3 mp3dec, mp3enc, oggdec, oggenc, flacdec, wavdec, pcmconv \- decode and encode audio files
39 .I "long or silly options"
55 These programs decode and encode various audio formats from and to
56 16-bit stereo PCM (little endian). The decoders read the compressed
57 audio data from standard input and produce PCM on standard output at
58 a sampling frequency of 44.1KHz.
61 decodes MPEG audio (layer 1, 2 and 3). The
63 option enables debug output to standard error.
70 but decode OGG Vorbis, FLAC lossless audio and PCM Wave.
72 The encoders read PCM on standard input and produce compressed audio
78 produce OGG Vorbis and MP3 audio. For
80 the MP3 file will use `constant bit-rate' (CBR) encoding by default,
81 but that can be changed via
83 (average bitrate desired, ABR)
86 (variable bitrate, VBR).
93 in Kb/s for VBR, default 32Kb/s.
95 set the exact bitrate in Kb/s, which defaults to 128Kb/s.
100 in Kb/s for VBR, default 256Kb/s.
116 forces mid/side stereo on all frames.
119 add CRC error protection (adds an additional 16 bits per frame to the stream).
120 This seems to break playback.
123 sets output quality to
132 set sampling frequency of input file (in KHz) to
137 use variable bitrate (VBR) encoding
140 set quality setting for VBR to
145 0 produces highest-quality and largest files, and
146 9 produces lowest-quality and smallest files.
148 .TF "\fB--resample sfreq \fP"
153 desired in Kb/s, instead of setting quality,
154 and generates ABR encoding.
156 .BI --resample " sfreq"
157 set sampling frequency of output file (in KHz) to
159 default is input sfreq.
174 mark as non-original (i.e. do not set the original bit)
180 disable sfb=21 cutoff
187 allow channels to have different blocktypes
190 disable Xing VBR informational tag
193 autoconvert from stereo to mono file for mono encoding
196 force byte-swapping of input (see
201 don't print progress report, VBR histograms
204 only use the ATH for masking
207 disable VBR histogram display
210 experimental voice mode
214 is a helper program used to convert various PCM sample formats. The
218 options specify the input and output format
222 is a concatinated string of the following parts:
226 sample format is little endian signed integer where
228 specifies the number of bits
231 unsigned little endian integer format
234 floating point format where
236 has to be 32 or 64 for single or double precisition
239 specifies the number of channels
242 gives the samplerate in Hz
245 The program reads samples from standard
246 input converting the data and writes the result to standard output
247 until it reached end of file or, if
249 was given, a number of
251 bytes have been consumed from input.
258 audio/mp3dec <foo.mp3 >/dev/audio
263 file as highest-quality MP3.
266 audio/mp3enc -q 0 -b 320
269 Create a fixed 128Kb/s MP3 file from a
274 audio/mp3enc -h <foo.wav >foo.mp3
277 Streaming from stereo 44.1KHz raw PCM data, encoding mono at 16KHz
282 dd -conv swab | audio/mp3enc -a -r -m m --resample 16 -b 24
285 .B /sys/src/cmd/audio
291 .B http://www.underbit.com/products/mad/
293 .B http://xiph.org/doc/
295 .B http://flac.sourceforge.net/documentation.html
297 It's another GNU behemoth, lightly tamed.