    adinrec

ADINREC(1)

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.

-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")

PORTAUDIO_DEV
    (portaudio V19) specify the name of capture device to use. See the instruction output of log at start up how to specify it.

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) 1991-2013 Kawahara Lab., Kyoto University

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

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

Copyright (c) 2005-2013 Julius project team, Nagoya Institute of Technology

LICENSE

The same as Julius.