Passed AudioMixer to AudioState::Config.

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.
  */
 
 #include <algorithm>
 #include <limits>
 #include <memory>
 #include <string>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/metrics_default.h"
 #include "webrtc/system_wrappers/include/rtp_to_ntp.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/drifting_clock.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_audio_device.h"
 #include "webrtc/test/fake_decoder.h"
@@ skipping 119 matching lines @@
   FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_filename,
                                     audio_rtp_speed);
   EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_));
   VoEBase::ChannelConfig config;
   config.enable_voice_pacing = true;
   int send_channel_id = voe_base->CreateChannel(config);
   int recv_channel_id = voe_base->CreateChannel();
 
   AudioState::Config send_audio_state_config;
   send_audio_state_config.voice_engine = voice_engine;
+  send_audio_state_config.audio_mixer = AudioMixerImpl::Create();
   Call::Config sender_config(&event_log_);
   sender_config.audio_state = AudioState::Create(send_audio_state_config);
   Call::Config receiver_config(&event_log_);
   receiver_config.audio_state = sender_config.audio_state;
   CreateCalls(sender_config, receiver_config);
 
 
   VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock());
 
   // Helper class to ensure we deliver correct media_type to the receiving call.
@@ skipping 85 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->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->StopPlayout(recv_channel_id));
   fake_audio_device.Stop();
 
   Stop();
@@ skipping 448 matching lines @@
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     test::FrameGeneratorCapturer* frame_generator_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc