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channel.h
Age
Commit message (
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Author
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
2013-07-03
Proper spacing for end-of-namespace comments.
pbos@webrtc.org
2013-06-07
Fix size_t to int conversion error on Win64.
andrew@webrtc.org
2013-06-05
Wire up Nack for Voe
pwestin@webrtc.org
2013-05-29
Breaking out RTP header parsing from the RTP module.
stefan@webrtc.org
2013-05-22
API to control target delay in NetEq jitter buffer. NetEq maintains the given...
turaj@webrtc.org
2013-05-21
Include files from webrtc/.. paths in voice_engine/
pbos@webrtc.org
2013-05-14
Remove const for plain data types in voice_engine/
pbos@webrtc.org
2013-04-11
Adding playout buffer status to the voe video sync
pwestin@webrtc.org
2013-04-09
WebRtc_Word32 -> int32_t in voice_engine/
pbos@webrtc.org
2013-04-03
Remove UDP transport API from VoE
pwestin@webrtc.org
2013-03-28
Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRT...
solenberg@webrtc.org
2013-03-13
Revert r3667 and r3665
pwestin@webrtc.org
2013-03-13
Removed the engine API:s related to transport such as SetSendDestination, the...
pwestin@webrtc.org
2013-03-12
Remove DTMF detection. Talk team has been in the loop and there is no need for
turaj@webrtc.org
2013-03-05
None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does n...
turaj@webrtc.org
2013-02-12
Implement initial delay. This CL allows clients of VoE to set an initial dela...
turaj@webrtc.org
2012-12-12
Add GetAudioFrame API to VoiceEngine.
roosa@google.com
2012-12-12
Add API to retreive last received RTP timestamp to VoiceEngine.
roosa@google.com
2012-12-11
VoE Changes to enable dual_streaming.
turaj@webrtc.org
2012-12-04
Revert 3231 - VoE Changes to enable dual_streaming.
perkj@webrtc.org
2012-12-03
VoE Changes to enable dual_streaming.
turaj@webrtc.org
2012-10-22
Move src/ -> webrtc/
andrew@webrtc.org