# \[Announce\] node-red-contrib-voice2json (beta)

**URL:** <https://discourse.nodered.org/t/announce-node-red-contrib-voice2json-beta/29066>\
**Category:** Share Your Nodes\
**Created:** [27 June 2020 13:43 UTC](https://discourse.nodered.org/t/announce-node-red-contrib-voice2json-beta/29066 "2020-06-27T13:43:57Z")\
**Posts on this page:** 1\
**Showing post:** 24

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**Author:** ![JGKK](https://sea2.discourse-cdn.com/flex026/user_avatar/discourse.nodered.org/jgkk/32/18515_2.png) [@JGKK](https://discourse.nodered.org/u/JGKK)\
**Post date:** [3 July 2020 14:34 UTC](https://discourse.nodered.org/t/announce-node-red-contrib-voice2json-beta/29066/24 "2020-07-03T14:34:33Z")

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> [@WhiteLion](#):
>
> I installed the Deepspeak german profile

I think the deepspeech model for german is not actually compatible with a pi. Because afaik voice2json uses this model: [GitHub - AASHISHAG/deepspeech-german: Automatic Speech Recognition (ASR) - German](https://github.com/AASHISHAG/deepspeech-german) which will not work for arm. Unfortunately deepspeech works with two different model formats where one of them doesn’t work on arm due to missing tensorflow features or something. I think only the english model works on the pi for deepspeech. I can’t really recommend it for now anyway see this post [Ways to control Node-RED with speech commands - #115 by JGKK](https://discourse.nodered.org/t/ways-to-control-node-red-with-speech-commands/5530/115) where i talk about it in more detail. For german I really recommend the Kaldi profile which i also use personally as it’s really fast. It’s the model from [GitHub - gooofy/zamia-speech: Open tools and data for cloudless automatic speech recognition](https://github.com/gooofy/zamia-speech) which is a great project that have some of the best open source models for kaldi in german and french.

> [@WhiteLion](#):
>
> voice2json transcribe-stream" did throw an "ALSA Error"

If you’d want to use transcribe stream from the commandline which our nodes don’t support for a couple of reasons you will actually need a microphone and a properly configured asound.conf file.  
We didn’t implement transcribe stream which is a combination of record-command and stt because it didn’t actually offer any performance improvements due to a couple of reasons. Every time you start transcribe-stream it has to load all the libraries for the stt part again which is one of the biggest slow down factors. I actually implemented a little trick in our stt node that wouldn’t work in transcribe-stream. We keep stt (transcribe-wav) running with all libraries already loaded once your first transcription was done. It will just idle in the background but this way on every subsequent transcription it works a lot faster. Also it’s a lot more stable and modular with record command and stt separated.

> [@WhiteLion](#):
>
> I didn t add slots to the slots tab

No you don’t need any slots at all. You could write everything just in the sentences.ini.

I hope this answers all of your questions for now.

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