Introduce RtpStreamReceiver and RtpStreamReceiverControllerInterface.

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 <vector>
 
 #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/rtp_packet_sink_interface.h"
 #include "webrtc/call/syncable.h"
 
 namespace webrtc {
 class PacketRouter;
 class RtcEventLog;
 class RtpPacketReceived;
+class RtpStreamReceiverControllerInterface;
+class RtpStreamReceiverInterface;
 
 namespace voe {
 class ChannelProxy;
 }  // namespace voe
 
 namespace internal {
 class AudioSendStream;
 
 class AudioReceiveStream final : public webrtc::AudioReceiveStream,
                                  public AudioMixer::Source,
-                                 public Syncable,
-                                 public RtpPacketSinkInterface {
+                                 public Syncable {
  public:
-  AudioReceiveStream(PacketRouter* packet_router,
+  AudioReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller,
+                     PacketRouter* packet_router,
                      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;
   int GetOutputLevel() const override;
   void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
   void SetGain(float gain) override;
   std::vector<webrtc::RtpSource> GetSources() const override;
 
-  // RtpPacketSinkInterface.
-  void OnRtpPacket(const RtpPacketReceived& packet) override;
+  // TODO(nisse): We don't formally implement RtpPacketSinkInterface, and this
+  // method shouldn't be needed. But it's currently used by the
+  // AudioReceiveStreamTest.ReceiveRtpPacket unittest. Figure out if that test
+  // shuld be refactored or deleted, and then delete this method.
+  void OnRtpPacket(const RtpPacketReceived& packet);
 
   // 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;
@@ skipping 11 matching lines @@
   int SetVoiceEnginePlayout(bool playout);
 
   rtc::ThreadChecker worker_thread_checker_;
   rtc::ThreadChecker module_process_thread_checker_;
   const webrtc::AudioReceiveStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   bool playing_ ACCESS_ON(worker_thread_checker_) = false;
 
+  std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_;
+
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_