Replace AudioReceiveStream::DeliverRtp with OnRtpPacket.

webrtc/test/mock_voe_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_TEST_MOCK_VOE_CHANNEL_PROXY_H_
 #define WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
 
 #include <string>
 
 #include "webrtc/test/gmock.h"
 #include "webrtc/voice_engine/channel_proxy.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
 
 namespace webrtc {
 namespace test {
 
 class MockVoEChannelProxy : public voe::ChannelProxy {
  public:
   MOCK_METHOD1(SetRTCPStatus, void(bool enable));
   MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
   MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
   MOCK_METHOD2(SetNACKStatus, void(bool enable, int max_packets));
@@ skipping 17 matching lines @@
   MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
   MOCK_METHOD2(SetSendTelephoneEventPayloadType, bool(int payload_type,
                                                       int payload_frequency));
   MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
   MOCK_METHOD2(SetBitrate, void(int bitrate_bps, int64_t probing_interval_ms));
   // TODO(solenberg): Talk the compiler into accepting this mock method:
   // MOCK_METHOD1(SetSink, void(std::unique_ptr<AudioSinkInterface> sink));
   MOCK_METHOD1(SetInputMute, void(bool muted));
   MOCK_METHOD1(RegisterExternalTransport, void(Transport* transport));
   MOCK_METHOD0(DeRegisterExternalTransport, void());
-  MOCK_METHOD3(ReceivedRTPPacket, bool(const uint8_t* packet,
-                                       size_t length,
-                                       const PacketTime& packet_time));
+  MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived& packet));
   MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
   MOCK_CONST_METHOD0(GetAudioDecoderFactory,
                      const rtc::scoped_refptr<AudioDecoderFactory>&());
   MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
   MOCK_METHOD1(SetRtcEventLog, void(RtcEventLog* event_log));
   MOCK_METHOD1(SetRtcpRttStats, void(RtcpRttStats* rtcp_rtt_stats));
   MOCK_METHOD1(EnableAudioNetworkAdaptor,
                void(const std::string& config_string));
   MOCK_METHOD0(DisableAudioNetworkAdaptor, void());
   MOCK_METHOD2(SetReceiverFrameLengthRange,
@@ skipping 19 matching lines @@
   MOCK_METHOD1(SetOpusDtx, bool(bool enable));
   MOCK_METHOD1(SetOpusMaxPlaybackRate, bool(int frequency_hz));
   MOCK_METHOD1(SetSendCodec, bool(const CodecInst& codec_inst));
   MOCK_METHOD2(SetSendCNPayloadType,
                bool(int type, PayloadFrequencies frequency));
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_