dmAudioRateConverterGetParams(3dm)

NAME
     dmAudioRateConverterGetParams - get rate converter parameter values


SYNOPSIS
     #include <dmedia/dm_audioutil.h>
     #include <dmedia/dm_audio.h>
     #include <dmedia/dm_params.h>

     DMstatus dmAudioRateConverterGetParams(DMaudiorateconverter handle,
                                   DMparams *params)


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


     params   List of parameters for query.


RETURNED VALUE
     Returns DM_SUCCESS or DM_FAILURE.


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


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

     DM_AUDIO_RC_ATOMIC_IN_LENGTH
     DM_AUDIO_RC_ATOMIC_OUT_LENGTH
     DM_AUDIO_RC_GROUP_DELAY

     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 DM_AUDIO_RC_ATOMIC_IN_LENGTH and
     DM_AUDIO_RC_ATOMIC_OUT_LENGTH, respectively.  See further description in
     dmAudioRateConvert(3dm).

     DM_AUDIO_RC_GROUP_DELAY 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
     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).