webrtc/media/engine/webrtcvoiceengine.cc - Issue 1891163002: Fix bug causing audio to stop being sent when AudioSendStreams are recreated.

Keyboard Shortcuts

	File
u :	up to issue
j / k :	jump to file after / before current file
J / K :	jump to next file with a comment after / before current file
	Side-by-side diff
i :	toggle intra-line diffs
e :	expand all comments
c :	collapse all comments
s :	toggle showing all comments
n / p :	next / previous diff chunk or comment
N / P :	next / previous comment
<Up> / <Down> :	next / previous line

	Issue
u :	up to list of issues
j / k :	jump to patch after / before current patch
o / <Enter> :	open current patch in side-by-side view
i :	open current patch in unified diff view

	Issue List
j / k :	jump to issue after / before current issue
o / <Enter> :	open current issue

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

Issue 1891163002: Fix bug causing audio to stop being sent when AudioSendStreams are recreated. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@51

Patch Set: Created 4 years, 8 months ago

Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.

Jump to:

View side-by-side diff with in-line comments

Download patch

Index: webrtc/media/engine/webrtcvoiceengine.cc

diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc

index 21094fde4d45af8baed7c5228237a5caba363e04..b899470b970c60f163d17d14079076f66c80c2ca 100644

--- a/webrtc/media/engine/webrtcvoiceengine.cc

+++ b/webrtc/media/engine/webrtcvoiceengine.cc

@@ -1116,6 +1116,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream

RTC_DCHECK(!stream_);

stream_ = call_->CreateAudioSendStream(config_);

RTC_CHECK(stream_);

+ UpdateSendState();

}

bool SendTelephoneEvent(int payload_type, int event, int duration_ms) {

« no previous file with comments | « no previous file | webrtc/media/engine/webrtcvoiceengine_unittest.cc » ('j') | no next file with comments »