使用 AutoModelForSpeechSeq2Seq 和 AutoProcessor 加载预训练的 Whisper 模型,并通过 pipeline 简化自动语音识别流程。的代码
时间: 2024-10-12 14:04:51 浏览: 25
在Hugging Face的Transformers库中,你可以使用`AutoModelForSpeechSeq2Seq`和`AutoProcessor`来加载预训练的Whisper模型并简化语音识别过程。以下是一个简化的Python代码示例:
```python
from transformers import WhisperTokenizerFast, WhisperForConditionalGeneration
# 加载预训练的Whisper模型和处理器
model_name = "whisper" # 更改为实际的模型名,如"facebook/whisper-base"
tokenizer = WhisperTokenizerFast.from_pretrained(model_name)
model = WhisperForConditionalGeneration.from_pretrained(model_name)
# 定义管道,用于音频到文本的转换
class SpeechToTextPipeline:
def __init__(self):
self.tokenizer = tokenizer
self.model = model
def __call__(self, audio_file, *args, **kwargs):
# 使用AudioSegment处理音频文件
from pydub import AudioSegment
audio = AudioSegment.from_file(audio_file, format="wav")
# 将音频转为Mel spectrogram
mel_spectrogram = ... # 这部分依赖于你的音频处理库,比如torchaudio或librosa
# 对Mel spectrogram编码成输入序列
input_ids = self.tokenizer(mel_spectrogram, return_tensors='pt', padding=True).input_ids
generated_text = self.model.generate(input_ids)[0] # 生成文本
# 文本解码
text = self.tokenizer.decode(generated_text, skip_special_tokens=True)
return text
# 创建pipeline实例并进行识别
pipeline = SpeechToTextPipeline()
transcribed_text = pipeline("path_to_your_audio.wav")
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