Add send transports to individual webrtc::Call streams.

webrtc/video/call.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 105 matching lines @@
   const rtc::scoped_ptr<ProcessThread> module_process_thread_;
   const rtc::scoped_ptr<ChannelGroup> channel_group_;
   volatile int next_channel_id_;
   Call::Config config_;
 
   // Needs to be held while write-locking |receive_crit_| or |send_crit_|. This
   // ensures that we have a consistent network state signalled to all senders
   // and receivers.
   rtc::CriticalSection network_enabled_crit_;
   bool network_enabled_ GUARDED_BY(network_enabled_crit_);
-  TransportAdapter transport_adapter_;
 
   rtc::scoped_ptr<RWLockWrapper> receive_crit_;
   std::map<uint32_t, AudioReceiveStream*> audio_receive_ssrcs_
       GUARDED_BY(receive_crit_);
   std::map<uint32_t, VideoReceiveStream*> video_receive_ssrcs_
       GUARDED_BY(receive_crit_);
   std::set<VideoReceiveStream*> video_receive_streams_
       GUARDED_BY(receive_crit_);
   std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
       GUARDED_BY(receive_crit_);
@@ skipping 16 matching lines @@
 
 namespace internal {
 
 Call::Call(const Call::Config& config)
     : num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
       module_process_thread_(ProcessThread::Create()),
       channel_group_(new ChannelGroup(module_process_thread_.get())),
       next_channel_id_(0),
       config_(config),
       network_enabled_(true),
-      transport_adapter_(nullptr),
       receive_crit_(RWLockWrapper::CreateRWLock()),
       send_crit_(RWLockWrapper::CreateRWLock()) {
-  DCHECK(config.send_transport != nullptr);
-
   DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   DCHECK_GE(config.bitrate_config.start_bitrate_bps,
             config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     DCHECK_GE(config.bitrate_config.max_bitrate_bps,
               config.bitrate_config.start_bitrate_bps);
   }
 
   Trace::CreateTrace();
   module_process_thread_->Start();
@@ skipping 68 matching lines @@
 webrtc::VideoSendStream* Call::CreateVideoSendStream(
     const webrtc::VideoSendStream::Config& config,
     const VideoEncoderConfig& encoder_config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
   LOG(LS_INFO) << "CreateVideoSendStream: " << config.ToString();
   DCHECK(!config.rtp.ssrcs.empty());
 
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
   VideoSendStream* send_stream = new VideoSendStream(
-      config_.send_transport, overuse_observer_proxy_.get(), num_cpu_cores_,
+      overuse_observer_proxy_.get(), num_cpu_cores_,
       module_process_thread_.get(), channel_group_.get(),
       rtc::AtomicOps::Increment(&next_channel_id_), config, encoder_config,
       suspended_video_send_ssrcs_);
 
   // This needs to be taken before send_crit_ as both locks need to be held
   // while changing network state.
   rtc::CritScope lock(&network_enabled_crit_);
   WriteLockScoped write_lock(*send_crit_);
   for (uint32_t ssrc : config.rtp.ssrcs) {
     DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
@@ skipping 39 matching lines @@
   delete send_stream_impl;
 }
 
 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
     const webrtc::VideoReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
   LOG(LS_INFO) << "CreateVideoReceiveStream: " << config.ToString();
   VideoReceiveStream* receive_stream = new VideoReceiveStream(
       num_cpu_cores_, channel_group_.get(),
       rtc::AtomicOps::Increment(&next_channel_id_), config,
-      config_.send_transport, config_.voice_engine);
+      config_.voice_engine);
 
   // This needs to be taken before receive_crit_ as both locks need to be held
   // while changing network state.
   rtc::CritScope lock(&network_enabled_crit_);
   WriteLockScoped write_lock(*receive_crit_);
   DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
       video_receive_ssrcs_.end());
   video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
   // TODO(pbos): Configure different RTX payloads per receive payload.
   VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it =
@@ skipping 213 matching lines @@
                                                    const uint8_t* packet,
                                                    size_t length) {
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length);
 }
 
 }  // namespace internal
 }  // namespace webrtc