Passed AudioMixer to AudioState::Config.

webrtc/video/video_quality_test.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.
  */
 #include "webrtc/video/video_quality_test.h"
 
 #include <stdio.h>
 #include <algorithm>
 #include <deque>
 #include <map>
 #include <sstream>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/include/cpu_info.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/layer_filtering_transport.h"
 #include "webrtc/test/run_loop.h"
 #include "webrtc/test/statistics.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/vcm_capturer.h"
 #include "webrtc/test/video_renderer.h"
@@ skipping 1295 matching lines @@
   // match the full stack tests.
   webrtc::RtcEventLogNullImpl event_log;
   Call::Config call_config(&event_log_);
   call_config.bitrate_config = params_.call.call_bitrate_config;
 
   ::VoiceEngineState voe;
   if (params_.audio.enabled) {
     CreateVoiceEngine(&voe, decoder_factory_);
     AudioState::Config audio_state_config;
     audio_state_config.voice_engine = voe.voice_engine;
+    audio_state_config.audio_mixer = AudioMixerImpl::Create();
     call_config.audio_state = AudioState::Create(audio_state_config);
   }
 
   std::unique_ptr<Call> call(Call::Create(call_config));
 
   // TODO(minyue): consider if this is a good transport even for audio only
   // calls.
   test::LayerFilteringTransport transport(
       params.pipe, call.get(), kPayloadTypeVP8, kPayloadTypeVP9,
       params.video.selected_tl, params_.ss.selected_sl);
@@ skipping 126 matching lines @@
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 
 }  // namespace webrtc