ref: 46b0adba843a5179fcabf0a7e98cbe08386a2e68
parent: e60dc72083fbaaf0dcd00d475e200dca56c0eb86
author: rrt <rrt>
date: Mon Dec 18 19:22:15 EST 2006
Remove generated files
--- a/sox.txt
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-SoX(1) Sound eXchange SoX(1)
-
-
-
-NAME
- sox - Sound eXchange : universal sound sample translator
-
-SYNOPSIS
- sox infile1 [ infile2 ... ] outfile
-
- sox [ global options ] [ format options ] infile1
- [ [ format options ] infile2 ... ] [ format options ] outfile
- [ effect [ effect options ] ... ]
-
- soxmix infile1 infile2 [ infile3 ... ] outfile
-
- soxmix [ global options ] [ format options ] infile1
- [ format options ] infile2
- [ [ format options ] infile3 ... ]
- [ format options ] outfile
- [ effect [ effect options ] ... ]
-
-DESCRIPTION
- SoX is a command line program that can convert most popular 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.
-
- If more than one input file is specified then they are concatenated
- into the output file. In this case, it has a restriction that all
- input files must be of the same data type and sample rates.
-
- soxmix is functionally the same as the command line program sox except
- that it takes two or more files as input and mixes the audio together
- to produce a single file as output. It has a restriction that all
- input files must be of the same data type and sample rates.
-
- There are two types of audio file formats that SoX can work with. The
- first are self-describing file formats. These contain a header that
- completely describe the characteristics of the audio data that follows.
-
- The second type are header-less data, or sometimes called raw data. A
- user must pass enough information to SoX on the command line so that it
- knows what type of data it contains.
-
- Audio data can usually be totally described by four characteristics:
-
- rate The sample rate is in samples per second. For example, CD
- sample rates are at 44100.
-
- data size The precision the data is stored in. Most popular are 8-bit
- bytes or 16-bit words.
-
- data encoding
- What encoding the data type uses. Examples are u-law, ADPCM,
- or signed linear data.
-
- channels How many channels are contained in the audio data. Mono and
- Stereo are the two most common.
-
- Please refer to the soxexam(1) manual page for a long description with
- examples on how to use SoX with various types of file formats.
-
-OPTIONS
- The option syntax is a little grotty, but in essence:
-
- sox file.au file.wav
-
- translates a sound file in SUN Sparc .AU format into a Microsoft .WAV
- file, while
-
- 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.
-
- The following will mix two sound files together to to produce a single
- sound file.
-
- soxmix music.wav voice.wav mixed.wav
-
- Filenames:
-
- SoX can be used as a part of pipe operations by using the special file-
- names of "-". If specified as an input name, it will read data from
- stdin. If specified as an output name, it will send data to stdout.
-
- Global options:
-
- -h Print version number and usage information.
-
- --help Same as -h
-
- --help-effect=name
- Prints usage information on the specifed effect. The name
- all can be used to disable usage on all effects.
-
- -p Run in preview mode and run fast. This will somewhat speed
- up SoX when the output format has a different number of chan-
- nels and a different rate than the input file. Currently,
- this defaults to using the rate effect instead of the resam-
- ple effect for sample rate changes.
-
- -q Run in quite mode when SoX wouldn’t otherwise do that.
- Inverse of -S option.
-
- -S Print status while processing audio data. Tells how much of
- audio data has been processed in terms of audio running time
- instead of samples.
-
- --version Print version number and exit.
-
- -V Print a description of processing phases. Useful for figur-
- ing out exactly how SoX
-
- is mangling your sound samples.
-
- Format options:
-
- Format options effect the input or output file that they immediately
- precede.
-
- Self describing input files can obtain all the format information
- directly from the header and so don’t generally need format options.
- Headerless input files lack this information and so format options must
- be used to inform SoX of the file’s data type, sample rate, and number
- of channels.
-
- By default, SoX attempts to write audio data using the same data type,
- sample rate, and channel count as the input data. If the user wants
- the output file to be of a different format then format options can be
- used to specify the differences.
-
- If an output file format doesn’t support the same data type, sample
- rate, or channel count as the input file format, then SoX will auto
- select the closest values it does support so that the user does not
- have to specify these format change options manually.
-
- -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 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 internally with default parameters.
-
- -e When specified after the last 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 specified effects to the input file. This is
- mainly useful with the stat effect but can be used.
-
- -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 format
- options.
- If the input and output files have different rates then a
- sample rate change effect must be ran. Since SoX has multi-
- ple rate changing effects, the user can specify which to use
- as an effect. If no rate change effect is specified then a
- default one will be chosen.
-
- -t filetype
- gives the file type of the sound sample file. Useful when
- file extension is not standard or can not be determeind by
- looking at the header of the file. See the section FILE
- TYPES for a list of supported file types.
-
- -v volume Change amplitude (floating point); less than 1.0 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 perceive volume logarithmically but this adjusts
- the amplitude linearly.
- As with other format options, the volume option effects the
- file its specified with. This is useful whe processing muti-
- ple input files as the volume adjustment can be specified for
- each input file or just once to adjust the output file. This
- can be compared to an audio mixer were you can control the
- volume of each input as well as a master volume (output
- side).
- soxmix defaults the value of the -v option for each input
- file to 1/input_file_count. This means if your mixing two
- input files together then each input file’s volume is
- adjusted by 0.5. This is done to prevent clipping of audio
- data during the mixing operation. Users will most likely not
- be happy with this large of a volume adjustment and can spec-
- ify the -v option to override this default value.
- Note: For the non-mixing case, see the stat effect for infor-
- mation on finding the maximum volume adjustment that can be
- done with this option without causing audio data to be
- clipped.
-
- -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-format
- floating-point data is not portable.
-
- -s/-u/-U/-A/-a/-i/-g/-f
- The sample data encoding is signed linear (2’s complement),
- unsigned linear, u-law (logarithmic), A-law (logarithmic),
- ADPCM, IMA_ADPCM, GSM, or Floating-point.
- U-law (actually shorthand for mu-law) and A-law are the U.S.
- and international standards for logarithmic telephone sound
- compression. When uncompressed u-law has roughly the preci-
- sion of 14-bit PCM audio and A-law has roughly the precision
- of 13-bit PCM audio.
- A-law and u-law data is sometimes encoded using a reversed
- bit-ordering (ie. MSB becomes LSB). Internally, SoX under-
- stands how to work with this encoding but there is currently
- no command line option to specify it. If you need this sup-
- port then you can use the psuedo file types of ".la" and
- ".lu" to inform sox of the encoding. See supported file
- types for more information.
- ADPCM is a form of sound compression that has a good compro-
- mise 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 ver-
- sion of ADPCM include G.726, MS ADPCM, and IMA ADPCM. The -a
- flag has different meanings in different file handlers. In
- .wav files it represents MS ADPCM files, in all 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 compression 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/-d
- The sample data size is in bytes, 16-bit words, 32-bit long
- words, or 64-bit double long (long long) words.
-
-FILE TYPES
- SoX attempts to determine the file type of input files automatically by
- looking at the header of the audio file. When it is unable to detect
- the file type or if its an output file then it uses the file extension
- of the file to determine what type of file format handler to use. This
- can be overridden by specifying the "-t" option on the command line.
-
- The input and output files may be read from standard in and out. This
- is done by specifying ’-’ as the filename.
-
- File formats which have headers are checked, if that header doesn’t
- seem right, the program exits with an appropriate message.
-
- The following file formats are supported:
-
-
- .8svx Amiga 8SVX musical instrument description format.
-
- .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
- description format. AIFF files are multimedia archives and
- can have multiple audio and picture chunks. You may need a
- separate archiver to work with them.
-
- .alsa ALSA /dev/snd/pcmCxDxp 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 ALSA /dev/snd/pcmCxDxp file and configure it to use the
- same data format as passed in to SoX. It works for both
- playing and recording sound samples. When playing sound
- files it attempts to set up the ALSA 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: sox infile -t alsa -w -s
- /dev/snd/pcmC0D0p
-
- .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 order. The .au handler can read these files
- but will not write them. Some .au files have valid AU head-
- ers 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 commercial packages
- on the Mac.
-
- .cdr CD-R
- CD-R files are used in mastering music on Compact Disks. The
- audio data on a CD-R disk is a raw audio file with a format
- of stereo 16-bit signed samples at a 44khz sample rate.
- There is a special blocking/padding oddity at the end of the
- audio file and is why it needs its own handler.
-
- .cvs Continuously Variable Slope Delta modulation
- 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. Subsequent lines contain two numeric data
- items: the 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 ana-
- lyzers or graph 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 telecommunications (GSM). Its good for
- its purpose, shrinking 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 formats.
-
- .hcom Macintosh HCOM files. These are (apparently) Mac FSSD files
- with some variant of Huffman compression. The Macintosh has
- wacky file formats and this format handler apparently doesn’t
- handle all the ones it should. Mac users will need 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.
-
- .mp3 MP3 Compressed Audio
- MP3 audio files come from the MPEG standards for audio and
- video compression. They are a lossy compression format that
- achieves good compression rates with a minimum amount of
- quality loss. Also see Ogg Vorbis for a similar format. MP3
- support in SoX is optional and requires access to either or
- both the external libmad and libmp3lame libraries. To see if
- there is support for Mp3 run sox -h and look for it under the
- list of supported file formats as "mp3".
-
-
- .nul Null file handler. This is a fake file hander that act as if
- its reading a stream of 0’s from a while or fake writing out-
- put to a file. This is not a very useful file handler in
- most cases. It might be useful in some scripts were you do
- not want to read or write from a real file but would like to
- specify a filename for consistency.
-
- .ogg Ogg Vorbis Compressed Audio.
- Ogg Vorbis is a open, patent-free CODEC designed for com-
- pressing music and streaming audio. It is similar to MP3,
- VQF, AAC, and other lossy formats. SoX can decode all types
- of Ogg Vorbis files, but can only encode at 128 kbps. Decod-
- ing is somewhat CPU intensive and encoding is very CPU inten-
- sive.
- Ogg Vorbis in SoX is optional and requires access to external
- Ogg Vorbis libraries. To see if there is support for Ogg
- Vorbis run sox -h and look for it under the list of supported
- file formats as "vorbis".
-
- 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 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 out-
- put values to use the highest quality samples your sound card
- can handle. Example: sox infile -t ossdsp -w -s /dev/dsp
-
- .prc Psion record.app
- Used in some Psion devices for System alarms. This format is
- newer then the .wve format that is used in some Psion
- devices.
-
- .sf IRCAM Sound Files.
- Sound Files are used by academic music software such as the
- CSound package, and the MixView sound sample editor.
-
- .sph
- SPHERE (SPeech HEader Resources) is a file format defined by
- NIST (National Institute of Standards and Technology) and is
- used with speech audio. SoX can read these files when they
- contain u-law and PCM data. It will ignore any header infor-
- mation that says the data is compressed using shorten com-
- pression and will treat the data as either u-law or PCM.
- This will allow SoX and the command line shorten program to
- be ran together using pipes to uncompress the data and then
- pass the result to SoX for processing.
-
- .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 them-
- selves. Currently loop points are ignored.
-
- .snd
- Under DOS this file format is the same as the .sndt format.
- Under all other platforms it is the same as the .au format.
-
- .sndt SoundTool files.
- This is an older DOS file format.
-
- 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 /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 audio driver to use the same format as
- the input file. It is suggested to always override the out-
- put values to use the highest quality samples your hardware
- can handle. Example: sox infile -t sunau -w -s /dev/audio or
- sox infile -t sunau -U -c 1 /dev/audio for older sun equip-
- ment.
-
- .txw Yamaha TX-16W sampler.
- A file format from a Yamaha sampling keyboard which wrote
- IBM-PC format 3.5" floppies. Handles reading of files which
- do not have the sample rate field set to one of the expected
- by looking at some other bytes in the attack/loop length
- fields, and defaulting to 33kHz if the sample rate is still
- unknown.
-
- .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 generated appropriately. On out-
- put, silence is not detected, nor are impossible sample
- rates. Note, this version now supports playing VOC files
- with multiple blocks and supports playing files containing u-
- law and A-law samples.
-
- vorbis See .ogg format.
-
- .vox A headerless file of Dialogic/OKI ADPCM audio data commonly
- comes with the extension .vox. This ADPCM data has 12-bit
- precision packed into only 4-bits.
-
- .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 formatting 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 bet-
- ter know what you are doing! Output format options will cause
- a format conversion, and the .wav will written appropriately.
- SoX currently can read PCM, ULAW, ALAW, MS ADPCM, and IMA (or
- DVI) ADPCM. It can write all of these formats including the
- ADPCM encoding. Big endian versions of RIFF files, called
- RIFX, can also be read and written. To write a RIFX file,
- use the -x option with the output file options.
-
- .wve Psion 8-bit A-law
- 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 encoding
- (signed, unsigned, etc.) of the sample file must be given.
- The number of channels defaults to 1.
-
- .ub, .sb, .uw, .sw, .ul, .al, .lu, .la, .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, al, lu, la and sl correspond to "unsigned byte", "signed
- byte", "unsigned word", "signed word", "u-law" (byte), "A-
- law" (byte), inverse bit order "u-law", inverse bit order "A-
- law", and "signed long". The sample rate defaults to 8000 hz
- if not explicitly set, and the number of channels 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.
-
- .auto This is a ‘‘meta-type’’ and is the default file type if the
- user does not specify one. This file type attempts to guess
- the real type by looking for magic words in the header. If
- the type can’t be guessed, the program exits with an error
- message. The input must be a plain file, not a pipe. This
- type can’t be used for output files.
-
-EFFECTS
- Multiple effects may be applied to the audio data by specifying them
- one after another at the end of the command line.
-
- avg [ -l | -r | -f | -b | -1 | -2 | -3 | -4 | n,n,...,n ]
- Reduce the number of channels by averaging the samples, or
- duplicate channels to increase the number of channels. This
- effect is automatically used when the number of input chan-
- nels differ from the number of output channels. When reduc-
- ing the number of channels it is possible to manually specify
- the avg effect and use the -l, -r, -f, -b, -1, -2, -3, -4,
- options to select only the left, right, front, back chan-
- nel(s) or specific channel for the output instead of averag-
- ing the channels. The -l, and -r options will do averaging
- in quad-channel files so select the exact channel to prevent
- this.
-
- The avg effect can also be invoked with up to 16 double-pre-
- cision numbers, seperated by commas, which specify the pro-
- portion (0.0 = 0% and 1.0 = 100%) of each input channel that
- is to be mixed into each output channel. In two-channel
- mode, 4 numbers are given: l->l, l->r, r->l, and r->r,
- respectively. In four-channel mode, the first 4 numbers give
- the proportions for the left-front output channel, as fol-
- lows: lf->lf, rf->lf, lb->lf, and rb->rf. The next 4 give
- the right-front output in the same order, then left-back and
- right-back.
-
- It is also possible to use the 16 numbers to expand or reduce
- the channel count; just specify 0 for unused channels.
-
- Finally, certain reduced combination of numbers can be
- specified for certain input/output channel combinations.
-
-
- In Ch Out Ch Num Mappings
- _____ ______ ___ _____________________________
- 2 1 2 l->l, r->l
- 2 2 1 adjust balance
- 4 1 4 lf->l, rf->l, lb->l, rb-l
- 4 2 2 lf->l&rf->r, lb->l&rb->r
- 4 4 1 adjust balance
- 4 4 2 front balance, back balance
-
-
- band [ -n ] center [ width ]
- Apply a band-pass filter. The frequency response drops loga-
- rithmically 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 amplitudes.
- Band defaults to a mode oriented to pitched signals, i.e.
- voice, singing, or instrumental music. The -n (for noise)
- option uses the alternate mode for un-pitched signals. Warn-
- ing: -n introduces a power-gain of about 11dB in the filter,
- so beware of output clipping. 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 effect with
- steeper shoulders.
-
- bandpass frequency bandwidth
- Butterworth bandpass filter. Description coming soon!
-
- bandreject frequency bandwidth
- Butterworth bandreject filter. Description coming soon!
-
- 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 milliseconds and
- the decay (relative to gain-in) with a modulation speed in Hz
- using depth in milliseconds. The modulation is either sinu-
- soidal (-s) or triangular (-t). Gain-out is the volume of
- the output.
-
- compand attack1,decay1[,attack2,decay2...]
-
- in-dB1,out-dB1[,in-dB2,out-dB2...]
-
- [gain [initial-volume [delay ] ] ]
- 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; attacks refer to increases in volume
- and decays refer to decreases. 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 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 function does not have to be monotonically rising.
- The special value -inf may be used to indicate that the input
- volume should be associated output 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 post-processing 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 per-
- mits the user to supply a nominal level initially, so that,
- for example, a very large gain is not applied to initial sig-
- nal 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.
-
- The fifth (optional) parameter is a delay in seconds. The
- input signal is analyzed immediately to control the compan-
- der, but it is delayed before being fed to the volume
- adjuster. Specifying a delay approximately equal to the
- attack/decay times allows the compander to effectively oper-
- ate in a "predictive" rather than a reactive mode.
-
- copy Copy the input file to the output file. This is the default
- effect if both files have the same sampling rate.
-
- dcshift shift [ limitergain ]
- DC Shift the audio data, with basic linear amplitude formula.
- This is most useful if your audio data tends to not be cen-
- tered around a value of 0. Shifting it back will allow you
- to get the most volume adjustments without clipping audio
- data.
- The first option is the dcshift value. It is a floating
- point number that indicates the amount to shift.
- An option limtergain value can be specified as well. It
- should have a value much less then 1.0 and is used only on
- peaks to prevent clipping.
-
- deemph Apply a treble attenuation shelving filter to samples in
- audio cd format. The frequency response of pre-emphasized
- recordings is rectified. The filtering is defined in the
- standard document ISO 908.
-
- earwax Makes sound easier to listen to on headphones. Adds audio-
- cues to samples in audio cd format so that when listened to
- on headphones the stereo image is moved from inside your head
- (standard for headphones) to outside and in front of the lis-
- tener (standard for speakers). See
- www.geocities.com/beinges for a full explanation.
-
- 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 decay (relative to gain-in)
- of that echo. Gain-out is the volume of the output.
-
- 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 echo. Gain-out is the volume of the out-
- put.
-
- fade [ type ] fade-in-length
-
- [ stop-time [ fade-out-length ] ]
- Add a fade effect to the beginning, end, or both of the audio
- data.
-
- For fade-ins, this starts from the first sample and ramps the
- volume of the audio from 0 to full volume over fade-in-length
- seconds. Specify 0 seconds if no fade-in is wanted.
-
- For fade-outs, the audio data will be truncated at the stop-
- time and the volume will be ramped from full volume down to 0
- starting at fade-out-length seconds before the stop-time. If
- fade-out-length is not specified, it defaults to the same
- value as fade-in-length. No fade-out is performed if the
- stop-time is not specified.
- All times can be specified in either periods of time or sam-
- ple counts. To specify time periods use the format
- hh:mm:ss.frac format. To specify using sample counts, spec-
- ify the number of samples and append the letter ’s’ to the
- sample count (for example 8000s).
- An optional type can be specified to change the type of enve-
- lope. Choices are q for quarter of a sinewave, h for half a
- sinewave, t for linear slope, l for logarithmic, and p for
- inverted parabola. The default is a linear slope.
-
- 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 fre-
- quency 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 unspeci-
- fied, or 0, or greater than or equal to the Nyquist fre-
- quency.
-
- The window-len, if unspecified, defaults to 128. Longer win-
- dows 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 Nuttall 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 milliseconds and the
- decay (relative to gain-in) with a modulation speed in Hz.
- The modulation is either sinodial (-s) or triangular (-t).
- Gain-out is the volume of the output.
-
- highp frequency
- Apply a single pole recursive high-pass filter. The fre-
- quency response drops logarithmically with I frequency 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 coming soon!
-
- lowp frequency
- Apply a single pole recursive low-pass filter. The frequency
- response drops logarithmically with frequency in the middle
- of the drop. The slope of the filter is quite gentle. See
- filter for a lowpass effect with sharper cutoff.
-
- lowpass frequency
- Butterworth lowpass filter. Description coming soon!
-
- 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.
-
- mcompand "attack1,decay1[,attack2,decay2...]
-
- in-dB1,out-dB1[,in-dB2,out-dB2...]
-
- [gain [initial-volume [delay ] ] ]" xover_freq
-
- Multi-band compander is similar to the single band compander
- but the audio file is first divided up into bands and then
- the compander is ran on each band. See the compand effect
- for definition of its options. Compand options are specified
- between double quotes and the crossover frequency for that
- band is specefied seperately with xover_fre. This can be
- repeated multiple times to create multiple bands.
-
- noiseprof [profile-file]
-
- noisered profile-file [threshold]
- Noise reduction filter with profiling. This filter is moder-
- ately effective at removing consistent background noise such
- as hiss or hum. To use it, first run the noiseprof effect on
- a section of silence (that is, a section which contains noth-
- ing but noise). The noiseprof effect will print a noise pro-
- file to profile-file, or to stdout if no profile-file is
- specified. If there is sound output on stdout then the pro-
- file will instead be directed to stderr.
-
- To actually remove the noise, run SoX again with the noisered
- filter. The filter needs one argument, profile-file, which
- contains the noise profile from noiseprof. thershold speci-
- fies how much noise should be removed, and may be between 0
- and 1 with a default of 0.5. Higher values will remove more
- noise but present a greater possibility of distorting the
- desired audio signal. Experiment with different threshold
- values to find the optimal one for your sample.
-
- 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 han-
- dle this. For instance, if the input contains 1 channel and
- the output contains 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 milliseconds and the
- decay (relative to gain-in) with a modulation speed in Hz.
- The modulation is either sinodial (-s) or triangular (-t).
- The decay should be less than 0.5 to avoid feedback. Gain-
- out is the volume of the output.
-
- pick [ -1 | -2 | -3 | -4 | -l | -r | -f | -b ]
- Pick a subset of channels to be copied into the output file.
- This effect is just an alias of the "avg" effect but is left
- here for historical reasons.
-
- pitch shift [ width interpole fade ]
- 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 be "cubic" or "linear". Default
- is "cubic". The fade option, can be "cos", "hamming", "lin-
- ear" or "trapezoid". Default is "cos".
-
- polyphase [ -w < nut / ham > ]
-
- [ -width < long / short / # > ]
-
- [ -cutoff # ]
- Translate input sampling rate to output sampling rate via
- polyphase interpolation, a DSP algorithm. This method is
- slow and uses lots of RAM, but gives much better results than
- rate.
-
- -w < nut / ham > : select either a Nuttal (~90 dB stopband)
- or Hamming (~43 dB stopband) window. Default is nut.
-
- -width long / short / # : specify the (approximate) width of
- the filter. long is 1024 samples; 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.
-
- -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 infor-
- mation on Nyquist frequency. If upsampling, then this is the
- fraction of the original signal that should go through. If
- downsampling, this is the fraction of the signal left after
- downsampling. Default is 0.95. Remember that this is a
- float.
-
-
- rate Translate input sampling rate to output sampling rate via
- linear interpolation to the Least Common Multiple of the two
- sampling rates. This is the default effect if the two files
- have different sampling rates and the preview options was
- specified. This is fast but noisy: the spectrum of the orig-
- inal sound will be shifted upwards and duplicated faintly
- when up-translating by a multiple.
-
- Lerp-ing is acceptable for cheap 8-bit sound hardware, but
- for CD-quality sound you should instead use either resample
- or polyphase. If you are wondering which rate changing
- effects to use, you will want to read a detailed analysis of
- all of them at http://leute.server.de/wilde/resample.html
-
- repeat count
- Repeats the audio data count times. Requires disk space to
- store the data to be repeated.
-
- 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 better 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 rejec-
- tion 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 in about 24 bits precision. The -qs,
- -q, or -ql options specify increased accuracy at the cost of
- lower execution speed. It is optional to specify rolloff and
- beta parameters when using the -q* options.
-
- Following is a table of the reasonable defaults which are
- built-in to SoX:
-
- Option Window rolloff beta interpolation
- ------ ------ ------- ---- -------------
- (none) 45 0.80 16 linear
- -qs 45 0.80 16 quadratic
- -q 75 0.875 16 quadratic
- -ql 149 0.94 16 quadratic
- ------ ------ ------- ---- -------------
-
- -qs, -q, or -ql use window lengths of 45, 75, or 149 samples,
- respectively, at the lower sample-rate of the two files.
- This means progressively sharper stop-band rejection, at pro-
- portionally slower execution times.
-
- rolloff refers to the cut-off frequency of the low pass fil-
- ter 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). Logi-
- cally, this is because the A/D converter needs at least 2
- samples to detect 1 cycle at the Nyquist frequency. Frequen-
- cies 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 fil-
- ter first to avoid these problems.
-
- Similar problems will happen in software when reducing the
- sample rate of an audio file (frequencies 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.
-
- The rolloff refers to how close to the Nyquist frequency this
- cutoff is, with closer being better. When increasing the
- sample rate of an audio file you would not expect to have any
- frequencies exist that are past the original Nyquist fre-
- quency. Because of resampling properties, it is common to
- have aliasing data created that is above the old Nyquist fre-
- quency. In that case the rolloff refers to how close to the
- original Nyquist frequency to use a highpass filter to remove
- this false data, with closer also being better.
-
- The beta parameter determines the type of filter window used.
- Any value greater than 2.0 is the beta for a Kaiser window.
- Beta <= 2.0 selects a Nuttall window. If unspecified, the
- default is a Kaiser window with beta 16.
-
- In the case of Kaiser window (beta > 2.0), lower betas pro-
- duce 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 cut-
- off 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 are, as indicated above,
- Kaiser window of length 45, rolloff 0.80, beta 16, linear
- interpolation.
-
- NOTE: -qs is only slightly slower, but more accurate for
- 16-bit or higher precision.
-
- NOTE: In many cases of up-sampling, no interpolation is
- needed, as exact filter coefficients can be computed in a
- reasonable amount of space. To be precise, this is done when
-
- input_rate < output_rate
- &&
- output_rate/gcd(input_rate,output_rate) <= 511
-
- reverb gain-out reverbe-time 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 reverberation.
- Gain-out is the volume of the output.
-
- reverse Reverse the sound sample completely. Included for finding
- Satanic subliminals.
-
- silence above_periods [ duration threshold[ d | % ]
-
- [ below_periods duration
-
- threshold[ d | % ]]
- Removes silence from the beginning, middle, or end of a sound
- file. Silence is anything below a specified threshold.
-
- The above_periods value is used to indicate if sound should
- be trimmed at the beginning of the audio file. A value of
- zero indicates no silence should be trimmed from the begin-
- ning. When specifing an non-zero above_periods, it trims
- audio up until it finds non-silence. Normally, when trimming
- silence from beginning of audio the above_periods will be 1
- but it can be increased to higher values to trim all data up
- to a specific count of non-silence periods. For example, if
- you had an audio file with two songs that each contained 2
- seconds of silence before the song, you could specify an
- above_period of 2 to strip out both silence periods and the
- first song.
-
- When above_periods is non-zero, you must also specify a dura-
- tion and threshold. Duration indications the amount of time
- that non-silence must be detected before it stops trimming
- data. By increasing the duration, burst of noise can be
- treated as silence and trimmed off.
-
- Threshold is used to indicate what sample value you should
- treat as silence. For digital audio, a value of 0 may be
- fine but for audio recorded from analog, you may wish to
- increase ths value to account for background noise.
-
- When optionally trimming silence from the end of a sound
- file, you specify a below_periods count. In this case,
- below_period means to remove all audio data after silence is
- detected. Normally, this will be a value 1 of but it can be
- increased to skip over periods of silence that are wanted.
- For example, if you have a song with 2 seconds of silence in
- the middle and 2 second at the end, you could set
- below_period to a value of 2 to skip over the silence in the
- middle of the audio file.
-
- For below_periods, duration specifies a period of silence
- that must exist before data is not copied any more. By spec-
- ifying a higher duration, silence that is wanted can be left
- in the audio. For example, if you have a song with an
- expected 1 second of silence in the middle and 2 seconds of
- silence at the end, a duration of 2 seconds could be used to
- skip over the middle silence.
-
- Unfortunetly, you must know the length of the silence at the
- end of your audio file to trim off silence reliably. A work
- around is to use the silence effect in combination with the
- reverse effect. By first reversing the audio, you can use
- the above_periods to reliably trim all audio from what looks
- like the front of the file. Then reverse the file again to
- get back to normal.
-
- To remove silence from the middle of a file, specify a
- below_periods that is negative. This value is then treated
- as a positive value and is also used to indicate the effect
- should restart processing as specified by the above_periods,
- making it suitable for removing periods of silence in the
- middle of the sound file.
-
- The period counts are in units of samples. Duration counts
- may be in the format of hh:mm:ss.frac, or the exact count of
- samples. Threshold numbers may be suffixed iwth d, or % to
- indicate the value is in decibels or a percentage of maximum
- value of the sample value (0% specifies pure digital
- silence).
-
- speed [ -c ] factor
- 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 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 doubles and pitch is one
- octave lower. If the optional -c parameter is used then the
- factor is specified in "cents".
-
- stat [ -s n ] [-rms ] [ -v ] [ -d ]
- 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 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 Adjustment:" 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.
-
-
- stretch factor [window fade shift fading]
- Time stretch file by a given factor. Change duration without
- affecting the pitch. factor of stretching: >1.0 lengthen,
- <1.0 shorten duration. window size is in ms. Default is
- 20ms. The fade option, can be "lin". shift ratio, in [0.0
- 1.0]. Default depends on stretch factor. 1.0 to shorten, 0.8
- to lengthen. The fading ratio, in [0.0 0.5]. The amount of a
- fade’s default depends on factor and shift.
-
- swap [ 1 2 | 1 2 3 4 ]
- Swap channels in multi-channel sound files. Optionally, you
- may specify the channel order you would like the output in.
- This defaults to output channel 2 and then 1 for stereo and
- 2, 1, 4, 3 for quad-channels. An interesting feature is that
- you may duplicate a given channel by overwriting another.
- This is done by repeating an output channel on the command
- line. For example, swap 2 2 will overwrite channel 1 with
- channel 2’s data; creating a stereo file with both channels
- containing the same audio data.
-
- synth [ length ] type mix [ freq [ -freq2 ]
-
- [ off ] [ ph ] [ p1 ] [ p2 ] [ p3 ]
- The synth effect will generate various types of audio data.
- Although this effect is used to generate audio data, an input
- file must be specified. The length of the input audio file
- determines the length of the output audio file.
- <length> length in sec or hh:mm:ss.frac, 0=inputlength,
- default=0
- <type> is sine, square, triangle, sawtooth, trapetz, exp,
- whitenoise, pinknoise, brownnoise, default=sine
- <mix> is create, mix, amod, default=create
- <freq> frequency at beginning in Hz, not used for noise..
- <freq2> frequency at end in Hz, not used for noise..
- <freq/2> can be given as %%n, where ’n’ is the number of half
- notes in respect to A (440Hz)
- <off> Bias (DC-offset) of signal in percent, default=0
- <ph> phase shift 0..100 shift phase 0..2*Pi, not used for
- noise..
- <p1> square: Ton/Toff, triangle+trapetz: rising slope time
- (0..100)
- <p2> trapetz: ON time (0..100)
- <p3> trapetz: falling slope position (0..100)
-
- trim start [ length ]
- Trim can trim off unwanted audio data from the beginning and
- end of the audio file. Audio samples are not sent to the
- output stream until the start location is reached.
- The optional length parameter tells the number of samples to
- output after the start sample and is used to trim off the
- back side of the audio data. Using a value of 0 for the
- start parameter will allow trimming off the back side only.
- Both options can be specified using either an amount of time
- and an exact count of samples. The format for specifying
- lengths in time is hh:mm:ss.frac. A start value of 1:30.5
- will not start until 1 minute, thirty and 1/2 seconds into
- the audio data. The format for specifying sample counts is
- the number of samples with the letter ’s’ appended to it. A
- value of 8000s will wait until 8000 samples are read before
- starting to process audio data.
-
- vibro speed [ depth ]
- Add the world-famous Fender Vibro-Champ sound effect to a
- sound sample by using a sine wave as the volume knob. Speed
- gives the Hertz value of 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 [ limitergain ] ]
- 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. If type is not specified then it defaults to
- amplitude.
- 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 volume
- to increase. Beware of clipping audio data when the gain is
- greater then 1.0. A negative value performs the same adjust-
- ment 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 changed logarithmically.
- 0.0 is constant while +6 doubles the amplitude.
- An optional limitergain value can be specified and should be
- a value much less then 1.0 (ie 0.05 or 0.02) and is used only
- on peaks to prevent clipping. Not specifying this parameter
- will cause no limiter to be used. In verbose mode, this
- effect will display the percentage of audio data that needed
- to be limited.
-
-BUGS
- Please report any bugs found in this version of SoX mailing list (sox-
- [email protected])
-
-SEE ALSO
- play(1), rec(1), soxexam(1)
-
- The SoX web page at http://sox.sourceforge.net/
-
-LICENSE
- Copyright 2006 by Chris Bagwell
-
- This program is free software; you can redistribute it and/or modify it
- under the terms of the GNU General Public License as published by the
- Free Software Foundation; either version 2, or (at your option) any
- later version.
-
- This program is distributed in the hope that it will be useful, but
- WITHOUT ANY WARRANTY; without even the implied warranty of MER-
- CHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General
- Public License for more details.
-
-AUTHORS
- Chris Bagwell ([email protected]).
-
- Additional Authors and contributors are listed in the Changelog file
- that is distributed with the source code.
-
-
-
-sox December 11, 2001 SoX(1)
--- a/soxexam.txt
+++ /dev/null
@@ -1,495 +1,0 @@
-SoX(1) SoX(1)
-
-
-
-NAME
- soxexam - SoX Examples (CHEAT SHEET)
-
-CONVERSIONS
- Introduction
-
- In general, SoX will attempt to take an input sound file format and
- convert it into a new file format using a similar data type and sample
- rate. For instance, "sox monkey.au monkey.wav" would try and convert
- the mono 8000Hz u-law sample .au file that comes with SoX to a 8000Hz
- u-law .wav file.
-
- If an output format doesn’t support the same data type as the input
- file then SoX will generally select a default data type to save it in.
- You can override the default data type selection by using command line
- options. This is also useful for producing an output file with higher
- or lower precision data and/or sample rate.
-
- Most file formats that contain headers can automatically be read in.
- When working with header-less file formats then a user must manually
- tell SoX the data type and sample rate using command line options.
-
- When working with header-less files (raw files), you may take advantage
- of the pseudo-file types of .ub, .uw, .sb, .sw, .ul, and .sl. By using
- these extensions on your filenames you will not have to specify the
- corresponding options on the command line.
-
- Precision
-
- The following data types and formats can be represented by their total
- uncompressed bit precision. When converting from one data type to
- another care must be taken to insure it has an equal or greater preci-
- sion. If not then the audio quality will be degraded. This is not
- always a bad thing when your working with things such as voice audio
- and are concerned about disk space or bandwidth of the audio data.
-
- Data Format Precision
- ___________ _________
- unsigned byte 8-bit
- signed byte 8-bit
- u-law 14-bit
- A-law 13-bit
- unsigned word 16-bit
- signed word 16-bit
- ADPCM 16-bit
- GSM 16-bit
- unsigned long 32-bit
- signed long 32-bit
- ___________ _________
-
- Examples
-
- Use the ’-V’ option on all your command lines. It makes SoX print out
- its idea of what is going on. ’-V’ is your friend.
-
- To convert from unsigned bytes at 8000 Hz to signed words at 8000 Hz:
-
- sox -r 8000 -c 1 filename.ub newfile.sw
-
- To convert from Apple’s AIFF format to Microsoft’s WAV format:
-
- sox filename.aiff filename.wav
-
- To convert from mono raw 8000 Hz 8-bit unsigned PCM data to a WAV file:
-
- sox -r 8000 -u -b -c 1 filename.raw filename.wav
-
- SoX may even be used to convert sample rates. Downconverting will
- reduce the bandwidth of a sample, but will reduce storage space on your
- disk. All such conversions are lossy and will introduce some noise.
- You should really pass your sample through a low pass filter prior to
- downconverting as this will prevent alias signals (which would sound
- like additional noise). For example to convert from a sample recorded
- at 11025 Hz to a u-law file at 8000 Hz sample rate:
-
- sox infile.wav -t au -r 8000 -U -b -c 1 outputfile.au
-
- To add a low-pass filter (note use of stdout for output of the first
- stage and stdin for input on the second stage):
-
- sox infile.wav -t raw -s -w -c 1 - lowpass 3700 |
- sox -t raw -r 11025 -s -w -c 1 - -t au -r 8000 -U -b -c 1 ofile.au
-
- If you hear some clicks and pops when converting to u-law or A-law,
- reduce the output level slightly, for example this will decrease it by
- 20%:
-
- sox infile.wav -t au -r 8000 -U -b -c 1 -v .8 outputfile.au
-
-
- SoX is great to use along with other command line programs by passing
- data between the programs using pipelines. The most common example is
- to use mpg123 to convert mp3 files in to wav files. The following com-
- mand line will do this:
-
- mpg123 -b 10000 -s filename.mp3 | sox -t raw -r 44100 -s -w -c 2 -
- filename.wav
-
- When working with totally unknown audio data then the "auto" file for-
- mat may be of use. It attempts to guess what the file type is and then
- you may save it into a known audio format.
-
- sox -V -t auto filename.snd filename.wav
-
- It is important to understand how the internals of SoX work with com-
- pressed audio including u-law, A-law, ADPCM, or GSM. SoX takes ALL
- input data types and converts them to uncompressed 32-bit signed data.
- It will then convert this internal version into the requested output
- format. This means additional noise can be introduced from decompress-
- ing data and then recompressing. If applying multiple effects to audio
- data, it is best to save the intermediate data as PCM data. After the
- final effect is performed, then you can specify it as a compressed out-
- put format. This will keep noise introduction to a minimum.
-
- The following example applies various effects to an 8000 Hz ADPCM input
- file and then end up with the final file as 44100 Hz ADPCM.
-
- sox firstfile.wav -r 44100 -s -w secondfile.wav
- sox secondfile.wav thirdfile.wav swap
- sox thirdfile.wav -a -b finalfile.wav mask
-
- Under a DOS shell, you can convert several audio files to an new output
- format using something similar to the following command line:
-
- FOR %X IN (*.RAW) DO sox -r 11025 -w -s -t raw $X $X.wav
-
-EFFECTS
- Special thanks goes to Juergen Mueller ([email protected]) for this
- write up on effects.
-
- Introduction:
-
- The core problem is that you need some experience in using effects in
- order to say "that any old sound file sounds with effects absolutely
- hip". There isn’t any rule-based system which tell you the correct set-
- ting of all the parameters for every effect. But after some time you
- will become an expert in using effects.
-
- Here are some examples which can be used with any music sample. (For a
- sample where only a single instrument is playing, extreme parameter
- setting may make well-known "typically" or "classical" sounds. Like-
- wise, for drums, vocals or guitars.)
-
- Single effects will be explained and some given parameter settings that
- can be used to understand the theory by listening to the sound file
- with the added effect.
-
- Using multiple effects in parallel or in series can result either in a
- very nice sound or (mostly) in a dramatic overloading in variations of
- sounds such that your ear may follow the sound but you will feel unsat-
- isfied. Hence, for the first time using effects try to compose them as
- minimally as possible. We don’t regard the composition of effects in
- the examples because too many combinations are possible and you really
- need a very fast machine and a lot of memory to play them in real-time.
-
- However, real-time playing of sounds will greatly speed up learning
- and/or tuning the parameter settings for your sounds in order to get
- that "perfect" effect.
-
- Basically, we will use the "play" front-end of SoX since it is easier
- to listen sounds coming out of the speaker or earphone instead of look-
- ing at cryptic data in sound files.
-
- For easy listening of file.xxx ("xxx" is any sound format):-
- play file.xxx effect-name effect-parameters
-
- Or more SoX-like (for "dsp" output on a UNIX/Linux computer):
-
- sox file.xxx -t ossdsp -w -s /dev/dsp effect-name effect-parame-
- ters
-
- or (for "au" output):
-
- sox file.xxx -t sunau -w -s /dev/audio effect-name effect-parame-
- ters
-
- And for date freaks:
-
- sox file.xxx file.yyy effect-name effect-parameters
-
- Additional options can be used. However, in this case, for real-time
- playing you’ll need a very fast machine.
-
- Notes:
-
- I played all examples in real-time on a Pentium 100 with 32 MB and
- Linux 2.0.30 using a self-recorded sample ( 3:15 min long in "wav" for-
- mat with 44.1 kHz sample rate and stereo 16 bit ). The sample should
- not contain any of the effects. However, if you take any recording of a
- sound track from radio or tape or CD, and it sounds like a live concert
- or ten people are playing the same rhythm with their drums or funky-
- grooves, then take any other sample. (Typically, less then four dif-
- ferent instruments and no synthesizer in the sample is suitable. Like-
- wise, the combination vocal, drums, bass and guitar.)
-
- Effects:
-
- Echo
-
- An echo effect can be naturally found in the mountains, standing some-
- where on a mountain and shouting a single word will result in one or
- more repetitions of the word (if not, turn a bit around and try again,
- or climb to the next mountain).
-
- However, the time difference between shouting and repeating is the
- delay (time), its loudness is the decay. Multiple echos can have dif-
- ferent delays and decays.
-
- It is very popular to use echos to play an instrument with itself
- together, like some guitar players (Brain May from Queen) or vocalists
- are doing. For music samples of more than one instrument, echo can be
- used to add a second sample shortly after the original one.
-
- This will sound as if you are doubling the number of instruments play-
- ing in the same sample:
-
- play file.xxx echo 0.8 0.88 60.0 0.4
-
- If the delay is very short, then it sound like a (metallic) robot play-
- ing music:
-
- play file.xxx echo 0.8 0.88 6.0 0.4
-
- Longer delay will sound like an open air concert in the mountains:
-
- play file.xxx echo 0.8 0.9 1000.0 0.3
-
- One mountain more, and:
-
- play file.xxx echo 0.8 0.9 1000.0 0.3 1800.0 0.25
-
- Echos
-
- Like the echo effect, echos stand for "ECHO in Sequel", that is the
- first echos takes the input, the second the input and the first echos,
- the third the input and the first and the second echos, ... and so on.
- Care should be taken using many echos (see introduction); a single
- echos has the same effect as a single echo.
-
- The sample will be bounced twice in symmetric echos:
-
- play file.xxx echos 0.8 0.7 700.0 0.25 700.0 0.3
-
- The sample will be bounced twice in asymmetric echos:
-
- play file.xxx echos 0.8 0.7 700.0 0.25 900.0 0.3
-
- The sample will sound as if played in a garage:
-
- play file.xxx echos 0.8 0.7 40.0 0.25 63.0 0.3
-
- Chorus
-
- The chorus effect has its name because it will often be used to make a
- single vocal sound like a chorus. But it can be applied to other
- instrument samples too.
-
- It works like the echo effect with a short delay, but the delay isn’t
- constant. The delay is varied using a sinusoidal or triangular modula-
- tion. The modulation depth defines the range the modulated delay is
- played before or after the delay. Hence the delayed sound will sound
- slower or faster, that is the delayed sound tuned around the original
- one, like in a chorus where some vocals are a bit out of tune.
-
- The typical delay is around 40ms to 60ms, the speed of the modulation
- is best near 0.25Hz and the modulation depth around 2ms.
-
- A single delay will make the sample more overloaded:
-
- play file.xxx chorus 0.7 0.9 55.0 0.4 0.25 2.0 -t
-
- Two delays of the original samples sound like this:
-
- play file.xxx chorus 0.6 0.9 50.0 0.4 0.25 2.0 -t 60.0 0.32 0.4
- 1.3 -s
-
- A big chorus of the sample is (three additional samples):
-
- play file.xxx chorus 0.5 0.9 50.0 0.4 0.25 2.0 -t 60.0 0.32 0.4
- 2.3 -t 40.0 0.3 0.3 1.3 -s
-
- Flanger
-
- The flanger effect is like the chorus effect, but the delay varies
- between 0ms and maximal 5ms. It sound like wind blowing, sometimes
- faster or slower including changes of the speed.
-
- The flanger effect is widely used in funk and soul music, where the
- guitar sound varies frequently slow or a bit faster.
-
- The typical delay is around 3ms to 5ms, the speed of the modulation is
- best near 0.5Hz.
-
- Now, let’s groove the sample:
-
- play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -s
-
- listen carefully between the difference of sinusoidal and triangular
- modulation:
-
- play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -t
-
- If the decay is a bit lower, than the effect sounds more popular:
-
- play file.xxx flanger 0.8 0.88 3.0 0.4 0.5 -t
-
- The drunken loudspeaker system:
-
- play file.xxx flanger 0.9 0.9 4.0 0.23 1.3 -s
-
- Reverb
-
- The reverb effect is often used in audience hall which are to small or
- contain too many many visitors which disturb (dampen) the reflection of
- sound at the walls. Reverb will make the sound be perceived as if it
- were in a large hall. You can try the reverb effect in your bathroom
- or garage or sport halls by shouting loud some words. You’ll hear the
- words reflected from the walls.
-
- The biggest problem in using the reverb effect is the correct setting
- of the (wall) delays such that the sound is realistic and doesn’t sound
- like music playing in a tin can or has overloaded feedback which
- destroys any illusion of playing in a big hall. To help you obtain
- realistic reverb effects, you should decide first how long the reverb
- should take place until it is not loud enough to be registered by your
- ears. This is be done by varying the reverb time "t". To simulate
- small halls, use 200ms. To simulate large halls, use 1000ms. Clearly,
- the walls of such a hall aren’t far away, so you should define its set-
- ting be given every wall its delay time. However, if the wall is to
- far away for the reverb time, you won’t hear the reverb, so the nearest
- wall will be best at "t/4" delay and the farthest at "t/2". You can try
- other distances as well, but it won’t sound very realistic. The walls
- shouldn’t stand to close to each other and not in a multiple integer
- distance to each other ( so avoid wall like: 200.0 and 202.0, or some-
- thing like 100.0 and 200.0 ).
-
- Since audience halls do have a lot of walls, we will start designing
- one beginning with one wall:
-
- play file.xxx reverb 1.0 600.0 180.0
-
- One wall more:
-
- play file.xxx reverb 1.0 600.0 180.0 200.0
-
- Next two walls:
-
- play file.xxx reverb 1.0 600.0 180.0 200.0 220.0 240.0
-
- Now, why not a futuristic hall with six walls:
-
- play file.xxx reverb 1.0 600.0 180.0 200.0 220.0 240.0 280.0
- 300.0
-
- If you run out of machine power or memory, then stop as many applica-
- tions as possible (every interrupt will consume a lot of CPU time which
- for bigger halls is absolutely necessary).
-
- Phaser
-
- The phaser effect is like the flanger effect, but it uses a reverb
- instead of an echo and does phase shifting. You’ll hear the difference
- in the examples comparing both effects (simply change the effect name).
- The delay modulation can be sinusoidal or triangular, preferable is the
- later for multiple instruments. For single instrument sounds, the sinu-
- soidal phaser effect will give a sharper phasing effect. The decay
- shouldn’t be to close to 1.0 which will cause dramatic feedback. A
- good range is about 0.5 to 0.1 for the decay.
-
- We will take a parameter setting as for the flanger before (gain-out is
- lower since feedback can raise the output dramatically):
-
- play file.xxx phaser 0.8 0.74 3.0 0.4 0.5 -t
-
- The drunken loudspeaker system (now less alcohol):
-
- play file.xxx phaser 0.9 0.85 4.0 0.23 1.3 -s
-
- A popular sound of the sample is as follows:
-
- play file.xxx phaser 0.89 0.85 1.0 0.24 2.0 -t
-
- The sample sounds if ten springs are in your ears:
-
- play file.xxx phaser 0.6 0.66 3.0 0.6 2.0 -t
-
- Compander
-
- The compander effect allows the dynamic range of a signal to be com-
- pressed or expanded. For most situations, the attack time (response to
- the music getting louder) should be shorter than the decay time because
- our ears are more sensitive to suddenly loud music than to suddenly
- soft music.
-
- For example, suppose you are listening to Strauss’ "Also Sprach
- Zarathustra" in a noisy environment such as a car. If you turn up the
- volume enough to hear the soft passages over the road noise, the loud
- sections will be too loud. You could try this:
-
- play file.xxx compand 0.3,1 -90,-90,-70,-70,-60,-20,0,0 -5 0 0.2
-
- The transfer function ("-90,...") says that very soft sounds between- -90 and -70 decibels (-90 is about the limit of 16-bit encoding) will
- remain unchanged. That keeps the compander from boosting the volume on
- "silent" passages such as between movements. However, sounds in the
- range -60 decibels to 0 decibels (maximum volume) will be boosted so
- that the 60-dB dynamic range of the original music will be compressed
- 3-to-1 into a 20-dB range, which is wide enough to enjoy the music but
- narrow enough to get around the road noise. The -5 dB output gain is
- needed to avoid clipping (the number is inexact, and was derived by
- experimentation). The 0 for the initial volume will work fine for a
- clip that starts with a bit of silence, and the delay of 0.2 has the
- effect of causing the compander to react a bit more quickly to sudden
- volume changes.
-
- Changing the Rate of Playback
-
- You can use stretch to change the rate of playback of an audio sample
- while preserving the pitch. For example to play at 1/2 the speed:
-
- play file.wav stretch 2
-
- To play a file at twice the speed:
-
- play file.wav stretch .5
-
- Other related options are "speed" to change the speed of play (and
- changing the pitch accordingly), and pitch, to alter the pitch of a
- sample. For example to speed a sample so it plays in 1/2 the time (for
- those Mickey Mouse voices):
-
- play file.wav speed 2
-
- To raise the pitch of a sample 1 while note (100 cents):
-
- play file.wav pitch 100
-
-
-
- Reducing noise in a recording
-
- First find a period of silence in your recording, such as the beginning
- or end of a piece. If the first 1.5 seconds of the recording are
- silent, do
-
-
- sox file.wav -t nul /dev/null trim 0 1.5 noiseprof /tmp/profile
-
- Next, use the noisered effect to actually reduce the noise:
-
-
- play file.wav noisered /tmp/profile
-
-
-
- Other effects (copy, rate, avg, stat, vibro, lowp, highp, band, reverb)
-
- The other effects are simple to use. However, an "easy to use manual"
- should be given here.
-
- More effects (to do !)
-
- There are a lot of effects around like noise gates, compressors, waw-
- waw, stereo effects and so on. They should be implemented, making SoX
- more useful in sound mixing techniques coming together with a great
- variety of different sound effects.
-
- Combining effects by using them in parallel or serially on different
- channels needs some easy mechanism which is stable for use in real-
- time.
-
- Really missing are the the changing of the parameters and start-
- ing/stopping of effects while playing samples in real-time!
-
- Good luck and have fun with all the effects!
-
- Juergen Mueller ([email protected])
-
-
-SEE ALSO
- sox(1), play(1), rec(1)
-
-AUTHOR
- Juergen Mueller ([email protected])
-
- Updates by Anonymous.
-
-
-
- December 11, 2001 SoX(1)