Stop using VoEVideoSync in Call/VideoReceiveStream.

webrtc/audio/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.
  */
 
 #ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
 
 #include <memory>
 
 #include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/audio/audio_state.h"
 #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;
 }  // namespace voe
 
 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,
                      const webrtc::AudioReceiveStream::Config& config,
                      const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
                      webrtc::RtcEventLog* event_log);
   ~AudioReceiveStream() override;
 
   // webrtc::AudioReceiveStream implementation.
   void Start() override;
   void Stop() override;
   webrtc::AudioReceiveStream::Stats GetStats() const override;
   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);
   bool DeliverRtp(const uint8_t* packet,
                   size_t length,
                   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;
   int SetVoiceEnginePlayout(bool playout);
 
   rtc::ThreadChecker thread_checker_;
   RemoteBitrateEstimator* const remote_bitrate_estimator_;
   const webrtc::AudioReceiveStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   bool playing_ ACCESS_ON(thread_checker_) = false;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_