[Freeswitch-users] Achieving streaming ASR

[Dave Horton]

Wouldn’t it be better to use their websockets interface? 

http://www.ibm.com/watson/developercloud/speech-to-text/api/v1/?curl#websockets <http://www.ibm.com/watson/developercloud/speech-to-text/api/v1/?curl#websockets>

Dave
On Oct 17, 2016, at 9:35 PM, Luca Pradovera <luca.pradovera at gmail.com> wrote:

Hello,
sorry, somehow my mail client had hidden this conversation.

IBM Watson (my current use case for this) supports a Session type of recognition, similar to HTTP Chunked upload, where the difference is that each chunk is sent as its own POST.

I understand that the goal here could/would be to get this working with FS alone, but even having the ability to get raw frames on disk would be enough for now.

Do you think that would be feasible?

Best regards,

Luca

On Tue, Oct 11, 2016 at 1:26 AM, Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
ASR is implemented like a media bug.

On Mon, Oct 10, 2016 at 3:34 PM, Stanislav Sinyagin <ssinyagin at gmail.com <mailto:ssinyagin at gmail.com>> wrote:
Brian, Mike, as far as I understand, they talk about speech to text, not the other way round. TTS is something completely different )



On 10 Oct 2016 18:37, "Brian West" <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
It can return as many as it wants, as it gets them from the speech server, there really is no limit in the ASR/TTS interface, We reuse handles in the MRCP interface module.

/b


On Mon, Oct 10, 2016 at 11:29 AM, Dave Horton <daveh at beachdognet.com <mailto:daveh at beachdognet.com>> wrote:
Just an FYI on a couple of points:

1) I’ve tried to implment support for Nuance NCS API as an asr_tts module.  tts works fine, but asr / transcription does not seem possible because the NCS API is not really suited for a streaming application.  Based on my experience (and I opened tickets with them to try to resolve, to no avail), their server does not handle streaming because when I start sending small-ish chunks of audio using chunked-encoding, they immediately return me a “no speech” result.  They seem to expect that you have a recorded file on disk that you can ship over in much larger chunks.

2) I’m not completely sure I agree that transcription fits into the asr part of the asr_tts modules.  Some of these cloud-based services support providing a stream of interim results while the speaker is talking, and this would be very useful to propogate up to the freeswitch application (or esl) but the asr_tts pattern seems to be to return a single speech result at the end of a segment of speech.

Dave


On Oct 10, 2016, at 11:54 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:

This should be implemented the way we implement other speech interfaces.


> On Oct 8, 2016, at 4:53 PM, Luca Pradovera <luca.pradovera at gmail.com <mailto:luca.pradovera at gmail.com>> wrote:
>
> Hello,
> following a discussion on IRC, I thought I would move this here so it is more async.
>
> The need is to be able to use Nuance/IBM Watson/Google Speech APIs in streaming mode for real-time transcription (essentially, captioning a video conference and getting a transcript of the proceedings).
>
> My first idea was to simply get the frames from a recording as it is being written, though there might be some obstacles there.
>
> It was suggested the media bugs infrastructure would work, at the C level, though I need to check which formats it supports.
>
> Do you have any other suggestions?
>
> Thank you!
>
> Luca

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