neulab/tldr huggingface dataset:
Some systems provide more than one type of (SoX-compatible) audio driver, e.g. ALSA & OSS, or SUNAU & AO.  Systems can also have more than one audio device (a.k.a. `sound card').  If more than one audio driver has been built-in to SoX, and the default selected by SoX when recording or playing is not the one that is wanted, then the AUDIODRIVER environment variable can be used to override the default.  For example (on many systems): set AUDIODRIVER=oss play ... The AUDIODEV environment variable can be used to override the default audio device, e.g. set AUDIODEV=/dev/dsp2 play ... sox ... -t oss or set AUDIODEV=hw:soundwave,1,2 play ... sox ... -t alsa Note that the way of setting environment variables varies from system to system - for some specific examples, see `SOX_OPTS' below.
When playing a file with a sample rate that is not supported by the audio output device, SoX will automatically invoke the rate effect to perform the necessary sample rate conversion.  For compatibility with old hardware, the default rate quality level is set to `low'. This can be changed by explicitly specifying the rate effect with a different quality level, e.g. play ... rate -m or by using the --play-rate-arg option (see below).
On some systems, SoX allows audio playback volume to be adjusted whilst using play.  Where supported, this is achieved by tapping the `v' & `V' keys during playback.
To help with setting a suitable recording level, SoX includes a peak- level meter which can be invoked (before making the actual recording) as follows: rec -n The recording level should be adjusted (using the system-provided mixer program, not SoX) so that the meter is at most occasionally full scale, and never `in the red' (an exclamation mark is shown).  See also -S below.
Accuracy Many file formats that compress audio discard some of the audio signal information whilst doing so. Converting to such a format and then converting back again will not produce an exact copy of the original audio.  This is the case for many formats used in telephony (e.g. A-law, GSM) where low signal bandwidth is more important than high audio fidelity, and for many formats used in portable music players (e.g. MP3, Vorbis) where adequate fidelity can be retained even with the large compression ratios that are needed to make portable players practical.
Formats that discard audio signal information are called `lossy'. Formats that do not are called `lossless'.  The term `quality' is used as a measure of how closely the original audio signal can be reproduced when using a lossy format.
Audio file conversion with SoX is lossless when it can be, i.e. when not using lossy compression, when not reducing the sampling rate or number of channels, and when the number of bits used in the destination format is not less than in the source format.  E.g.  converting from an 8-bit PCM format to a 16-bit PCM format is lossless but converting from an 8-bit PCM format to (8-bit) A-law isn't.
N.B.  SoX converts all audio files to an internal uncompressed format before performing any audio processing. This means that manipulating a file that is stored in a lossy format can cause further losses in audio fidelity.  E.g. with sox long.mp3 short.mp3 trim 10 SoX first decompresses the input MP3 file, then applies the trim effect, and finally creates the output MP3 file by re-compressing the audio - with a possible reduction in fidelity above that which occurred when the input file was created.  Hence, if what is ultimately desired is lossily compressed audio, it is highly recommended to perform all audio processing using lossless file formats and then convert to the lossy format only at the final stage.
N.B.  Applying multiple effects with a single SoX invocation will, in general, produce more accurate results than those produced using multiple SoX invocations.
Dithering Dithering is a technique used to maximise the dynamic range of audio stored at a particular bit-depth. Any distortion introduced by quantisation is decorrelated by adding a small amount of white noise to the signal.  In most cases, SoX can determine whether the selected processing requires dither and will add it during output formatting if appropriate.
Specifically, by default, SoX automatically adds TPDF dither when the output bit-depth is less than 24 and any of the following are true:
•   bit-depth reduction has been specified explicitly using a command- line option
•   the output file format supports only bit-depths lower than that of the input file format
•   an effect has increased effective bit-depth within the internal processing chain
For example, adjusting volume with vol 0.25 requires two additional bits in which to losslessly store its results (since 0.25 decimal equals 0.01 binary).  So if the input file bit-depth is 16, then SoX's internal representation will utilise 18 bits after processing this volume change. In order to store the output at the same depth as the input, dithering is used to remove the additional bits.
Use the -V option to see what processing SoX has automatically added. The -D option may be given to override automatic dithering.  To invoke dithering manually (e.g. to select a noise-shaping curve), see the dither effect.
Clipping Clipping is distortion that occurs when an audio signal level (or `volume') exceeds the range of the chosen representation.  In most cases, clipping is undesirable and so should be corrected by adjusting the level prior to the point (in the processing chain) at which it occurs.
In SoX, clipping could occur, as you might expect, when using the vol or gain effects to increase the audio volume. Clipping could also occur with many other effects, when converting one format to another, and even when simply playing the audio.
Playing an audio file often involves resampling, and processing by analogue components can introduce a small DC offset and/or amplification, all of which can produce distortion if the audio signal level was initially too close to the clipping point.
For these reasons, it is usual to make sure that an audio file's signal level has some `headroom', i.e. it does not exceed a particular level below the maximum possible level for the given representation.  Some standards bodies recommend as much as 9dB headroom, but in most cases, 3dB (≈ 70% linear) is enough.  Note that this wisdom seems to have been lost in modern music production; in fact, many CDs, MP3s, etc.  are now mastered at levels above 0dBFS i.e. the audio is clipped as delivered.
SoX's stat and stats effects can assist in determining the signal level in an audio file. The gain or vol effect can be used to prevent clipping, e.g. sox dull.wav bright.wav gain -6 treble +6 guarantees that the treble boost will not clip.
If clipping occurs at any point during processing, SoX will display a warning message to that effect.
See also -G and the gain and norm effects.
Input File Combining SoX's input combiner can be configured (see OPTIONS below) to combine multiple files using any of the following methods: `concatenate', `sequence', `mix', `mix-power', `merge', or `multiply'.  The default method is `sequence' for play, and `concatenate' for rec and sox.
For all methods other than `sequence', multiple input files must have the same sampling rate. If necessary, separate SoX invocations can be used to make sampling rate adjustments prior to combining.
If the `concatenate' combining method is selected (usually, this will be by default) then the input files must also have the same number of channels.  The audio from each input will be concatenated in the order given to form the output file.
The `sequence' combining method is selected automatically for play.  It is similar to `concatenate' in that the audio from each input file is sent serially to the output file. However, here the output file may be closed and reopened at the corresponding transition between input files. This may be just what is needed when sending different types of audio to an output device, but is not generally useful when the output is a normal file.
If either the `mix' or `mix-power' combining method is selected then two or more input files must be given and will be mixed together to form the output file.  The number of channels in each input file need not be the same, but SoX will issue a warning if they are not and some channels in the output file will not contain audio from every input file.  A mixed audio file cannot be un-mixed without reference to the original input files.
If the `merge' combining method is selected then two or more input files must be given and will be merged together to form the output file.  The number of channels in each input file need not be the same.  A merged audio file comprises all of the channels from all of the input files. Un- merging is possible using multiple invocations of SoX with the remix effect.  For example, two mono files could be merged to form one stereo file. The first and second mono files would become the left and right channels of the stereo file.
The `multiply' combining method multiplies the sample values of corresponding channels (treated as numbers in the interval -1 to +1).  If the number of channels in the input files is not the same, the missing channels are considered to contain all zero.
When combining input files, SoX applies any specified effects (including, for example, the vol volume adjustment effect) after the audio has been combined. However, it is often useful to be able to set the volume of (i.e. `balance') the inputs individually, before combining takes place.
For all combining methods, input file volume adjustments can be made manually using the -v option (below) which can be given for one or more input files. If it is given for only some of the input files then the others receive no volume adjustment.  In some circumstances, automatic volume adjustments may be applied (see below).
The -V option (below) can be used to show the input file volume adjustments that have been selected (either manually or automatically).
There are some special considerations that need to made when mixing input files:
Unlike the other methods, `mix' combining has the potential to cause clipping in the combiner if no balancing is performed.  In this case, if manual volume adjustments are not given, SoX will try to ensure that clipping does not occur by automatically adjusting the volume (amplitude) of each input signal by a factor of ¹/n, where n is the number of input files.  If this results in audio that is too quiet or otherwise unbalanced then the input file volumes can be set manually as described above. Using the norm effect on the mix is another alternative.
If mixed audio seems loud enough at some points but too quiet in others then dynamic range compression should be applied to correct this - see the compand effect.
With the `mix-power' combine method, the mixed volume is approximately equal to that of one of the input signals.  This is achieved by balancing using a factor of ¹/√n instead of ¹/n.  Note that this balancing factor does not guarantee that clipping will not occur, but the number of clips will usually be low and the resultant distortion is generally imperceptible.
Output Files SoX's default behaviour is to take one or more input files and write them to a single output file.
This behaviour can be changed by specifying the pseudo-effect `newfile' within the effects list.  SoX will then enter multiple output mode.
In multiple output mode, a new file is created when the effects prior to the `newfile' indicate they are done.  The effects chain listed after `newfile' is then started up and its output is saved to the new file.
In multiple output mode, a unique number will automatically be appended to the end of all filenames.  If the filename has an extension then the number is inserted before the extension.  This behaviour can be customized by placing a %n anywhere in the filename where the number should be substituted.  An optional number can be placed after the % to indicate a minimum fixed width for the number.
Multiple output mode is not very useful unless an effect that will stop the effects chain early is specified before the `newfile'. If end of file is reached before the effects chain stops itself then no new file will be created as it would be empty.
The following is an example of splitting the first 60 seconds of an input file into two 30 second files and ignoring the rest. sox song.wav ringtone%1n.wav trim 0 30 : newfile : trim 0 30
Stopping SoX Usually SoX will complete its processing and exit automatically once it has read all available audio data from the input files.
If desired, it can be terminated earlier by sending an interrupt signal to the process (usually by pressing the keyboard interrupt key which is normally Ctrl-C).  This is a natural requirement in some circumstances, e.g. when using SoX to make a recording.  Note that when using SoX to play multiple files, Ctrl-C behaves slightly differently: pressing it once causes SoX to skip to the next file; pressing it twice in quick succession causes SoX to exit.
Another option to stop processing early is to use an effect that has a time period or sample count to determine the stopping point. The trim effect is an example of this.  Once all effects chains have stopped then SoX will also stop.
Filenames can be simple file names, absolute or relative path names, or URLs (input files only).  Note that URL support requires that wget(1) is available.
Note: Giving SoX an input or output filename that is the same as a SoX effect-name will not work since SoX will treat it as an effect specification.  The only work-around to this is to avoid such filenames. This is generally not difficult since most audio filenames have a filename `extension', whilst effect-names do not.
Special Filenames The following special filenames may be used in certain circumstances in place of a normal filename on the command line:
-      SoX can be used in simple pipeline operations by using the special filename `-' which, if used as an input filename, will cause SoX will read audio data from `standard input' (stdin), and which, if used as the output filename, will cause SoX will send audio data to `standard output' (stdout).  Note that when using this option for the output file, and sometimes when using it for an input file, the file-type (see -t below) must also be given.
"|program [options] ..." This can be used in place of an input filename to specify the the given program's standard output (stdout) be used as an input file. Unlike - (above), this can be used for several inputs to one SoX command.  For example, if `genw' generates mono WAV formatted signals to its standard output, then the following command makes a stereo file from two generated signals: sox -M "|genw --imd -" "|genw --thd -" out.wav For headerless (raw) audio, -t (and perhaps other format options) will need to be given, preceding the input command.
"wildcard-filename" Specifies that filename `globbing' (wild-card matching) should be performed by SoX instead of by the shell.  This allows a single set of file options to be applied to a group of files.  For example, if the current directory contains three `vox' files, file1.vox, file2.vox, and file3.vox, then play --rate 6k *.vox will be expanded by the `shell' (in most environments) to play --rate 6k file1.vox file2.vox file3.vox which will treat only the first vox file as having a sample rate of 6k.  With play --rate 6k "*.vox" the given sample rate option will be applied to all three vox files.
-p, --sox-pipe This can be used in place of an output filename to specify that the SoX command should be used as in input pipe to another SoX command.  For example, the command: play "|sox -n -p synth 2" "|sox -n -p synth 2 tremolo 10" stat plays two `files' in succession, each with different effects.
-p is in fact an alias for `-t sox -'.
-d, --default-device This can be used in place of an input or output filename to specify that the default audio device (if one has been built into SoX) is to be used.  This is akin to invoking rec or play (as described above).
-n, --null This can be used in place of an input or output filename to specify that a `null file' is to be used.  Note that here, `null file' refers to a SoX-specific mechanism and is not related to any operating-system mechanism with a similar name.
Using a null file to input audio is equivalent to using a normal audio file that contains an infinite amount of silence, and as such is not generally useful unless used with an effect that specifies a finite time length (such as trim or synth).
Using a null file to output audio amounts to discarding the audio and is useful mainly with effects that produce information about the audio instead of affecting it (such as noiseprof or stat).
The sampling rate associated with a null file is by default 48 kHz, but, as with a normal file, this can be overridden if desired using command-line format options (see below).
Supported File & Audio Device Types See soxformat(7) for a list and description of the supported file formats and audio device drivers.
Global Options These options can be specified on the command line at any point before the first effect name.
The SOX_OPTS environment variable can be used to provide alternative default values for SoX's global options.  For example: SOX_OPTS="--buffer 20000 --play-rate-arg -hs --temp /mnt/temp" Note that setting SOX_OPTS can potentially create unwanted changes in the behaviour of scripts or other programs that invoke SoX.  SOX_OPTS might best be used for things (such as in the given example) that reflect the environment in which SoX is being run.  Enabling options such as --no-clobber as default might be handled better using a shell alias since a shell alias will not affect operation in scripts etc.
One way to ensure that a script cannot be affected by SOX_OPTS is to clear SOX_OPTS at the start of the script, but this of course loses the benefit of SOX_OPTS carrying some system-wide default options.  An alternative approach is to explicitly invoke SoX with default option values, e.g. SOX_OPTS="-V --no-clobber" ... sox -V2 --clobber $input $output ... Note that the way to set environment variables varies from system to system. Here are some examples:
Unix bash: export SOX_OPTS="-V --no-clobber" Unix csh: setenv SOX_OPTS "-V --no-clobber" MS-DOS/MS-Windows: set SOX_OPTS=-V --no-clobber MS-Windows GUI: via Control Panel : System : Advanced : Environment Variables
Mac OS X GUI: Refer to Apple's Technical Q&A QA1067 document.
--buffer BYTES, --input-buffer BYTES Set the size in bytes of the buffers used for processing audio (default 8192).  --buffer applies to input, effects, and output processing; --input-buffer applies only to input processing (for which it overrides --buffer if both are given).
Be aware that large values for --buffer will cause SoX to be become slow to respond to requests to terminate or to skip the current input file.
--clobber Don't prompt before overwriting an existing file with the same name as that given for the output file.  This is the default behaviour.
--combine concatenate|merge|mix|mix-power|multiply|sequence Select the input file combining method; for some of these, short options are available: -m selects `mix', -M selects `merge', and -T selects `multiply'.
See Input File Combining above for a description of the different combining methods.
-D, --no-dither Disable automatic dither - see `Dithering' above.  An example of why this might occasionally be useful is if a file has been converted from 16 to 24 bit with the intention of doing some processing on it, but in fact no processing is needed after all and the original 16 bit file has been lost, then, strictly speaking, no dither is needed if converting the file back to 16 bit.  See also the stats effect for how to determine the actual bit depth of the audio within a file.
--effects-file FILENAME Use FILENAME to obtain all effects and their arguments.  The file is parsed as if the values were specified on the command line.  A new line can be used in place of the special : marker to separate effect chains.  For convenience, such markers at the end of the file are normally ignored; if you want to specify an empty last effects chain, use an explicit : by itself on the last line of the file.  This option causes any effects specified on the command line to be discarded.
-G, --guard Automatically invoke the gain effect to guard against clipping. E.g. sox -G infile -b 16 outfile rate 44100 dither -s is shorthand for sox infile -b 16 outfile gain -h rate 44100 gain -rh dither -s See also -V, --norm, and the gain effect.
-h, --help Show version number and usage information.
--help-effect NAME Show usage information on the specified effect.  The name all can be used to show usage on all effects.
--help-format NAME Show information about the specified file format.  The name all can be used to show information on all formats.
--i, --info Only if given as the first parameter to sox, behave as soxi(1).
-m|-M  Equivalent to --combine mix and --combine merge, respectively.
--magic If SoX has been built with the optional `libmagic' library then this option can be given to enable its use in helping to detect audio file types.
--multi-threaded | --single-threaded By default, SoX is `single threaded'.  If the --multi-threaded option is given however then SoX will process audio channels for most multi-channel effects in parallel on hyper-threading/multi- core architectures. This may reduce processing time, though sometimes it may be necessary to use this option in conjunction with a larger buffer size than is the default to gain any benefit from multi-threaded processing (e.g. 131072; see --buffer above).
--no-clobber Prompt before overwriting an existing file with the same name as that given for the output file.
N.B.  Unintentionally overwriting a file is easier than you might think, for example, if you accidentally enter sox file1 file2 effect1 effect2 ... when what you really meant was play file1 file2 effect1 effect2 ... then, without this option, file2 will be overwritten.  Hence, using this option is recommended. SOX_OPTS (above), a `shell' alias, script, or batch file may be an appropriate way of permanently enabling it.
--norm[=dB-level] Automatically invoke the gain effect to guard against clipping and to normalise the audio. E.g. sox --norm infile -b 16 outfile rate 44100 dither -s is shorthand for sox infile -b 16 outfile gain -h rate 44100 gain -nh dither -s Optionally, the audio can be normalized to a given level (usually) below 0 dBFS: sox --norm=-3 infile outfile
See also -V, -G, and the gain effect.
--play-rate-arg ARG Selects a quality option to be used when the `rate' effect is automatically invoked whilst playing audio.  This option is typically set via the SOX_OPTS environment variable (see above).
--plot gnuplot|octave|off If not set to off (the default if --plot is not given), run in a mode that can be used, in conjunction with the gnuplot program or the GNU Octave program, to assist with the selection and configuration of many of the transfer-function based effects.  For the first given effect that supports the selected plotting program, SoX will output commands to plot the effect's transfer function, and then exit without actually processing any audio. E.g. sox --plot octave input-file -n highpass 1320 > highpass.plt octave highpass.plt
-q, --no-show-progress Run in quiet mode when SoX wouldn't otherwise do so.  This is the opposite of the -S option.
-R     Run in `repeatable' mode.  When this option is given, where applicable, SoX will embed a fixed time-stamp in the output file (e.g.  AIFF) and will `seed' pseudo random number generators (e.g. dither) with a fixed number, thus ensuring that successive SoX invocations with the same inputs and the same parameters yield the same output.
--replay-gain track|album|off Select whether or not to apply replay-gain adjustment to input files.  The default is off for sox and rec, album for play where (at least) the first two input files are tagged with the same Artist and Album names, and track for play otherwise.
-S, --show-progress Display input file format/header information, and processing progress as input file(s) percentage complete, elapsed time, and remaining time (if known; shown in brackets), and the number of samples written to the output file.  Also shown is a peak-level meter, and an indication if clipping has occurred.  The peak-level meter shows up to two channels and is calibrated for digital audio as follows (right channel shown):
dB FSD   Display   dB FSD   Display -25     -          -11     ==== -23     =           -9     ====- -21     =-          -7     ===== -19     ==          -5     =====- -17     ==-         -3     ====== -15     ===         -1     =====! -13     ===-
A three-second peak-held value of headroom in dBs will be shown to the right of the meter if this is below 6dB.
This option is enabled by default when using SoX to play or record audio.
-T     Equivalent to --combine multiply.
--temp DIRECTORY Specify that any temporary files should be created in the given DIRECTORY.  This can be useful if there are permission or free- space problems with the default location. In this case, using `--temp .' (to use the current directory) is often a good solution.
--version Show SoX's version number and exit.
-V[level] Set verbosity. This is particularly useful for seeing how any automatic effects have been invoked by SoX.
SoX displays messages on the console (stderr) according to the following verbosity levels:
0      No messages are shown at all; use the exit status to determine if an error has occurred.
1      Only error messages are shown.  These are generated if SoX cannot complete the requested commands.
2      Warning messages are also shown.  These are generated if SoX can complete the requested commands, but not exactly according to the requested command parameters, or if clipping occurs.
3      Descriptions of SoX's processing phases are also shown. Useful for seeing exactly how SoX is processing your audio.
4 and above Messages to help with debugging SoX are also shown.
By default, the verbosity level is set to 2 (shows errors and warnings). Each occurrence of the -V option increases the verbosity level by 1.  Alternatively, the verbosity level can be set to an absolute number by specifying it immediately after the -V, e.g.  -V0 sets it to 0.
Input File Options These options apply only to input files and may precede only input filenames on the command line.
--ignore-length Override an (incorrect) audio length given in an audio file's header. If this option is given then SoX will keep reading audio until it reaches the end of the input file.
-v, --volume FACTOR Intended for use when combining multiple input files, this option adjusts the volume of the file that follows it on the command line by a factor of FACTOR. This allows it to be `balanced' w.r.t. the other input files.  This is a linear (amplitude) adjustment, so a number less than 1 decreases the volume and a number greater than 1 increases it.  If a negative number is given then in addition to the volume adjustment, the audio signal will be inverted.
See also the norm, vol, and gain effects, and see Input File Balancing above.
Input & Output File Format Options These options apply to the input or output file whose name they immediately precede on the command line and are used mainly when working with headerless file formats or when specifying a format for the output file that is different to that of the input file.
-b BITS, --bits BITS The number of bits (a.k.a. bit-depth or sometimes word-length) in each encoded sample.  Not applicable to complex encodings such as MP3 or GSM.  Not necessary with encodings that have a fixed number of bits, e.g.  A/μ-law, ADPCM.
For an input file, the most common use for this option is to inform SoX of the number of bits per sample in a `raw' (`headerless') audio file.  For example sox -r 16k -e signed -b 8 input.raw output.wav converts a particular `raw' file to a self-describing `WAV' file.
For an output file, this option can be used (perhaps along with -e) to set the output encoding size.  By default (i.e. if this option is not given), the output encoding size will (providing it is supported by the output file type) be set to the input encoding size.  For example sox input.cdda -b 24 output.wav converts raw CD digital audio (16-bit, signed-integer) to a 24-bit (signed-integer) `WAV' file.
-c CHANNELS, --channels CHANNELS The number of audio channels in the audio file. This can be any number greater than zero.
For an input file, the most common use for this option is to inform SoX of the number of channels in a `raw' (`headerless') audio file.  Occasionally, it may be useful to use this option with a `headered' file, in order to override the (presumably incorrect) value in the header - note that this is only supported with certain file types.  Examples: sox -r 48k -e float -b 32 -c 2 input.raw output.wav converts a particular `raw' file to a self-describing `WAV' file. play -c 1 music.wav interprets the file data as belonging to a single channel regardless of what is indicated in the file header.  Note that if the file does in fact have two channels, this will result in the file playing at half speed.
For an output file, this option provides a shorthand for specifying that the channels effect should be invoked in order to change (if necessary) the number of channels in the audio signal to the number given.  For example, the following two commands are equivalent: sox input.wav -c 1 output.wav bass -b 24 sox input.wav      output.wav bass -b 24 channels 1 though the second form is more flexible as it allows the effects to be ordered arbitrarily.
-e ENCODING, --encoding ENCODING The audio encoding type.  Sometimes needed with file-types that support more than one encoding type. For example, with raw, WAV, or AU (but not, for example, with MP3 or FLAC).  The available encoding types are as follows:
signed-integer PCM data stored as signed (`two's complement') integers. Commonly used with a 16 or 24 -bit encoding size.  A value of 0 represents minimum signal power.
unsigned-integer PCM data stored as unsigned integers.  Commonly used with an 8-bit encoding size.  A value of 0 represents maximum signal power.
floating-point PCM data stored as IEEE 753 single precision (32-bit) or double precision (64-bit) floating-point (`real') numbers. A value of 0 represents minimum signal power.
a-law  International telephony standard for logarithmic encoding to 8 bits per sample.  It has a precision equivalent to roughly 13-bit PCM and is sometimes encoded with reversed bit-ordering (see the -X option).
u-law, mu-law North American telephony standard for logarithmic encoding to 8 bits per sample.  A.k.a. μ-law.  It has a precision equivalent to roughly 14-bit PCM and is sometimes encoded with reversed bit-ordering (see the -X option).
oki-adpcm OKI (a.k.a. VOX, Dialogic, or Intel) 4-bit ADPCM; it has a precision equivalent to roughly 12-bit PCM.  ADPCM is a form of audio compression that has a good compromise between audio quality and encoding/decoding speed.
ima-adpcm IMA (a.k.a. DVI) 4-bit ADPCM; it has a precision equivalent to roughly 13-bit PCM.
ms-adpcm Microsoft 4-bit ADPCM; it has a precision equivalent to roughly 14-bit PCM.
gsm-full-rate GSM is currently used for the vast majority of the world's digital wireless telephone calls.  It utilises several audio formats with different bit-rates and associated speech quality.  SoX has support for GSM's original 13kbps `Full Rate' audio format.  It is usually CPU-intensive to work with GSM audio.
Encoding names can be abbreviated where this would not be ambiguous; e.g. `unsigned-integer' can be given as `un', but not `u' (ambiguous with `u-law').
For an input file, the most common use for this option is to inform SoX of the encoding of a `raw' (`headerless') audio file (see the examples in -b and -c above).
For an output file, this option can be used (perhaps along with -b) to set the output encoding type  For example sox input.cdda -e float output1.wav
sox input.cdda -b 64 -e float output2.wav convert raw CD digital audio (16-bit, signed-integer) to floating- point `WAV' files (single & double precision respectively).
By default (i.e. if this option is not given), the output encoding type will (providing it is supported by the output file type) be set to the input encoding type.
--no-glob Specifies that filename `globbing' (wild-card matching) should not be performed by SoX on the following filename.  For example, if the current directory contains the two files `five-seconds.wav' and `five*.wav', then play --no-glob "five*.wav" can be used to play just the single file `five*.wav'.
-r, --rate RATE[k] Gives the sample rate in Hz (or kHz if appended with `k') of the file.
For an input file, the most common use for this option is to inform SoX of the sample rate of a `raw' (`headerless') audio file (see the examples in -b and -c above).  Occasionally it may be useful to use this option with a `headered' file, in order to override the (presumably incorrect) value in the header - note that this is only supported with certain file types.  For example, if audio was recorded with a sample-rate of say 48k from a source that played back a little, say 1.5%, too slowly, then sox -r 48720 input.wav output.wav effectively corrects the speed by changing only the file header (but see also the speed effect for the more usual solution to this problem).
For an output file, this option provides a shorthand for specifying that the rate effect should be invoked in order to change (if necessary) the sample rate of the audio signal to the given value.  For example, the following two commands are equivalent: sox input.wav -r 48k output.wav bass -b 24 sox input.wav        output.wav bass -b 24 rate 48k though the second form is more flexible as it allows rate options to be given, and allows the effects to be ordered arbitrarily.
-t, --type FILE-TYPE Gives the type of the audio file.  For both input and output files, this option is commonly used to inform SoX of the type a `headerless' audio file (e.g. raw, mp3) where the actual/desired type cannot be determined from a given filename extension.  For example: another-command | sox -t mp3 - output.wav
sox input.wav -t raw output.bin It can also be used to override the type implied by an input filename extension, but if overriding with a type that has a header, SoX will exit with an appropriate error message if such a header is not actually present.
See soxformat(7) for a list of supported file types.
-L, --endian little -B, --endian big -x, --endian swap These options specify whether the byte-order of the audio data is, respectively, `little endian', `big endian', or the opposite to that of the system on which SoX is being used.  Endianness applies only to data encoded as floating-point, or as signed or unsigned integers of 16 or more bits.  It is often necessary to specify one of these options for headerless files, and sometimes necessary for (otherwise) self-describing files.  A given endian-setting option may be ignored for an input file whose header contains a specific endianness identifier, or for an output file that is actually an audio device.
N.B.  Unlike other format characteristics, the endianness (byte, nibble, & bit ordering) of the input file is not automatically used for the output file; so, for example, when the following is run on a little-endian system: sox -B audio.s16 trimmed.s16 trim 2 trimmed.s16 will be created as little-endian; sox -B audio.s16 -B trimmed.s16 trim 2 must be used to preserve big-endianness in the output file.
The -V option can be used to check the selected orderings.
-N, --reverse-nibbles Specifies that the nibble ordering (i.e. the 2 halves of a byte) of the samples should be reversed; sometimes useful with ADPCM- based formats.
N.B.  See also N.B. in section on -x above.
-X, --reverse-bits Specifies that the bit ordering of the samples should be reversed; sometimes useful with a few (mostly headerless) formats.
Output File Format Options These options apply only to the output file and may precede only the output filename on the command line.
--add-comment TEXT Append a comment in the output file header (where applicable).
--comment TEXT Specify the comment text to store in the output file header (where applicable).
SoX will provide a default comment if this option (or --comment-file) is not given. To specify that no comment should be stored in the output file, use --comment "" .
--comment-file FILENAME Specify a file containing the comment text to store in the output file header (where applicable).
-C, --compression FACTOR The compression factor for variably compressing output file formats.  If this option is not given then a default compression factor will apply.  The compression factor is interpreted differently for different compressing file formats.  See the description of the file formats that use this option in soxformat(7) for more information.
In addition to converting, playing and recording audio files, SoX can be used to invoke a number of audio `effects'.  Multiple effects may be applied by specifying them one after another at the end of the SoX command line, forming an `effects chain'.  Note that applying multiple effects in real-time (i.e. when playing audio) is likely to require a high performance computer. Stopping other applications may alleviate performance issues should they occur.
Some of the SoX effects are primarily intended to be applied to a single instrument or `voice'.  To facilitate this, the remix effect and the global SoX option -M can be used to isolate then recombine tracks from a multi-track recording.
Multiple Effects Chains A single effects chain is made up of one or more effects.  Audio from the input runs through the chain until either the end of the input file is reached or an effect in the chain requests to terminate the chain.
SoX supports running multiple effects chains over the input audio.  In this case, when one chain indicates it is done processing audio, the audio data is then sent through the next effects chain.  This continues until either no more effects chains exist or the input has reached the end of the file.
An effects chain is terminated by placing a : (colon) after an effect. Any following effects are a part of a new effects chain.
It is important to place the effect that will stop the chain as the first effect in the chain.  This is because any samples that are buffered by effects to the left of the terminating effect will be discarded.  The amount of samples discarded is related to the --buffer option and it should be kept small, relative to the sample rate, if the terminating effect cannot be first.  Further information on stopping effects can be found in the Stopping SoX section.
There are a few pseudo-effects that aid using multiple effects chains. These include newfile which will start writing to a new output file before moving to the next effects chain and restart which will move back to the first effects chain.  Pseudo-effects must be specified as the first effect in a chain and as the only effect in a chain (they must have a : before and after they are specified).
The following is an example of multiple effects chains.  It will split the input file into multiple files of 30 seconds in length.  Each output filename will have unique number in its name as documented in the Output Files section. sox infile.wav output.wav trim 0 30 : newfile : restart
Common Notation And Parameters In the descriptions that follow, brackets [ ] are used to denote parameters that are optional, braces { } to denote those that are both optional and repeatable, and angle brackets < > to denote those that are repeatable but not optional.  Where applicable, default values for optional parameters are shown in parenthesis ( ).
The following parameters are used with, and have the same meaning for, several effects:
center[k] See frequency.
frequency[k] A frequency in Hz, or, if appended with `k', kHz.
gain   A power gain in dB.  Zero gives no gain; less than zero gives an attenuation.
position A position within the audio stream; the syntax is [=|+|-]timespec, where timespec is a time specification (see below).  The optional first character indicates whether the timespec is to be interpreted relative to the start (=) or end (-) of audio, or to the previous position if the effect accepts multiple position arguments (+).  The audio length must be known for end-relative locations to work; some effects do accept -0 for end-of-audio, though, even if the length is unknown.  Which of =, +, - is the default depends on the effect and is shown in its syntax as, e.g., position(+).
Examples: =2:00 (two minutes into the audio stream), -100s (one hundred samples before the end of audio), +0:12+10s (twelve seconds and ten samples after the previous position), -0.5+1s (one sample less than half a second before the end of audio).
width[h|k|o|q] Used to specify the band-width of a filter.  A number of different methods to specify the width are available (though not all for every effect).  One of the characters shown may be appended to select the desired method as follows:
Method    Notes h      Hz k     kHz o   Octaves q   Q-factor   See [2]
For each effect that uses this parameter, the default method (i.e. if no character is appended) is the one that it listed first in the first line of the effect's description.
Most effects that expect an audio position or duration in a parameter, i.e. a time specification, accept either of the following two forms:
[[hours:]minutes:]seconds[.frac][t] A specification of `1:30.5' corresponds to one minute, thirty and ½ seconds.  The t suffix is entirely optional (however, see the silence effect for an exception).  Note that the component values do not have to be normalized; e.g., `1:23:45', `83:45', `79:0285', `1:0:1425', `1::1425' and `5025' all are legal and equivalent to each other.
sampless Specifies the number of samples directly, as in `8000s'.  For large sample counts, e notation is supported: `1.7e6s' is the same as `1700000s'.
Time specifications can also be chained with + or - into a new time specification where the right part is added to or subtracted from the left, respectively: `3:00-200s' means two hundred samples less than three minutes.
To see if SoX has support for an optional effect, enter sox -h and look for its name under the list: `EFFECTS'.
Supported Effects Note: a categorised list of the effects can be found in the accompanying `README' file.
allpass frequency[k] width[h|k|o|q] Apply a two-pole all-pass filter with central frequency (in Hz) frequency, and filter-width width.  An all-pass filter changes the audio's frequency to phase relationship without changing its frequency to amplitude relationship.  The filter is described in detail in [1].
This effect supports the --plot global option.
band [-n] center[k] [width[h|k|o|q]] Apply a band-pass filter.  The frequency response drops logarithmically around the center frequency.  The width parameter 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 audio, i.e. voice, singing, or instrumental music.  The -n (for noise) option uses the alternate mode for un-pitched audio (e.g. percussion). Warning: -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 also sinc for a bandpass filter with steeper shoulders.
bandpass|bandreject [-c] frequency[k] width[h|k|o|q] Apply a two-pole Butterworth band-pass or band-reject filter with central frequency frequency, and (3dB-point) band-width width. The -c option applies only to bandpass and selects a constant skirt gain (peak gain = Q) instead of the default: constant 0dB peak gain.  The filters roll off at 6dB per octave (20dB per decade) and are described in detail in [1].
These effects support the --plot global option.
bandreject frequency[k] width[h|k|o|q] Apply a band-reject filter.  See the description of the bandpass effect for details.
bass|treble gain [frequency[k] [width[s|h|k|o|q]]] Boost or cut the bass (lower) or treble (upper) frequencies of the audio using a two-pole shelving filter with a response similar to that of a standard hi-fi's tone-controls.  This is also known as shelving equalisation (EQ).
gain gives the gain at 0 Hz (for bass), or whichever is the lower of ∼22 kHz and the Nyquist frequency (for treble).  Its useful range is about -20 (for a large cut) to +20 (for a large boost). Beware of Clipping when using a positive gain.
If desired, the filter can be fine-tuned using the following optional parameters:
frequency sets the filter's central frequency and so can be used to extend or reduce the frequency range to be boosted or cut.  The default value is 100 Hz (for bass) or 3 kHz (for treble).
width determines how steep is the filter's shelf transition.  In addition to the common width specification methods described above, `slope' (the default, or if appended with `s') may be used. The useful range of `slope' is about 0.3, for a gentle slope, to 1 (the maximum), for a steep slope; the default value is 0.5.
The filters are described in detail in [1].
See also equalizer for a peaking equalisation effect.
bend [-f frame-rate(25)] [-o over-sample(16)] { start- position(+),cents,end-position(+) } Changes pitch by specified amounts at specified times.  Each given triple: start-position,cents,end-position specifies one bend. cents is the number of cents (100 cents = 1 semitone) by which to bend the pitch. The other values specify the points in time at which to start and end bending the pitch, respectively.
The pitch-bending algorithm utilises the Discrete Fourier Transform (DFT) at a particular frame rate and over-sampling rate. The -f and -o parameters may be used to adjust these parameters and thus control the smoothness of the changes in pitch.
For example, an initial tone is generated, then bent three times, yielding four different notes in total: play -n synth 2.5 sin 667 gain 1 \ bend .35,180,.25  .15,740,.53  0,-520,.3 Here, the first bend runs from 0.35 to 0.6, and the second one from 0.75 to 1.28 seconds.  Note that the clipping that is produced in this example is deliberate; to remove it, use gain -5 in place of gain 1.
See also pitch.
biquad b0 b1 b2 a0 a1 a2 Apply a biquad IIR filter with the given coefficients. Where b* and a* are the numerator and denominator coefficients respectively.
See http://en.wikipedia.org/wiki/Digital_biquad_filter (where a0 = 1).
channels CHANNELS Invoke a simple algorithm to change the number of channels in the audio signal to the given number CHANNELS: mixing if decreasing the number of channels or duplicating if increasing the number of channels.
The channels effect is invoked automatically if SoX's -c option specifies a number of channels that is different to that of the input file(s).  Alternatively, if this effect is given explicitly, then SoX's -c option need not be given.  For example, the following two commands are equivalent: sox input.wav -c 1 output.wav bass -b 24 sox input.wav      output.wav bass -b 24 channels 1 though the second form is more flexible as it allows the effects to be ordered arbitrarily.
See also remix for an effect that allows channels to be mixed/selected arbitrarily.
chorus gain-in gain-out <delay decay speed depth -s|-t> Add a chorus effect to the audio.  This can make a single vocal sound like a chorus, but can also be applied to instrumentation.
Chorus resembles an echo effect with a short delay, but whereas with echo the delay is constant, with chorus, it is varied using sinusoidal or triangular modulation.  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 slightly off key.  See [3] for more discussion of the chorus effect.
Each four-tuple parameter 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 sinusoidal (-s) or triangular (-t).  Gain-out is the volume of the output.