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]] ...
Almost all SoX functionality is available using just the sox command, however, 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) |
To show how this works in practise, here is a selection of examples of how SoX might be used. The simple
sox recital.au recital.wav
translates an audio file in Sun AU format to a Microsoft WAV file, whilst
sox recital.au -r 12k -b 8 -c 1 recital.wav vol 0.7 dither
performs the same format translation, but also changes the audio
sampling rate & sample size, down-mixes to mono, and applies
the vol and dither effects.
sox -r 8k -u -b 8 -c 1 voice-memo.raw voice-memo.wav
converts `raw' (a.k.a. `headerless') audio to a self-descibing file format,
sox slow.aiff fixed.aiff speed 1.027
adjusts audio speed,
sox short.au long.au longer.au
concatenates two audio files, and
sox -m music.mp3 voice.wav mixed.flac
mixes together two audio files.
play dqThe Moonbeams/Greatest/*.oggdq bass +3
plays a collection of audio files whilst applying a bass boosting effect,
play -n -c1 synth sin %-12 sin %-9 sin %-5 sin %-2 fade q 0.1 1 0.1
plays a synthesised `A minor seventh' chord with a pipe-organ sound,
rec -c 2 test.aiff trim 0 10
records 10 seconds of stereo audio, and
rec -M take1.aiff take1-dub.aiff
records a new track in a multi-track recording.
rec -r 44100 -2 -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 : restart
records a stream of audio such as LP/cassette and splits in to multiple
audio files at points with 2 seconds of silence. Also does not start
recording until it detects audio is playing and stops after it sees
10 minutes of silence.
N.B. Detailed explanations of how to use all SoX parameters, file formats, and effects can be found below in this manual, and in soxformat(7).
The following four characteristics are sufficient to describe the format of audio data such that it can be processed with SoX:
The term `bit-rate' is sometimes used as an overall measure of an audio format and may incorporate elements of all of the above.
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:
| 1. | Command-line format options. | |
| 2. | The contents of the file header. | |
| 3. | The filename extension. |
To set the output file format, SoX will use, in order of precedence and as given or available:
| 1. | Command-line format options. | |
| 2. | The filename extension. | |
| 3. |
The input file format characteristics, or the closest
to them that is supported by the output file type.
|
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.
set AUDIODRIVER=oss
play ...
For
rec,
play,
and
sox,
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:0
play ...
sox ... -t alsa
(Note that the syntax of the
set
command may vary from system to system.)
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, here, the default rate quality level is set to `low'; however, this can be changed if desired, by explicitly specifing the rate effect with a different quality level, e.g.
play ... rate -m
or by setting the environment varible
PLAY_RATE_ARG
to the desired quality option, e.g.
set PLAY_RATE_ARG=-m
play ...
(Note that the syntax of the
set
command may vary from system to system.)
To help with setting a suitable recording level, SoX includes a simple VU 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).
Formats that discard audio signal information are called `lossy', and formats that do not, `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 recompressing 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; hence this is also recommended.
In SoX, clipping could occur, as you might expect, when using the vol effect to increase the audio volume, but 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 re-sampling, and processing by analogue components that 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 does not exceed around 70% of the maximum (linear) range available, as this will avoid the majority of clipping problems. SoX's stat effect 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.au bright.au gain -6 treble +6
guarantees that the treble boost will not clip.
If clipping occurs at any point during processing, then SoX will display a warning message to that effect.
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, however, 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 orignal 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.
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. So here, if manual volume adjustments are not given, to ensure that clipping does not occur, SoX will automatically adjust the volume (amplitude) of each input signal by a factor of S1/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 through the mixed audio 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 appropriately equal to that of one of the input signals. This is achieved by balancing using a factor of S1/srn instead of S1/n. Note that this balancing factor does not guarantee that no clipping will occur, however, in many cases, the number of clips will be low and the resultant distortion imperceptable.
This behavior 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 behavior 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 in to 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 usually 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; however, this is generally not difficult since most audio filenames have a filename `extension', whilst effect-names do not.
sox -M -t wav "|genw --imd -" -t wav "|genw --thd -" out.wavIf -t is not given then the signal is assumed (and checked) to be in SoX's native .sox format (see -p below and soxformat(7)).
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).
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.
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 strongly recommended; a `shell' alias, script, or batch file
may be an appropriate way of permanently enabling it.
See Input File Combining above for a description of the different combining methods.
sox --plot octave input-file -n highpass 1320 > plot.m
octave plot.m
|
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.
See also the stat effect for information on how to find the maximum volume of an audio file; this can be used to help select suitable values for this option.
See also Input File Balancing above.
Alternatively, some effects (e.g. synth, remix) determine what will be the number of output channels; in this case, neither this option nor the mixer effect is necessary.
Note that explicitly specifying other encoding types (e.g. MP3, FLAC) is not necessary since they can be inferred from the file type or header.
If the input and output files have different rates then a sample rate change effect must be run. Since SoX has multiple rate changing effects, the user can specify which to use as an effect. If no rate change effect is specified then the rate effect will be chosen by default.
The -t option 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 normal 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.s2 trimmed.s2 trim 2
trimmed.s2 will be created as little-endian;
sox -B audio.s2 -B trimmed.s2 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.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 dqdq .
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 chain 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 reach end of file.
A effects chain is terminated by placing a : (colon) after an effect. Any following effects are apart 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 keep small, relative to the sample rate, if the terminating effect can not 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 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 effect's first line of 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 filter for a bandpass filter with steeper shoulders.
These effects support the --plot global option.
See also filter 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 [1].
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, yeilding four different notes in total:
play -n synth 2.5 sin 667 gain 1 \
bend .35,180,.25 .15,740,.53 0,-520,.3
Note that the clipping that is produced in this example is deliberate;
to remove it, use
gain -5
in place of
gain 1.
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.
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 -t
Two 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 -s
A 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.au asz-car.au compand 0.3,1 6:-70,-60,-20 -5 -90 0.2
The 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.
This effect supports the --plot global option (for the transfer function).
See also mcompand for a multiple-band companding effect.
sox infile outfile norm -i contrast
The first option is the dcshift value. It is a floating point number that indicates the amount to shift.
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 out.au synth 5 sin %0 50 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 deemph
and then burn track1-deemph.wav to CD, or
play track1-deemph.wav
or simply
play track1.wav deemph
The 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 sin %-21.5 sin %-14.5 sin %-9.5 sin %-5.5 \
sin %-2.5 sin %2.5 gain -5.4 fade h 0.008 2 1.5 \
delay 0 .27 .54 .76 1.01 1.3 remix - fade h 0.1 2.72 2.5
By default, the amount of noise added is ±½ bit for RPDF, ±1 bit for TPDF; the optional depth parameter (0.5 to 1) is a (linear or voltage) multiplier of this amount. Reducing this value reduces the audibility of the added white noise, but correspondingly creates residual quantization noise, so it should not normally be changed.
This effect should not be followed by any other effect that affects the audio.
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 0.4
If the delay is very short, then it sound like a (metallic) robot playing
music:
play lead.aiff echo 0.8 0.88 6 0.4
A longer delay will sound like an open air concert in the mountains:
play lead.aiff echo 0.8 0.9 1000 0.3
One 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 0.3
The sample will be bounced twice in asymmetric echos:
play lead.aiff echos 0.8 0.7 700 0.25 900 0.3
The 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 [1].
This effect supports the --plot global option.
See also bass and treble for shelving equalisation effects.
For fade-ins, this starts from the first sample and ramps the volume of the audio from 0 to full volume over fade-in-length seconds. Specify 0 seconds if no fade-in is wanted.
For fade-outs, the audio will be truncated at stop-time and the volume will be ramped from full volume down to 0 starting at fade-out-length seconds before the stop-time. If fade-out-length is not specified, it defaults to the same value as fade-in-length. No fade-out is performed if 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').
An optional type can be specified to change the type of envelope. Choices are q for quarter of a sine wave, h for half a sine wave, t for linear slope, l for logarithmic, and p for inverted parabola. The default is logarithmic.
A low-pass filter is obtained by leaving low unspecified, or 0. A high-pass filter is obtained by leaving high unspecified, or 0, or greater than or equal to the Nyquist frequency.
The window-len, if unspecified, defaults to 128. Longer windows give a sharper cut-off, smaller windows a more gradual cut-off.
The beta parameter determines the type of filter window used. Any value greater than 2 is the beta for a Kaiser window. Beta < 2 selects a Blackman-Nuttall window. If unspecified, the default is a Kaiser window with beta 16.
In the case of Kaiser window (beta > 2), lower betas produce a somewhat faster transition from pass-band to stop-band, at the cost of noticeable artifacts. A beta of 16 is the default, beta less than 10 is not recommended. If you want a sharper cut-off, don't use low beta's, use a longer sample window. A Blackman-Nuttall window is selected by specifying any `beta' < 2, and the Blackman-Nuttall window has somewhat steeper cut-off than the default Kaiser window. You will probably not need to use the beta parameter at all, unless you are just curious about comparing the effects of Blackman-Nuttall vs. Kaiser windows.
This effect supports the --plot global option.
All parameters are optional (right to left).
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See also the vol effect.
These effects support the --plot global option.
See also filter for filters with a steeper roll-off.
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 filter 8000- 32 100 mcompand \
dq0.005,0.1 -47,-40,-34,-34,-17,-33dq 100 \
dq0.003,0.05 -47,-40,-34,-34,-17,-33dq 400 \
dq0.000625,0.0125 -47,-40,-34,-34,-15,-33dq 1600 \
dq0.0001,0.025 -47,-40,-34,-34,-31,-31,-0,-30dq 6400 \
dq0,0.025 -38,-31,-28,-28,-0,-25dq \
gain 15 highpass 22 highpass 22 filter -17500 256 \
gain 9 lowpass -1 17801
The 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 preemphasis.
See also compand for a single-band companding effect.
This effect is automatically used when the number of input channels differ from the number of output channels. When reducing the number of channels it is possible to manually specify the mixer effect and 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.
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See also remix for a mixing effect that handles any number of channels.
sox speech.au -n trim 0 1.5 noiseprof speech.noise-profile
To 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.au cleaned.au noisered speech.noise-profile 0.3
How 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.au -n trim 0 1 noiseprof | play noisy.au noisered
To create a normalised copy of an audio file,
sox infile outfile norm
can be used, though note that if `infile' has a simple encoding (e.g.
PCM), then
sox infile outfile vol `sox infile -n stat -v 2>&1`
(on systems that support this construct) might be quicker to execute
(though perhaps not to type!) as it doesn't require a temporary file.
For a more complex example, suppose that `effect1' performs some unknown or unpredictable attenuation and that `effect2' requires up to 10dB of headroom, then
sox infile outfile effect1 norm -10 effect2 norm
gives both effect2 and the output file the highest possible signal
levels.
Normally, audio is normalised based on the level of the channel with the highest peak level, which means that whilst all channels are adjusted, only one channel attains the normalised level. If the -i option is given, then each channel is treated individually and will attain the normalised level.
If the -b option is given (with a multi-channel audio file), then the audio channels will be balanced; i.e. the RMS level of each channel will be normalised to that of the channel with the highest RMS level. This can be used, for example, to correct stereo imbalance. Note that -b can cause clipping.
In most cases, norm -3 should be the maximum level used at the output file (to leave headroom for playback-resampling, etc.). See also the discussions of Clipping and Replay Gain above.
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.
For example:
play snare.flac phaser 0.8 0.74 3 0.4 0.5 -t
Gentler:
play snare.flac phaser 0.9 0.85 4 0.23 1.3 -s
A popular sound:
play snare.flac phaser 0.89 0.85 1 0.24 2 -t
More 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.
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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'.
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.au -r 48k output.au bass -3
sox input.au output.au bass -3 rate 48k
though 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:
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N.B. Override options can not 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 noticable 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 orignal 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 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 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%.
Examples:
sox input.wav -b 16 output.wav rate -s -a 44100 dither
default (high) quality resampling; overrides: steep filter, allow
aliasing; to 44.1kHz sample rate; dither output to 16-bit WAV
file.
sox input.wav -b 24 output.aiff rate -v -L -b 90 48k
very high quality resampling; overrides: linear phase, band-width 90%;
to 48k sample rate; store output to 24-bit AIFF file.
| * | * | * |
The pitch, speed and tempo effects all use the rate effect at their core.
See also resample, polyphase and rabbit for other sample-rate changing effects.
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.au output.au remix 6 7 8 0
creates 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.au output.au remix 1-3,7 3
creates 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.au output.au 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 S1/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:
| Letter | Volume number | Notes |
| p | power adjust in dB | 0 = no change |
| i | power adjust in dB |
As `p', but invert the audio
|
| v | voltage multiplier |
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, S1/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.au output.au remix 1,2 3,4v0.8
results in channel level multipliers of 0.5,0.5 1,0.8, whereas
sox input.au output.au remix -a 1,2 3,4v0.8
results 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.au output.au remix -m 1,2 3,4v0.8
results 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.au is stereo, then
sox input.au output.au remix 1,2i
is a mono equivalent of the
oops
effect.
If the -p option is given, then any automatic S1/n scaling is replaced by S1/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.au containing six audio channels were given, the script would produce six output files: input-01.au, input-02.au, ..., input-06.au.
See also mixer and swap for similar effects.
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 [3] 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.au gain -3 pad 0 3 reverb
This effect supports the --plot global option.
Removes silence from the beginning, middle, or end of the audio. Silence is anything below 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%
By default, the speed change is performed by resampling with the rate effect 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.
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 intensity) of the pixels in the X-Y plane.
This effect supports only one channel; for multi-channel input files, use either SoX's -c 1 option with the output file (to obtain a spectrogram on the mix-down), or the remix n effect to select a particular channel. Be aware though, that both of these methods affect the audio in the effects chain.
The maximum width of the spectrogram is 999 pixels; if the audio length and the given -x number are such that this would be exceeded, then the spectrogram (and the effects chain) will be truncated. To move the spectrogram to a point later in the audio stream, first invoke the trim effect; e.g.
sox audio.ogg -n trim 1:00 spectrogramstarts the spectrogram at 1 minute through the audio.
To increase the frequency resolution without increasing the height of the spectrogram, the rate effect may be invoked to reduce the sampling rate of the signal before invoking spectrogram; e.g.
sox audio.ogg -r 4k -n rate spectrogramallows detailed analysis of frequencies up to 2kHz (half the sampling rate).
sox -n -n synth 6 tri 10k:14k spectrogram -z 100 -w k
Append the following to the `chime' example in the
delay
effect to see its spectrogram:
rate 2k spectrogram -x 200 -Z -15 -w k
For the ability to perform off-line processing of spectral data, see the
stat
effect.
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).
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.au part1.au trim 0 30.130
(5 ms excess, after the first verse starts)
sox long.au part2.au trim 1:03.422
(5 ms excess plus 5 ms leeway, before the second verse starts)
sox part1.au part2.au just-right.au splice 30.130
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.au trim `expr $2 - $e - $l`s \
`expr $3 - $2 + $e + $l + $e`s
sox "$1" part1.au trim 0 `expr $4 + $e`s
sox "$1" part2.au trim `expr $4 + $3 - $2 - $e - $l`s
sox part1.au piece.au part2.au "$5" splice \
`expr $4 + $e`s \
`expr $4 + $e + $3 - $2 + $e + $l + $e`s
In the above Bourne shell script,
two splices are used to `copy and paste' audio.
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 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).
| * | * | * |
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, and leeway should be set to zero.
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 | |
| Length (seconds) | n÷r | |
| Scaled by | See -s below. | |
| Maximum amplitude | max(xk) |
The maximum sample value in the audio; usually this will be a positive number.
|
| Minimum amplitude | min(xk) |
The minimum sample value in the audio; usually this will be a negative number.
|
| Midline amplitude | ½min(xk)+½max(xk) | |
| Mean norm | S1/nSigma|xk| |
The average of the absolute value of each sample in the audio.
|
| Mean amplitude | S1/nSigmaxk |
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
dcshift
effect).
|
| RMS amplitude | sr(S1/nSigmaxkS2) |
The level of a D.C. signal that would have the same power
as the audio's average power.
|
| Maximum delta | max(|xk-xk-1|) | |
| Minimum delta | min(|xk-xk-1|) | |
| Mean delta | S1/n-1Sigma|xk-xk-1| | |
| RMS delta | sr(S1/n-1Sigma(xk-xk-1)S2) | |
| Rough frequency | In Hz. | |
| Volume Adjustment |
The parameter to the
vol
effect which would make the audio as loud as possible without clipping.
Note: See the discussion on
Clipping
above for reasons why it is rarely a good idea actually to do this.
|
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.
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 remix 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.au synth 3 sine 300-3300
and this produces an 8 kHz version:
sox -r 8000 -n output.au synth 3 sine 300-3300
Multiple 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.au synth 3 sine 300-3300 brownnoise
The following example shows how two synth effects can be cascaded
to create a more complex waveform:
sox -n output.au synth 0.5 sine 200-500 rs
synth 0.5 sine fmod 700-100
Frequencies can also be given as a number of musical semitones relative
to `middle A' (440 Hz) by prefixing a `%' character; for example, the
following could be used to help tune a guitar's `E' strings:
play -n synth sine %-17
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 most cases, you will want to follow this effect with the gain
effect to prevent this from happening. (See also
Clipping
above.)
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, pinknoise, brownnoise; 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); for both, default=%0. 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).
p2 (trapezium): the percentage through each cycle at which `falling' begins; default=50. exp: the amplitude in percent; default=100.
p3 (trapezium): the percentage through each cycle at which `falling' ends; default=60.
By default, linear searches are used to find the best overlapping points; if the optional -q parameter is given, tree searches are used instead, giving a quicker, but possibly lower quality, result.
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. 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), 50 ms may give a better result. When changing the tempo of speech, a segment size of around 30 ms often works well.
The optional search parameter gives the audio length in milliseconds (default 14) over which the algorithm will search for overlapping points. Larger values use more processing time and do not necessarily produce better results.
The optional overlap parameter gives the segment overlap length in milliseconds (default 12).
See also speed for an effect that changes tempo and pitch together, and pitch for an effect that changes pitch without changing tempo.
Note: This effect is a special case of the synth effect.
The optional length parameter tells the number of samples to output after the start sample and is used to trim off the back side of the audio. Using a value of 0 for the start parameter will allow trimming off the back side only.
Both options can be specified using either an amount of time or an exact count of samples. The format for specifying lengths in time is hh:mm:ss.frac. A start value of 1:30.5 will not start until 1 minute, thirty and 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 will wait until 8000 samples are read before starting to process audio.
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 [4] 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 compand for a dynamic-range compression/expansion/limiting effect.
The direction is a value from -1 to 1. -1 represents far left and 1 represents far right.
If the -w parameter is nut, then a Blackman-Nuttall (~90 dB stop-band) window will be used; ham selects a Hamming (~43 dB stop-band) window. The default is Blackman-Nuttall.
The -width parameter specifies the (approximate) width of the filter. The default is 1024 samples, which produces reasonable results.
The -cut-off value (c) specifies the filter cut-off frequency in terms of fraction of frequency bandwidth, also know as the Nyquist frequency. See the resample effect for further information on Nyquist frequency. If up-sampling, then this is the fraction of the original signal that should go through. If down-sampling, this is the fraction of the signal left after down-sampling. The default is 0.95.
See also rate, rabbit and resample for other sample-rate changing effects.
See http://www.mega-nerd.com/SRC for details of the algorithms. Algorithms 0 through 2 are progressively faster and lower quality versions of the sinc algorithm; the default is -c0. Algorithm 3 is zero-order hold, and 4 is linear interpolation.
See also rate, polyphase and resample for other sample-rate changing effects, and see resample for more discussion of resampling.
By default, linear interpolation of the filter coefficients is used, with a window width about 45 samples at the lower of the two rates. This gives an accuracy of about 16 bits, but insufficient stop-band rejection in the case that you want to have roll-off greater than about 0.8 of the Nyquist frequency.
The -q* options will change the default values for roll-off and beta as well as use quadratic interpolation of filter coefficients, resulting in about 24 bits precision. The -qs, -q, or -ql options specify increased accuracy at the cost of lower execution speed. It is optional to specify roll-off and beta parameters when using the -q* options.
Following is a table of the reasonable defaults which are built-in to SoX:
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-qs, -q, or -ql use window lengths of 45, 75, or 149 samples, respectively, at the lower sample-rate of the two files. This means progressively sharper stop-band rejection, at proportionally slower execution times.
rolloff refers to the cut-off frequency of the low pass filter and is given in terms of the Nyquist frequency for the lower sample rate. rolloff therefore should be something between 0 and 1, in practise 0.8-0.95. The defaults are indicated above.
The Nyquist frequency is equal to half the sample rate. Logically, this is because the A/D converter needs at least 2 samples to detect 1 cycle at the Nyquist frequency. Frequencies higher then the Nyquist will actually appear as lower frequencies to the A/D converter and is called aliasing. Normally, A/D converts run the signal through a lowpass filter first to avoid these problems.
Similar problems will happen in software when reducing the sample rate of an audio file (frequencies above the new Nyquist frequency can be aliased to lower frequencies). Therefore, a good resample effect will remove all frequency information above the new Nyquist frequency.
The rolloff refers to how close to the Nyquist frequency this cut-off is, with closer being better. When increasing the sample rate of an audio file you would not expect to have any frequencies exist that are past the original Nyquist frequency. Because of resampling properties, it is common to have aliasing artifacts created above the old Nyquist frequency. In that case the rolloff refers to how close to the original Nyquist frequency to use a highpass filter to remove these artifacts, with closer also being better.
The beta, if unspecified, defaults to 16. This selects a Kaiser window. You can select a Blackman-Nuttall window by specifying anything < 2 here. For more discussion of beta, look under the filter effect.
Default parameters are, as indicated above, Kaiser window of length 45, roll-off 0.80, beta 16, linear interpolation.
Note: -qs is only slightly slower, but more accurate for 16-bit or higher precision.
See also rate, polyphase and rabbit for other sample-rate changing effects. There is a detailed analysis of resample and polyphase at http://leute.server.de/wilde/resample.html; see rabbit for a pointer to its own documentation.
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.