| Index: webrtc/media/engine/webrtcvoiceengine.cc
|
| diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
|
| index 21f12105f1db6745f8c70c6167e8210e023e305e..cef7cdf2becc2cfa786ee5e9e1414f9b752d148c 100644
|
| --- a/webrtc/media/engine/webrtcvoiceengine.cc
|
| +++ b/webrtc/media/engine/webrtcvoiceengine.cc
|
| @@ -1121,6 +1121,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
|
| RTC_DCHECK(!stream_);
|
| stream_ = call_->CreateAudioSendStream(config_);
|
| RTC_CHECK(stream_);
|
| + UpdateSendState();
|
| }
|
|
|
| bool SendTelephoneEvent(int payload_type, int event, int duration_ms) {
|
|
|