ref: 556f4b5081288f19f89873b5a8353151b1ab4498
dir: /sox.txt/
SoX(1) SoX(1) NAME sox - Sound eXchange : universal sound sample translator SYNOPSIS sox infile outfile sox infile outfile [ effect [ effect options ... ] ] sox infile -e effect [ effect options ... ] sox [ general options ] [ format options ] infile [ for- mat options ] outfile [ effect [ effect options ... ] ] General options: [ -e ] [ -h ] [ -p ] [ -v volume ] [ -V ] Format options: [ -t filetype ] [ -r rate ] [ -s/-u/-U/-A/-a/-i/-g ] [ -b/-w/-l/-f/-d/-D ] [ -c channels ] [ -x ] Effects: avg [ -l | -r ] band [ -n ] center [ width ] bandpass frequency bandwidth bandreject frequency bandwidth check chorus gain-in gain out delay decay speed depth -s | -t [ delay decay speed depth -s | -t ] compand attack1,decay1[,attack2,decay2...] in-dB1,out-dB1[,in-dB2,out-dB2...] [gain] [initial-volume] copy cut deemph echo gain-in gain-out delay decay [ delay decay ...] echos gain-in gain-out delay decay [ delay decay ...] filter [ low ]-[ high ] [ window-len [ beta ]] flanger gain-in gain-out delay decay speed -s | -t highp center highpass frequency lowp center lowpass frequency map mask pan direction phaser gain-in gain-out delay decay speed -s | -t pick pitch shift [ width interpole fade ] polyphase [ -w < nut / ham > ] [ -width < long / short / # > ] [ -cutoff # ] rate resample reverb gain-out reverb-time delay [ delay ... ] reverse speed factor split stat [ debug | -v ] December 10, 1999 1 SoX(1) SoX(1) stretch [ factor [ window fade shift fading ] swap [ 1 2 3 4 ] vibro speed [ depth ] vol gain [ type ] DESCRIPTION SoX is a command line program that can convert most popu- lar audio files to most other popular audio file formats. It can optionally apply a sound effect to the file during this translation. There are two types of audio files formats that SoX can work with. The first are self-describing file formats. These contain a header that completely describe the char- acteristics of the audio data that follows. The second type are headerless data, or sometimes called raw data. A user must pass enough information to SoX on the command line so that it knows what type of data it contains. Audio data can usually be totally described by four char- acteristics: rate The sample rate is in samples per second. For example, CD sample rates are at 44100. data type What format the data is stored in. Most popular are 8-bit or 16-bit words. data format What encoding the data type uses. Examples are u-law, ADPCM, or signed linear data. channels How many channels are contained in the audio data. Mono and Stereo are the two most common. Please refer to the soxexam(1) manual page for a long description with examples on how to use sox with various types of file formats. OPTIONS The option syntax is a little grotty, but in essence: sox file.au file.voc translates a sound file in SUN Sparc .AU format into a SoundBlaster .VOC file, while sox -v 0.5 file.au -r 12000 file.voc rate does the same format translation but also lowers the amplitude by 1/2 and changes the sampling rate from 8000 hertz to 12000 hertz via the rate sound effect loop. December 10, 1999 2 SoX(1) SoX(1) Format options: Format options effect the audio samples that they immedi- ately percede. If they are placed before the input file name then they effect the input data. If they are placed before the output file name then they will effect the out- put data. By taking advantage of this, you can override a input file's currupted header or produce an output file that is totally different style then the input file. -t filetype gives the type of the sound sample file. -r rate Give sample rate in Hertz of file. To cause the output file to have a different sample rate than the input file, include this option with the appropriate rate value along with the output options. If the input and output files have different rates then a sample rate change effect must be ran. If a sample rate changing effect is not specified then a default one will be used with its default parameters. -s/-u/-U/-A/-a/-i/-g The sample data format is signed linear (2's complement), unsigned linear, U-law (logarith- mic), A-law (logarithmic), ADPCM, IMA_ADPCM, or GSM. U-law and A-law are the U.S. and interna- tional standards for logarithmic telephone sound compression. ADPCM is form of sound compression that has a good compromise between good sound quality and fast encoding/decoding time. IMA_ADPCM is also a form of adpcm compression, slightly simpler and slightly lower fidelity than Microsoft's flavor of ADPCM. IMA_ADPCM is also called DVI_ADPCM. GSM is a standard used for telephone sound compression in European countries and its gaining popularity because of its quality. -b/-w/-l/-f/-d/-D The sample data type is in bytes, 16-bit words, 32-bit longwords, 32-bit floats, 64-bit double floats, or 80-bit IEEE floats. Floats and dou- ble floats are in native machine format. -x The sample data is in XINU format; that is, it comes from a machine with the opposite word order than yours and must be swapped according to the word-size given above. Only 16-bit and 32-bit integer data may be swapped. Machine- format floating-point data is not portable. IEEE floats are a fixed, portable format. December 10, 1999 3 SoX(1) SoX(1) -c channels The number of sound channels in the data file. This may be 1, 2, or 4; for mono, stereo, or quad sound data. To cause the output file to have a different number of channels than the input file, include this option with the appro- raite value with the output file options. If the input and output file have a different num- ber of channels then the avg effect must be used. If the avg effect is not specified on the command line it will be invoked with default parameters. General options: -e When used after the input file (so that it applies to the output file) it allows you to avoid giving an output filename and will not produce an output file. It will apply any spec- ified effects to the input file. This is mainly useful with the stat effect but can be used with others. -h Print version number and usage information. -p Run in preview mode and run fast. This will somewhat speed up sox when the output format has a different number of channels and a different rate than the input file. The order that the effects are run in will be arranged for maximum speed and not quality. -v volume Change amplitude (floating point); less than 1.0 decreases, greater than 1.0 increases. Note: we perceive volume logarithmically, not linearly. Note: see the stat effect. -V Print a description of processing phases. Use- ful for figuring out exactly how sox is mangling your sound samples. FILE TYPES SoX uses the file extension of the input and output file to determine what type of file format to use. This can be overriden by specifying the "-t" option on the command line. The input and output files may be read from standard in and out. This is done by specifing '-' as the filename. File formats which have headers are checked, if that header doesn't seem right, the program exits with an appropriate message. December 10, 1999 4 SoX(1) SoX(1) The following file formats are supported: .8svx Amiga 8SVX musical instrument description for- mat. .aiff AIFF files used on Apple IIc/IIgs and SGI. Note: the AIFF format supports only one SSND chunk. It does not support multiple sound chunks, or the 8SVX musical instrument descrip- tion format. AIFF files are multimedia archives and and can have multiple audio and picture chunks. You may need a separate archiver to work with them. .au SUN Microsystems AU files. There are apparently many types of .au files; DEC has invented its own with a different magic number and word order. The .au handler can read these files but will not write them. Some .au files have valid AU headers and some do not. The latter are probably original SUN u-law 8000 hz samples. These can be dealt with using the .ul format (see below). .avr Audio Visual Research The AVR format is produced by a number of com- mercial packages on the Mac. .cdr CD-R CD-R files are used in mastering music Compact Disks. The file format is, as you might expect, raw stereo raw unsigned samples at 44khz. But, there's some blocking/padding oddity in the for- mat, so it needs its own handler. .cvs Continuously Variable Slope Delta modulation Used to compress speech audio for applications such as voice mail. .dat Text Data files These files contain a textual representation of the sample data. There is one line at the beginning that contains the sample rate. Subse- quent lines contain two numeric data items: the time since the beginning of the sample and the sample value. Values are normalized so that the maximum and minimum are 1.00 and -1.00. This file format can be used to create data files for external programs such as FFT analyzers or graph routines. SoX can also convert a file in this format back into one of the other file formats. .gsm GSM 06.10 Lossy Speech Compression December 10, 1999 5 SoX(1) SoX(1) A standard for compressing speech which is used in the Global Standard for Mobil telecommunica- tions (GSM). Its good for its purpose, shrink- ing audio data size, but it will introduce lots of noise when a given sound sample is encoded and decoded multiple times. This format is used by some voice mail applications. It is rather CPU intensive. GSM in sox is optional and requires access to an external GSM library. To see if there is support for gsm run sox -h and look for it under the list of supported file formats. .hcom Macintosh HCOM files. These are (apparently) Mac FSSD files with some variant of Huffman com- pression. The Macintosh has wacky file formats and this format handler apparently doesn't han- dle all the ones it should. Mac users will need your usual arsenal of file converters to deal with an HCOM file under Unix or DOS. .maud An Amiga format An IFF-conform sound file type, registered by MS MacroSystem Computer GmbH, published along with the "Toccata" sound-card on the Amiga. Allows 8bit linear, 16bit linear, A-Law, u-law in mono and stereo. ossdsp OSS /dev/dsp device driver This is a pseudo-file type and can be optionally compiled into Sox. Run sox -h to see if you have support for this file type. When this driver is used it allows you to open up the OSS /dev/dsp file and configure it to use the same data type as passed in to Sox. It works for both playing and recording sound samples. When playing sound files it attempts to set up the OSS driver to use the same format as the input file. It is suggested to always override the output values to use the highest quality samples your sound card can handle. Example: -t ossdsp -w -s /dev/dsp .sf IRCAM Sound Files. SoundFiles are used by academic music software such as the CSound package, and the MixView sound sample editor. .smp Turtle Beach SampleVision files. SMP files are for use with the PC-DOS package SampleVision by Turtle Beach Softworks. This package is for communication to several MIDI samplers. All sample rates are supported by the package, although not all are supported by the December 10, 1999 6 SoX(1) SoX(1) samplers themselves. Currently loop points are ignored. sunau Sun /dev/audio device driver This is a pseudo-file type and can be optionally compiled into Sox. Run sox -h to see if you have support for this file type. When this driver is used it allows you to open up a Sun /dev/audio file and configure it to use the same data type as passed in to Sox. It works for both playing and recording sound samples. When playing sound files it attempts to set up the audio driver to use the same format as the input file. It is suggested to always override the output values to use the highest quality samples your hardware can handle. Example: -t sunau -w -s /dev/audio or -t sunau -U -c 1 /dev/audio for older sun equipment. .txw Yamaha TX-16W sampler. A file format from a Yamaha sampling keyboard which wrote IBM-PC format 3.5" floppies. Han- dles reading of files which do not have the sam- ple rate field set to one of the expected by looking at some other bytes in the attack/loop length fields, and defaulting to 33kHz if the sample rate is still unknown. .vms More info to come. Used to compress speech audio for applications such as voice mail. .voc Sound Blaster VOC files. VOC files are multi-part and contain silence parts, looping, and different sample rates for different chunks. On input, the silence parts are filled out, loops are rejected, and sample data with a new sample rate is rejected. Silence with a different sample rate is gener- ated appropriately. On output, silence is not detected, nor are impossible sample rates. .wav Microsoft .WAV RIFF files. These appear to be very similar to IFF files, but not the same. They are the native sound file format of Windows. (Obviously, Windows was of such incredible importance to the computer industry that it just had to have its own sound file format.) Normally .wav files have all for- matting information in their headers, and so do not need any format options specified for an input file. If any are, they will override the file header, and you will be warned to this effect. You had better know what you are doing! December 10, 1999 7 SoX(1) SoX(1) Output format options will cause a format con- version, and the .wav will written appropri- ately. Sox currently can read PCM, ULAW, ALAW, MS ADPCM, and IMA (or DVI) ADPCM. It can write all of these formats including (NEW!) the ADPCM encoding. .wve Psion 8-bit alaw These are 8-bit a-law 8khz sound files used on the Psion palmtop portable computer. .raw Raw files (no header). The sample rate, size (byte, word, etc), and encoding (signed, unsigned, etc.) of the sample file must be given. The number of channels defaults to 1. .ub, .sb, .uw, .sw, .ul, .sl These are several suffices which serve as a shorthand for raw files with a given size and encoding. Thus, ub, sb, uw, sw, ul and sl cor- respond to "unsigned byte", "signed byte", "unsigned word", "signed word", "ulaw" (byte), and "signed long". The sample rate defaults to 8000 hz if not explicitly set, and the number of channels (as always) defaults to 1. There are lots of Sparc samples floating around in u-law format with no header and fixed at a sample rate of 8000 hz. (Certain sound management software cheerfully ignores the headers.) Similarly, most Mac sound files are in unsigned byte format with a sample rate of 11025 or 22050 hz. .auto This is a ``meta-type'': specifying this type for an input file triggers some code that tries to guess the real type by looking for magic words in the header. If the type can't be guessed, the program exits with an error mes- sage. The input must be a plain file, not a pipe. This type can't be used for output files. EFFECTS Only one effect from the palette may be applied to a sound sample. To do multiple effects you'll need to run sox in a pipeline. avg [ -l | -r ] Reduce the number of channels by averaging the samples, or duplicate channels to increase the number of channels. This effect is automati- cally 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 avg effect and use the December 10, 1999 8 SoX(1) SoX(1) -l and -r options to select only the left or right channel for the output instead of averag- ing the two channels. band [ -n ] center [ width ] Apply a band-pass filter. The frequency response drops logarithmically around the center frequency. The width gives the slope of the drop. The frequencies at center + width and center - width will be half of their original amplitudes. Band defaults to a mode oriented to pitched signals, i.e. voice, singing, or instru- mental music. The -n (for noise) option uses the alternate mode for un-pitched signals. Warning: -n introduces a power-gain of about 11dB in the filter, so beware of output clip- ping. Band introduces noise in the shape of the filter, i.e. peaking at the center frequency and settling around it. See filter for a bandpass effect with steeper shoulders. bandpass frequency bandwidth Butterworth bandpass filter. Description coming soon! bandreject frequency bandwidth Butterworth bandreject filter. Description com- ing soon! chorus gain-in gain-out delay decay speed depth -s | -t [ delay decay speed depth -s | -t ... ] Add a chorus to a sound sample. Each quadtuple delay/decay/speed/depth gives the delay in mil- liseconds and the decay (relative to gain-in) with a modulation speed in Hz using depth in milliseconds. The modulation is either sinodial (-s) or triangular (-t). Gain-out is the volume of the output. compand attack1,decay1[,attack2,decay2...] in-dB1,out-dB1[,in-dB2,out-dB2...] [gain] [initial-volume] Compand (compress or expand) the dynamic range of a sample. The attack and decay time specify the integration time over which the absolute value of the input signal is integrated to determine its volume. Where more than one pair of attack/decay parameters are specified, each channel is treated separately and the number of pairs must agree with the number of input chan- nels. The second parameter is a list of points December 10, 1999 9 SoX(1) SoX(1) on the compander's transfer function specified in dB relative to the maximum possible signal amplitude. The input values must be in a strictly increasing order but the transfer func- tion does not have to be monotonically rising. The special value -inf may be used to indicate that the input volume should be associated out- put volume. The points -inf,-inf and 0,0 are assumed; the latter may be overridden, but the former may not. The third (optional) parameter is a postprocessing gain in dB which is applied after the compression has taken place; the fourth (optional) parameter is an initial volume to be assumed for each channel when the effect starts. This permits the user to supply a nomi- nal level initially, so that, for example, a very large gain is not applied to initial signal levels before the companding action has begun to operate: it is quite probable that in such an event, the output would be severely clipped while the compander gain properly adjusts itself. copy Copy the input file to the output file. This is the default effect if both files have the same sampling rate. cut loopnumber Extract loop #N from a sample. deemph Apply a treble attenuation shelving filter to samples in audio cd format. The frequency response of pre-emphasized recordings is recti- fied. The filtering is defined in the standard document ISO 908. echo gain-in gain-out delay decay [ delay decay ... ] Add echoing to a sound sample. Each delay/decay part gives the delay in milliseconds and the decay (relative to gain-in) of that echo. Gain- out is the volume of the output. echos gain-in gain-out delay decay [ delay decay ... ] Add a sequence of echos to a sound sample. Each delay/decay part gives the delay in milliseconds and the decay (relative to gain-in) of that echo. Gain-out is the volume of the output. filter [ low ]-[ high ] [ window-len [ beta ] ] Apply a Sinc-windowed lowpass, highpass, or bandpass filter of given window length to the signal. low refers to the frequency of the lower 6dB corner of the filter. high refers to the frequency of the upper 6dB corner of the December 10, 1999 10 SoX(1) SoX(1) filter. A lowpass filter is obtained by leaving low unspecified, or 0. A highpass 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 cutoff, smaller windows a more gradual cutoff. The beta, if unspecified, defaults to 16. This selects a Kaiser window. You can select a Nut- tall window by specifying anything <= 2.0 here. For more discussion of beta, look under the resample effect. flanger gain-in gain-out delay decay speed -s | -t Add a flanger to a sound sample. Each triple delay/decay/speed gives the delay in millisec- onds and the decay (relative to gain-in) with a modulation speed in Hz. The modulation is either sinodial (-s) or triangular (-t). Gain- out is the volume of the output. highp center Apply a high-pass filter. The frequency response drops logarithmically with center fre- quency in the middle of the drop. The slope of the filter is quite gentle. See filter for a highpass effect with sharper cutoff. highpass frequency Butterworth highpass filter. Description com- ming soon! lowp center Apply a low-pass filter. The frequency response drops logarithmically with center frequency in the middle of the drop. The slope of the filter is quite gentle. See filter for a lowpass effect with sharper cutoff. lowpass frequency Butterworth lowpass filter. Description coming soon! map Display a list of loops in a sample, and miscel- laneous loop info. mask Add "masking noise" to signal. This effect deliberately adds white noise to a sound in order to mask quantization effects, created by December 10, 1999 11 SoX(1) SoX(1) the process of playing a sound digitally. It tends to mask buzzing voices, for example. It adds 1/2 bit of noise to the sound file at the output bit depth. pan direction Pan the sound of an audio file from one channel to another. This is done by changing the volume of the input channels so that it fade's out on one channel and fades-in on another. If the number of input channels is different then the number of output channels then this effect tries to intellegently handle this. For instance, if the input contains 1 channel and the output con- tains 2 channels, then it will create the miss- ing channel itself. The direction is a value from -1.0 to 1.0. -1.0 represents far left and 1.0 represents far right. Numbers in between will start the pan effect without totally muting the opposite channel. phaser gain-in gain-out delay decay speed -s | -t Add a phaser to a sound sample. Each triple delay/decay/speed gives the delay in millisec- onds and the decay (relative to gain-in) with a modulation speed in Hz. The modulation is either sinodial (-s) or triangular (-t). The decay should be less than 0.5 to avoid feedback. Gain-out is the volume of the output. pick Select the left or right channel of a stereo sample, or one of four channels in a quadro- phonic sample. pitch shift [ width interpole fade ] Change the pitch of file without affecting its duration by cross-fading shifted samples. shift is given in cents. Use a positive value to shift to treble, negative value to shift to bass. Default shift is 0. width of window is in ms. Default width is 20ms. Try 30ms to lower pitch, and 10ms to raise pitch. interpole option, can be "cubic" or "linear". Default is "cubic". The fade option, can be "cos", "hamming", "linear" or "trapezoid". Default is "cos". polyphase [ -w < nut / ham > ] [ -width < long / short / # > ] [ -cutoff # ] Translate input sampling rate to output sampling rate via polyphase interpolation, a DSP algo- rithm. This method is slow and uses lots of December 10, 1999 12 SoX(1) SoX(1) RAM, but gives much better results than rate. -w < nut / ham > : select either a Nuttal (~90 dB stopband) or Hamming (~43 dB stopband) win- dow. Default is nut. -width long / short / # : specify the (approxi- mate) width of the filter. long is 1024 sam- ples; short is 128 samples. Alternatively, an exact number can be used. Default is long. The short option is not recommended, as it produces poor quality results. -cutoff # : specify the filter cutoff frequency in terms of fraction of bandwidth. If upsam- pling, then this is the fraction of the original signal that should go through. If downsampling, this is the fraction of the signal left after downsampling. Default is 0.95. Remember that this is a float. rate Translate input sampling rate to output sampling rate via linear interpolation to the Least Com- mon Multiple of the two sampling rates. This is the default effect if the two files have differ- ent sampling rates and the preview options was specified. This is fast but noisy: the spectrum of the original sound will be shifted upwards and duplicated faintly when up-translating by a multiple. Lerp-ing is acceptable for cheap 8-bit sound hardware, but for CD-quality sound you should instead use either resample or polyphase. If you are wondering which of SoX's rate changing effects to use, you will want to read a detailed analysis of all of them at http://eakaw2.et.tu-dresden.de/~wilde/resam- ple/resample.html [Nov,1999: These tests need to be updated for sox-12.17, which has bugfixes to the resample and polyphase code.] resample [ -qs | -q | -ql ] [ rolloff [ beta ] ] Translate input sampling rate to output sampling rate via simulated analog filtration. This method is slower than rate, but gives much bet- ter results. The -qs, -q, or -ql options specify increased accuracy at the cost of lower execution speed. By default, linear interpolation is used, with a window width about 45 samples at the lower rate. This gives an accuracy of about 16 bits, but insufficient stopband rejection in the case that you want to have rolloff greater than about 0.80 of the Nyquist frequency. The -q* options use quadratic interpolation of filter coefficients, resulting in about 24 bits precision. December 10, 1999 13 SoX(1) SoX(1) Following is a table of the reasonable defaults which are built-in to sox: Option Window rolloff beta interpolation ------ ------ ------- ---- ------------- (none) 45 0.80 16 linear -qs 45 0.80 16 quadratic -q 75 0.875 16 quadratic -ql 149 0.94 16 quadratic ------ ------ ------- ---- ------------- -qs, -q, or -ql use window lengths of 45, 75, or 149 samples, respectively, at the lower sample- rate of the two files. This means progressively sharper stop-band rejection, at 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 practice 0.8-0.95. The defaults are indicated above. The beta parameter determines the type of filter window used. Any value greater than 2.0 is the beta for a Kaiser window. Beta <= 2.0 selects a Nuttall window. If unspecified, the default is a Kaiser window with beta 16. In the case of Kaiser window (beta > 2.0), lower betas produce a somewhat faster transition from passband to stopband, at the cost of noticeable artifacts. A beta of 16 is the default, beta less than 10 is not recommended. If you want a sharper cutoff, don't use low beta's, use a longer sample window. A Nuttall window is selected by specifying any 'beta' <= 2, and the Nuttall window has somewhat steeper cutoff than the default Kaiser window. You will probably not need to use the beta parameter at all, unless you are just curious about comparing the effects of Nuttall vs. Kaiser windows. This is the default effect if the two files have different sampling rates. Default parameters are, as indicated above, Kaiser window of length 45, rolloff 0.80, beta 16, linear interpolation. NOTE: -qs is only slightly slower, but more accurate for 16-bit or higher precision. NOTE: In many cases of up-sampling, no interpo- lation is needed, as exact filter coefficients can be computed in a reasonable amount of space. To be precise, this is done when December 10, 1999 14 SoX(1) SoX(1) input_rate < output_rate && output_rate/gcd(input_rate,output_rate) <= 511 reverb gain-out delay [ delay ... ] Add reverberation to a sound sample. Each delay is given in milliseconds and its feedback is depending on the reverb-time in milliseconds. Each delay should be in the range of half to quarter of reverb-time to get a realistic rever- beration. Gain-out is the volume of the output. reverse Reverse the sound sample completely. Included for finding Satanic subliminals. speed factor Speed up or down the sound, as a magnetic tape with a speed control. It affects both pitch and time. A factor of 1.0 means no change, and is the default. 2.0 doubles speed, thus time length is cut by a half and pitch is one octave higher. 0.5 halves speed thus time length dou- bles and pitch is one octave lower. split Turn a mono sample into a stereo sample by copy- ing the input channel to the left and right channels. stat [ debug | -v ] Do a statistical check on the input file, and print results on the standard error file. stat may copy the file untouched from input to out- put, if you select an output file. The "Volume Adjustment:" field in the statistics gives you the argument to the -v number which will make the sample as loud as possible without clipping. There is an optional parameter -v that will print out the "Volume Adjustment:" field's value and return. This could be of use in scripts to auto convert the volume. There is an also an optional parameter debug that will place sox into debug mode and print out a hex dump of the sound file from the internal buffer that is in 32-bit signed PCM data. This is mainly only of use in tracking down endian problems that creep in to sox on cross-platform versions. stretch factor [window fade shift fading] Time stretch file by a given factor. Change duration without affecting the pitch. factor of stretching: >1.0 lengthen, <1.0 shorten dura- tion. window size is in ms. Default is 20ms. The fade option, can be "lin". shift ratio, in [0.0 1.0]. Default depends on stretch factor. December 10, 1999 15 SoX(1) SoX(1) 1.0 to shorten, 0.8 to lengthen. The fading ratio, in [0.0 0.5]. The amount of a fade's default depends on factor and shift. swap [ 1 2 3 4 ] Swap channels in multi-channel sound files. In files with more than 2 channels you may specify the order that the channels should be rearranged in. vibro speed [ depth ] Add the world-famous Fender Vibro-Champ sound effect to a sound sample by using a sine wave as the volume knob. Speed gives the Hertz value of the wave. This must be under 30. Depth gives the amount the volume is cut into by the sine wave, ranging 0.0 to 1.0 and defaulting to 0.5. vol gain [ type ] The vol effect is much like the command line option -v. It allows you to adjust the volume of an input file and allows you to specify the adjustment in relation to amplitude, power, or dB. When type is amplitude then a linear change of the amplitude is performed based on the gain. Therefore, a value of 1.0 will keep the volume the same, 0.0 to < 1.0 will cause the volume to decrease and values of > 1.0 will cause the vol- ume to increase. Beware of clipping audio data when the gain is greater then 1.0. A negative value performs the same adjustment while also changing the phase. When type is power then a value of 1.0 also means no change in volume. When type is dB the amplitude is change loga- rithmically. 0.0 is constant while +6 doubles the amplitude. Sox enforces certain effects. If the two files have dif- ferent sampling rates, the requested effect must be one of copy, or rate, If the two files have different numbers of channels, the avg effect must be requested. BUGS The syntax is horrific. Thats the breaks when trying to handle all things from the command line. Please report any bugs found in this version of sox to Chris Bagwell (cbagwell@sprynet.com) FILES SEE ALSO play(1), rec(1), soxexam(1) December 10, 1999 16 SoX(1) SoX(1) NOTICES The version of Sox that accompanies this manual page is support by Chris Bagwell (cbagwell@sprynet.com). Please refer any questions regarding it to this address. You may obtain the latest version at the the web site http://home.sprynet.com/~cbagwell/sox.html December 10, 1999 17