Introduce RtpStreamReceiver and RtpStreamReceiverControllerInterface.

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 30 matching lines @@
   audio_decode_stats.decoded_plc = 123;
   audio_decode_stats.decoded_cng = 456;
   audio_decode_stats.decoded_plc_cng = 789;
   audio_decode_stats.decoded_muted_output = 987;
   return audio_decode_stats;
 }
 
 const int kChannelId = 2;
 const uint32_t kRemoteSsrc = 1234;
 const uint32_t kLocalSsrc = 5678;
+#if 0
 const size_t kOneByteExtensionHeaderLength = 4;
 const size_t kOneByteExtensionLength = 4;
+#endif
 const int kAudioLevelId = 3;
 const int kTransportSequenceNumberId = 4;
 const int kJitterBufferDelay = -7;
 const int kPlayoutBufferDelay = 302;
 const unsigned int kSpeechOutputLevel = 99;
 const CallStatistics kCallStats = {
     345,  678,  901, 234, -12, 3456, 7890, 567, 890, 123};
 const CodecInst kCodecInst = {
     123, "codec_name_recv", 96000, -187, 0, -103};
 const NetworkStatistics kNetworkStats = {
@@ skipping 98 matching lines @@
   PacketRouter packet_router_;
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
   MockRtcEventLog event_log_;
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   rtc::scoped_refptr<MockAudioMixer> audio_mixer_;
   AudioReceiveStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
 };
 
+#if 0
 void BuildOneByteExtension(std::vector<uint8_t>::iterator it,
                            int id,
                            uint32_t extension_value,
                            size_t value_length) {
   const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
   ByteWriter<uint16_t>::WriteBigEndian(&(*it), kRtpOneByteHeaderExtensionId);
   it += 2;
 
   ByteWriter<uint16_t>::WriteBigEndian(&(*it), kOneByteExtensionLength / 4);
   it += 2;
@@ skipping 30 matching lines @@
   std::vector<uint8_t> packet;
   const size_t kRtcpSrLength = 28;  // In bytes.
   packet.resize(kRtcpSrLength);
   packet[0] = 0x80;  // Version 2.
   packet[1] = 0xc8;  // PT = 200, SR.
   // Length in number of 32-bit words - 1.
   ByteWriter<uint16_t>::WriteBigEndian(&packet[2], 6);
   ByteWriter<uint32_t>::WriteBigEndian(&packet[4], kLocalSsrc);
   return packet;
 }
+#endif
 }  // namespace
 
 TEST(AudioReceiveStreamTest, ConfigToString) {
   AudioReceiveStream::Config config;
   config.rtp.remote_ssrc = kRemoteSsrc;
   config.rtp.local_ssrc = kLocalSsrc;
   config.voe_channel_id = kChannelId;
   config.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
   EXPECT_EQ(
       "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, nack: "
       "{rtp_history_ms: 0}, extensions: [{uri: "
       "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 3}]}, "
       "rtcp_send_transport: null, voe_channel_id: 2}",
       config.ToString());
 }
 
+#if 0
 TEST(AudioReceiveStreamTest, ConstructDestruct) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
       helper.packet_router(),
       helper.config(), helper.audio_state(), helper.event_log());
 }
 
 TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
   ConfigHelper helper;
   helper.config().rtp.transport_cc = true;
@@ skipping 102 matching lines @@
       helper.packet_router(),
       helper.config(), helper.audio_state(), helper.event_log());
 
   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();
 }
+#endif
 }  // namespace test
 }  // namespace webrtc