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  man pages->IRIX man pages -> dmedia/dmAudioRateConverterSetParams (3d)              


dmAudioRateConverterSetParams(3dm)	    dmAudioRateConverterSetParams(3dm)

NAME    [Toc]    [Back]

     dmAudioRateConverterSetParams - set rate converter	parameter values

SYNOPSIS    [Toc]    [Back]

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

     DMstatus dmAudioRateConverterSetParams(DMaudiorateconverter handle,
				   DMparams *params)

PARAMETER    [Toc]    [Back]

     handle   DMaudiorateconverter structure, created by

     params   List of parameters for specification/query. Null (0) value ok.

RETURNED VALUE    [Toc]    [Back]

     Returns DM_SUCCESS	or DM_FAILURE.

DESCRIPTION    [Toc]    [Back]

     dmAudioRateConverterSetParams(3dm)	sets state of converter	with params.
     Unrecognized parameters are ignored.

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

     DM_AUDIO_RC_INPUT_RATE    [Toc]    [Back]

     Audio Util	rate conversion	input, output rate parameters (rate given in
     Hz).  The corresponding values for	the rates are doubles.	The jitterfree
 algorithm supports sampling rate conversion between any two of the
     following rates in	the set	{8000, 11025, 16000, 22050, 32000, 44100,
     48000} (sample frames/second or Hertz).  Polynomial_order_{1,3}
     algorithms	support	arbitrary input	and output rate.

     DM_AUDIO_RC_ALGORITHM    [Toc]    [Back]

									Page 1

dmAudioRateConverterSetParams(3dm)	    dmAudioRateConverterSetParams(3dm)

     Currently,	the algorithms supported are jitter-free, which	is at least
     one integer interpolator /	integer	decimator stage, polynomial-order-1,
     which is linear interpolation between 2 proximal samples, and
     polynomial-order-3, which is rate convert block by	3rd order polynomial
     interpolation between four	proximal samples.


     The parameter is specific to the jitter free rate converter filter
     stopband attenuation.  The	values are limited to set {78, 96, 120}. The
     values correspond to the minimum attenuation level	of the out-of-band
     frequencies, those	frequencies which do not fit in	the smaller of the
     output or input bandwidths.  Higher values	correspond to better noise


     The parameter is specific to the jitter free rate converter filter
     transition	bandwidth.  The	values are limited to set {1, 10, 20}. The
     values correspond to the %	of the high frequency response of the output
     signal lost in the	conversion process.  Lower percentages correspond to
     better high frequency response.

     Parameters	for query only:

     DM_AUDIO_RC_ATOMIC_IN_LENGTH    [Toc]    [Back]

     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.  Query these parameters on a
     configured	converter can also be done by using


     The 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 he	filtered signal	contains more

									Page 2

dmAudioRateConverterSetParams(3dm)	    dmAudioRateConverterSetParams(3dm)

     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

     Query DM_AUDIO_RC_GROUP_DELAY on a	configured converter can also be done
     by	using dmAudioRateConverterGetParams(3dm).

NOTE    [Toc]    [Back]

     dmAudioRateConverterSetParams(3dm)	calls dmAudioRateConverterReset(3dm)

     Files converted using jitter-free algorithm with decibels set to 96 and
     bandwidth set to 10 will sound most excellent.  For the highest quality
     conversion	algorithm, however, you	can set	decibels to 120	and bandwidth
     to	1.  By decreasing decibels and increasing bandwidth, you increase the
     speed of the rate conversion at a loss of high frequency response.

     The following table shows the relationships between the bandwidth
     parameter,	and the	pass band ripple:

     bandwidth	  % final     passband
     parameter	  bandwidth   ripple

     20		  20%	      +/- 0.05	 dB
     10		  10%	      +/- 0.05	 dB
      1		  1%	      +/- 0.0005 dB

SEE ALSO    [Toc]    [Back]

     dmAudioRateConverterCreate(3dm), dmAudioRateConvert(3dm),

									PPPPaaaaggggeeee 3333
[ Back ]
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