3 mp3dec, mp3enc, oggdec, oggenc, flacdec, sundec, wavdec, pcmconv \- decode and encode audio files
41 .I "long or silly options"
57 These programs decode and encode various audio formats from and to
58 16-bit stereo PCM (little endian). The decoders read the compressed
59 audio data from standard input and produce PCM on standard output at
60 a sampling frequency of 44.1KHz.
63 decodes MPEG audio (layer 1, 2 and 3). The
65 option enables debug output to standard error.
73 but decode OGG Vorbis, FLAC lossless audio, Sun audio and RIFF wave.
75 The encoders read PCM on standard input and produce compressed audio
81 produce OGG Vorbis and MP3 audio. For
83 the MP3 file will use `constant bit-rate' (CBR) encoding by default,
84 but that can be changed via
86 (average bitrate desired, ABR)
89 (variable bitrate, VBR).
96 in Kb/s for VBR, default 32Kb/s.
98 set the exact bitrate in Kb/s, which defaults to 128Kb/s.
103 in Kb/s for VBR, default 256Kb/s.
119 forces mid/side stereo on all frames.
122 add CRC error protection (adds an additional 16 bits per frame to the stream).
123 This seems to break playback.
126 sets output quality to
135 set sampling frequency of input file (in KHz) to
140 use variable bitrate (VBR) encoding
143 set quality setting for VBR to
148 0 produces highest-quality and largest files, and
149 9 produces lowest-quality and smallest files.
151 .TF "\fB--resample sfreq \fP"
156 desired in Kb/s, instead of setting quality,
157 and generates ABR encoding.
159 .BI --resample " sfreq"
160 set sampling frequency of output file (in KHz) to
162 default is input sfreq.
177 mark as non-original (i.e. do not set the original bit)
183 disable sfb=21 cutoff
190 allow channels to have different blocktypes
193 disable Xing VBR informational tag
196 autoconvert from stereo to mono file for mono encoding
199 force byte-swapping of input (see
204 don't print progress report, VBR histograms
207 only use the ATH for masking
210 disable VBR histogram display
213 experimental voice mode
217 is a helper program used to convert various PCM sample formats. The
221 options specify the input and output format
225 is a concatinated string of the following parts:
229 sample format is little-endian signed integer where
231 specifies the number of bits
234 unsigned little-endian integer format
237 singed big-endian integer format
240 unsigned big-endian integer format
243 floating point format where
245 has to be 32 or 64 for single- or double-precisition
254 specifies the number of channels
257 gives the samplerate in Hz
260 The program reads samples from standard
261 input converting the data and writes the result to standard output
262 until it reached end of file or, if
264 was given, a number of
266 bytes have been consumed from input.
273 audio/mp3dec <foo.mp3 >/dev/audio
278 file as highest-quality MP3.
281 audio/mp3enc -q 0 -b 320
284 Create a fixed 128Kb/s MP3 file from a
289 audio/mp3enc -h <foo.wav >foo.mp3
292 Streaming from stereo 44.1KHz raw PCM data, encoding mono at 16KHz
297 dd -conv swab | audio/mp3enc -a -r -m m --resample 16 -b 24
300 .B /sys/src/cmd/audio
306 .B http://www.underbit.com/products/mad/
308 .B http://xiph.org/doc/
310 .B http://flac.sourceforge.net/documentation.html
312 It's another GNU behemoth, lightly tamed.