Log - bc46bf22e776d2f1372c6e79e1f9b4bdb7edd17b - platform/external/webrtc

bc46bf2 common_audio: Explicit cast in WebRtcSpl_NormW16 on ARM by Bjorn Volcker · 10 years ago
0194d32 Add WebRtcAudioManager to peerconnection_jar library by Alex Glaznev · 10 years ago
65f74a1 Revert "Suppress data races in libjingle_peerconnection_unittest" by Tommi · 10 years ago
2c9c83d Remove non-functional asynchronous resampling mode. by Andrew MacDonald · 10 years ago
45c6449 Introduce CodecManager and move code from AudioCodingModuleImpl by Henrik Lundin · 10 years ago
f7b9cf5 Suppress "EndToEndTest::ReceivedFecPacketsNotNacked" on Asan, Tsan by Minyue Li · 10 years ago
842a4a6 Add locks to Start(), Stop() methods in ProcessThread. by Tommi · 10 years ago
22e209d Introduce AudioCodingModuleImpl::current_encoder_ by Henrik Lundin · 10 years ago
582f80e Clamp decoder sample rate to 32000 in iSAC by Henrik Lundin · 10 years ago
1ecfd55 videoadapter_unittest.cc: Revert removal of '#if defined(HAVE_WEBRTC_VIDEO)' by Magnus Jedvert · 10 years ago
451b614 Fix gyp path for bwe simulator include. by Stefan Holmer · 10 years ago
8e9c67e Suppress data races in libjingle_peerconnection_unittest by Henrik Kjellander · 10 years ago
9f52448 Roll chromium_revision 4d63ee8..719b839 (322012:322539) by Henrik Kjellander · 10 years ago
6b3ccfc GN: Cleanup no longer needed libvpx config. by Henrik Kjellander · 10 years ago
819011c Additional suppression for TSan deadlock detection by Henrik Kjellander · 10 years ago
dfd53fe Raise streams for SetMaxSendBitrates above 2000k. by Peter Boström · 10 years ago
53eda3d Add tests for r8811. by Peter Boström · 10 years ago
b3fc48b Update the notice about the slow Chromium sync. by Henrik Kjellander · 10 years ago
1d36003 Suppress TSan errors triggered when deadlock detection is enabled. by Henrik Kjellander · 10 years ago
9ff73f5 Final minor fix in WebRtcAudioManager by henrika · 10 years ago
424694c audio_processing/agc: Put entire method set_output_will_be_muted() under lock by Bjorn Volcker · 10 years ago
75a0255 Handle borked Android cameras gracefully. by Per · 10 years ago
8324b52 Adding playout volume control to WebRtcAudioTrack.java. by henrika · 10 years ago
8ed6a4b Remove unused non-standard capture stats. by Peter Boström · 10 years ago
3954e1d Remove unused implementations in cricket::VideoFrame by Magnus Jedvert · 10 years ago
7100dcd Adding "usedtx" as Opus codec parameter. by Minyue Li · 10 years ago
bef8d2d Add a lock to NSSContext to fix data race by Jiayang Liu · 10 years ago
b8cfa68 Update speed setting in VP9. by Marco · 10 years ago
74d9ed7 Report send codec name in GetStats(). by Peter Boström · 10 years ago
d6f4c25 Reject streams reusing simulcast or RTX SSRCs. by Peter Boström · 10 years ago
a990784 AcmReceiver: index decoders by payload type instead of ACM codec ID by Jelena Marusic · 10 years ago
9b5f96e Add some sanity CHECKs to webrtc::Call. by Peter Boström · 10 years ago
c79f7ed Fix build error introduced by r8864. by Stefan Holmer · 10 years ago
e590416 Moving the pacer and the pacer thread to ChannelGroup. by Stefan Holmer · 10 years ago
5225dd8 If audio ptime is negotiated in SDP, then we would set the audio codec with negotiated packet size if it's allowed. If the negotiated packet size is not supported by the working codec, then we would use the next smallest size. by Brave Yao · 10 years ago
dfa3605 Reparent Nonlinear beamformer under beamforming interface. by Michael Graczyk · 10 years ago
bf395c1 Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android by Bjorn Volcker · 10 years ago
caae5d4 Bye request should use POST not GET by Chuck Hays · 10 years ago
190c3ca Register sample rate of Audio RED in RTPPayloadRegistry. by Minyue Li · 10 years ago
79064e5 Fix crash on decode found by fuzz tester. by Stefan Holmer · 10 years ago
3fbf99c Refactor common_audio/vad: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT by Bjorn Volcker · 10 years ago
855acf7 Remove video from WebRTC Android example. by Per · 10 years ago
d4362cd Reject StreamParams with RTX SSRCs not in ssrcs. by Peter Boström · 10 years ago
a49f515 Roll chromium_revision da9a1c0..4d63ee8 (321718:322012) by Henrik Kjellander · 10 years ago
1ccd8b4 Refactor common_audio/signal_processing: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT by Bjorn Volcker · 10 years ago
245989b Address comments from cr 43769004. by Tommi · 10 years ago
0e209b0 Update bundle behavior to match BundlePolicy spec in http://rtcweb-wg.github.io/jsep/. by Donald Curtis · 10 years ago
e61c64d Delete NullVideoRenderer by Magnus Jedvert · 10 years ago
07a4ba5 Simulcast settings for 1080p. Using same bit rates for all 3 modes since only one is used in reality, and the plan is to unify them. by Niklas Enbom · 10 years ago
ac27e20 Delete VideoAdapter::AdaptFrame by Magnus Jedvert · 10 years ago
45636ec Post Git switch: Update codereview.settings and remove drover.properties by Henrik Kjellander · 10 years ago
68a5418 Enable PENDING_REF_PREFIX in codereview.settings. by Henrik Kjellander · 10 years ago
4d14592 rtc::Buffer: Restore length method for backwards compatibility by kwiberg@webrtc.org · 10 years ago
deafa7b Remove I420VideoFrame::SwapFrame by magjed@webrtc.org · 10 years ago
2d2a30c Remove I420VideoFrame::CloneFrame by magjed@webrtc.org · 10 years ago
0b52ceb Improve logging and add DCHECKs in codec database. by pbos@webrtc.org · 10 years ago
eebcab5 rtc::Buffer: Rename length to size, for conformance with the STL by kwiberg@webrtc.org · 10 years ago
e815290 Update README instructions for Android AppRTCDemo. by glaznev@webrtc.org · 10 years ago
a5f6fb5 Permit single-stream max bitrates above 2000k. by pbos@webrtc.org · 10 years ago
a197a5e Update libsrtp includes in preparation of roll into Chromium. by jiayl@webrtc.org · 10 years ago
a3ffc56 Allow setting thread priorities in Chromium on all but linux platforms. by tommi@webrtc.org · 10 years ago
39fc1d3 Disable PeerConnectionClientTest.testLoopbackVp9 by henrik.lundin@webrtc.org · 10 years ago
0b44b58 Limit disabling of PeerConnectionEndToEndTest.Call to Windows by henrik.lundin@webrtc.org · 10 years ago
64eb2ff iOS library build script by tkchin@webrtc.org · 10 years ago
9509fbf Split EventWrapper in twain. by tommi@webrtc.org · 10 years ago
82e8ae4 Disable PeerConnectionEndToEndTest.Call in libjingle_peerconnection_unittest by henrik.lundin@webrtc.org · 10 years ago
2b4ce3a Convert webrtc/video/ abort/assert to CHECK/DCHECK. by pbos@webrtc.org · 10 years ago
41d2bef Limit RED audio payload to narrow band. by minyue@webrtc.org · 10 years ago
1596a4f Temporarily disable SetPriority when building with Chromium. by tommi@webrtc.org · 10 years ago
d4e7d49 Scaler: Recycle allocations using buffer pool. by magjed@webrtc.org · 10 years ago
09b6ff9 Disable PLC for iSAC by henrik.lundin@webrtc.org · 10 years ago
ee0c5af Remove unused version.py script. by kjellander@webrtc.org · 10 years ago
aa0bbab Fix build failure by jmarusic@webrtc.org · 10 years ago
a4bef3e AcmReceiver: use std::map instead of an array to keep the list of decoders by jmarusic@webrtc.org · 10 years ago
3335a4f Prevent asserting on unset start bitrate. by pbos@webrtc.org · 10 years ago
50ed0d9 Roll chromium_revision 6311617..da9a1c0 (321517:321718) by kjellander@webrtc.org · 10 years ago
e5e92bd Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows (fix) by kjellander@webrtc.org · 10 years ago
cfde27e Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows. by kjellander@webrtc.org · 10 years ago
38492c5 Re-land 8810 "- Add a SetPriority method to ThreadWr..." by tommi@webrtc.org · 10 years ago
90a1cb4 Revert 8810 "- Add a SetPriority method to ThreadWrapper" by tommi@webrtc.org · 10 years ago
b789f62 Re-land 8809 "Set WebRtcVideoEngine2 as the WebRtcMe..." by tommi@webrtc.org · 10 years ago
0c34001 Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine." by tommi@webrtc.org · 10 years ago
346a64b Mac would force bluetooth playout working with 8kHz/1ch if capturing/rendering shares the same device, e.g. changing from 44.1kHz/2ch as default. by braveyao@webrtc.org · 10 years ago
4553941 Document the 'int' return value of Resampler methods. by wtc@chromium.org · 10 years ago
3200a64 Minor fix for MIPS Android build. by andrew@webrtc.org · 10 years ago
4ddc938 Support VP8 hardware encoding and decoding on IA devices. by glaznev@webrtc.org · 10 years ago
b9557a9 Fix code to handle crashes for non-VP8. by pbos@webrtc.org · 10 years ago
b6817d7 - Add a SetPriority method to ThreadWrapper by tommi@webrtc.org · 10 years ago
66df3cf Set WebRtcVideoEngine2 as the WebRtcMediaEngine. by pbos@webrtc.org · 10 years ago
8296ec5 Fix heap-use-after-free in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
a3209a2 Release buffer pool in Vp8DecoderImpl::Release(). by pbos@webrtc.org · 10 years ago
8904290 Make screenshare target bitrate experiment always on by pbos@webrtc.org · 10 years ago
d9c5024 Roll chromium_revision bd49b12..6311617 (320783:321517) by kjellander@webrtc.org · 10 years ago
9f9ea7e Clean up webrtc external capture. by perkj@webrtc.org · 10 years ago
443ad40 Remove FullStackTest frame pointer handles. by pbos@webrtc.org · 10 years ago
6231fb6 Prevent crashes when copying a zero-size frame. by pbos@webrtc.org · 10 years ago
6069032 Refactor audio_coding/isac: removed usage of macro WEBRTC_SPL_LSHIFT_W32 by bjornv@webrtc.org · 10 years ago
4ab23d0 Refactor audio_coding/ilbc: removes usage of macro WEBRTC_SPL_LSHIFT_W32 by bjornv@webrtc.org · 10 years ago
bd8c865 Remove build-time beamformer flags. by andrew@webrtc.org · 10 years ago
04c5098 Add the Ooura FFT to RealFourier. by andrew@webrtc.org · 10 years ago