Log - 5614cf16e71e59e4785deca05249d9a06f746d66 - platform/external/webrtc

5614cf1 audio_processing: Use fixed aggregation window in delay metrics by bjornv@webrtc.org · 10 years ago
6e25182 Whitespace change after enabling gnumbd by kjellander@webrtc.org · 10 years ago
ccd608e Whitespace change for git updater by kjellander@webrtc.org · 10 years ago
0bc73a1 Whitespace change to trigger git updater by kjellander@webrtc.org · 10 years ago
f68ffca Add PRESUBMIT check for GYP files including source files above itself. by kjellander@webrtc.org · 10 years ago
76e5e20 Roll chromium_revision 4664fe0..9070a80 (312733:313233) by kjellander@webrtc.org · 10 years ago
273fbbb Update StreamDataCounter with FEC bytes. by asapersson@webrtc.org · 10 years ago
70117a8 AEC: Implements a new function for calculating delay metrics by bjornv@webrtc.org · 10 years ago
fc5ad95 Reland of: "Implement elapsed time and capture start NTP time estimation." revision @8139 by magjed@webrtc.org · 10 years ago
8501ee6 Support VP8 HW decoding on devices with Exynos codec. by glaznev@webrtc.org · 10 years ago
df9a41d Fix bug in GetREDStatus(): it doesn't actually return the current status. by pkasting@chromium.org · 10 years ago
82415e3 Update AppRTCDemo to use renamed GAE messages. by glaznev@webrtc.org · 10 years ago
041035b Add an AudioRingBuffer class wrapper for the ring_buffer.h C interface. by andrew@webrtc.org · 10 years ago
4dba2e9 Consolidate anonymous namespace content and file-static methods to all be in the by pkasting@chromium.org · 10 years ago
d7e34e1 Make it easier to use external libyuv + cleanup GYP files. by kjellander@webrtc.org · 10 years ago
d25c034 Refactor common_audio/vad: Removed usage of macro WEBRTC_SPL_MUL_16_16() by bjornv@webrtc.org · 10 years ago
04cd466 Move ThreadChecker into rtc_base_approved. by tommi@webrtc.org · 10 years ago
38d11b8 Enable encoder multi-threading for VP9. by marpan@webrtc.org · 10 years ago
6f200b5 Temporarily revert r8147 ("Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h") by kwiberg@webrtc.org · 10 years ago
b6fab2b Introduce rtc::CheckedDivExact by henrik.lundin@webrtc.org · 10 years ago
19eb4e4 Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h by kwiberg@webrtc.org · 10 years ago
995b4c9 Remove win_asan trybot from PRESUBMIT.py by kjellander@webrtc.org · 10 years ago
acb8085 Roll chromium_revision c086b4e..4664fe0 (312108:312733) by kjellander@webrtc.org · 10 years ago
7519de5 Revert 8136 "Remove frame copy in ViEExternalRendererImpl::Rende..." by tkchin@webrtc.org · 10 years ago
0f98844 Revert 8139 "Implement elapsed time and capture start NTP time e..." by tkchin@webrtc.org · 10 years ago
dacdd94 Reland r7980: by jiayl@webrtc.org · 10 years ago
8919cfe Change a GYP reference to cpufeatures.gypi by fdegans@chromium.org · 10 years ago
ad3ee2c Implement elapsed time and capture start NTP time estimation. by pbos@webrtc.org · 10 years ago
a02d768 Disable DtmfSenderTest.InsertDtmfWithCommaAsDelay due to flakiness by kjellander@webrtc.org · 10 years ago
456f014 Re-allowing RED in voice engine. by minyue@webrtc.org · 10 years ago
182ea46 Remove frame copy in ViEExternalRendererImpl::RenderFrame by magjed@webrtc.org · 10 years ago
73ee453 Switch to use range based loops in the BWE simulation framework. by stefan@webrtc.org · 10 years ago
36d5c3c Leave BIO_METHOD non-const. by davidben@webrtc.org · 10 years ago
586f2ed Change GetStreamBySsrc to not copy StreamParams. by tommi@webrtc.org · 10 years ago
7e5b380 Fix a crash in AllocationSequence. Internal bug 19074679. by jiayl@webrtc.org · 10 years ago
ff108fe Revert 8125 "Modify some tests to never use DTX disable mode" by kjellander@webrtc.org · 10 years ago
b40c7bb Change sprintf use in talk samples to snprintf by jlmiller@webrtc.org · 10 years ago
ea1c842 Correct GetDriveType error handling. by jlmiller@webrtc.org · 10 years ago
043db24 Modify some tests to never use DTX disable mode by henrik.lundin@webrtc.org · 10 years ago
e5251ad Integrate send-side BWE into simulation framework. by stefan@webrtc.org · 10 years ago
cfd82df Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. by asapersson@webrtc.org · 10 years ago
3dd33a6 Fix bug in thresholds for bitrate probing and adjust thresholds to allow a larger dispersion and concentration for successful probes. by stefan@webrtc.org · 10 years ago
fbd37bd Make iSAC SWB own its decoder by henrik.lundin@webrtc.org · 10 years ago
cceb166 Fix a use-after-free when sending queued messages is aborted for blocked channel. by jiayl@webrtc.org · 10 years ago
e65d9d9 Fix an unitialized variable warning. by andrew@webrtc.org · 10 years ago
c429b82 GN: Prepare to remove webrtc_base target by kjellander@webrtc.org · 10 years ago
c78d81a Re-land "Support 48kHz in AEC" by aluebs@webrtc.org · 10 years ago
e81c5d6 Fix TransientDetectorTest in modules_unittests on Android ARM64 by aluebs@webrtc.org · 10 years ago
11af039 Disable AcmSenderBitExactnessOldApi.Opus_stereo_20ms_voip on ARM64. by minyue@webrtc.org · 10 years ago
df7b65b Change CreateOrGetReportBlockInformation to have one return path. by asapersson@webrtc.org · 10 years ago
f938922 Simplify and guard access to WindowsRealTimeClock. by pbos@webrtc.org · 10 years ago
4fb7e25 Update StatsReport and by extension StatsCollector to reduce data copying. by tommi@webrtc.org · 10 years ago
f66a6b2 Remove unnecessary dependencies from webrtc_all target. by kjellander@webrtc.org · 10 years ago
e7358ea Only report fraction of lost packets if report_block_stats has been updated. by asapersson@webrtc.org · 10 years ago
9ffd8fe Indentation changes. by asapersson@webrtc.org · 10 years ago
fedb9ea Correct the class name in peerconnection_jni.cc. by braveyao@webrtc.org · 10 years ago
5f93d0a Update libjingle license statements at top of talk files for consistency by jlmiller@webrtc.org · 10 years ago
cbacd9e Bump to version 41. by tnakamura@webrtc.org · 10 years ago
7dba786 Setting Opus target application. by minyue@webrtc.org · 10 years ago
853049f Move internal capture+render to build_with_chromium==0 condition by kjellander@webrtc.org · 10 years ago
511ab3e Roll chromium_revision a6eafec..c086b4e by kjellander@webrtc.org · 10 years ago
ee0c100 Revert 8080 "Support 48kHz in AEC" by tina.legrand@webrtc.org · 10 years ago
f88f88e Remove webrtc/base/compile_assert.h by kwiberg@webrtc.org · 10 years ago
9691b36 Cleanup for Rtp Rtcp API test. by changbin.shao@intel.com · 10 years ago
8e327c4 Update StatsCollector's interface in preparation of more changes. by tommi@webrtc.org · 10 years ago
43e54e3 Revert 8095 "Update StatsCollector's interface in preparation of..." by tommi@webrtc.org · 10 years ago
5b76fd7 Update StatsCollector's interface in preparation of more changes. by tommi@webrtc.org · 10 years ago
474e36e Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps. by stefan@webrtc.org · 10 years ago
f9d3555 Fixing LD_LIBRARY_PATH, improving safety for libjingle java unit test. by phoglund@webrtc.org · 10 years ago
ce3ac53 Adding TRYSERVER_PROJECT to codereview.settings. by kjellander@webrtc.org · 10 years ago
018c087 Add /talk/examples/androidtests/{bin,gen} to .gitignore. by kjellander@webrtc.org · 10 years ago
a32d154 Disable tests failing on Android ARM64 (Nexus9). by kjellander@webrtc.org · 10 years ago
ff9462e Disable WebRtcVideoMediaChannelSimulcastTest::SimulcastSend_* on tsan. by sprang@webrtc.org · 10 years ago
2624b1e Remove unused private data member engine_id_ by tommi@webrtc.org · 10 years ago
fe672e3 release the turn allocation by sending a refresh request with lifetime 0 by pthatcher@webrtc.org · 10 years ago
d7de120 Re-enable the messagequeue unittests. These were commented out at one point but never reenabled. by decurtis@webrtc.org · 10 years ago
a1aea10 Revert r8076 "Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps." by stefan@webrtc.org · 10 years ago
4ba1e44 Remove unnecessary remote bitrate estimator build rule which serves no purpose. by andresp@webrtc.org · 10 years ago
487a444 Add stats collection for the data channel. by decurtis@webrtc.org · 10 years ago
357469d Fixes reference counting problem when a TransportProxy points to a Transport prior to creating channels. by decurtis@webrtc.org · 10 years ago
ef2a5dd Update AppRTCDemo UI. by tkchin@webrtc.org · 10 years ago
64d3c4b Support 48kHz in AEC by aluebs@webrtc.org · 10 years ago
89aa276 Fix a case where empty candidate id is used by guoweis@webrtc.org · 10 years ago
d82f55d Only adapt AGC when the desired signal is present by aluebs@webrtc.org · 10 years ago
3e42a8a Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps. by stefan@webrtc.org · 10 years ago
32e8528 Log configs when creating video streams in Call. by pbos@webrtc.org · 10 years ago
1f67b53 Remove dual stream functionality in ACM by henrik.lundin@webrtc.org · 10 years ago
9ce01e6 Clean unnecessary workaround for chromium import. by andresp@webrtc.org · 10 years ago
0800db7 Add percentage of fec packets and recovered media packets to histogram stats: by asapersson@webrtc.org · 10 years ago
61c1247 Fix a case where empty candidate id is used by guoweis@webrtc.org · 10 years ago
6c38552 Add WebRtcIsacfix_AllpassFilter2FixDec16Neon()'s intrinsics version. by andrew@webrtc.org · 10 years ago
5a92b78 Add beamforming to audioproc_float utility. by mgraczyk@chromium.org · 10 years ago
6b63015 Move ring_buffer to common_audio. by andrew@webrtc.org · 10 years ago
fd630a5 Add BundlePolicy to RTCConfiguration. Don't change any behavior. Just make it possible to make progress in Chromium while we work on the behavior. by pthatcher@webrtc.org · 10 years ago
693e01c Fix searching for DirectX SDK during GN build. by kjellander@webrtc.org · 10 years ago
f1c8b90 Remove WebRtcVideoEncoderFactory2. by pbos@webrtc.org · 10 years ago
e5a31e1 Revert removing of compile_assert.h. by turaj@webrtc.org · 10 years ago
85fa94d Exclude EndToEndTest.SendsAndReceivesH264 for Dr Memory. by kjellander@webrtc.org · 10 years ago
387841a Improved fairness simulation by starting the flows 20 seconds apart. by stefan@webrtc.org · 10 years ago
f18fba2 Implement SimulcastEncoderAdapter support. by pbos@webrtc.org · 10 years ago