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3b1aacbe6d
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87 Commits

Author SHA1 Message Date
Georgi Gerganov
3b1aacbe6d talk : talk with AI in the terminal 2022-12-10 16:51:58 +02:00
Georgi Gerganov
56822621a8 twitch.sh : various fixes and polishing 
- check if streamlink is installed
- fix audio chunking
- change default threads to 4
 2022-12-08 19:20:04 +02:00
keyehzy
9e5f3ddc16 Allow for Twitch.tv live transcription 
We rely on streamlink library to give us a stream, then we proceed similarly to
the radio livestream example.
 2022-12-08 19:20:04 +02:00
Georgi Gerganov
47afb93c3c
yt-wsp.sh : improve usage instructions 2022-12-07 22:12:08 +02:00
Georgi Gerganov
575c53dc41
yt-wsp.sh : fix usage instruction + comment 2022-12-07 21:12:55 +02:00
Georgi Gerganov
faa85f9840 livestream.sh : remove obsolete comment 2022-12-07 04:41:43 +02:00
Georgi Gerganov
9fe7306f4b
models : add the new "large" model release by OpenAI 
The old "large" model is now renamed "large-v1".
If you have been using it, make sure to rename it and download the new
"large" model for best results.
 2022-12-06 18:48:57 +02:00
Georgi Gerganov
57e0e6b700
livestream : handle ffmpeg errors gracefully and stabilize transcript 2022-12-01 20:49:09 +02:00
Georgi Gerganov
4f7363077f
livestream : minor changes 2022-12-01 19:47:58 +02:00
semiformal-net
093c840dee
livestream : fix losing words across audio chunk (#195) 
* improve livestream script

* Update examples/livestream.sh

Co-authored-by: Georgi Gerganov <ggerganov@gmail.com>

Co-authored-by: Paul Edwards <paul.edwards@semiformal.net>
Co-authored-by: Georgi Gerganov <ggerganov@gmail.com>
 2022-12-01 19:18:22 +02:00
Georgi Gerganov
4698dcdb52 whisper : add mechanism for aborting the whisper_full() computation 2022-11-27 20:42:45 +02:00
Georgi Gerganov
164df0d447
whisper.objc : fix context + broken readme links 2022-11-27 10:52:27 +02:00
Georgi Gerganov
e266cb0723
whisper.objc : add real-time processing (#97) 
Similar to the "stream" app
 2022-11-26 18:32:46 +02:00
Georgi Gerganov
c207eed431
whisper.objc : fix build warnings 2022-11-26 16:27:04 +02:00
Georgi Gerganov
a425365b82
yt-wsp.sh : script to easily transcribe VODs 
Thanks to @DaniruKun
ref: https://gist.github.com/DaniruKun/96f763ec1a037cc92fe1a059b643b818

Usage:

  cd whisper.cpp
  make

  ./examples/yt-wsp.sh <video-url>
 2022-11-26 12:54:42 +02:00
Georgi Gerganov
68ecadbbc9
command.wasm : add voice assistant example for the Web (#171) 
Same as the command-line tool "command", but runs in the browser

Also, added helper script "extra/deploy-wasm.sh" and fixed some timing
constants for the WASM examples.
 2022-11-26 11:40:06 +02:00
Georgi Gerganov
c536ff4005
minor : add comment for using "generate_karaoke.sh" 2022-11-26 10:22:42 +02:00
Georgi Gerganov
cb70b07db5
livestream.sh : simple tool to transcribe audio livestreams (#185) 2022-11-26 10:05:37 +02:00
Georgi Gerganov
3c390ffe38
stream.wasm : add web-based real-time transcription (#112) 2022-11-25 23:57:46 +02:00
Georgi Gerganov
be16dfa038
whisper.wasm : do not block page while processing (close #86) 2022-11-25 23:07:42 +02:00
Georgi Gerganov
0f619b52ce
main : add stereo-channel-based diarization (#64) 
Not tested - I don't have stereo dialog audio
 2022-11-25 22:08:58 +02:00
Georgi Gerganov
1246dd023e
command : add demonstration video 2022-11-25 20:23:58 +02:00
Georgi Gerganov
0be27bbd92
command : fix build + fix README + add bold printing 2022-11-25 19:53:50 +02:00
Georgi Gerganov
bc88eb13c6
examples : add "command" tool (#171) 2022-11-25 19:36:57 +02:00
Georgi Gerganov
b8ce25dec1
refactoring : more readable code 2022-11-25 19:28:04 +02:00
Georgi Gerganov
e4805d9601
wasm : refactor wasm example + reuse fetch mechanism 2022-11-24 23:13:26 +02:00
Georgi Gerganov
ff36415a86
talk.wasm : update video link + some minor fixes 2022-11-24 20:15:24 +02:00
Georgi Gerganov
025ff465b6
Update README.md 
Use a less cringy video to demo talk.wasm lol
 2022-11-24 20:09:45 +02:00
Georgi Gerganov
abce28ea99
talk.wasm : move to https://whisper.ggerganov.com/talk 
This way, we can share the same models across different WASM examples
and not have to download them for each page
 2022-11-24 18:24:06 +02:00
Georgi Gerganov
454b91de16
main : fix dangling pointer when using stdin for input (#65) 2022-11-24 17:53:51 +02:00
Georgi Gerganov
d7024cf9dc
main, stream : remove --verbose flag (#178) 2022-11-24 17:52:04 +02:00
Georgi Gerganov
37422ed733
talk.wasm : add audio pre-processing + bump memory 2022-11-24 00:34:00 +02:00
Georgi Gerganov
be3b720f96
talk.wasm : refactoring + update README.md 2022-11-24 00:08:57 +02:00
Georgi Gerganov
49706a658a
minor : updates few prints + fix buttons in whisper.wasm 2022-11-23 17:19:21 +02:00
Georgi Gerganov
e5dcdabbb8
unicode : fix character replacement (thanks to @tamo) 2022-11-23 08:24:29 +02:00
Georgi Gerganov
dad109c3f1
close #109 : add fetching of the model over HTTP (whisper.wasm) 2022-11-22 22:48:56 +02:00
Georgi Gerganov
326573de9a
talk.wasm : final touches 2022-11-22 22:22:17 +02:00
Georgi Gerganov
9aea96f774
talk.wasm : polishing + adding many AI personalities 2022-11-22 20:10:20 +02:00
Georgi Gerganov
385236d1d3
stream : "-kc" now enables context keeping from previous segment (#90) 
By default, the context keeping is disabled
 2022-11-22 18:21:15 +02:00
M. Eren Akbiyik
63ae03b8e0
Prompt previous tokens for streaming (#163) 
* feat: prompt previous tokens for streaming

I used a vector pointer instead of vector itself because it gave weird errors, and why not

* convert vector to use with C api

* feat: remove old refs, check for prompt size

* feat: use better way of getting the pointer
 2022-11-22 18:10:35 +02:00
Georgi Gerganov
78116f8eda
talk.wasm : update README.md 2022-11-21 22:42:29 +02:00
Georgi Gerganov
a4dfbeecf9
talk.wasm : GPT-2 meets Whisper in WebAssembly (#155) 
* talk : initial real-time transcription in the browser

* talk : polishing the UI

* talk : ready for beta testing

* talk.wasm : rename example
 2022-11-21 22:20:42 +02:00
Georgi Gerganov
f2df9bd768 stream : add "max_tokens" cli arg 
Controls the max tokens per segment for the stream example
 2022-11-20 21:22:41 +02:00
Georgi Gerganov
fb8d77f760 stream : add "audio_ctx" parameter 
Used to overwrite the audio context size of the Encoder.
For example, setting "audio_ctx = 512" will make it run about 3 times
faster, processing about 10s of audio, instead of 30s.

The transcription quality drops, but this can be used for real-time
streaming purposes where performance is important.
 2022-11-20 21:22:41 +02:00
Georgi Gerganov
62b5ff875c stream : add "max_tokens" parameter 
Used to limit the number of tokens in a segment.
Useful to battle with word repetition when using partial encoder context
 2022-11-20 21:22:41 +02:00
Georgi Gerganov
d351771a4b stream : add "single_segment" option 
Force the entire audio chunk to be transcribed into a single segment
 2022-11-20 21:22:41 +02:00
Georgi Gerganov
c058aaf22e stream : partial encoder experiments 2022-11-20 21:22:41 +02:00
Georgi Gerganov
83c742f1a7 whisper : add option to speed up the audio tempo by x2 
Using a Phase Vocoder for speeding up the audio tempo by scaling down
the frequencies in the frequency domain.

This reduces the computation in the Encoder by a factor of 2.
The transcription accuracy is degraded, but for slow to normal speech -
it seems to be still very good.

I think this can find application for real-time transcription - i.e. the
"stream" example.
 2022-11-13 16:25:43 +02:00
Alan
7519eabf65 Adds support for stdin wav input 2022-11-09 20:37:23 +02:00
Georgi Gerganov
c30bffc8a5
ref #22 : add "duration" option 
Can be used to partially process a recording
 2022-11-07 20:14:52 +02:00