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adinrec - Online in the Cloud

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This is the command adinrec that can be run in the OnWorks free hosting provider using one of our multiple free online workstations such as Ubuntu Online, Fedora Online, Windows online emulator or MAC OS online emulator



- record audio device and save one utterance to a file


adinrec [options...] {filename}


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).


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)

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)

This option enables DC offset removal.

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

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.

-NA devicename
Host name for DatLink server input (-input netaudio).

-adport port_number
With -input adinnet, specify adinnet port number to listen. (default: 5530)

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.


Device name string for ALSA. (default: "default")

Device name string for OSS. (default: "/dev/dsp")

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.

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