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_); |