学习目标：如何使用whisper

---

学习内容一：whisper 转文字

1.1 使用whisper.load_model()方法下载，加载

model=whisper.load_model(参数)

• name 需要加载的模型，如上图
• device：默认有个方法，有显存使用显存，没有使用cpu
• download_root:下载的根目录，默认使用~/.cache/whisper
• in_memory： 是否将模型权重预加载到主机内存中

---

返回值
model : Whisper
Whisper语音识别模型实例

def load_model(name: str,device: Optional[Union[str, torch.device]] = None,download_root: str = None,in_memory: bool = False,
) -> Whisper:"""Load a Whisper ASR modelParameters----------name : strone of the official model names listed by `whisper.available_models()`, orpath to a model checkpoint containing the model dimensions and the model state_dict.device : Union[str, torch.device]the PyTorch device to put the model intodownload_root: strpath to download the model files; by default, it uses "~/.cache/whisper"in_memory: boolwhether to preload the model weights into host memoryReturns-------model : WhisperThe Whisper ASR model instance"""if device is None:device = "cuda" if torch.cuda.is_available() else "cpu"if download_root is None:default = os.path.join(os.path.expanduser("~"), ".cache")download_root = os.path.join(os.getenv("XDG_CACHE_HOME", default), "whisper")if name in _MODELS:checkpoint_file = _download(_MODELS[name], download_root, in_memory)alignment_heads = _ALIGNMENT_HEADS[name]elif os.path.isfile(name):checkpoint_file = open(name, "rb").read() if in_memory else namealignment_heads = Noneelse:raise RuntimeError(f"Model {name} not found; available models = {available_models()}")with (io.BytesIO(checkpoint_file) if in_memory else open(checkpoint_file, "rb")) as fp:checkpoint = torch.load(fp, map_location=device)del checkpoint_filedims = ModelDimensions(**checkpoint["dims"])model = Whisper(dims)model.load_state_dict(checkpoint["model_state_dict"])if alignment_heads is not None:model.set_alignment_heads(alignment_heads)return model.to(device)

1.2 使用实例对文件进行转录

result = model.transcribe(file_path)

def transcribe(model: "Whisper",audio: Union[str, np.ndarray, torch.Tensor],*,verbose: Optional[bool] = None,temperature: Union[float, Tuple[float, ...]] = (0.0, 0.2, 0.4, 0.6, 0.8, 1.0),compression_ratio_threshold: Optional[float] = 2.4,logprob_threshold: Optional[float] = -1.0,no_speech_threshold: Optional[float] = 0.6,condition_on_previous_text: bool = True,initial_prompt: Optional[str] = None,word_timestamps: bool = False,prepend_punctuations: str = "\"'“¿([{-",append_punctuations: str = "\"'.。,，!！?？:：”)]}、",**decode_options,
):"""将音频转换为文本。参数：- model: Whisper模型- audio: 音频文件路径、NumPy数组或PyTorch张量- verbose: 是否打印详细信息，默认为None- temperature: 温度参数，默认为(0.0, 0.2, 0.4, 0.6, 0.8, 1.0)- compression_ratio_threshold: 压缩比阈值，默认为2.4- logprob_threshold: 对数概率阈值，默认为-1.0- no_speech_threshold: 无语音信号阈值，默认为0.6- condition_on_previous_text: 是否根据先前的文本进行解码，默认为True- initial_prompt: 初始提示，默认为None- word_timestamps: 是否返回单词时间戳，默认为False- prepend_punctuations: 前缀标点符号，默认为"\"'“¿([{-"- append_punctuations: 后缀标点符号，默认为"\"'.。,，!！?？:：”)]}、"- **decode_options: 其他解码选项返回：- 转录得到的文本"""

1.3 实战

建议load_model添加参数

• download_root:下载的根目录，默认使用~/.cache/whisper
  transcribe方法添加参数
• word_timestamps=True

import whisper
import arrow# 定义模型、音频地址、录音开始时间
def excute(model_name,file_path,start_time):model = whisper.load_model(model_name)result = model.transcribe(file_path,word_timestamps=True)for segment in result["segments"]:now = arrow.get(start_time)start = now.shift(seconds=segment["start"]).format("YYYY-MM-DD HH:mm:ss")end = now.shift(seconds=segment["end"]).format("YYYY-MM-DD HH:mm:ss")print("【"+start+"->" +end+"】："+segment["text"])if __name__ == '__main__':excute("large","/root/autodl-tmp/no/test.mp3","2022-10-24 16:23:00")

学习内容二：语者分离（pyannote.audio）pyannote.audio是huggingface开源音色包

第一步：安装依赖

pip install pyannote.audio

第二步：创建key

https://huggingface.co/settings/tokens

第三步：测试pyannote.audio

• 创建实例：Pipeline.from_pretrained(参数)
• 使用GPU加速：import torch # 导入torch库
  pipeline.to(torch.device(“cuda”))
• 实例转化音频pipeline(“test.wav”)

---

from_pretrained(参数)

• cache_dir:路径或str，可选模型缓存目录的路径。默认/pyannote"当未设置时。

---

pipeline(参数)

• file_path：录音文件
• num_speakers：几个说话者，可以不带

from pyannote.audio import Pipeline
pipeline = Pipeline.from_pretrained("pyannote/speaker-diarization@2.1", use_auth_token="申请的key")# send pipeline to GPU (when available)
import torch
device='cuda' if torch.cuda.is_available() else 'cpu'
pipeline.to(torch.device(device))# apply pretrained pipeline
diarization = pipeline("test.wav")
print(diarization)
# print the result
for turn, _, speaker in diarization.itertracks(yield_label=True):print(f"start={turn.start:.1f}s stop={turn.end:.1f}s speaker_{speaker}")
# start=0.2s stop=1.5s speaker_0
# start=1.8s stop=3.9s speaker_1
# start=4.2s stop=5.7s speaker_0
# ...

学习内容三：整合

这里要借助一个开源代码，用于整合以上两种产生的结果

报错No module named 'pyannote_whisper'
如果你使用使用AutoDL平台，你可以使用学术代理加速

source /etc/network_turbo

git clone https://github.com/yinruiqing/pyannote-whisper.git
cd pyannote-whisper
pip install -r requirements.txt

这个错误可能是由于缺少或不正确安装了所需的 sndfile 库。sndfile 是一个用于处理音频文件的库，它提供了多种格式的读写支持。

你可以尝试安装 sndfile 库，方法如下：

在 Ubuntu 上，使用以下命令安装：sudo apt-get install libsndfile1-dev
在 CentOS 上，使用以下命令安装：sudo yum install libsndfile-devel
在 macOS 上，使用 Homebrew 安装：brew install libsndfile
然后重新执行如上指令

在项目里面写代码就可以了,或者复制代码里面的pyannote_whisper.utils模块代码

import os
import whisper
from pyannote.audio import Pipeline
from pyannote_whisper.utils import diarize_text
import concurrent.futures
import subprocess
import torch
print("正在加载声纹模型")
pipeline = Pipeline.from_pretrained("pyannote/speaker-diarization@2.1",use_auth_token="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")
output_dir = '/root/autodl-tmp/no/out'
print("正在whisper模型")
model = whisper.load_model("large", device="cuda")# MP3转化为wav
def convert_to_wav(path):new_path = ''if path[-3:] != 'wav':new_path = '.'.join(path.split('.')[:-1]) + '.wav'try:subprocess.call(['ffmpeg', '-i', path, new_path, '-y', '-an'])except:return path, 'Error: Could not convert file to .wav'else:new_path = ''return new_path, Nonedef process_audio(file_path):file_path, retmsg = convert_to_wav(file_path)print(f"===={file_path}=======")asr_result = model.transcribe(file_path, initial_prompt="语音转换")pipeline.to(torch.device('cuda'))diarization_result = pipeline(file_path, num_speakers=2)final_result = diarize_text(asr_result, diarization_result)output_file = os.path.join(output_dir, os.path.basename(file_path)[:-4] + '.txt')with open(output_file, 'w') as f:for seg, spk, sent in final_result:line = f'{seg.start:.2f} {seg.end:.2f} {spk} {sent}\n'f.write(line)if not os.path.exists(output_dir):os.makedirs(output_dir)wave_dir = '/root/autodl-tmp/no'# 获取当前目录下所有wav文件名
wav_files = [os.path.join(wave_dir, file) for file in os.listdir(wave_dir) if file.endswith('.mp3')]# 处理每个wav文件
# with concurrent.futures.ThreadPoolExecutor(max_workers=1) as executor:
#     executor.map(process_audio, wav_files)
for wav_file in wav_files:process_audio(wav_file)
print('处理完成！')