Replace AudioReceiveStream::DeliverRtp with OnRtpPacket.

webrtc/voice_engine/channel.h

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 35 matching lines @@
 class FileWrapper;
 class PacketRouter;
 class ProcessThread;
 class RateLimiter;
 class ReceiveStatistics;
 class RemoteNtpTimeEstimator;
 class RtcEventLog;
 class RTPPayloadRegistry;
 class RtpReceiver;
 class RTPReceiverAudio;
+class RtpPacketReceived;
 class RtpRtcp;
 class TelephoneEventHandler;
 class VoERTPObserver;
 class VoiceEngineObserver;
 
 struct CallStatistics;
 struct ReportBlock;
 struct SenderInfo;
 
 namespace voe {
@@ skipping 128 matching lines @@
   int SetOpusDtx(bool enable_dtx);
   int GetOpusDtx(bool* enabled);
   bool EnableAudioNetworkAdaptor(const std::string& config_string);
   void DisableAudioNetworkAdaptor();
   void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                    int max_frame_length_ms);
 
   // VoENetwork
   int32_t RegisterExternalTransport(Transport* transport);
   int32_t DeRegisterExternalTransport();
-  int32_t ReceivedRTPPacket(const uint8_t* received_packet,
-                            size_t length,
-                            const PacketTime& packet_time);
+  void OnRtpPacket(const RtpPacketReceived& packet);
   int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
 
   // VoEFile
   int StartPlayingFileLocally(const char* fileName,
                               bool loop,
                               FileFormats format,
                               int startPosition,
                               float volumeScaling,
                               int stopPosition,
                               const CodecInst* codecInst);
@@ skipping 303 matching lines @@
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_