New class RtpDemuxer and RtpPacketSinkInterface, use in Call.

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 20ed4613a7ede16d1ad3aa8897cb957c8c282c34..d0b5a4d1cbf1464fb69ac5f916ad56a88b6fef19 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -19,6 +19,7 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/audio_receive_stream.h"
+#include "webrtc/call/rtp_demuxer.h"
 #include "webrtc/call/syncable.h"
 
 namespace webrtc {
@@ -35,7 +36,8 @@ class AudioSendStream;
 
 class AudioReceiveStream final : public webrtc::AudioReceiveStream,
                                  public AudioMixer::Source,
-                                 public Syncable {
+                                 public Syncable,
+                                 public RtpPacketSinkInterface {
  public:
   AudioReceiveStream(PacketRouter* packet_router,
                      const webrtc::AudioReceiveStream::Config& config,
@@ -52,8 +54,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   void SetGain(float gain) override;
   std::vector<webrtc::RtpSource> GetSources() const override;
 
-  // TODO(nisse): Intended to be part of an RtpPacketReceiver interface.
-  void OnRtpPacket(const RtpPacketReceived& packet);
+  // RtpPacketSinkInterface.
+  void OnRtpPacket(const RtpPacketReceived& packet) override;
 
   // AudioMixer::Source
   AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,