ref: b3524a9a9831d9d3dd49bfa8ee1d3ad55c1ff2a2
parent: 350589c6b0fc22600f7e18d46ed798e0f9ca03dd
author: cbagwell <cbagwell>
date: Mon Jul 24 20:50:39 EDT 2000
Syncing up txt manual pages with latest nroff source.
--- a/sox.txt
+++ b/sox.txt
@@ -52,7 +52,7 @@
[ -width < long / short / # > ]
[ -cutoff # ]
rate
- resample
+ resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
reverb gain-out reverb-time delay [ delay ... ]
reverse
speed factor
@@ -61,7 +61,7 @@
- December 10, 1999 1
+ July 24, 2000 1
@@ -71,7 +71,7 @@
stretch [ factor [ window fade shift fading ]
- swap [ 1 2 3 4 ]
+ swap [ 1 2 | 1 2 3 4 ]
vibro speed [ depth ]
vol gain [ type ]
@@ -127,7 +127,7 @@
- December 10, 1999 2
+ July 24, 2000 2
@@ -139,11 +139,11 @@
Format options:
Format options effect the audio samples that they immedi-
- ately percede. If they are placed before the input file
+ ately precede. If they are placed before the input file
name then they effect the input data. If they are placed
before the output file name then they will effect the out-
put data. By taking advantage of this, you can override a
- input file's currupted header or produce an output file
+ input file's corrupted header or produce an output file
that is totally different style then the input file.
-t filetype
@@ -193,7 +193,7 @@
- December 10, 1999 3
+ July 24, 2000 3
@@ -208,7 +208,7 @@
quad sound data. To cause the output file to
have a different number of channels than the
input file, include this option with the appro-
- raite value with the output file options. If
+ priate value with the output file options. If
the input and output file have a different num-
ber of channels then the avg effect must be
used. If the avg effect is not specified on the
@@ -246,20 +246,20 @@
FILE TYPES
SoX uses the file extension of the input and output file
to determine what type of file format to use. This can be
- overriden by specifying the "-t" option on the command
+ overridden by specifying the "-t" option on the command
line.
The input and output files may be read from standard in
- and out. This is done by specifing '-' as the filename.
+ and out. This is done by specifying '-' as the filename.
- File formats which have headers are checked, if that
- header doesn't seem right, the program exits with an
+ File formats which have headers are checked, if that
+ header doesn't seem right, the program exits with an
appropriate message.
- December 10, 1999 4
+ July 24, 2000 4
@@ -271,127 +271,127 @@
The following file formats are supported:
- .8svx Amiga 8SVX musical instrument description for-
+ .8svx Amiga 8SVX musical instrument description for-
mat.
- .aiff AIFF files used on Apple IIc/IIgs and SGI.
- Note: the AIFF format supports only one SSND
+ .aiff AIFF files used on Apple IIc/IIgs and SGI.
+ Note: the AIFF format supports only one SSND
chunk. It does not support multiple sound
- chunks, or the 8SVX musical instrument descrip-
+ chunks, or the 8SVX musical instrument descrip-
tion format. AIFF files are multimedia archives
- and and can have multiple audio and picture
- chunks. You may need a separate archiver to
+ and and can have multiple audio and picture
+ chunks. You may need a separate archiver to
work with them.
.au SUN Microsystems AU files. There are apparently
- many types of .au files; DEC has invented its
- own with a different magic number and word
+ many types of .au files; DEC has invented its
+ own with a different magic number and word
order. The .au handler can read these files but
- will not write them. Some .au files have valid
- AU headers and some do not. The latter are
- probably original SUN u-law 8000 hz samples.
- These can be dealt with using the .ul format
+ will not write them. Some .au files have valid
+ AU headers and some do not. The latter are
+ probably original SUN u-law 8000 hz samples.
+ These can be dealt with using the .ul format
(see below).
.avr Audio Visual Research
- The AVR format is produced by a number of com-
+ The AVR format is produced by a number of com-
mercial packages on the Mac.
.cdr CD-R
- CD-R files are used in mastering music Compact
- Disks. The file format is, as you might expect,
- raw stereo raw unsigned samples at 44khz. But,
- there's some blocking/padding oddity in the for-
- mat, so it needs its own handler.
+ CD-R files are used in mastering music on Com-
+ pact Disks. The audio data on a CD-R disk is a
+ raw audio file with a format of stereo 16-bit
+ signed samples at a 44khz sample rate. There is
+ a special blocking/padding oddity at the end of
+ the audio file and is why it needs its own han-
+ dler.
.cvs Continuously Variable Slope Delta modulation
- Used to compress speech audio for applications
+ Used to compress speech audio for applications
such as voice mail.
.dat Text Data files
- These files contain a textual representation of
- the sample data. There is one line at the
+ These files contain a textual representation of
+ the sample data. There is one line at the
beginning that contains the sample rate. Subse-
- quent lines contain two numeric data items: the
- time since the beginning of the sample and the
+ quent lines contain two numeric data items: the
+ time since the beginning of the sample and the
sample value. Values are normalized so that the
- maximum and minimum are 1.00 and -1.00. This
+ maximum and minimum are 1.00 and -1.00. This
file format can be used to create data files for
external programs such as FFT analyzers or graph
- routines. SoX can also convert a file in this
- format back into one of the other file formats.
+ routines. SoX can also convert a file in this
+ format back into one of the other file formats.
- .gsm GSM 06.10 Lossy Speech Compression
+ July 24, 2000 5
- December 10, 1999 5
-
SoX(1) SoX(1)
- A standard for compressing speech which is used
- in the Global Standard for Mobil telecommunica-
- tions (GSM). Its good for its purpose, shrink-
- ing audio data size, but it will introduce lots
- of noise when a given sound sample is encoded
+ .gsm GSM 06.10 Lossy Speech Compression
+ A standard for compressing speech which is used
+ in the Global Standard for Mobil telecommunica-
+ tions (GSM). Its good for its purpose, shrink-
+ ing audio data size, but it will introduce lots
+ of noise when a given sound sample is encoded
and decoded multiple times. This format is used
- by some voice mail applications. It is rather
- CPU intensive. GSM in sox is optional and
- requires access to an external GSM library. To
- see if there is support for gsm run sox -h and
- look for it under the list of supported file
+ by some voice mail applications. It is rather
+ CPU intensive. GSM in sox is optional and
+ requires access to an external GSM library. To
+ see if there is support for gsm run sox -h and
+ look for it under the list of supported file
formats.
- .hcom Macintosh HCOM files. These are (apparently)
+ .hcom Macintosh HCOM files. These are (apparently)
Mac FSSD files with some variant of Huffman com-
- pression. The Macintosh has wacky file formats
- and this format handler apparently doesn't han-
+ pression. The Macintosh has wacky file formats
+ and this format handler apparently doesn't han-
dle all the ones it should. Mac users will need
- your usual arsenal of file converters to deal
+ your usual arsenal of file converters to deal
with an HCOM file under Unix or DOS.
.maud An Amiga format
An IFF-conform sound file type, registered by MS
- MacroSystem Computer GmbH, published along with
- the "Toccata" sound-card on the Amiga. Allows
- 8bit linear, 16bit linear, A-Law, u-law in mono
+ MacroSystem Computer GmbH, published along with
+ the "Toccata" sound-card on the Amiga. Allows
+ 8bit linear, 16bit linear, A-Law, u-law in mono
and stereo.
ossdsp OSS /dev/dsp device driver
This is a pseudo-file type and can be optionally
- compiled into Sox. Run sox -h to see if you
- have support for this file type. When this
- driver is used it allows you to open up the OSS
- /dev/dsp file and configure it to use the same
- data type as passed in to Sox. It works for
- both playing and recording sound samples. When
- playing sound files it attempts to set up the
- OSS driver to use the same format as the input
- file. It is suggested to always override the
+ compiled into Sox. Run sox -h to see if you
+ have support for this file type. When this
+ driver is used it allows you to open up the OSS
+ /dev/dsp file and configure it to use the same
+ data type as passed in to Sox. It works for
+ both playing and recording sound samples. When
+ playing sound files it attempts to set up the
+ OSS driver to use the same format as the input
+ file. It is suggested to always override the
output values to use the highest quality samples
- your sound card can handle. Example: -t ossdsp
+ your sound card can handle. Example: -t ossdsp
-w -s /dev/dsp
.sf IRCAM Sound Files.
- SoundFiles are used by academic music software
- such as the CSound package, and the MixView
+ Sound Files are used by academic music software
+ such as the CSound package, and the MixView
sound sample editor.
.smp Turtle Beach SampleVision files.
- SMP files are for use with the PC-DOS package
- SampleVision by Turtle Beach Softworks. This
- package is for communication to several MIDI
- samplers. All sample rates are supported by the
- package, although not all are supported by the
+ SMP files are for use with the PC-DOS package
+ SampleVision by Turtle Beach Softworks. This
+ package is for communication to several MIDI
+ samplers. All sample rates are supported by the
- December 10, 1999 6
+ July 24, 2000 6
@@ -400,64 +400,64 @@
SoX(1) SoX(1)
- samplers themselves. Currently loop points are
+ package, although not all are supported by the
+ samplers themselves. Currently loop points are
ignored.
sunau Sun /dev/audio device driver
This is a pseudo-file type and can be optionally
- compiled into Sox. Run sox -h to see if you
- have support for this file type. When this
- driver is used it allows you to open up a Sun
+ compiled into Sox. Run sox -h to see if you
+ have support for this file type. When this
+ driver is used it allows you to open up a Sun
/dev/audio file and configure it to use the same
- data type as passed in to Sox. It works for
- both playing and recording sound samples. When
- playing sound files it attempts to set up the
+ data type as passed in to Sox. It works for
+ both playing and recording sound samples. When
+ playing sound files it attempts to set up the
audio driver to use the same format as the input
- file. It is suggested to always override the
+ file. It is suggested to always override the
output values to use the highest quality samples
- your hardware can handle. Example: -t sunau -w
+ your hardware can handle. Example: -t sunau -w
-s /dev/audio or -t sunau -U -c 1 /dev/audio for
older sun equipment.
.txw Yamaha TX-16W sampler.
- A file format from a Yamaha sampling keyboard
- which wrote IBM-PC format 3.5" floppies. Han-
+ A file format from a Yamaha sampling keyboard
+ which wrote IBM-PC format 3.5" floppies. Han-
dles reading of files which do not have the sam-
- ple 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
+ ple 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.
.vms More info to come.
- Used to compress speech audio for applications
+ Used to compress speech audio for applications
such as voice mail.
.voc Sound Blaster VOC files.
- 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 gener-
- ated appropriately. On output, silence is not
+ 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 gener-
+ ated appropriately. On output, silence is not
detected, nor are impossible sample rates.
.wav Microsoft .WAV RIFF files.
- These appear to be very similar to IFF files,
- but not the same. They are the native sound
+ 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
+ of such incredible importance to the computer
+ industry that it just had to have its own sound
file format.) Normally .wav files have all for-
- matting information in their headers, and so do
- not need any format options specified for an
- input file. If any are, they will override the
- file header, and you will be warned to this
- effect. You had better know what you are doing!
+ matting information in their headers, and so do
+ not need any format options specified for an
+ input file. If any are, they will override the
+ file header, and you will be warned to this
- December 10, 1999 7
+ July 24, 2000 7
@@ -466,64 +466,64 @@
SoX(1) SoX(1)
- Output format options will cause a format con-
- version, and the .wav will written appropri-
- ately. Sox currently can read PCM, ULAW, ALAW,
- MS ADPCM, and IMA (or DVI) ADPCM. It can write
+ effect. You had better know what you are doing!
+ Output format options will cause a format con-
+ version, and the .wav will written appropri-
+ ately. Sox currently can read PCM, ULAW, ALAW,
+ MS ADPCM, and IMA (or DVI) ADPCM. It can write
all of these formats including (NEW!) the ADPCM
encoding.
.wve Psion 8-bit alaw
- These are 8-bit a-law 8khz sound files used on
+ These are 8-bit a-law 8khz sound files used on
the Psion palmtop portable computer.
.raw Raw files (no header).
- The sample rate, size (byte, word, etc), and
+ The sample rate, size (byte, word, etc), and
encoding (signed, unsigned, etc.) of the sample
- file must be given. The number of channels
+ file must be given. The number of channels
defaults to 1.
.ub, .sb, .uw, .sw, .ul, .sl
- These are several suffices which serve as a
- shorthand for raw files with a given size and
- encoding. Thus, ub, sb, uw, sw, ul and sl cor-
- respond to "unsigned byte", "signed byte",
- "unsigned word", "signed word", "ulaw" (byte),
- and "signed long". The sample rate defaults to
+ These are several suffices which serve as a
+ shorthand for raw files with a given size and
+ encoding. Thus, ub, sb, uw, sw, ul and sl cor-
+ respond to "unsigned byte", "signed byte",
+ "unsigned word", "signed word", "ulaw" (byte),
+ and "signed long". 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
+ 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
+ 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.
- .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 mes-
- sage. The input must be a plain file, not a
+ .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 mes-
+ sage. The input must be a plain file, not a
pipe. This type can't be used for output files.
EFFECTS
Only one effect from the palette may be applied to a sound
- sample. To do multiple effects you'll need to run sox in
+ sample. To do multiple effects you'll need to run sox in
a pipeline.
avg [ -l | -r ]
- Reduce the number of channels by averaging the
- samples, or duplicate channels to increase the
- number of channels. This effect is automati-
- cally used when the number of input channels
+ Reduce the number of channels by averaging the
+ samples, or duplicate channels to increase the
+ number of channels. This effect is automati-
+ cally used when the number of input channels
differ from the number of output channels. When
- reducing the number of channels it is possible
- to manually specify the avg effect and use the
+ reducing the number of channels it is possible
- December 10, 1999 8
+ July 24, 2000 8
@@ -532,29 +532,30 @@
SoX(1) SoX(1)
- -l and -r options to select only the left or
- right channel for the output instead of averag-
+ to manually specify the avg effect and use the
+ -l and -r options to select only the left or
+ right channel for the output instead of averag-
ing the two channels.
band [ -n ] center [ width ]
- Apply a band-pass filter. The frequency
+ Apply a band-pass filter. The frequency
response drops logarithmically around the center
- frequency. The width gives the slope of the
- drop. The frequencies at center + width and
- center - width will be half of their original
+ frequency. The width gives the slope of the
+ drop. The frequencies at center + width and
+ center - width will be half of their original
amplitudes. Band defaults to a mode oriented to
pitched signals, i.e. voice, singing, or instru-
- mental music. The -n (for noise) option uses
- the alternate mode for un-pitched signals.
- Warning: -n introduces a power-gain of about
- 11dB in the filter, so beware of output clip-
+ mental music. The -n (for noise) option uses
+ the alternate mode for un-pitched signals.
+ Warning: -n introduces a power-gain of about
+ 11dB in the filter, so beware of output clip-
ping. Band introduces noise in the shape of the
filter, i.e. peaking at the center frequency and
- settling around it. See filter for a bandpass
+ settling around it. See filter for a bandpass
effect with steeper shoulders.
bandpass frequency bandwidth
- Butterworth bandpass filter. Description coming
+ Butterworth bandpass filter. Description coming
soon!
bandreject frequency bandwidth
@@ -564,10 +565,10 @@
chorus gain-in gain-out delay decay speed depth
-s | -t [ delay decay speed depth -s | -t ... ]
- Add a chorus to a sound sample. Each quadtuple
- delay/decay/speed/depth gives the delay in mil-
- liseconds and the decay (relative to gain-in)
- with a modulation speed in Hz using depth in
+ Add a chorus to a sound sample. Each quadtuple
+ delay/decay/speed/depth gives the delay in mil-
+ liseconds 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.
@@ -577,19 +578,18 @@
in-dB1,out-dB1[,in-dB2,out-dB2...]
[gain] [initial-volume]
- Compand (compress or expand) the dynamic range
- of a sample. The attack and decay time specify
- the integration time over which the absolute
- value of the input signal is integrated to
- determine its volume. Where more than one pair
- of attack/decay parameters are specified, each
- channel is treated separately and the number of
- pairs must agree with the number of input chan-
- nels. The second parameter is a list of points
+ Compand (compress or expand) the dynamic range
+ of a sample. The attack and decay time specify
+ the integration time over which the absolute
+ value of the input signal is integrated to
+ determine its volume. Where more than one pair
+ of attack/decay parameters are specified, each
+ channel is treated separately and the number of
+ pairs must agree with the number of input
- December 10, 1999 9
+ July 24, 2000 9
@@ -598,45 +598,46 @@
SoX(1) SoX(1)
- on the compander's transfer function specified
- in dB relative to the maximum possible signal
- amplitude. The input values must be in a
+ channels. The second parameter is a list of
+ points on the compander's transfer function
+ specified in dB relative to the maximum possible
+ signal amplitude. The input values must be in a
strictly increasing order but the transfer func-
- tion does not have to be monotonically rising.
- The special value -inf may be used to indicate
- that the input volume should be associated out-
- put volume. The points -inf,-inf and 0,0 are
- assumed; the latter may be overridden, but the
- former may not. The third (optional) parameter
- is a postprocessing gain in dB which is applied
+ tion does not have to be monotonically rising.
+ The special value -inf may be used to indicate
+ that the input volume should be associated out-
+ put volume. The points -inf,-inf and 0,0 are
+ assumed; the latter may be overridden, but the
+ former may not. The third (optional) parameter
+ is a postprocessing gain in dB which is applied
after the compression has taken place; the
fourth (optional) parameter is an initial volume
- to be assumed for each channel when the effect
+ to be assumed for each channel when the effect
starts. This permits the user to supply a nomi-
- nal level initially, so that, for example, a
+ nal level initially, so that, for example, a
very large gain is not applied to initial signal
levels before the companding action has begun to
- operate: it is quite probable that in such an
- event, the output would be severely clipped
- while the compander gain properly adjusts
+ operate: it is quite probable that in such an
+ event, the output would be severely clipped
+ while the compander gain properly adjusts
itself.
copy Copy the input file to the output file. This is
- the default effect if both files have the same
+ the default effect if both files have the same
sampling rate.
cut loopnumber
Extract loop #N from a sample.
- deemph Apply a treble attenuation shelving filter to
+ deemph Apply a treble attenuation shelving filter to
samples in audio cd format. The frequency
- response of pre-emphasized recordings is recti-
- fied. The filtering is defined in the standard
+ response of pre-emphasized recordings is recti-
+ fied. The filtering is defined in the standard
document ISO 908.
echo gain-in gain-out delay decay [ delay decay ... ]
Add echoing to a sound sample. Each delay/decay
- part gives the delay in milliseconds and the
+ part gives the delay in milliseconds and the
decay (relative to gain-in) of that echo. Gain-
out is the volume of the output.
@@ -643,19 +644,18 @@
echos gain-in gain-out delay decay [ delay decay ... ]
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
+ and the decay (relative to gain-in) of that
echo. Gain-out is the volume of the output.
filter [ low ]-[ high ] [ window-len [ beta ] ]
Apply a Sinc-windowed lowpass, highpass, or
- bandpass filter of given window length to the
- signal. low refers to the frequency of the
- lower 6dB corner of the filter. high refers to
- the frequency of the upper 6dB corner of the
+ bandpass filter of given window length to the
+ signal. low refers to the frequency of the
+ lower 6dB corner of the filter. high refers to
- December 10, 1999 10
+ July 24, 2000 10
@@ -664,64 +664,64 @@
SoX(1) SoX(1)
+ the frequency of the upper 6dB corner of the
filter.
- A lowpass filter is obtained by leaving low
- unspecified, or 0. A highpass filter is
- obtained by leaving high unspecified, or 0, or
- greater than or equal to the Nyquist frequency.
+ A lowpass filter is obtained by leaving low
+ unspecified, or 0. A highpass filter is
+ obtained by leaving high unspecified, or 0, or
+ greater than or equal to the Nyquist frequency.
The window-len, if unspecified, defaults to 128.
- Longer windows give a sharper cutoff, smaller
+ Longer windows give a sharper cutoff, smaller
windows a more gradual cutoff.
- The beta, if unspecified, defaults to 16. This
- selects a Kaiser window. You can select a Nut-
- tall window by specifying anything <= 2.0 here.
- For more discussion of beta, look under the
+ The beta, if unspecified, defaults to 16. This
+ selects a Kaiser window. You can select a Nut-
+ tall window by specifying anything <= 2.0 here.
+ For more discussion of beta, look under the
resample effect.
flanger gain-in gain-out delay decay speed -s | -t
- Add a flanger to a sound sample. Each triple
- delay/decay/speed gives the delay in millisec-
- onds and the decay (relative to gain-in) with a
+ Add a flanger to a sound sample. Each triple
+ delay/decay/speed gives the delay in millisec-
+ onds and the decay (relative to gain-in) with a
modulation speed in Hz. The modulation is
- either sinodial (-s) or triangular (-t). Gain-
+ either sinodial (-s) or triangular (-t). Gain-
out is the volume of the output.
highp center
- Apply a high-pass filter. The frequency
- response drops logarithmically with center fre-
- quency in the middle of the drop. The slope of
- the filter is quite gentle. See filter for a
+ Apply a high-pass filter. The frequency
+ response drops logarithmically with center fre-
+ quency in the middle of the drop. The slope of
+ the filter is quite gentle. See filter for a
highpass effect with sharper cutoff.
highpass frequency
- Butterworth highpass filter. Description com-
+ Butterworth highpass filter. Description com-
ming soon!
lowp center
Apply a low-pass filter. The frequency response
- drops logarithmically with center frequency in
+ drops logarithmically with center frequency in
the middle of the drop. The slope of the filter
- is quite gentle. See filter for a lowpass
+ is quite gentle. See filter for a lowpass
effect with sharper cutoff.
lowpass frequency
- Butterworth lowpass filter. Description coming
+ Butterworth lowpass filter. Description coming
soon!
map Display a list of loops in a sample, and miscel-
laneous loop info.
- mask Add "masking noise" to signal. This effect
- deliberately adds white noise to a sound in
- order to mask quantization effects, created by
+ mask Add "masking noise" to signal. This effect
+ deliberately adds white noise to a sound in
- December 10, 1999 11
+ July 24, 2000 11
@@ -730,50 +730,51 @@
SoX(1) SoX(1)
- 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
+ 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.
pan direction
- Pan the sound of an audio file from one channel
+ Pan the sound of an audio file from one channel
to another. This is done by changing the volume
- of the input channels so that it fade's out on
- one channel and fades-in on another. If the
- number of input channels is different then the
+ of the input channels so that it fades out on
+ one channel and fades-in on another. If the
+ number of input channels is different then the
number of output channels then this effect tries
- to intellegently handle this. For instance, if
+ to intelligently handle this. For instance, if
the input contains 1 channel and the output con-
- tains 2 channels, then it will create the miss-
- ing channel itself. The direction is a value
- from -1.0 to 1.0. -1.0 represents far left and
- 1.0 represents far right. Numbers in between
+ tains 2 channels, then it will create the miss-
+ ing channel itself. The direction is a value
+ from -1.0 to 1.0. -1.0 represents far left and
+ 1.0 represents far right. Numbers in between
will start the pan effect without totally muting
the opposite channel.
phaser gain-in gain-out delay decay speed -s | -t
- Add a phaser to a sound sample. Each triple
- delay/decay/speed gives the delay in millisec-
- onds and the decay (relative to gain-in) with a
+ Add a phaser to a sound sample. Each triple
+ delay/decay/speed gives the delay in millisec-
+ onds and the decay (relative to gain-in) with a
modulation speed in Hz. The modulation is
- either sinodial (-s) or triangular (-t). The
+ 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.
- pick Select the left or right channel of a stereo
- sample, or one of four channels in a quadro-
+ pick Select the left or right channel of a stereo
+ sample, or one of four channels in a quadro-
phonic sample.
pitch shift [ width interpole fade ]
- Change the pitch of file without affecting its
+ Change the pitch of file without affecting its
duration by cross-fading shifted samples. shift
is given in cents. Use a positive value to shift
- to treble, negative value to shift to bass.
- Default shift is 0. width of window is in ms.
- Default width is 20ms. Try 30ms to lower pitch,
- and 10ms to raise pitch. interpole option, can
+ to treble, negative value to shift to bass.
+ Default shift is 0. width of window is in ms.
+ Default width is 20ms. Try 30ms to lower pitch,
+ and 10ms to raise pitch. interpole option, can
be "cubic" or "linear". Default is "cubic". The
- fade option, can be "cos", "hamming", "linear"
+ fade option, can be "cos", "hamming", "linear"
or "trapezoid". Default is "cos".
polyphase [ -w < nut / ham > ]
@@ -782,12 +783,11 @@
[ -cutoff # ]
Translate input sampling rate to output sampling
- rate via polyphase interpolation, a DSP algo-
- rithm. This method is slow and uses lots of
+ rate via polyphase interpolation, a DSP
- December 10, 1999 12
+ July 24, 2000 12
@@ -796,44 +796,49 @@
SoX(1) SoX(1)
+ algorithm. This method is slow and uses lots of
RAM, but gives much better results than rate.
- -w < nut / ham > : select either a Nuttal (~90
- dB stopband) or Hamming (~43 dB stopband) win-
+
+ -w < nut / ham > : select either a Nuttal (~90
+ dB stopband) or Hamming (~43 dB stopband) win-
dow. Default is nut.
- -width long / short / # : specify the (approxi-
- mate) width of the filter. long is 1024 sam-
- ples; short is 128 samples. Alternatively, an
+
+ -width long / short / # : specify the (approxi-
+ mate) width of the filter. long is 1024 sam-
+ ples; short is 128 samples. Alternatively, an
exact number can be used. Default is long. The
- short option is not recommended, as it produces
+ short option is not recommended, as it produces
poor quality results.
- -cutoff # : specify the filter cutoff frequency
- in terms of fraction of bandwidth. If upsam-
- pling, then this is the fraction of the original
- 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.
+ -cutoff # : specify the filter cutoff frequency
+ in terms of fraction of bandwidth, also know as
+ the Nyquist frequency. Please see the resample
+ effect for further information on Nyquist fre-
+ quency. If upsampling, then this is the frac-
+ tion of the original 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.
+
rate Translate input sampling rate to output sampling
- rate via linear interpolation to the Least Com-
+ rate via linear interpolation to the Least Com-
mon Multiple of the two sampling rates. This is
the default effect if the two files have differ-
- ent sampling rates and the preview options was
+ ent sampling rates and the preview options was
specified. 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 resample or
- polyphase. If you are wondering which of SoX's
- rate changing effects to use, you will want to
- read a detailed analysis of all of them at
- http://eakaw2.et.tu-dresden.de/~wilde/resam-
- ple/resample.html [Nov,1999: These tests need to
- be updated for sox-12.17, which has bugfixes to
- the resample and polyphase code.]
+ 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 resample or polyphase. If
+ you are wondering which rate changing effects to
+ use, you will want to read a detailed analysis
+ of all of them at http://eakaw2.et.tu-dres-
+ den.de/~wilde/resample/resample.html
+
resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
Translate input sampling rate to output sampling
rate via simulated analog filtration. This
@@ -840,20 +845,15 @@
method is slower than rate, but gives much bet-
ter results.
- The -qs, -q, or -ql options specify increased
- accuracy at the cost of lower execution speed.
By default, linear interpolation is used, with a
- window width about 45 samples at the lower rate.
- This gives an accuracy of about 16 bits, but
- insufficient stopband rejection in the case that
- you want to have rolloff greater than about 0.80
- of the Nyquist frequency. The -q* options use
- quadratic interpolation of filter coefficients,
- resulting in about 24 bits precision.
+ window width about 45 samples at the lower of
+ the two rate. This gives an accuracy of about
+ 16 bits, but insufficient stopband rejection in
+ the case that you want to have rolloff greater
- December 10, 1999 13
+ July 24, 2000 13
@@ -862,8 +862,20 @@
SoX(1) SoX(1)
- Following is a table of the reasonable defaults
+ than about 0.80 of the Nyquist frequency.
+
+ The -q* options will change the default values
+ for rolloff and beta as well as use quadratic
+ interpolation of filter coefficients, resulting
+ in about 24 bits precision. The -qs, -q, or -ql
+ options specify increased accuracy at the cost
+ of lower execution speed. It is optional to
+ specify rolloff and beta parameters when using
+ the -q* options.
+
+ Following is a table of the reasonable defaults
which are built-in to sox:
+
Option Window rolloff beta interpolation
------ ------ ------- ---- -------------
(none) 45 0.80 16 linear
@@ -871,19 +883,51 @@
-q 75 0.875 16 quadratic
-ql 149 0.94 16 quadratic
------ ------ ------- ---- -------------
+
-qs, -q, or -ql use window lengths of 45, 75, or
- 149 samples, respectively, at the lower sample-
+ 149 samples, respectively, at the lower sample-
rate of the two files. This means progressively
- sharper stop-band rejection, at proportionally
+ sharper stop-band rejection, at proportionally
slower execution times.
- rolloff refers to the cut-off frequency of the
- low pass filter and is given in terms of the
- Nyquist frequency for the lower sample rate.
- rolloff therefore should be something between 0.
- and 1., in practice 0.8-0.95. The defaults are
- indicated above.
+ rolloff refers to the cut-off frequency of the
+ low pass filter and is given in terms of the
+ Nyquist frequency for the lower sample rate.
+ rolloff therefore should be something between
+ 0.0 and 1.0, in practice 0.8-0.95. The defaults
+ are indicated above.
+ The Nyquist frequency is 1/2 of the sample rate.
+ This refers to the fact that an audio file can
+ only represent frequencies up to 1/2 of the sam-
+ ple rate. Therefore, when reducing the sample
+ rate of an audio file a filter will remove all
+ frequency information above the new Nyquist fre-
+ quency. The rolloff refers to how close to the
+ Nyquist frequency this cutoff is, with closer
+ being better. When increasing the sample rate
+ of an audio file you would not expect to have
+ any frequencies exist that are past the original
+ Nyquist frequency. Because of filter proper-
+ ties, it is common to have false frequency data
+ created that is above the old Nyquist frequency.
+ In that case the rolloff refers to how close to
+ the original Nyquist frequency to use a highpass
+ filter to remove this false data, with closer
+ also being better.
+
+
+
+
+ July 24, 2000 14
+
+
+
+
+
+SoX(1) SoX(1)
+
+
The beta parameter determines the type of filter
window used. Any value greater than 2.0 is the
beta for a Kaiser window. Beta <= 2.0 selects a
@@ -917,17 +961,6 @@
can be computed in a reasonable amount of space.
To be precise, this is done when
-
-
- December 10, 1999 14
-
-
-
-
-
-SoX(1) SoX(1)
-
-
input_rate < output_rate
&&
output_rate/gcd(input_rate,output_rate) <= 511
@@ -949,6 +982,18 @@
time. A factor of 1.0 means no change, and is
the default. 2.0 doubles speed, thus time
length is cut by a half and pitch is one octave
+
+
+
+ July 24, 2000 15
+
+
+
+
+
+SoX(1) SoX(1)
+
+
higher. 0.5 halves speed thus time length dou-
bles and pitch is one octave lower.
@@ -982,66 +1027,73 @@
tion. window size is in ms. Default is 20ms.
The fade option, can be "lin". shift ratio, in
[0.0 1.0]. Default depends on stretch factor.
+ 1.0 to shorten, 0.8 to lengthen. The fading
+ ratio, in [0.0 0.5]. The amount of a fade's
+ default depends on factor and shift.
+ swap [ 1 2 | 1 2 3 4 ]
+ Swap channels in multi-channel sound files.
+ Optionally, you may specify the channel order
+ you would like the output in. This defaults to
+ output channel 2 and then 1 for stereo and 2, 1,
+ 4, 3 for quad-channels. An interesting feature
+ is that you may duplicate a given channel by
+ overwriting another. This is done by repeating
+ an output channel on the command line. For
+ example, swap 2 2 will overwrite channel 1 with
+ channel 2's data; creating a stereo file with
+ both channels containing the same audio data.
+ vibro speed [ depth ]
+ Add the world-famous Fender Vibro-Champ sound
+ effect to a sound sample by using a sine wave as
+ the volume knob. Speed gives the Hertz value of
- December 10, 1999 15
+ July 24, 2000 16
-SoX(1) SoX(1)
- 1.0 to shorten, 0.8 to lengthen. The fading
- ratio, in [0.0 0.5]. The amount of a fade's
- default depends on factor and shift.
+SoX(1) SoX(1)
- swap [ 1 2 3 4 ]
- Swap channels in multi-channel sound files. In
- files with more than 2 channels you may specify
- the order that the channels should be rearranged
- in.
- vibro speed [ depth ]
- Add the world-famous Fender Vibro-Champ sound
- effect to a sound sample by using a sine wave as
- the volume knob. Speed gives the Hertz value of
- the wave. This must be under 30. Depth gives
- the amount the volume is cut into by the sine
- wave, ranging 0.0 to 1.0 and defaulting to 0.5.
+ the wave. This must be under 30. Depth gives
+ the amount the volume is cut into by the sine
+ wave, ranging 0.0 to 1.0 and defaulting to 0.5.
vol gain [ type ]
- The vol effect is much like the command line
- option -v. It allows you to adjust the volume
- of an input file and allows you to specify the
- adjustment in relation to amplitude, power, or
+ The vol effect is much like the command line
+ option -v. It allows you to adjust the volume
+ of an input file and allows you to specify the
+ adjustment in relation to amplitude, power, or
dB. When type is amplitude then a linear change
of the amplitude is performed based on the gain.
- Therefore, a value of 1.0 will keep the volume
- the same, 0.0 to < 1.0 will cause the volume to
+ Therefore, a value of 1.0 will keep the volume
+ the same, 0.0 to < 1.0 will cause the volume to
decrease and values of > 1.0 will cause the vol-
- ume to increase. Beware of clipping audio data
- when the gain is greater then 1.0. A negative
- value performs the same adjustment while also
+ ume to increase. Beware of clipping audio data
+ when the gain is greater then 1.0. A negative
+ value performs the same adjustment while also
changing the phase.
- When type is power then a value of 1.0 also
+ When type is power then a value of 1.0 also
means no change in volume.
- When type is dB the amplitude is change loga-
- rithmically. 0.0 is constant while +6 doubles
+ When type is dB the amplitude is change loga-
+ rithmically. 0.0 is constant while +6 doubles
the amplitude.
- Sox enforces certain effects. If the two files have dif-
+ Sox enforces certain effects. If the two files have dif-
ferent sampling rates, the requested effect must be one of
- copy, or rate, If the two files have different numbers of
+ copy, or rate, If the two files have different numbers of
channels, the avg effect must be requested.
BUGS
- The syntax is horrific. Thats the breaks when trying to
+ The syntax is horrific. Thats the breaks when trying to
handle all things from the command line.
- Please report any bugs found in this version of sox to
+ Please report any bugs found in this version of sox to
Chris Bagwell (cbagwell@sprynet.com)
FILES
@@ -1048,23 +1100,11 @@
SEE ALSO
play(1), rec(1), soxexam(1)
-
-
-
- December 10, 1999 16
-
-
-
-
-
-SoX(1) SoX(1)
-
-
NOTICES
- The version of Sox that accompanies this manual page is
- support by Chris Bagwell (cbagwell@sprynet.com). Please
+ The version of Sox that accompanies this manual page is
+ support by Chris Bagwell (cbagwell@sprynet.com). Please
refer any questions regarding it to this address. You may
- obtain the latest version at the the web site
+ obtain the latest version at the the web site
http://home.sprynet.com/~cbagwell/sox.html
@@ -1077,46 +1117,6 @@
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
- December 10, 1999 17
+ July 24, 2000 17