Switch voice transport to use Call and Stream instead of VoENetwork.

webrtc/audio/audio_receive_stream.cc

 /*
  *  Copyright (c) 2015 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 48 matching lines @@
   }
   ss << ']';
   ss << ", transport_cc: " << (transport_cc ? "on" : "off");
   ss << '}';
   return ss.str();
 }
 
 std::string AudioReceiveStream::Config::ToString() const {
   std::stringstream ss;
   ss << "{rtp: " << rtp.ToString();
-  ss << ", receive_transport: "
-     << (receive_transport ? "(Transport)" : "nullptr");
   ss << ", rtcp_send_transport: "
      << (rtcp_send_transport ? "(Transport)" : "nullptr");
   ss << ", voe_channel_id: " << voe_channel_id;
   if (!sync_group.empty()) {
     ss << ", sync_group: " << sync_group;
   }
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
 AudioReceiveStream::AudioReceiveStream(
     CongestionController* congestion_controller,
     const webrtc::AudioReceiveStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state)
     : config_(config),
       audio_state_(audio_state),
       rtp_header_parser_(RtpHeaderParser::Create()) {
   LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
   RTC_DCHECK(congestion_controller);
   RTC_DCHECK(rtp_header_parser_);
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
+
+  channel_proxy_->RegisterExternalTransport(config.rtcp_send_transport);
+
   for (const auto& extension : config.rtp.extensions) {
     if (extension.name == RtpExtension::kAudioLevel) {
       channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id);
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAudioLevel, extension.id);
       RTC_DCHECK(registered);
     } else if (extension.name == RtpExtension::kAbsSendTime) {
       channel_proxy_->SetReceiveAbsoluteSenderTimeStatus(true, extension.id);
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAbsoluteSendTime, extension.id);
@@ skipping 12 matching lines @@
       congestion_controller->packet_router());
   if (UseSendSideBwe(config)) {
     remote_bitrate_estimator_ =
         congestion_controller->GetRemoteBitrateEstimator(true);
   }
 }
 
 AudioReceiveStream::~AudioReceiveStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
+  channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
   if (remote_bitrate_estimator_) {
     remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
   }
 }
 
 void AudioReceiveStream::Start() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 void AudioReceiveStream::Stop() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 void AudioReceiveStream::SignalNetworkState(NetworkState state) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   // TODO(solenberg): Tests call this function on a network thread, libjingle
   // calls on the worker thread. We should move towards always using a network
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
-  return false;
+  return channel_proxy_->ReceivedRTCPPacket(packet, length);
 }
 
 bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
                                     size_t length,
                                     const PacketTime& packet_time) {
   // TODO(solenberg): Tests call this function on a network thread, libjingle
   // calls on the worker thread. We should move towards always using a network
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
   RTPHeader header;
   if (!rtp_header_parser_->Parse(packet, length, &header)) {
     return false;
   }
 
   // Only forward if the parsed header has one of the headers necessary for
   // bandwidth estimation. RTP timestamps has different rates for audio and
   // video and shouldn't be mixed.
   if (remote_bitrate_estimator_ &&
       header.extension.hasTransportSequenceNumber) {
     int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
     if (packet_time.timestamp >= 0)
       arrival_time_ms = (packet_time.timestamp + 500) / 1000;
     size_t payload_size = length - header.headerLength;
     remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
                                               header, false);
   }
-  return true;
+
+  return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
 }
 
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   webrtc::AudioReceiveStream::Stats stats;
   stats.remote_ssrc = config_.rtp.remote_ssrc;
   ScopedVoEInterface<VoECodec> codec(voice_engine());
 
   webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
   webrtc::CodecInst codec_inst = {0};
@@ skipping 49 matching lines @@
 
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc