| { |
| "models": [ |
| // Configuration for the built-in models. You can remove any of these |
| // if you don't want to use the default models. |
| { |
| "name": "tiny", |
| "url": "tiny" |
| }, |
| { |
| "name": "base", |
| "url": "base" |
| }, |
| { |
| "name": "small", |
| "url": "small" |
| }, |
| { |
| "name": "medium", |
| "url": "medium" |
| }, |
| { |
| "name": "large", |
| "url": "large" |
| }, |
| { |
| "name": "large-v1", |
| "url": "large-v1" |
| }, |
| { |
| "name": "large-v2", |
| "url": "large-v2" |
| }, |
| { |
| "name": "large-v3", |
| "url": "large-v3" |
| }, |
| // Uncomment to add custom Japanese models |
| //{ |
| // "name": "whisper-large-v2-mix-jp", |
| // "url": "vumichien/whisper-large-v2-mix-jp", |
| // // The type of the model. Can be "huggingface" or "whisper" - "whisper" is the default. |
| // // HuggingFace models are loaded using the HuggingFace transformers library and then converted to Whisper models. |
| // "type": "huggingface", |
| //}, |
| //{ |
| // "name": "local-model", |
| // "url": "path/to/local/model", |
| //}, |
| //{ |
| // "name": "remote-model", |
| // "url": "https://example.com/path/to/model", |
| //} |
| ], |
| // Configuration options that will be used if they are not specified in the command line arguments. |
|
|
| // * WEBUI options * |
|
|
| // Maximum audio file length in seconds, or -1 for no limit. Ignored by CLI. |
| "input_audio_max_duration": -1, |
| // True to share the app on HuggingFace. |
| "share": false, |
| // The host or IP to bind to. If None, bind to localhost. |
| "server_name": null, |
| // The port to bind to. |
| "server_port": 7860, |
| // The number of workers to use for the web server. Use -1 to disable queueing. |
| "queue_concurrency_count": 1, |
| // Whether or not to automatically delete all uploaded files, to save disk space |
| "delete_uploaded_files": true, |
|
|
| // * General options * |
|
|
| // The default implementation to use for Whisper. Can be "whisper" or "faster-whisper". |
| // Note that you must either install the requirements for faster-whisper (requirements-fasterWhisper.txt) |
| // or whisper (requirements.txt) |
| "whisper_implementation": "whisper", |
|
|
| // The default model name. |
| "default_model_name": "medium", |
| // The default VAD. |
| "default_vad": "silero-vad", |
| // A commma delimited list of CUDA devices to use for parallel processing. If None, disable parallel processing. |
| "vad_parallel_devices": "", |
| // The number of CPU cores to use for VAD pre-processing. |
| "vad_cpu_cores": 1, |
| // The number of seconds before inactivate processes are terminated. Use 0 to close processes immediately, or None for no timeout. |
| "vad_process_timeout": 1800, |
| // True to use all available GPUs and CPU cores for processing. Use vad_cpu_cores/vad_parallel_devices to specify the number of CPU cores/GPUs to use. |
| "auto_parallel": false, |
| // Directory to save the outputs (CLI will use the current directory if not specified) |
| "output_dir": null, |
| // The path to save model files; uses ~/.cache/whisper by default |
| "model_dir": null, |
| // Device to use for PyTorch inference, or Null to use the default device |
| "device": null, |
| // Whether to print out the progress and debug messages |
| "verbose": true, |
| // Whether to perform X->X speech recognition ('transcribe') or X->English translation ('translate') |
| "task": "transcribe", |
| // Language spoken in the audio, specify None to perform language detection |
| "language": null, |
| // The window size (in seconds) to merge voice segments |
| "vad_merge_window": 5, |
| // The maximum size (in seconds) of a voice segment |
| "vad_max_merge_size": 30, |
| // The padding (in seconds) to add to each voice segment |
| "vad_padding": 1, |
| // Whether or not to prepend the initial prompt to each VAD segment (prepend_all_segments), or just the first segment (prepend_first_segment) |
| "vad_initial_prompt_mode": "prepend_first_segment", |
| // The window size of the prompt to pass to Whisper |
| "vad_prompt_window": 3, |
| // Temperature to use for sampling |
| "temperature": 0, |
| // Number of candidates when sampling with non-zero temperature |
| "best_of": 5, |
| // Number of beams in beam search, only applicable when temperature is zero |
| "beam_size": 5, |
| // Optional patience value to use in beam decoding, as in https://arxiv.org/abs/2204.05424, the default (1.0) is equivalent to conventional beam search |
| "patience": 1, |
| // Optional token length penalty coefficient (alpha) as in https://arxiv.org/abs/1609.08144, uses simple length normalization by default |
| "length_penalty": null, |
| // Comma-separated list of token ids to suppress during sampling; '-1' will suppress most special characters except common punctuations |
| "suppress_tokens": "-1", |
| // Optional text to provide as a prompt for the first window |
| "initial_prompt": null, |
| // If True, provide the previous output of the model as a prompt for the next window; disabling may make the text inconsistent across windows, but the model becomes less prone to getting stuck in a failure loop |
| "condition_on_previous_text": true, |
| // Whether to perform inference in fp16; True by default |
| "fp16": true, |
| // The compute type used by faster-whisper. Can be "int8". "int16" or "float16". |
| "compute_type": "auto", |
| // Temperature to increase when falling back when the decoding fails to meet either of the thresholds below |
| "temperature_increment_on_fallback": 0.2, |
| // If the gzip compression ratio is higher than this value, treat the decoding as failed |
| "compression_ratio_threshold": 2.4, |
| // If the average log probability is lower than this value, treat the decoding as failed |
| "logprob_threshold": -1.0, |
| // If the probability of the <no-speech> token is higher than this value AND the decoding has failed due to `logprob_threshold`, consider the segment as silence |
| "no_speech_threshold": 0.6, |
|
|
| // (experimental) extract word-level timestamps and refine the results based on them |
| "word_timestamps": false, |
| // if word_timestamps is True, merge these punctuation symbols with the next word |
| "prepend_punctuations": "\"\'“¿([{-", |
| // if word_timestamps is True, merge these punctuation symbols with the previous word |
| "append_punctuations": "\"\'.。,，!！?？:：”)]}、", |
| // (requires --word_timestamps True) underline each word as it is spoken in srt and vtt |
| "highlight_words": false, |
|
|
| // Diarization settings |
| "auth_token": null, |
| // Whether to perform speaker diarization |
| "diarization": false, |
| // The number of speakers to detect |
| "diarization_speakers": 2, |
| // The minimum number of speakers to detect |
| "diarization_min_speakers": 1, |
| // The maximum number of speakers to detect |
| "diarization_max_speakers": 8, |
| // The number of seconds before inactivate processes are terminated. Use 0 to close processes immediately, or None for no timeout. |
| "diarization_process_timeout": 60, |
| } |
|
|
