Index: webrtc/call/call.cc |
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc |
index cdb4f5d1a612ab3f8e1f6019e0d1e97abf573efa..eda209a01e76d826e3f75783101d4e33478153a1 100644 |
--- a/webrtc/call/call.cc |
+++ b/webrtc/call/call.cc |
@@ -145,7 +145,6 @@ Call::Call(const Call::Config& config) |
network_enabled_(true), |
receive_crit_(RWLockWrapper::CreateRWLock()), |
send_crit_(RWLockWrapper::CreateRWLock()) { |
- RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); |
RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0); |
RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps, |
config.bitrate_config.min_bitrate_bps); |
@@ -199,7 +198,8 @@ 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); |
if (!network_enabled_) |
send_stream->SignalNetworkState(kNetworkDown); |
{ |