Introduce RtpStreamReceiver and RtpStreamReceiverControllerInterface.

webrtc/voice_engine/channel_proxy.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_VOICE_ENGINE_CHANNEL_PROXY_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
 
 #include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/race_checker.h"
 #include "webrtc/base/thread_checker.h"
+#include "webrtc/call/rtp_packet_sink_interface.h"
 #include "webrtc/voice_engine/channel_manager.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 
 #include <memory>
 #include <string>
 #include <vector>
 
 namespace webrtc {
 
 class AudioSinkInterface;
@@ skipping 13 matching lines @@
 
 class Channel;
 
 // This class provides the "view" of a voe::Channel that we need to implement
 // webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two
 // purposes:
 //  1. Allow mocking just the interfaces used, instead of the entire
 //     voe::Channel class.
 //  2. Provide a refined interface for the stream classes, including assumptions
 //     on return values and input adaptation.
-class ChannelProxy {
+class ChannelProxy : public RtpPacketSinkInterface {
  public:
   ChannelProxy();
   explicit ChannelProxy(const ChannelOwner& channel_owner);
   virtual ~ChannelProxy();
 
   virtual bool SetEncoder(int payload_type,
                           std::unique_ptr<AudioEncoder> encoder);
   virtual void ModifyEncoder(
       rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier);
 
@@ skipping 23 matching lines @@
                                                 int payload_frequency);
   virtual bool SendTelephoneEventOutband(int event, int duration_ms);
   virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);
   virtual void SetRecPayloadType(int payload_type,
                                  const SdpAudioFormat& format);
   virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
   virtual void SetInputMute(bool muted);
   virtual void RegisterExternalTransport(Transport* transport);
   virtual void DeRegisterExternalTransport();
-  virtual void OnRtpPacket(const RtpPacketReceived& packet);
+
+  // Implements RtpPacketSinkInterface
+  void OnRtpPacket(const RtpPacketReceived& packet) override;
   virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
   virtual const rtc::scoped_refptr<AudioDecoderFactory>&
       GetAudioDecoderFactory() const;
   virtual void SetChannelOutputVolumeScaling(float scaling);
   virtual void SetRtcEventLog(RtcEventLog* event_log);
   virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
       int sample_rate_hz,
       AudioFrame* audio_frame);
   virtual int NeededFrequency() const;
   virtual void SetTransportOverhead(int transport_overhead_per_packet);
@@ skipping 27 matching lines @@
   rtc::RaceChecker audio_thread_race_checker_;
   rtc::RaceChecker video_capture_thread_race_checker_;
   ChannelOwner channel_owner_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);
 };
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_