Initial implementation of RtpTransportControllerReceive and related interfaces.

webrtc/audio/audio_receive_stream_unittest.cc

 /*
  *  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.
  */
 
@@ skipping 236 matching lines @@
       kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
   PacketTime packet_time(5678000, 0);
 
   RtpPacketReceived parsed_packet;
   ASSERT_TRUE(parsed_packet.Parse(&rtp_packet[0], rtp_packet.size()));
   parsed_packet.set_arrival_time_ms((packet_time.timestamp + 500) / 1000);
 
   EXPECT_CALL(*helper.channel_proxy(),
               OnRtpPacket(testing::Ref(parsed_packet)));
 
-  recv_stream.OnRtpPacket(parsed_packet);
+  recv_stream.OnRtpPacketReceive(&parsed_packet);
 }
 
 TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) {
   ConfigHelper helper;
   helper.config().rtp.transport_cc = true;
   internal::AudioReceiveStream recv_stream(
       helper.packet_router(),
       helper.config(), helper.audio_state(), helper.event_log());
 
   std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport();
@@ skipping 78 matching lines @@
 
   EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(0));
   EXPECT_CALL(helper.voice_engine(), StopPlayout(_));
   EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream))
       .WillOnce(Return(true));
 
   recv_stream.Start();
 }
 }  // namespace test
 }  // namespace webrtc