dmAudioRateConverterGetParams(3dm)          dmAudioRateConverterGetParams(3dm)



NAME
     dmAudioRateConverterGetParams - get rate converter parameter values


SYNOPSIS
     #include <dmedia/dmaudioutil.h>
     #include <dmedia/dmaudio.h>
     #include <dmedia/dmparams.h>

     DMstatus dmAudioRateConverterGetParams(DMaudiorateconverter handle,
                                   DMparams *params)


PARAMETER
     handle   DMaudiorateconverter structure, created by
              dmAudioRateConverterCreate(3dm).


     params   List of parameters for query.


RETURNED VALUE
     Returns DMSUCCESS or DMFAILURE.


DESCRIPTION
     dmAudioRateConverterGetParams(3dm) gets state of converter with params.


     The set of parameters of params for query are, defined in
     dmedia/dm_audioutil.h:

     DMAUDIORCATOMICINLENGTH
     DMAUDIORCATOMICOUTLENGTH
     DMAUDIORCGROUPDELAY

     The rate convert algorithm processes blocks of a fixed length determined
     by the conversion process parameters.  The input and output lengths must
     be a multiple of DMAUDIORCATOMICINLENGTH and
     DMAUDIORCATOMICOUTLENGTH, respectively.  See further description in
     dmAudioRateConvert(3dm).

     DMAUDIORCGROUPDELAY measured in output samples.  These sampling rate
     conversion algorithms use filter operations that convolve a N past input
     samples with a filter M-coefficient array to create N+M-1 output samples.
     Note that the filtered signal contains more samples than the unfiltered
     signal.

     We use linear phase filters with a constant group delay G = (M-1)/2
     samples.  In real time operation, convolution simply delays the output by
     G samples.  In file conversion expecting N samples in and out, the output
     signal is offset by G samples from the first sample thus omits the last G



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dmAudioRateConverterGetParams(3dm)          dmAudioRateConverterGetParams(3dm)



     samples from the output file.  Multiple conversions compound the delays
     and data loss.

     SGI audio applications manage the above problems by omitting the first G
     samples while computing more than N+G samples.  In practice, G is rounded
     to the nearest sample and thus sample time alignment is occasionally
     ahead or behind one sample.  G specifies the number of output samples to
     omit.


NOTE
SEE ALSO
     dmAudioRateConverterCreate(3dm), dmAudioRateConverterSetParams(3dm),
     dmAudioRateConvert(3dm).









































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