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

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 76 matching lines @@
           EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1);
           EXPECT_CALL(*channel_proxy_,
               SetReceiveAbsoluteSenderTimeStatus(true, kAbsSendTimeId))
                   .Times(1);
           EXPECT_CALL(*channel_proxy_,
               SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
                   .Times(1);
           EXPECT_CALL(*channel_proxy_, EnableReceiveTransportSequenceNumber(
                                            kTransportSequenceNumberId))
               .Times(1);
-          EXPECT_CALL(*channel_proxy_, SetCongestionControlObjects(
-                                           nullptr, nullptr, &packet_router_))
+          EXPECT_CALL(*channel_proxy_,
+                      RegisterReceiverCongestionControlObjects(&packet_router_))
               .Times(1);
           EXPECT_CALL(congestion_controller_, packet_router())
               .WillOnce(Return(&packet_router_));
-          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.local_ssrc = kLocalSsrc;
     stream_config_.rtp.remote_ssrc = kRemoteSsrc;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
@@ skipping 209 matching lines @@
   EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq);
   EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc);
   EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng);
   EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_,
             stats.capture_start_ntp_time_ms);
 }
 }  // namespace test
 }  // namespace webrtc