Base A/V synchronization on sync_labels.

webrtc/video/bitrate_estimator_tests.cc

Index: webrtc/video/bitrate_estimator_tests.cc
diff --git a/webrtc/video/bitrate_estimator_tests.cc b/webrtc/video/bitrate_estimator_tests.cc
index 6e71ca24efe7d8b9269b96235af8df03ce6fcfb9..d7b0b374d35c12eb0733485d79c2a08b3cc00eb2 100644
--- a/webrtc/video/bitrate_estimator_tests.cc
+++ b/webrtc/video/bitrate_estimator_tests.cc
@@ -204,6 +204,9 @@ class BitrateEstimatorTest : public test::CallTest {
       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(