minyue@webrtc.org
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74aaf29a0f
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Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.
The filter is an exponential filter borrowed from video coding module.
The method is written in a new class called PacketLossProtector (not sure if the name is nice), which can be used in the future for more sophisticated logic.
BUG=
R=henrika@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6709 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-07-16 21:28:26 +00:00 |
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pbos@webrtc.org
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034f004a4f
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WebRtc_Word32 => int32_t in video_coding/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1203008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3778 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-08 11:13:29 +00:00 |
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stefan@webrtc.org
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84cd8e39cf
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Disable frame dropper for screenshare mode.
BUG=1466
Review URL: https://webrtc-codereview.appspot.com/1170004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3629 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-03-07 13:12:32 +00:00 |
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stefan@webrtc.org
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eb91792cfd
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Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings.
Review URL: https://webrtc-codereview.appspot.com/1105007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3528 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-02-18 14:40:18 +00:00 |
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