minyue@webrtc.org
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52bc4f4797
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Revert 7807 "Removing unused opus wrapper APIs."
> Removing unused opus wrapper APIs.
>
> WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().
>
> WebRtcOpus_DecodePlcMaster/Slave() are also removed.
>
> BUG=
> R=henrik.lundin@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/28139004
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7809 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-12-04 11:00:50 +00:00 |
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minyue@webrtc.org
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e54a6342dd
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Removing unused opus wrapper APIs.
WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().
WebRtcOpus_DecodePlcMaster/Slave() are also removed.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7807 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-12-04 08:47:25 +00:00 |
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minyue@webrtc.org
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0040a6ef97
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This is a setup to solve
https://code.google.com/p/webrtc/issues/detail?id=1906
In particular, we add an API to call Opus's set maximum bandwidth to prevent the encoder from coding audio content beyond this bandwidth so as to increase computation and transmission efficiency (without affecting sampling rate).
BUG=
R=henrik.lundin@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6817 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-08-04 14:41:57 +00:00 |
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