import torch
from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
# config
model_id = "kotoba-tech/kotoba-whisper-v1.0"
torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32
device = "cuda:0" if torch.cuda.is_available() else "cpu"
# load model
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,
torch_dtype=torch_dtype,
device=device,
)
audio_file = "./abc.mp3"
result = pipe(audio_file)
print(result["text"])
#音声ファイルから日本語を抽出
#ここではabc.mp3を用意
import torch
from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
from datasets import load_dataset, Audio
# config
model_id = "kotoba-tech/kotoba-whisper-v1.0"
torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32
device = "cuda:0" if torch.cuda.is_available() else "cpu"
# load model
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,
torch_dtype=torch_dtype,
device=device,
)
# load sample audio & downsample to 16kHz
dataset = load_dataset("japanese-asr/ja_asr.reazonspeech_test", split="test")
dataset = dataset.cast_column("audio", Audio(sampling_rate=16000))
sample = dataset[0]["audio"]
# run inference
result = pipe(sample)
print(result["text"])
日本語音声認識に特化したWhisperである kotoba-whisper-v1.0を早速試してみた | DevelopersIO
はじめに 昨日公開された日本語音声認識に特化した「Kotoba-Whisper」のkotoba-whisper-v1.0を試してみました。 本モデルは、OpenAIの「Whisper large-v3」を教師モデルとして …