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Unified Diff: webrtc/media/engine/webrtcvoiceengine.cc

Issue 1881793006: Fix bug causing audio to stop being sent when AudioSendStreams are recreated. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Better test case Created 4 years, 8 months ago
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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) {
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