webrtc/api/webrtcsession_unittest.cc - Issue 1775033002: Changed defaults for CreateAnswer in non-constraint mode

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/api/webrtcsession_unittest.cc

Issue 1775033002: Changed defaults for CreateAnswer in non-constraint mode (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: First landable version of patch Created 4 years, 9 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/api/webrtcsession_unittest.cc

diff --git a/webrtc/api/webrtcsession_unittest.cc b/webrtc/api/webrtcsession_unittest.cc

index 87ff927f7a990ff2ab9ba1584966181ecbab22c8..2c9a99b5e08a247cea0eaf567f30a9e65b464899 100644

--- a/webrtc/api/webrtcsession_unittest.cc

+++ b/webrtc/api/webrtcsession_unittest.cc

@@ -518,7 +518,7 @@ class WebRtcSessionTest

void GetOptionsForOffer(

const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,

cricket::MediaSessionOptions* session_options) {

- ASSERT_TRUE(ExtractMediaSessionOptions(rtc_options, session_options));

+ ASSERT_TRUE(ExtractMediaSessionOptions(rtc_options, true, session_options));

perkj_webrtc 2016/03/10 15:35:24 Hum. weird that ExtractMediaSessionOptions has bee

hta-webrtc 2016/03/10 16:22:05 Yep, I wondered if I could stop exporting it at al

AddStreamsToOptions(session_options);

if (rtc_options.offer_to_receive_audio ==

« webrtc/api/peerconnection_unittest.cc ('K') | « webrtc/api/peerconnectioninterface_unittest.cc ('k') | no next file » | no next file with comments »