ref: d82b300c148b52471603d9b77cd46d0955946f09
dir: /sys/man/1/audio/
.TH AUDIO 1 .SH NAME mp3dec, mp3enc, oggdec, oggenc, flacdec, flacenc, sundec, wavdec, vocdec, pcmconv, mixfs \- decode and encode audio files .SH SYNOPSIS .B audio/mp3dec [ .B -s .I seconds ] [ .B -d ] .br .B audio/oggdec [ .B -s .I seconds ] .br .B audio/flacdec [ .B -s .I seconds ] .br .B audio/wavdec [ .B -s .I seconds ] .br .B audio/sundec .br .B audio/opusdec [ .B -h ] [ .I options ] .PP .B audio/oggenc .br .B audio/mp3enc [ .B -hprv ] [ .B -b .I bitrate ] [ .B -B .I bitrate ] [ .B -m .I mode ] [ .B -q q ] [ .B -s .I sfreq ] [ .B -V .I q ] [ .I "long or silly options" ] .br .B audio/flacenc [ .B -b .I bitspersample ] [ .B -c .I channels ] [ .B -l .I compresslevel ] [ .B -s .I sfreq ] [ .B -P .I padding ] [ .B -T .I field=value ] .br .B audio/opusenc [ .B -h ] [ .I options ] .PP .B audio/pcmconv [ .B -i .I fmt ] [ .B -o .I fmt ] [ .B -l .I length ] .PP .B audio/mixfs [ .B -D ] [ .B -s .I srvname ] [ .B -m .I mtpt ] .PP .SH 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. .PP .I Mp3dec decodes MPEG audio (layer 1, 2 and 3). The .B -d option enables debug output to standard error. .I Oggdec, .I flacdec, .I sunwdec and .I wavdec are like .I mp3dec but decode OGG Vorbis, FLAC lossless audio, Sun audio and RIFF wave. .SS Decoding options .TF "\fB-s seconds \fP" .TP .BI -s " seconds" seek to a specific position in seconds before decoding. .SS .PP The encoders read PCM on standard input and produce compressed audio on standard output. .PP .IR Flacenc , .IR oggenc , .IR opusenc , and .I mp3enc produce FLAC, OGG Vorbis, OGG Opus and MP3 audio. For .I mp3enc, the MP3 file will use `constant bit-rate' (CBR) encoding by default, but that can be changed via .B --abr (average bitrate desired, ABR) or .BR -v (variable bitrate, VBR). .I Opusenc uses VBR by default. .PP .IR Flacenc , .I opusenc and .I oggenc accept raw PCM in the same byte order as .B /dev/audio (little-endian), while .I mp3enc -r expects big-endian. .SS Encoding options .TF "\fB-b \fP" .TP .B -b set minimum allowed .I bitrate in Kb/s for VBR, default 32Kb/s. For CBR, set the exact bitrate in Kb/s, which defaults to 128Kb/s. .TP .B -B set maximum allowed .I bitrate in Kb/s for VBR, default 256Kb/s. .TP .BI -h same as .LR "-q 2" . .TP .B -m .I mode may be (s)tereo, (j)oint, (f)orce or (m)ono (default j). .B force forces mid/side stereo on all frames. .TP .B -p add CRC error protection (adds an additional 16 bits per frame to the stream). This seems to break playback. .TP .B -q sets output quality to .I q (see .BR -V ). .TP .B -r input is raw pcm .TP .B -s set sampling frequency of input file (in KHz) to .IR sfreq , default is 44.1. .TP .B -v use variable bitrate (VBR) encoding .TP .B -V set quality setting for VBR to .IR q . Default .I q is 4; 0 produces highest-quality and largest files, and 9 produces lowest-quality and smallest files. .SS Long options .TF "\fB--resample sfreq \fP" .TP .BI --abr " bitrate" sets average .I bitrate desired in Kb/s, instead of setting quality, and generates ABR encoding. .TP .BI --resample " sfreq" set sampling frequency of output file (in KHz) to .IR sfreq , default is input sfreq. .TP .BI --mp3input " input" is an MP3 file . .SS Silly options .TF --nohist .TP .B -f same as .LR "-q 7" . Such a deal. .TP .B -o mark as non-original (i.e. do not set the original bit) .TP .B -c mark as copyright .TP .B -k disable sfb=21 cutoff .TP .BI -e " emp" de-emphasis n/5/c (default n) .TP .B -d allow channels to have different blocktypes .TP .B -t disable Xing VBR informational tag .TP .B -a autoconvert from stereo to mono file for mono encoding .TP .B -x force byte-swapping of input (see .IR dd (1) instead) .TP .B -S don't print progress report, VBR histograms .TP .B --athonly only use the ATH for masking .TP .B --nohist disable VBR histogram display .TP .B --voice experimental voice mode .SS Like .IR mp3enc , opusenc accepts a number of long options to configure encoding parameters, input options and metadata. By default, it assumes raw PCM in the same byte order as .B /dev/audio with a 44.1KHz sampling rate and two channels. Similarly, .I opusdec will output that same format by default. Known knobs for both programs can be displayed with the .B -h option. .SS .I Pcmconv is a helper program used to convert various PCM sample formats. The .B -i and .B -o options specify the input and output format .I fmt of the conversion. .I Fmt is a concatenated string of the following parts: .TP .BI s # sample format is little-endian signed integer where .I # specifies the number of bits .TP .BI u # unsigned little-endian integer format .TP .BI S # signed big-endian integer format .TP .BI U # unsigned big-endian integer format .TP .BI f # floating point format where .I # has to be 32 or 64 for single- or double-precision .TP .B a8 8-bit a-law format .TP .B µ8 8-bit µ-law format .TP .BI c # specifies the number of channels .TP .BI r # gives the samplerate in Hz .PP 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 .B -l was given, a number of .I length bytes have been consumed from input. .PP .I Mixfs is a fileserver serving a single .B audio file which allows simultaneous playback of audio streams. When run, it binds over .B /dev/audio and mixes the audio samples that are written to it. A service name .I srvname can be given with the .B -s option which gets posted to .BR /srv . By default, .I mixfs mounts itself on .B /mnt/mix and then binds .B /mnt/mix/audio over .BR /dev . A alternative mountpoint .I mtpt can be specified with the .B -m option. The .B -D option causes .B 9p debug messages to be written to file-descriptor 2. .SH EXAMPLE Play back an .L .mp3 .IP .EX audio/mp3dec <foo.mp3 >/dev/audio .EE .LP Encode a .L .wav file as highest-quality MP3. .IP .EX audio/mp3enc -q 0 -b 320 <foo.wav >foo.mp3 .EE .LP Create a fixed 128Kb/s MP3 file from a .L .wav file. .IP .EX audio/mp3enc -h <foo.wav >foo.mp3 .EE .LP Streaming from stereo 44.1KHz raw PCM data, encoding mono at 16KHz (you may not need .IR dd ): .IP .EX dd -conv swab | audio/mp3enc -a -r -m m --resample 16 -b 24 .EE .LP Transcoding a FLAC file to Ogg Opus with slightly higher quality settings than default: .IP .EX audio/flacdec <foo.flac | audio/opusenc --music --downmix-stereo --bitrate 128 >foo.opus .EE .SH SOURCE .B /sys/src/cmd/audio .SH "SEE ALSO" .IR play (1), .IR zuke (1) .br .B http://www.underbit.com/products/mad/ .br .B http://xiph.org/doc/ .br .B http://flac.sourceforge.net/documentation.html .SH HISTORY .I Pcmconv first appeared in 9front (December, 2012). .I Mixfs first appeared in 9front (December, 2013). .I Flacenc first appeared in 9front (January, 2021).