📄 sox.txt
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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 ALSA /dev/snd/pcmCxDxp file and configure it to use the same data format as passed in to SoX. 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: sox infile -t alsa -w -s /dev/snd/pcmC0D0p .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 head- ers 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 commercial packages on the Mac. .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 and is why 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. 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 ana- lyzers 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 A standard for compressing speech which is used in the Global Standard for Mobil telecommunications (GSM). Its 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. 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 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. .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. .mp3 MP3 Compressed Audio MP3 audio files come from the MPEG standards for audio and video compression. They are a lossy compression format that achieves good compression rates with a minimum amount of quality loss. Also see Ogg Vorbis for a similar format. MP3 support in 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 sox -h and look for it under the list of supported file formats as "mp3". .nul Null file handler. This is a fake file hander that act as if its reading a stream of 0’s from a while or fake writing out- put to a file. This is not a very useful file handler in most cases. It might be useful in some scripts were you do not want to read or write from a real file but would like to specify a filename for consistency. .ogg Ogg Vorbis Compressed Audio. Ogg Vorbis is a open, patent-free CODEC designed for com- pressing music and streaming audio. It is similar to MP3, VQF, AAC, and other lossy formats. SoX can decode all types of Ogg Vorbis files, but can only encode at 128 kbps. Decod- ing is somewhat CPU intensive and encoding is very CPU inten- sive. Ogg Vorbis in SoX is optional and requires access to external Ogg Vorbis libraries. To see if there is support for Ogg Vorbis run sox -h and look for it under the list of supported file formats as "vorbis". 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 format 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 out- put values to use the highest quality samples your sound card can handle. Example: sox infile -t ossdsp -w -s /dev/dsp .prc Psion record.app Used in some Psion devices for System alarms. This format is newer then the .wve format that is used in some Psion devices. .sf IRCAM Sound Files. Sound Files are used by academic music software such as the CSound package, and the MixView sound sample editor. .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 infor- mation that says the data is compressed using shorten com- pression and will treat the data as either u-law or PCM. This will allow SoX and the command line shorten program to be ran together using pipes to uncompress the data and then pass the result to SoX for processing. .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 them- selves. Currently loop points are ignored. .snd Under DOS this file format is the same as the .sndt format. Under all other platforms it is the same as the .au format. .sndt SoundTool files. This is an older DOS file format. 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 out- put values to use the highest quality samples your hardware can handle. Example: sox infile -t sunau -w -s /dev/audio or sox infile -t sunau -U -c 1 /dev/audio for older sun equip- ment. .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. .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 generated appropriately. On out- put, 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. vorbis See .ogg format. 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. .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 formatting information in their headers, and so do not need any format options spec- ified 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 .wav will written appro- priately. 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 A-law 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, .al, .lu, .la, .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, al, lu, la and sl 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. .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 message. The input must be a plain file, not a pipe. This type can’t be used for output files.EFFECTS Multiple effects may be applied to the audio data by specifying them one after another at the end of the command line. avg [ -l | -r | -f | -b | -1 | -2 | -3 | -4 | n,n,...,n ] 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 chan- nels differ from the number of output channels. When reduc- ing the number of channels it is possible to manually specify the avg effect and use the -l, -r, -f, -b, -1, -2, -3, -4, options to select only the left, right, front, back chan- nel(s) or specific channel for the output instead of averag- ing the channels. The -l, and -r 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-pre- cision numbers, seperated by commas, which specify the pro- portion (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 fol- lows: 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 speci- fied for certain input/output channel combinations. In Ch Out Ch Num Mappings _____ ______ ___ _____________________________ 2 1 2 l->l, r->l 2 2 1 adjust balance 4 1 4 lf->l, rf->l, lb->l, rb-l 4 2 2 lf->l&rf->r, lb->l&rb->r 4 4 1 adjust balance 4 4 2 front balance, back balance band [ -n ] center [ width ] Apply a band-pass filter. The frequency response drops loga- rithmically 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 instrumental music. The -n (for noise) option uses the alternate mode for un-pitched signals. Warn- ing: -n introduces a power-gain of about 11dB in the filter, so beware of output clipping. Band introduces noise in the shape of the filter, i.e. peaking at the center frequency and settling around it. See filter for a bandpass effect with steeper shoulders.
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