index
:
external/chromium_org/third_party/webrtc.git
idea133-weekly-release
l-preview
lollipop-cts-release
lollipop-dev
lollipop-mr1-cts-release
lollipop-mr1-dev
lollipop-mr1-fi-release
lollipop-mr1-release
lollipop-mr1-wfc-release
lollipop-release
lollipop-wear-release
main
master
master-soong
ub-webview-m40-release
summary
refs
log
tree
commit
diff
log msg
author
committer
range
path:
root
/
voice_engine
Age
Commit message (
Expand
)
Author
2014-07-16
Raw packet loss rate reported by RTP_RTCP module may vary too drastically ove...
minyue@webrtc.org
2014-07-11
Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.
tommi@webrtc.org
2014-07-11
Remove the send-side cname getter APIs from voice and video engine.
stefan@webrtc.org
2014-07-08
Some refactoring inside rtp_rtcp/.
pbos@webrtc.org
2014-06-30
Add ExperimentalNs support in Config
aluebs@webrtc.org
2014-06-23
GN: Add BUILD.gn files + kjellander to OWNERS
kjellander@webrtc.org
2014-06-13
Pass GYP DEPTH variable to isolate.
kjellander@webrtc.org
2014-06-10
Add kjellander@webrtc.org as OWNER for *.isolate
kjellander@webrtc.org
2014-06-05
Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
wu@webrtc.org
2014-06-05
Have RTX be enabled by setting an RTX payload type instead of by setting an R...
stefan@webrtc.org
2014-06-04
Android: cleanup gtest_target_type conditions.
henrike@webrtc.org
2014-05-30
Add a Reset() method to AudioFrame.
andrew@webrtc.org
2014-05-28
This CL is to adding feedback of packet loss rate to encoder in voice engine....
minyue@webrtc.org
2014-05-23
1. Make a clear distinction between codec internal FEC and RED, confusing men...
minyue@webrtc.org
2014-05-21
Switch to using base/constructormagic.h and remove system_wrappers/interface/...
henrike@webrtc.org
2014-05-21
Revert 6202 "Switch to using base/constructormagic.h and remove ..."
mcasas@webrtc.org
2014-05-20
Calculate capture ntp timestamp in local timebase for decoded audio frame.
wu@webrtc.org
2014-05-20
Switch to using base/constructormagic.h and remove system_wrappers/interface/...
henrike@webrtc.org
2014-05-20
VoEVolumeTest: Enabled Linux flaky tests
bjornv@webrtc.org
2014-05-20
Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size
minyue@webrtc.org
2014-05-19
Add interface to propagate audio capture timestamp to the renderer.
wu@webrtc.org
2014-05-16
Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*.
solenberg@webrtc.org
2014-05-14
Remove the use of AudioFrame::energy_ from AudioProcessing and VoE.
andrew@webrtc.org
2014-05-14
Add webrtc field trials API.
andresp@webrtc.org
2014-05-14
Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation ...
henrika@webrtc.org
2014-05-14
Removes parts of the webrtc::VoEVolumeControl sub API as part of a clean-up o...
henrika@webrtc.org
2014-05-14
VoEVolumeTest: Adds error return tests.
bjornv@webrtc.org
2014-05-14
Make vie/voe_auto_test accept non-supported flags without error.
kjellander@webrtc.org
2014-05-13
Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255
bjornv@webrtc.org
2014-05-12
Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation ...
henrika@webrtc.org
2014-05-12
Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up o...
henrika@webrtc.org
2014-05-12
Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operat...
henrika@webrtc.org
2014-05-09
Removes parts of the webrtc::VoEHardware sub API (relanding)
henrika@webrtc.org
2014-05-09
Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..."
henrika@webrtc.org
2014-05-09
Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up o...
henrika@webrtc.org
2014-05-05
Allow the RTP level indicator computation to work at any sample rate.
andrew@webrtc.org
2014-05-02
Removed NetworkTest.CanSwitchToExternalTransport since it tests an unsupporte...
henrika@webrtc.org
2014-04-30
Only clamp to 16 kHz when AECM is enabled.
andrew@webrtc.org
2014-04-25
Replace scoped_array<T> with scoped_ptr<T[]>.
andrew@webrtc.org
2014-04-24
* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.
wu@webrtc.org
2014-04-23
Remove ACM1/ACM2 switching from VoiceEngine tests
henrik.lundin@webrtc.org
2014-04-22
Support arbitrary input/output rates and downmixing in AudioProcessing.
andrew@webrtc.org
2014-04-22
Reland "Stop using ACM factory in VoiceEngine"
henrik.lundin@webrtc.org
2014-04-22
Revert "Stop using ACM factory in VoiceEngine"
henrik.lundin@webrtc.org
2014-04-22
Stop using ACM factory in VoiceEngine
henrik.lundin@webrtc.org
2014-04-22
Reland "Make VoiceEngine choose ACM2 by default""
henrik.lundin@webrtc.org
2014-04-19
Resampler modifications in preparation for arbitrary audioproc rates.
andrew@webrtc.org
2014-04-17
Removes parts of the VoEBase sub API as part of a clean-up operation where th...
henrika@webrtc.org
2014-04-17
Removes VoECodec sub API as part of a clean-up operation where the goal is to...
henrika@webrtc.org
2014-04-17
Revert "Make VoiceEngine choose ACM2 by default"
henrik.lundin@webrtc.org
[next]