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gerrit-public.fairphone.software
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platform
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external
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webrtc
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5614cf16e71e59e4785deca05249d9a06f746d66
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
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