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Author
2014-08-04
Update makefiles after merge of Chromium at 287308
Android Chromium Automerger
2014-07-29
Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...
Android Chromium Automerger
2014-07-24
Make sure padding is sent on the first sending RTP module.
mflodman@webrtc.org
2014-07-22
The lastest commit on this file was in
minyue@webrtc.org
2014-07-22
Remove remains of WEBRTC_NO_STL.
andresp@webrtc.org
2014-07-21
MIPS optimizations for ISAC (patch #2)
andrew@webrtc.org
2014-07-18
This is to re-open an earlier CL
minyue@webrtc.org
2014-07-18
Runtime guard for iOS7 property.
tkchin@webrtc.org
2014-07-18
Fix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS.
tkchin@webrtc.org
2014-07-18
This is related to an earlier CL of enabling Opus 48 kHz.
minyue@webrtc.org
2014-07-18
AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float
kwiberg@webrtc.org
2014-07-17
Reduce runtime of RingBufferTest by a factor of 100.
andrew@webrtc.org
2014-07-17
Use _numMixedParticipants instead of audioFrameList->size() to determine if t...
wu@webrtc.org
2014-07-17
Fix issue where padding is sent before media with undefined timestamps if not...
stefan@webrtc.org
2014-07-17
Remove unused ExperimentalNS API in AudioProcessing
aluebs@webrtc.org
2014-07-17
AudioBuffer: Eliminate the SplitChannelBuffer class
kwiberg@webrtc.org
2014-07-17
Simplify AudioBuffer::mixed_low_pass_data API
aluebs@webrtc.org
2014-07-17
AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameter
kwiberg@webrtc.org
2014-07-17
Add unit test for MediaFile WAV file writing
kwiberg@webrtc.org
2014-07-17
Fixes up rtc so that it compiles on iOS 8 SDK.
tkchin@webrtc.org
2014-07-16
r6709 lacks a change in BUILD.gn
minyue@webrtc.org
2014-07-16
Raw packet loss rate reported by RTP_RTCP module may vary too drastically ove...
minyue@webrtc.org
2014-07-16
Compile-time guard for iOS7 specific property.
tkchin@webrtc.org
2014-07-16
Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...
Android Chromium Automerger
2014-07-16
Remove old padding path in RTPSender.
pbos@webrtc.org
2014-07-16
nrsh1 is written before tmp321 is read, so needs to be earlyclobber
kwiberg@webrtc.org
2014-07-15
Fix an invalid memory access due to typo in win/cursor.cc.
jiayl@webrtc.org
2014-07-15
After an audio interruption the audio unit no longer invokes its render callb...
tkchin@webrtc.org
2014-07-15
Improvements to the pacer where it lost some budget due to truncation errors.
stefan@webrtc.org
2014-07-15
Fix breakage introduced by r6691.
pbos@webrtc.org
2014-07-15
Make RTCP sender report send media bytes.
pbos@webrtc.org
2014-07-15
Remove the VPM denoiser.
pbos@webrtc.org
2014-07-14
Fix deadlock in Android stopCapture() call.
glaznev@webrtc.org
2014-07-13
GN: Fix include paths for WebRTC in Chromium build.
kjellander@webrtc.org
2014-07-11
Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 .
tommi@webrtc.org
2014-07-11
Remove always-true expression.
tommi@webrtc.org
2014-07-11
Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.
tommi@webrtc.org
2014-07-11
Thread annotate RTCPSender.
pbos@webrtc.org
2014-07-11
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
stefan@webrtc.org
2014-07-11
Document that channels are stored contiguously in AudioBuffer
aluebs@webrtc.org
2014-07-11
Remove unnecessary build message.
tommi@webrtc.org
2014-07-11
Remove the send-side cname getter APIs from voice and video engine.
stefan@webrtc.org
2014-07-10
Update makefiles after merge of Chromium at 282385
Android Chromium Automerger
2014-07-10
Count total bytes sent in RTPSender::Bytes().
pbos@webrtc.org
2014-07-10
Fix data race in VCMTiming::ResetDecodeTime.
pbos@webrtc.org
2014-07-10
Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module i...
andresp@webrtc.org
2014-07-10
audio_processing: Updates aec_core_sse2.c with changes made to aec_common.h
bjornv@webrtc.org
2014-07-10
Neon version of SubbandCoherence()
bjornv@webrtc.org
2014-07-10
Neon version of rftbsub_128()
bjornv@webrtc.org
2014-07-10
Revert "Remove remains of WEBRTC_NO_STL." (rev 6641).
andresp@webrtc.org
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