AUDIO(1) AUDIO(1) 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 [ -b bitspersample ] [ -c channels ] [ -l compresslevel ] [ -s sfreq ] [ -P padding ] [ -T field=value ] audio/pcmconv [ -i fmt ] [ -o fmt ] [ -l length ] audio/mixfs [ -D ] [ -s srvname ] [ -m mtpt ] 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 com- pressed 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). Flacenc and oggenc accept raw PCM in the same byte order as /dev/audio (little-endian), while mp3enc -r expects big- AUDIO(1) AUDIO(1) endian. 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 play- back. -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 pro- duces 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 sfreq set sampling frequency of output file (in KHz) to sfreq, default is input sfreq. --mp3input input is 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 --athonly only use the ATH for masking --nohist disable VBR histogram display --voice experimental voice mode Pcmconv is a helper program used to convert various PCM sam- ple formats. The -i and -o options specify the input and AUDIO(1) AUDIO(1) 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 over /dev. A 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 AUDIO(1) AUDIO(1) 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).