Initial implementation of RtpTransportControllerReceive and related interfaces.

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 adac88387613a43a7f6220567a75c8e881fe313c..f53a138298ce517ca25e7cbb201be8436777ecd4 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -18,7 +18,10 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/audio_receive_stream.h"
+#include "webrtc/call/rtp_transport_controller_receive.h"
 #include "webrtc/call/syncable.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
 
 namespace webrtc {
 class PacketRouter;
@@ -32,7 +35,8 @@ class ChannelProxy;
 namespace internal {
 class AudioSendStream;
 
-class AudioReceiveStream final : public webrtc::AudioReceiveStream,
+class AudioReceiveStream final : public webrtc::RtpPacketReceiverInterface,
+                                 public webrtc::AudioReceiveStream,
                                  public AudioMixer::Source,
                                  public Syncable {
  public:
@@ -50,8 +54,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
   void SetGain(float gain) override;
 
-  // TODO(nisse): Intended to be part of an RtpPacketReceiver interface.
-  void OnRtpPacket(const RtpPacketReceived& packet);
+  // webrtc::RtpPacketReceiverInterface implementation
+  bool OnRtpPacketReceive(RtpPacketReceived* packet) override;
 
   // AudioMixer::Source
   AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
@@ -78,6 +82,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   rtc::ThreadChecker worker_thread_checker_;
   rtc::ThreadChecker module_process_thread_checker_;
   const webrtc::AudioReceiveStream::Config config_;
+  RtpHeaderExtensionMap rtp_header_extensions_;
+
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;