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ADINREC

adinrec

NAME

adinrec
− record audio device and save one utterance to a file

SYNOPSIS

adinrec [options...] {filename}

DESCRIPTION

adinrec opens an audio stream, detects an utterance input and store it to a specified file. The utterance detection is done by level and zero−cross thresholds. Default input device is microphone, but other audio input source, including Julius A/D−in plugin, can be used by using "−input" option.

The audio format is 16 bit, 1 channel, in Microsoft WAV format. If the given filename already exists, it will be overridden.

If filename is "−" , the captured data will be streamed into standard out, with no header (raw format).

OPTIONS

adinrec uses JuliusLib and adopts Julius options. Below is a list of valid options.

adinrec specific options
−freq
Hz

Set sampling rate in Hz. (default: 16,000)

−raw

Output in raw file format.

JuliusLib options
−input
{mic|rawfile|adinnet|stdin|netaudio|esd|alsa|oss}

Choose speech input source. Specify ’file’ or ’rawfile’ for waveform file. On file input, users will be prompted to enter the file name from stdin.

´mic’ is to get audio input from a default live microphone device, and ’adinnet’ means receiving waveform data via tcpip network from an adinnet client. ’netaudio’ is from DatLink/NetAudio input, and ’stdin’ means data input from standard input.

At Linux, you can choose API at run time by specifying alsa, oss and esd.

−chunk_size samples

Audio fragment size in number of samples. (default: 1000)

−lv thres

Level threshold for speech input detection. Values should be in range from 0 to 32767. (default: 2000)

−zc thres

Zero crossing threshold per second. Only input that goes over the level threshold (−lv) will be counted. (default: 60)

−headmargin msec

Silence margin at the start of speech segment in milliseconds. (default: 300)

−tailmargin msec

Silence margin at the end of speech segment in milliseconds. (default: 400)

−zmean

This option enables DC offset removal.

−smpFreq Hz

Set sampling rate in Hz. (default: 16,000)

−48

Record input with 48kHz sampling, and down−sample it to 16kHz on−the−fly. This option is valid for 16kHz model only. The down−sampling routine was ported from sptk. (Rev. 4.0)

−NA devicename

Host name for DatLink server input (−input netaudio).

−adport port_number

With −input adinnet, specify adinnet port number to listen. (default: 5530)

−nostrip

Julius by default removes successive zero samples in input speech data. This option stop it.

−C jconffile

Load a jconf file at here. The content of the jconffile will be expanded at this point.

−plugindir dirlist

Specify which directories to load plugin. If several direcotries exist, specify them by colon−separated list.

ENVIRONMENT VARIABLES

ALSADEV

Device name string for ALSA. (default: "default")

AUDIODEV

Device name string for OSS. (default: "/dev/dsp")

LATENCY_MSEC

Input latency of microphone input in milliseconds. Smaller value will shorten latency but sometimes make process unstable. Default value will depend on the running OS.

SEE ALSO

julius ( 1 ) ,
adintool
( 1 )

COPYRIGHT

Copyright (c) 1997−2000 Information−technology Promotion Agency, Japan

Copyright (c) 1991−2008 Kawahara Lab., Kyoto University

Copyright (c) 2000−2005 Shikano Lab., Nara Institute of Science and Technology

Copyright (c) 2005−2008 Julius project team, Nagoya Institute of Technology

LICENSE

The same as Julius.

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