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.

                                                   10/02/2008                                         ADINREC(1)