Replace AudioConferenceMixer with AudioMixer.

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 101 matching lines @@
           EXPECT_CALL(*channel_proxy_, GetAudioDecoderFactory())
               .WillOnce(ReturnRef(decoder_factory_));
           testing::Expectation expect_set =
               EXPECT_CALL(*channel_proxy_, SetRtcEventLog(&event_log_))
                   .Times(1);
           EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
               .Times(1)
               .After(expect_set);
           return channel_proxy_;
         }));
-    EXPECT_CALL(voice_engine_, StopPlayout(kChannelId)).WillOnce(Return(0));

[ossu, 2016/10/25 14:13:33]

Why won't StopPlayout be called after this change?

[the sun, 2016/10/27 10:06:45]

Because it is called from the dtor, but now the call will only happen if the
stream has actually been started.

[aleloi, 2016/11/01 15:17:35]

Thanks, sorry for being late!

     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.local_ssrc = kLocalSsrc;
     stream_config_.rtp.remote_ssrc = kRemoteSsrc;
     stream_config_.rtp.nack.rtp_history_ms = 300;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
     stream_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
@@ skipping 234 matching lines @@
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
       helper.congestion_controller(), helper.config(), helper.audio_state(),
       helper.event_log());
   EXPECT_CALL(*helper.channel_proxy(),
       SetChannelOutputVolumeScaling(FloatEq(0.765f)));
   recv_stream.SetGain(0.765f);
 }
 }  // namespace test
 }  // namespace webrtc