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
2 years ago
a2ecd54455
models : add instructions for using HF fine-tuned models
2 years ago
128aaadb93
whisper : improve printfs
2 years ago
454b91de16
main : fix dangling pointer when using stdin for input ( #65 )
2 years ago
d7024cf9dc
main, stream : remove --verbose flag ( #178 )
2 years ago
37422ed733
talk.wasm : add audio pre-processing + bump memory
2 years ago
be3b720f96
talk.wasm : refactoring + update README.md
2 years ago
00f46dbc1d
models : add usage comments to the HF convert script ( #157 )
2 years ago
5698bddbc9
models : fix HF fine-tuned model conversion script ( #157 )
...
It works now
2 years ago
388e9f79ad
ggml : fix the fix
2 years ago
35cd29ce1f
ggml : fix cross-compile Linux -> Window with mingw ( #168 )
2 years ago
a156a358ca
Revert "update README.md"
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This reverts commit 6a84147113
2 years ago
6a84147113
update README.md
2 years ago
804f36aa2c
ggml: change inline ggml_fp16_to_fp32, ggml_fp16_t ggml_fp32_to_fp16
2 years ago
4b2f51b479
add gprof option
2 years ago
800ae5b808
fix AVX,AVX2,FMA,F16C detection on Linux and add flags for OpenBLAS
2 years ago
83456076f0
add AVX support
2 years ago
3df6c14fca
Build with OpenBLAS and SDL2 on windows
2 years ago
d64d6ca3fd
models : minor changes to the HF convert script ( #157 )
2 years ago
93482d0373
models : add "convert-h5-to-ggml.py" script ( #157 )
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Converts transformers models to ggml.
Although the conversion is successful, it does not work for some reason.
Not sure why
2 years ago
49706a658a
minor : updates few prints + fix buttons in whisper.wasm
2 years ago
363a2dadec
Update README.md
2 years ago
623a486056
Update README.md
2 years ago
2f596f5b33
Find libopenblas.dll.a on windows
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"lib" is needed for windows.
With this change, you can build whisper.cpp with OpenBLAS's prebuilt DLL.
1. extract a zip from https://github.com/xianyi/OpenBLAS/releases
2. copy the headers in (openblas)/include to the root directory of whisper.cpp
3. invoke cmake with -DCMAKE_LIBRARY_PATH=(openblas)\lib -DWHISPER_SUPPORT_OPENBLAS=ON
4. copy (openblas)/bin/libopenblas.dll to the same directory of whisper.dll after msbuild
https://github.com/ggerganov/whisper.cpp/issues/89#issuecomment-1324391258
2 years ago
e5dcdabbb8
unicode : fix character replacement (thanks to @tamo)
2 years ago
dad109c3f1
close #109 : add fetching of the model over HTTP (whisper.wasm)
2 years ago
326573de9a
talk.wasm : final touches
2 years ago
9aea96f774
talk.wasm : polishing + adding many AI personalities
2 years ago
385236d1d3
stream : "-kc" now enables context keeping from previous segment ( #90 )
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By default, the context keeping is disabled
2 years ago
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
2 years ago
78116f8eda
talk.wasm : update README.md
2 years ago
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
2 years ago
2e311a2917
Update README.md
2 years ago
2065572a11
ggml : fix Windows build
2 years ago
5c2176e314
ci : add Windows build
2 years ago
f2df9bd768
stream : add "max_tokens" cli arg
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Controls the max tokens per segment for the stream example
2 years ago
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.
2 years ago
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
2 years ago
d351771a4b
stream : add "single_segment" option
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Force the entire audio chunk to be transcribed into a single segment
2 years ago
c058aaf22e
stream : partial encoder experiments
2 years ago
2ba66360c9
fix: free ggml_context ( close #149 ) ( #150 )
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* fix: free ggml_context
* ggml : free the model's contexts in whisper_free()
Co-authored-by: Georgi Gerganov <ggerganov@gmail.com>
2 years ago
e70e5c8b53
models : simplify the conversion script
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"transformers" dependency is not actually needed
2 years ago
55a0e1a64e
Update download-ggml-model.sh
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follow curl redirect to new hosting site
2 years ago
864a78a8d0
models : change default hosting to Hugging Face
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My Linode is running out of monthly bandwidth due to the big interest in
the project
2 years ago
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.
2 years ago
41b48ab7f1
make : add libwhisper.so target ( #144 )
2 years ago
a728be9cdb
Add WHISPER_NO_AVX and WHISPER_NO_AVX2 to CMakeLists ( #136 )
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* Check for AVX and AVX2 on Darwin
* Add AVX options to CMakeLists
2 years ago
46a68fb9b5
minor : remove one more redundant line
2 years ago
ccd56a9c5b
minor : fix double float32 conversion in python script
2 years ago
3500ce8727
ref #40 : start working on the documentation
2 years ago
Georgi Gerganov