Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
2ae140ae7e34bd1e102ca832a14b7b0adbb72d15
/
pc
/
remoteaudiosource.h
d367921
Configure media flow correctly with Unified Plan
by Steve Anton
· 7 years ago
6077675
Change RtpReceivers to interact with the media channel directly
by Steve Anton
· 7 years ago
3b80aac
Fix flaky memory leak in RemoteAudioSource
by Steve Anton
· 7 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago
[Renamed from webrtc/pc/remoteaudiosource.h]
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 7 years ago
a80c16a
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 7 years ago
c3771cc
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 7 years ago
7bb87ee
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
by ossu
· 8 years ago
[Renamed (94%) from webrtc/api/remoteaudiosource.h]
a69d973
Move webrtc/audio_*.h to webrtc/api/call
by kjellander
· 8 years ago
ba29c6a
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
by Taylor Brandstetter
· 8 years ago
3784b4a
Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ )
by tkchin
· 8 years ago
2d54917
Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
by Taylor Brandstetter
· 8 years ago
1a7162d
Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ )
by deadbeef
· 8 years ago
bc58319
Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
by Taylor Brandstetter
· 8 years ago
1a018dc
Prevent a voice channel from sending data before a source is set.
by Taylor Brandstetter
· 8 years ago
7ffeab5
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
by kjellander@webrtc.org
· 9 years ago
7324eb9
Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )
by kjellander
· 9 years ago
99b345c
Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.
by kjellander@webrtc.org
· 9 years ago
b24317b
Fix license headers in webrtc/api.
by kjellander
· 9 years ago
15583c1
Move talk/app/webrtc to webrtc/api
by Henrik Kjellander
· 9 years ago
[Renamed (93%) from talk/app/webrtc/remoteaudiosource.h]
a96e2d7
Move talk/media to webrtc/media
by kjellander
· 9 years ago
6eca7e3
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
by tommi
· 9 years ago
f888bb5
Support for unmixed remote audio into tracks.
by Tommi
· 9 years ago
14665ff
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
by kjellander@webrtc.org
· 10 years ago
5f93d0a
Update libjingle license statements at top of talk files for consistency
by jlmiller@webrtc.org
· 10 years ago
d4e598d
(Auto)update libjingle 72097588-> 72159069
by buildbot@webrtc.org
· 10 years ago
b9a088b
Update talk to 61538839.
by wu@webrtc.org
· 11 years ago
0de2950
Revert 5545 "Update libjingle to 61514460"
by wu@webrtc.org
· 11 years ago
e749c9e
Update libjingle to 61514460
by xians@webrtc.org
· 11 years ago