Move ownership of PacketRouter from CongestionController to Call.

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.
  */
 
 #include "webrtc/audio/audio_receive_stream.h"
 
 #include <string>
 #include <utility>
 
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/timeutils.h"
-#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_neteq_stats.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 #include "webrtc/voice_engine/include/voe_volume_control.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
-namespace {
-
-bool UseSendSideBwe(const webrtc::AudioReceiveStream::Config& config) {
-  if (!config.rtp.transport_cc) {
-    return false;
-  }
-  for (const auto& extension : config.rtp.extensions) {
-    if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
-      return true;
-    }
-  }
-  return false;
-}
-}  // namespace
 
 std::string AudioReceiveStream::Config::Rtp::ToString() const {
   std::stringstream ss;
   ss << "{remote_ssrc: " << remote_ssrc;
   ss << ", local_ssrc: " << local_ssrc;
   ss << ", transport_cc: " << (transport_cc ? "on" : "off");
   ss << ", nack: " << nack.ToString();
   ss << ", extensions: [";
   for (size_t i = 0; i < extensions.size(); ++i) {
     ss << extensions[i].ToString();
@@ skipping 14 matching lines @@
   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,
+    PacketRouter* packet_router,
+    RemoteBitrateEstimator* remote_bitrate_estimator,
     const webrtc::AudioReceiveStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     webrtc::RtcEventLog* event_log)
-    : config_(config),
+    : remote_bitrate_estimator_(remote_bitrate_estimator),
+      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(packet_router);
+  RTC_DCHECK(remote_bitrate_estimator);
   RTC_DCHECK(rtp_header_parser_);
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->SetRtcEventLog(event_log);
   channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
   // TODO(solenberg): Config NACK history window (which is a packet count),
   // using the actual packet size for the configured codec.
   channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0,
                                 config_.rtp.nack.rtp_history_ms / 20);
@@ skipping 18 matching lines @@
     } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
       channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id);
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionTransportSequenceNumber, extension.id);
       RTC_DCHECK(registered);
     } else {
       RTC_NOTREACHED() << "Unsupported RTP extension.";
     }
   }
   // Configure bandwidth estimation.
-  channel_proxy_->RegisterReceiverCongestionControlObjects(
-      congestion_controller->packet_router());
-  if (UseSendSideBwe(config)) {
-    remote_bitrate_estimator_ =
-        congestion_controller->GetRemoteBitrateEstimator(true);
-  }
+  channel_proxy_->RegisterReceiverCongestionControlObjects(packet_router);
 }
 
 AudioReceiveStream::~AudioReceiveStream() {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
   if (playing_) {
     Stop();
   }
   channel_proxy_->DisassociateSendChannel();
   channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
-  if (remote_bitrate_estimator_) {
-    remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
-  }
+  remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
 }
 
 void AudioReceiveStream::Start() {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   if (playing_) {
     return;
   }
 
   int error = SetVoiceEnginePlayout(true);
   if (error != 0) {
@@ skipping 118 matching lines @@
   // 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_ &&
+  if (config_.rtp.transport_cc &&
       header.extension.hasTransportSequenceNumber) {
     int64_t arrival_time_ms = rtc::TimeMillis();
     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);
   }
 
   return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
@@ skipping 29 matching lines @@
   ScopedVoEInterface<VoEBase> base(voice_engine());
   if (playout) {
     return base->StartPlayout(config_.voe_channel_id);
   } else {
     return base->StopPlayout(config_.voe_channel_id);
   }
 }
 
 }  // namespace internal
 }  // namespace webrtc