Georgi Gerganov
9fe7306f4b
models : add the new "large" model release by OpenAI
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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 )
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* 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 )
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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
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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 )
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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 )
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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
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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
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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 )
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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 )
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* 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 )
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* 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
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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
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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
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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
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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
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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
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Can be used to partially process a recording
2022-11-07 20:14:52 +02:00
Georgi Gerganov
c71363f14c
examples : add simple script for generating Karaoke video
2022-11-06 09:22:50 +02:00
Georgi Gerganov
d42cf6d0df
Update README.md
2022-11-04 22:26:08 +02:00
Georgi Gerganov
ef47d77492
main : fix generated bash script
2022-11-04 18:30:38 +02:00
Georgi Gerganov
d5afebd37c
whisper : token-level timestamp refactoring ( #49 , #120 )
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This turned out pretty good overall. The algorithm has been moved from
main.cpp to whisper.cpp and can be reused for all subtitles types. This
means that now you can specify the maximum length of the generated
lines. Simply provide the "-ml" argument specifying the max length in
number of characters
2022-11-02 21:45:54 +02:00
Georgi Gerganov
6fb98370ba
main : add some comments for the word-level timestamp algorithm
2022-11-01 22:35:21 +02:00
Georgi Gerganov
0729da9a3b
main : fix some edge cases for word-level timestamps
2022-11-01 22:09:25 +02:00
