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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).

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.

License

       The same as Julius.

                                                   10/02/2008                                         ADINREC(1)

Name

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

Options

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

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

        -raw
          Output in raw file format.

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

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

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

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

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

        -zmean
          This option enables DC offset removal.

        -smpFreqHz
          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)

        -NAdevicename
          Host name for DatLink server input (-inputnetaudio).

        -adportport_number
          With -inputadinnet, specify adinnet port number to listen. (default: 5530)

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

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

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

See Also

julius ( 1 ) ,
        adintool ( 1 )

Synopsis

adinrec [options...] {filename}

See Also