sox [global-options] [format-options] infile1 [[format-options] infile2] ... [format-options] outfile [effect [effect-options]] ... play [global-options] [format-options] infile1 [[format-options] infile2] ... [format-options] [effect [effect-options]] ... rec [global-options] [format-options] outfile [effect [effect-options]] ...
All SoX functionality is available using just the sox command. To simplify playing and recording audio, if SoX is invoked as play, the output file is automatically set to be the default sound device, and if invoked as rec, the default sound device is used as an input source. Additionally, the soxi(1) command provides a convenient way to just query audio file header information.
The heart of SoX is a library called libSoX. Those interested in extending SoX or using it in other programs should refer to the libSoX manual page: libsox(3).
SoX is a command-line audio processing tool, particularly suited to making quick, simple edits and to batch processing. If you need an interactive, graphical audio editor, use audacity(1).
The overall SoX processing chain can be summarised as follows:
|Input(s) → Combiner → Effects → Output(s)|
Note however, that on the SoX command line, the positions of the Output(s) and the Effects are swapped w.r.t. the logical flow just shown. Note also that whilst options pertaining to files are placed before their respective file name, the opposite is true for effects. To show how this works in practice, here is a selection of examples of how SoX might be used. The simple
sox recital.au recital.wavtranslates an audio file in Sun AU format to a Microsoft WAV file, whilst
sox recital.au -b 16 recital.wav channels 1 rate 16k fade 3 normperforms the same format translation, but also applies four effects (down-mix to one channel, sample rate change, fade-in, nomalize), and stores the result at a bit-depth of 16.
sox -r 16k -e signed -b 8 -c 1 voice-memo.raw voice-memo.wavconverts `raw' (a.k.a. `headerless') audio to a self-describing file format,
sox slow.aiff fixed.aiff speed 1.027adjusts audio speed,
sox short.wav long.wav longer.wavconcatenates two audio files, and
sox -m music.mp3 voice.wav mixed.flacmixes together two audio files.
play "The Moonbeams/Greatest/*.ogg" bass +3plays a collection of audio files whilst applying a bass boosting effect,
play -n -c1 synth sin %-12 sin %-9 sin %-5 sin %-2 fade h 0.1 1 0.1plays a synthesised `A minor seventh' chord with a pipe-organ sound,
rec -c 2 radio.aiff trim 0 30:00records half an hour of stereo audio, and
play -q take1.aiff & rec -M take1.aiff take1-dub.aiff(with POSIX shell and where supported by hardware) records a new track in a multi-track recording. Finally,
rec -r 44100 -b 16 -s -p silence 1 0.50 0.1% 1 10:00 0.1% | \ sox -p song.ogg silence 1 0.50 0.1% 1 2.0 0.1% : \ newfile : restartrecords a stream of audio such as LP/cassette and splits in to multiple audio files at points with 2 seconds of silence. Also, it does not start recording until it detects audio is playing and stops after it sees 10 minutes of silence.
N.B. The above is just an overview of SoX's capabilities; detailed explanations of how to use all SoX parameters, file formats, and effects can be found below in this manual, in soxformat(7), and in soxi(1).
The following four characteristics are used to describe the format of audio data such that it can be processed with SoX:
The term `bit-rate' is a measure of the amount of storage occupied by an encoded audio signal over a unit of time. It can depend on all of the above and is typically denoted as a number of kilo-bits per second (kbps). An A-law telephony signal has a bit-rate of 64 kbps. MP3-encoded stereo music typically has a bit-rate of 128-196 kbps. FLAC-encoded stereo music typically has a bit-rate of 550-760 kbps.
Most self-describing formats also allow textual `comments' to be embedded in the file that can be used to describe the audio in some way, e.g. for music, the title, the author, etc.
One important use of audio file comments is to convey `Replay Gain' information. SoX supports applying Replay Gain information, but not generating it. Note that by default, SoX copies input file comments to output files that support comments, so output files may contain Replay Gain information if some was present in the input file. In this case, if anything other than a simple format conversion was performed then the output file Replay Gain information is likely to be incorrect and so should be recalculated using a tool that supports this (not SoX).
The soxi(1) command can be used to display information from audio file headers.
To determine the format of an input file, SoX will use, in order of precedence and as given or available:
To set the output file format, SoX will use, in order of precedence and as given or available:
For all files, SoX will exit with an error if the file type cannot be determined. Command-line format options may need to be added or changed to resolve the problem.
rec new-file.wavThese two commands are functionally equivalent to
sox existing-file.wav -dand
sox -d new-file.wavOf course, further options and effects (as described below) can be added to the commands in either form.
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 ossor
set AUDIODEV=hw:soundwave,1,2 play ... sox ... -t alsaNote 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 -mor 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 -nThe 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.
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 10SoX 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.
Specifically, by default, SoX automatically adds TPDF dither when the output bit-depth is less than 24 and any of the following are true:
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.
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 +6guarantees 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.
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/srn 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.
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
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.
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.
sox -M "|genw --imd -" "|genw --thd -" out.wavFor headerless (raw) audio, -t (and perhaps other format options) will need to be given, preceding the input command.
play --rate 6k *.voxwill be expanded by the `shell' (in most environments) to
play --rate 6k file1.vox file2.vox file3.voxwhich 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.
play "|sox -n -p synth 2" "|sox -n -p synth 2 tremolo 10" statplays two `files' in succession, each with different effects.
-p is in fact an alias for `-t sox -'.
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).
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:
export SOX_OPTS="-V --no-clobber"Unix csh:
setenv SOX_OPTS "-V --no-clobber"MS-DOS/MS-Windows:
set SOX_OPTS=-V --no-clobberMS-Windows GUI: via Control Panel : System : Advanced : Environment Variables
Mac OS X GUI: Refer to Apple's Technical Q&A QA1067 document.
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.
See Input File Combining above for a description of the different combining methods.
sox -G infile -b 16 outfile rate 44100 dither -sis shorthand for
sox infile -b 16 outfile gain -h rate 44100 gain -rh dither -sSee also -V, --norm, and the gain effect.
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.
sox --norm infile -b 16 outfile rate 44100 dither -sis shorthand for
sox infile -b 16 outfile gain -h rate 44100 gain -nh dither -sOptionally, 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.
sox --plot octave input-file -n highpass 1320 > highpass.plt octave highpass.plt
|dB FSD||Display||dB FSD||Display|
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.
SoX displays messages on the console (stderr) according to the following verbosity levels:
See also the norm, vol, and gain effects, and see Input File Balancing above.
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.wavconverts 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.wavconverts raw CD digital audio (16-bit, signed-integer) to a 24-bit (signed-integer) `WAV' file.
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.wavconverts a particular `raw' file to a self-describing `WAV' file.
play -c 1 music.wavinterprets 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 1though the second form is more flexible as it allows the effects to be ordered arbitrarily.
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.wavconvert 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.
play --no-glob "five*.wav"can be used to play just the single file `five*.wav'.
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.waveffectively 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 48kthough the second form is more flexible as it allows rate options to be given, and allows the effects to be ordered arbitrarily.
another-command | sox -t mp3 - output.wav sox input.wav -t raw output.binIt 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
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 2trimmed.s16 will be created as little-endian;
sox -B audio.s16 -B trimmed.s16 trim 2must be used to preserve big-endianness in the output file.
The -V option can be used to check the selected orderings.
N.B. See also N.B. in section on -x above.
N.B. See also N.B. in section on -x above.
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 "" .
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.
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
The following parameters are used with, and have the same meaning for, several effects:
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.
To see if SoX has support for an optional effect, enter sox -h and look for its name under the list: `EFFECTS'.
This effect supports the --plot global option.
This effect supports the --plot global option.
See also sinc for a bandpass filter with steeper shoulders.
These effects support the --plot global option.
See also sinc for a bandpass filter with steeper shoulders.
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 .
These effects support the --plot global option.
See also equalizer for a peaking equalisation effect.
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,.3Note 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.
See http://en.wikipedia.org/wiki/Digital_biquad_filter (where a0 = 1).
This effect supports the --plot global option.
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 1though 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 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  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.
A typical delay is around 40ms to 60ms; the modulation speed is best near 0.25Hz and the modulation depth around 2ms. For example, a single delay:
play guitar1.wav chorus 0.7 0.9 55 0.4 0.25 2 -tTwo delays of the original samples:
play guitar1.wav chorus 0.6 0.9 50 0.4 0.25 2 -t \ 60 0.32 0.4 1.3 -sA fuller sounding chorus (with three additional delays):
play guitar1.wav chorus 0.5 0.9 50 0.4 0.25 2 -t \ 60 0.32 0.4 2.3 -t 40 0.3 0.3 1.3 -s
Compand (compress or expand) the dynamic range of the audio.
The attack and decay parameters (in seconds) determine the time over which the instantaneous level of the input signal is averaged to determine its volume; attacks refer to increases in volume and decays refer to decreases. For most situations, the attack time (response to the music getting louder) should be shorter than the decay time because the human ear is more sensitive to sudden loud music than sudden soft music. Where more than one pair of attack/decay parameters are specified, each input channel is companded separately and the number of pairs must agree with the number of input channels. Typical values are 0.3,0.8 seconds.
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. If omitted, the value of out-dB1 defaults to the same value as in-dB1; levels below in-dB1 are not companded (but may have gain applied to them). The point 0,0 is assumed but may be overridden (by 0,out-dBn). If the list is preceded by a soft-knee-dB value, then the points at where adjacent line segments on the transfer function meet will be rounded by the amount given. Typical values for the transfer function are 6:-70,-60,-20.
The third (optional) parameter is an additional gain in dB to be applied at all points on the transfer function and allows easy adjustment of the overall gain.
The fourth (optional) parameter is an initial level to be assumed for each channel when companding starts. This permits the user to supply a nominal level initially, so that, for example, a very large gain is not applied to initial signal levels before the companding action has begun to operate: it is quite probable that in such an event, the output would be severely clipped while the compander gain properly adjusts itself. A typical value (for audio which is initially quiet) is -90 dB.
The fifth (optional) parameter is a delay in seconds. The input signal is analysed immediately to control the compander, 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 operate in a `predictive' rather than a reactive mode. A typical value is 0.2 seconds.
The following example might be used to make a piece of music with both quiet and loud passages suitable for listening to in a noisy environment such as a moving vehicle:
sox asz.wav asz-car.wav compand 0.3,1 6:-70,-60,-20 -5 -90 0.2The transfer function (`6:-70,...') says that very soft sounds (below -70dB) will remain unchanged. This will stop the compander from boosting the volume on `silent' passages such as between movements. However, sounds in the range -60dB to 0dB (maximum volume) will be boosted so that the 60dB dynamic range of the original music will be compressed 3-to-1 into a 20dB range, which is wide enough to enjoy the music but narrow enough to get around the road noise. The `6:' selects 6dB soft-knee companding. The -5 (dB) output gain is needed to avoid clipping (the number is inexact, and was derived by experimentation). The -90 (dB) for the initial volume will work fine for a clip that starts with near silence, and the delay of 0.2 (seconds) has the effect of causing the compander to react a bit more quickly to sudden volume changes.
In the next example, compand is being used as a noise-gate for when the noise is at a lower level than the signal:
play infile compand .1,.2 -inf,-50.1,-inf,-50,-50 0 -90 .1Here is another noise-gate, this time for when the noise is at a higher level than the signal (making it, in some ways, similar to squelch):
play infile compand .1,.1 -45.1,-45,-inf,0,-inf 45 -90 .1This effect supports the --plot global option (for the transfer function).
See also mcompand for a multiple-band companding effect.
See also the compand and mcompand effects.
The given dcshift value is a floating point number in the range of ±2 that indicates the amount to shift the audio (which is in the range of ±1).
An optional limitergain can be specified as well. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is used only on peaks to prevent clipping.
An alternative approach to removing a DC offset (albeit with a short delay) is to use the highpass filter effect at a frequency of say 10Hz, as illustrated in the following example:
sox -n dc.wav synth 5 sin %0 50 sox dc.wav fixed.wav highpass 10
Pre-emphasis was applied in the mastering of some CDs issued in the early 1980s. These included many classical music albums, as well as now sought-after issues of albums by The Beatles, Pink Floyd and others. Pre-emphasis should be removed at playback time by a de-emphasis filter in the playback device. However, not all modern CD players have this filter, and very few PC CD drives have it; playing pre-emphasised audio without the correct de-emphasis filter results in audio that sounds harsh and is far from what its creators intended.
With the deemph effect, it is possible to apply the necessary de-emphasis to audio that has been extracted from a pre-emphasised CD, and then either burn the de-emphasised audio to a new CD (which will then play correctly on any CD player), or simply play the correctly de-emphasised audio files on the PC. For example:
sox track1.wav track1-deemph.wav deemphand then burn track1-deemph.wav to CD, or
play track1-deemph.wavor simply
play track1.wav deemphThe de-emphasis filter is implemented as a biquad; its maximum deviation from the ideal response is only 0.06dB (up to 20kHz).
This effect supports the --plot global option.
See also the bass and treble shelving equalisation effects.
play -n synth -j 3 sin %3 sin %-2 sin %-5 sin %-9 \ sin %-14 sin %-21 fade h .01 2 1.5 delay \ 1.3 1 .76 .54 .27 remix - fade h 0 2.7 2.5 norm -1and this plays a guitar chord:
play -n synth pl G2 pl B2 pl D3 pl G3 pl D4 pl G4 \ delay 0 .05 .1 .15 .2 .25 remix - fade 0 4 .1 norm -1
The -S option selects a slightly `sloped' TPDF, biased towards higher frequencies. It can be used at any sampling rate but below ~~22k, plain TPDF is probably better, and above ~~ 37k, noise-shaped is probably better.
The -a option enables a mode where dithering (and noise-shaping if applicable) are automatically enabled only when needed. The most likely use for this is when applying fade in or out to an already dithered file, so that the redithering applies only to the faded portions. However, auto dithering is not fool-proof, so the fades should be carefully checked for any noise modulation; if this occurs, then either re-dither the whole file, or use trim, fade, and concatencate.
The -p option allows overriding the target precision.
If the SoX global option -R option is not given, then the pseudo-random number generator used to generate the white noise will be `reseeded', i.e. the generated noise will be different between invocations.
This effect should not be followed by any other effect that affects the audio.
See also the `Dithering' section above.
No decimation filter is applied. If the input is not a properly bandlimited baseband signal, aliasing will occur. This may be desirable, e.g., for frequency translation.
For a general resampling effect with anti-aliasing, see rate. See also upsample.
Each given delay decay pair gives the delay in milliseconds and the decay (relative to gain-in) of that echo. Gain-out is the volume of the output. For example: This will make it sound as if there are twice as many instruments as are actually playing:
play lead.aiff echo 0.8 0.88 60 0.4If the delay is very short, then it sound like a (metallic) robot playing music:
play lead.aiff echo 0.8 0.88 6 0.4A longer delay will sound like an open air concert in the mountains:
play lead.aiff echo 0.8 0.9 1000 0.3One mountain more, and:
play lead.aiff echo 0.8 0.9 1000 0.3 1800 0.25
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; a single echos has the same effect as a single echo.
The sample will be bounced twice in symmetric echos:
play lead.aiff echos 0.8 0.7 700 0.25 700 0.3The sample will be bounced twice in asymmetric echos:
play lead.aiff echos 0.8 0.7 700 0.25 900 0.3The sample will sound as if played in a garage:
play lead.aiff echos 0.8 0.7 40 0.25 63 0.3
frequency gives the filter's central frequency in Hz, width, the band-width, and gain the required gain or attenuation in dB. Beware of Clipping when using a positive gain.
In order to produce complex equalisation curves, this effect can be given several times, each with a different central frequency.
The filter is described in detail in .
This effect supports the --plot global option.
See also bass and treble for shelving equalisation effects.
An optional type can be specified to select the shape of the fade curve: q for quarter of a sine wave, h for half a sine wave, t for linear (`triangular') slope, l for logarithmic, and p for inverted parabola. The default is logarithmic.
A fade-in starts from the first sample and ramps the signal level from 0 to full volume over fade-in-length seconds. Specify 0 seconds if no fade-in is wanted.
For fade-outs, the audio will be truncated at stop-time and the signal level 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 stop-time is not specified. If the file length can be determined from the input file header and length-changing effects are not in effect, then 0 may be specified for stop-time to indicate the usual case of a fade-out that ends at the end of the input audio stream.
All times can be specified in either periods of time or sample counts. To specify time periods use the format hh:mm:ss.frac format. To specify using sample counts, specify the number of samples and append the letter `s' to the sample count (for example `8000s').
See also the splice effect.
sox infile outfile fir 0.0195 -0.082 0.234 0.891 -0.145 0.043
sox infile outfile fir coefs.txtwith coefs.txt containing
# HP filter # freq=10000 1.2311233052619888e-01 -4.4777096106211783e-01 5.1031563346705155e-01 -6.6502926320995331e-02 ...
This effect supports the --plot global option.
All parameters are optional (right to left).
|delay||0 - 30||0||Base delay in milliseconds.|
|depth||0 - 10||2||Added swept delay in milliseconds.|
|regen||-95 - 95||0||
Percentage regeneration (delayed signal feedback).
|width||0 - 100||71||
Percentage of delayed signal mixed with original.
|speed||0.1 - 10||0.5||Sweeps per second (Hz).|
|shape||sin||Swept wave shape: sine|triangle.|
|phase||0 - 100||25||
Swept wave percentage phase-shift for multi-channel (e.g. stereo) flange;
0 = 100 = same phase on each channel.
Digital delay-line interpolation: linear|quadratic.
Without other options, gain-dB is used to adjust the signal power level by the given number of dB: positive amplifies (beware of Clipping), negative attenuates. With other options, the gain-dB amplification or attenuation is (logically) applied after the processing due to those options.
Given the -e option, the levels of the audio channels of a multi-channel file are `equalised', i.e. gain is applied to all channels other than that with the highest peak level, such that all channels attain the same peak level (but, without also giving -n, the audio is not `normalised').
The -B (balance) option is similar to -e, but with -B, the RMS level is used instead of the peak level. -B might be used to correct stereo imbalance caused by an imperfect record turntable cartridge. Note that unlike -e, -B might cause some clipping.
-b is similar to -B but has clipping protection, i.e. if necessary to prevent clipping whilst balancing, attenuation is applied to all channels. Note, however, that in conjunction with -n, -B and -b are synonymous.
The -r option is used in conjunction with a prior invocation of gain with the -h option - see below for details.
The -n option normalises the audio to 0dB FSD; it is often used in conjunction with a negative gain-dB to the effect that the audio is normalised to a given level below 0dB. For example,
sox infile outfile gain -nnormalises to 0dB, and
sox infile outfile gain -n -3normalises to -3dB.
The -l option invokes a simple limiter, e.g.
sox infile outfile gain -l 6will apply 6dB of gain but never clip. Note that limiting more than a few dBs more than occasionally (in a piece of audio) is not recommended as it can cause audible distortion. See the compand effect for a more capable limiter.
The -h option is used to apply gain to provide head-room for subsequent processing. For example, with
sox infile outfile gain -h bass +66dB of attenuation will be applied prior to the bass boosting effect thus ensuring that it will not clip. Of course, with bass, it is obvious how much headroom will be needed, but with other effects (e.g. rate, dither) it is not always as clear. Another advantage of using gain -h rather than an explicit attenuation, is that if the headroom is not used by subsequent effects, it can be reclaimed with gain -r, for example:
sox infile outfile gain -h bass +6 rate 44100 gain -rThe above effects chain guarantees never to clip nor amplify; it attenuates if necessary to prevent clipping, but by only as much as is needed to do so.
Output formatting (dithering and bit-depth reduction) also requires headroom (which cannot be `reclaimed'), e.g.
sox infile outfile gain -h bass +6 rate 44100 gain -rh ditherHere, the second gain invocation, reclaims as much of the headroom as it can from the preceding effects, but retains as much headroom as is needed for subsequent processing. The SoX global option -G can be given to automatically invoke gain -h and gain -r.
See also the norm and vol effects.
These effects support the --plot global option.
See also sinc for filters with a steeper roll-off.
This is used in many matrix coding schemes and for analytic signal generation. The process is often written as a multiplication by i (or j), the imaginary unit.
An odd-tap Hilbert transform filter has a bandpass characteristic, attenuating the lowest and highest frequencies. Its bandwidth can be controlled by the number of filter taps, which can be specified with -n. By default, the number of taps is chosen for a cutoff frequency of about 75 Hz.
This effect supports the --plot global option.
See also the gain effect.
The multi-band compander is similar to the single-band compander but the audio is first divided into bands using Linkwitz-Riley cross-over filters and a separately specifiable compander run on each band. See the compand effect for the definition of its parameters. Compand parameters are specified between double quotes and the crossover frequency for that band is given by crossover-freq; these can be repeated to create multiple bands.
For example, the following (one long) command shows how multi-band companding is typically used in FM radio:
play track1.wav gain -3 sinc 8000- 29 100 mcompand \ "0.005,0.1 -47,-40,-34,-34,-17,-33" 100 \ "0.003,0.05 -47,-40,-34,-34,-17,-33" 400 \ "0.000625,0.0125 -47,-40,-34,-34,-15,-33" 1600 \ "0.0001,0.025 -47,-40,-34,-34,-31,-31,-0,-30" 6400 \ "0,0.025 -38,-31,-28,-28,-0,-25" \ gain 15 highpass 22 highpass 22 sinc -n 255 -b 16 -17500 \ gain 9 lowpass -1 17801The audio file is played with a simulated FM radio sound (or broadcast signal condition if the lowpass filter at the end is skipped). Note that the pipeline is set up with US-style 75us pre-emphasis.
See also compand for a single-band companding effect.
sox speech.wav -n trim 0 1.5 noiseprof speech.noise-profileTo actually remove the noise, run SoX again, this time with the noisered effect; noisered will reduce noise according to a noise profile (which was generated by noiseprof), from profile-file, or from stdin if no profile-file or if `-' is given. E.g.
sox speech.wav cleaned.wav noisered speech.noise-profile 0.3How much noise should be removed is specified by amount - a number between 0 and 1 with a default of 0.5. Higher numbers will remove more noise but present a greater likelihood of removing wanted components of the audio signal. Before replacing an original recording with a noise-reduced version, experiment with different amount values to find the optimal one for your audio; use headphones to check that you are happy with the results, paying particular attention to quieter sections of the audio.
On most systems, the two stages - profiling and reduction - can be combined using a pipe, e.g.
sox noisy.wav -n trim 0 1 noiseprof | play noisy.wav noisered
See also delay for an effect that can add silence at the beginning of the audio on a channel-by-channel basis.
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 sinusoidal (-s) - preferable for multiple instruments, or triangular (-t) - gives single instruments a sharper phasing effect. The decay should be less than 0.5 to avoid feedback, and usually no less than 0.1. Gain-out is the volume of the output.
play snare.flac phaser 0.8 0.74 3 0.4 0.5 -tGentler:
play snare.flac phaser 0.9 0.85 4 0.23 1.3 -sA popular sound:
play snare.flac phaser 0.89 0.85 1 0.24 2 -tMore severe:
play snare.flac phaser 0.6 0.66 3 0.6 2 -t
shift gives the pitch shift as positive or negative `cents' (i.e. 100ths of a semitone). See the tempo effect for a description of the other parameters.
See also the bend, speed, and tempo effects.
|-q||quick||n/a||~=30 @ Fs/4||
playback on ancient hardware
playback on old hardware
16-bit mastering (use with dither)
|-v||very high||95%||175||24-bit mastering|
where Band-width is the percentage of the audio frequency band that is preserved and Rej dB is the level of noise rejection. Increasing levels of resampling quality come at the expense of increasing amounts of time to process the audio. If no quality option is given, the quality level used is `high' (but see `Playing & Recording Audio' above regarding playback).
The `quick' algorithm uses cubic interpolation; all others use band-limited interpolation. By default, all algorithms have a `linear' phase response; for `medium', `high' and `very high', the phase response is configurable (see below).
The rate effect is invoked automatically if SoX's -r option specifies a rate that is different to that of the input file(s). Alternatively, if this effect is given explicitly, then SoX's -r option need not be given. 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 48kthough the second command is more flexible as it allows rate options to be given, and allows the effects to be ordered arbitrarily.
Warning: technically detailed discussion follows.
The simple quality selection described above provides settings that satisfy the needs of the vast majority of resampling tasks. Occasionally, however, it may be desirable to fine-tune the resampler's filter response; this can be achieved using override options, as detailed in the following table:
|-M/-I/-L||Phase response = minimum/intermediate/linear|
|-s||Steep filter (band-width = 99%)|
|-a||Allow aliasing/imaging above the pass-band|
|-b 74-99.7||Any band-width %|
Any phase response (0 = minimum, 25 = intermediate, 50 = linear, 100 = maximum)
N.B. Override options cannot be used with the `quick' or `low' quality algorithms.
All resamplers use filters that can sometimes create `echo' (a.k.a. `ringing') artefacts with transient signals such as those that occur with `finger snaps' or other highly percussive sounds. Such artefacts are much more noticeable to the human ear if they occur before the transient (`pre-echo') than if they occur after it (`post-echo'). Note that frequency of any such artefacts is related to the smaller of the original and new sampling rates but that if this is at least 44.1kHz, then the artefacts will lie outside the range of human hearing.
A phase response setting may be used to control the distribution of any transient echo between `pre' and `post': with minimum phase, there is no pre-echo but the longest post-echo; with linear phase, pre and post echo are in equal amounts (in signal terms, but not audibility terms); the intermediate phase setting attempts to find the best compromise by selecting a small length (and level) of pre-echo and a medium lengthed post-echo.
Minimum, intermediate, or linear phase response is selected using the -M, -I, or -L option; a custom phase response can be created with the -p option. Note that phase responses between `linear' and `maximum' (greater than 50) are rarely useful.
A resampler's band-width setting determines how much of the frequency content of the original signal (w.r.t. the original sample rate when up-sampling, or the new sample rate when down-sampling) is preserved during conversion. The term `pass-band' is used to refer to all frequencies up to the band-width point (e.g. for 44.1kHz sampling rate, and a resampling band-width of 95%, the pass-band represents frequencies from 0Hz (D.C.) to circa 21kHz). Increasing the resampler's band-width results in a slower conversion and can increase transient echo artefacts (and vice versa).
The -s `steep filter' option changes resampling band-width from the default 95% (based on the 3dB point), to 99%. The -b option allows the band-width to be set to any value in the range 74-99.7 %, but note that band-width values greater than 99% are not recommended for normal use as they can cause excessive transient echo.
If the -a option is given, then aliasing/imaging above the pass-band is allowed. For example, with 44.1kHz sampling rate, and a resampling band-width of 95%, this means that frequency content above 21kHz can be distorted; however, since this is above the pass-band (i.e. above the highest frequency of interest/audibility), this may not be a problem. The benefits of allowing aliasing/imaging are reduced processing time, and reduced (by almost half) transient echo artefacts. Note that if this option is given, then the minimum band-width allowable with -b increases to 85%.
sox input.wav -b 16 output.wav rate -s -a 44100 dither -sdefault (high) quality resampling; overrides: steep filter, allow aliasing; to 44.1kHz sample rate; noise-shaped dither to 16-bit WAV file.
sox input.wav -b 24 output.aiff rate -v -I -b 90 48kvery high quality resampling; overrides: intermediate phase, band-width 90%; to 48k sample rate; store output to 24-bit AIFF file.
The pitch and speed effects use the rate effect at their core.
Select and mix input audio channels into output audio channels. Each output channel is specified, in turn, by a given out-spec: a list of contributing input channels and volume specifications.
Note that this effect operates on the audio channels within the SoX effects processing chain; it should not be confused with the -m global option (where multiple files are mix-combined before entering the effects chain).
An out-spec contains comma-separated input channel-numbers and hyphen-delimited channel-number ranges; alternatively, 0 may be given to create a silent output channel. For example,
sox input.wav output.wav remix 6 7 8 0creates an output file with four channels, where channels 1, 2, and 3 are copies of channels 6, 7, and 8 in the input file, and channel 4 is silent. Whereas
sox input.wav output.wav remix 1-3,7 3creates a (somewhat bizarre) stereo output file where the left channel is a mix-down of input channels 1, 2, 3, and 7, and the right channel is a copy of input channel 3.
Where a range of channels is specified, the channel numbers to the left and right of the hyphen are optional and default to 1 and to the number of input channels respectively. Thus
sox input.wav output.wav remix -performs a mix-down of all input channels to mono.
By default, where an output channel is mixed from multiple (n) input channels, each input channel will be scaled by a factor of ¹/n. Custom mixing volumes can be set by following a given input channel or range of input channels with a vol-spec (volume specification). This is one of the letters p, i, or v, followed by a volume number, the meaning of which depends on the given letter and is defined as follows:
|p||power adjust in dB||0 = no change|
|i||power adjust in dB||
As `p', but invert the audio
1 = no change, 0.5 ~= 6dB attenuation, 2 ~= 6dB gain, -1 = invert
If an out-spec includes at least one vol-spec then, by default, ¹/n scaling is not applied to any other channels in the same out-spec (though may be in other out-specs). The -a (automatic) option however, can be given to retain the automatic scaling in this case. For example,
sox input.wav output.wav remix 1,2 3,4v0.8results in channel level multipliers of 0.5,0.5 1,0.8, whereas
sox input.wav output.wav remix -a 1,2 3,4v0.8results in channel level multipliers of 0.5,0.5 0.5,0.8.
The -m (manual) option disables all automatic volume adjustments, so
sox input.wav output.wav remix -m 1,2 3,4v0.8results in channel level multipliers of 1,1 1,0.8.
The volume number is optional and omitting it corresponds to no volume change; however, the only case in which this is useful is in conjunction with i. For example, if input.wav is stereo, then
sox input.wav output.wav remix 1,2iis a mono equivalent of the oops effect.
If the -p option is given, then any automatic ¹/n scaling is replaced by ¹/srn (`power') scaling; this gives a louder mix but one that might occasionally clip.
One use of the remix effect is to split an audio file into a set of files, each containing one of the constituent channels (in order to perform subsequent processing on individual audio channels). Where more than a few channels are involved, a script such as the following (Bourne shell script) is useful:
#!/bin/sh chans=`soxi -c "$1"` while [ $chans -ge 1 ]; do chans0=`printf %02i $chans` # 2 digits hence up to 99 chans out=`echo "$1"|sed "s/\(.*\)\.\(.*\)/\1-$chans0.\2/"` sox "$1" "$out" remix $chans chans=`expr $chans - 1` doneIf a file input.wav containing six audio channels were given, the script would produce six output files: input-01.wav, input-02.wav, ..., input-06.wav.
See also the swap effect.
Add reverberation to the audio using the `freeverb' algorithm. A reverberation effect is sometimes desirable for concert halls that are too small or contain so many people that the hall's natural reverberance is diminished. Applying a small amount of stereo reverb to a (dry) mono signal will usually make it sound more natural. See  for a detailed description of reverberation.
Note that this effect increases both the volume and the length of the audio, so to prevent clipping in these domains, a typical invocation might be:
play dry.wav gain -3 pad 0 3 reverbThe -w option can be given to select only the `wet' signal, thus allowing it to be processed further, independently of the `dry' signal. E.g.
play -m voice.wav "|sox voice.wav -p reverse reverb -w reverse"for a reverse reverb effect.
This effect supports the --plot global option.
Removes silence from the beginning, middle, or end of the audio. `Silence' is determined by a specified threshold.
The above-periods value is used to indicate if audio should be trimmed at the beginning of the audio. A value of zero indicates no silence should be trimmed from the beginning. When specifying 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 audio 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 duration and threshold. Duration indications the amount of time that non-silence must be detected before it stops trimming audio. 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 the value to account for background noise.
When optionally trimming silence from the end of the audio, you specify a below-periods count. In this case, below-period means to remove all audio 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.
For below-periods, duration specifies a period of silence that must exist before audio is not copied any more. By specifying 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.
Unfortunately, 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 audio.
The option -l indicates that below-periods duration length of audio should be left intact at the beginning of each period of silence. For example, if you want to remove long pauses between words but do not want to remove the pauses completely.
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 with d to indicate the value is in decibels, or % to indicate a percentage of maximum value of the sample value (0% specifies pure digital silence).
The following example shows how this effect can be used to start a recording that does not contain the delay at the start which usually occurs between `pressing the record button' and the start of the performance:
rec parameters filename other-effects silence 1 5 2%
sinc 3k sinc -4k sinc 3k-4k sinc 4k-3kcreate a high-pass, low-pass, band-pass, and band-reject filter respectively.
The default stop-band attenuation of 120dB can be overridden with -a; alternatively, the kaiser-window `beta' parameter can be given directly with -b.
The default transition band-width of 5% of the total band can be overridden with -t (and tbw in Hertz); alternatively, the number of filter taps can be given directly with -n.
If both freqHP and freqLP are given, then a -t or -n option given to the left of the frequencies applies to both frequencies; one of these options given to the right of the frequencies applies only to freqLP.
The -p, -M, -I, and -L options control the filter's phase response; see the rate effect for details.
This effect supports the --plot global option.
The spectrogram is rendered in a Portable Network Graphic (PNG) file, and shows time in the X-axis, frequency in the Y-axis, and audio signal magnitude in the Z-axis. Z-axis values are represented by the colour (or optionally the intensity) of the pixels in the X-Y plane. If the audio signal contains multiple channels then these are shown from top to bottom starting from channel 1 (which is the left channel for stereo audio).
For example, if `my.wav' is a stereo file, then with
sox my.wav -n spectrograma spectrogram of the entire file will be created in the file `spectrogram.png'. More often though, analysis of a smaller portion of the audio is required; e.g. with
sox my.wav -n remix 2 trim 20 30 spectrogramthe spectrogram shows information only from the second (right) channel, and of thirty seconds of audio starting from twenty seconds in. To analyse a small portion of the frequency domain, the rate effect may be used, e.g.
sox my.wav -n rate 6k spectrogramallows detailed analysis of frequencies up to 3kHz (half the sampling rate) i.e. where the human auditory system is most sensitive. With
sox my.wav -n trim 0 10 spectrogram -x 600 -y 200 -z 100the given options control the size of the spectrogram's X, Y & Z axes (in this case, the spectrogram area of the produced image will be 600 by 200 pixels in size and the Z-axis range will be 100 dB). Note that the produced image includes axes legends etc. and so will be a little larger than the specified spectrogram size. In this example:
sox -n -n synth 6 tri 10k:14k spectrogram -z 100 -w kaiseran analysis `window' with high dynamic range is selected to best display the spectrogram of a swept triangular wave. For a smilar example, append the following to the `chime' command in the description of the delay effect (above):
rate 2k spectrogram -X 200 -Z -10 -w kaiserOptions are also avaliable to control the appearance (colour-set, brightness, contrast, etc.) and filename of the spectrogram; e.g. with
sox my.wav -n spectrogram -m -l -o print.pnga spectrogram is created suitable for printing on a `black and white' printer.
sox input.mp3 output.wav -n spectrogram -d 1:00 statscreates a spectrogram showing the first minute of the audio, whilst
the stats effect is applied to the entire audio signal.
See also -X for an alternative way of setting the X-axis resolution.
sox input.aiff output.wav spectrogram -S 1:00creates a spectrogram showing all but the first minute of the audio (the output file however, receives the entire audio stream).
Technically, the speed effect only changes the sample rate information, leaving the samples themselves untouched. The rate effect is invoked automatically to resample to the output sample rate, using its default quality/speed. For higher quality or higher speed resampling, in addition to the speed effect, specify the rate effect with the desired quality option.
See also the bend, pitch, and tempo effects.
One of the options -h, -t, or -q may be given to select the fade envelope as half-cosine wave (the default), triangular (a.k.a. linear), or quarter-cosine wave respectively.
To perform a splice, first use the trim effect to select the audio sections to be joined together. As when performing a tape splice, the end of the section to be spliced onto should be trimmed with a small excess (default 0.005 seconds) of audio after the ideal joining point. The beginning of the audio section to splice on should be trimmed with the same excess (before the ideal joining point), plus an additional leeway (default 0.005 seconds). SoX should then be invoked with the two audio sections as input files and the splice effect given with the position at which to perform the splice - this is length of the first audio section (including the excess).
The following diagram uses the tape analogy to illustrate the splice operation. The effect simulates the diagonal cuts and joins the two pieces:
length1 excess -----------><---> _________ : : _________________ \ : : :\ ` \ : : : \ ` \: : : \ ` * : : * - - * \ : : :\ ` \ : : : \ ` _______________\: : : \_____`____ : : : : <---> <-----> excess leewaywhere * indicates the joining points.
For example, a long song begins with two verses which start (as determined e.g. by using the play command with the trim (start) effect) at times 0:30.125 and 1:03.432. The following commands cut out the first verse:
sox too-long.wav part1.wav trim 0 30.130(5 ms excess, after the first verse starts)
sox too-long.wav part2.wav trim 1:03.422(5 ms excess plus 5 ms leeway, before the second verse starts)
sox part1.wav part2.wav just-right.wav splice 30.130For another example, the SoX command
play "|sox -n -p synth 1 sin %1" "|sox -n -p synth 1 sin %3"generates and plays two notes, but there is a nasty click at the transition; the click can be removed by splicing instead of concatenating the audio, i.e. by appending splice 1 to the command. (Clicks at the beginning and end of the audio can be removed by preceding the splice effect with fade q .01 2 .01).
Provided your arithmetic is good enough, multiple splices can be performed with a single splice invocation. For example:
#!/bin/sh # Audio Copy and Paste Over # acpo infile copy-start copy-stop paste-over-start outfile # All times measured in samples. rate=`soxi -r "$1"` e=`expr $rate '*' 5 / 1000` # Using default excess l=$e # and leeway. sox "$1" piece.wav trim `expr $2 - $e - $l`s \ `expr $3 - $2 + $e + $l + $e`s sox "$1" part1.wav trim 0 `expr $4 + $e`s sox "$1" part2.wav trim `expr $4 + $3 - $2 - $e - $l`s sox part1.wav piece.wav part2.wav "$5" splice \ `expr $4 + $e`s \ `expr $4 + $e + $3 - $2 + $e + $l + $e`sIn the above Bourne shell script, two splices are used to `copy and paste' audio.
It is also possible to use this effect to perform general cross-fades, e.g. to join two songs. In this case, excess would typically be an number of seconds, the -q option would typically be given (to select an `equal power' cross-fade), and leeway should be zero (which is the default if -q is given). For example, if f1.wav and f2.wav are audio files to be cross-faded, then
sox f1.wav f2.wav out.wav splice -q $(soxi -D f1.wav),3cross-fades the files where the point of equal loudness is 3 seconds before the end of f1.wav, i.e. the total length of the cross-fade is 2 × 3 = 6 seconds (Note: the $(...) notation is POSIX shell).
The information is output to the `standard error' (stderr) stream and is calculated, where n is the duration of the audio in samples, c is the number of audio channels, r is the audio sample rate, and xk represents the PCM value (in the range -1 to +1 by default) of each successive sample in the audio, as follows:
|Samples read||n×c|| |
|Scaled by||See -s below.|
The maximum sample value in the audio; usually this will be a positive number.
The minimum sample value in the audio; usually this will be a negative number.
The average of the absolute value of each sample in the audio.
The average of each sample in the audio. If this figure is non-zero, then it indicates the
presence of a D.C. offset (which could be removed using the
The level of a D.C. signal that would have the same power
as the audio's average power.
|Rough frequency||In Hz.|
The parameter to the
effect which would make the audio as loud as possible without clipping.
Note: See the discussion on
above for reasons why it is rarely a good idea actually to do this.
Note that the delta measurements are not applicable for multi-channel audio.
The -s option can be used to scale the input data by a given factor. The default value of scale is 2147483647 (i.e. the maximum value of a 32-bit signed integer). 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.
The -v option displays only the `Volume Adjustment' value.
The -freq option calculates the input's power spectrum (4096 point DFT) instead of the statistics listed above. This should only be used with a single channel audio file.
The -d option displays a hex dump of the 32-bit signed PCM data audio in SoX's internal buffer. This is mainly used to help track down endian problems that sometimes occur in cross-platform versions of SoX.
See also the stats effect.
For example, for a typical well-mastered stereo music file:
Overall Left Right
|DC offset 0.000803 -0.000391 0.000803|
|Min level -0.750977 -0.750977 -0.653412|
|Max level 0.708801 0.708801 0.653534|
|Pk lev dB -2.49 -2.49 -3.69|
|RMS lev dB -19.41 -19.13 -19.71|
|RMS Pk dB -13.82 -13.82 -14.38|
|RMS Tr dB -85.25 -85.25 -82.66|
|Crest factor - 6.79 6.32|
|Flat factor 0.00 0.00 0.00|
|Pk count 2 2 2|
|Bit-depth 16/16 16/16 16/16|
|Num samples 7.72M|
|Length s 174.973|
|Scale max 1.000000|
|Window s 0.050|
DC offset, Min level, and Max level are shown, by default, in the range ±1. If the -b (bits) options is given, then these three measurements will be scaled to a signed integer with the given number of bits; for example, for 16 bits, the scale would be -32768 to +32767. The -x option behaves the same way as -b except that the signed integer values are displayed in hexadecimal. The -s option scales the three measurements by a given floating-point number.
Pk lev dB and RMS lev dB are standard peak and RMS level measured in dBFS. RMS Pk dB and RMS Tr dB are peak and trough values for RMS level measured over a short window (default 50ms).
Crest factor is the standard ratio of peak to RMS level (note: not in dB).
Flat factor is a measure of the flatness (i.e. consecutive samples with the same value) of the signal at its peak levels (i.e. either Min level, or Max level). Pk count is the number of occasions (not the number of samples) that the signal attained either Min level, or Max level.
The right-hand Bit-depth figure is the standard definition of bit-depth i.e. bits less significant than the given number are fixed at zero. The left-hand figure is the number of most significant bits that are fixed at zero (or one for negative numbers) subtracted from the right-hand figure (the number subtracted is directly related to Pk lev dB).
For multi-channel audio, an overall figure for each of the above measurements is given and derived from the channel figures as follows: DC offset: maximum magnitude; Max level, Pk lev dB, RMS Pk dB, Bit-depth: maximum; Min level, RMS Tr dB: minimum; RMS lev dB, Flat factor, Pk count: average; Crest factor: not applicable.
Length s is the duration in seconds of the audio, and Num samples is equal to the sample-rate multiplied by Length. Scale Max is the scaling applied to the first three measurements; specifically, it is the maximum value that could apply to Max level. Window s is the length of the window used for the peak and trough RMS measurements.
See also the stat effect.
factor of stretching: >1 lengthen, <1 shorten duration. window size is in ms. Default is 20ms. The fade option, can be `lin'. shift ratio, in [0 1]. Default depends on stretch factor. 1 to shorten, 0.8 to lengthen. The fading ratio, in [0 0.5]. The amount of a fade's default depends on factor and shift.
See also the tempo effect.
Though this effect is used to generate audio, an input file must still be given, the characteristics of which will be used to set the synthesised audio length, the number of channels, and the sampling rate; however, since the input file's audio is not normally needed, a `null file' (with the special name -n) is often given instead (and the length specified as a parameter to synth or by another given effect that can has an associated length).
For example, the following produces a 3 second, 48kHz, audio file containing a sine-wave swept from 300 to 3300 Hz:
sox -n output.wav synth 3 sine 300-3300and this produces an 8 kHz version:
sox -r 8000 -n output.wav synth 3 sine 300-3300Multiple channels can be synthesised by specifying the set of parameters shown between braces multiple times; the following puts the swept tone in the left channel and adds `brown' noise in the right:
sox -n output.wav synth 3 sine 300-3300 brownnoiseThe following example shows how two synth effects can be cascaded to create a more complex waveform:
play -n synth 0.5 sine 200-500 synth 0.5 sine fmod 700-100Frequencies can also be given in `scientific' note notation, or, by prefixing a `%' character, as a number of semitones relative to `middle A' (440 Hz). For example, the following could be used to help tune a guitar's low `E' string:
play -n synth 4 pluck %-29or with a (Bourne shell) loop, the whole guitar:
for n in E2 A2 D3 G3 B3 E4; do play -n synth 4 pluck $n repeat 2; doneSee the delay effect (above) and the reference to `SoX scripting examples' (below) for more synth examples.
N.B. This effect generates audio at maximum volume (0dBFS), which means that there is a high chance of clipping when using the audio subsequently, so in many cases, you will want to follow this effect with the gain effect to prevent this from happening. (See also Clipping above.) Note that, by default, the synth effect incorporates the functionality of gain -h (see the gain effect for details); synth's -n option may be given to disable this behaviour.
A detailed description of each synth parameter follows:
len is the length of audio to synthesise expressed as a time or as a number of samples; 0=inputlength, default=0.
The format for specifying lengths in time is hh:mm:ss.frac. The format for specifying sample counts is the number of samples with the letter `s' appended to it.
type is one of sine, square, triangle, sawtooth, trapezium, exp, [white]noise, tpdfnoise pinknoise, brownnoise, pluck; default=sine.
combine is one of create, mix, amod (amplitude modulation), fmod (frequency modulation); default=create.
freq/freq2 are the frequencies at the beginning/end of synthesis in Hz or, if preceded with `%', semitones relative to A (440 Hz); alternatively, `scientific' note notation (e.g. E2) may be used. The default frequency is 440Hz. By default, the tuning used with the note notations is `equal temperament'; the -j KEY option selects `just intonation', where KEY is an integer number of semitones relative to A (so for example, -9 or 3 selects the key of C), or a note in scientific notation.
If freq2 is given, then len must also have been given and the generated tone will be swept between the given frequencies. The two given frequencies must be separated by one of the characters `:', `+', `/', or `-'. This character is used to specify the sweep function as follows:
off is the bias (DC-offset) of the signal in percent; default=0.
ph is the phase shift in percentage of 1 cycle; default=0. Not used for noise.
p1 is the percentage of each cycle that is `on' (square), or `rising' (triangle, exp, trapezium); default=50 (square, triangle, exp), default=10 (trapezium), or sustain (pluck); default=40.
p2 (trapezium): the percentage through each cycle at which `falling' begins; default=50. exp: the amplitude in multiples of 2dB; default=50, or tone-1 (pluck); default=20.
p3 (trapezium): the percentage through each cycle at which `falling' ends; default=60, or tone-2 (pluck); default=90.
By default, linear searches are used to find the best overlapping points. If the optional -q parameter is given, tree searches are used instead. This makes the effect work more quickly, but the result may not sound as good. However, if you must improve the processing speed, this generally reduces the sound quality less than reducing the search or overlap values.
The -m option is used to optimize default values of segment, search and overlap for music processing.
The -s option is used to optimize default values of segment, search and overlap for speech processing.
The -l option is used to optimize default values of segment, search and overlap for `linear' processing that tends to cause more noticeable distortion but may be useful when factor is close to 1.
If -m, -s, or -l is specified, the default value of segment will be calculated based on factor, while default search and overlap values are based on segment. Any values you provide still override these default values.
factor gives the ratio of new tempo to the old tempo, so e.g. 1.1 speeds up the tempo by 10%, and 0.9 slows it down by 10%.
The optional segment parameter selects the algorithm's segment size in milliseconds. If no other flags are specified, the default value is 82 and is typically suited to making small changes to the tempo of music. For larger changes (e.g. a factor of 2), 41 ms may give a better result. The -m, -s, and -l flags will cause the segment default to be automatically adjusted based on factor. For example using -s (for speech) with a tempo of 1.25 will calculate a default segment value of 32.
The optional search parameter gives the audio length in milliseconds over which the algorithm will search for overlapping points. If no other flags are specified, the default value is 14.68. Larger values use more processing time and may or may not produce better results. A practical maximum is half the value of segment. Search can be reduced to cut processing time at the risk of degrading output quality. The -m, -s, and -l flags will cause the search default to be automatically adjusted based on segment.
The optional overlap parameter gives the segment overlap length in milliseconds. Default value is 12, but -m, -s, or -l flags automatically adjust overlap based on segment size. Increasing overlap increases processing time and may increase quality. A practical maximum for overlap is the value of search, with overlap typically being (at least) a little smaller then search.
See also speed for an effect that changes tempo and pitch together, pitch and bend for effects that change pitch only, and stretch for an effect that changes tempo using a different algorithm.
If a position is preceded by an equals or minus sign, it is interpreted relative to the beginning or the end of the audio, respectively. (The audio length must be known for end-relative locations to work.) Otherwise, it is considered an offset from the last position, or from the start of audio for the first parameter. Using a value of 0 for the first position parameter allows copying from the beginning of the audio.
All parameters can be specified using either an amount of time or an exact count of samples. The format for specifying lengths in time is hh:mm:ss.frac. A value of 1:30.5 for the first parameter will not start until 1 minute, thirty and ½ seconds into the audio. The format for specifying sample counts is the number of samples with the letter `s' appended to it. A value of 8000s for the first parameter will wait until 8000 samples are read before starting to process audio.
sox infile outfile trim 0 10will copy the first ten seconds, while
play infile trim 12:34 =15:00 -2:00will play from 12 minutes 34 seconds into the audio up to 15 minutes into the audio (i.e. 2 minutes and 26 seconds long), then resume playing two minutes before the end of audio.
For a general resampling effect with anti-aliasing, see rate. See also downsample.
play speech.wav norm vadto trim from the front,
play speech.wav norm reverse vad reverseto trim from the back, and
play speech.wav norm vad reverse vad reverseto trim from both ends. The use of the norm effect is recommended, but remember that neither reverse nor norm is suitable for use with streamed audio.
Default values are shown in parenthesis.
The amount to change the volume is given by gain which is interpreted, according to the given type, as follows: if type is amplitude (or is omitted), then gain is an amplitude (i.e. voltage or linear) ratio, if power, then a power (i.e. wattage or voltage-squared) ratio, and if dB, then a power change in dB.
When type is amplitude or power, a gain of 1 leaves the volume unchanged, less than 1 decreases it, and greater than 1 increases it; a negative gain inverts the audio signal in addition to adjusting its volume.
When type is dB, a gain of 0 leaves the volume unchanged, less than 0 decreases it, and greater than 0 increases it.
See  for a detailed discussion on electrical (and hence audio signal) voltage and power ratios.
Beware of Clipping when the increasing the volume.
The gain and the type parameters can be concatenated if desired, e.g. vol 10dB.
An optional limitergain value can be specified and should be a value much less than 1 (e.g. 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 the audio that needed to be limited.
See also gain for a volume-changing effect with different capabilities, and compand for a dynamic-range compression/expansion/limiting effect.
When reducing the number of channels it is possible to use the -l, -r, -f, -b, -1, -2, -3, -4, options to select only the left, right, front, back channel(s) or specific channel for the output instead of averaging the channels. The -l, and -r options will do averaging in quad-channel files so select the exact channel to prevent this.
The mixer effect can also be invoked with up to 16 numbers, separated by commas, which specify the proportion (0 = 0% and 1 = 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 follows: 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|
|4||1||4||lf → l, rf → l, lb → l, rb → l|
|4||2||2||lf → l&rf → r, lb → l&rb → r|
|4||4||2||front balance, back balance|
This effect has been superseded by the remix effect that handles any number of channels.
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 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details.