Use separate rtp module lists for send and receive in PacketRouter.

webrtc/audio/audio_send_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 65 matching lines @@
           EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
           EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
           EXPECT_CALL(*channel_proxy_,
               SetSendAbsoluteSenderTimeStatus(true, kAbsSendTimeId)).Times(1);
           EXPECT_CALL(*channel_proxy_,
               SetSendAudioLevelIndicationStatus(true, kAudioLevelId)).Times(1);
           EXPECT_CALL(*channel_proxy_, EnableSendTransportSequenceNumber(
                                            kTransportSequenceNumberId))
               .Times(1);
           EXPECT_CALL(*channel_proxy_,
-                      SetCongestionControlObjects(
+                      RegisterSenderCongestionControlObjects(
                           congestion_controller_.pacer(),
                           congestion_controller_.GetTransportFeedbackObserver(),
                           congestion_controller_.packet_router()))
               .Times(1);
-          EXPECT_CALL(*channel_proxy_,
-                      SetCongestionControlObjects(nullptr, nullptr, nullptr))
+          EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects())
               .Times(1);
           return channel_proxy_;
         }));
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.ssrc = kSsrc;
     stream_config_.rtp.c_name = kCName;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
@@ skipping 133 matching lines @@
       static_cast<internal::AudioState*>(helper.audio_state().get());
   VoiceEngineObserver* voe_observer =
       static_cast<VoiceEngineObserver*>(internal_audio_state);
   voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
   EXPECT_TRUE(send_stream.GetStats().typing_noise_detected);
   voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
   EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
 }
 }  // namespace test
 }  // namespace webrtc