Log - 812dd11f8c83f391bb9efa1ebcee9411475e46be - platform/external/webrtc

812dd11 Add baseline generation/verification to BWE test framework. by solenberg@webrtc.org · 11 years ago
499631c Utility class for reading/writing network-byte-ordered integers. by sprang@webrtc.org · 11 years ago
37968a9 Change BitrateStats to more generalized RateStatistics by sprang@webrtc.org · 11 years ago
b613b5a Set local SSRC for VideoReceiveStream. by pbos@webrtc.org · 11 years ago
5ecdef1 Do not use recursive calling in NetEq test tools by henrik.lundin@webrtc.org · 11 years ago
e003455 RTCPeerConnection(objc): avoid leaking ICE candidate on addition. by fischman@webrtc.org · 11 years ago
8418e96 Fixing NetEq tests for new Opus version by tina.legrand@webrtc.org · 11 years ago
54e8bfa Apprtc demo: add DSCP support. by braveyao@webrtc.org · 11 years ago
03c7a35 Fixing long lines in apprtc.py. by phoglund@webrtc.org · 11 years ago
e1fc3f2 Disable check for all sent SSRCs being valid. by pbos@webrtc.org · 11 years ago
bd41a84 This CL adds an API to enable robust validation of delay estimates. by bjornv@webrtc.org · 11 years ago
b627f67 Fixes a crash in the pacer where it fails to find a normal prio packet if there are no high prio packets, given that the queue has grown too large. by stefan@webrtc.org · 11 years ago
1f7c8d8 Lock frame in ViECapturer::IncomingFrameI420. by pbos@webrtc.org · 11 years ago
13d38a1 Set up SSRCs correctly after switching codec. by pbos@webrtc.org · 11 years ago
d1a1c35 Recommit CL5184 by bjornv@webrtc.org · 11 years ago
c8f76dd Refactor Remote Estimators Test into a more reusable form. by solenberg@webrtc.org · 11 years ago
82eb3a6 Revert 5184 "Small refactoring change in delay_estimator." by bjornv@webrtc.org · 11 years ago
eea079a Small refactoring change in delay_estimator. by bjornv@webrtc.org · 11 years ago
19a40ff Ensure that no packet stays in the pacer queue for longer than 2 seconds. by stefan@webrtc.org · 11 years ago
b3ea3af Create default implementation to fix issue in libjingle by sprang@webrtc.org · 11 years ago
4070935 Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
c7ff8f9 Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api. by asapersson@webrtc.org · 11 years ago
bd51d93 LSan suppressions for libjingle_peerconnection_unittest by kjellander@webrtc.org · 11 years ago
7f95998 Remove const in vie_rtp_rtcp, where there is conflict with by sprang@webrtc.org · 11 years ago
d89b52a Faster implementation of BitRateStats. by mikhal@webrtc.org · 11 years ago
326bcff Updated WebRTC version to 3.47 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
4e3161d Style-option file for clang-format. by pbos@webrtc.org · 11 years ago
3260f10 Made video quality toolchain more configurable. by phoglund@webrtc.org · 11 years ago
47fadba Add include stdlib.h to files using abs. by stefan@webrtc.org · 11 years ago
4ab4fc0 Add test for automatically disabling padding when no video is being captured. by stefan@webrtc.org · 11 years ago
b5bc098 Clear empty video frames in unittest so DrMemory will allow them to be read without an uninitialized read error. by fbarchard@google.com · 11 years ago
aa74b5d Add success/error callback to set sdp calls. by wu@webrtc.org · 11 years ago
5272eb8 Don't register iSAC-swb and iSAC-fb in NetEqDecodingTest. by turaj@webrtc.org · 11 years ago
e839da0 Fix MouseCursor to MouseCursorShape conversion in ScreenCapturerWin. by sergeyu@chromium.org · 11 years ago
78b41a0 Fix issues with sequence number wrap-around in jitter statistics. by turaj@webrtc.org · 11 years ago
832bd74 libyuv r874 for build improvements on ios/android, and improved YUV scale performance. by fbarchard@google.com · 11 years ago
b43202d Disable PeerConnectionEndToEndTest for tsanv2 build. by wu@webrtc.org · 11 years ago
1e8c93c Distinguish instances of ACM1 from ACM2 by a version string. This is fpr testing purposes and will be removed when the experiment is done and ACM1 is fade out. by turaj@webrtc.org · 11 years ago
2ffb149 Replace VideoFrameI420 with I420VideoFrame. by pbos@webrtc.org · 11 years ago
b0ed8f8 Don't reset the AEC filter in extended mode. by andrew@webrtc.org · 11 years ago
9e85c01 Add exception handling when configuring MediaCodc in order to prevent break in the new sdk release. by dwkang@webrtc.org · 11 years ago
9fe3603 Renaming ViEEncoderObserver::VideoSuspended by henrik.lundin@webrtc.org · 11 years ago
484ee96 Protect reads of ViEEncoder::video_suspended_. by pbos@webrtc.org · 11 years ago
1977960 AppRTCDemo(ios): remove codesigning hack now that gyp signs by default. by fischman@webrtc.org · 11 years ago
ef2d554 Increase size of pacer window to 500 ms as that better matches the encoder. by stefan@webrtc.org · 11 years ago
331d440 Connect pacer/padding to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
ffe1b17 Lock access to ModuleRtpRtcpImpl::simulcast_. by pbos@webrtc.org · 11 years ago
2c46f8d Rename DestroyStream methods to include Video. by pbos@webrtc.org · 11 years ago
6f6ba6e Fix issues with sequence number wrap-around in jitter statistics by henrik.lundin@webrtc.org · 11 years ago
b3cc78d Add -Wnon-virtual-dtor warning for C++ code. by pbos@webrtc.org · 11 years ago
72964bd Make interface destructor virtual by sprang@webrtc.org · 11 years ago
8d02f5d Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
54a0551 Increase run-time for full stack test for the rtt to be added reliably to the delay measurement. by asapersson@webrtc.org · 11 years ago
425e1d0 Typo in vie_autotest_win.cc by braveyao@webrtc.org · 11 years ago
a750044 Fixes a crash in VoE when unregistering JNI hooks. by henrike@webrtc.org · 11 years ago
364f204 Update talk to 56698267. by wu@webrtc.org · 11 years ago
dc50aae Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
b24d335 Added ViE API for getting overuse measure. by asapersson@webrtc.org · 11 years ago
d29d4e9 Deliver I420VideoFrames from VideoRender module. by pbos@webrtc.org · 11 years ago
1ae1d0c Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module). by asapersson@webrtc.org · 11 years ago
27326b6 Rename newapi::Transport::SendRTP()->SendRtp(). by pbos@webrtc.org · 11 years ago
ce90eff Rename RTP-extension constants. by pbos@webrtc.org · 11 years ago
53c8573 Rename video streams' start/stop methods. by pbos@webrtc.org · 11 years ago
5a63655 Rename Call::Create{Receive,Send}Stream(). by pbos@webrtc.org · 11 years ago
0b72f58 Add experimental noise suppression dummy API. by aluebs@webrtc.org · 11 years ago
5d85819 Fix DesktopAndCursorComposer to restore frames to the original state. by sergeyu@chromium.org · 11 years ago
7a05ae5 Adding back main() to the test. Now it is possible to choose between ACM1 and ACM2, furthermore, the test can simulate a channel with packet loss and FEC can be activated. Packet loss pattern is based on channel implementation in Channel{.cc,.h}, which currently is a determenistic pattern with 1 every 3rd packet is discarded. by turaj@webrtc.org · 11 years ago
9c5fb76 Exclude AV-sync test from Valgrind platforms. by pbos@webrtc.org · 11 years ago
ce8e093 Rename AutoMute to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
28bf50f Fix test broken with r5128. by stefan@webrtc.org · 11 years ago
b082ade Hook up audio/video sync to Call. by stefan@webrtc.org · 11 years ago
4cfa605 Fix breakage after introducing new test. by stefan@webrtc.org · 11 years ago
69969e2 Improve Call tests for RTX. by stefan@webrtc.org · 11 years ago
6e95d7a Increment RTP timestamps for padding packets by henrik.lundin@webrtc.org · 11 years ago
6488761 Implement VideoSendStream::SetCodec(). by pbos@webrtc.org · 11 years ago
183c727 Disable datachannel_unittest.cc by sergeyu@chromium.org · 11 years ago
a23f0ca Update talk to 56619788 by sergeyu@chromium.org · 11 years ago
e872285 Disable all vie_auto_tests on Linux for now (take 2) by kjellander@webrtc.org · 11 years ago
c848985 Disable all automated vie_auto_tests on Linux for now by kjellander@webrtc.org · 11 years ago
9b82f5a Fix for RTX in combination with pacing. by stefan@webrtc.org · 11 years ago
03f3370 Inject config when creating channels to override the existing one. by turaj@webrtc.org · 11 years ago
e8433eb Reimplementing NetEq4's AudioVector by henrik.lundin@webrtc.org · 11 years ago
3859951 Parse next RTCP XR report block after an unsupported block type. by asapersson@webrtc.org · 11 years ago
3e42726 Reducing opus_test runtime to pass Android test by minyue@webrtc.org · 11 years ago
e03cafa MIPS optimizations for AECM audio processing module by andrew@webrtc.org · 11 years ago
b073010 Move audio_processing dependencies to a variable. by andrew@webrtc.org · 11 years ago
57eb858 Remove ".." from include_dirs in build/common. by pbos@webrtc.org · 11 years ago
6e908b3 Remove unnecessary include_dirs from audio_processing. by andrew@webrtc.org · 11 years ago
00ed170 Roll libvpx 225010:232686. by marpan@webrtc.org · 11 years ago
5973f3a Remove unneeded includes from trace_posix.cc. by andrew@webrtc.org · 11 years ago
48df381 Fix for making sure that the packet in order checks are done prior to updating the last received packet state. by stefan@webrtc.org · 11 years ago
bff9620 Fix log build error for Chromium builds. by henrikg@webrtc.org · 11 years ago
4c828e1 Remove update_resources.py as it's no longer used. by kjellander@webrtc.org · 11 years ago
f1a4817 Replace disabled logging with a restricted logging mode. by andrew@webrtc.org · 11 years ago
5adc897 Updated WebRTC version to 3.46 by elham@webrtc.org · 11 years ago
a7855a8 Fix for xgetbv on Visual Studio 2010. by fbarchard@google.com · 11 years ago
bde3056 Fix for video_processor_intergration_tests to run in parallel. by marpan@webrtc.org · 11 years ago
c4225b6 Update getUserMedia W3C conformance tests. by kjellander@webrtc.org · 11 years ago
8bad50e Sending status fix for module. by asapersson@webrtc.org · 11 years ago
16d6254 Update talk to 56183333. by wu@webrtc.org · 11 years ago