A key service in the building of interactive voice systems.
Now, there are many online services with similar functionality, but most of them bring the risks of data leaks, randomly latencies on request processing, and requires a paid subscription.
This software is free from the enumerated shortcomings and enables to build complete local speech-to-text service.
This is a fast, lightweight solution with simple web-api which doesn't require some special hardware (such as GPU) and manage to operate on regular servers with appropriate performance.
Written in pure C, with libs:
mpg123,
alphacephei framework.
You won't depend on any online services, and spend money for subscription
There are open models for various languages
There are tools for it
Allows to defined the context dictionary (bag of accessible words) an use it during recognition
Allows to generate speakers signature vector for further identification
IBM x3550-M3 or event Core2 Quad with 2Gb RAM (like my local PBX for example)
There is already a ready-made module for integration with FreeSWITCH.
for details see: mod_sivr_stt
Takes a positive effect on performance and memory consumption
Enable to make easy integration with various applications
- wav
- mp3
- l16
- Linux
Example #1 (simple request)
Request:
curl http://127.0.0.1:8801/v1/transcriptions -X POST -H "Authorization: Bearer secret" -H "Content-Type: multipart/form-data" -F language="en" -F smodel="small" -F file="@test.mp3"
Response (json):
{
"text" : "hello world"
}
Example #2 (with speake identify)
Request:
curl http://127.0.0.1:8801/v1/transcriptions -X POST -H "Authorization: Bearer secret" -H "Content-Type: multipart/form-data" -F language="en" -F smodel="small" -F vmodel="default" -F file="@test.mp3"
Response (json):
{
"spk" : [-0.644623, 1.023342, 2.575434, 0.623447, -0.602342, 1.0234234 -1.4824234 -0.021242, 0.824297, -0.152424, ... ],
"spk_frames" : 81,
"text" : "hello world"
}