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webrtc_m130/webrtc/modules/audio_coding
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turaj@webrtc.org 2e6b7e938f In streaming mode it is preferable to fade to silence when sender stops sending, or long period of packet loss.
test=try bots.
Review URL: https://webrtc-codereview.appspot.com/1272004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3771 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-06 00:08:11 +00:00
..
codecs
WebRtc_Word -> stdint in audio_coding/g711/
2013-03-21 13:38:29 +00:00
main
In streaming mode it is preferable to fade to silence when sender stops sending, or long period of packet loss.
2013-04-06 00:08:11 +00:00
neteq
Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM.
2013-02-12 21:42:18 +00:00
neteq4
G722-stereo has been missing when creating AudioDecoder.
2013-03-27 20:42:48 +00:00
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