Commit
•
9067154
1
Parent(s):
4a37dab
add code
Browse files- utils/vad.py +290 -0
utils/vad.py
ADDED
@@ -0,0 +1,290 @@
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1 |
+
import bisect
|
2 |
+
import functools
|
3 |
+
import os
|
4 |
+
import warnings
|
5 |
+
|
6 |
+
from typing import List, NamedTuple, Optional
|
7 |
+
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8 |
+
import numpy as np
|
9 |
+
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10 |
+
|
11 |
+
# The code below is adapted from https://github.com/snakers4/silero-vad.
|
12 |
+
class VadOptions(NamedTuple):
|
13 |
+
"""VAD options.
|
14 |
+
|
15 |
+
Attributes:
|
16 |
+
threshold: Speech threshold. Silero VAD outputs speech probabilities for each audio chunk,
|
17 |
+
probabilities ABOVE this value are considered as SPEECH. It is better to tune this
|
18 |
+
parameter for each dataset separately, but "lazy" 0.5 is pretty good for most datasets.
|
19 |
+
min_speech_duration_ms: Final speech chunks shorter min_speech_duration_ms are thrown out.
|
20 |
+
max_speech_duration_s: Maximum duration of speech chunks in seconds. Chunks longer
|
21 |
+
than max_speech_duration_s will be split at the timestamp of the last silence that
|
22 |
+
lasts more than 100ms (if any), to prevent aggressive cutting. Otherwise, they will be
|
23 |
+
split aggressively just before max_speech_duration_s.
|
24 |
+
min_silence_duration_ms: In the end of each speech chunk wait for min_silence_duration_ms
|
25 |
+
before separating it
|
26 |
+
window_size_samples: Audio chunks of window_size_samples size are fed to the silero VAD model.
|
27 |
+
WARNING! Silero VAD models were trained using 512, 1024, 1536 samples for 16000 sample rate.
|
28 |
+
Values other than these may affect model performance!!
|
29 |
+
speech_pad_ms: Final speech chunks are padded by speech_pad_ms each side
|
30 |
+
"""
|
31 |
+
|
32 |
+
threshold: float = 0.5
|
33 |
+
min_speech_duration_ms: int = 250
|
34 |
+
max_speech_duration_s: float = float("inf")
|
35 |
+
min_silence_duration_ms: int = 2000
|
36 |
+
window_size_samples: int = 1024
|
37 |
+
speech_pad_ms: int = 400
|
38 |
+
|
39 |
+
|
40 |
+
def get_speech_timestamps(
|
41 |
+
audio: np.ndarray,
|
42 |
+
vad_options: Optional[VadOptions] = None,
|
43 |
+
**kwargs,
|
44 |
+
) -> List[dict]:
|
45 |
+
"""This method is used for splitting long audios into speech chunks using silero VAD.
|
46 |
+
|
47 |
+
Args:
|
48 |
+
audio: One dimensional float array.
|
49 |
+
vad_options: Options for VAD processing.
|
50 |
+
kwargs: VAD options passed as keyword arguments for backward compatibility.
|
51 |
+
|
52 |
+
Returns:
|
53 |
+
List of dicts containing begin and end samples of each speech chunk.
|
54 |
+
"""
|
55 |
+
if vad_options is None:
|
56 |
+
vad_options = VadOptions(**kwargs)
|
57 |
+
|
58 |
+
threshold = vad_options.threshold
|
59 |
+
min_speech_duration_ms = vad_options.min_speech_duration_ms
|
60 |
+
max_speech_duration_s = vad_options.max_speech_duration_s
|
61 |
+
min_silence_duration_ms = vad_options.min_silence_duration_ms
|
62 |
+
window_size_samples = vad_options.window_size_samples
|
63 |
+
speech_pad_ms = vad_options.speech_pad_ms
|
64 |
+
|
65 |
+
if window_size_samples not in [512, 1024, 1536]:
|
66 |
+
warnings.warn(
|
67 |
+
"Unusual window_size_samples! Supported window_size_samples:\n"
|
68 |
+
" - [512, 1024, 1536] for 16000 sampling_rate"
|
69 |
+
)
|
70 |
+
|
71 |
+
sampling_rate = 16000
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72 |
+
min_speech_samples = sampling_rate * min_speech_duration_ms / 1000
|
73 |
+
speech_pad_samples = sampling_rate * speech_pad_ms / 1000
|
74 |
+
max_speech_samples = (
|
75 |
+
sampling_rate * max_speech_duration_s
|
76 |
+
- window_size_samples
|
77 |
+
- 2 * speech_pad_samples
|
78 |
+
)
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79 |
+
min_silence_samples = sampling_rate * min_silence_duration_ms / 1000
|
80 |
+
min_silence_samples_at_max_speech = sampling_rate * 98 / 1000
|
81 |
+
|
82 |
+
audio_length_samples = len(audio)
|
83 |
+
|
84 |
+
model = get_vad_model()
|
85 |
+
state = model.get_initial_state(batch_size=1)
|
86 |
+
|
87 |
+
speech_probs = []
|
88 |
+
for current_start_sample in range(0, audio_length_samples, window_size_samples):
|
89 |
+
chunk = audio[current_start_sample : current_start_sample + window_size_samples]
|
90 |
+
if len(chunk) < window_size_samples:
|
91 |
+
chunk = np.pad(chunk, (0, int(window_size_samples - len(chunk))))
|
92 |
+
speech_prob, state = model(chunk, state, sampling_rate)
|
93 |
+
speech_probs.append(speech_prob)
|
94 |
+
|
95 |
+
triggered = False
|
96 |
+
speeches = []
|
97 |
+
current_speech = {}
|
98 |
+
neg_threshold = threshold - 0.15
|
99 |
+
|
100 |
+
# to save potential segment end (and tolerate some silence)
|
101 |
+
temp_end = 0
|
102 |
+
# to save potential segment limits in case of maximum segment size reached
|
103 |
+
prev_end = next_start = 0
|
104 |
+
|
105 |
+
for i, speech_prob in enumerate(speech_probs):
|
106 |
+
if (speech_prob >= threshold) and temp_end:
|
107 |
+
temp_end = 0
|
108 |
+
if next_start < prev_end:
|
109 |
+
next_start = window_size_samples * i
|
110 |
+
|
111 |
+
if (speech_prob >= threshold) and not triggered:
|
112 |
+
triggered = True
|
113 |
+
current_speech["start"] = window_size_samples * i
|
114 |
+
continue
|
115 |
+
|
116 |
+
if (
|
117 |
+
triggered
|
118 |
+
and (window_size_samples * i) - current_speech["start"] > max_speech_samples
|
119 |
+
):
|
120 |
+
if prev_end:
|
121 |
+
current_speech["end"] = prev_end
|
122 |
+
speeches.append(current_speech)
|
123 |
+
current_speech = {}
|
124 |
+
# previously reached silence (< neg_thres) and is still not speech (< thres)
|
125 |
+
if next_start < prev_end:
|
126 |
+
triggered = False
|
127 |
+
else:
|
128 |
+
current_speech["start"] = next_start
|
129 |
+
prev_end = next_start = temp_end = 0
|
130 |
+
else:
|
131 |
+
current_speech["end"] = window_size_samples * i
|
132 |
+
speeches.append(current_speech)
|
133 |
+
current_speech = {}
|
134 |
+
prev_end = next_start = temp_end = 0
|
135 |
+
triggered = False
|
136 |
+
continue
|
137 |
+
|
138 |
+
if (speech_prob < neg_threshold) and triggered:
|
139 |
+
if not temp_end:
|
140 |
+
temp_end = window_size_samples * i
|
141 |
+
# condition to avoid cutting in very short silence
|
142 |
+
if (window_size_samples * i) - temp_end > min_silence_samples_at_max_speech:
|
143 |
+
prev_end = temp_end
|
144 |
+
if (window_size_samples * i) - temp_end < min_silence_samples:
|
145 |
+
continue
|
146 |
+
else:
|
147 |
+
current_speech["end"] = temp_end
|
148 |
+
if (
|
149 |
+
current_speech["end"] - current_speech["start"]
|
150 |
+
) > min_speech_samples:
|
151 |
+
speeches.append(current_speech)
|
152 |
+
current_speech = {}
|
153 |
+
prev_end = next_start = temp_end = 0
|
154 |
+
triggered = False
|
155 |
+
continue
|
156 |
+
|
157 |
+
if (
|
158 |
+
current_speech
|
159 |
+
and (audio_length_samples - current_speech["start"]) > min_speech_samples
|
160 |
+
):
|
161 |
+
current_speech["end"] = audio_length_samples
|
162 |
+
speeches.append(current_speech)
|
163 |
+
|
164 |
+
for i, speech in enumerate(speeches):
|
165 |
+
if i == 0:
|
166 |
+
speech["start"] = int(max(0, speech["start"] - speech_pad_samples))
|
167 |
+
if i != len(speeches) - 1:
|
168 |
+
silence_duration = speeches[i + 1]["start"] - speech["end"]
|
169 |
+
if silence_duration < 2 * speech_pad_samples:
|
170 |
+
speech["end"] += int(silence_duration // 2)
|
171 |
+
speeches[i + 1]["start"] = int(
|
172 |
+
max(0, speeches[i + 1]["start"] - silence_duration // 2)
|
173 |
+
)
|
174 |
+
else:
|
175 |
+
speech["end"] = int(
|
176 |
+
min(audio_length_samples, speech["end"] + speech_pad_samples)
|
177 |
+
)
|
178 |
+
speeches[i + 1]["start"] = int(
|
179 |
+
max(0, speeches[i + 1]["start"] - speech_pad_samples)
|
180 |
+
)
|
181 |
+
else:
|
182 |
+
speech["end"] = int(
|
183 |
+
min(audio_length_samples, speech["end"] + speech_pad_samples)
|
184 |
+
)
|
185 |
+
|
186 |
+
return speeches
|
187 |
+
|
188 |
+
|
189 |
+
def collect_chunks(audio: np.ndarray, chunks: List[dict]) -> np.ndarray:
|
190 |
+
"""Collects and concatenates audio chunks."""
|
191 |
+
if not chunks:
|
192 |
+
return np.array([], dtype=np.float32)
|
193 |
+
|
194 |
+
return np.concatenate([audio[chunk["start"] : chunk["end"]] for chunk in chunks])
|
195 |
+
|
196 |
+
|
197 |
+
class SpeechTimestampsMap:
|
198 |
+
"""Helper class to restore original speech timestamps."""
|
199 |
+
|
200 |
+
def __init__(self, chunks: List[dict], sampling_rate: int, time_precision: int = 2):
|
201 |
+
self.sampling_rate = sampling_rate
|
202 |
+
self.time_precision = time_precision
|
203 |
+
self.chunk_end_sample = []
|
204 |
+
self.total_silence_before = []
|
205 |
+
|
206 |
+
previous_end = 0
|
207 |
+
silent_samples = 0
|
208 |
+
|
209 |
+
for chunk in chunks:
|
210 |
+
silent_samples += chunk["start"] - previous_end
|
211 |
+
previous_end = chunk["end"]
|
212 |
+
|
213 |
+
self.chunk_end_sample.append(chunk["end"] - silent_samples)
|
214 |
+
self.total_silence_before.append(silent_samples / sampling_rate)
|
215 |
+
|
216 |
+
def get_original_time(
|
217 |
+
self,
|
218 |
+
time: float,
|
219 |
+
chunk_index: Optional[int] = None,
|
220 |
+
) -> float:
|
221 |
+
if chunk_index is None:
|
222 |
+
chunk_index = self.get_chunk_index(time)
|
223 |
+
|
224 |
+
total_silence_before = self.total_silence_before[chunk_index]
|
225 |
+
return round(total_silence_before + time, self.time_precision)
|
226 |
+
|
227 |
+
def get_chunk_index(self, time: float) -> int:
|
228 |
+
sample = int(time * self.sampling_rate)
|
229 |
+
return min(
|
230 |
+
bisect.bisect(self.chunk_end_sample, sample),
|
231 |
+
len(self.chunk_end_sample) - 1,
|
232 |
+
)
|
233 |
+
|
234 |
+
|
235 |
+
@functools.lru_cache
|
236 |
+
def get_vad_model():
|
237 |
+
"""Returns the VAD model instance."""
|
238 |
+
asset_dir = os.path.join(os.path.dirname(__file__), "assets")
|
239 |
+
path = os.path.join(asset_dir, "silero_vad.onnx")
|
240 |
+
return SileroVADModel(path)
|
241 |
+
|
242 |
+
|
243 |
+
class SileroVADModel:
|
244 |
+
def __init__(self, path):
|
245 |
+
try:
|
246 |
+
import onnxruntime
|
247 |
+
except ImportError as e:
|
248 |
+
raise RuntimeError(
|
249 |
+
"Applying the VAD filter requires the onnxruntime package"
|
250 |
+
) from e
|
251 |
+
|
252 |
+
opts = onnxruntime.SessionOptions()
|
253 |
+
opts.inter_op_num_threads = 1
|
254 |
+
opts.intra_op_num_threads = 1
|
255 |
+
opts.log_severity_level = 4
|
256 |
+
|
257 |
+
self.session = onnxruntime.InferenceSession(
|
258 |
+
path,
|
259 |
+
providers=["CPUExecutionProvider"],
|
260 |
+
sess_options=opts,
|
261 |
+
)
|
262 |
+
|
263 |
+
def get_initial_state(self, batch_size: int):
|
264 |
+
h = np.zeros((2, batch_size, 64), dtype=np.float32)
|
265 |
+
c = np.zeros((2, batch_size, 64), dtype=np.float32)
|
266 |
+
return h, c
|
267 |
+
|
268 |
+
def __call__(self, x, state, sr: int):
|
269 |
+
if len(x.shape) == 1:
|
270 |
+
x = np.expand_dims(x, 0)
|
271 |
+
if len(x.shape) > 2:
|
272 |
+
raise ValueError(
|
273 |
+
f"Too many dimensions for input audio chunk {len(x.shape)}"
|
274 |
+
)
|
275 |
+
if sr / x.shape[1] > 31.25:
|
276 |
+
raise ValueError("Input audio chunk is too short")
|
277 |
+
|
278 |
+
h, c = state
|
279 |
+
|
280 |
+
ort_inputs = {
|
281 |
+
"input": x,
|
282 |
+
"h": h,
|
283 |
+
"c": c,
|
284 |
+
"sr": np.array(sr, dtype="int64"),
|
285 |
+
}
|
286 |
+
|
287 |
+
out, h, c = self.session.run(None, ort_inputs)
|
288 |
+
state = (h, c)
|
289 |
+
|
290 |
+
return out, state
|