On the one hand, your problem becomes more than just sending the audio stream, but also receiving the transcription in real time and then sending it to whatever endpoint needs it, cause Asterisk ain't doing anything with it. My program get a correct respon from google when the flac file recorded manual by using windows's sound recorder and convert it using a software converter. I personally see little use in going over the hurdles of building and Asterisk module to do real.time transcription for a server that doesn't do video, but if I were to do this I would try ARI instead of the module.īut then of course, I don't know your use case.ĭo you need the transcripts while the audio is being streamed or is a transcription after the call ends enough? Google Chrome is a browser that combines a minimal design with sophisticated technology to make the web faster, safer, and easier. Streams (which is what I think you want to do) If my memory serves me well, there are many entries for this in and asterisk forums.
#Sample google speech to text api install
The crux of the matter is not Asterisk, but google's API. Install the Google Cloud Speech-to-Text API client library for C First, create a simple C console application that you will use to run Speech-to-Text API samples: dotnet new console -n SpeechToTextApiDemo You should see the application created and dependencies resolved: The template 'Console Application' was created successfully. Unfortunately, there is no standard C sdk for google speech.
In this example you sent it a complete audio file, but you can also use. Modules: afaik, modules in Asterisk are in C. The Speech API supports both synchronous and asynchronous speech to text transcription.
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