Replace AudioConferenceMixer with AudioMixer.

webrtc/call/call_perf_tests.cc

 /*
  *  Copyright (c) 2013 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 254 matching lines @@
   EXPECT_EQ(1u, video_receive_streams_.size());
   observer.set_receive_stream(video_receive_streams_[0]);
   DriftingClock drifting_clock(clock_, video_ntp_speed);
   CreateFrameGeneratorCapturerWithDrift(&drifting_clock, video_rtp_speed,
                                         kDefaultFramerate, kDefaultWidth,
                                         kDefaultHeight);
 
   Start();
 
   fake_audio_device.Start();
-  EXPECT_EQ(0, voe_base->StartPlayout(recv_channel_id));
+  audio_receive_stream->Start();
   EXPECT_EQ(0, voe_base->StartReceive(recv_channel_id));
   EXPECT_EQ(0, voe_base->StartSend(send_channel_id));
 
   EXPECT_TRUE(observer.Wait())
       << "Timed out while waiting for audio and video to be synchronized.";
 
   EXPECT_EQ(0, voe_base->StopSend(send_channel_id));
   EXPECT_EQ(0, voe_base->StopReceive(recv_channel_id));
   EXPECT_EQ(0, voe_base->StopPlayout(recv_channel_id));

[the sun, 2016/10/27 10:06:46]

audio_receive_stream->Stop();

   fake_audio_device.Stop();
 
   Stop();
   video_send_transport.StopSending();
   audio_send_transport.StopSending();
   receive_transport.StopSending();
 
   DestroyStreams();
 
   sender_call_->DestroyAudioSendStream(audio_send_stream);
@@ skipping 434 matching lines @@
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     test::FrameGeneratorCapturer* frame_generator_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc