aconvert(1) aconvert(1)
NAME
aconvert - convert audio data
SYNOPSIS
aconvert [options] [inp] [out]
DESCRIPTION
The aconvert program is used to convert between different audio file for-
mats, sampling rates and sample sizes.
OPTIONS
aconvert accepts the following command line options:
-a Auto Gain Control (AGC) enable.
-b Byte swap input data.
-c Copy raw header information from input to output when a header is
skipped using the -j option.
-f order
The specified order will be used for the FIR filter when converting
between different sampling rates (default is 10).
-g gain The specified gain will be used between input and output files
(default is 1.0).
-j bytes
The specified number of bytes will be skipped from the input file
for raw input data (default is 0).
-m units
No more than the specified number of units will be input. A value
of 0 implies no limit. This number is also used to generate a
specific number of samples for ``fake'' inputs (sweep, tone and
white).
-s Output statistics on standard error.
-v Enable silence compression of input (vox). This will only work on
single channel input.
INP/OUT
The inp and out arguments are of the form:
[name][,option...]
Where name may be the desired filename or - for stdin/stdout (default is
stdin/stdout if the name is left off). option may be any of:
b[its]=num Specifies bit width of 1 sample. Default is 8.
c[hans]=num[:mix...]
Specifies input/output channels. num is the total number of
channels either feeding the input or feeding the output
(after input mixing). Each :mix entry specifies how the
input channels are mixed into an output channel.
For example: chans=4:12:34 specifies that there are 4 chan-
nels which will be mixed down to 2 channels. The first
will be mixed from channels 1 and 2, while the second will
be mixed from channels 3 and 4. If you only specify the num
value on input, all of the channels will be preserved (i.e.
c=4 is the same as c=4:1:2:3:4).
If you specify more channels on output than on input, chan-
nels will be duplicated. For example: chans=2:1:2:12 will
take a stero stream add a third channel which is the mix-
ture of streams 1 and 2. Whenever possible, down-mix/drop
any channels on input (rather than on output)... this is
MUCH more efficient.
If you do not specify the number of channels on output, all
channels will be mixed into a single output channel. The
default is 1 channel in and one channel out.
e[xpon]=num FFT Peak enhancement factor. Default is 0.6
[f[ormat]=]fmt File format. Legal values are:
ascii data is read/written in ASCII. 8bit data is in hex, 16bit
data is in decimal and FFT vectors are in floating point.
htk Hidden Markov Toolkit format.
raw no header (default).
sphere NIST standard format (e.g., Timit database).
i[incr]=samples Specifies increment between FFT windows. Default is 160.
p[arams]=num Specifies number of parameters (coefficients) per FFT vec-
tor. Default is 11. This number actually includes Energy +
Peak-to-Peak + Zero-Crossings + Actual-Coeffs.
[r[ate]=]frequency
Sample frequency in kilohertz. Default is 8.0.
s[amples]=samples
Specifies number of samples in an FFT window. Default is
320.
[t[ype]=]type Type of data as listed below:
Type Inp/Out Description
adpcm I/O 2,3 or [4] bits
alaw I/O 8 bit data
cepstrum O ener+ptp+zc+coeffs
ima I/O 4 bits
linear I/O 16 bit data
melcep O ener+ptp+zc+coeffs
plp O ener+ptp+zc+coeffs
rasta O ener+ptp+zc+coeffs
sweep I fake linear for -m samples
tone I fake linear for -m samples
ulaw I/O 8 bit data
white I fake linear for -m samples
EXAMPLES
Here are different ways to read from an 8 bit, 8khz, raw ULAW file inp.snd
and create a 16 bit, 44.1khz, linear SPHERE format stereo audio file
out.snd:
aconvert inp.snd,t=ulaw,r=8 out.au,t=linear,f=sphere,r=44.1,c=1:1:1
aconvert inp.snd,ulaw,8 out.au,linear,sphere,44.1,c=1:1:1
aconvert inp.snd out.au,linear,sphere,44.1,c=1:1:1
aconvert inp.snd -,linear,sphere,44.1,c=1:1:1 >out.au
aconvert - -,linear,sphere,44.1,c=1:1:1 <inp.snd >out.au
RETURN VALUE
none specified.
SEE ALSO
AF(1)
BUGS
melcep and cepstrum have not been implemented yet.
COPYRIGHT
Copyright 1993-1994, Digital Equipment Corporation.
See AF(1) for a full statement of rights and permissions.
AUTHORS
Dave Wecker, Cambridge Research Lab, Digital Equipment Corporation.