Automatic Speech Recognition
Transformers
Safetensors
PyTorch
mini_whisper
text2text-generation
custom_code
Instructions to use NeuraCraft/MiniWhisper-ASR with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use NeuraCraft/MiniWhisper-ASR with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("automatic-speech-recognition", model="NeuraCraft/MiniWhisper-ASR", trust_remote_code=True)# Load model directly from transformers import AutoModelForSeq2SeqLM model = AutoModelForSeq2SeqLM.from_pretrained("NeuraCraft/MiniWhisper-ASR", trust_remote_code=True, device_map="auto") - Notebooks
- Google Colab
- Kaggle
File size: 9,738 Bytes
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import numpy as np
import torch
class MiniWhisperFeatureExtractor:
"""Feature extractor matching Whisper exactly."""
model_input_names = ["input_features"]
def __init__(
self,
feature_size=80,
sampling_rate=16000,
hop_length=160,
chunk_length=30,
n_fft=400,
padding_value=0.0,
dither=0.0,
return_attention_mask=False,
**kwargs,
):
self.feature_size = feature_size
self.sampling_rate = sampling_rate
self.hop_length = hop_length
self.chunk_length = chunk_length
self.n_fft = n_fft
self.n_samples = chunk_length * sampling_rate
self.nb_max_frames = self.n_samples // hop_length
self.padding_value = padding_value
self.dither = dither
self.return_attention_mask = return_attention_mask
self.mel_filters = self._mel_filter_bank()
def _mel_filter_bank(self):
"""Create mel filter bank matching Whisper's parameters."""
try:
from torchaudio.functional import melscale_fbanks
n_freq_bins = 1 + self.n_fft // 2
mel_filters = melscale_fbanks(
n_freqs=n_freq_bins,
f_min=0.0,
f_max=8000.0,
n_mels=self.feature_size,
sample_rate=self.sampling_rate,
norm="slaney",
mel_scale="slaney",
)
return mel_filters.numpy()
except ImportError:
return self._numpy_mel_filter_bank()
def _numpy_mel_filter_bank(self):
"""NumPy fallback for mel filter bank."""
def hz_to_mel(hz):
return 2595 * np.log10(1 + hz / 700)
def mel_to_hz(mel):
return 700 * (10 ** (mel / 2595) - 1)
n_freq_bins = 1 + self.n_fft // 2
fmin, fmax = 0.0, 8000.0
mel_min = hz_to_mel(fmin)
mel_max = hz_to_mel(fmax)
mel_points = np.linspace(mel_min, mel_max, self.feature_size + 2)
hz_points = mel_to_hz(mel_points)
bin_points = np.floor((self.n_fft + 1) * hz_points / self.sampling_rate).astype(int)
filters = np.zeros((self.feature_size, n_freq_bins))
for i in range(self.feature_size):
left, center, right = bin_points[i], bin_points[i + 1], bin_points[i + 2]
for j in range(left, center):
filters[i, j] = (j - left) / (center - left)
for j in range(center, right):
filters[i, j] = (right - j) / (right - center)
filters = filters / (filters.sum(axis=1, keepdims=True) + 1e-10)
return filters
def _np_extract_fbank_features(self, waveform_batch, device="cpu"):
"""NumPy implementation matching Whisper exactly."""
log_spec_batch = []
for waveform in waveform_batch:
# Apply dither
if self.dither != 0.0:
waveform = waveform + self.dither * np.random.randn(*waveform.shape).astype(np.float32)
# Compute STFT
n_frames = 1 + (len(waveform) - self.n_fft) // self.hop_length
spec = np.zeros((self.n_fft // 2 + 1, n_frames), dtype=np.float32)
window = np.hanning(self.n_fft).astype(np.float32)
for i in range(n_frames):
start = i * self.hop_length
frame = waveform[start:start + self.n_fft]
if len(frame) < self.n_fft:
frame = np.pad(frame, (0, self.n_fft - len(frame)))
spec[:, i] = np.abs(np.fft.rfft(frame * window)) ** 2
# Apply mel filters: mel_filters is [n_mels, n_freqs]
mel_spec = self.mel_filters @ spec
# Log10 with floor
log_spec = np.log10(np.maximum(mel_spec, 1e-10))
# Whisper uses: np.maximum(log_spec, log_spec.max() - 8.0)
log_spec = np.maximum(log_spec, log_spec.max() - 8.0)
# Normalize: (log_spec + 4.0) / 4.0
log_spec = (log_spec + 4.0) / 4.0
# Remove last frame to match Whisper
log_spec = log_spec[:, :-1]
log_spec_batch.append(log_spec)
return np.array(log_spec_batch)
def _torch_extract_fbank_features(self, waveform, device="cpu"):
"""PyTorch implementation matching Whisper exactly."""
waveform = torch.from_numpy(waveform).to(device, torch.float32)
window = torch.hann_window(self.n_fft, device=device)
if self.dither != 0.0:
waveform += self.dither * torch.randn(waveform.shape, dtype=waveform.dtype, device=waveform.device)
stft = torch.stft(waveform, self.n_fft, self.hop_length, window=window, return_complex=True)
magnitudes = stft[..., :-1].abs() ** 2 # Remove last frame
# mel_filters shape: [n_mels, n_freqs] from mel_filter_bank
mel_filters = torch.from_numpy(self.mel_filters).to(device, torch.float32)
# Transpose and matmul: mel_filters.T @ magnitudes
mel_spec = mel_filters.T @ magnitudes
log_spec = torch.clamp(mel_spec, min=1e-10).log10()
if waveform.dim() == 2:
max_val = log_spec.max(dim=2, keepdim=True)[0].max(dim=1, keepdim=True)[0]
log_spec = torch.maximum(log_spec, max_val - 8.0)
else:
log_spec = torch.maximum(log_spec, log_spec.max() - 8.0)
log_spec = (log_spec + 4.0) / 4.0
if device != "cpu":
log_spec = log_spec.detach().cpu()
return log_spec.numpy()
@staticmethod
def zero_mean_unit_var_norm(input_values, attention_mask, padding_value=0.0):
if attention_mask is not None:
attention_mask = np.array(attention_mask, np.int32)
normed = []
for vector, length in zip(input_values, attention_mask.sum(-1)):
normed_slice = (vector - vector[:length].mean()) / (np.sqrt(vector[:length].var() + 1e-7))
if length < normed_slice.shape[0]:
normed_slice[length:] = padding_value
normed.append(normed_slice)
else:
normed = [(x - x.mean()) / (np.sqrt(x.var() + 1e-7)) for x in input_values]
return normed
def __call__(
self,
raw_speech,
sampling_rate=None,
truncation=True,
return_tensors="pt",
return_attention_mask=None,
do_normalize=False,
**kwargs,
):
if sampling_rate is not None and sampling_rate != self.sampling_rate:
raise ValueError(f"Expected {self.sampling_rate}, got {sampling_rate}")
is_batched = isinstance(raw_speech, list) or (isinstance(raw_speech, np.ndarray) and raw_speech.ndim > 1)
if not is_batched:
raw_speech = [raw_speech]
# Convert to numpy float32
processed = []
for audio in raw_speech:
if isinstance(audio, np.ndarray) and audio.dtype == np.float64:
audio = audio.astype(np.float32)
elif not isinstance(audio, np.ndarray):
audio = np.array(audio, dtype=np.float32)
# Pad or truncate
if len(audio) > self.n_samples:
if truncation:
audio = audio[:self.n_samples]
elif len(audio) < self.n_samples:
audio = np.pad(audio, (0, self.n_samples - len(audio)), constant_values=self.padding_value)
processed.append(audio)
waveform_batch = np.stack(processed, axis=0)
# Use torch if available
if torch.cuda.is_available():
input_features = self._torch_extract_fbank_features(waveform_batch, device="cuda")
# Result is [batch, n_mels, n_frames]
else:
input_features = self._np_extract_fbank_features(waveform_batch)
result = {"input_features": input_features}
if return_attention_mask is None:
return_attention_mask = self.return_attention_mask
if return_attention_mask:
result["attention_mask"] = np.ones(input_features.shape[0], dtype=np.int64)
if return_tensors == "pt":
result["input_features"] = torch.from_numpy(input_features).float()
if "attention_mask" in result:
result["attention_mask"] = torch.from_numpy(result["attention_mask"]).long()
return result
def to_dict(self):
return {
"feature_size": self.feature_size,
"sampling_rate": self.sampling_rate,
"hop_length": self.hop_length,
"n_fft": self.n_fft,
"chunk_length": self.chunk_length,
"padding_value": self.padding_value,
"dither": self.dither,
}
def save_pretrained(self, save_directory):
import json, os
os.makedirs(save_directory, exist_ok=True)
with open(os.path.join(save_directory, "preprocessor_config.json"), "w") as f:
json.dump(self.to_dict(), f, indent=2)
@classmethod
def from_pretrained(cls, save_directory):
import json, os
path = os.path.join(save_directory, "preprocessor_config.json")
if os.path.exists(path):
with open(path) as f:
return cls(**json.load(f))
return cls()
__all__ = ["MiniWhisperFeatureExtractor"] |