3 mp3dec, mp3enc, oggdec, oggenc, flacdec, sundec, wavdec, pcmconv, mixfs \- decode and encode audio files
41 .I "long or silly options"
68 These programs decode and encode various audio formats from and to
69 16-bit stereo PCM (little endian). The decoders read the compressed
70 audio data from standard input and produce PCM on standard output at
71 a sampling frequency of 44.1KHz.
74 decodes MPEG audio (layer 1, 2 and 3). The
76 option enables debug output to standard error.
84 but decode OGG Vorbis, FLAC lossless audio, Sun audio and RIFF wave.
86 The encoders read PCM on standard input and produce compressed audio
92 produce OGG Vorbis and MP3 audio. For
94 the MP3 file will use `constant bit-rate' (CBR) encoding by default,
95 but that can be changed via
97 (average bitrate desired, ABR)
100 (variable bitrate, VBR).
107 in Kb/s for VBR, default 32Kb/s.
109 set the exact bitrate in Kb/s, which defaults to 128Kb/s.
114 in Kb/s for VBR, default 256Kb/s.
130 forces mid/side stereo on all frames.
133 add CRC error protection (adds an additional 16 bits per frame to the stream).
134 This seems to break playback.
137 sets output quality to
146 set sampling frequency of input file (in KHz) to
151 use variable bitrate (VBR) encoding
154 set quality setting for VBR to
159 0 produces highest-quality and largest files, and
160 9 produces lowest-quality and smallest files.
162 .TF "\fB--resample sfreq \fP"
167 desired in Kb/s, instead of setting quality,
168 and generates ABR encoding.
170 .BI --resample " sfreq"
171 set sampling frequency of output file (in KHz) to
173 default is input sfreq.
175 .BI --mp3input " input"
187 mark as non-original (i.e. do not set the original bit)
193 disable sfb=21 cutoff
200 allow channels to have different blocktypes
203 disable Xing VBR informational tag
206 autoconvert from stereo to mono file for mono encoding
209 force byte-swapping of input (see
214 don't print progress report, VBR histograms
217 only use the ATH for masking
220 disable VBR histogram display
223 experimental voice mode
226 is a helper program used to convert various PCM sample formats. The
230 options specify the input and output format
234 is a concatenated string of the following parts:
237 sample format is little-endian signed integer where
239 specifies the number of bits
242 unsigned little-endian integer format
245 singed big-endian integer format
248 unsigned big-endian integer format
251 floating point format where
253 has to be 32 or 64 for single- or double-precision
262 specifies the number of channels
265 gives the samplerate in Hz
267 The program reads samples from standard
268 input converting the data and writes the result to standard output
269 until it reached end of file or, if
271 was given, a number of
273 bytes have been consumed from input.
276 is a fileserver serving a single
278 file which allows simultaneous playback of audio streams. When
281 and mixes the audio samples that are written to it.
284 can be given with the
286 option which gets posted to
296 A alternative mountpoint
298 can be specified with the
305 debug messages to be written to file-descriptor 2.
311 audio/mp3dec <foo.mp3 >/dev/audio
316 file as highest-quality MP3.
319 audio/mp3enc -q 0 -b 320
322 Create a fixed 128Kb/s MP3 file from a
327 audio/mp3enc -h <foo.wav >foo.mp3
330 Streaming from stereo 44.1KHz raw PCM data, encoding mono at 16KHz
335 dd -conv swab | audio/mp3enc -a -r -m m --resample 16 -b 24
338 .B /sys/src/cmd/audio
344 .B http://www.underbit.com/products/mad/
346 .B http://xiph.org/doc/
348 .B http://flac.sourceforge.net/documentation.html
351 first appeared in 9front (December, 2012).
353 first appeared in 9front (December, 2013).