|
import torch |
|
from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline |
|
from datasets import load_dataset |
|
pip install accelerate |
|
|
|
device = "cuda:0" if torch.cuda.is_available() else "cpu" |
|
torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32 |
|
|
|
model_id = "openai/whisper-large-v3" |
|
|
|
model = AutoModelForSpeechSeq2Seq.from_pretrained( |
|
model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True |
|
) |
|
model.to(device) |
|
|
|
processor = AutoProcessor.from_pretrained(model_id) |
|
|
|
pipe = pipeline( |
|
"automatic-speech-recognition", |
|
model=model, |
|
tokenizer=processor.tokenizer, |
|
feature_extractor=processor.feature_extractor, |
|
max_new_tokens=128, |
|
chunk_length_s=30, |
|
batch_size=16, |
|
return_timestamps=True, |
|
torch_dtype=torch_dtype, |
|
device=device, |
|
) |
|
|
|
dataset = load_dataset("distil-whisper/librispeech_long", "clean", split="validation") |
|
sample = dataset[0]["audio"] |
|
|
|
result = pipe(sample) |
|
print(result["text"]) |
|
|