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ADINREC(1) General Commands Manual ADINREC(1)

adinrec
 

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.
12/19/2013