aifcresample(1) aifcresample(1)
aifcresample  perform sampling rate conversion on an AIFFC file
aifcresample [options] rate rate infile.aifc outfile.aifc
rate rate
specifies the sampling rate for the output file outfile.aifc. The
value rate must be one of the following (sample frames per second):
8000 11025 16000 22050 32000 44100 48000
The sampling rate specified for the output file must be different
from the sampling rate of the input file.
dynamic decibels
specifies the minimum attenuation (in decibels) of the alias/image
artifacts generated by the sampling rate conversion. The loss in
dynamic range due to the rate conversion process will be no worse
than this value.
The value for decibels must be one of: 78, 96, or 120. Default
value is 96. A higher value for decibels gives higher quality rate
conversion.
taper bandwidth
specifies the percentage of the final bandwidth tapered off. A
smaller percentage of tapered bandwidth corresponds to a greater
high frequency content. Conversions to low sampling rates should
use higher values for bandwidth.
The value for bandwidth must be chosen from: 1, 10, or 20. Default
value is 10. A lower value for bandwidth gives higher quality rate
conversion.
verbose
causes the program to print out messages periodically which
indicate how much of the data in the original file has been
converted.
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aifcresample(1) aifcresample(1)
aifcresample is a commandline program which reads an AIFFC (or AIFF)
digital audio input file, converts the audio data to new sampling rate,
and writes out a new AIFFC file containing the resampled data.
aifcresample uses highquality rate conversion algorithms with linear
phase filters to perform resampling between any two of the following
rates: 8000, 11025, 16000, 22050, 32000, 44100, 48000 sample frames per
second (or Hz).
The speed of the rate conversion algorithm depends on the values
specified for bandwidth, decibels, and the original and new sampling
rates.
If the input AIFFC file parses correctly, and the output file is written
with no errors, aifcresample returns 0. If there is an error,
aifcresample returns 1.
The AIFFC file format specification is published by Apple Computer Inc.
Files converted using 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
tapered
20 20% +/ 0.05 dB
10 10% +/ 0.05 dB
1 1% +/ 0.0005 dB
Due to convolution remnants at the beginning of the output file's audio
data, there may be some inaccuracy in rescaled loop points or markers.
The leading convolution remnant may displace some valid audio data at the
end of the file. The magnitude of this inaccuracy depends on the rate
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aifcresample(1) aifcresample(1)
conversion bandwidth parameters and input and output sampling rates.
aifcresample simply multiplies existing loop points by the ratio
output/input sampling rate.
Gints Klimanis and Scott Porter, Silicon Graphics Inc.
playaifc(1), recordaifc(1),
aifc2aiff(1), aifccompress(1), aifcdecompress(1),
dmconvert(1), mediaconvert(1)
AIFFC File Format Specification
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