| Index: webrtc/test/call_test.cc
|
| diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
|
| index be437430b62f5c6243a3e5b9e4142b4a35291a4c..4f83b6d06ba6d5eeeee6687d50a4ba6c3d1c6167 100644
|
| --- a/webrtc/test/call_test.cc
|
| +++ b/webrtc/test/call_test.cc
|
| @@ -10,6 +10,7 @@
|
| #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"
|
| @@ -46,6 +47,7 @@ void CallTest::RunBaseTest(BaseTest* test) {
|
| 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);
|
| @@ -54,6 +56,7 @@ void CallTest::RunBaseTest(BaseTest* test) {
|
| 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);
|
|
|