RtpPacketReceiver base class and OnRtpPacket, with a pre-parsed RTP packet.

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 0792e243b06871172259eb5ca049f336b1da6ab9..d46e167bdb1e915320d869ed219ee30c575a47d1 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -18,6 +18,7 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/audio_receive_stream.h"
+#include "webrtc/call/rtp_packet_receiver.h"
 #include "webrtc/call/syncable.h"
 
 namespace webrtc {
@@ -32,6 +33,7 @@ namespace internal {
 class AudioSendStream;
 
 class AudioReceiveStream final : public webrtc::AudioReceiveStream,
+                                 public webrtc::RtpPacketReceiver,
                                  public AudioMixer::Source,
                                  public Syncable {
  public:
@@ -48,6 +50,10 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
   void SetGain(float gain) override;
 
+  // Implements RtpPacketReceiver
+  bool OnRtpPacket(const RtpPacketReceived& packet) override;
+  const RtpConfig& rtp_config() const override;
+
   // AudioMixer::Source
   AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
                                        AudioFrame* audio_frame) override;
@@ -63,19 +69,16 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   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;
 
  private:
   VoiceEngine* voice_engine() const;
   AudioState* audio_state() const;
   int SetVoiceEnginePlayout(bool playout);
-
   rtc::ThreadChecker worker_thread_checker_;
   rtc::ThreadChecker module_process_thread_checker_;
   const webrtc::AudioReceiveStream::Config config_;
+  const RtpConfig rtp_config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;