Log - d024f759a86a9a50ade7e864b1b673d75f5bf160 - platform/external/webrtc

d024f75 clear asm code and unused functions in audio processing module by andrew@webrtc.org · 10 years ago
c492231 Removes talk/xmllite, talk/xmpp and talk/p2p as they are no longer used by gyp/gn builds. by henrike@webrtc.org · 10 years ago
d819803 Wire up DSCP support in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
83d4804 Put send-side bwe probing under finch experiment. by stefan@webrtc.org · 10 years ago
957e802 Refactor SetDefaultEncoderConfig to work on existing codecs. by pbos@webrtc.org · 10 years ago
a5d29fc Add unit to dropped frames. by pbos@webrtc.org · 10 years ago
bd495fa .gitignore updates by kjellander@webrtc.org · 10 years ago
3c1970f (Auto)update libjingle 79414100-> 79428003 by buildbot@webrtc.org · 10 years ago
188d3b2 Enable VP9 video codec support on webrtcvideoengine behind a field trial. by andresp@webrtc.org · 10 years ago
f85dbce Reapply "Advertise G722 as 8 kHz rather than 16 kHz"" by henrik.lundin@webrtc.org · 10 years ago
d105cc8 Change dummy address to use 0.0.0.0 instead of :: by perkj@webrtc.org · 10 years ago
d42a3ad Remove partially defined WebRtcRTPHeader from Parse(). by pbos@webrtc.org · 10 years ago
a2ef4fe Prevent a lot of VideoSendStream reconfigures. by pbos@webrtc.org · 10 years ago
82775b1 Refactor webrtcvideoengines to have the default list of supported codecs being generated in runtime. by andresp@webrtc.org · 10 years ago
5e16066 Reland Volume buttons in AppRTCDemo should affect output audio volume (part I). by henrika@webrtc.org · 10 years ago
332331f Use uint16s for port numbers in webrtc/p2p/base. by pkasting@chromium.org · 10 years ago
d89b69a Fix WebRTC Win64 + BoringSSL build. by henrike@webrtc.org · 10 years ago
dd43bbe Volume buttons in AppRTCDemo should affect output audio volume (part II). by henrika@webrtc.org · 10 years ago
dced5d7 Revert "Advertise G722 as 8 kHz rather than 16 kHz" by henrik.lundin@webrtc.org · 10 years ago
34bda43 (Auto)update libjingle 79326895-> 79329222 by buildbot@webrtc.org · 10 years ago
e5421e9 Volume buttons in AppRTCDemo should affect output audio volume. by henrika@webrtc.org · 10 years ago
fd0efb6 Remove deprecated PeerConnection APIs. by perkj@webrtc.org · 10 years ago
19b4741 Removing unused method GetDefaultVideoEncoderConfig. by andresp@webrtc.org · 10 years ago
931e3da Log formatting fix for VideoEncoderConfig. by pbos@webrtc.org · 10 years ago
0ef890a (Auto)update libjingle 79285346-> 79320771 by buildbot@webrtc.org · 10 years ago
6340acd AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. by mcasas@webrtc.org · 10 years ago
1dcca40 Advertise G722 as 8 kHz rather than 16 kHz by henrik.lundin@webrtc.org · 10 years ago
8b2058e Remove the state_ member from AudioDecoder by kwiberg@webrtc.org · 10 years ago
32022c6 Revert 7642 "Fix memcheck and dr memory after flakiness dashboar..." by kjellander@webrtc.org · 10 years ago
724fbaf Fix memcheck and dr memory after flakiness dashboard deployment. by kjellander@webrtc.org · 10 years ago
7e4a05e Exclude SendsAndReceivesVP9 for linux-memcheck. by marpan@webrtc.org · 10 years ago
53bed75 Change DrMemory exclusion to match changed test name. by andrew@webrtc.org · 10 years ago
f6b7c7e Exclude SendsAndReceivesVP9 for WinDrMemory. by marpan@webrtc.org · 10 years ago
e1745cb Adjust parameter in vp9 rate control test. by marpan@webrtc.org · 10 years ago
5f1e2e4 Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test. by marpan@webrtc.org · 10 years ago
ee9d61c This fixes a small memory leak (found using Xcode/Instruments on iOS) in by tkchin@webrtc.org · 10 years ago
6a364fe Remove uses of build date/time. by pbos@webrtc.org · 10 years ago
0bae1fa Wire up bandwidth stats to the new API and webrtcvideoengine2. by stefan@webrtc.org · 10 years ago
a22a628 (Auto)update libjingle 79205306-> 79244016 by buildbot@webrtc.org · 10 years ago
72fd339 Restore old behavior for Android in fileutils.cc by kjellander@webrtc.org · 10 years ago
f6e1600 Roll chromium_revision d3db2ff..375f736 by kjellander@webrtc.org · 10 years ago
dc86624 Fix android_clang build. by glaznev@webrtc.org · 10 years ago
368215d Revert 7623 "Remove the state_ member from AudioDecoder" by niklas.enbom@webrtc.org · 10 years ago
8a232f6 Revert 7625 "Don't use DCHECK when you need the side effects..." by niklas.enbom@webrtc.org · 10 years ago
795d003 (Auto)update libjingle 79200114-> 79205306 by buildbot@webrtc.org · 10 years ago
8125744 Cleanup RTCVideoRenderer interface. by tkchin@webrtc.org · 10 years ago
b8425bc Don't use DCHECK when you need the side effects... by kwiberg@webrtc.org · 10 years ago
45ecf4c (Auto)update libjingle 79169148-> 79192489 by buildbot@webrtc.org · 10 years ago
9e52558 Remove the state_ member from AudioDecoder by kwiberg@webrtc.org · 10 years ago
7c29e8c Add support for VP9 in webrtc::Call and video_loopback. by stefan@webrtc.org · 10 years ago
d839e0a Reduce to 2 probes when probing for initial bandwidth. by stefan@webrtc.org · 10 years ago
db26247 Add UMA for measuring the diff between the BWE at 2 seconds compared to the BWE at 20 seconds when the BWE should have converged. by stefan@webrtc.org · 10 years ago
8944c9d AppRTCDemoActivity: use differnet Themes for different API levels by mcasas@webrtc.org · 10 years ago
d367321 Add kjellander as PRESUBMIT.py OWNER by kjellander@webrtc.org · 10 years ago
dcebf2d Reworked paced sender queue by sprang@webrtc.org · 10 years ago
fad9aec Remove protected files from talk/PRESUBMIT.py. by pbos@webrtc.org · 10 years ago
88ef632 Falling back on single-stream on multiple SSRC. by pbos@webrtc.org · 10 years ago
28af641 Presubmit was not whitelisting libjingle_tests.gyp or sound.gyp due to a missing comma leading to a concatenation of the two strings in the whitelist. by henrike@webrtc.org · 10 years ago
b3265ac Adds support for finch experiments to video_loopback. by stefan@webrtc.org · 10 years ago
52b42cb Fix problem with late packets in NetEq by henrik.lundin@webrtc.org · 10 years ago
09cc686 Delete VideoReceiveStream channels in destructor. by pbos@webrtc.org · 10 years ago
6de75ca Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16 by kwiberg@webrtc.org · 10 years ago
c78cf97 Remove the useless dummy state parameter to WebRtcG711_* by kwiberg@webrtc.org · 10 years ago
b5d045e ReAdd PeerConnectionInterface::AddStream to fix Chrome build. by perkj@webrtc.org · 10 years ago
18de6f9 Change the PeerConnection proxy templates to use blocking method calls instead of using Thread::Send. by tommi@webrtc.org · 10 years ago
721ef63 Remove the codec_type_ member from AudioDecoder by kwiberg@webrtc.org · 10 years ago
c2dd5ee Prepare for removal of PeerConnectionObserver::OnError. by perkj@webrtc.org · 10 years ago
f37145f Enables AIMD control by default. by stefan@webrtc.org · 10 years ago
b0f4b3d Improving error message from neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
a663d90 (Auto)update libjingle 79104430-> 79104922 by buildbot@webrtc.org · 10 years ago
5f38c8d Android AppRTCDemo improvements: by glaznev@webrtc.org · 10 years ago
5804936 Add format members to AudioConverter for DCHECKing. by andrew@webrtc.org · 10 years ago
e451b75 Update rate control parameter in vp9 test. by marpan@webrtc.org · 10 years ago
4765ca5 Roll chromium_revision: 28d1981..d3db2ff by marpan@webrtc.org · 10 years ago
f866b2d Restore the void return type on WriteWavHeader. by andrew@webrtc.org · 10 years ago
b81e304 replace inline assembly WebRtcNsx_AnalysisUpdate by intrinsics. by andrew@webrtc.org · 10 years ago
f947180 Add Opus support to neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
96a9325 Implement external decoder support in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
548b228 Add UMA metrics for the initial (after two seconds) packet loss, round-trip time and bandwidth estimate of a WebRTC call. by stefan@webrtc.org · 10 years ago
96dc685 Add stats for video: by asapersson@webrtc.org · 10 years ago
2236267 Disable PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate under MSan by henrik.lundin@webrtc.org · 10 years ago
bf09976 Add more sanity checks to workaround the unidentified problem that CaptureThread is still running while related resouces are destroyed already. by braveyao@webrtc.org · 10 years ago
ed45896 Adjust/increase rate control thresold for a vp9 test. by marpan@webrtc.org · 10 years ago
5b88317 Add VP9 codec to VCM and vie_auto_test. by marpan@webrtc.org · 10 years ago
5072e0f Update Android projects to API level 21. by kjellander@webrtc.org · 10 years ago
818c9f9 replace inline assembly WebRtcNsx_SynthesisUpdateNeon by intrinsics. by andrew@webrtc.org · 10 years ago
a3ed713 Add a WavReader counterpart to WavWriter. by andrew@webrtc.org · 10 years ago
c2c94a9 Change default JVM location to /usr/lib/jvm/java-7-openjdk-amd64 by kjellander@webrtc.org · 10 years ago
78c222b Update all .isolate files for the new format. by kjellander@webrtc.org · 10 years ago
8a130c1 Update Android projects to API level 20. by kjellander@webrtc.org · 10 years ago
053c6ab Fix N7 camera aspect ratio. by glaznev@webrtc.org · 10 years ago
508c916 Build fix for MIPS32R6. by andrew@webrtc.org · 10 years ago
cc476aa Fix a name collision with Android libc++ by andrew@webrtc.org · 10 years ago
b7ed779 Implement conference-mode temporal-layer screencast. by pbos@webrtc.org · 10 years ago
3bf3d23 Configure A/V sync in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
4abadab Simplify bwe tests. by stefan@webrtc.org · 10 years ago
2dc6f31 Adapting bitrate according to maxplaybackrate for Opus. by minyue@webrtc.org · 10 years ago
8328e7c Revert "Revert part of r7561, "Refactor audio conversion functions."" by andrew@webrtc.org · 10 years ago
14146e4 arm64 iOS build. by tkchin@webrtc.org · 10 years ago
50ca986 Improve the logging when a TCP connection is deleted. by jiayl@webrtc.org · 10 years ago