Revert of Adding the ability to create an RtpSender without a track.

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 41 matching lines @@
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/system_wrappers/interface/field_trial.h"
 
 namespace {
 
 using webrtc::DataChannel;
 using webrtc::MediaConstraintsInterface;
 using webrtc::MediaStreamInterface;
 using webrtc::PeerConnectionInterface;
-using webrtc::RtpSenderInterface;
 using webrtc::StreamCollection;
 using webrtc::StunConfigurations;
 using webrtc::TurnConfigurations;
 typedef webrtc::PortAllocatorFactoryInterface::StunConfiguration
     StunConfiguration;
 typedef webrtc::PortAllocatorFactoryInterface::TurnConfiguration
     TurnConfiguration;
 
 static const char kDefaultStreamLabel[] = "default";
 static const char kDefaultAudioTrackLabel[] = "defaulta0";
@@ skipping 286 matching lines @@
              : std::vector<cricket::StreamParams>();
 }
 
 bool IsValidOfferToReceiveMedia(int value) {
   typedef PeerConnectionInterface::RTCOfferAnswerOptions Options;
   return (value >= Options::kUndefined) &&
          (value <= Options::kMaxOfferToReceiveMedia);
 }
 
 // Add the stream and RTP data channel info to |session_options|.
-void AddSendStreams(
-    cricket::MediaSessionOptions* session_options,
-    const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
-    const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
-        rtp_data_channels) {
+void SetStreams(cricket::MediaSessionOptions* session_options,
+                rtc::scoped_refptr<StreamCollection> streams,
+                const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
+                    rtp_data_channels) {
   session_options->streams.clear();
-  for (const auto& sender : senders) {
-    session_options->AddSendStream(sender->media_type(), sender->id(),
-                                   sender->stream_id());
+  if (streams != nullptr) {
+    for (size_t i = 0; i < streams->count(); ++i) {
+      MediaStreamInterface* stream = streams->at(i);
+      // For each audio track in the stream, add it to the MediaSessionOptions.
+      for (const auto& track : stream->GetAudioTracks()) {
+        session_options->AddSendStream(cricket::MEDIA_TYPE_AUDIO, track->id(),
+                                       stream->label());
+      }
+      // For each video track in the stream, add it to the MediaSessionOptions.
+      for (const auto& track : stream->GetVideoTracks()) {
+        session_options->AddSendStream(cricket::MEDIA_TYPE_VIDEO, track->id(),
+                                       stream->label());
+      }
+    }
   }
 
   // Check for data channels.
   for (const auto& kv : rtp_data_channels) {
     const DataChannel* channel = kv.second;
     if (channel->state() == DataChannel::kConnecting ||
         channel->state() == DataChannel::kOpen) {
       // |streamid| and |sync_label| are both set to the DataChannel label
       // here so they can be signaled the same way as MediaStreams and Tracks.
       // For MediaStreams, the sync_label is the MediaStream label and the
@@ skipping 264 matching lines @@
 rtc::scoped_refptr<StreamCollectionInterface>
 PeerConnection::local_streams() {
   return local_streams_;
 }
 
 rtc::scoped_refptr<StreamCollectionInterface>
 PeerConnection::remote_streams() {
   return remote_streams_;
 }
 
+// TODO(deadbeef): Create RtpSenders immediately here, even if local
+// description hasn't yet been set.
 bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
   if (IsClosed()) {
     return false;
   }
   if (!CanAddLocalMediaStream(local_streams_, local_stream)) {
     return false;
   }
 
   local_streams_->AddStream(local_stream);
 
+  // Find tracks that have already been configured in SDP. This can occur if a
+  // local session description that contains the MSID of these tracks is set
+  // before AddLocalStream is called. It can also occur if the local session
+  // description is not changed and RemoveLocalStream  is called and later
+  // AddLocalStream is called again with the same stream.
   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());
+    const TrackInfo* track_info =
+        FindTrackInfo(local_audio_tracks_, local_stream->label(), track->id());
+    if (track_info) {
+      CreateAudioSender(local_stream, track.get(), track_info->ssrc);
     }
   }
   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());
+    const TrackInfo* track_info =
+        FindTrackInfo(local_video_tracks_, local_stream->label(), track->id());
+    if (track_info) {
+      CreateVideoSender(local_stream, track.get(), track_info->ssrc);
     }
   }
 
   stats_->AddStream(local_stream);
   observer_->OnRenegotiationNeeded();
   return true;
 }
 
 // TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
-// indefinitely, when we have unified plan SDP.
+// indefinitely.
 void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
   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;
+    const TrackInfo* track_info =
+        FindTrackInfo(local_audio_tracks_, local_stream->label(), track->id());
+    if (track_info) {
+      DestroyAudioSender(local_stream, track.get(), track_info->ssrc);
     }
-    (*sender)->Stop();
-    senders_.erase(sender);
   }
   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;
+    const TrackInfo* track_info =
+        FindTrackInfo(local_video_tracks_, local_stream->label(), track->id());
+    if (track_info) {
+      DestroyVideoSender(local_stream, track.get());
     }
-    (*sender)->Stop();
-    senders_.erase(sender);
   }
 
   local_streams_->RemoveStream(local_stream);
 
   if (IsClosed()) {
     return;
   }
   observer_->OnRenegotiationNeeded();
 }
 
@@ skipping 10 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());
 }
 
-rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
-    const std::string& kind) {
-  RtpSenderInterface* new_sender;
-  if (kind == MediaStreamTrackInterface::kAudioTrackKind) {
-    new_sender = new AudioRtpSender(session_.get(), stats_.get());
-  } else if (kind == MediaStreamTrackInterface::kVideoTrackKind) {
-    new_sender = new VideoRtpSender(session_.get());
-  } else {
-    LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
-    return rtc::scoped_refptr<RtpSenderInterface>();
-  }
-  senders_.push_back(new_sender);
-  return RtpSenderProxy::Create(signaling_thread(), new_sender);
-}
-
 std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
     const {
   std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders;
   for (const auto& sender : senders_) {
     senders.push_back(RtpSenderProxy::Create(signaling_thread(), sender.get()));
   }
   return senders;
 }
 
 std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
@@ skipping 496 matching lines @@
   auto it = FindReceiverForTrack(video_track);
   if (it == receivers_.end()) {
     LOG(LS_WARNING) << "RtpReceiver for track with id " << video_track->id()
                     << " doesn't exist.";
   } else {
     (*it)->Stop();
     receivers_.erase(it);
   }
 }
 
+void PeerConnection::CreateAudioSender(MediaStreamInterface* stream,
+                                       AudioTrackInterface* audio_track,
+                                       uint32_t ssrc) {
+  senders_.push_back(new AudioRtpSender(audio_track, ssrc, session_.get()));
+  stats_->AddLocalAudioTrack(audio_track, ssrc);
+}
+
+void PeerConnection::CreateVideoSender(MediaStreamInterface* stream,
+                                       VideoTrackInterface* video_track,
+                                       uint32_t ssrc) {
+  senders_.push_back(new VideoRtpSender(video_track, ssrc, session_.get()));
+}
+
+// TODO(deadbeef): Keep RtpSenders around even if track goes away in local
+// description.
+void PeerConnection::DestroyAudioSender(MediaStreamInterface* stream,
+                                        AudioTrackInterface* audio_track,
+                                        uint32_t ssrc) {
+  auto it = FindSenderForTrack(audio_track);
+  if (it == senders_.end()) {
+    LOG(LS_WARNING) << "RtpSender for track with id " << audio_track->id()
+                    << " doesn't exist.";
+    return;
+  } else {
+    (*it)->Stop();
+    senders_.erase(it);
+  }
+  stats_->RemoveLocalAudioTrack(audio_track, ssrc);
+}
+
+void PeerConnection::DestroyVideoSender(MediaStreamInterface* stream,
+                                        VideoTrackInterface* video_track) {
+  auto it = FindSenderForTrack(video_track);
+  if (it == senders_.end()) {
+    LOG(LS_WARNING) << "RtpSender for track with id " << video_track->id()
+                    << " doesn't exist.";
+    return;
+  } else {
+    (*it)->Stop();
+    senders_.erase(it);
+  }
+}
+
 void PeerConnection::OnIceConnectionChange(
     PeerConnectionInterface::IceConnectionState new_state) {
   RTC_DCHECK(signaling_thread()->IsCurrent());
   // After transitioning to "closed", ignore any additional states from
   // WebRtcSession (such as "disconnected").
   if (IsClosed()) {
     return;
   }
   ice_connection_state_ = new_state;
   observer_->OnIceConnectionChange(ice_connection_state_);
@@ skipping 55 matching lines @@
   signaling_thread()->Post(this, MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
 }
 
 bool PeerConnection::GetOptionsForOffer(
     const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
     cricket::MediaSessionOptions* session_options) {
   if (!ConvertRtcOptionsForOffer(rtc_options, session_options)) {
     return false;
   }
 
-  AddSendStreams(session_options, senders_, rtp_data_channels_);
+  SetStreams(session_options, local_streams_, rtp_data_channels_);
   // Offer to receive audio/video if the constraint is not set and there are
   // send streams, or we're currently receiving.
   if (rtc_options.offer_to_receive_audio == RTCOfferAnswerOptions::kUndefined) {
     session_options->recv_audio =
         session_options->HasSendMediaStream(cricket::MEDIA_TYPE_AUDIO) ||
         !remote_audio_tracks_.empty();
   }
   if (rtc_options.offer_to_receive_video == RTCOfferAnswerOptions::kUndefined) {
     session_options->recv_video =
         session_options->HasSendMediaStream(cricket::MEDIA_TYPE_VIDEO) ||
@@ skipping 12 matching lines @@
 
 bool PeerConnection::GetOptionsForAnswer(
     const MediaConstraintsInterface* constraints,
     cricket::MediaSessionOptions* session_options) {
   session_options->recv_audio = false;
   session_options->recv_video = false;
   if (!ParseConstraintsForAnswer(constraints, session_options)) {
     return false;
   }
 
-  AddSendStreams(session_options, senders_, rtp_data_channels_);
+  SetStreams(session_options, local_streams_, rtp_data_channels_);
   session_options->bundle_enabled =
       session_options->bundle_enabled &&
       (session_options->has_audio() || session_options->has_video() ||
        session_options->has_data());
 
   // RTP data channel is handled in MediaSessionOptions::AddStream. SCTP streams
   // are not signaled in the SDP so does not go through that path and must be
   // handled here.
   if (session_->data_channel_type() == cricket::DCT_SCTP) {
     session_options->data_channel_type = cricket::DCT_SCTP;
   }
   return true;
 }
 
 void PeerConnection::UpdateRemoteStreamsList(
     const cricket::StreamParamsVec& streams,
     cricket::MediaType media_type,
     StreamCollection* new_streams) {
   TrackInfos* current_tracks = GetRemoteTracks(media_type);
 
   // Find removed tracks. I.e., tracks where the track id or ssrc don't match
-  // the new StreamParam.
+  // the
+  // new StreamParam.
   auto track_it = current_tracks->begin();
   while (track_it != current_tracks->end()) {
     const TrackInfo& info = *track_it;
     const cricket::StreamParams* params =
         cricket::GetStreamBySsrc(streams, info.ssrc);
     if (!params || params->id != info.track_id) {
       OnRemoteTrackRemoved(info.stream_label, info.track_id, media_type);
       track_it = current_tracks->erase(track_it);
     } else {
       ++track_it;
@@ skipping 181 matching lines @@
       current_tracks->push_back(TrackInfo(stream_label, track_id, ssrc));
       OnLocalTrackSeen(stream_label, track_id, params.first_ssrc(), media_type);
     }
   }
 }
 
 void PeerConnection::OnLocalTrackSeen(const std::string& stream_label,
                                       const std::string& track_id,
                                       uint32_t ssrc,
                                       cricket::MediaType media_type) {
-  RtpSenderInterface* sender = FindSenderById(track_id);
-  if (!sender) {
-    LOG(LS_WARNING) << "An unknown RtpSender with id " << track_id
-                    << " has been configured in the local description.";
+  MediaStreamInterface* stream = local_streams_->find(stream_label);
+  if (!stream) {
+    LOG(LS_WARNING) << "An unknown local MediaStream with label "
+                    << stream_label << " has been configured.";
     return;
   }
 
-  if (sender->media_type() != media_type) {
-    LOG(LS_WARNING) << "An RtpSender has been configured in the local"
-                    << " description with an unexpected media type.";
-    return;
+  if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+    AudioTrackInterface* audio_track = stream->FindAudioTrack(track_id);
+    if (!audio_track) {
+      LOG(LS_WARNING) << "An unknown local AudioTrack with id , " << track_id
+                      << " has been configured.";
+      return;
+    }
+    CreateAudioSender(stream, audio_track, ssrc);
+  } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+    VideoTrackInterface* video_track = stream->FindVideoTrack(track_id);
+    if (!video_track) {
+      LOG(LS_WARNING) << "An unknown local VideoTrack with id , " << track_id
+                      << " has been configured.";
+      return;
+    }
+    CreateVideoSender(stream, video_track, ssrc);
+  } else {
+    RTC_DCHECK(false && "Invalid media type");
   }
-
-  sender->set_stream_id(stream_label);
-  sender->SetSsrc(ssrc);
 }
 
 void PeerConnection::OnLocalTrackRemoved(const std::string& stream_label,
                                          const std::string& track_id,
                                          uint32_t ssrc,
                                          cricket::MediaType media_type) {
-  RtpSenderInterface* sender = FindSenderById(track_id);
-  if (!sender) {
-    // This is the normal case. I.e., RemoveStream has been called and the
+  MediaStreamInterface* stream = local_streams_->find(stream_label);
+  if (!stream) {
+    // This is the normal case. I.e., RemoveLocalStream has been called and the
     // SessionDescriptions has been renegotiated.
     return;
   }
-
-  // A sender has been removed from the SessionDescription but it's still
-  // associated with the PeerConnection. This only occurs if the SDP doesn't
-  // match with the calls to CreateSender, AddStream and RemoveStream.
-  if (sender->media_type() != media_type) {
-    LOG(LS_WARNING) << "An RtpSender has been configured in the local"
-                    << " description with an unexpected media type.";
-    return;
+  // A track has been removed from the SessionDescription but the MediaStream
+  // is still associated with PeerConnection. This only occurs if the SDP
+  // doesn't match with the calls to AddLocalStream and RemoveLocalStream.
+  if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+    AudioTrackInterface* audio_track = stream->FindAudioTrack(track_id);
+    if (!audio_track) {
+      return;
+    }
+    DestroyAudioSender(stream, audio_track, ssrc);
+  } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+    VideoTrackInterface* video_track = stream->FindVideoTrack(track_id);
+    if (!video_track) {
+      return;
+    }
+    DestroyVideoSender(stream, video_track);
+  } else {
+    RTC_DCHECK(false && "Invalid media type.");
   }
-
-  sender->SetSsrc(0);
 }
 
 void PeerConnection::UpdateLocalRtpDataChannels(
     const cricket::StreamParamsVec& streams) {
   std::vector<std::string> existing_channels;
 
   // Find new and active data channels.
   for (const cricket::StreamParams& params : streams) {
     // |it->sync_label| is actually the data channel label. The reason is that
     // we use the same naming of data channels as we do for
@@ skipping 197 matching lines @@
       InternalCreateDataChannel(label, &config));
   if (!channel.get()) {
     LOG(LS_ERROR) << "Failed to create DataChannel from the OPEN message.";
     return;
   }
 
   observer_->OnDataChannel(
       DataChannelProxy::Create(signaling_thread(), channel));
 }
 
-RtpSenderInterface* PeerConnection::FindSenderById(const std::string& id) {
-  auto it =
-      std::find_if(senders_.begin(), senders_.end(),
-                   [id](const rtc::scoped_refptr<RtpSenderInterface>& sender) {
-                     return sender->id() == id;
-                   });
-  return it != senders_.end() ? it->get() : nullptr;
-}
-
 std::vector<rtc::scoped_refptr<RtpSenderInterface>>::iterator
 PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) {
   return std::find_if(
       senders_.begin(), senders_.end(),
       [track](const rtc::scoped_refptr<RtpSenderInterface>& sender) {
         return sender->track() == track;
       });
 }
 
 std::vector<rtc::scoped_refptr<RtpReceiverInterface>>::iterator
@@ skipping 37 matching lines @@
 DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
   for (const auto& channel : sctp_data_channels_) {
     if (channel->id() == sid) {
       return channel;
     }
   }
   return nullptr;
 }
 
 }  // namespace webrtc