dmAudioRateConverterGetParams(3dm)	    dmAudioRateConverterGetParams(3dm)

NAME    [Toc]    [Back]

     dmAudioRateConverterGetParams - get rate converter	parameter values

SYNOPSIS    [Toc]    [Back]

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

     DMstatus dmAudioRateConverterGetParams(DMaudiorateconverter handle,
				   DMparams *params)

PARAMETER    [Toc]    [Back]

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


     params   List of parameters for query.

RETURNED VALUE    [Toc]    [Back]

     Returns DM_SUCCESS	or DM_FAILURE.

DESCRIPTION    [Toc]    [Back]

     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    [Toc]    [Back]
     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



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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    [Toc]    [Back]

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


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