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@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@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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