Log - 13d38fbe904cd99acebf77068e93df34725eeb47 - platform/external/webrtc

a6a699a Sent bitrate stats are incorrect if FlexFEC is configured: by asapersson · 8 years ago
6b272c5 RtpReceiver: Add RegisterReceivePayload function for VideoCodec by magjed · 8 years ago
5de9b6a Move helpers_ios.cc/.h by solenberg · 8 years ago
0928a3c Reland of Split out target rtc_media_base from rtc_media (patchset #1 id:1 of https://codereview.webrtc.org/2508163002/ ) by magjed · 8 years ago
33c81d0 Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #7 id:240001 of https://codereview.webrtc.org/2524923002/ ) by magjed · 8 years ago
69b627d Move smoothing filter to common audio and exp_filter to base/analytics. by minyue · 8 years ago
b881254 Remove RTPPayloadStrategy and simplify RTPPayloadRegistry by magjed · 8 years ago
56124bd Send audio and video codecs to RTPPayloadRegistry by magjed · 8 years ago
b7374db Fix parsing padding byte in rtp header extension by danilchap · 8 years ago
bf67663 Rename "Audio playout level" to "Audio level" on the Y-axis of the event log graph. by ivoc · 8 years ago
3c3aef4 Revert of Reland "Move smoothing filter to common audio". (patchset #5 id:100001 of https://codereview.webrtc.org/2520003005/ ) by minyue · 8 years ago
223641f Reland "Move smoothing filter to common audio". by minyue · 8 years ago
b365b80 Revert of Modify the paths of the resource files to point to chromium/src/tools/... (patchset #1 id:1 of https://codereview.webrtc.org/2528893002/ ) by ehmaldonado · 8 years ago
d8ae20b Modify the paths of the resource files to point to chromium/src/tools/... by ehmaldonado · 8 years ago
3cfb3ef Added a perf test for the residual echo detector. by ivoc · 8 years ago
37a2111 Increase the threshold for RunPlayoutAndRecordingInFullDuplex. Again. by ehmaldonado · 8 years ago
3edc7f0 AGC: Add a histogram for new level by henrik.lundin · 8 years ago
c42d376 DataChannelInterface: Remove default implementation of methods. by hbos · 8 years ago
464d50f Set rtc_use_memcheck=true for the FYI bot. by ehmaldonado · 8 years ago
ed8c8ed Add rtc_use_memcheck flag, update MB and GN to handle it, and add gni files listing the runtime deps by ehmaldonado · 8 years ago
d44d0ba For VPN network, use the underlying network type as its type. by honghaiz · 8 years ago
4dfb8ce Make the default value of rtcp-mux policy to required. by zhihuang · 8 years ago
e02407a Add myself to WATCHLIST for api/. by solenberg · 8 years ago
42eee12 RTCPeerConnectionStats: Removed fixed TODO comments. by hbos · 8 years ago
08be780 Reland of Allow custom metrics implementations on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2516403002/ ) by sakal · 8 years ago
817208b Re-enables AudioDeviceTest.StartStopPlayout on Android by henrika · 8 years ago
8b64628 Add fps reduction API to SurfaceViewRenderer. by sakal · 8 years ago
4fe3b8d Add framelistener functionality to SurfaceViewRenderer. by sakal · 8 years ago
1c82884 Remove binding framebuffer from GlTextureFrameBuffer.setSize. by sakal · 8 years ago
8e321c8 CQ: Disable android_more_configs trybot by Henrik Kjellander · 8 years ago
0c5a154 Try to deflake VideoSendStream tests with FlexFEC. by brandtr · 8 years ago
0adb828 RTCCodecStats[1] added. by hbos · 8 years ago
71caaca Split avfoundationcapturer classes in separate files. by denicija · 8 years ago
90ea736 Add DesktopFrame rotation functions by zijiehe · 8 years ago
e2b1501 Start probes only after network is connected. by Sergey Ulanov · 8 years ago
1c062bf Fix module/desktop_capture compilation on iOS by Sergey Ulanov · 8 years ago
c1dd1a5 Really disable Opus complexity tests on Android by henrik.lundin · 8 years ago
d661e9c WebRTC: Replace ProjectRootPath by ResourcePath by ehmaldonado · 8 years ago
10165ab Unify VideoCodecType to/from string functionality by magjed · 8 years ago
2d60e53 H264 encoder: Include QP information in encoded images by magjed · 8 years ago
e60f020 iOS AppRTCMobile: Fix SDP video codec reordering for multiple H264 profiles by magjed · 8 years ago
8271d04 This CL introduces the new functionality for setting by peah · 8 years ago
30a12fb AGC: Add a histogram for clipping adjustment by henrik.lundin · 8 years ago
24d812d DEPS: Specify WebRTC hooks and add a few dependencies by kjellander · 8 years ago
ab6996d Enable QP parsing from CABAC bitstreams by kthelgason · 8 years ago
04c0722 Replace AudioConferenceMixer with AudioMixer. by aleloi · 8 years ago
b426040 Add Full HD and 4K camera resolutions to AppRTCMobile Android. by sakal · 8 years ago
2df1ab4 MB: Add Win32 SyzyASan (swarming) config. by ehmaldonado · 8 years ago
17338d4 Created an AudioMixer mock in webrtc/api/test. by aleloi · 8 years ago
0eb1960 ComfortNoise: Calculate used scale factor in Q13 by ossu · 8 years ago
58f90a7 Disable Opus complexity tests on Android by henrik.lundin · 8 years ago
03d5fb1 Let MediaSession generate a FlexFEC SSRC when FlexFEC is active. by brandtr · 8 years ago
0dbb6f5 Fix the standard deviation calculation in the level controller perf tests. by ivoc · 8 years ago
820f578 RTCInboundRTPStreamStats's [fir/pli/nack]_count are collected for video. by hbos · 8 years ago
468da7c Wire up FlexFEC in VideoEngine2. by brandtr · 8 years ago
d848a56 DEPS: Cleanup extra_gyp_flag and extra_gitignore.py by kjellander · 8 years ago
875862c Let Opus increase complexity for low bitrates by henrik.lundin · 8 years ago
b1e6d5e Set surface view surface size to minimum of the layout size and frame size. by sakal · 8 years ago
f6acc2a Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/ by magjed · 8 years ago
0ce6aaf Move androidvideotracksource from api under api/android/jni. by sakal · 8 years ago
f723312 Add an empty libjingle_peerconnection_metrics_default_jni target. by sakal · 8 years ago
9688e38 Add support for FEC-FR semantics in StreamParams. by brandtr · 8 years ago
96385e0 iOS: Add FlexFEC-03 field trial. by brandtr · 8 years ago
fb94cd6 build_ios_libs.sh: Add command line bitcode option. by tkchin · 8 years ago
7a07f13 Fix TimeCallback used by BoringSSL. by deadbeef · 8 years ago
1b0e3aa Remove deprecated CroppingWindowCapturer::Create by zijiehe · 8 years ago
2874796 RTCStats operator== bugfix by hbos · 8 years ago
f570a28 Revert of Allow custom metrics implementations on Android. (patchset #11 id:260001 of https://codereview.webrtc.org/2403463002/ ) by philipel · 8 years ago
ab102f1 Update gtest-parallel and introduce gtest-parallel-wrapper. by ehmaldonado · 8 years ago
de609b2 Allow custom metrics implementations on Android. by sakal · 8 years ago
e718606 Make magjed@ owner of webrtc/api/android/ by magjed · 8 years ago
64d6ff7 In VoiceEngine, the settings for APM are applied in such a way that by peah · 8 years ago
40217c3 Initial rate allocation should not use fps = 0 by sprang · 8 years ago
57c1ad3 Don't declare function arguments of array type by kwiberg · 8 years ago
cc7bf88 Revert of Roll chromium_revision 5e821a778b..80ff2be807 (432715:433495) (patchset #1 id:1 of https://codereview.webrtc.org/2517933002/ ) by kjellander · 8 years ago
6280960 Correctly pass drawn frame size when layout aspect ratio is used in EglRenderer. by sakal · 8 years ago
96c1587 RtpPacket::payload() return rtc::ArrayView instead of raw pointer by danilchap · 8 years ago
fe09560 Roll chromium_revision 5e821a778b..80ff2be807 (432715:433495) by buildbot · 8 years ago
f880285 iOS: Cleanup buildbot JSON files + bump iOS version to 10.0 by kjellander · 8 years ago
3898944 Remove unused files linux.cc/.h and linuxfdwalk.c/.h. by solenberg · 8 years ago
2184155 Add more logging in ScreenCapturerIntegrationTest by zijiehe · 8 years ago
ed9dccf Revert of Remove unused HttpClient class. (patchset #1 id:1 of https://codereview.webrtc.org/2511883005/ ) by honghaiz · 8 years ago
4a698f6 Remove unused HttpClient class. by solenberg · 8 years ago
01af3a3 Remove unused dbus.cc/.h and related things. by solenberg · 8 years ago
90c024f Move FirewallSocketServer to test code. by nisse · 8 years ago
00f2ee0 Changed the way we find the ProjectRootPath. by ehmaldonado · 8 years ago
dedaf1c Modify audio_processing_unittest to use ResourcePath instead of ProjectRootPath. by ehmaldonado · 8 years ago
bbc747c Delete WindowPicker class and subclasses. by nisse · 8 years ago
76b3049 Changed the interface AudioMixer::RemoveSource to have a void return type. by aleloi · 8 years ago
a28780e Introduce ArrayView::subview function to return portion of the original view by danilchap · 8 years ago
509e4fe Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) by magjed · 8 years ago
d7ac0a9 Revert of Move smoothing filter to common audio. (patchset #3 id:60001 of https://codereview.webrtc.org/2484153002/ ) by magjed · 8 years ago
a82395b Move smoothing filter to common audio. by michaelt · 8 years ago
610c454 Add Datachannel support to Android AppRTCMobile by hekra01 · 8 years ago
1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
7b9feee Fix PayloadRouter::OnEncodedImage() to handle errors properly. by sergeyu · 8 years ago
81c3a03 Added a callback function OnAddTrack to PeerConnectionObserver by zhihuang · 8 years ago
5b93db2 iOS: Add AudioSendSideBwe field trial. by tkchin · 8 years ago
eacbaea Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) by magjed · 8 years ago
0d0d753 Revert of Split out target rtc_media_base from rtc_media (patchset #3 id:40001 of https://codereview.webrtc.org/2471573003/ ) by magjed · 8 years ago