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This is a setup to solve

Browse Source 
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
This commit is contained in:
minyue@webrtc.org
2014-08-04 14:41:57 +00:00
parent 84b9e1e9d9
commit 0040a6ef97
5 changed files with 110 additions and 16 deletions
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21
webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
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@ -72,6 +72,27 @@ int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate);
*/
int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate);
/****************************************************************************
* WebRtcOpus_SetMaxBandwidth(...)
*
* Configures the maximum bandwidth for encoding. This can be taken as a hint
* about the maximum output bandwidth that the receiver is capable to render,
* due to hardware limitations. Sending signals with higher audio bandwidth
* results in higher than necessary network usage and encoding complexity.
*
* Input:
* - inst : Encoder context
* - bandwidth : Maximum encoding bandwidth in Hz.
* This parameter can take any value, but values
* other than Opus typical bandwidths: 4000, 6000,
* 8000, 12000, and 20000 will be rounded up (values
* greater than 20000 will be rounded down) to
* these values.
* Return value : 0 - Success
* -1 - Error
*/
int16_t WebRtcOpus_SetMaxBandwidth(OpusEncInst* inst, int32_t bandwidth);
/* TODO(minyue): Check whether an API to check the FEC and the packet loss rate
* is needed. It might not be very useful since there are not many use cases and
* the caller can always maintain the states. */