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.
164 .BI --mp3input " input"
176 mark as non-original (i.e. do not set the original bit)
182 disable sfb=21 cutoff
189 allow channels to have different blocktypes
192 disable Xing VBR informational tag
195 autoconvert from stereo to mono file for mono encoding
198 force byte-swapping of input (see
203 don't print progress report, VBR histograms
206 only use the ATH for masking
209 disable VBR histogram display
212 experimental voice mode
215 is a helper program used to convert various PCM sample formats. The
219 options specify the input and output format
223 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 singed big-endian integer format
237 unsigned big-endian integer format
240 floating point format where
242 has to be 32 or 64 for single- or double-precisition
251 specifies the number of channels
254 gives the samplerate in Hz
256 The program reads samples from standard
257 input converting the data and writes the result to standard output
258 until it reached end of file or, if
260 was given, a number of
262 bytes have been consumed from input.
268 audio/mp3dec <foo.mp3 >/dev/audio
273 file as highest-quality MP3.
276 audio/mp3enc -q 0 -b 320
279 Create a fixed 128Kb/s MP3 file from a
284 audio/mp3enc -h <foo.wav >foo.mp3
287 Streaming from stereo 44.1KHz raw PCM data, encoding mono at 16KHz
292 dd -conv swab | audio/mp3enc -a -r -m m --resample 16 -b 24
295 .B /sys/src/cmd/audio
301 .B http://www.underbit.com/products/mad/
303 .B http://xiph.org/doc/
305 .B http://flac.sourceforge.net/documentation.html
307 It's another GNU behemoth, lightly tamed.