ref: a479f7063aa32e45d0378b233727a44414ddcc00
parent: 03c73e6b0a38243191a3c20a896ca9b68c89a789
author: cbagwell <cbagwell>
date: Sat Oct 30 14:20:11 EDT 1999
Clearing up channel and rate options descriptions.
--- a/sox.1
+++ b/sox.1
@@ -136,7 +136,10 @@
gives the type of the sound sample file.
.TP 10
\fB-r \fIrate\fR
-Give sample rate in Hertz of file. If the input and output files have
+Give sample rate in Hertz of file. To cause the output file to have
+a different sample rate then 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.
@@ -170,10 +173,13 @@
.TP 10
\fB-c \fIchannels\fR
The number of sound channels in the data file.
-This may be 1, 2, or 4; for mono, stereo, or quad sound data. If an
-input and output file have a different number of channels then the
-average effect must be used. If it is not specified on the command line
-it will be invoked with default parameters.
+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 then the input
+file, include this option with the approraite value 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 with default parameters.
.PP
General options:
.TP 10
--- a/sox.txt
+++ b/sox.txt
@@ -93,37 +93,37 @@
-t filetype
gives the type of the sound sample file.
- -r rate Give sample rate in Hertz of file. 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 Give sample rate in Hertz of file. To cause the
+ output file to have a different sample rate then
+ 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.
-s/-u/-U/-A/-a/-g
- The sample data is signed linear (2's comple-
- ment), unsigned linear, U-law (logarithmic), A-
- law (logarithmic), ADPCM, or GSM. U-law and A-
+ The sample data is signed linear (2's comple-
+ ment), unsigned linear, U-law (logarithmic), A-
+ law (logarithmic), ADPCM, or GSM. U-law and A-
law are the U.S. and international 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. GSM is a standard used
+ logarithmic telephone sound compression. ADPCM
+ is form of sound compression that has a good
+ compromise between good sound quality and fast
+ encoding/decoding time. GSM is a standard used
for telephone sound compression in European
- countries and its gaining popularity because of
+ countries and its gaining popularity because of
its quality.
-b/-w/-l/-f/-d/-D
- The sample data is in bytes, 16-bit words,
- 32-bit longwords, 32-bit floats, 64-bit double
- floats, or 80-bit IEEE floats. Floats and dou-
+ The sample data 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.
- -x The sample data is in XINU format; that is, it
- comes from a machine with the opposite word
- order than yours and must be swapped according
- to the word-size given above. Only 16-bit and
- 32-bit integer data may be swapped. Machine-
+ -x The sample data is in XINU format; that is, it
+ comes from a machine with the opposite word
@@ -136,63 +136,63 @@
SoX(1) SoX(1)
+ order than yours and must be swapped according
+ to the word-size given above. Only 16-bit and
+ 32-bit integer data may be swapped. Machine-
format floating-point data is not portable.
IEEE floats are a fixed, portable format.
-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. If an input and output file
- have a different number of channels then the
- average effect must be used. If it is not spec-
- ified on the command line it will be invoked
- with default parameters.
+ 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 then the
+ input file, include this option with the appro-
+ raite 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.
General options:
- -e after the input file allows you to avoid giving
+ -e after the input file allows you to avoid giving
an output file and just name an effect. This is
- mainly useful with the stat effect but can be
+ mainly useful with the stat effect but can be
used with others.
-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 then the input file. The order that the
- effects are run in will be arranged for maximum
+ a different number of channels and a different
+ rate then the input file. The order that the
+ effects are run in will be arranged for maximum
speed and not quality.
-v volume Change amplitude (floating point); less than 1.0
decreases, greater than 1.0 increases. Note: we
- perceive volume logarithmically, not linearly.
+ perceive volume logarithmically, not linearly.
Note: see the stat effect.
- -V Print a description of processing phases. Use-
+ -V Print a description of processing phases. Use-
ful for figuring out exactly how sox is mangling
your sound samples.
- The input and output files may be standard input and out-
- put. This is specified by '-'. The -t type option must
- be given in this case, else sox will not know the format
+ The input and output files may be standard input and out-
+ put. This is specified by '-'. The -t type option must
+ be given in this case, else sox will not know the format
of the given file. The -t, -r, -s/-u/-U/-A,
- -b/-w/-l/-f/-d/-D and -x options refer to the input data
- when given before the input file name. After, they refer
+ -b/-w/-l/-f/-d/-D and -x options refer to the input data
+ when given before the input file name. After, they refer
to the output data.
- If you don't give an output file name, sox will just read
- the input file. This is useful for validating structured
- file formats; the stat effect may also be used via the -e
- option.
+ If you don't give an output file name, sox will just read
+ the input file. This is useful for validating structured
-FILE TYPES
- Sox needs to know the formats of the input and output
- files. File formats which have headers are checked, if
- that header doesn't seem right, the program exits with an
-
June 28, 1999 3
@@ -202,60 +202,60 @@
SoX(1) SoX(1)
- appropriate message. Currently, raw (no header) binary
- and textual data, Amiga 8SVX, Apple/SGI AIFF, SPARC .AU
- (w/header), AVR, NeXT .SND, CD-R, CVSD, GSM 06.10, Mac
- HCOM, Sound Tools MAUD, OSS device drivers, Turtle Beach
- .SMP, Sound Blaster, Sndtool, and Sounder, Sun Audio
- device driver, Yamaha TX-16W Sampler, IRCAM Sound Files,
- Creative Labs VOC, Psion .WVE, and Microsoft RIFF/WAV are
+ file formats; the stat effect may also be used via the -e
+ option.
+
+FILE TYPES
+ Sox needs to know the formats of the input and output
+ files. File formats which have headers are checked, if
+ that header doesn't seem right, the program exits with an
+ appropriate message. Currently, raw (no header) binary
+ and textual data, Amiga 8SVX, Apple/SGI AIFF, SPARC .AU
+ (w/header), AVR, NeXT .SND, CD-R, CVSD, GSM 06.10, Mac
+ HCOM, Sound Tools MAUD, OSS device drivers, Turtle Beach
+ .SMP, Sound Blaster, Sndtool, and Sounder, Sun Audio
+ device driver, Yamaha TX-16W Sampler, IRCAM Sound Files,
+ Creative Labs VOC, Psion .WVE, and Microsoft RIFF/WAV 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
+ 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,
+ raw stereo raw unsigned samples at 44khz. But,
there's some blocking/padding oddity in the for-
mat, so it needs its own handler.
.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
- 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
@@ -268,63 +268,63 @@
SoX(1) SoX(1)
- maximum and minimum are 1.00 and -1.00. This
+ .dat Text Data files
+ 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
+ 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.
+ 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
- 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
+ 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 psuedo-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
- -w -s /dev/dsp
- .sf IRCAM Sound Files.
- SoundFiles are used by academic music software
- such as the CSound package, and the MixView
- sound sample editor.
-
June 28, 1999 5
@@ -334,63 +334,63 @@
SoX(1) SoX(1)
+ 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 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
- samplers themselves. Currently loop points are
+ 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 psuedo-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
- detected, nor are impossible sample rates.
+ 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
- .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
-
June 28, 1999 6
@@ -400,63 +400,63 @@
SoX(1) SoX(1)
+ 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
+ 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. Note that it is possible to write data
- of a type that cannot be specified by the .wav
- header, and you will be warned that you a writ-
- ing a bad file ! Sox currently can read PCM,
- ULAW, ALAW, MS ADPCM, and IMA (or DVI) ADPCM.
- It can output all of these formats except the
+ Output format options will cause a format con-
+ version, and the .wav will written appropri-
+ ately. Note that it is possible to write data
+ of a type that cannot be specified by the .wav
+ header, and you will be warned that you a writ-
+ ing a bad file ! Sox currently can read PCM,
+ ULAW, ALAW, MS ADPCM, and IMA (or DVI) ADPCM.
+ It can output all of these formats except the
ADPCM styles.
.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
- style (signed, unsigned, etc.) of the sample
- file must be given. The number of channels
+ The sample rate, size (byte, word, etc), and
+ style (signed, unsigned, etc.) of the sample
+ file must be given. The number of channels
defaults to 1.
.ub, .sb, .uw, .sw, .ul
- These are several suffices which serve as a
- shorthand for raw files with a given size and
- style. Thus, ub, sb, uw, sw, and ul correspond
- to "unsigned byte", "signed byte", "unsigned
- word", "signed word", and "ulaw" (byte). The
- sample rate defaults to 8000 hz if not explic-
+ These are several suffices which serve as a
+ shorthand for raw files with a given size and
+ style. Thus, ub, sb, uw, sw, and ul correspond
+ to "unsigned byte", "signed byte", "unsigned
+ word", "signed word", and "ulaw" (byte). The
+ sample rate defaults to 8000 hz if not explic-
itly 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
+ 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
+ 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
- pipe. This type can't be used for output files.
+ .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
-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.
-
June 28, 1999 7
@@ -466,39 +466,48 @@
SoX(1) SoX(1)
+ 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-
+ 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 samples dif-
- fer then 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-
+ fer then 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
+ mental music. The -n (for noise) option uses
the alternate mode for un-pitched signals. Band
- introduces noise in the shape of the filter,
- i.e. peaking at the center frequency and set-
+ introduces noise in the shape of the filter,
+ i.e. peaking at the center frequency and set-
tling around it.
chorus gain-in gain-out delay decay speed deptch
-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.
@@ -508,18 +517,9 @@
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
- on the compander's transfer function specified
- in dB relative to the maximum possible signal
- amplitude. The input values must be in a
+ Compand (compress or expand) the dynamic range
+ of a sample. The attack and decay time specify
+ the integration time over which the absolute
@@ -532,42 +532,51 @@
SoX(1) SoX(1)
+ 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
+ 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.
@@ -574,21 +583,12 @@
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.
- 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-
- out is the volume of the output.
-
-
June 28, 1999 9
@@ -598,15 +598,23 @@
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-
+ 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
+ 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.
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.
@@ -613,25 +621,25 @@
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
- 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
+ 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.
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.
polyphase [ -w < num / ham > ]
@@ -640,18 +648,10 @@
[ -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 algo-
+ rithm. This method is slow and uses lots of
RAM, but gives much better results then rate.
- -w < nut / ham > : select either a Nuttal (~90
- dB stopband) or Hamming (~43 dB stopband) win-
- dow. Warning: Nuttall windows require 2x length
- than Hamming windows. Default is nut.
- -width long / short / # : specify the width of
- the filter. long is 1024 samples; short is 128
- samples. Alternatively, an exact number can be
- used. Default is long.
- -cutoff # : specify the filter cutoff frequency
+ -w < nut / ham > : select either a Nuttal (~90
@@ -664,28 +664,36 @@
SoX(1) SoX(1)
- in terms of fraction of bandwidth. If upsam-
- pling, then this is the fraction of the orignal
+ dB stopband) or Hamming (~43 dB stopband) win-
+ dow. Warning: Nuttall windows require 2x length
+ than Hamming windows. Default is nut.
+ -width long / short / # : specify the width of
+ the filter. long is 1024 samples; short is 128
+ samples. Alternatively, an exact number can be
+ used. Default is long.
+ -cutoff # : specify the filter cutoff frequency
+ in terms of fraction of bandwidth. If upsam-
+ pling, then this is the fraction of the orignal
signal that should go through. If downsampling,
- this is the fraction of the signal left after
- downsampling. Default is 0.95. Remember that
+ this is 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
+ 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 ues, you will want to
- read a detailed analysis of all of them at
+ 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 ues, you will want to
+ read a detailed analysis of all of them at
http://eakaw2.et.tu-dresden.de/~andreas/resam-
ple/resample.html
@@ -692,35 +700,27 @@
resample [ 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. rolloff refers to the cut-off fre-
- quency of the low pass filter and is given in
- terms of the Nyquist frequency for the lower
- sample rate. rolloff therefor should be some-
- thing between 0. and 1., in practice 0.8-0.95.
- beta trades stop band rejection against transi-
- tion width from passband to stop band. Larger
+ quency of the low pass filter and is given in
+ terms of the Nyquist frequency for the lower
+ sample rate. rolloff therefor should be some-
+ thing between 0. and 1., in practice 0.8-0.95.
+ beta trades stop band rejection against transi-
+ tion width from passband to stop band. Larger
beta means a slower transition and greater stop-
band rejection. beta should be at least greater
- than 2. The default is rollof 0.8, beta 17.5,
- which is rather conservative with respect to
- aliasing. Lower beta and higher rolloff values
- preserve more high frequency signal energy, but
- introduce measurable artifacts. This is the
- default effect if the two files have different
+ than 2. The default is rollof 0.8, beta 17.5,
+ which is rather conservative with respect to
+ aliasing. Lower beta and higher rolloff values
+ preserve more high frequency signal energy, but
+ introduce measurable artifacts. This is the
+ default effect if the two files have different
sampling rates.
- reverb gain-out delay [ delay ... ]
- Add reverbation to a sound sample. Each delay
- is given in milliseconds and its feedback is
- depending on the reverb-time in milliseconds.
- Each delay should be in the range of half to
- quarter of reverb-time to get a realistic rever-
- bation. Gain-out is the volume of the output.
-
June 28, 1999 11
@@ -730,60 +730,60 @@
SoX(1) SoX(1)
- reverse Reverse the sound sample completely. Included
+ reverb gain-out delay [ delay ... ]
+ Add reverbation to a sound sample. Each delay
+ is given in milliseconds and its feedback is
+ depending on the reverb-time in milliseconds.
+ Each delay should be in the range of half to
+ quarter of reverb-time to get a realistic rever-
+ bation. Gain-out is the volume of the output.
+
+ reverse Reverse the sound sample completely. Included
for finding Satanic subliminals.
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 [ debug | -v ]
- Do a statistical check on the input file, and
- print results on the standard error file. stat
- may copy the file untouched from input to out-
- put, if you select an output file. The "Volume
- Adjustment:" field in the statistics gives you
- the argument to the -v number which will make
+ Do a statistical check on the input file, and
+ print results on the standard error file. stat
+ may copy the file untouched from input to out-
+ put, if you select an output file. 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.
- There is an optional parameter -v that will
+ There is an optional parameter -v that will
print out the "Volume Adjustment:" field's value
- and return. This could be of use in scripts to
- auto convert the volume. There is an also an
- optional parameter debug that will place sox
- into debug mode and print out a hex dump of the
- sound file from the internal buffer that is in
- 32-bit signed PCM data. This is mainly only of
- use in tracking down endian problems that creep
+ and return. This could be of use in scripts to
+ auto convert the volume. There is an also an
+ optional parameter debug that will place sox
+ into debug mode and print out a hex dump of the
+ sound file from the internal buffer that is in
+ 32-bit signed PCM data. This is mainly only of
+ use in tracking down endian problems that creep
in to sox on cross-platform versions.
swap [ 1 2 3 4 ]
- Swap channels in multi-channel sound files. In
- files with more than 2 channels you may specify
+ 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
+ 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.
- 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. It's very tempting to include a
- default system that allows an effect name as the program
- name and just pipes a sound sample from standard input to
- standard output, but the problem of inputting the sample
- rates makes this unworkable.
- Please report any bugs found in this version of sox to
- Chris Bagwell (cbagwell@sprynet.com)
@@ -796,37 +796,37 @@
SoX(1) SoX(1)
+BUGS
+ The syntax is horrific. It's very tempting to include a
+ default system that allows an effect name as the program
+ name and just pipes a sound sample from standard input to
+ standard output, but the problem of inputting the sample
+ rates makes this unworkable.
+
+ Please report any bugs found in this version of sox to
+ Chris Bagwell (cbagwell@sprynet.com)
+
FILES
SEE ALSO
play(1), rec(1)
NOTICES
- The echoplex effect is: Copyright (C) 1989 by Jef
+ The echoplex effect is: Copyright (C) 1989 by Jef
Poskanzer.
Permission to use, copy, modify, and distribute this soft-
ware and its documentation for any purpose and without fee
- is hereby granted, provided that the above copyright
- notice appear in all copies and that both that copyright
- notice and this permission notice appear in supporting
- documentation. This software is provided "as is" without
+ is hereby granted, provided that the above copyright
+ notice appear in all copies and that both that copyright
+ notice and this permission notice appear in supporting
+ documentation. This software is provided "as is" without
express or implied warranty.
- The version of Sox that accompanies this manual page is
- support by Chris Bagwell (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
-
-
-
-
-
-
-
-
-
-