adinrec ADINREC(1) 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. -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)