sox.1

linux下录音程序

or
.I -t sunau -U -c 1 /dev/audio
for older sun equipment.
.TP 10
.B .txw
Yamaha TX-16W sampler.
.br
A file format from a Yamaha sampling keyboard which wrote IBM-PC
format 3.5" floppies.  Handles reading of files which do not have
the sample rate field set to one of the expected by looking at some
other bytes in the attack/loop length fields, and defaulting to
33kHz if the sample rate is still unknown.
.TP 10
.B .vms
More info to come.
.br
Used to compress speech audio for applications such as voice mail.
.TP 10
.B .voc
Sound Blaster VOC files.
.br
VOC files are multi-part and contain silence parts, looping, and
different sample rates for different chunks.
On input, the silence parts are filled out, loops are rejected,
and sample data with a new sample rate is rejected.
Silence with a different sample rate is generated appropriately.
On output, silence is not detected, nor are impossible sample rates.
.TP 10
.B .wav
Microsoft .WAV RIFF files.
.br
These appear to be very similar to IFF files,
but not the same.  
They are the native sound file format of Windows.
(Obviously, Windows was of such incredible importance
to the computer industry that it just had to have its own 
sound file format.)
Normally \fB.wav\fR files have all formatting information
in their headers, and so do not need any format options
specified for an input file. If any are, they will
overide the file header, and you will be warned to this effect.
You had better know what you are doing! Output format
options will cause a format conversion, and the \fB.wav\fR
will written appropriately.  Note that it is possible to
write data of a type that cannot be specified by
the \fB.wav\fR header, and you will be warned that
you a writing a bad file !
Sox currently can read PCM, ULAW, ALAW, MS ADPCM, and IMA (or DVI) ADPCM.
It can output all of these formats except the ADPCM styles.
.TP 10
.B .wve
Psion 8-bit alaw
.br
These are 8-bit a-law 8khz sound files used on the
Psion palmtop portable computer.
.TP 10
.B .raw
Raw files (no header).
.br
The sample rate, size (byte, word, etc), 
and style (signed, unsigned, etc.)
of the sample file must be given.
The number of channels defaults to 1.
.TP 10
.B ".ub, .sb, .uw, .sw, .ul"
These are several suffices which serve as
a shorthand for raw files with a given size and style.
Thus, \fBub, sb, uw, sw,\fR and \fBul\fR
correspond to "unsigned byte", "signed byte",
"unsigned word", "signed word", and "ulaw" (byte).
The sample rate defaults to 8000 hz if not explicitly set,
and the number of channels (as always) defaults to 1.
There are lots of Sparc samples floating around in u-law format
with no header and fixed at a sample rate of 8000 hz.
(Certain sound management software cheerfully ignores the headers.)
Similarly, most Mac sound files are in unsigned byte format with
a sample rate of 11025 or 22050 hz.
.TP 10
.B .auto
This is a ``meta-type'': specifying this type for an input file
triggers some code that tries to guess the real type by looking for
magic words in the header.  If the type can't be guessed, the program
exits with an error message.  The input must be a plain file, not a
pipe.  This type can't be used for output files.
.SH EFFECTS
Only one effect from the palette may be applied to a sound sample.
To do multiple effects you'll need to run 
.I sox 
in a pipeline.
.TP 10
avg [ \fI-l\fR | \fI-r\fR ]
Reduce the number of channels by averaging the samples,
or duplicate channels to increase the number of channels.
Valid combinations are 1 - 2, 1 - 4, 2 - 4, 4 - 2, 4 - 1,
2 - 1. The \fI-l\fR or \fI-r\fR option averages from
just left or right channels/duplicates to just the left
or right channels.
.TP 10
band \fB[ \fI-n \fB] \fIcenter \fB[ \fIwidth\fB ]
Apply a band-pass filter.
The frequency response drops logarithmically
around the
.I center
frequency.
The
.I width
gives the slope of the drop.
The frequencies at 
.I "center + width"
and
.I "center - width"
will be half of their original amplitudes.
.B Band
defaults to a mode oriented to pitched signals,
i.e. voice, singing, or instrumental music.
The 
.I -n
(for noise) option uses the alternate mode
for un-pitched signals.
.B Band
introduces noise in the shape of the filter,
i.e. peaking at the 
.I center
frequency and settling around it.
.TP
chorus \fIgain-in gain-out delay decay speed deptch 
.TP 10
       -s \fR| \fI-t [ \fIdelay decay speed depth -s \fR| \fI-t ... \fR]
Add a chorus to a sound sample.  Each quadtuple
delay/decay/speed/depth gives the delay in milliseconds
and the decay (relative to gain-in) with a modulation
speed in Hz using depth in milliseconds.
The modulation is either sinodial (-s) or triangular
(-t).  Gain-out is the volume of the output.
.TP 10
copy
Copy the input file to the output file.
This is the default effect if both files have the same 
sampling rate, or the rates are "close".
.TP 10
cut \fIloopnumber
Extract loop #N from a sample.
.TP 10
deemph
Apply a treble attenuation shelving filter to samples in
audio cd format.  The frequency response of pre-emphasized
recordings is rectified.  The filtering is defined in the
standard document ISO 908.
.TP 10
echo \fIgain-in gain-out delay decay \fR[ \fIdelay decay ... \fR]
Add echoing to a sound sample.
Each delay/decay part gives the delay in milliseconds 
and the decay (relative to gain-in) of that echo.
Gain-out is the volume of the output.
.TP 10
echos \fIgain-in gain-out delay decay \fR[ \fIdelay decay ... \fR]
Add a sequence of echos to a sound sample.
Each delay/decay part gives the delay in milliseconds 
and the decay (relative to gain-in) of that echo.
Gain-out is the volume of the output.
.TP 10
flanger \fIgain-in gain-out delay decay speed -s \fR| \fI-t
Add a flanger to a sound sample.  Each triple
delay/decay/speed gives the delay in milliseconds
and the decay (relative to gain-in) with a modulation
speed in Hz.
The modulation is either sinodial (-s) or triangular
(-t).  Gain-out is the volume of the output.
.TP 10
highp \fIcenter
Apply a high-pass filter.
The frequency response drops logarithmically with 
.I center
frequency in the middle of the drop.
The slope of the filter is quite gentle.
.TP 10
lowp \fIcenter
Apply a low-pass filter.
The frequency response drops logarithmically with 
.I center
frequency in the middle of the drop.
The slope of the filter is quite gentle.
.TP 10
map 
Display a list of loops in a sample,
and miscellaneous loop info.
.TP 10
mask
Add "masking noise" to signal.
This effect deliberately adds white noise to a sound 
in order to mask quantization effects,
created by the process of playing a sound digitally.
It tends to mask buzzing voices, for example.
It adds 1/2 bit of noise to the sound file at the
output bit depth.
.TP 10
phaser \fIgain-in gain-out delay decay speed -s \fR| \fI-t
Add a phaser to a sound sample.  Each triple
delay/decay/speed gives the delay in milliseconds
and the decay (relative to gain-in) with a modulation
speed in Hz.
The modulation is either sinodial (-s) or triangular
(-t).  The decay should be less than 0.5 to avoid
feedback.  Gain-out is the volume of the output.
.TP 10
pick
Select the left or right channel of a stereo sample,
or one of four channels in a quadrophonic sample.
.TP
polyphase [ \fI-w \fR< \fInum\fR / \fIham\fR > ] 
.TP
          [ \fI -width \fR< \fI long \fR / \fIshort \fR / \fI# \fR> ] 
.TP 10
          [ \fI-cutoff # \fR ]
Translate input sampling rate to output sampling rate via polyphase
interpolation, a DSP algorithm.  This method is slow and uses lots
of RAM, but gives much better results then 
.B rate.
.br
-w < nut / ham > : select either a Nuttal (~90 dB stopband) or Hamming
(~43 dB stopband) window.
.B Warning:
Nuttall windows require 2x length than Hamming windows.  Default is
.I nut.
.br
-width long / short / # : specify the width of the filter.
.I long
is 1024 samples;
.I short
is 128 samples.  Alternatively, an exact number can be used.  Default is
.I long.
.br
-cutoff # : specify the filter cutoff frequency in terms of fraction of
bandwidth.  If upsampling, then this is the fraction of the orignal signal
that should go through.  If downsampling, this is the fraction of the
signal left after downsampling.  Default is 0.95.  Remember that
this is a float.

.TP 10
rate
Translate input sampling rate to output sampling rate
via linear interpolation to the Least Common Multiple
of the two sampling rates.
This is the default effect 
if the two files have different sampling rates.
This is fast but noisy:
the spectrum of the original sound will be shifted upwards
and duplicated faintly when up-translating by a multiple.
Lerp-ing is acceptable for cheap 8-bit sound hardware,
but for CD-quality sound you should instead use either
.B resample
or
.B polyphase.
If you are wondering which of
.B Sox's
rate changing effects to ues, you will want to read a
detailed analysis of all of them at http://usa.ece.cmu.edu/Sox/
.TP 10
resample [ \fIrolloff\fR [ \fIbeta\fR ] ]
Translate input sampling rate to output sampling rate
via simulated analog filtration.
This method is slow and uses lots of RAM,
but gives much better results then
.B rate 
(This has empirically been shown to be false.  The
resample algorthym needs to be updated from its original source).
.TP 10
reverb \fIgain-out delay \fR[ \fIdelay ... \fR]
Add reverbation to a sound sample.  Each delay is given 
in milliseconds and its feedback is depending on the
reverb-time in milliseconds.  Each delay should be in 
the range of half to quarter of reverb-time to get
a realistic reverbation.  Gain-out is the volume of the
output.
.TP 10
reverse 
Reverse the sound sample completely.
Included for finding Satanic subliminals.
.TP 10
split
Turn a mono sample into a stereo sample by copying
the input channel to the left and right channels.
.TP 10
stat [ debug | -v ]
Do a statistical check on the input file,
and print results on the standard error file.
.B stat
may copy the file untouched from input to output,
if you select an output file.  
The "Volume Adjustment:" field in the statistics
gives you the argument to the
.B -v
.I number
which will make the sample as loud as possible without clipping. 
There is an optional parameter
.B -v
that will print out the "Volume Adjustment:" field's value and
return.  This could be of use in scripts to auto convert the
volume.  There is an also an optional parameter
.B debug
that will place sox into debug mode and print out a hex dump of the
sound file from the internal buffer that is in 32-bit signed PCM data.
This is mainly only of use in tracking down endian problems that
creep in to sox on cross-platform versions.
.TP 10
vibro \fIspeed \fB [ \fIdepth\fB ]
Add the world-famous Fender Vibro-Champ sound
effect to a sound sample by using
a sine wave as the volume knob.
.B Speed 
gives the Hertz value of the wave.
This must be under 30.
.B Depth
gives the amount the volume is cut into
by the sine wave,
ranging 0.0 to 1.0 and defaulting to 0.5.
.P
.I Sox
enforces certain effects.
If the two files have different sampling
rates, the requested effect must be one of
.B copy,
or
.B rate,
." or
." .B resample.
If the two files have different numbers of channels,
the 
.B avg
." or other channel mixing
effect must be requested.
.SH BUGS
The syntax is horrific.
It's very tempting to include a default system that allows
an effect name as the program name
and just pipes a sound sample from standard input 
to standard output, but the problem of inputting the
sample rates makes this unworkable.
.P
Please report any bugs found in this version of sox to Chris Bagwell ([e-mail])
.SH FILES
.SH SEE ALSO
.BR play (1) ,
.BR rec (1)
.SH NOTICES
The echoplex effect is:
Copyright (C) 1989 by Jef Poskanzer.

Permission to use, copy, modify, and distribute this software and its
documentation for any purpose and without fee is hereby granted, provided
that the above copyright notice appear in all copies and that both that
copyright notice and this permission notice appear in supporting
documentation.  This software is provided "as is" without express or
implied warranty.

The version of Sox that accompanies this manual page is support by 
Chris Bagwell ([e-mail]).  Please refer any questions 
regarding it to this address.  You may obtain the latest version at the 
the web site http://home.sprynet.com/sprynet/cbagwell/projects.html