Always call RemoteBitrateEstimator::IncomingPacket from 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.
  */
 
@@ skipping 52 matching lines @@
   return ss.str();
 }
 
 namespace internal {
 AudioReceiveStream::AudioReceiveStream(
     PacketRouter* packet_router,
     RemoteBitrateEstimator* remote_bitrate_estimator,
     const webrtc::AudioReceiveStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     webrtc::RtcEventLog* event_log)
     : remote_bitrate_estimator_(remote_bitrate_estimator),

[brandtr, 2017/01/30 12:03:53]

Can |remote_bitrate_estimator_| be completely removed from this class?

[nisse-webrtc, 2017/01/30 14:56:57]

I plan that for the next cl. There's a call to RemoveStream which also needs to
be moved over to Call.

       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(packet_router);
   RTC_DCHECK(remote_bitrate_estimator);
   RTC_DCHECK(rtp_header_parser_);
 
@@ skipping 179 matching lines @@
                                     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 (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);
 }
 
 AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
     int sample_rate_hz,
     AudioFrame* audio_frame) {
   return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
 }
 
 int AudioReceiveStream::PreferredSampleRate() const {
@@ skipping 20 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