[Freeswitch-users] Achieving streaming ASR

[Brian West]

ASR is implemented like a media bug.

On Mon, Oct 10, 2016 at 3:34 PM, Stanislav Sinyagin <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> 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>
>> 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> 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>
>>> 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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>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
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>>
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>
> _________________________________________________________________________
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-- 

*Brian West*
brian at freeswitch.org


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