Streaming For ASR
Hello, I have been working on designs for two different projects, where both of them would need to use the IBM Watson streaming ASR service.
Based on our discussion at AstriDevCon, I know there is currently no support for that. However, there may be some workarounds I am not aware of.
Would it be possible to write out the audio frames as they get recorded?
Watson supports 16 bit signed little endian audio, IIRC, but there are a few other raw formats available.
The goal is to eventually fund the development of a small module to get something a bit better working, but for now I would be happy with a solution I can build a prototype on.
Thanks to everybody!
Luca
8 thoughts on - Streaming For ASR
That’s the main problem I was discussing at AstriDevCon. There’s no way currently to stream the frames to a service.
At the moment I’m saving to a file and then sending that file once recording has finished.
Matt Riddell wrote:
The UnicastRTP channel driver allows you to send RTP to a specific target address with media. Combined with Chanspy (or Snoop channels in ARI) you can duplicate audio from a channel and send it off to where you want.
So originate a new channel, make one leg a UnicastRTP and the other a chanspy to spy on the channel you’re interested in transcribing?
Cheers,
Matt Riddell
Matt Riddell wrote:
Theoretically, yes. Orchestrating stuff could get complicated but that is a way to get media out.
There is also EAGI, not very flexible but still an option.
Hello,
(sorry for not continuing the thread, I had set the list to digest).
Would UnicastRTP be able to output u-law frames directly? If so, I think that is all I need. Does anyone know what the EAGI output is? Raw RTP?
Best regards,
Luca
Joshua Colp did a great writeup that may work for your situation:
http://www.joshua-colp.com/broadcasting-asterisk-conferences/ <http://www.joshua-colp.com/broadcasting-asterisk-conferences/>
I’m still working on mine 🙂
Matt Riddell wrote:
I did indeed, although I think the options have since changed.
channel originate UnicastRTP/127.0.0.1:5001//c(g722) extension 1000@test
Would be the current.
As well running from git is currently needed as a bug was fixed which would cause a crash in UnicastRTP.