ref: 063fd456b2db1518c3bd58bd378a8dedb8bc0a57
dir: /sox.1/
.de Sh .br .ne 5 .PP \fB\\$1\fR .PP .. .de Sp .if t .sp .5v .if n .sp .. .TH SoX 1 "November 14, 2006" "sox" "Sound eXchange" .SH NAME sox \- Sound eXchange : universal sound sample translator .SH SYNOPSIS .P \fBsox\fR \fIinfile1\fR [ \fIinfile2\fR ... ] \fIoutfile\fR .P \fBsox\fR [ \fIglobal options\fR ] [ \fIformat options\fR ] \fIinfile1\fR .br [ [ \fIformat options\fR ] \fIinfile2\fR ... ] [ \fIformat options\fR ] \fIoutfile\fR .br [ \fIeffect\fR [ \fIeffect options\fR ] ... ] .P \fBsoxmix\fR \fIinfile1 infile2\fR [ \fIinfile3\fR ... ] outfile\fR .P \fBsoxmix\fR [ \fIglobal options\fR ] [ \fIformat options\fR ] \fIinfile1\fR .br [ \fIformat options\fR ] \fIinfile2\fR .br [ [ \fIformat options\fR ] \fIinfile3\fR ... ] .br [ \fIformat options\fR ] \fIoutfile\fR .br [ \fIeffect\fR [ \fIeffect options\fR ] ... ] .SH DESCRIPTION The .I SoX audio file transformer can read and write most popular audio formats and optionally apply effects to them; it includes a simple audio synthesiser, and on unix-like systems, can also play and record sound files. .P Multiple input files can be combined to form the output file using one of three methods: `concatenate', `mix', or `merge'. There is currently the restriction that multiple input files must have the same number of channels and the same sample rate (though not necessarily the same file format). .P The default combining method for \fBsox\fR is `concatenate'; \fBsoxmix\fR is an alias for \fBsox\fR for which the the default combining method is `mix'. .P Exit status is 0 for no error, 1 if there is a problem with the command-line arguments, and 2 if an error occurs during file processing. .P \fBFile Format Types\fR .br There are two types of audio file format that .I SoX can work with. The first is "self-describing". Such formats include a header that completely describe the characteristics of the audio data that follows. The header may also allow the inclusion of textual "comments" that can be used to describe the audio in some way, e.g. for music, the title, the author, etc. .P The second type is headerless data, often called raw data. For this type, a user must pass enough information to .I SoX on the command line so that it knows what type of data it contains, though in some cases, the audio filename extension may imply some of this information. .P Audio data can usually be totally described by four characteristics: .TP 10 rate The sample rate is in samples per second. For example, CDs use 44,100 samples per second. .TP 10 data size The precision the data is stored in. Most popular are 8-bit bytes or 16-bit words. .TP 10 data encoding What encoding the data type uses. Examples are u-law, ADPCM, or signed linear data. .TP 10 channels The number of audio channels contained in the file. 1 ("mono") and 2 ("stereo") are widely used. .P Refer to the .B soxexam(1) manual page for a long description with examples on how to use SoX with various file formats. .P \fBFormat Conversion\fR .br Converting an audio file from one format to another with .I SoX is "lossless" (i.e. converting back again would yield an exact copy of the original audio signal) where it can be, i.e. when not using "lossy" compression (e.g. A-law, MP3, etc.) and 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 a 24-bit PCM format to a 16-bit PCM format isn't. When performing a lossy conversion, .I SoX uses rounding to retain as much accuracy as possible in the audio signal. .P \fBClipping\fR .br Clipping is distortion that occurs when an audio signal level exceeds the range of the chosen representation. Clipping is nearly always undesirable and so should usually be corrected by adjusting the audio volume prior to the point at which clipping occurs. In \fISoX\fR, clipping could occur, as you might expect, when using the .I 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 reampling, 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 intially too close to the clipping point. For these reasons, it is usual to make sure that a digital 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. \fISoX\fR's .I stat effect can assist here by displaying the signal level in an audio file. If clipping occurs at any point during processing, then .I SoX will display a warning message to that effect. .SH OPTIONS The option syntax is somewhat complex, but in essence: .P .br sox file.au file.wav .P .br translates a sound file in SUN Sparc .AU format into a Microsoft .WAV file, while .P .br sox file.au -r 12000 -1 file.wav vol 0.5 dither .P .br does the same format translation but also changes the sampling rate to 12000 Hz, the sample size to 1 byte (8 bits), and applies the \fBvol\fR and \fBdither\fR sound effects to the audio data; .P .br sox short.au long.au longer.au .P .br concatenates two sound files to produce a single file, whilst .P .br sox -m music.mp3 voice.wav mixed.flac .P .br mixes together two sound files. .PP \fBSpecial Filenames\fR .TP 10 \fB-\fR SoX can be used in pipeline operations by using the special filename "-" which, if used in place of input filename, will cause .I SoX will read data from stdin, and which, if used in place of output filename, will cause .I SoX will send data to stdout. .TP 10 \fB-n\fR This can be used in place of an input or output filename to specify that the "null" file type should be used. See .B null below for further information. .TP 10 \fB-e\fR This is just an alias of .I -n but is left here for historical reasons. .PP \fBGlobal Options\fR .TP 10 \fB\-h\fR, \fB\-\-help\fR Print version number and usage information. .TP 10 \fB--help-effect=name\fR Display usage information on the specified effect. The name \fBall\fR can be used to display usage on all effects. .TP 10 \fB\-m\fR, \fB\-\-mix\fR Set the input file combining method to `mix'. Two or more input files must be given, and will be mixed together (instead of concatenated) to form the output file. See also \fBInput File Balancing\fR below. .TP 10 \fB\-M\fR, \fB\-\-merge\fR Set the input file combining method to `merge'. Two or more input files must be given, and will be merged together (instead of concatenated) to form the output file. This can be used for example to merge two mono files into one stereo file; the first and second mono files become the left and right channels of the stereo file. .TP 10 \fB-o\fR Run in a mode that can be used, in conjunction with the GNU Octave program, to assist with the selection and configuration of many of the filtering effects. For the first given effect that supports the \fI-o\fR option, SoX will output Octave commands to plot the effect's transfer function, and then exit without actually processing any audio. E.g. sox -o input-file -n highpass 1320 > plot.m .br octave plot.m .TP 10 \fB-q\fR Run in quiet mode when SoX wouldn't otherwise do so. Inverse of \fB-S\fR option. .TP \fB-S\fR Display status while processing audio data. Shows how much of audio data has been processed in terms of audio running time instead of samples. .TP 10 \fB--version\fR Print version number and exit. .IP "\fB\-V[level]\fP" Set verbosity. .I SoX prints messages to the console (stderr) according to the following verbosity levels: .IP .RS .IP 0 No messages are printed at all; use the exit status to determine if an error has ocurred. .IP 1 Only error messages are printed. These are generated if .I SoX cannot complete the requested commands. .IP 2 Warning messages are also printed. These are generated if .I SoX can complete the requested commands, but not exactly according to the requested command parameters, or if clipping occurs. .IP 3 Descriptions of .I SoX's processing phases are also printed. Useful for figuring out exactly how .I SoX is mangling your sound samples. .IP "4 and above" Messages to help with debugging .I SoX are also printed. .RE .IP By default, the verbosity level is set to 2. Each occurrence of the \fI-V\fR option increases the verbosity level by 1. Alternatively, the verbosity level can be set to an absolute number by specifying it immediately after the .I -V e.g. .I -V0 sets it to 0. .IP .PP \fBInput File Balancing\fR .br When multiple input files are given, \fISoX\fR applies any specified effects (including, for example, volume adjustment) after the audio has been combined. However, as with a traditional audio mixer, it is useful to be able to set the volume of (i.e. `balance') the inputs individually, before combining takes place. If the selected combining method is `mix' then, to guarantee that clipping does not occur at the mixing stage, \fISoX\fR defaults to adjusting the amplitude of each input signal by a factor of 1/n, where n is the number of input files; if this results in audio that is perceived to be too quiet, then the volume adjustments can be set manually instead. For the other combining methods, the default behaviour is for no input volume adjustments. Manual input file volume adjustment is achieved using the following option which, as with format options, 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 (regardless of combining method): .TP 10 \fB-v \fIvolume\fR Adjust volume by a factor of \fIvolume\fR. This is a linear (amplitude) adjustment, so a number less than 1 decreases the volume; greater than 1 increases it. If a negative number is given, then in addition to the volume adjustment, the audio signal will be inverted. See the \fBstat\fR effect for information on how to find the maximum volume of an audio file to help with setting suitable values for this option. .P The \fB-V\fR option will show what input file volume adjustments have been selected (either manually or automatically). .PP \fBInput And Output File Format Options\fR .br These options apply to the input or output file that they immediately precede. .PP Self describing input files can contain all the format information in the header and so don't generally need format options. Headerless input files lack this information and so format options must be used to inform SoX of the file's data type, sample rate, and number of channels. .PP By default, SoX attempts to write audio data using the same data type, sample rate, and channel count as the input data. If the user wants the output file to be of a different format then format options can be used to specify the differences. .PP If an output file format doesn't support the same data type, sample rate, or channel count as the input file format, then SoX will automatically select the closest values it does support so that the user does not have to specify these format change options manually. .TP 10 \fB-c \fIchannels\fR The number of sound channels in the data file. This may be 1, 2, or 4; for mono, stereo, or quad sound data. To cause the output file to have a different number of channels than the input file, include this option with the output file options. If the input and output file have a different number of channels then the avg effect must be used. If the avg effect is not specified on the command line it will be invoked internally with default parameters. .TP 10 \fB-r \fIrate\fR Gives the sample rate in Hz of the file. To cause the output file to have a different sample rate than the input file, include this option as a part of the output format options. .br 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 a default one will be chosen. .TP 10 \fB-t \fIfiletype\fR gives the file type of the sound sample file. Useful when file extension is not standard or can not be determined by looking at the header of the file. The .I -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, .I SoX will exit with an appropriate error message if such a header is not actually present. See the section \fRFILE TYPES\fR for a list of supported file types. .TP 10 \fB-x\fR The sample data 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, 24-bit, and 32-bit integer data may be swapped. Machine-format floating-point data is not portable. .TP 10 \fB-s/-u/-U/-A/-a/-i/-g/-f\fR The sample data encoding is signed linear (2's complement), unsigned linear, u-law (logarithmic), A-law (logarithmic), ADPCM, IMA_ADPCM, GSM, or Floating-point. U-law (actually shorthand for mu-law) and A-law are the U.S. and international standards for logarithmic telephone sound compression. When uncompressed u-law has roughly the precision of 14-bit PCM audio and A-law has roughly the precision of 13-bit PCM audio. A-law and u-law data is sometimes encoded using a reversed bit-ordering (i.e. MSB becomes LSB). Internally, SoX understands how to work with this encoding but there is currently no command line option to specify it. If you need this support then you can use the pseudo file types of ".la" and ".lu" to inform SoX of the encoding. See supported file types for more information. ADPCM is a form of sound compression that has a good compromise between good sound quality and fast encoding/decoding time. It is used for telephone sound compression and places were full fidelity is not as important. When uncompressed it has roughly the precision of 16-bit PCM audio. Popular version of ADPCM include G.726, MS ADPCM, and IMA ADPCM. The \fB-a\fR flag has different meanings in different file handlers. In \fB.wav\fR files it represents MS ADPCM files, in all others it means G.726 ADPCM. IMA ADPCM is a specific form of ADPCM compression, slightly simpler and slightly lower fidelity than Microsoft's flavor of ADPCM. IMA ADPCM is also called DVI ADPCM. GSM is a standard used for telephone sound compression in European countries and it's gaining popularity because of its quality. It usually is CPU intensive to work with GSM audio data. .TP 10 \fB-1/-2/-3/-4/-8\fR The sample data size is 1, 2, 3, 4, or 8 bytes; i.e 8, 16, 24, 32, or 64 bits. .TP 10 \fB-b/-w/-l/-d\fR Aliases for -1/-2/-4/-8. Abbreviations of: byte, word, long word, double long (long long) word. .PP \fBOutput File Format Options\fR .br These options apply to and may precede only the output file. .TP 10 \fB--comment \fItext\fR Specify the comment text to store in the output file header (where applicable). .TP 10 \fB--comment-file \fIfilename\fR Specify a file containing the comment text to store in the output file header (where applicable). .TP 10 \fB-C \fIcompression-factor\fR The compression factor for variably compressing output file formats. If this option is not given, then a default compression factor will apply. The compression factor is interpreted differently for different compressing file formats. See the description of the file formats that use this parameter for more information. .SH FILE TYPES .B Determining The File Type .br .I SoX uses the following method to determine the type of audio to use for each input file and the output file: If .I -n or .I -t has been given, then the associated or given type will be used. (Note that if using stdin or stdout ("\fI-\fR"), then .I -t must be given.) Otherwise, .I SoX will try first using the file header (input files only), and then the filename extension to determine the file type. If the file type cannot be determined, then .I SoX will exit with an error. .P .B Supported File Types .br Note: a file type that can be determined by filename extension is listed with its name preceded by a dot. .PP .TP 10 .B .8svx Amiga 8SVX musical instrument description format. .TP 10 .B .aiff AIFF files used on Apple IIc/IIgs and SGI. Note: the AIFF format supports only one SSND chunk. It does not support multiple sound chunks, or the 8SVX musical instrument description format. AIFF files are multimedia archives and can have multiple audio and picture chunks. You may need a separate archiver to work with them. .TP 10 .B .aifc AIFF-C (not compressed, linear), defined in DAVIC 1.4 Part 9 Annex B. This format is referred from ARIB STD-B24, which is specified for Japanese data broadcasting. Any private chunks are not supported. .br Note: The infile is processed as .aiff currently. .TP 10 .B alsa ALSA default device driver. This is a pseudo-file type and can be optionally compiled into SoX. Run .B sox -h to see if you have support for this file type. When this driver is used it allows you to open up the ALSA /dev/snd/pcmCxDxp file and configure it to use the same data format as passed in to \fBSoX\fR. It works for both playing and recording sound samples. When playing sound files it attempts to set up the ALSA driver to use the same format as the input file. It is suggested to always override the output values to use the highest quality samples your sound card can handle. Example: .I sox infile -t alsa default .TP 10 .B .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 .B .ul format (see below). .br It is possible to override .au file header information with the .B -r and .B -c options, in which case .I SoX will issue a warning to that effect. .TP 10 .B .avr Audio Visual Research. The AVR format is produced by a number of commercial packages on the Mac. .TP 10 .B .cdr CD-R. CD-R files are used in mastering music on Compact Disks. The audio data on a CD-R disk is a raw audio file with a format of stereo 16-bit signed samples at a 44kHz sample rate. There is a special blocking/padding oddity at the end of the audio file, which is why it needs its own handler. .TP 10 .B .cvs Continuously Variable Slope Delta modulation. Used to compress speech audio for applications such as voice mail. .TP 10 .B .dat Text Data files. These files contain a textual representation of the sample data. There is one line at the beginning that contains the sample rate. Subsequent lines contain two numeric data items: the time since the beginning of the first sample and the sample value. Values are normalized so that the maximum and minimum are 1.00 and -1.00. This file format can be used to create data files for external programs such as FFT analysers or graph routines. SoX can also convert a file in this format back into one of the other file formats. .TP 10 .B .flac Free Lossless Audio Codec compressed audio .br FLAC is an open, patent-free CODEC designed for compressing music. It is similar to MP3 and Ogg Vorbis, but lossless, meaning that audio is compressed in FLAC without any loss in quality. .B SoX can decode native FLAC files (.flac) but not Ogg FLAC files (.ogg). [But see .B .ogg below for information relating to support for Ogg Vorbis files.] .B SoX has rudimentary support for writing FLAC files: it can encode to native FLAC using compression levels 0 to 8. 8 is the default compression level and gives the best (but slowest) compression; 0 gives the least (but fastest) compression. The compression level can be selected using the .B -C option (see above) with a whole number from 0 to 8. Note that Replay Gain information is not used by .B SoX if present in FLAC input files and is not generated by .B SoX for FLAC output files, however .B SoX will copy input file "comments" (which can be used to hold Replay Gain information) to output files that support comments, so FLAC output files may contain Replay Gain information if some was present in the input file. In this case the Replay Gain information in the output file is likely to be incorrect and so should be recalculated using a tool that supports this (not .B SoX ). .br FLAC support in .B SoX is optional and requires optional FLAC libraries. To see if there is support for FLAC run \fBsox -h\fR and look for it under the list of supported file formats as "flac". .TP 10 .B .gsm GSM 06.10 Lossy Speech Compression. A standard for compressing speech which is used in the Global Standard for Mobile telecommunications (GSM). It's good for its purpose, shrinking audio data size, but it will introduce lots of noise when a given sound sample is encoded and decoded multiple times. This format is used by some voice mail applications. It is rather CPU intensive. .br GSM in .B SoX is optional and requires access to an external GSM library. To see if there is support for gsm run \fBsox -h\fR and look for it under the list of supported file formats. .TP 10 .B .hcom Macintosh HCOM files. These are (apparently) Mac FSSD files with some variant of Huffman compression. The Macintosh has wacky file formats and this format handler apparently doesn't handle all the ones it should. Mac users will need your usual arsenal of file converters to deal with an HCOM file under Unix or DOS. .TP 10 .B .maud An IFF-conformant 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. .TP 10 .B .mp3 MP3 Compressed Audio. MP3 (MPEG Layer 3) is part of the MPEG standards for audio and video compression. It is a lossy compression format that achieves good compression rates with little quality loss. Also see Ogg Vorbis for a similar format. MP3 support in .B SoX is optional and requires access to either or both the external libmad and libmp3lame libraries. To see if there is support for Mp3 run \fBsox -h\fR and look for it under the list of supported file formats as "mp3". .TP 10 .B null Null file type. This is a special file type that can be used when normal file reading or writing is not needed to use a particular effect. It is selected by using the special filename .I -n in place of an input or output filename. Using this file type 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 \fBtrim\fR or \fBsynth\fR). Using this type 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 \fBnoiseprof\fR or \fBstat\fR). The number of channels and the sampling rate associated with a null file are by default 2 and 44.1kHz respectively, but these can be overriden if necessary by using appropriate \fBFormat Options\fR. One other use of the null file type is to use it in conjunction with .I -V to display information from the audio file header without having to read any further into the audio file. E.g. .B sox -V *.wav -n will display header information for each "wav" file in the current directory. .TP 10 .B .ogg Ogg Vorbis Compressed Audio. Ogg Vorbis is a open, patent-free CODEC designed for compressing music and streaming audio. It is a lossy compression format (similar to MP3, VQF & AAC) that achieves good compression rates with a minimum amount of quality loss. Also see MP3 for a similar format. .B SoX can decode all types of Ogg Vorbis files, and can encode at different compression levels/qualities given as a number from -1 (highest compression/lowest quality) to 10 (lowest compression, highest quality). By default the encoding quality level is 3 (which gives an encoded rate of approx. 112kbps), but this can be changed using the .B -C option (see above) with a number from -1 to 10; fractional numbers (e.g. 3.6) are also allowed. Decoding is somewhat CPU intensive and encoding is very CPU intensive. Ogg Vorbis in .B SoX is optional and requires access to external Ogg Vorbis libraries. To see if there is support for Ogg Vorbis run \fBsox -h\fR and look for it under the list of supported file formats as "vorbis". .TP 10 .B ossdsp OSS /dev/dsp device driver. This is a pseudo-file type and can be optionally compiled into SoX. Run .B sox -h to see if you have support for this file type. When this driver is used it allows you to open up the OSS /dev/dsp file and configure it to use the same data format as passed in to \fBSoX\fR. 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: .I sox infile -t ossdsp -w -s /dev/dsp .TP 10 .B .prc Psion Record. Used in some Psion devices for System alarms and recordings made by the built-in Record application. This format is newer then the .wve format that is used in some Psion devices. .TP 10 .B .sf IRCAM Sound Files. Sound Files are used by academic music software such as the CSound package, and the MixView sound sample editor. .TP 10 .B .sph SPHERE (SPeech HEader Resources) is a file format defined by NIST (National Institute of Standards and Technology) and is used with speech audio. SoX can read these files when they contain u-law and PCM data. It will ignore any header information that says the data is compressed using \fIshorten\fR compression and will treat the data as either u-law or PCM. This will allow SoX and the command line \fIshorten\fR program to be ran together using pipes to encompasses the data and then pass the result to SoX for processing. .TP 10 .B .smp Turtle Beach SampleVision files. SMP files are for use with the PC-DOS package SampleVision by Turtle Beach Softworks. This package is for communication to several MIDI samplers. All sample rates are supported by the package, although not all are supported by the samplers themselves. Currently loop points are ignored. .TP 10 .B .snd Under DOS this file format is the same as the \fB.sndt\fR format. Under all other platforms it is the same as the \fB.au\fR format. .TP 10 .B .sndt SoundTool files. This is an older DOS file format. .TP 10 .B sunau Sun /dev/audio device driver. This is a pseudo-file type and can be optionally compiled into SoX. Run .B 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 .B 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: .I sox infile -t sunau -w -s /dev/audio or .I sox infile -t sunau -U -c 1 /dev/audio for older sun equipment. .TP 10 .B .txw Yamaha TX-16W sampler. A file format from a Yamaha sampling keyboard which wrote IBM-PC format 3.5\" floppies. Handles reading of files which do not have the sample rate field set to one of the expected by looking at some other bytes in the attack/loop length fields, and defaulting to 33kHz if the sample rate is still unknown. .TP 10 .B .vms (More info to come.) Used to compress speech audio for applications such as voice mail. .TP 10 .B .voc Sound Blaster VOC files. VOC files are multi-part and contain silence parts, looping, and different sample rates for different chunks. On input, the silence parts are filled out, loops are rejected, and sample data with a new sample rate is rejected. Silence with a different sample rate is generated appropriately. On output, silence is not detected, nor are impossible sample rates. Note, this version now supports playing VOC files with multiple blocks and supports playing files containing u-law and A-law samples. .TP 10 .B vorbis See .B .ogg format. .TP 10 .B .vox A headerless file of Dialogic/OKI ADPCM audio data commonly comes with the extension .vox. This ADPCM data has 12-bit precision packed into only 4-bits. .TP 10 .B .wav Microsoft .WAV RIFF files. This is the native sound file format of Windows, and widely used for uncompressed sound. Normally \fB.wav\fR files have all formatting information in their headers, and so do not need any format options specified for an input file. If any are, they will override the file header, and you will be warned to this effect. You had better know what you are doing! Output format options will cause a format conversion, and the \fB.wav\fR will written appropriately. SoX currently can read PCM, ULAW, ALAW, MS ADPCM, and IMA (or DVI) ADPCM. It can write all of these formats including the ADPCM encoding. Big endian versions of RIFF files, called RIFX, can also be read and written. To write a RIFX file, use the .I -x option with the output file options. .TP 10 .B .wve Psion 8-bit A-law. Used on older Psion PDAs. .TP 10 .B .xa Maxis XA files .br These are 16-bit ADPCM sound files used by Maxis games. Writing .xa files is currently not supported, although adding write support should not be very difficult. .TP 10 .B .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. .TP 10 .B ".ub, .sb, .uw, .sw, .ul, .al, .lu, .la, .sl" These are several suffices which serve as a shorthand for raw files with a given size and encoding. Thus, \fBub, sb, uw, sw, ul, al, lu, la\fR and \fBsl\fR correspond to "unsigned byte", "signed byte", "unsigned word", "signed word", "u-law" (byte), "A-law" (byte), inverse bit order "u-law", inverse bit order "A-law", and "signed long". The sample rate defaults to 8000 Hz if not explicitly set, and the number of channels defaults to 1. There are lots of Sparc samples floating around in u-law format with no header and fixed at a sample rate of 8000 Hz. (Certain sound management software cheerfully ignores the headers.) Similarly, most Mac sound files are in unsigned byte format with a sample rate of 11025 or 22050 Hz. .SH EFFECTS Multiple effects may be applied to the audio data by specifying them one after another at the end of the command line. .TP 10 avg [ \fI-l\fR | \fI-r\fR | \fI-f\fR | \fI-b\fR | \fI-1\fR | \fI-2\fR | \fI-3\fR | \fI-4\fR | \fIn,n,...,n\fR ] Reduce the number of channels by averaging the samples, or duplicate channels to increase the number of channels. This effect is automatically used when the number of input 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 \fI-l\fR, \fI-r\fR, \fI-f\fR, \fI-b\fR, \fI-1\fR, \fI-2\fR, \fI-3\fR, \fI-4\fR, options to select only the left, right, front, back channel(s) or specific channel for the output instead of averaging the channels. The \fI-l\fR, and \fI-r\fR options will do averaging in quad-channel files so select the exact channel to prevent this. The avg effect can also be invoked with up to 16 double-precision numbers, separated by commas, which specify the proportion (0.0 = 0% and 1.0 = 100%) of each input channel that is to be mixed into each output channel. In two-channel mode, 4 numbers are given: l->l, l->r, r->l, and r->r, respectively. In four-channel mode, the first 4 numbers give the proportions for the left-front output channel, as follows: lf->lf, rf->lf, lb->lf, and rb->rf. The next 4 give the right-front output in the same order, then left-back and right-back. It is also possible to use the 16 numbers to expand or reduce the channel count; just specify 0 for unused channels. Finally, certain reduced combination of numbers can be specified for certain input/output channel combinations. In Ch Out Ch Num Mappings .br _____ ______ ___ _____________________________ .b4 2 1 2 l->l, r->l .br 2 2 1 adjust balance .br 4 1 4 lf->l, rf->l, lb->l, rb-l .br 4 2 2 lf->l&rf->r, lb->l&rb->r .br 4 4 1 adjust balance .br 4 4 2 front balance, back balance .br .TP 10 band \fB[ \fI-n \fB] \fIcenter \fB[ \fIwidth\fB ] Apply a band-pass filter. The frequency response drops logarithmically around the .I center frequency. The .I width gives the slope of the drop. The frequencies at .I "center + width" and .I "center - width" will be half of their original amplitudes. .B Band defaults to a mode oriented to pitched signals, i.e. voice, singing, or instrumental music. The .I -n (for noise) option uses the alternate mode for un-pitched signals. .B Warning: .I -n introduces a power-gain of about 11dB in the filter, so beware of output clipping. .B Band introduces noise in the shape of the filter, i.e. peaking at the .I center frequency and settling around it. This effect supports the \fI-o\fR option (see above). See \fBfilter\fR for a bandpass filter with steeper shoulders. .TP 10 bandpass|bandreject \fIfrequency bandwidth\fR Apply a two-pole Butterworth band-pass or band-reject filter with central frequency (in Hz) \fIfrequency\fR, and bandwidth (in Hz, and as determined by the 3dB points) \fIbandwidth\fR. The filter rolls off at 6dB per octave (20dB per decade). These effects support the \fI-o\fR option (see above). .TP 10 bandreject \fIfrequency bandwidth\fR See \fI bandpass\fR. .TP 10 bass|treble \fIgain\fR [\fIfrequency\fR] [\fIslope\fR] Boost or cut the bass (lower) or treble (upper) frequencies of the audio signal using a two-pole shelving filter with (by default) a response similar to that of a standard hi-fi's (Baxandall) tone controls. \fIgain\fR gives the dB gain at 0Hz (for \fIbass\fR), or whichever is the lower of ~22kHz and the Nyquist frequency (for \fItreble\fR). Its useful range is about -20.0 (for a large cut) to +20.0 (for a large boost). N.B. When using a positive \fIgain\fR, in order to prevent clipping, it may be necessary to precede this effect with a suitable attenuation using the \fI-v\fR option or the \fIvol\fR effect. SoX will display a warning message should clipping occur. If desired, the filter can be fine-tuned using the following optional parameters (in either order): \fIfrequency\fR sets the filter's center frequency and so can be used to extend or reduce the frequency range to be boosted or cut. The default value is 100Hz (for \fIbass\fR) or 3kHz (for \fItreble\fR). \fIslope\fR is a number between 0 and 1 that determines how steep the filter's shelf transition is. Its useful range is about 0.3 (for a gentle slope) to 1 (for a steep slope). The default value is 0.5. The \fIbass\fR and \fItreble\fR effects support the \fI-o\fR option (see above). .TP chorus \fIgain-in gain-out delay decay speed depth .TP 10 -s \fR| \fI-t [ \fIdelay decay speed depth -s \fR| \fI-t ... \fR] Add a chorus to a sound sample. Each four-tuple 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. .TP compand \fIattack1,decay1\fR[,\fIattack2,decay2\fR...] .TP \fIin-dB1,out-dB1\fR[,\fIin-dB2,out-dB2\fR...] .TP 10 [\fIgain\fR [\fIinitial-volume\fR [\fIdelay\fR ] ] ] Compand (compress or expand) the dynamic range of a sample. The attack and decay time specify the integration time over which the absolute value of the input signal is integrated to determine its volume; attacks refer to increases in volume and decays refer to decreases. Where more than one pair of attack/decay parameters are specified, each channel is treated separately and the number of pairs must agree with the number of input channels. The second parameter is a list of points on the compander's transfer function specified in dB relative to the maximum possible signal amplitude. The input values must be in a strictly increasing order but the transfer function does not have to be monotonically rising. The special value \fI-inf\fR may be used to indicate that the input volume should be associated output volume. The points \fI-inf,-inf\fR and \fI0,0\fR are assumed; the latter may be overridden, but the former may not. The third (optional) parameter is a post-processing gain in dB which is applied after the compression has taken place; the fourth (optional) parameter is an initial volume to be assumed for each channel when the effect starts. This 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. 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. .TP 10 copy Copy the input file to the output file. This is the default effect if both files have the same sampling rate. .TP 10 dcshift \fIshift\fR [ \fIlimitergain\fR ] DC Shift the audio data, with basic linear amplitude formula. This is most useful if your audio data tends to not be centered around a value of 0. Shifting it back will allow you to get the most volume adjustments without clipping audio data. The first option is the \fIdcshift\fR value. It is a floating point number that indicates the amount to shift. An option limitergain value can be specified as well. It should have a value much less then 1.0 and is used only on peaks to prevent clipping. .TP 10 deemph Apply a treble attenuation shelving filter to samples in audio CD format. The frequency response of pre-emphasized recordings is rectified. The filtering is defined in the standard document ISO 908. This effect supports the \fI-o\fR option (see above). .TP 10 dither [\fIdepth\fR] Dithering deliberately adds white noise to the signal in order to mask audible quantization effects that can occur if the output sample size is less than 24 bits. By default, the amount of noise added is 1/2 bit; the optional \fIdepth\fR parameter is a (linear or voltage) multiplier of this amount. This effect should not be followed by any other effect that affects the audio. .TP 10 earwax Makes sound easier to listen to on headphones. Adds audio-cues to samples in audio CD format so that when listened to on headphones the stereo image is moved from inside your head (standard for headphones) to outside and in front of the listener (standard for speakers). See http://www.geocities.com/beinges for a full explanation. .TP 10 echo \fIgain-in gain-out delay decay \fR[ \fIdelay decay ... \fR] 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. .TP 10 echos \fIgain-in gain-out delay decay \fR[ \fIdelay decay ... \fR] 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. .TP 10 equalizer \fIcentral\-frequency\fR \fIQ\fR \fIgain\fR Apply an equalizer effect which allows you to modify the amplitude (\fIgain\fR) of a signal at and around (\fIQ\-factor\fR) a central frequency (\fIcentral\-frequency\fR), leaving all other frequencies untouched (unlike regular bandpass/bandreject filters). \fIcentral\-frequency\fR gives the central frequency in Hz, \fIQ\fR is the Q\-factor (see http://en.wikipedia.org/wiki/Q_factor), and \fIgain\fR is the gain or attenuation in dB. This effect supports the \fI-o\fR option (see above). .TP 10 fade [ \fItype\fR ] \fIfade-in-length\fR [ \fIstop-time\fR [ \fIfade-out-length\fR ] ] Add a fade effect to the beginning, end, or both of the audio data. For fade-ins, this starts from the first sample and ramps the volume of the audio from 0 to full volume over \fIfade-in-length\fR seconds. Specify 0 seconds if no fade-in is wanted. For fade-outs, the audio data will be truncated at the stop-time and the volume will be ramped from full volume down to 0 starting at \fIfade-out-length\fR seconds before the \fIstop-time\fR. If fade-out-length is not specified, it defaults to the same value as fade-in-length. No fade-out is performed if the stop-time is not specified. All times can be specified in either periods of time or 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 \fItype\fR 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 a linear slope. .TP 10 filter [ \fIlow\fR ]-[ \fIhigh\fR ] [ \fIwindow-len\fR [ \fIbeta\fR ] ] Apply a Sinc-windowed lowpass, highpass, or bandpass filter of given window length to the signal. \fIlow\fR refers to the frequency of the lower 6dB corner of the filter. \fIhigh\fR refers to the frequency of the upper 6dB corner of the filter. A low-pass filter is obtained by leaving \fIlow\fR unspecified, or 0. A high-pass filter is obtained by leaving \fIhigh\fR unspecified, or 0, or greater than or equal to the Nyquist frequency. The \fIwindow-len\fR, if unspecified, defaults to 128. Longer windows give a sharper cutoff, smaller windows a more gradual cutoff. The \fIbeta\fR, if unspecified, defaults to 16. This selects a Kaiser window. You can select a Nuttall window by specifying anything <= 2.0 here. For more discussion of beta, look under the \fBresample\fR effect. .TP 10 flanger [\fIdelay depth regen width speed shape phase interp\fR] Apply a flanging effect to the signal. All parameters are optional (right to left). PARAM RANGE DEFAULT DESCRIPTION .RS .TP 21 \fIdelay\fR 0 10 0 Base delay in milliseconds. .TP 21 \fIdepth\fR 0 10 2 Added swept delay in milliseconds. .TP 21 \fIregen\fR -95 +95 0 Percentage regeneration (delayed signal feedback). .TP 21 \fIwidth\fR 0 100 71 Percentage of delayed signal mixed with original. .TP 21 \fIspeed\fR 0.1 10 0.5 Sweeps per second (Hz). .TP 21 \fIshape\fR -- sin Swept wave shape: sine | triangle. .TP 21 \fIphase\fR 0 100 25 Swept wave percentage phase-shift for multi-channel (e.g. stereo) flange; 0 = 100 = same phase on each channel. .TP 21 \fIinterp\fR -- lin Delay-line interpolation: linear | quadratic. .RE .TP 10 highp|lowp \fIfrequency\fR Apply a single-pole recursive high-pass or low-pass filter with 3dB point \fIfrequency\fR. The filters roll off at 6dB per octave (20dB per decade). These effects support the \fI-o\fR option (see above). See \fBfilter\fR for filters with a sharper cutoff. .TP 10 highpass|lowpass \fIfrequency\fR Apply a two-pole Butterworth high-pass or low-pass filter with 3dB point \fIfrequency\fR. The filters roll off at 12dB per octave (40dB per decade). These effects support the \fI-o\fR option (see above). .TP 10 lowp \fIfrequency\fR See the description of the \fIhighp\fR effect for details. .TP 10 lowpass \fIfrequency\fB See the description of the \fIhighpass\fR effect for details. .TP 10 mask [\fIdepth\fR] This effect is just an alias of the "dither" effect but is left here for historical reasons. .TP mcompand "\fIattack1,decay1\fR[,\fIattack2,decay2\fR...] .TP \fIin-dB1,out-dB1\fR[,\fIin-dB2,out-dB2\fR...] .TP 10 [\fIgain\fR [\fIinitial-volume\fR [\fIdelay\fR ] ] ]" \fIxover_freq\fR Multi-band compander is similar to the single band compander but the audio file is first divided up into bands and then the compander is run on each band. See the \fBcompand\fR effect for the definition of its options. Compand options are specified between double quotes and the crossover frequency for that band is specified separately with \fIxover_fre\fR. This can be repeated multiple times to create multiple bands. .TP noiseprof [\fIprofile-file\fR] .TP 10 noisered \fIprofile-file\fR [\fIthreshold\fR] Noise reduction filter with profiling. This filter is moderately effective at removing consistent background noise such as hiss or hum. To use it, first run the \fBnoiseprof\fR effect on a section of silence (that is, a section which contains nothing but noise). The \fBnoiseprof\fR effect will print a noise profile to \fIprofile-file\fR, or to stdout if no \fIprofile-file\fR is specified. If there is sound output on stdout then the profile will instead be directed to stderr. To actually remove the noise, run SoX again with the \fInoisered\fR filter. The filter needs one argument, \fIprofile-file\fR, which contains the noise profile from noiseprof. \fIthreshold\fR specifies how much noise should be removed, and may be between 0 and 1 with a default of 0.5. Higher values will remove more noise but present a greater possibility of distorting the desired audio signal. Experiment with different threshold values to find the optimal one for your sample. .TP 10 pan \fIdirection\fB Pan the sound of an audio file from one channel to another. This is done by changing the volume of the input channels so that it fades out on one channel and fades-in on another. If the number of input channels is different then the number of output channels then this effect tries to intelligently handle this. For instance, if the input contains 1 channel and the output contains 2 channels, then it will create the missing channel itself. The .I 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. .TP 10 phaser \fIgain-in gain-out delay decay speed\fR < -s | -t > Add a phaser to a sound sample. Each triple delay/decay/speed gives the delay in milliseconds and the decay (relative to gain-in) with a modulation speed in Hz. The modulation is either sinusoidal (-s) or triangular (-t). The decay should be less than 0.5 to avoid feedback. Gain-out is the volume of the output. .TP 10 pick [ \fI-1\fR | \fI-2\fR | \fI-3\fR | \fI-4\fR | \fI-l\fR | \fI-r\fR | \fI-f\fR | \fI-b\fR ] Pick a subset of channels to be copied into the output file. This effect is just an alias of the "avg" effect but is left here for historical reasons. .TP 10 pitch \fIshift [ width interpole fade ]\fB Change the pitch of file without affecting its duration by cross-fading shifted samples. .I shift is given in cents. Use a positive value to shift to treble, negative value to shift to bass. Default shift is 0. .I width of window is in ms. Default width is 20ms. Try 30ms to lower pitch, and 10ms to raise pitch. .I interpole option, can be "cubic" or "linear". Default is "cubic". The .I fade option, can be "cos", "hamming", "linear" or "trapezoid". Default is "cos". .TP polyphase [ \fI-w \fR< \fInut\fR / \fIham\fR > ] .TP [ \fI -width \fR< \fI long \fR / \fIshort \fR / \fI# \fR> ] .TP 10 [ \fI-cutoff # \fR ] Translate input sampling rate to output sampling rate via polyphase interpolation, a DSP algorithm. This method is relatively slow and memory intensive. .br -w < nut / ham > : select either a Nuttall (~90 dB stopband) or Hamming (~43 dB stopband) window. Default is .I nut. .br -width long / short / # : specify the (approximate) width of the filter. .I long is 1024 samples; .I short is 128 samples. Alternatively, an exact number can be used. Default is .I long. The .I short option is .B not recommended, as it produces poor quality results. .br -cutoff # : specify the filter cutoff frequency in terms of fraction of frequency bandwidth, also know as the Nyquist frequency. See the \fIresample\fR 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. Default is 0.95. Remember that this is a float. .TP 10 rabbit [ \fI-c0\fR | \fI-c1\fR | \fI-c2\fR | \fI-c3\fR | \fI-c4\fR ] Resample using libsamplerate, aka Secret Rabbit Code. This effect is optional and must have been selected at compile time of \fISoX\fR. See http://www.mega-nerd.com/SRC/ for details of the algorithm. Algorithms 0 through 2 are progressively faster and lower quality versions of the sinc algorithm; the default is \fI-c0\fR, which is probably the best quality algorithm for general use currently available in SoX. Algorithm 3 is zero-order hold, and 4 is linear interpolation. See the \fIresample\fR effect for more discussion of resampling. .TP 10 rate Does the same as \fBresample\fR with no arguments; it exists for backwards compatibility. .TP 10 repeat \fIcount\fR Repeats the audio data \fIcount\fR times. Requires disk space to store the data to be repeated. .TP 10 resample [ \fI-qs\fR | \fI-q\fR | \fI-ql\fR ] [ \fIrolloff\fR [ \fIbeta\fR ] ] Translate input sampling rate to output sampling rate via simulated analog filtration. Other rate changing effects available are \fBpolyphase\fR and \fBrabbit\fR. There is a detailed analysis of \fBresample\fR and \fBpolyphase\fR at http://leute.server.de/wilde/resample.html; see \fBrabbit\fR for a pointer to its own documentation. By default, linear interpolation 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 stopband rejection in the case that you want to have rolloff greater than about 0.80 of the Nyquist frequency. The \fI-q*\fR options will change the default values for rolloff and beta as well as use quadratic interpolation of filter coefficients, resulting in about 24 bits precision. The \fI-qs\fR, \fI-q\fR, or \fI-ql\fR options specify increased accuracy at the cost of lower execution speed. It is optional to specify rolloff and beta parameters when using the \fI-q*\fR options. Following is a table of the reasonable defaults which are built-in to SoX: .br \fBOption Window rolloff beta interpolation\fR .br \fB------ ------ ------- ---- -------------\fR .br (none) 45 0.80 16 linear .br -qs 45 0.80 16 quadratic .br -q 75 0.875 16 quadratic .br -ql 149 0.94 16 quadratic .br \fB------ ------ ------- ---- -------------\fR \fI-qs\fR, \fI-q\fR, or \fI-ql\fR 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. \fIrolloff\fR 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 0.0 and 1.0, in practice 0.8-0.95. The defaults are indicated above. The \fINyquist frequency\fR is equal to (sample rate / 2). 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 highpass 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 \fIrolloff\fR refers to how close to the Nyquist frequency this cutoff is, with closer being better. When increasing the sample rate of an audio file you would not expect to have any frequencies exist that are past the original Nyquist frequency. Because of resampling properties, it is common to have aliasing data created that is above the old Nyquist frequency. In that case the \fIrolloff\fR refers to how close to the original Nyquist frequency to use a highpass filter to remove this false data, with closer also being better. The \fIbeta\fR 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. \fBNOTE:\fR \fI-qs\fR is only slightly slower, but more accurate for 16-bit or higher precision. \fBNOTE:\fR In many cases of up-sampling, no interpolation is needed, as exact filter coefficients can be computed in a reasonable amount of space. To be precise, this is done when .br input_rate < output_rate .br && .br output_rate/gcd(input_rate,output_rate) <= 511 .TP 10 reverb \fIgain-out reverb-time delay \fR[ \fIdelay ... \fR] Add reverberation to a sound sample. Each delay is given in milliseconds and its feedback is depending on the reverb-time in milliseconds. Each delay should be in the range of half to quarter of reverb-time to get a realistic reverberation. Gain-out is the volume of the output. .TP 10 reverse Reverse the sound sample completely. Included for finding Satanic subliminals. .TP 10 silence \fIabove_periods\fR [ \fIduration threshold\fR[ \fId\fR | \fI%\fR ] [ \fIbelow_periods duration threshold\fR[ \fId\fR | \fI%\fR ]] Removes silence from the beginning, middle, or end of a sound file. Silence is anything below a specified threshold. The \fIabove_periods\fR value is used to indicate if sound should be trimmed at the beginning of the audio file. A value of zero indicates no silence should be trimmed from the beginning. When specifying an non-zero \fIabove_periods\fR, it trims audio up until it finds non-silence. Normally, when trimming silence from beginning of audio the \fIabove_periods\fR will be 1 but it can be increased to higher values to trim all data up to a specific count of non-silence periods. For example, if you had an audio file with two songs that each contained 2 seconds of silence before the song, you could specify an \fIabove_period\fR of 2 to strip out both silence periods and the first song. When \fIabove_periods\fR is non-zero, you must also specify a \fIduration\fR and \fIthreshold\fR. \fIDuration\fR indications the amount of time that non-silence must be detected before it stops trimming data. By increasing the duration, burst of noise can be treated as silence and trimmed off. \fIThreshold\fR 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 a sound file, you specify a \fIbelow_periods\fR count. In this case, \fIbelow_period\fR means to remove all audio data after silence is detected. Normally, this will be a value 1 of but it can be increased to skip over periods of silence that are wanted. For example, if you have a song with 2 seconds of silence in the middle and 2 second at the end, you could set below_period to a value of 2 to skip over the silence in the middle of the audio file. For \fIbelow_periods\fR, \fIduration\fR specifies a period of silence that must exist before data 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 \fIsilence\fR effect in combination with the \fIreverse\fR effect. By first reversing the audio, you can use the \fIabove_periods\fR 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 \fIbelow_periods\fR 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 \fIabove_periods\fR, making it suitable for removing periods of silence in the middle of the sound file. The \fIperiod\fR counts are in units of samples. \fIDuration\fR counts may be in the format of hh:mm:ss.frac, or the exact count of samples. \fIThreshold\fR 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). .TP 10 speed \fIfactor\fR[\fIc\fR] Adjust the audio speed (pitch and tempo together). \fIfactor\fR is either the ratio of the new speed to the old speed: greater than 1 speeds up, less than 1 slows down, or, if appended with `\fIc\fR', the number of cents (i.e. 100ths of a semitone) by which the pitch (and tempo) should be adjusted: greater than 0 increases, less than 0 decreases. By default, the speed change is performed by the \fBresample\fR effect with its default parameters. For higher quality resampling, in addition to the \fBspeed\fR effect, specify either the \fBresample\fR or the \fBrabbit\fR effect with appropriate parameters. .TP 10 stat [ \fI-s N\fB ] [\fI-rms\fB ] [\fI-freq\fB ] [ \fI-v\fB ] [ \fI-d\fB ] Do a statistical check on the input file, and print results on the standard error file. Audio data is passed unmodified from input to output file unless used along with the .B -n option. The "Volume Adjustment:" field in the statistics gives you the argument to the .B -v .I number which will make the sample as loud as possible without clipping. The option .B -v will print out the "Volume Adjustment:" field's value only and return. This could be of use in scripts to auto convert the volume. The .B -s n option is used to scale the input data by a given factor. The default value of n is the max value of a signed long variable (0x7fffffff). Internal effects always work with signed long PCM data and so the value should relate to this fact. The .B -rms option will convert all output average values to \fIroot mean square\fR format. The .B -freq option calculates the input's power spectrum and prints it to standard error. There is also an optional parameter .B -d that will print out a hex dump of the sound file from the internal buffer that is in 32-bit signed PCM data. This is mainly only of use in tracking down endian problems that creep in to SoX on cross-platform versions. .TP 10 stretch \fIfactor [window fade shift fading]\fB Time stretch file by a given factor. Change duration without affecting the pitch. .I factor of stretching: >1.0 lengthen, <1.0 shorten duration. .I window size is in ms. Default is 20ms. The .I fade option, can be "lin". .I shift ratio, in [0.0 1.0]. Default depends on stretch factor. 1.0 to shorten, 0.8 to lengthen. The .I fading ratio, in [0.0 0.5]. The amount of a fade's default depends on factor and shift. .TP 10 swap [ \fI1 2\fB | \fI1 2 3 4\fB ] Swap channels in multi-channel sound files. Optionally, you may specify the channel order you would like the output in. This defaults to output channel 2 and then 1 for stereo and 2, 1, 4, 3 for quad-channels. An interesting feature is that you may duplicate a given channel by overwriting another. This is done by repeating an output channel on the command line. For example, swap 2 2 will overwrite channel 1 with channel 2's data; creating a stereo file with both channels containing the same audio data. .TP 10 synth [\fIlen\fR] {[\fItype] [combine\fR] [\fIfreq\fR[\fI-freq2\fR]] [\fIoff\fR] [\fIph\fR] [\fIp1\fR] [\fIp2\fR] [\fIp3\fR]} This effect can be used to generate fixed or swept frequency audio tones with various wave shapes, or to generate wideband noise of various "colours". Multiple synth effects can be cascaded to produce more complex waveforms; at each stage it is possible to choose whether the generated waveform will be mixed with, or modulated onto the output from the previous stage. Audio for each channel in a multi-channel sound file can be synthesised independently. Though this effect is used to generate audio data, an input file must still be specified. This can be used to set the synthesised audio length, the number of channels, and the sampling rate, however since the input file's audio data is not needed, the .I null file "\fI-n\fR" is usually used instead (and the length specified as a parameter to \fIsynth\fR). For example, the following produces a 3 second, 44.1kHz, stereo audio file containing a sine-wave swept from 300 to 3300 Hz. sox -n output.au synth 3 sine 300-3300 This produces an 8kHz mono version: sox -r 8000 -c 1 -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 .5 sine 200-500 synth .5 sine fmod 700-100 Frequencies can also specied in terms of musical semitones relative to "middle A" (440Hz); the following could be used to help tune a guitar's "low E" string (on a system that supports \fBalsa\fR): sox -n -t alsa default synth sine %-5 N.B. This effect generates audio at maximum volume, 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 \fBvol\fR effect to select a suitable attenuation. A detailed description of each .I synth parameter follows: \fIlen\fR 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. \fItype\fR is one of sine, square, triangle, sawtooth, trapezium, exp, [white]noise, pinknoise, brownnoise; default=sine \fIcombine\fR is one of create, mix, amod (amplitude modulation), fmod (frequency modulation); default=create \fIfreq\fR/\fIfreq2\fR are the frequencies at the beginning/end of synthesis in Hz or, if prepended with '%', semitones relative to A (440Hz); for both, default=%0. Not used for noise. \fIoff\fR is the bias (DC-offset) of the signal in percent; default=0. \fIph\fR is the phase shift in percentage of 1 cycle; default=0. Not used for noise. \fIp1\fR is the percentage of each cycle that is "on" (square), or "rising" (triangle, exp, trapezium); default=50 (square, triangle, exp), default=10 (trapezium). \fIp2\fR trapezium: the percentage through each cycle at which "falling" begins; default=50. exp: the amplitude in percent; default=100. \fIp3\fR trapezium: the percentage through each cycle at which "falling" ends; default=60. .TP 10 treble \fIgain\fR [\fIfrequency\fR] [\fIslope\fR] See the description of the \fIbass\fR effect for details. .TP 10 trim \fIstart\fR [ \fIlength\fR ] Trim can trim off unwanted audio data from the beginning and end of the audio file. Audio samples are not sent to the output stream until the \fIstart\fR location is reached. The optional \fIlength\fR parameter tells the number of samples to output after the \fIstart\fR sample and is used to trim off the back side of the audio data. Using a value of 0 for the \fIstart\fR 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 1/2 seconds into the audio data. The format for specifying sample counts is the number of samples with the letter 's' appended to it. A value of 8000s will wait until 8000 samples are read before starting to process audio data. .TP 10 vibro \fIspeed \fB [ \fIdepth\fB ] Add the world-famous Fender Vibro-Champ sound effect to a sound sample by using a sine wave as the volume knob. .B Speed gives the Hertz value of the wave. This must be under 30. .B Depth gives the amount the volume is cut into by the sine wave, ranging 0.0 to 1.0 and defaulting to 0.5. .TP 10 vol \fIgain\fR [ \fItype\fB [ \fIlimitergain\fR ] ] The vol effect is much like the command line option -v. It allows you to adjust the volume of an input file and allows you to specify the adjustment in relation to amplitude, power, or dB. If \fItype\fR is not specified then it defaults to \fIamplitude\fR. When type is .I amplitude then a linear change of the amplitude is performed based on the gain. Therefore, a value of 1.0 will keep the volume the same, 0.0 to < 1.0 will cause the volume to decrease and values of > 1.0 will cause the volume to increase. Beware of clipping audio data when the gain is greater then 1.0. A negative value performs the same adjustment while also changing the phase. When type is .I power then a value of 1.0 also means no change in volume. When type is .I dB the amplitude is changed logarithmically. 0.0 is constant while +6 doubles the amplitude. An optional \fIlimitergain\fR value can be specified and should be a value much less then 1.0 (i.e. 0.05 or 0.02) and is used only on peaks to prevent clipping. Not specifying this parameter will cause no limiter to be used. In verbose mode, this effect will display the percentage of audio data that needed to be limited. .SH BUGS Please report any bugs found in this version of SoX to the mailing list (sox-users@lists.sourceforge.net). .SH SEE ALSO .BR play (1), .BR rec (1), .BR soxexam (1) .LP The SoX web page at http://sox.sourceforge.net/ .SH LICENSE Copyright 1991 Lance Norskog and Sundry Contributors. Copyright 1998-2006 by Chris Bagwell and SoX Contributors. .LP 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. .LP 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. .SH AUTHORS Chris Bagwell (cbagwell@users.sourceforge.net). .P Additional authors and contributors are listed in the AUTHORS file that is distributed with the source code.