NAME

	mp3dec, mp3enc, oggdec, oggenc, flacdec, flacenc, sundec, wavdec, pcmconv, mixfs – decode and encode audio files

SYNOPSIS

audio/mp3dec [ –s seconds ] [ –d ] audio/oggdec [ –s seconds ] audio/flacdec [ –s seconds ] audio/wavdec [ –s seconds ] audio/sundec audio/oggenc audio/mp3enc [ –hprv ] [ –b bitrate ] [ –B bitrate ] [ –m mode ] [ –q q ] [ –s sfreq ] [ –V q ] [ long or silly options ] audio/flacenc [ –i fmt ] [ –l compresslevel ] [ –P padding ] [ –T field=value ] audio/pcmconv [ –i fmt ] [ –o fmt ] [ –l length ] audio/mixfs [ –D ] [ –s srvname ] [ –m mtpt ] [ /dev/audio ]

DESCRIPTION

These programs decode and encode various audio formats from and to 16–bit stereo PCM (little endian). The decoders read the compressed audio data from standard input and produce PCM on standard output at a sampling frequency of 44.1KHz. Mp3dec decodes MPEG audio (layer 1, 2 and 3). The –d option enables debug output to standard error. Oggdec, flacdec, sunwdec and wavdec are like mp3dec but decode OGG Vorbis, FLAC lossless audio, Sun audio and RIFF wave. Decoding options –s seconds   seek to a specific position in seconds before decoding. The encoders read PCM on standard input and produce compressed audio on standard output. Flacenc, oggenc and mp3enc produce FLAC, OGG Vorbis and MP3 audio. For mp3enc, the MP3 file will use `constant bit–rate' (CBR) encoding by default, but that can be changed via ––abr (average bitrate desired, ABR) or –v (variable bitrate, VBR). Oggenc accept raw PCM in the same byte order as /dev/audio (little–endian), while mp3enc –r expects big–endian. Flacenc by default expects raw PCM in the same format as /dev/audio, but also supports signed integer samples of bit widths 4 to 32, either little– or big–endian, one to eight channels and arbitrary samplerates, see –i option of pcmconv. Encoding options –b   set minimum allowed bitrate in Kb/s for VBR, default 32Kb/s. For CBR, set the exact bitrate in Kb/s, which defaults to 128Kb/s. –B   set maximum allowed bitrate in Kb/s for VBR, default 256Kb/s. –h   same as –q 2. –m   mode may be (s)tereo, (j)oint, (f)orce or (m)ono (default j). force forces mid/side stereo on all frames. –p   add CRC error protection (adds an additional 16 bits per frame to the stream). This seems to break playback. –q   sets output quality to q (see –V). –r   input is raw pcm –s   set sampling frequency of input file (in KHz) to sfreq, default is 44.1. –v   use variable bitrate (VBR) encoding –V   set quality setting for VBR to q. Default q is 4; 0 produces highest–quality and largest files, and 9 produces lowest–quality and smallest files. Long options ––abr bitrate      sets average bitrate desired in Kb/s, instead of setting quality, and generates ABR encoding. ––resample sfreqset sampling frequency of output file (in KHz) to sfreq, default is input sfreq. ––mp3input inputis an MP3 file Silly options –f         same as –q 7. Such a deal. –o         mark as non–original (i.e. do not set the original bit) –c         mark as copyright –k         disable sfb=21 cutoff –e emp     de–emphasis n/5/c (default n) –d         allow channels to have different blocktypes –t         disable Xing VBR informational tag –a         autoconvert from stereo to mono file for mono encoding –x         force byte–swapping of input (see dd(1) instead) –S         don't print progress report, VBR histograms ––athonlyonly use the ATH for masking ––nohist   disable VBR histogram display ––voice    experimental voice mode Pcmconv is a helper program used to convert various PCM sample formats. The –i and –o options specify the input and output format fmt of the conversion. Fmt is a concatenated string of the following parts: s#    sample format is little–endian signed integer where # specifies the number of bits u#    unsigned little–endian integer format S#    signed big–endian integer format U#    unsigned big–endian integer format f#    floating point format where # has to be 32 or 64 for single– or double–precision a8    8–bit a–law format µ8    8–bit µ–law format c#    specifies the number of channels r#    gives the samplerate in Hz The program reads samples from standard input converting the data and writes the result to standard output until it reached end of file or, if –l was given, a number of length bytes have been consumed from input. Mixfs is a fileserver serving a single audio file which allows simultaneous playback of audio streams. When run, it binds over /dev/audio and mixes the audio samples that are written to it. A service name srvname can be given with the –s option which gets posted to /srv. By default, mixfs mounts itself on /mnt/mix and then binds /mnt/mix/audio and /mnt/mix/volume over /dev. /dev/volume from the parent namespace is proxied with an additional control "mix" which is used to set the output volume of the mixer. Another additional control "dev" can be used to switch between audio devices. Mixfs will resample incoming audio to the format of the audio device output if it does not match the default (s16c2r44100). An alternative mountpoint mtpt can be specified with the –m option. The –D option causes 9p debug messages to be written to file–descriptor 2.

EXAMPLE

	Play back an .mp3 audio/mp3dec <foo.mp3 >/dev/audio Encode a .wav file as highest–quality MP3. audio/mp3enc –q 0 –b 320 <foo.wav >foo.mp3 Create a fixed 128Kb/s MP3 file from a .wav file. audio/mp3enc –h <foo.wav >foo.mp3 Streaming from stereo 44.1KHz raw PCM data, encoding mono at 16KHz (you may not need dd): dd –conv swab | audio/mp3enc –a –r –m m ––resample 16 –b 24

SOURCE

	/sys/src/cmd/audio

SEE ALSO

	play(1), zuke(1) http://www.underbit.com/products/mad/ http://xiph.org/doc/ http://flac.sourceforge.net/documentation.html

HISTORY

	Pcmconv first appeared in 9front (December, 2012). Mixfs first appeared in 9front (December, 2013). Flacenc first appeared in 9front (January, 2021).