| Index: webrtc/call/call.cc
|
| diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
|
| index 3969bc6b517e97c2f51bcd35a64af8cc9a40654b..77741d1158a93fc2808e277aa34a143cc3fe7da5 100644
|
| --- a/webrtc/call/call.cc
|
| +++ b/webrtc/call/call.cc
|
| @@ -203,7 +203,8 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
|
| const webrtc::AudioSendStream::Config& config) {
|
| TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
|
| RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
|
| - AudioSendStream* send_stream = new AudioSendStream(config);
|
| + AudioSendStream* send_stream =
|
| + new AudioSendStream(config, config_.voice_engine);
|
| {
|
| rtc::CritScope lock(&network_enabled_crit_);
|
| WriteLockScoped write_lock(*send_crit_);
|
|
|