Automatic Speech Recognition
Transformers
PyTorch
whisper
audio
speech
wav2vec2
Eval Results (legacy)
Instructions to use devasheeshG/whisper_medium_fp16_transformers with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use devasheeshG/whisper_medium_fp16_transformers with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("automatic-speech-recognition", model="devasheeshG/whisper_medium_fp16_transformers")# Load model directly from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq processor = AutoProcessor.from_pretrained("devasheeshG/whisper_medium_fp16_transformers") model = AutoModelForSpeechSeq2Seq.from_pretrained("devasheeshG/whisper_medium_fp16_transformers") - Notebooks
- Google Colab
- Kaggle
| from transformers import ( | |
| WhisperForConditionalGeneration, | |
| WhisperProcessor, | |
| WhisperConfig, | |
| ) | |
| import torch | |
| import ffmpeg | |
| import torch | |
| import torch.nn.functional as F | |
| import numpy as np | |
| import os | |
| # load_audio and pad_or_trim functions | |
| SAMPLE_RATE = 16000 | |
| CHUNK_LENGTH = 30 # 30-second chunks | |
| N_SAMPLES = CHUNK_LENGTH * SAMPLE_RATE # 480000 samples in a 30-second chunk | |
| # audio = whisper.load_audio('test.wav') | |
| 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: | |
| # This launches a subprocess to decode audio while down-mixing and resampling as necessary. | |
| # Requires the ffmpeg CLI and `ffmpeg-python` package to be installed. | |
| 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(np.float32) / 32768.0 | |
| return np.frombuffer(out, np.int16).flatten().astype(dtype) / 32768.0 | |
| # audio = whisper.pad_or_trim(audio) | |
| 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", # torch.device("cuda" if torch.cuda.is_available() else "cpu") | |
| ): | |
| 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, | |
| # device_map=DEVICE, # 'balanced', 'balanced_low_0', 'sequential', 'cuda', 'cpu' | |
| low_cpu_mem_usage=True, | |
| ) | |
| # Move model to GPU | |
| 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() | |