mirror of
https://github.com/snakers4/silero-vad.git
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10
README.md
10
README.md
@@ -15,7 +15,7 @@ This repository also includes Number Detector and Language classifier [models](h
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<br/>
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<p align="center">
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<img src="https://user-images.githubusercontent.com/36505480/145563071-681b57e3-06b5-4cd0-bdee-e2ade3d50a60.png" />
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<img src="https://user-images.githubusercontent.com/36505480/198026365-8da383e0-5398-4a12-b7f8-22c2c0059512.png" />
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</p>
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<details>
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@@ -35,11 +35,11 @@ https://user-images.githubusercontent.com/36505480/144874384-95f80f6d-a4f1-42cc-
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- **Fast**
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One audio chunk (30+ ms) [takes](https://github.com/snakers4/silero-vad/wiki/Performance-Metrics#silero-vad-performance-metrics) around **1ms** to be processed on a single CPU thread. Using batching or GPU can also improve performance considerably. Under certain conditions ONNX may even run up to 2-3x faster.
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One audio chunk (30+ ms) [takes](https://github.com/snakers4/silero-vad/wiki/Performance-Metrics#silero-vad-performance-metrics) less than **1ms** to be processed on a single CPU thread. Using batching or GPU can also improve performance considerably. Under certain conditions ONNX may even run up to 4-5x faster.
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- **Lightweight**
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JIT model is less than one megabyte in size.
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JIT model is around one megabyte in size.
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- **General**
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@@ -47,11 +47,11 @@ https://user-images.githubusercontent.com/36505480/144874384-95f80f6d-a4f1-42cc-
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- **Flexible sampling rate**
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Silero VAD [supports](https://github.com/snakers4/silero-vad/wiki/Quality-Metrics#sample-rate-comparison) **8000 Hz** and **16000 Hz** (PyTorch JIT) and **16000 Hz** (ONNX) [sampling rates](https://en.wikipedia.org/wiki/Sampling_(signal_processing)#Sampling_rate).
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Silero VAD [supports](https://github.com/snakers4/silero-vad/wiki/Quality-Metrics#sample-rate-comparison) **8000 Hz** and **16000 Hz** [sampling rates](https://en.wikipedia.org/wiki/Sampling_(signal_processing)#Sampling_rate).
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- **Flexible chunk size**
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Model was trained on audio chunks of different lengths. **30 ms**, **60 ms** and **100 ms** long chunks are supported directly, others may work as well.
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Model was trained on **30 ms**. Longer chunks are supported directly, others may work as well.
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- **Highly Portable**
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68
utils_vad.py
68
utils_vad.py
@@ -9,7 +9,7 @@ languages = ['ru', 'en', 'de', 'es']
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class OnnxWrapper():
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def __init__(self, path, force_onnx_cpu):
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def __init__(self, path, force_onnx_cpu=False):
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import numpy as np
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global np
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import onnxruntime
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@@ -21,12 +21,9 @@ class OnnxWrapper():
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self.session.inter_op_num_threads = 1
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self.reset_states()
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self.sample_rates = [8000, 16000]
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def reset_states(self):
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self._h = np.zeros((2, 1, 64)).astype('float32')
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self._c = np.zeros((2, 1, 64)).astype('float32')
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def __call__(self, x, sr: int):
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def _validate_input(self, x, sr: int):
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if x.dim() == 1:
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x = x.unsqueeze(0)
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if x.dim() > 2:
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@@ -37,23 +34,62 @@ class OnnxWrapper():
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x = x[::step]
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sr = 16000
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if x.shape[0] > 1:
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raise ValueError("Onnx model does not support batching")
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if sr not in [16000]:
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raise ValueError(f"Supported sample rates: {[16000]}")
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if sr not in self.sample_rates:
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raise ValueError(f"Supported sampling rates: {self.sample_rates} (or multiply of 16000)")
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if sr / x.shape[1] > 31.25:
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raise ValueError("Input audio chunk is too short")
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ort_inputs = {'input': x.numpy(), 'h0': self._h, 'c0': self._c}
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ort_outs = self.session.run(None, ort_inputs)
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out, self._h, self._c = ort_outs
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return x, sr
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out = torch.tensor(out).squeeze(2)[:, 1] # make output type match JIT analog
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def reset_states(self, batch_size=1):
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self._h = np.zeros((2, batch_size, 64)).astype('float32')
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self._c = np.zeros((2, batch_size, 64)).astype('float32')
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self._last_sr = 0
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self._last_batch_size = 0
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def __call__(self, x, sr: int):
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x, sr = self._validate_input(x, sr)
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batch_size = x.shape[0]
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if not self._last_batch_size:
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self.reset_states(batch_size)
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if (self._last_sr) and (self._last_sr != sr):
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self.reset_states(batch_size)
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if (self._last_batch_size) and (self._last_batch_size != batch_size):
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self.reset_states(batch_size)
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if sr in [8000, 16000]:
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ort_inputs = {'input': x.numpy(), 'h': self._h, 'c': self._c, 'sr': np.array(sr)}
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ort_outs = self.session.run(None, ort_inputs)
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out, self._h, self._c = ort_outs
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else:
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raise ValueError()
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self._last_sr = sr
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self._last_batch_size = batch_size
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out = torch.tensor(out)
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return out
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def audio_forward(self, x, sr: int, num_samples: int = 512):
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outs = []
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x, sr = self._validate_input(x, sr)
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if x.shape[1] % num_samples:
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pad_num = num_samples - (x.shape[1] % num_samples)
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x = torch.nn.functional.pad(x, (0, pad_num), 'constant', value=0.0)
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self.reset_states(x.shape[0])
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for i in range(0, x.shape[1], num_samples):
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wavs_batch = x[:, i:i+num_samples]
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out_chunk = self.__call__(wavs_batch, sr)
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outs.append(out_chunk)
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stacked = torch.cat(outs, dim=1)
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return stacked.cpu()
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class Validator():
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def __init__(self, url, force_onnx_cpu):
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@@ -128,7 +164,7 @@ def get_speech_timestamps(audio: torch.Tensor,
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sampling_rate: int = 16000,
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min_speech_duration_ms: int = 250,
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min_silence_duration_ms: int = 100,
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window_size_samples: int = 1536,
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window_size_samples: int = 512,
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speech_pad_ms: int = 30,
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return_seconds: bool = False,
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visualize_probs: bool = False):
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