ref: 8abc793d95d141a162b56730d2061c49ca0515b8
parent: 631a9484ffda7cc636b4096041bd0bc4005cb929
author: cbagwell <cbagwell>
date: Thu Aug 24 22:58:21 EDT 2000
Converting updated man pages into text.
--- a/TODO
+++ b/TODO
@@ -9,19 +9,11 @@
o Add support for mixing multiple inputs from command line.
- o Endian checks are probably invalid on 64-bit machines. Need to use
- something a more constant size the "int". Probably should make it
- at least a util function/macro that can be overriden at compile time.
+ o Endian checks are probably invalid on 64-bit machines.
o Enhance sox for better interactive support. This includes updating
effect parameters in real time and ablity to start/stop/scan
file handlers in real time.
-
- o More handlers! Everyone who adds sound hardware to a computer has
- the urge to come up with their own file format.
-
- o More effects! I don't know DSP at all. A Pitch Shifter is high
- on the list. Spectrum analyser, etc, using FFTW lib.
o Loop support for all formats that know about it. WAV has
some support for loops ("cue-points") but no support for--- a/sox.txt
+++ b/sox.txt
@@ -9,55 +9,55 @@
SYNOPSIS
sox infile outfile
- sox infile outfile [ effect [ effect options ... ] ]
- sox infile -e effect [ effect options ... ]
- sox [ general options ] [ format options ] infile [ for-
- mat options ] outfile [ effect [ effect options ... ] ]
- General options: [ -e ] [ -h ] [ -p ] [ -v volume ] [ -V ]
+ sox [ general options ] [ format options ] infile
+ -e effect [ effect options ]
- Format options: [ -t filetype ] [ -r rate ] [
- -s/-u/-U/-A/-a/-i/-g ] [ -b/-w/-l/-f/-d/-D ] [ -c channels
- ] [ -x ]
+ sox [ general options ] [ format options ] infile
+ [ format options ] outfile
+ [ effect [ effect options ] ... ]
+ General options:
+ [ -h ] [ -p ] [ -v volume ] [ -V ]
+
+ Format options:
+ [ -t filetype ] [ -r rate ] [ -s/-u/-U/-A/-a/-i/-g ]
+ [ -b/-w/-l/-f/-d/-D ]
+ [ -c channels ] [ -x ] [ -e ]
+
Effects:
- avg [ -l | -r ]
- band [ -n ] center [ width ]
- bandpass frequency bandwidth
- bandreject frequency bandwidth
- check
- chorus gain-in gain out delay decay speed depth
- -s | -t [ delay decay speed depth -s | -t ]
- compand attack1,decay1[,attack2,decay2...]
- in-dB1,out-dB1[,in-dB2,out-dB2...]
- [gain] [initial-volume]
- copy
- cut
- deemph
- echo gain-in gain-out delay decay [ delay decay ...]
- echos gain-in gain-out delay decay [ delay decay ...]
- filter [ low ]-[ high ] [ window-len [ beta ]]
- flanger gain-in gain-out delay decay speed -s | -t
- highp center
- highpass frequency
- lowp center
- lowpass frequency
- map
- mask
- pan direction
- phaser gain-in gain-out delay decay speed -s | -t
- pick
- pitch shift [ width interpole fade ]
- polyphase [ -w < nut / ham > ]
- [ -width < long / short / # > ]
- [ -cutoff # ]
- rate
- resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
- reverb gain-out reverb-time delay [ delay ... ]
- reverse
- speed factor
- split
- stat [ -s n ] [-rms ] [ -v ] [ -d ]
+ avg [ -l | -r ]
+ band [ -n ] center [ width ]
+ bandpass frequency bandwidth
+ bandreject frequency bandwidth
+ chorus gain-in gain out delay decay speed depth
+ -s | -t [ delay decay speed depth -s | -t ]
+ compand attack1,decay1[,attack2,decay2...]
+ in-dB1,out-dB1[,in-dB2,out-dB2...]
+ [ gain ] [ initial-volume ]
+ copy
+ cut
+ deemph
+ echo gain-in gain-out delay decay [ delay decay ...]
+ echos gain-in gain-out delay decay [ delay decay ...]
+ filter [ low ]-[ high ] [ window-len [ beta ]]
+ flanger gain-in gain-out delay decay speed < -s | -t >
+ highp center
+ highpass frequency
+ lowp center
+ lowpass frequency
+ map
+ mask
+ pan direction
+ phaser gain-in gain-out delay decay speed < -s | -t >
+ pick
+ pitch shift [ width interpole fade ]
+ polyphase [ -w < nut / ham > ]
+ [ -width < long / short / # > ]
+ [ -cutoff # ]
+ rate
+ resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
+ reverb gain-out reverb-time delay [ delay ... ]
@@ -70,16 +70,21 @@
SoX(1) SoX(1)
- stretch [ factor [ window fade shift fading ]
- swap [ 1 2 | 1 2 3 4 ]
- vibro speed [ depth ]
- vol gain [ type ]
+ reverse
+ speed factor
+ split
+ stat [ -s n ] [ -rms ] [ -v ] [ -d ]
+ stretch [ factor [ window fade shift fading ]
+ swap [ 1 2 | 1 2 3 4 ]
+ vibro speed [ depth ]
+ vol gain [ type ]
DESCRIPTION
- SoX is a command line program that can convert most popu-
- lar audio files to most other popular audio file formats.
- It can optionally apply a sound effect to the file during
- this translation.
+ SoX is a command line program that can convert most popu-
+ lar audio files to most other popular audio file formats.
+ It can optionally change the audio sample data type and
+ apply one or more sound effects to the file during this
+ translation.
There are two types of audio files formats that SoX can
work with. The first are self-describing file formats.
@@ -97,10 +102,10 @@
rate The sample rate is in samples per second. For
example, CD sample rates are at 44100.
- data type What format the data is stored in. Most popular
- are 8-bit or 16-bit words.
+ data size The precision the data is stored in. Most popu-
+ lar are 8-bit bytes or 16-bit words.
- data format
+ data encoding
What encoding the data type uses. Examples are
u-law, ADPCM, or signed linear data.
@@ -114,19 +119,14 @@
OPTIONS
The option syntax is a little grotty, but in essence:
- sox file.au file.voc
+ sox file.au file.wav
translates a sound file in SUN Sparc .AU format into a
- SoundBlaster .VOC file, while
+ Microsoft .WAV file, while
- sox -v 0.5 file.au -r 12000 file.voc rate
- does the same format translation but also lowers the
- amplitude by 1/2 and changes the sampling rate from 8000
- hertz to 12000 hertz via the rate sound effect loop.
-
July 24, 2000 2
@@ -136,48 +136,84 @@
SoX(1) SoX(1)
+ sox -v 0.5 file.au -r 12000 file.wav mask
+
+ does the same format translation but also lowers the
+ amplitude by 1/2, changes the sampling rate to 12000
+ hertz, and applies the mask sound effect to the audio
+ data.
+
Format options:
- Format options effect the audio samples that they immedi-
- ately precede. If they are placed before the input file
- name then they effect the input data. If they are placed
+ Format options effect the audio samples that they immedi-
+ ately preceed. 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 corrupted header or produce an output file
- that is totally different style then the input file.
+ input file's corrupted header or produce an output file
+ that is totally different style then the input file. It
+ is also how sox is informed about the format of raw input
+ data.:w
-t filetype
- gives the type of the sound sample file.
+ gives the type of the sound sample file. Useful
+ when file extension is not standard or for spec-
+ ifying the .auto file type.
- -r rate Give sample rate in Hertz of file. To cause the
- output file to have a different sample rate than
- the input file, include this option with the
- appropriate rate value along with the output
- options. If the input and output files have
- different rates then a sample rate change effect
- must be ran. If a sample rate changing effect
- is not specified then a default one will be used
- with its default parameters.
+ -r rate Gives the sample rate in Hertz of the file. To
+ cause the output file to have a different sample
+ rate than the input file, include this option as
+ a part of the output options.
+ If the input and output files have different
+ rates then a sample rate change effect must be
+ ran. If a sample rate changing effect is not
+ specified then a default one will internally be
+ ran by sox using its default parameters.
-s/-u/-U/-A/-a/-i/-g
- The sample data format is signed linear (2's
+ The sample data encoding is signed linear (2's
complement), unsigned linear, U-law (logarith-
mic), A-law (logarithmic), ADPCM, IMA_ADPCM, or
- GSM. U-law and A-law are the U.S. and interna-
- tional standards for logarithmic telephone sound
- compression. ADPCM is form of sound compression
- that has a good compromise between good sound
- quality and fast encoding/decoding time.
- IMA_ADPCM is also a form of adpcm compression,
- slightly simpler and slightly lower fidelity
- than Microsoft's flavor of ADPCM. IMA_ADPCM is
- also called DVI_ADPCM. GSM is a standard used
- for telephone sound compression in European
- countries and its gaining popularity because of
- its quality.
+ GSM.
+ U-law (actually shorthand for mu-law) and A-law
+ are the U.S. and international standards for
+ logarithmic telephone sound compression. When
+ uncompressed it has roughly the precision of
+ 12-byte PCM audio.
+ ADPCM is form of sound compression that has a
+ good compromise between good sound quality and
+ fast encoding/decoding time. It is used for
+ telephone sound compression and places were full
+ fidelity is not as important. When uncompressed
+ it has roughly the precision of 16-bit PCM
+ audio. Popular version of ADPCM include G.726,
+ MS ADPCM, and IMA ADPCM. The -a flag has dif-
+ ferent meanings in different file handlers. In
+ .wav files it represents MS ADPCM files, in all
+
+
+ July 24, 2000 3
+
+
+
+
+
+SoX(1) SoX(1)
+
+
+ others it means G.726 ADPCM. IMA ADPCM is a
+ specific form of adpcm compression, slightly
+ simpler and slightly lower fidelity than
+ Microsoft's flavor of ADPCM. IMA ADPCM is also
+ called DVI ADPCM.
+ GSM is a standard used for telephone sound com-
+ pression in European countries and its gaining
+ popularity because of its quality. It usually
+ is CPU intensive to work with GSM audio data.
+
-b/-w/-l/-f/-d/-D
- The sample data type is in bytes, 16-bit words,
+ The sample data size is in bytes, 16-bit words,
32-bit longwords, 32-bit floats, 64-bit double
floats, or 80-bit IEEE floats. Floats and dou-
ble floats are in native machine format.
@@ -190,54 +226,58 @@
format floating-point data is not portable.
IEEE floats are a fixed, portable format.
-
-
-
- July 24, 2000 3
-
-
-
-
-
-SoX(1) SoX(1)
-
-
-c channels
The number of sound channels in the data file.
This may be 1, 2, or 4; for mono, stereo, or
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-
- 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
- command line it will be invoked with default
- parameters.
+ input file, include this option with the output
+ file options. If the input and output file have
+ a different number of channels then the avg
+ effect must be used. If the avg effect is not
+ specified on the command line it will be invoked
+ internally with default parameters.
- General options:
-
- -e When used after the input file (so that it
- applies to the output file) it allows you to
- avoid giving an output filename and will not
+ -e When used after the input filename (so that it
+ applies to the output file) it allows you to
+ avoid giving an output filename and will not
produce an output file. It will apply any spec-
ified effects to the input file. This is mainly
useful with the stat effect but can be used with
others.
+ General options:
+
-h Print version number and usage information.
- -p Run in preview mode and run fast. This will
+ -p Run in preview mode and run fast. This will
somewhat speed up sox when the output format has
- a different number of channels and a different
- rate than the input file. The order that the
- effects are run in will be arranged for maximum
- speed and not quality.
+ a different number of channels and a different
+ rate than the input file. Currently, this
+ defaults to using the rate effect instead of the
+ resample for sample rate changes.
+
+
+ July 24, 2000 4
+
+
+
+
+
+SoX(1) SoX(1)
+
+
-v volume Change amplitude (floating point); less than 1.0
- decreases, greater than 1.0 increases. Note: we
- perceive volume logarithmically, not linearly.
- Note: see the stat effect.
+ decreases, greater than 1.0 increases. May use
+ a negative number to invert the phase of the
+ audio data. It is interesting to note that we
+ percieve volume logarithmically but this adjusts
+ the amplitude linearly.
+ Note: see the stat effect for information on
+ finding the maximum value that can be used with
+ this option without causing audio data be be
+ clipped.
-V Print a description of processing phases. Use-
ful for figuring out exactly how sox is mangling
@@ -256,18 +296,6 @@
header doesn't seem right, the program exits with an
appropriate message.
-
-
-
- July 24, 2000 4
-
-
-
-
-
-SoX(1) SoX(1)
-
-
The following file formats are supported:
@@ -279,9 +307,9 @@
chunk. It does not support multiple sound
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
- work with them.
+ 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
@@ -294,6 +322,18 @@
(see below).
.avr Audio Visual Research
+
+
+
+ July 24, 2000 5
+
+
+
+
+
+SoX(1) SoX(1)
+
+
The AVR format is produced by a number of com-
mercial packages on the Mac.
@@ -315,25 +355,15 @@
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
- sample value. Values are normalized so that the
- 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.
+ time since the beginning of the first sample and
+ the sample value. Values are normalized so that
+ the 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 analyz-
+ ers or graph routines. SoX can also convert a
+ file in this format back into one of the other
+ file formats.
-
-
- July 24, 2000 5
-
-
-
-
-
-SoX(1) SoX(1)
-
-
.gsm GSM 06.10 Lossy Speech Compression
A standard for compressing speech which is used
in the Global Standard for Mobil telecommunica-
@@ -342,220 +372,232 @@
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
- formats.
+ CPU intensive.
+ GSM in sox is optional and requires access to an
+ external GSM library. To see if there is sup-
+ port 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
- 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
- output values to use the highest quality samples
- your sound card can handle. Example: -t ossdsp
- -w -s /dev/dsp
- .sf IRCAM Sound Files.
- 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
+ July 24, 2000 6
- July 24, 2000 6
+SoX(1) SoX(1)
+ 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.
-SoX(1) SoX(1)
+ 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 format as passed in to /fBSoX. 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 -w -s /dev/dsp
+ .sf IRCAM Sound Files.
+ Sound Files are used by academic music software
+ such as the CSound package, and the MixView
+ sound sample editor.
- package, although not all are supported by the
- samplers themselves. Currently loop points are
+ .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
+ 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-
- 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
- sample rate is still unknown.
+ 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
- .vms More info to come.
- 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
- 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
- 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 .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
+ July 24, 2000 7
- July 24, 2000 7
+SoX(1) SoX(1)
+ 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.
-SoX(1) SoX(1)
+ .vms More info to come.
+ 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
+ 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
+ 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 .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!
- 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
+ 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
- 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.
+ 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),
- .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
- 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
- differ from the number of output channels. When
- reducing the number of channels it is possible
+ July 24, 2000 8
- July 24, 2000 8
+SoX(1) SoX(1)
+ 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
+ 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.
-SoX(1) SoX(1)
+ .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
+ Multiple effects may be applied to the audio data by spec-
+ ifying them one after another at the end of the command
+ line.
- 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-
+ 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
+ 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
+ -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
+
+
+
+ July 24, 2000 9
+
+
+
+
+
+SoX(1) SoX(1)
+
+
soon!
bandreject frequency bandwidth
@@ -565,10 +607,10 @@
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.
@@ -578,66 +620,66 @@
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
+ 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
+ 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
+ after the compression has taken place; the
+ fourth (optional) parameter is an initial volume
+ 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
+ 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
+ itself.
- July 24, 2000 9
+ July 24, 2000 10
+
SoX(1) SoX(1)
- 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
- after the compression has taken place; the
- fourth (optional) parameter is an initial volume
- 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
- 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
- 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.
@@ -644,137 +686,137 @@
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
+ 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
+ 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.
+ The window-len, if unspecified, defaults to 128.
+ Longer windows give a sharper cutoff, smaller
+ windows a more gradual cutoff.
- July 24, 2000 10
+ 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
+ modulation speed in Hz. The modulation is
-SoX(1) SoX(1)
+ July 24, 2000 11
- 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.
- The window-len, if unspecified, defaults to 128.
- 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
- resample effect.
+SoX(1) SoX(1)
- 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
- 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
+ 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.
+ pan direction
+ 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 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 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
+ 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
- July 24, 2000 11
+ July 24, 2000 12
-SoX(1) SoX(1)
- 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.
+SoX(1) SoX(1)
- pan direction
- 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 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 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
- 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
+ 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 > ]
@@ -783,59 +825,58 @@
[ -cutoff # ]
Translate input sampling rate to output sampling
- rate via polyphase interpolation, a DSP
+ rate via polyphase interpolation, a DSP algo-
+ rithm. 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-
+ 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
+ exact number can be used. Default is long. The
+ short option is not recommended, as it produces
+ poor quality results.
- July 24, 2000 12
+ -cutoff # : specify the filter cutoff frequency
+ in terms of fraction of frequency bandwidth,
+ also know as the Nyquist frequency. Please see
+ the resample effect for further information on
+ Nyquist frequency. If upsampling, then this is
+ the fraction of the original signal that should
+ go through. If downsampling, this is the frac-
+ tion of the signal left after downsampling.
+ Default is 0.95. Remember that this is a float.
+ July 24, 2000 13
-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-
- 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
- exact number can be used. Default is long. The
- short option is not recommended, as it produces
- poor quality results.
+SoX(1) SoX(1)
- -cutoff # : specify the filter cutoff frequency
- in terms of fraction of frequency bandwidth,
- also know as the Nyquist frequency. Please see
- the resample effect for further information on
- Nyquist frequency. If upsampling, then this is
- the fraction of the original signal that should
- go through. If downsampling, this is the frac-
- tion 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
+ 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
+ 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
+ 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
@@ -842,38 +883,26 @@
resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
Translate input sampling rate to output sampling
rate via simulated analog filtration. This
- method is slower than rate, but gives much bet-
+ method is slower than rate, but gives much bet-
ter results.
By default, linear interpolation is used, with a
- 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
-
-
-
- July 24, 2000 13
-
-
-
-
-
-SoX(1) SoX(1)
-
-
+ 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
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
+ 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
+ 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
+ Following is a table of the reasonable defaults
which are built-in to sox:
Option Window rolloff beta interpolation
@@ -885,90 +914,102 @@
------ ------ ------- ---- -------------
-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
- 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
- 0.0 and 1.0, in practice 0.8-0.95. The defaults
- are indicated above.
- The Nyquist frequency is equal to (sample rate /
- 2). Logically, this is because the A/D con-
- verter needs at least 2 samples to detect 1
- cycle at the Nyquist frequency. Frequencies
- higher then the Nyquist will actually appear as
- lower frequencies to the A/D converter and is
- called aliasing. Normally, A/D converts run the
- signal through a highpass filter first to avoid
- these problems.
- Similar problems will happen in software when
- reducing the sample rate of an audio file (fre-
- quencies above the new Nyquist frequency can be
- aliased to lower frequencies). Therefore, a
- good resample effect will remove all frequency
- information above the new Nyquist frequency.
+ July 24, 2000 14
- The rolloff refers to how close to the Nyquist
- frequency this cutoff is, with closer being
- July 24, 2000 14
+SoX(1) SoX(1)
+ 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
+ 0.0 and 1.0, in practice 0.8-0.95. The defaults
+ are indicated above.
-SoX(1) SoX(1)
+ The Nyquist frequency is equal to (sample rate /
+ 2). Logically, this is because the A/D con-
+ verter needs at least 2 samples to detect 1
+ cycle at the Nyquist frequency. Frequencies
+ higher then the Nyquist will actually appear as
+ lower frequencies to the A/D converter and is
+ called aliasing. Normally, A/D converts run the
+ signal through a highpass filter first to avoid
+ these problems.
+ Similar problems will happen in software when
+ reducing the sample rate of an audio file (fre-
+ quencies above the new Nyquist frequency can be
+ aliased to lower frequencies). Therefore, a
+ good resample effect will remove all frequency
+ information above the new Nyquist frequency.
- better. When increasing the sample rate of an
+ The rolloff refers to how close to the Nyquist
+ frequency this cutoff is, with closer being bet-
+ ter. When increasing the sample rate of an
audio file you would not expect to have any fre-
- quencies exist that are past the original
- Nyquist frequency. Because of resampling prop-
- erties, it is common to have alaising data cre-
- ated that is above the old Nyquist frequency.
- In that case the rolloff refers to how close to
+ quencies exist that are past the original
+ Nyquist frequency. Because of resampling prop-
+ erties, it is common to have alaising data cre-
+ ated 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
+ filter to remove this false data, with closer
also being better.
The beta parameter determines the type of filter
- window used. Any value greater than 2.0 is the
+ window used. Any value greater than 2.0 is the
beta for a Kaiser window. Beta <= 2.0 selects a
- Nuttall window. If unspecified, the default is
+ Nuttall window. If unspecified, the default is
a Kaiser window with beta 16.
In the case of Kaiser window (beta > 2.0), lower
- betas produce a somewhat faster transition from
- passband to stopband, at the cost of noticeable
- artifacts. A beta of 16 is the default, beta
- less than 10 is not recommended. If you want a
- sharper cutoff, don't use low beta's, use a
+ betas produce a somewhat faster transition from
+ passband to stopband, at the cost of noticeable
+ artifacts. A beta of 16 is the default, beta
+ less than 10 is not recommended. If you want a
+ sharper cutoff, don't use low beta's, use a
longer sample window. A Nuttall window is
- selected by specifying any 'beta' <= 2, and the
- Nuttall window has somewhat steeper cutoff than
- the default Kaiser window. You will probably
- not need to use the beta parameter at all,
- unless you are just curious about comparing the
+ selected by specifying any 'beta' <= 2, and the
+
+
+
+ July 24, 2000 15
+
+
+
+
+
+SoX(1) SoX(1)
+
+
+ Nuttall window has somewhat steeper cutoff than
+ the default Kaiser window. You will probably
+ not need to use the beta parameter at all,
+ unless you are just curious about comparing the
effects of Nuttall vs. Kaiser windows.
This is the default effect if the two files have
- different sampling rates. Default parameters
+ different sampling rates. Default parameters
are, as indicated above, Kaiser window of length
45, rolloff 0.80, beta 16, linear interpolation.
- NOTE: -qs is only slightly slower, but more
+ NOTE: -qs is only slightly slower, but more
accurate for 16-bit or higher precision.
- NOTE: In many cases of up-sampling, no interpo-
- lation is needed, as exact filter coefficients
+ NOTE: In many cases of up-sampling, no interpo-
+ lation is needed, as exact filter coefficients
can be computed in a reasonable amount of space.
To be precise, this is done when
@@ -978,145 +1019,104 @@
reverb gain-out delay [ delay ... ]
Add reverberation 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
+ 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 rever-
+ beration. Gain-out is the volume of the output.
-
-
- July 24, 2000 15
-
-
-
-
-
-SoX(1) SoX(1)
-
-
- reverberation. Gain-out is the volume of the
- output.
-
- reverse Reverse the sound sample completely. Included
+ reverse Reverse the sound sample completely. Included
for finding Satanic subliminals.
speed factor
- Speed up or down the sound, as a magnetic tape
+ Speed up or down the sound, as a magnetic tape
with a speed control. It affects both pitch and
- time. A factor of 1.0 means no change, and is
+ 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
- higher. 0.5 halves speed thus time length dou-
+ length is cut by a half and pitch is one octave
+ higher. 0.5 halves speed thus time length dou-
bles and pitch is one octave lower.
split Turn a mono sample into a stereo sample by copy-
- ing the input channel to the left and right
+ ing the input channel to the left and right
channels.
stat [ -s n ] [-rms ] [ -v ] [ -d ]
- Do a statistical check on the input file, and
+ Do a statistical check on the input file, and
print results on the standard error file. Audio
- data is passed unmodified from input to output
+ data is passed unmodified from input to output
file unless used along with the -e option.
The "Volume Adjustment:" field in the statistics
- gives you the argument to the -v number which
- will make the sample as loud as possible without
- clipping.
- The option -v will print out the "Volume Adjust-
- ment:" field's value only and return. This
- could be of use in scripts to auto convert the
- volume.
- The -s n option is used to scale the input data
- by a given factor. The default value of n is
- the max value of a signed long variable
- (0x7fffffff). Internal effects always work with
- signed long PCM data and so the value should
- relate to this fact.
- The -rms option will convert all output average
- values to root mean square format.
+ July 24, 2000 16
- There is also an optional parameter -d that will
- 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.
+SoX(1) SoX(1)
- July 24, 2000 16
+ gives you the argument to the -v number which
+ will make the sample as loud as possible without
+ clipping.
+ The option -v will print out the "Volume Adjust-
+ ment:" field's value only and return. This
+ could be of use in scripts to auto convert the
+ volume.
+ The -s n option is used to scale the input data
+ by a given factor. The default value of n is
+ the max value of a signed long variable
+ (0x7fffffff). Internal effects always work with
+ signed long PCM data and so the value should
+ relate to this fact.
+ The -rms option will convert all output average
+ values to root mean square format.
+ There is also an optional parameter -d that will
+ 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.
-SoX(1) SoX(1)
-
stretch factor [window fade shift fading]
- Time stretch file by a given factor. Change
+ Time stretch file by a given factor. Change
duration without affecting the pitch. factor of
- stretching: >1.0 lengthen, <1.0 shorten dura-
- 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
+ stretching: >1.0 lengthen, <1.0 shorten dura-
+ 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
+ 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
+ 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
+ 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.
- 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
- 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
- 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
- changing the phase.
- 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
- the amplitude.
- Sox enforces certain effects. If the two files have dif-
- ferent sampling rates, the requested effect must be one of
-
-
July 24, 2000 17
@@ -1126,15 +1126,42 @@
SoX(1) SoX(1)
- polyphase, rate, or resample. If the two files have dif-
- ferent numbers of channels, the avg or other channel mix-
+ 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.
+
+ 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
+ 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
+ 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
+ changing the phase.
+ 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
+ the amplitude.
+
+ Sox enforces certain effects. If the two files have dif-
+ ferent sampling rates, the requested effect must be one of
+ polyphase, rate, or resample. If the two files have dif-
+ ferent numbers of channels, the avg or other channel mix-
ing 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
@@ -1142,38 +1169,11 @@
play(1), rec(1), soxexam(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
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