Provided by: julius_4.2.2-0ubuntu2_amd64 
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)