Passed AudioMixer to AudioState::Config.

webrtc/test/call_test.cc

 /*
  *  Copyright (c) 2014 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/test/call_test.h"
 
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
+#include "webrtc/test/call_test.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 
 namespace webrtc {
 namespace test {
 
 namespace {
 const int kVideoRotationRtpExtensionId = 4;
 }
 
@@ skipping 17 matching lines @@
 void CallTest::RunBaseTest(BaseTest* test) {
   num_video_streams_ = test->GetNumVideoStreams();
   num_audio_streams_ = test->GetNumAudioStreams();
   num_flexfec_streams_ = test->GetNumFlexfecStreams();
   RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
   Call::Config send_config(test->GetSenderCallConfig());
   if (num_audio_streams_ > 0) {
     CreateVoiceEngines();
     AudioState::Config audio_state_config;
     audio_state_config.voice_engine = voe_send_.voice_engine;
+    audio_state_config.audio_mixer = AudioMixerImpl::Create();
     send_config.audio_state = AudioState::Create(audio_state_config);
   }
   CreateSenderCall(send_config);
   if (test->ShouldCreateReceivers()) {
     Call::Config recv_config(test->GetReceiverCallConfig());
     if (num_audio_streams_ > 0) {
       AudioState::Config audio_state_config;
       audio_state_config.voice_engine = voe_recv_.voice_engine;
+      audio_state_config.audio_mixer = AudioMixerImpl::Create();
       recv_config.audio_state = AudioState::Create(audio_state_config);
     }
     CreateReceiverCall(recv_config);
   }
   test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
   receive_transport_.reset(test->CreateReceiveTransport());
   send_transport_.reset(test->CreateSendTransport(sender_call_.get()));
 
   if (test->ShouldCreateReceivers()) {
     send_transport_->SetReceiver(receiver_call_->Receiver());
@@ skipping 426 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc