Log - bf22be902e8968aa98d7be572266f876561728b7 - platform/external/webrtc

bf22be9 Roll chromium_revision 2b5aa49038..d74a300097 (434640:434704) by buildbot · 8 years ago
fd87f4a Opus: Move complexity variable out of conditional build flag by henrik.lundin · 8 years ago
1bc3146 Disable more VideoProcessorIntegrationTest tests on Linux 32-bit by Henrik Kjellander · 8 years ago
bb58435 Fix potential synchronization issues with framelisteners in EglRenderer. by sakal · 8 years ago
266f0a4 Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment. by philipel · 8 years ago
d1aaaaa Remove surface size mismatch logic from EglRenderer. by sakal · 8 years ago
6287e82 Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ ) by ossu · 8 years ago
7703b27 Disable PeerConnectionEndToEndTest.CallWithLegacySdp on Asan bots. by philipel · 8 years ago
ceecea4 Pass selected cricket::VideoCodec down to internal H264 encoder by magjed · 8 years ago
20dce34 Fixed bug in PacketBuffer to correctly detect new complete frames after ClearTo has been called. by philipel · 8 years ago
e1a13f8 MB: Remove a --target-devices-file flag for JUnit tests on android. by ehmaldonado · 8 years ago
a8eb756 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
9abbf5a Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
ffc6118 Don't cache video codec list in VideoEngine2. by brandtr · 8 years ago
ec1a670 Only create |remote_rate| when needed in RemoteBitrateEstimatorSingleStream. by Rasmus Brandt · 8 years ago
fb4a37a Add memcheck suppressions. by ehmaldonado · 8 years ago
26fa6b2 Revert of Bug in ExtractFrame API (extracts frames incorrectly) (patchset #9 id:130001 of https://codereview.webrtc.org/2529923002/ ) by charujain · 8 years ago
566cba1 Roll chromium_revision 5c22c2afac..2b5aa49038 (434448:434640) by buildbot · 8 years ago
b7636b4 Fixed bug in ExtractFrameFromY4mFile API which was not extracting the frames correctly. by charujain · 8 years ago
2f58ec8 Add I420Buffer::Copy method taking plane pointers as input. by nisse · 8 years ago
e441bdb Cleanup RtpSender hiding RtpHeaderExtensionLength function. by danilchap · 8 years ago
2fedf9c Smooth BWE and pass it to Audio Network Adaptor. by michaelt · 8 years ago
847f689 Roll chromium_revision 5e821a778b..5c22c2afac (432715:434448) by kjellander · 8 years ago
deb95f3 Change rtc::TimeNanos and rtc::TimeMicros return value from uint64_t to int64_t. by nisse · 8 years ago
71b9b58 Revert of Move ADM specific Android files into modules/audio_device/android/ (patchset #2 id:20001 of https://codereview.webrtc.org/2533573002/ ) by solenberg · 8 years ago
e8d8a2b Move ADM specific Android files into modules/audio_device/android/ by solenberg · 8 years ago
e69a1a9 Reland of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:1 of https://codereview.webrtc.org/2529143002/ ) by magjed · 8 years ago
d7e6ccb Revert of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:40001 of https://codereview.webrtc.org/2525693003/ ) by magjed · 8 years ago
c7805db Fix perf regression in screenshare temporal layer bitrate allocation by sprang · 8 years ago
fd34d30 iOS HW encoder: Enable H264 High profile support by magjed · 8 years ago
bdbc4b7 Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload by magjed · 8 years ago
1da1a09 Android HW encoder: Set constrained baseline as the profile by magjed · 8 years ago
03d6b08 Get rid of webrtc/base/latebindingsymboltable* by ehmaldonado · 8 years ago
f3feeff Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #1 id:1 of https://codereview.webrtc.org/2528993002/ ) by magjed · 8 years ago
0fa164a Make Valgrind memcheck work in swarming. by ehmaldonado · 8 years ago
5732910 Revert of CQ: Disable android_more_configs trybot (patchset #1 id:1 of https://codereview.webrtc.org/2522953003/ ) by ehmaldonado · 8 years ago
76622ce Adding a unit test for RMSLevel by henrik.lundin · 8 years ago
293bc2a Add 'Update LASTCHANGE' hook to DEPS by ehmaldonado · 8 years ago
5f7226f Turn off error resilience for vp8 for no temporal layers if nack is enabled. by asapersson · 8 years ago
5dfac56 Keep all codec parameters in VideoReceiveStream::Decoder by magjed · 8 years ago
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