Stop using VoEVideoSync in Call/VideoReceiveStream.

webrtc/audio/audio_receive_stream.h

Index: webrtc/audio/audio_receive_stream.h
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index 7dfc5d6bb86ca723cba13a6c92bc7c6a34c3648e..a2074c1af56e76c068c35442d5793998ab4957e8 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -18,13 +18,13 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/audio_receive_stream.h"
-#include "webrtc/call/audio_state.h"
+#include "webrtc/call/syncable.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
 namespace webrtc {
+class PacketRouter;
 class RemoteBitrateEstimator;
 class RtcEventLog;
-class PacketRouter;
 
 namespace voe {
 class ChannelProxy;
@@ -34,7 +34,8 @@ namespace internal {
 class AudioSendStream;
 
 class AudioReceiveStream final : public webrtc::AudioReceiveStream,
-                                 public AudioMixer::Source {
+                                 public AudioMixer::Source,
+                                 public Syncable {
  public:
   AudioReceiveStream(PacketRouter* packet_router,
                      RemoteBitrateEstimator* remote_bitrate_estimator,
@@ -50,6 +51,18 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
   void SetGain(float gain) override;
 
+  // AudioMixer::Source
+  AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
+                                       AudioFrame* audio_frame) override;
+  int Ssrc() const override;
+  int PreferredSampleRate() const override;
+
+  // Syncable
+  int id() const override;
+  rtc::Optional<Syncable::Info> GetInfo() const override;
+  uint32_t GetPlayoutTimestamp() const override;
+  void SetMinimumPlayoutDelay(int delay_ms) override;
+
   void AssociateSendStream(AudioSendStream* send_stream);
   void SignalNetworkState(NetworkState state);
   bool DeliverRtcp(const uint8_t* packet, size_t length);
@@ -58,12 +71,6 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
                   const PacketTime& packet_time);
   const webrtc::AudioReceiveStream::Config& config() const;
 
-  // AudioMixer::Source
-  AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
-                                       AudioFrame* audio_frame) override;
-  int PreferredSampleRate() const override;
-  int Ssrc() const override;
-
  private:
   VoiceEngine* voice_engine() const;
   AudioState* audio_state() const;