Remove base channel for video receivers.

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 97 matching lines @@
 
   void SetBitrateControllerConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config);
 
   void ConfigureSync(const std::string& sync_group)
       EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
 
   const int num_cpu_cores_;
   const rtc::scoped_ptr<ProcessThread> module_process_thread_;
   const rtc::scoped_ptr<ChannelGroup> channel_group_;
-  const int base_channel_id_;
   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_;
 
@@ skipping 22 matching lines @@
 Call* Call::Create(const Call::Config& config) {
   return new internal::Call(config);
 }
 
 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())),
-      base_channel_id_(0),
-      next_channel_id_(base_channel_id_ + 1),
+      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();
 
-  // TODO(pbos): Remove base channel when CreateReceiveChannel no longer
-  // requires one.
-  CHECK(channel_group_->CreateSendChannel(base_channel_id_, 0,
-                                          &transport_adapter_, num_cpu_cores_,
-                                          std::vector<uint32_t>(), true));
-
   if (config.overuse_callback) {
     overuse_observer_proxy_.reset(
         new CpuOveruseObserverProxy(config.overuse_callback));
   }
 
   SetBitrateControllerConfig(config_.bitrate_config);
 }
 
 Call::~Call() {
   CHECK_EQ(0u, video_send_ssrcs_.size());
   CHECK_EQ(0u, video_send_streams_.size());
   CHECK_EQ(0u, audio_receive_ssrcs_.size());
   CHECK_EQ(0u, video_receive_ssrcs_.size());
   CHECK_EQ(0u, video_receive_streams_.size());
 
-  channel_group_->DeleteChannel(base_channel_id_);
   module_process_thread_->Stop();
   Trace::ReturnTrace();
 }
 
 PacketReceiver* Call::Receiver() { return this; }
 
 webrtc::AudioSendStream* Call::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   return nullptr;
 }
@@ skipping 100 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_, base_channel_id_, channel_group_.get(),
+      num_cpu_cores_, channel_group_.get(),
       rtc::AtomicOps::Increment(&next_channel_id_), config,
       config_.send_transport, 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;
@@ skipping 215 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