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afSetConversionParams(3dm)			    afSetConversionParams(3dm)


NAME    [Toc]    [Back]

     afSetConversionParams, afGetConversionParams - set/get the	parameters
     associated	with format conversion for a specified audio track via
     dmParams

SYNOPSIS    [Toc]    [Back]

     #include <dmedia/dm_audio.h>
     #include <dmedia/audiofile.h>

     DMstatus afSetConversionParams(AFfilehandle file, int track,
				       DMparams	*params)

     DMstatus afGetConversionParams(AFfilehandle file, int track,
				       DMparams	*params)

PARAMETER    [Toc]    [Back]

     file     is an AFfilehandle structure, previously created by a call to
	      afOpenFile(3dm) or its equivalent.

     track    is an integer which identifies an	audio track in file.

	      Since all	currently supported file formats contain only one
	      audio track, the value AF_DEFAULT_TRACK should be	used here for
	      now.

     params   a	DMparams list, previously created by a call to
	      dmParamsCreate(3dm).

RETURN VALUE    [Toc]    [Back]

     afSetConversionParams() and afGetConversionParams() return	DM_SUCCESS on
     success, otherwise	they return DM_FAILURE.

DESCRIPTION    [Toc]    [Back]

     When some type of format conversion takes place during afReadFrames() or
     afWriteFrames() due to a difference between the track and the virtual
     format, the precise mechanism for this conversion may be controlled via a
     set of parameters.

     afSetConversionParams() sets all parameters associated with the
     conversion	of audio data to or from an audio track	from or	to an audio
     buffer.

     afGetConversionParams() allows an application to retrieve the complete
     set of conversion parameters currently contained within an	AFfilehandle
     structure for a given audio track.

     The following parameters are defined in dmedia/dm_audio.h and may be set
     or	retrieved via params:

     DM_AUDIO_RC_ALGORITHM    [Toc]    [Back]
	     The converson algorithm to	be used	if rate	conversion is being
	     performed.	 Its possible values are:



									Page 1






afSetConversionParams(3dm)			    afSetConversionParams(3dm)



     DM_AUDIO_RC_JITTER_FREE   Jitter-Free is the highest quality rate
			       conversion available, and is the	default	for
			       all sampling rate ratios	which it can support.
			       This includes all standard rate conversions
			       (i.e., 44.1k to 8k, 48k to 22.05k, etc.).

     DM_AUDIO_RC_POLYNOMIAL_ORDER_3    [Toc]    [Back]
			       A third-order polynomial	interpolating
			       algorithm.  It is the default for all sampling
			       rate ratios not covered by Jitter-Free.

     DM_AUDIO_RC_POLYNOMIAL_ORDER_1    [Toc]    [Back]
			       A simple	linear-interpolating algorithm.	 This
			       is the least CPU-intensive, but also generated
			       the most	distortion.

     DM_AUDIO_RC_JITTER_FREE_STOPBAND_ATTENUATION    [Toc]    [Back]
	     The parameter is specific to the jitter free rate converter
	     algorithm.	 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 suppression.  Supported values
	     are:

		 DM_AUDIO_RC_JITTER_FREE_STOPBAND_ATTENUATION_78_DB (default)
		 DM_AUDIO_RC_JITTER_FREE_STOPBAND_ATTENUATION_96_DB
		 DM_AUDIO_RC_JITTER_FREE_STOPBAND_ATTENUATION_120_DB

     DM_AUDIO_RC_JITTER_FREE_TRANSITION_BANDWIDTH    [Toc]    [Back]
	     This parameter is specific	to the jitter free rate	converter
	     algorithm.	 The values correspond to the percent of the high
	     frequency response	of the output signal lost in the conversion
	     process.  Lower percentages correspond to better high frequency
	     response.	Supported values are:

		 DM_AUDIO_RC_JITTER_FREE_TRANSITION_BANDWIDTH_1_PERCENT
		 DM_AUDIO_RC_JITTER_FREE_TRANSITION_BANDWIDTH_10_PERCENT
		 DM_AUDIO_RC_JITTER_FREE_TRANSITION_BANDWIDTH_20_PERCENT (default)

     DM_AUDIO_DITHER_ALGORITHM    [Toc]    [Back]
	     When performing conversions from larger into smaller data types
	     (i.e., floating point to 16-bit integer or	16-bit integer to 8-
	     bit integer), a dithering algorithm may be	applied	to reduce the
	     quantization error	distortion inherant in such conversions.  This
	     parameter is ignored if such a conversion is not happening.
	     Supported values are:

		 DM_AUDIO_DITHER_NONE (default)
		 DM_AUDIO_DITHER_LSB_TPDF (LSB TPDF dither)






									Page 2






afSetConversionParams(3dm)			    afSetConversionParams(3dm)



     DM_AUDIO_CHANNEL_MATRIX    [Toc]    [Back]
	     Allows specification of input to output channel routing via a
	     matrix of values.

	     This is a type DMfloatarray which must be set and retrieved via
	     dmParamsSetFloatArrayafSetChannelMatrix(3dm)	manual page
	     for a detailed explanation	of this.

SEE ALSO    [Toc]    [Back]

      
      
     afWriteFrames(3dm), afInitFormatParams(3dm),
     afSetVirtualFormatParams(3dm), dmACSetParams(3dm)


									PPPPaaaaggggeeee 3333
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