Base A/V synchronization on sync_labels.

webrtc/audio_receive_stream.h

 /*
  *  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 25 matching lines @@
       // Synchronization source (stream identifier) to be received.
       uint32_t remote_ssrc = 0;
 
       // Sender SSRC used for sending RTCP (such as receiver reports).
       uint32_t local_ssrc = 0;
 
       // RTP header extensions used for the received stream.
       std::vector<RtpExtension> extensions;
     } rtp;
 
+    // Underlying VoiceEngine handle, used to map AudioReceiveStream to
+    // lower-level components. Temporarily used while VoiceEngine channels are
+    // created outside of Call.
+    int voe_channel_id = -1;
+
+    // Identifier for an A/V synchronization group. Empty string to disable.
+    // TODO(pbos): Synchronize streams in a sync group, not just one video
+    // stream to one audio stream. Tracked by issue webrtc:4762.
+    std::string sync_group;
+
     // Decoders for every payload that we can receive. Call owns the
     // AudioDecoder instances once the Config is submitted to
     // Call::CreateReceiveStream().
     // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
     std::map<uint8_t, AudioDecoder*> decoder_map;
   };
 
   virtual Stats GetStats() const = 0;
 
  protected:
   virtual ~AudioReceiveStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_RECEIVE_STREAM_H_