Base A/V synchronization on sync_labels.

webrtc/video/audio_receive_stream.cc

Index: webrtc/video/audio_receive_stream.cc
diff --git a/webrtc/video/audio_receive_stream.cc b/webrtc/video/audio_receive_stream.cc
index f5383f43196231985077960c6e4ad4d34fec428d..7884972bc590ee1b4b0720e623959e3ade497e1e 100644
--- a/webrtc/video/audio_receive_stream.cc
+++ b/webrtc/video/audio_receive_stream.cc
@@ -34,6 +34,9 @@ std::string AudioReceiveStream::Config::Rtp::ToString() const {
 std::string AudioReceiveStream::Config::ToString() const {
   std::stringstream ss;
   ss << "{rtp: " << rtp.ToString();
+  ss << ", channel_id: " << channel_id;
+  if (!sync_group.empty())
+    ss << ", sync_group: " << sync_group;
   ss << '}';
   return ss.str();
 }
@@ -45,6 +48,7 @@ AudioReceiveStream::AudioReceiveStream(
     : remote_bitrate_estimator_(remote_bitrate_estimator),
       config_(config),
       rtp_header_parser_(RtpHeaderParser::Create()) {
+  CHECK(config.channel_id != -1);

[the sun, 2015/06/11 11:52:03]

Why not DCHECK()? If the channel doesn't exist, subsequent calls to VoE will
fail anyway.

[pbos-webrtc, 2015/06/11 14:48:53]

Done.

   DCHECK(remote_bitrate_estimator_ != nullptr);
   DCHECK(rtp_header_parser_ != nullptr);
   for (const auto& ext : config.rtp.extensions) {