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audio-source - for transcribe-stream ? #12
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Transcribe-stream would expect something like this: |
Thank you @johanneskropf for explanations and ideas :) I'm definitely trying, as you understood, to speed up the whole thing (so writing a full wav file to the disk and reading it after is not an option), keeping it simple. As mosquitto_sub can listen continuously to a topic where a publisher streams raw wav after having detected an hotword, piping it directly to voice2json would be the best thing to do to my mind. My target is a MQTT, network-enabled, service running something like : mosquitto_sub (raw wav) | voice2json transcribe-stream | voice2json recognize-intent | mosquitto_pub (intent) where mosquitto_sub would listen indefinitely to what is posted to hermes/audioServer/<SITE_ID>/audioFrame by snips-satellite (for the moment, and after that by something opensourced that will replace it) So I've got :
|
Yes it expects raw audio only. It actually uses webrtc vad internally i think to determine when a command was finished speaking. It does it pretty much the same way that record-command does. It records until it thinks the user stopped speaking or a timeout that is defined in your profile.yml is reached. |
You're right, with your detailed explanations I understand now why I definitely have to use transcribe-wav. Thank you very much ! I now just have to find spare time to do it :) |
Hello @synesthesiam, and thanks for your amazing work !
I am trying to stream from MQTT to transcribe-stream, but I can't.
When I try to transcribe-stream from stdin :
sox -t wav /tmp/test.wav -t wav - | /usr/bin/voice2json --debug transcribe-stream --audio-source -
I get that :
but I don't understand when I spoke about stdout ?
Regards,
Romain
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