Base A/V synchronization on sync_labels.

webrtc/video/bitrate_estimator_tests.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.
  */
 #include <functional>
@@ skipping 186 matching lines @@
           test_->encoder_config_.streams[0].width,
           test_->encoder_config_.streams[0].height,
           30,
           Clock::GetRealTimeClock()));
       send_stream_->Start();
       frame_generator_capturer_->Start();
 
       if (receive_audio) {
         AudioReceiveStream::Config receive_config;
         receive_config.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0];
+        // Bogus non-default id, every receive stream should correspond to an
+        // underlying channel.

[hta-webrtc, 2015/07/06 19:04:00]

This comment is not clear. Under what conditions is the voe_channel_id bogus,
and is zero always a bogus value?

Possible rewording:

"We set voe_channel_id to 0, which is bogus but legal. In production, the
voe_channel_id is always set to the id of an underlying channel."

[pbos-webrtc, 2015/07/09 10:21:49]

Rephrased as:

// Bogus non-default id to prevent hitting a DCHECK when creating the
// AudioReceiveStream. Every receive stream has to correspond to an  
// underlying channel id.                                            

+        receive_config.voe_channel_id = 0;
         receive_config.rtp.extensions.push_back(
             RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
         audio_receive_stream_ = test_->receiver_call_->CreateAudioReceiveStream(
             receive_config);
       } else {
         VideoReceiveStream::Decoder decoder;
         decoder.decoder = &fake_decoder_;
         decoder.payload_type =
             test_->send_config_.encoder_settings.payload_type;
         decoder.payload_name =
@@ skipping 153 matching lines @@
   receiver_trace_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   receiver_trace_.PushExpectedLogLine(
       "RemoteBitrateEstimatorFactory: Instantiating.");
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
 }
 }  // namespace webrtc