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21cb767db7
This commit imports libresample for use in Asterisk. It also adds a new codec module, codec_resample. This module uses libresample to re-sample signed linear audio between 8 kHz and 16 kHz. It also provides an alternative for converting between 16 kHz G.722 and 8 kHz signed linear when using G.722, which will likely be useful as some people have complained about volume issues when the current codec_g722 converts to 8 kHz signed linear. But, to test this, you will have to disable the g722-to-slin and g722-to-slin16 translators in codec_g722.c. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
45 lines
1.1 KiB
C
45 lines
1.1 KiB
C
/**********************************************************************
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resample.h
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Real-time library interface by Dominic Mazzoni
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Based on resample-1.7:
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http://www-ccrma.stanford.edu/~jos/resample/
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License: LGPL - see the file LICENSE.txt for more information
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**********************************************************************/
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#ifndef LIBRESAMPLE_INCLUDED
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#define LIBRESAMPLE_INCLUDED
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#ifdef __cplusplus
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extern "C" {
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#endif /* __cplusplus */
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void *resample_open(int highQuality,
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double minFactor,
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double maxFactor);
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void *resample_dup(const void *handle);
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int resample_get_filter_width(const void *handle);
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int resample_process(void *handle,
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double factor,
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float *inBuffer,
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int inBufferLen,
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int lastFlag,
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int *inBufferUsed,
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float *outBuffer,
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int outBufferLen);
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void resample_close(void *handle);
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#ifdef __cplusplus
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} /* extern "C" */
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#endif /* __cplusplus */
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#endif /* LIBRESAMPLE_INCLUDED */
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