| from transformers import ( |
| WhisperForConditionalGeneration, |
| WhisperProcessor, |
| WhisperConfig, |
| ) |
| import torch |
| import ffmpeg |
| import torch |
| import torch.nn.functional as F |
| import numpy as np |
| import os |
|
|
| |
| SAMPLE_RATE = 16000 |
| CHUNK_LENGTH = 30 |
| N_SAMPLES = CHUNK_LENGTH * SAMPLE_RATE |
|
|
|
|
| |
| def load_audio(file: str, sr: int = SAMPLE_RATE, start_time: int = 0, dtype=np.float16): |
| """ |
| Load an audio file into a numpy array at the specified sampling rate. |
| """ |
| try: |
| |
| |
| out, _ = ( |
| ffmpeg.input(file, ss=start_time, threads=0) |
| .output("-", format="s16le", acodec="pcm_s16le", ac=1, ar=sr) |
| .run(cmd=["ffmpeg", "-nostdin"], capture_stdout=True, capture_stderr=True) |
| ) |
| except ffmpeg.Error as e: |
| raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e |
|
|
| |
| return np.frombuffer(out, np.int16).flatten().astype(dtype) / 32768.0 |
|
|
|
|
| |
| def pad_or_trim(array, length: int = N_SAMPLES, *, axis: int = -1): |
| """ |
| Pad or trim the audio array to N_SAMPLES, as expected by the encoder. |
| """ |
| if torch.is_tensor(array): |
| if array.shape[axis] > length: |
| array = array.index_select( |
| dim=axis, index=torch.arange(length, device=array.device) |
| ) |
|
|
| if array.shape[axis] < length: |
| pad_widths = [(0, 0)] * array.ndim |
| pad_widths[axis] = (0, length - array.shape[axis]) |
| array = F.pad(array, [pad for sizes in pad_widths[::-1] for pad in sizes]) |
| else: |
| if array.shape[axis] > length: |
| array = array.take(indices=range(length), axis=axis) |
|
|
| if array.shape[axis] < length: |
| pad_widths = [(0, 0)] * array.ndim |
| pad_widths[axis] = (0, length - array.shape[axis]) |
| array = np.pad(array, pad_widths) |
|
|
| return array |
|
|
|
|
| class Model: |
| def __init__( |
| self, |
| model_name_or_path: str, |
| cuda_visible_device: str = "0", |
| device: str = "cuda", |
| ): |
| os.environ["CUDA_VISIBLE_DEVICES"] = cuda_visible_device |
| self.DEVICE = device |
|
|
| self.processor = WhisperProcessor.from_pretrained(model_name_or_path) |
| self.tokenizer = self.processor.tokenizer |
|
|
| self.config = WhisperConfig.from_pretrained(model_name_or_path) |
|
|
| self.model = WhisperForConditionalGeneration( |
| config=self.config |
| ).from_pretrained( |
| pretrained_model_name_or_path=model_name_or_path, |
| torch_dtype=self.config.torch_dtype, |
| |
| low_cpu_mem_usage=True, |
| ) |
|
|
| |
| if self.model.device.type != self.DEVICE: |
| print(f"Moving model to {self.DEVICE}") |
| self.model = self.model.to(self.DEVICE) |
| self.model.eval() |
|
|
| else: |
| print(f"Model is already on {self.DEVICE}") |
| self.model.eval() |
|
|
| print("dtype of model acc to config: ", self.config.torch_dtype) |
| print("dtype of loaded model: ", self.model.dtype) |
|
|
| def transcribe( |
| self, audio, language: str = "english", skip_special_tokens: bool = True |
| ) -> str: |
| input_features = ( |
| self.processor(audio, sampling_rate=SAMPLE_RATE, return_tensors="pt") |
| .input_features.half() |
| .to(self.DEVICE) |
| ) |
| with torch.no_grad(): |
| predicted_ids = self.model.generate( |
| input_features, |
| num_beams=1, |
| language=language, |
| task="transcribe", |
| use_cache=True, |
| is_multilingual=True, |
| return_timestamps=True, |
| ) |
|
|
| transcription = self.tokenizer.batch_decode( |
| predicted_ids, skip_special_tokens=skip_special_tokens |
| )[0] |
| return transcription.strip() |
|
|