Support VAD filter (#95)
* Support VAD filter * Generalize function collect_samples * Define AudioSegment class * Only pass prompt and prefix to the first chunk * Add dict argument vad_parameters * Fix isort format * Rename method * Update README * Add shortcut when the chunk offset is 0 * Reword readme * Fix end property * Concatenate the speech chunks * Cleanup diff * Increase default speech pad * Update README * Increase default speech pad
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268
faster_whisper/vad.py
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268
faster_whisper/vad.py
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import bisect
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import functools
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import os
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import warnings
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from typing import List, Optional
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import numpy as np
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from faster_whisper.utils import get_assets_path
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# The code below is adapted from https://github.com/snakers4/silero-vad.
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def get_speech_timestamps(
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audio: np.ndarray,
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*,
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threshold: float = 0.5,
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min_speech_duration_ms: int = 250,
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max_speech_duration_s: float = float("inf"),
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min_silence_duration_ms: int = 2000,
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window_size_samples: int = 1024,
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speech_pad_ms: int = 200,
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) -> List[dict]:
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"""This method is used for splitting long audios into speech chunks using silero VAD.
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Args:
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audio: One dimensional float array.
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threshold: Speech threshold. Silero VAD outputs speech probabilities for each audio chunk,
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probabilities ABOVE this value are considered as SPEECH. It is better to tune this
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parameter for each dataset separately, but "lazy" 0.5 is pretty good for most datasets.
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min_speech_duration_ms: Final speech chunks shorter min_speech_duration_ms are thrown out.
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max_speech_duration_s: Maximum duration of speech chunks in seconds. Chunks longer
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than max_speech_duration_s will be split at the timestamp of the last silence that
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lasts more than 100s (if any), to prevent agressive cutting. Otherwise, they will be
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split aggressively just before max_speech_duration_s.
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min_silence_duration_ms: In the end of each speech chunk wait for min_silence_duration_ms
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before separating it
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window_size_samples: Audio chunks of window_size_samples size are fed to the silero VAD model.
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WARNING! Silero VAD models were trained using 512, 1024, 1536 samples for 16000 sample rate.
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Values other than these may affect model perfomance!!
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speech_pad_ms: Final speech chunks are padded by speech_pad_ms each side
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Returns:
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List of dicts containing begin and end samples of each speech chunk.
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"""
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if window_size_samples not in [512, 1024, 1536]:
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warnings.warn(
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"Unusual window_size_samples! Supported window_size_samples:\n"
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" - [512, 1024, 1536] for 16000 sampling_rate"
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)
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sampling_rate = 16000
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min_speech_samples = sampling_rate * min_speech_duration_ms / 1000
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speech_pad_samples = sampling_rate * speech_pad_ms / 1000
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max_speech_samples = (
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sampling_rate * max_speech_duration_s
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- window_size_samples
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- 2 * speech_pad_samples
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)
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min_silence_samples = sampling_rate * min_silence_duration_ms / 1000
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min_silence_samples_at_max_speech = sampling_rate * 98 / 1000
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audio_length_samples = len(audio)
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model = get_vad_model()
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state = model.get_initial_state(batch_size=1)
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speech_probs = []
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for current_start_sample in range(0, audio_length_samples, window_size_samples):
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chunk = audio[current_start_sample : current_start_sample + window_size_samples]
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if len(chunk) < window_size_samples:
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chunk = np.pad(chunk, (0, int(window_size_samples - len(chunk))))
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speech_prob, state = model(chunk, state, sampling_rate)
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speech_probs.append(speech_prob)
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triggered = False
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speeches = []
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current_speech = {}
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neg_threshold = threshold - 0.15
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# to save potential segment end (and tolerate some silence)
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temp_end = 0
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# to save potential segment limits in case of maximum segment size reached
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prev_end = next_start = 0
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for i, speech_prob in enumerate(speech_probs):
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if (speech_prob >= threshold) and temp_end:
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temp_end = 0
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if next_start < prev_end:
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next_start = window_size_samples * i
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if (speech_prob >= threshold) and not triggered:
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triggered = True
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current_speech["start"] = window_size_samples * i
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continue
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if (
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triggered
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and (window_size_samples * i) - current_speech["start"] > max_speech_samples
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):
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if prev_end:
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current_speech["end"] = prev_end
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speeches.append(current_speech)
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current_speech = {}
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# previously reached silence (< neg_thres) and is still not speech (< thres)
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if next_start < prev_end:
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triggered = False
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else:
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current_speech["start"] = next_start
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prev_end = next_start = temp_end = 0
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else:
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current_speech["end"] = window_size_samples * i
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speeches.append(current_speech)
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current_speech = {}
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prev_end = next_start = temp_end = 0
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triggered = False
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continue
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if (speech_prob < neg_threshold) and triggered:
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if not temp_end:
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temp_end = window_size_samples * i
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# condition to avoid cutting in very short silence
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if (window_size_samples * i) - temp_end > min_silence_samples_at_max_speech:
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prev_end = temp_end
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if (window_size_samples * i) - temp_end < min_silence_samples:
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continue
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else:
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current_speech["end"] = temp_end
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if (
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current_speech["end"] - current_speech["start"]
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) > min_speech_samples:
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speeches.append(current_speech)
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current_speech = {}
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prev_end = next_start = temp_end = 0
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triggered = False
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continue
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if (
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current_speech
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and (audio_length_samples - current_speech["start"]) > min_speech_samples
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):
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current_speech["end"] = audio_length_samples
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speeches.append(current_speech)
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for i, speech in enumerate(speeches):
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if i == 0:
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speech["start"] = int(max(0, speech["start"] - speech_pad_samples))
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if i != len(speeches) - 1:
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silence_duration = speeches[i + 1]["start"] - speech["end"]
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if silence_duration < 2 * speech_pad_samples:
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speech["end"] += int(silence_duration // 2)
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speeches[i + 1]["start"] = int(
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max(0, speeches[i + 1]["start"] - silence_duration // 2)
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)
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else:
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speech["end"] = int(
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min(audio_length_samples, speech["end"] + speech_pad_samples)
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)
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speeches[i + 1]["start"] = int(
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max(0, speeches[i + 1]["start"] - speech_pad_samples)
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)
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else:
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speech["end"] = int(
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min(audio_length_samples, speech["end"] + speech_pad_samples)
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)
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return speeches
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def collect_chunks(audio: np.ndarray, chunks: List[dict]) -> np.ndarray:
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"""Collects and concatenates audio chunks."""
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if not chunks:
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return np.array([], dtype=np.float32)
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return np.concatenate([audio[chunk["start"] : chunk["end"]] for chunk in chunks])
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class SpeechTimestampsMap:
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"""Helper class to restore original speech timestamps."""
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def __init__(self, chunks: List[dict], sampling_rate: int, time_precision: int = 2):
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self.sampling_rate = sampling_rate
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self.time_precision = time_precision
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self.chunk_end_sample = []
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self.total_silence_before = []
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previous_end = 0
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silent_samples = 0
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for chunk in chunks:
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silent_samples += chunk["start"] - previous_end
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previous_end = chunk["end"]
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self.chunk_end_sample.append(chunk["end"] - silent_samples)
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self.total_silence_before.append(silent_samples / sampling_rate)
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def get_original_time(
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self,
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time: float,
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chunk_index: Optional[int] = None,
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) -> float:
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if chunk_index is None:
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chunk_index = self.get_chunk_index(time)
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total_silence_before = self.total_silence_before[chunk_index]
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return round(total_silence_before + time, self.time_precision)
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def get_chunk_index(self, time: float) -> int:
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sample = int(time * self.sampling_rate)
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return bisect.bisect(self.chunk_end_sample, sample)
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@functools.lru_cache
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def get_vad_model():
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"""Returns the VAD model instance."""
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path = os.path.join(get_assets_path(), "silero_vad.onnx")
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return SileroVADModel(path)
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class SileroVADModel:
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def __init__(self, path):
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try:
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import onnxruntime
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except ImportError as e:
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raise RuntimeError(
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"Applying the VAD filter requires the onnxruntime package"
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) from e
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opts = onnxruntime.SessionOptions()
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opts.inter_op_num_threads = 1
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opts.intra_op_num_threads = 1
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opts.log_severity_level = 4
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self.session = onnxruntime.InferenceSession(
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path,
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providers=["CPUExecutionProvider"],
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sess_options=opts,
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)
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def get_initial_state(self, batch_size: int):
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h = np.zeros((2, batch_size, 64), dtype=np.float32)
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c = np.zeros((2, batch_size, 64), dtype=np.float32)
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return h, c
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def __call__(self, x, state, sr: int):
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if len(x.shape) == 1:
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x = np.expand_dims(x, 0)
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if len(x.shape) > 2:
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raise ValueError(
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f"Too many dimensions for input audio chunk {len(x.shape)}"
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)
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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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h, c = state
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ort_inputs = {
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"input": x,
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"h": h,
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"c": c,
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"sr": np.array(sr, dtype="int64"),
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}
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out, h, c = self.session.run(None, ort_inputs)
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state = (h, c)
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return out, state
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