Log - bdcf38c89446b1b464a646414f6cd7573a190bd1 - platform/external/webrtc

bdcf38c cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class by magjed@webrtc.org · 11 years ago
ad0e71c Update mock_frame_dropper.h to use size_t by kjellander@webrtc.org · 11 years ago
4591fbd Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 11 years ago
edc6e57 Support loopback mode and command line execution by glaznev@webrtc.org · 11 years ago
6ff3ac1 Fix problems if first packet into NetEq is rejected by henrik.lundin@webrtc.org · 11 years ago
ed91068 Create a NetEq test for when the first incoming payload type is unknown by henrik.lundin@webrtc.org · 11 years ago
049e4ec Change default values for CpuOveruseOptions. by asapersson@webrtc.org · 11 years ago
f58b455 cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. by magjed@webrtc.org · 11 years ago
40af3a5 Revert "Add DCHECK to ensure that NetEq's packet buffer is not empty" by henrik.lundin@webrtc.org · 11 years ago
6f6ef72 Add DCHECK to ensure that NetEq's packet buffer is not empty by henrik.lundin@webrtc.org · 11 years ago
2176db3 AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. (re-land) by henrika@webrtc.org · 11 years ago
c56814f Roll chromium_revision 91f1781..d8c9041 by kjellander@webrtc.org · 11 years ago
087da13 Add empty 3 band splitting filter API by aluebs@webrtc.org · 11 years ago
2656bf8 Fix ExpectedQueueTimeMs() to avoid truncation or overflow. by pkasting@chromium.org · 11 years ago
930e004 Add jmi field for packets discarded due to network error by guoweis@webrtc.org · 11 years ago
c72a22c Add preliminary empty file videoframefactory.cc by magjed@webrtc.org · 11 years ago
f5b56fb Annotate COMPILE_ASSERT with __attribute__((unused)). by pbos@webrtc.org · 11 years ago
4ef22d1 Setting Opus FEC as default by minyue@webrtc.org · 11 years ago
966a708 Use RtpFileSource in NetEqDecodingTest by henrik.lundin@webrtc.org · 11 years ago
4ec19e3 Revert 7707 "cricket::VideoAdapter: Drop frames before spending ..." by tommi@webrtc.org · 11 years ago
858dbbc cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. by magjed@webrtc.org · 11 years ago
6a782c2 Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases. by henrike@webrtc.org · 11 years ago
be05c74 Wrap the splitting filter in its own class by aluebs@webrtc.org · 11 years ago
67c2247 Disable EndToEnd.GetStats test. by pbos@webrtc.org · 11 years ago
a73d746 Revert 7702 "cricket::VideoAdapter: Drop frames before spending ..." by magjed@webrtc.org · 11 years ago
bbd8cad cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. by magjed@webrtc.org · 11 years ago
ece3890 Report total bitrate for all streams in GetStats. by pbos@webrtc.org · 11 years ago
35c1ace Revert 7698 "WebRtcVideoMediaChannel::SetSendParams: Don't cap r..." by magjed@webrtc.org · 11 years ago
a1f5b96 Remove unnecessary copying of libjingle resource files. by kjellander@webrtc.org · 11 years ago
52da44b WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution by magjed@webrtc.org · 11 years ago
49ff40e Make SetREMBData accept vector of SSRCs. by pbos@webrtc.org · 11 years ago
a9c2d45 Fix and enable CanReceiveFec test. by pbos@webrtc.org · 11 years ago
ee30082 Set correct sample rate in far_frame in audioproc tool. by bjornv@webrtc.org · 11 years ago
52bb521 Update isolate files for Android APK tests. by kjellander@webrtc.org · 11 years ago
312614a Add jmi field for packets discarded due to network error by guoweis@webrtc.org · 11 years ago
90b9b08 Fix a platform check to use WEBRTC_WIN instead of OS_WIN. by jiayl@webrtc.org · 11 years ago
6ca6190 Fix a SCTP message reordering issue in datachannel.cc. by jiayl@webrtc.org · 11 years ago
ea73ff7 webrtc::Scaler: Preserve aspect ratio by magjed@webrtc.org · 11 years ago
0b3d89b VideoSendStreamTest.SwapsI420VideoFrames: Initialize frame memory to avoid drmemory errors by magjed@webrtc.org · 11 years ago
14ea50a Change the static_library("webrtc") to a source set in the GN build. by kjellander@webrtc.org · 11 years ago
0e37b89 replace inline assembly WebRtcAecm_CalcLinearEnergiesNeon by intrinsics. by andrew@webrtc.org · 11 years ago
e497be3 replace inline assembly WebRtcAecm_StoreAdaptiveChannelNeon by intrinsics. by andrew@webrtc.org · 11 years ago
0e71070 Use ScreenCapturer to capture the whole and clip to the window rect when the shared window is on the top. by jiayl@webrtc.org · 11 years ago
a367aea Bump to version 40 by tnakamura@webrtc.org · 11 years ago
f7c5d4f Revert 7679 "webrtc::Scaler: Preserve aspect ratio" by magjed@webrtc.org · 11 years ago
525baea Add PROJECT to codereview.settings by kjellander@webrtc.org · 11 years ago
944fb57 Roll chromium_revision 375f736..91f1781 by kjellander@webrtc.org · 11 years ago
809986b webrtc::Scaler: Preserve aspect ratio by magjed@webrtc.org · 11 years ago
cd621a8 Add thread annotations to overuse_frame_detector class. by asapersson@webrtc.org · 11 years ago
8038d42 Follow-up fixes for G722 by henrik.lundin@webrtc.org · 11 years ago
1431e4d Revert 7675 "Make an AudioEncoder subclass for iSAC" by turaj@webrtc.org · 11 years ago
05feff0 Make an AudioEncoder subclass for iSAC by kwiberg@webrtc.org · 11 years ago
33045ab Change from talk/p2p (r7664) "(Auto)update libjingle 79414100-> 79428003". by henrike@webrtc.org · 11 years ago
43e033e Change from talk/p2p (r7572): "Improve the logging when a TCP connection is deleted." by henrike@webrtc.org · 11 years ago
4ffc734 replace inline assembly WebRtcAecm_ResetAdaptiveChannelNeon by intrinsics. by andrew@webrtc.org · 11 years ago
d024f75 clear asm code and unused functions in audio processing module by andrew@webrtc.org · 11 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 · 11 years ago
d819803 Wire up DSCP support in WebRtcVideoEngine2. by pbos@webrtc.org · 11 years ago
83d4804 Put send-side bwe probing under finch experiment. by stefan@webrtc.org · 11 years ago
957e802 Refactor SetDefaultEncoderConfig to work on existing codecs. by pbos@webrtc.org · 11 years ago
a5d29fc Add unit to dropped frames. by pbos@webrtc.org · 11 years ago
bd495fa .gitignore updates by kjellander@webrtc.org · 11 years ago
3c1970f (Auto)update libjingle 79414100-> 79428003 by buildbot@webrtc.org · 11 years ago
188d3b2 Enable VP9 video codec support on webrtcvideoengine behind a field trial. by andresp@webrtc.org · 11 years ago
f85dbce Reapply "Advertise G722 as 8 kHz rather than 16 kHz"" by henrik.lundin@webrtc.org · 11 years ago
d105cc8 Change dummy address to use 0.0.0.0 instead of :: by perkj@webrtc.org · 11 years ago
d42a3ad Remove partially defined WebRtcRTPHeader from Parse(). by pbos@webrtc.org · 11 years ago
a2ef4fe Prevent a lot of VideoSendStream reconfigures. by pbos@webrtc.org · 11 years ago
82775b1 Refactor webrtcvideoengines to have the default list of supported codecs being generated in runtime. by andresp@webrtc.org · 11 years ago
5e16066 Reland Volume buttons in AppRTCDemo should affect output audio volume (part I). by henrika@webrtc.org · 11 years ago
332331f Use uint16s for port numbers in webrtc/p2p/base. by pkasting@chromium.org · 11 years ago
d89b69a Fix WebRTC Win64 + BoringSSL build. by henrike@webrtc.org · 11 years ago
dd43bbe Volume buttons in AppRTCDemo should affect output audio volume (part II). by henrika@webrtc.org · 11 years ago
dced5d7 Revert "Advertise G722 as 8 kHz rather than 16 kHz" by henrik.lundin@webrtc.org · 11 years ago
34bda43 (Auto)update libjingle 79326895-> 79329222 by buildbot@webrtc.org · 11 years ago
e5421e9 Volume buttons in AppRTCDemo should affect output audio volume. by henrika@webrtc.org · 11 years ago
fd0efb6 Remove deprecated PeerConnection APIs. by perkj@webrtc.org · 11 years ago
19b4741 Removing unused method GetDefaultVideoEncoderConfig. by andresp@webrtc.org · 11 years ago
931e3da Log formatting fix for VideoEncoderConfig. by pbos@webrtc.org · 11 years ago
0ef890a (Auto)update libjingle 79285346-> 79320771 by buildbot@webrtc.org · 11 years ago
6340acd AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. by mcasas@webrtc.org · 11 years ago
1dcca40 Advertise G722 as 8 kHz rather than 16 kHz by henrik.lundin@webrtc.org · 11 years ago
8b2058e Remove the state_ member from AudioDecoder by kwiberg@webrtc.org · 11 years ago
32022c6 Revert 7642 "Fix memcheck and dr memory after flakiness dashboar..." by kjellander@webrtc.org · 11 years ago
724fbaf Fix memcheck and dr memory after flakiness dashboard deployment. by kjellander@webrtc.org · 11 years ago
7e4a05e Exclude SendsAndReceivesVP9 for linux-memcheck. by marpan@webrtc.org · 11 years ago
53bed75 Change DrMemory exclusion to match changed test name. by andrew@webrtc.org · 11 years ago
f6b7c7e Exclude SendsAndReceivesVP9 for WinDrMemory. by marpan@webrtc.org · 11 years ago
e1745cb Adjust parameter in vp9 rate control test. by marpan@webrtc.org · 11 years ago
5f1e2e4 Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test. by marpan@webrtc.org · 11 years ago
ee9d61c This fixes a small memory leak (found using Xcode/Instruments on iOS) in by tkchin@webrtc.org · 11 years ago
6a364fe Remove uses of build date/time. by pbos@webrtc.org · 11 years ago
0bae1fa Wire up bandwidth stats to the new API and webrtcvideoengine2. by stefan@webrtc.org · 11 years ago
a22a628 (Auto)update libjingle 79205306-> 79244016 by buildbot@webrtc.org · 11 years ago
72fd339 Restore old behavior for Android in fileutils.cc by kjellander@webrtc.org · 11 years ago
f6e1600 Roll chromium_revision d3db2ff..375f736 by kjellander@webrtc.org · 11 years ago
dc86624 Fix android_clang build. by glaznev@webrtc.org · 11 years ago
368215d Revert 7623 "Remove the state_ member from AudioDecoder" by niklas.enbom@webrtc.org · 11 years ago
8a232f6 Revert 7625 "Don't use DCHECK when you need the side effects..." by niklas.enbom@webrtc.org · 11 years ago
795d003 (Auto)update libjingle 79200114-> 79205306 by buildbot@webrtc.org · 11 years ago