Chromium Code Reviews| Index: webrtc/media/engine/webrtcvoiceengine.cc |
| diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc |
| index 96f9ed70d403c6c464cc9b015f83190630a0b331..b95cb6f2f298912c8a463d21310902fef54a154e 100644 |
| --- a/webrtc/media/engine/webrtcvoiceengine.cc |
| +++ b/webrtc/media/engine/webrtcvoiceengine.cc |
| @@ -280,7 +280,10 @@ void GetOpusConfig(const AudioCodec& codec, |
| webrtc::AudioState::Config MakeAudioStateConfig(VoEWrapper* voe_wrapper) { |
| webrtc::AudioState::Config config; |
| config.voice_engine = voe_wrapper->engine(); |
| - config.audio_mixer = webrtc::AudioMixerImpl::Create(); |
| + if (voe_wrapper->AudioMixer()) |
|
aleloi
2016/12/05 14:03:08
Please use brackets for one-line blocks consistent
GeorgeZ
2016/12/05 17:56:18
Done.
|
| + config.audio_mixer = voe_wrapper->AudioMixer(); |
| + else |
| + config.audio_mixer = webrtc::AudioMixerImpl::Create(); |
| return config; |
| } |
| @@ -540,14 +543,19 @@ bool WebRtcVoiceEngine::ToCodecInst(const AudioCodec& in, |
| WebRtcVoiceEngine::WebRtcVoiceEngine( |
| webrtc::AudioDeviceModule* adm, |
| - const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory) |
| - : WebRtcVoiceEngine(adm, decoder_factory, new VoEWrapper()) { |
| + const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, |
| + rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) |
| + : WebRtcVoiceEngine(adm, |
| + decoder_factory, |
| + audio_mixer, |
| + new VoEWrapper(audio_mixer)) { |
| audio_state_ = webrtc::AudioState::Create(MakeAudioStateConfig(voe())); |
| } |
| WebRtcVoiceEngine::WebRtcVoiceEngine( |
| webrtc::AudioDeviceModule* adm, |
| const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, |
| + rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, |
| VoEWrapper* voe_wrapper) |
| : adm_(adm), decoder_factory_(decoder_factory), voe_wrapper_(voe_wrapper) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |