[Asterisk-Users] Voice detection

[Brad Waite]

Steve,

I don't have any real experience in DSP methodologies, although I have picked up 
on the high-level theories in my research.  However, I am *very* strong-willed 
in the "Where there's a will, there's a way" category.  :)

Here's my current thought:

Sphinx is an open source STT library that can work in real-time (specifically 
sphinx2).  Could we not pipe the called party's audio into it and then look for 
a given period of time with no text output?

I also found this site, 
http://www.cs.wpi.edu/~claypool/courses/525-S01/projects/proj1/ where a prof's 
got a project for students that fits perfectly with what I'd like to do.  He 
mentions Rabiner and Sambur's algorithm (from 1975) for detecting isolated 
speech endpoints.

Brad

Steve Underwood wrote:

> Hi Brad,
> 
> If you want to detect that a sound is voice, rather than something else, 
> it isn't easy. There is information around on the Internet about 
> methods, but I have never tried them and don't know how well they work. 
> Unless you have some understanding of DSP I wouldn't bother trying. On 
> the other hand, if you do have some DSP expertise it might be a fun 
> thing to try.
> 
> Regards,
> Steve
> 
> Brad Waite wrote:
> 
>> Does anyone know if there's public voice detection algorithms 
>> available?  I've scoured the net for the last hour or so, and I can't 
>> come up with anything except a few proprietary or embedded solutions.
>>
>> I know dsp.c uses goertzel algorithms for DTMF detection, but how does 
>> one detect voice?
>>
>> I dunno, maybe detecting voice isn't the way to go.  I want to begin 
>> playback of a file after a phone/answering machine has answered.
> 
> 
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>