {
     LOG(LS_ERROR) << "GetStats - observer is NULL.";
     return false;
   }
 
   stats_->UpdateStats(level);
@@ skipping 317 matching lines @@
       break;
   }
 }
 
 void PeerConnection::OnAddRemoteStream(MediaStreamInterface* stream) {
   stats_->AddStream(stream);
   observer_->OnAddStream(stream);
 }
 
 void PeerConnection::OnRemoveRemoteStream(MediaStreamInterface* stream) {
-  stream_handler_container_->RemoveRemoteStream(stream);
   observer_->OnRemoveStream(stream);
 }
 
 void PeerConnection::OnAddDataChannel(DataChannelInterface* data_channel) {
   observer_->OnDataChannel(DataChannelProxy::Create(signaling_thread(),
                                                     data_channel));
 }
 
 void PeerConnection::OnAddRemoteAudioTrack(MediaStreamInterface* stream,
                                            AudioTrackInterface* audio_track,
                                            uint32 ssrc) {
-  stream_handler_container_->AddRemoteAudioTrack(stream, audio_track, ssrc);
+  AddRtpReceiver(new AudioRtpReceiver(audio_track, ssrc, session_.get()));
 }
 
 void PeerConnection::OnAddRemoteVideoTrack(MediaStreamInterface* stream,
                                            VideoTrackInterface* video_track,
                                            uint32 ssrc) {
-  stream_handler_container_->AddRemoteVideoTrack(stream, video_track, ssrc);
+  AddRtpReceiver(new VideoRtpReceiver(video_track, ssrc, session_.get()));
 }
 
+// TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
+// description.

[pthatcher1, 2015/09/23 14:47:39]

Well, in unified plan m-lines can never go away.  So this TODO is really
"Implement Unified Plan, in which case RtpReceivers will never be removed."

 void PeerConnection::OnRemoveRemoteAudioTrack(
     MediaStreamInterface* stream,
     AudioTrackInterface* audio_track) {
-  stream_handler_container_->RemoveRemoteTrack(stream, audio_track);
+  auto it = FindRtpReceiverPairForTrack(audio_track);
+  if (it == rtp_receiver_pairs_.end()) {
+    LOG(LS_WARNING) << "RtpReceiver for track with id " << audio_track->id()
+                    << " doesn't exist.";
+  } else {
+    it->first->DetachFromProvider();
+    rtp_receiver_pairs_.erase(it);
+  }
 }
 
 void PeerConnection::OnRemoveRemoteVideoTrack(
     MediaStreamInterface* stream,
     VideoTrackInterface* video_track) {
-  stream_handler_container_->RemoveRemoteTrack(stream, video_track);
+  auto it = FindRtpReceiverPairForTrack(video_track);
+  if (it == rtp_receiver_pairs_.end()) {
+    LOG(LS_WARNING) << "RtpReceiver for track with id " << video_track->id()
+                    << " doesn't exist.";
+  } else {
+    it->first->DetachFromProvider();
+    rtp_receiver_pairs_.erase(it);
+  }
 }
+
 void PeerConnection::OnAddLocalAudioTrack(MediaStreamInterface* stream,
                                           AudioTrackInterface* audio_track,
                                           uint32 ssrc) {
-  stream_handler_container_->AddLocalAudioTrack(stream, audio_track, ssrc);
+  AddRtpSender(new AudioRtpSender(audio_track, ssrc, 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);
+  AddRtpSender(new VideoRtpSender(video_track, ssrc, session_.get()));
 }
 
+// TODO(deadbeef): Keep RtpSenders around even if track goes away in local
+// description.
 void PeerConnection::OnRemoveLocalAudioTrack(MediaStreamInterface* stream,
                                              AudioTrackInterface* audio_track,
                                              uint32 ssrc) {
-  stream_handler_container_->RemoveLocalTrack(stream, audio_track);
+  auto it = FindRtpSenderPairForTrack(audio_track);
+  if (it == rtp_sender_pairs_.end()) {
+    LOG(LS_WARNING) << "RtpSender for track with id " << audio_track->id()
+                    << " doesn't exist.";
+    return;
+  } else {
+    it->first->DetachFromProvider();
+    rtp_sender_pairs_.erase(it);
+  }
   stats_->RemoveLocalAudioTrack(audio_track, ssrc);
 }
 
 void PeerConnection::OnRemoveLocalVideoTrack(MediaStreamInterface* stream,
                                              VideoTrackInterface* video_track) {
-  stream_handler_container_->RemoveLocalTrack(stream, video_track);
+  auto it = FindRtpSenderPairForTrack(video_track);
+  if (it == rtp_sender_pairs_.end()) {
+    LOG(LS_WARNING) << "RtpSender for track with id " << video_track->id()
+                    << " doesn't exist.";
+    return;
+  } else {
+    it->first->DetachFromProvider();
+    rtp_sender_pairs_.erase(it);
+  }
 }
 
 void PeerConnection::OnRemoveLocalStream(MediaStreamInterface* stream) {
-  stream_handler_container_->RemoveLocalStream(stream);
 }
 
 void PeerConnection::OnIceConnectionChange(
     PeerConnectionInterface::IceConnectionState new_state) {
   ASSERT(signaling_thread()->IsCurrent());
   ice_connection_state_ = new_state;
   observer_->OnIceConnectionChange(ice_connection_state_);
 }
 
 void PeerConnection::OnIceGatheringChange(
@@ skipping 29 matching lines @@
     observer_->OnIceConnectionChange(ice_connection_state_);
     if (ice_gathering_state_ != kIceGatheringComplete) {
       ice_gathering_state_ = kIceGatheringComplete;
       observer_->OnIceGatheringChange(ice_gathering_state_);
     }
   }
   observer_->OnSignalingChange(signaling_state_);
   observer_->OnStateChange(PeerConnectionObserver::kSignalingState);
 }
 
+void PeerConnection::AddRtpSender(BaseRtpSender* sender) {
+  rtp_sender_pairs_.push_back(RtpSenderPair(
+      sender, RtpSenderProxy::Create(signaling_thread(), sender)));
+}
+
+void PeerConnection::AddRtpReceiver(BaseRtpReceiver* receiver) {
+  rtp_receiver_pairs_.push_back(RtpReceiverPair(
+      receiver, RtpReceiverProxy::Create(signaling_thread(), receiver)));
+}
+
+PeerConnection::RtpSenderPairList::iterator
+PeerConnection::FindRtpSenderPairForTrack(MediaStreamTrackInterface* track) {
+  return std::find_if(rtp_sender_pairs_.begin(), rtp_sender_pairs_.end(),
+                      [track](const RtpSenderPair& pair) {
+                        return pair.first->track() == track;
+                      });
+}
+
+PeerConnection::RtpReceiverPairList::iterator
+PeerConnection::FindRtpReceiverPairForTrack(MediaStreamTrackInterface* track) {
+  return std::find_if(rtp_receiver_pairs_.begin(), rtp_receiver_pairs_.end(),
+                      [track](const RtpReceiverPair& pair) {
+                        return pair.first->track() == track;
+                      });
+}
+
 }  // namespace webrtc