Upload silero_vad.py

modules/vad/silero_vad.py

@@ -0,0 +1,277 @@
+# Adapted from https://github.com/SYSTRAN/faster-whisper/blob/master/faster_whisper/vad.py
+
+from faster_whisper.vad import VadOptions, get_vad_model
+import numpy as np
+from typing import BinaryIO, Union, List, Optional, Tuple
+import warnings
+import bisect
+import faster_whisper
+from faster_whisper.transcribe import SpeechTimestampsMap
+import gradio as gr
+
+from modules.whisper.data_classes import *
+
+
+class SileroVAD:
+    def __init__(self):
+        self.sampling_rate = 16000
+        self.window_size_samples = 512
+        self.model = None
+
+    def run(self,
+            audio: Union[str, BinaryIO, np.ndarray],
+            vad_parameters: VadOptions,
+            progress: gr.Progress = gr.Progress()
+            ) -> Tuple[np.ndarray, List[dict]]:
+        """
+        Run VAD
+
+        Parameters
+        ----------
+        audio: Union[str, BinaryIO, np.ndarray]
+            Audio path or file binary or Audio numpy array
+        vad_parameters:
+            Options for VAD processing.
+        progress: gr.Progress
+            Indicator to show progress directly in gradio.
+
+        Returns
+        ----------
+        np.ndarray
+            Pre-processed audio with VAD
+        List[dict]
+            Chunks of speeches to be used to restore the timestamps later
+        """
+
+        sampling_rate = self.sampling_rate
+
+        if not isinstance(audio, np.ndarray):
+            audio = faster_whisper.decode_audio(audio, sampling_rate=sampling_rate)
+
+        duration = audio.shape[0] / sampling_rate
+        duration_after_vad = duration
+
+        if vad_parameters is None:
+            vad_parameters = VadOptions()
+        elif isinstance(vad_parameters, dict):
+            vad_parameters = VadOptions(**vad_parameters)
+        speech_chunks = self.get_speech_timestamps(
+            audio=audio,
+            vad_options=vad_parameters,
+            progress=progress
+        )
+
+        audio = self.collect_chunks(audio, speech_chunks)
+        duration_after_vad = audio.shape[0] / sampling_rate
+
+        return audio, speech_chunks
+
+    def get_speech_timestamps(
+        self,
+        audio: np.ndarray,
+        vad_options: Optional[VadOptions] = None,
+        progress: gr.Progress = gr.Progress(),
+        **kwargs,
+    ) -> List[dict]:
+        """This method is used for splitting long audios into speech chunks using silero VAD.
+
+        Args:
+          audio: One dimensional float array.
+          vad_options: Options for VAD processing.
+          kwargs: VAD options passed as keyword arguments for backward compatibility.
+          progress: Gradio progress to indicate progress.
+
+        Returns:
+          List of dicts containing begin and end samples of each speech chunk.
+        """
+
+        if self.model is None:
+            self.update_model()
+
+        if vad_options is None:
+            vad_options = VadOptions(**kwargs)
+
+        threshold = vad_options.threshold
+        neg_threshold = vad_options.neg_threshold
+        min_speech_duration_ms = vad_options.min_speech_duration_ms
+        max_speech_duration_s = vad_options.max_speech_duration_s
+        min_silence_duration_ms = vad_options.min_silence_duration_ms
+        window_size_samples = self.window_size_samples
+        speech_pad_ms = vad_options.speech_pad_ms
+        min_speech_samples = self.sampling_rate * min_speech_duration_ms / 1000
+        speech_pad_samples = self.sampling_rate * speech_pad_ms / 1000
+        max_speech_samples = (
+                self.sampling_rate * max_speech_duration_s
+                - window_size_samples
+                - 2 * speech_pad_samples
+        )
+        min_silence_samples = self.sampling_rate * min_silence_duration_ms / 1000
+        min_silence_samples_at_max_speech = self.sampling_rate * 98 / 1000
+
+        audio_length_samples = len(audio)
+
+        padded_audio = np.pad(
+            audio, (0, window_size_samples - audio.shape[0] % window_size_samples)
+        )
+        speech_probs = self.model(padded_audio.reshape(1, -1)).squeeze(0)
+
+        triggered = False
+        speeches = []
+        current_speech = {}
+        if neg_threshold is None:
+            neg_threshold = max(threshold - 0.15, 0.01)
+
+        # to save potential segment end (and tolerate some silence)
+        temp_end = 0
+        # to save potential segment limits in case of maximum segment size reached
+        prev_end = next_start = 0
+
+        for i, speech_prob in enumerate(speech_probs):
+            if (speech_prob >= threshold) and temp_end:
+                temp_end = 0
+                if next_start < prev_end:
+                    next_start = window_size_samples * i
+
+            if (speech_prob >= threshold) and not triggered:
+                triggered = True
+                current_speech["start"] = window_size_samples * i
+                continue
+
+            if (
+                    triggered
+                    and (window_size_samples * i) - current_speech["start"] > max_speech_samples
+            ):
+                if prev_end:
+                    current_speech["end"] = prev_end
+                    speeches.append(current_speech)
+                    current_speech = {}
+                    # previously reached silence (< neg_thres) and is still not speech (< thres)
+                    if next_start < prev_end:
+                        triggered = False
+                    else:
+                        current_speech["start"] = next_start
+                    prev_end = next_start = temp_end = 0
+                else:
+                    current_speech["end"] = window_size_samples * i
+                    speeches.append(current_speech)
+                    current_speech = {}
+                    prev_end = next_start = temp_end = 0
+                    triggered = False
+                    continue
+
+            if (speech_prob < neg_threshold) and triggered:
+                if not temp_end:
+                    temp_end = window_size_samples * i
+                # condition to avoid cutting in very short silence
+                if (window_size_samples * i) - temp_end > min_silence_samples_at_max_speech:
+                    prev_end = temp_end
+                if (window_size_samples * i) - temp_end < min_silence_samples:
+                    continue
+                else:
+                    current_speech["end"] = temp_end
+                    if (
+                            current_speech["end"] - current_speech["start"]
+                    ) > min_speech_samples:
+                        speeches.append(current_speech)
+                    current_speech = {}
+                    prev_end = next_start = temp_end = 0
+                    triggered = False
+                    continue
+
+        if (
+                current_speech
+                and (audio_length_samples - current_speech["start"]) > min_speech_samples
+        ):
+            current_speech["end"] = audio_length_samples
+            speeches.append(current_speech)
+
+        for i, speech in enumerate(speeches):
+            if i == 0:
+                speech["start"] = int(max(0, speech["start"] - speech_pad_samples))
+            if i != len(speeches) - 1:
+                silence_duration = speeches[i + 1]["start"] - speech["end"]
+                if silence_duration < 2 * speech_pad_samples:
+                    speech["end"] += int(silence_duration // 2)
+                    speeches[i + 1]["start"] = int(
+                        max(0, speeches[i + 1]["start"] - silence_duration // 2)
+                    )
+                else:
+                    speech["end"] = int(
+                        min(audio_length_samples, speech["end"] + speech_pad_samples)
+                    )
+                    speeches[i + 1]["start"] = int(
+                        max(0, speeches[i + 1]["start"] - speech_pad_samples)
+                    )
+            else:
+                speech["end"] = int(
+                    min(audio_length_samples, speech["end"] + speech_pad_samples)
+                )
+
+        return speeches
+
+    def update_model(self):
+        self.model = get_vad_model()
+
+    @staticmethod
+    def collect_chunks(audio: np.ndarray, chunks: List[dict]) -> np.ndarray:
+        """Collects and concatenates audio chunks."""
+        if not chunks:
+            return np.array([], dtype=np.float32)
+
+        return np.concatenate([audio[chunk["start"]: chunk["end"]] for chunk in chunks])
+
+    @staticmethod
+    def format_timestamp(
+        seconds: float,
+        always_include_hours: bool = False,
+        decimal_marker: str = ".",
+    ) -> str:
+        assert seconds >= 0, "non-negative timestamp expected"
+        milliseconds = round(seconds * 1000.0)
+
+        hours = milliseconds // 3_600_000
+        milliseconds -= hours * 3_600_000
+
+        minutes = milliseconds // 60_000
+        milliseconds -= minutes * 60_000
+
+        seconds = milliseconds // 1_000
+        milliseconds -= seconds * 1_000
+
+        hours_marker = f"{hours:02d}:" if always_include_hours or hours > 0 else ""
+        return (
+            f"{hours_marker}{minutes:02d}:{seconds:02d}{decimal_marker}{milliseconds:03d}"
+        )
+
+    def restore_speech_timestamps(
+        self,
+        segments: List[Segment],
+        speech_chunks: List[dict],
+        sampling_rate: Optional[int] = None,
+    ) -> List[Segment]:
+        if sampling_rate is None:
+            sampling_rate = self.sampling_rate
+
+        ts_map = SpeechTimestampsMap(speech_chunks, sampling_rate)
+
+        for segment in segments:
+            if segment.words:
+                words = []
+                for word in segment.words:
+                    # Ensure the word start and end times are resolved to the same chunk.
+                    middle = (word.start + word.end) / 2
+                    chunk_index = ts_map.get_chunk_index(middle)
+                    word.start = ts_map.get_original_time(word.start, chunk_index)
+                    word.end = ts_map.get_original_time(word.end, chunk_index)
+                    words.append(word)
+
+                segment.start = words[0].start
+                segment.end = words[-1].end
+                segment.words = words
+
+            else:
+                segment.start = ts_map.get_original_time(segment.start)
+                segment.end = ts_map.get_original_time(segment.end)
+
+        return segments
+