Fix bug causing audio to stop being sent when AudioSendStreams are recreated.
BUG=webrtc:5772
Review URL: https://codereview.webrtc.org/1881793006
Cr-Commit-Position: refs/heads/master@{#12347}
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 21f1210..cef7cdf 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1121,6 +1121,7 @@
RTC_DCHECK(!stream_);
stream_ = call_->CreateAudioSendStream(config_);
RTC_CHECK(stream_);
+ UpdateSendState();
}
bool SendTelephoneEvent(int payload_type, int event, int duration_ms) {