Log - fae640003cf028477819309777ecd29a74c25eff - platform/external/webrtc

749f660 Enable SSRC 0 in MediaChannel methods by Saurav Das · 4 years, 8 months ago
428dcb2 Remove SetLatency/GetLatency from MediaSourceInterface API level by Ruslan Burakov · 5 years ago
493a650 Propagate base minimum delay from video jitter buffer to webrtc/api. by Ruslan Burakov · 5 years ago
7ea4605 Add latency to remote source api. by Ruslan Burakov · 5 years ago
10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from 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 · 8 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 · 8 years ago
99b345c Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 8 years ago
b24317b Fix license headers in webrtc/api. by kjellander · 8 years ago
15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 8 years ago[Renamed (93%) from talk/app/webrtc/remoteaudiosource.h]
a96e2d7 Move talk/media to webrtc/media by kjellander · 8 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 · 9 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 · 10 years ago
0de2950 Revert 5545 "Update libjingle to 61514460" by wu@webrtc.org · 10 years ago
e749c9e Update libjingle to 61514460 by xians@webrtc.org · 10 years ago