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
/
channel.h
Age
Commit message (
Expand
)
Author
2014-10-10
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
xians@webrtc.org
2014-10-01
Reland "Remove DTMF status methods from Voice Engine" r7276
henrik.lundin@webrtc.org
2014-09-23
Revert "Remove DTMF status methods from Voice Engine" r7276
henrik.lundin@webrtc.org
2014-09-23
Remove DTMF status methods from Voice Engine
henrik.lundin@webrtc.org
2014-09-22
Remove Get/SetNetEQPlayoutMode APIs
henrik.lundin@webrtc.org
2014-09-11
Calculating round-trip-time in send-only channel in VoE.
minyue@webrtc.org
2014-09-03
Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate
minyue@webrtc.org
2014-09-02
Fix race in Voice Engine's Channel where it accesses RemoteNtpTimeEstimator f...
stefan@webrtc.org
2014-08-12
Adding SetOpusMaxBandwidth in VoE and ACM
minyue@webrtc.org
2014-07-16
Raw packet loss rate reported by RTP_RTCP module may vary too drastically ove...
minyue@webrtc.org
2014-07-11
Remove the send-side cname getter APIs from voice and video engine.
stefan@webrtc.org
2014-06-05
Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
wu@webrtc.org
2014-05-28
This CL is to adding feedback of packet loss rate to encoder in voice engine....
minyue@webrtc.org
2014-05-20
Calculate capture ntp timestamp in local timebase for decoded audio frame.
wu@webrtc.org
2014-05-19
Add interface to propagate audio capture timestamp to the renderer.
wu@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::VoERTP_RTCP sub API as part of a clean-up operat...
henrika@webrtc.org
2014-05-05
Allow the RTP level indicator computation to work at any sample rate.
andrew@webrtc.org
2014-04-24
* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.
wu@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
2014-04-14
Removes VoECallReport sub API as part of a clean-up operation where the goal ...
henrika@webrtc.org
2014-04-03
Consolidate audio conversion from Channel and TransmitMixer.
andrew@webrtc.org
2014-03-24
VoE changes to allow forwarding of packets from VoE to ViE BWE.
solenberg@webrtc.org
2014-03-18
Resolves TSan v2 warnings in voe_auto_test.
henrika@webrtc.org
2014-03-06
Help to land 7969005 on behalf of solenberg. The review and try is done in 79...
wu@webrtc.org
2014-02-19
Removes VoERTP_RTCP::InsertExtraRTPPacket.
henrika@webrtc.org
2014-02-18
Remove external encryption API for VoE.
solenberg@webrtc.org
2014-01-07
Remove the requirement to call set_sample_rate_hz and friends.
andrew@webrtc.org
2013-12-19
Add callbacks for receive channel RTCP statistics.
sprang@webrtc.org
2013-12-13
Fix jitter buffer delay estimate.
turaj@webrtc.org
2013-12-13
Update talk to 58174641 together with http://review.webrtc.org/4319005/.
wu@webrtc.org
2013-11-08
Fix for making sure that the packet in order checks are done prior to updatin...
stefan@webrtc.org
2013-10-17
Fix tsan failures in channel.cc regarding to the volume settings.
wu@webrtc.org
2013-09-23
Remove deprecated AudioCodingModule::Destroy.
andrew@webrtc.org
2013-09-18
Small refactoring of AudioProcessing use in channel.cc.
andrew@webrtc.org
2013-09-12
This issue is related to
minyue@webrtc.org
2013-09-06
Adds support for combining RTX and FEC/RED.
stefan@webrtc.org
2013-08-21
Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a...
stefan@webrtc.org
2013-08-21
Revert 4582 "Reverts a second set of reverts caused by a bug in ..."
henrike@webrtc.org
2013-08-21
Reverts a second set of reverts caused by a bug in a dependency.
stefan@webrtc.org
2013-08-15
Update talk to 50918584.
wu@webrtc.org
2013-07-31
Merge r4374 from stable to trunk.
xians@webrtc.org
2013-07-31
Merge r4394 from stable to trunk.
xians@webrtc.org
2013-07-31
Merge r4326 from stable to trunk.
xians@webrtc.org
2013-07-16
Revert r4301
tnakamura@webrtc.org
2013-07-15
Revert r4322 "Support sending multiple report blocks and keeping track of sta...
elham@webrtc.org
2013-07-10
Support sending multiple report blocks and keeping track of statistics on sev...
stefan@webrtc.org
2013-07-05
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
stefan@webrtc.org
[next]