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openai/whisper-small

Whisper is a pre-trained model for automatic speech recognition (ASR) and speech translation. It was trained on 680k hours of labelled data and demonstrates a strong ability to generalize to many datasets and domains without the need for fine-tuning. The model is based on a Transformer architecture and uses a large-scale weak supervision technique.

Whisper is a pre-trained model for automatic speech recognition (ASR) and speech translation. It was trained on 680k hours of labelled data and demonstrates a strong ability to generalize to many datasets and domains without the need for fine-tuning. The model is based on a Transformer architecture and uses a large-scale weak supervision technique.

Public
$0.0005/sec

HTTP/cURL API

 

Input fields

audiostring

audio to transcribe


taskstring

task to perform

Default value: transcribe

Allowed values: transcribetranslate


languagestring

language that the audio is in; uses detected language if None


temperaturenumber

temperature to use for sampling

Default value: 0


patiencenumber

patience value to use in beam decoding

Default value: 1


suppress_tokensstring

token ids to suppress during sampling

Default value: -1


initial_promptstring

optional text to provide as a prompt for the first window.


condition_on_previous_textboolean

provide the previous output of the model as a prompt for the next window

Default value: true


temperature_increment_on_fallbacknumber

temperature to increase when falling back when the decoding fails to meet either of the thresholds below

Default value: 0.2


compression_ratio_thresholdnumber

gzip compression ratio threshold

Default value: 2.4


logprob_thresholdnumber

average log probability threshold

Default value: -1


no_speech_thresholdnumber

probability of the <|nospeech|> token threshold

Default value: 0.6


webhookfile

The webhook to call when inference is done, by default you will get the output in the response of your inference request

Input Schema

Output Schema


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