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AUDIORECORD(1) General Commands Manual AUDIORECORD(1) NAME audiorecord - record audio files SYNOPSIS audiorecord [-afhqV] [-B buffersize] [-b balance] [-c channels] [-d device] [-e encoding] [-F format] [-i info] [-m monvol] [-P precision] [-p port] [-s rate] [-t time] [-v volume] file DESCRIPTION The audiorecord program copies the audio device to the named audiofile or, if the file name is -, to the standard output. The output file will contain either a Sun/NeXT audio header, a RIFF/WAVE audio header or no header at all. Sun output files using a linear PCM encoding are written with big-endian signed samples, possibly after converting these from little-endian or unsigned samples. RIFF/WAVE files are written in little-endian, signed samples, also converting if necessary. The default output is Sun/NeXT format, but if the output file file ends with a .wav file extension it will be written as RIFF/WAVE. OPTIONS The following options are available: -a Append to the specified file, rather than overwriting. -B buffersize Set the audio device read buffer size to buffersize. The default value is the record.buffer_size of the audio device. -b balance Set the balance to balance. This value must be between 0 and 63. -c channels Set number of channels to channels. -d device Set the audio device to be device. The default is /dev/sound. -e encoding Set encoding to either "alaw", "ulaw", or "linear", or any other value reported by audioctl encodings. The default encoding is "ulaw". If the output format is "sun", the file will contain slinear_be samples, if it is "wav", then slinear_le, independent of the argument to -e. Setting the argument to -e still may be important since it is used in an ioctl(2) call to the kernel to choose the kind of data provided. -F format Set the output header format to format. Currently supported formats are "sun", "wav", and "none" for Sun/NeXT audio, WAV, and no header, respectively. -f Force. Normally when appending to audiofiles using the -a option, the sample rates must match. The -f option will allow a discrepancy to be ignored. -h Print a help message. -i info If supported by the -F format, add the string info to the output header. -m monvol Set the monitor volume. -P precision Set the precision. This value is the number of bits per sample, and is normally either "8" or "16", though the values "4", "24", and "32" are also valid. -p port Set the input port to port. The valid values of port are "cd", "internal-cd", "mic", and "line". -q Be quiet. -s rate Set the sampling rate. This value is per-second. Typical values are 8000, 44100, and 48000, which are the telephone, CD Audio, and DAT Audio default sampling rates. -t time Sets the maximum amount of time to record. Format is [hh:]mm:ss[.dddddd]. -V Be verbose. -v volume Set the volume (gain) to volume. This value must be between 0 and 255. ENVIRONMENT AUDIOCTLDEVICE the audio control device to be used. AUDIODEVICE the audio device to be used. SEE ALSO audioctl(1), audioplay(1), aria(4), audio(4), audioamd(4), auich(4), autri(4), auvia(4), clcs(4), clct(4), cmpci(4), eap(4), emuxki(4), esm(4), eso(4), ess(4), fms(4), gus(4), guspnp(4), neo(4), sb(4), sv(4), wss(4), yds(4), ym(4) HISTORY The audiorecord program was first seen in SunOS 5. It was first made available in NetBSD 1.4. RIFF/WAVE support, and support for converting signed/unsigned and big/little-endian samples was first made available in NetBSD 1.6. AUTHORS The audiorecord program was written by Matthew R. Green <email@example.com>. BUGS WAV big-endian samples are converted to little-endian, rather than a RIFX header being written. NetBSD 8.0 December 30, 2010 NetBSD 8.0