import torch import torchaudio from typing import Callable, List import torch.nn.functional as F import warnings languages = ['ru', 'en', 'de', 'es'] class OnnxWrapper(): def __init__(self, path, force_onnx_cpu=False): import numpy as np global np import onnxruntime opts = onnxruntime.SessionOptions() opts.inter_op_num_threads = 1 opts.intra_op_num_threads = 1 if force_onnx_cpu and 'CPUExecutionProvider' in onnxruntime.get_available_providers(): self.session = onnxruntime.InferenceSession(path, providers=['CPUExecutionProvider'], sess_options=opts) else: self.session = onnxruntime.InferenceSession(path, sess_options=opts) self.reset_states() self.sample_rates = [8000, 16000] def _validate_input(self, x, sr: int): if x.dim() == 1: x = x.unsqueeze(0) if x.dim() > 2: raise ValueError(f"Too many dimensions for input audio chunk {x.dim()}") if sr != 16000 and (sr % 16000 == 0): step = sr // 16000 x = x[:,::step] sr = 16000 if sr not in self.sample_rates: raise ValueError(f"Supported sampling rates: {self.sample_rates} (or multiply of 16000)") if sr / x.shape[1] > 31.25: raise ValueError("Input audio chunk is too short") return x, sr def reset_states(self, batch_size=1): self._h = np.zeros((2, batch_size, 64)).astype('float32') self._c = np.zeros((2, batch_size, 64)).astype('float32') self._last_sr = 0 self._last_batch_size = 0 def __call__(self, x, sr: int): x, sr = self._validate_input(x, sr) batch_size = x.shape[0] if not self._last_batch_size: self.reset_states(batch_size) if (self._last_sr) and (self._last_sr != sr): self.reset_states(batch_size) if (self._last_batch_size) and (self._last_batch_size != batch_size): self.reset_states(batch_size) if sr in [8000, 16000]: ort_inputs = {'input': x.numpy(), 'h': self._h, 'c': self._c, 'sr': np.array(sr, dtype='int64')} ort_outs = self.session.run(None, ort_inputs) out, self._h, self._c = ort_outs else: raise ValueError() self._last_sr = sr self._last_batch_size = batch_size out = torch.tensor(out) return out def audio_forward(self, x, sr: int, num_samples: int = 512): outs = [] x, sr = self._validate_input(x, sr) if x.shape[1] % num_samples: pad_num = num_samples - (x.shape[1] % num_samples) x = torch.nn.functional.pad(x, (0, pad_num), 'constant', value=0.0) self.reset_states(x.shape[0]) for i in range(0, x.shape[1], num_samples): wavs_batch = x[:, i:i+num_samples] out_chunk = self.__call__(wavs_batch, sr) outs.append(out_chunk) stacked = torch.cat(outs, dim=1) return stacked.cpu() class Validator(): def __init__(self, url, force_onnx_cpu): self.onnx = True if url.endswith('.onnx') else False torch.hub.download_url_to_file(url, 'inf.model') if self.onnx: import onnxruntime if force_onnx_cpu and 'CPUExecutionProvider' in onnxruntime.get_available_providers(): self.model = onnxruntime.InferenceSession('inf.model', providers=['CPUExecutionProvider']) else: self.model = onnxruntime.InferenceSession('inf.model') else: self.model = init_jit_model(model_path='inf.model') def __call__(self, inputs: torch.Tensor): with torch.no_grad(): if self.onnx: ort_inputs = {'input': inputs.cpu().numpy()} outs = self.model.run(None, ort_inputs) outs = [torch.Tensor(x) for x in outs] else: outs = self.model(inputs) return outs def read_audio(path: str, sampling_rate: int = 16000): wav, sr = torchaudio.load(path) if wav.size(0) > 1: wav = wav.mean(dim=0, keepdim=True) if sr != sampling_rate: transform = torchaudio.transforms.Resample(orig_freq=sr, new_freq=sampling_rate) wav = transform(wav) sr = sampling_rate assert sr == sampling_rate return wav.squeeze(0) def save_audio(path: str, tensor: torch.Tensor, sampling_rate: int = 16000): torchaudio.save(path, tensor.unsqueeze(0), sampling_rate, bits_per_sample=16) def init_jit_model(model_path: str, device=torch.device('cpu')): torch.set_grad_enabled(False) model = torch.jit.load(model_path, map_location=device) model.eval() return model def make_visualization(probs, step): import pandas as pd pd.DataFrame({'probs': probs}, index=[x * step for x in range(len(probs))]).plot(figsize=(16, 8), kind='area', ylim=[0, 1.05], xlim=[0, len(probs) * step], xlabel='seconds', ylabel='speech probability', colormap='tab20') def get_speech_timestamps(audio: torch.Tensor, model, threshold: float = 0.5, sampling_rate: int = 16000, min_speech_duration_ms: int = 250, max_speech_duration_s: float = float('inf'), min_silence_duration_ms: int = 100, window_size_samples: int = 512, speech_pad_ms: int = 30, return_seconds: bool = False, visualize_probs: bool = False, progress_tracking_callback: Callable[[float], None] = None): """ This method is used for splitting long audios into speech chunks using silero VAD Parameters ---------- audio: torch.Tensor, one dimensional One dimensional float torch.Tensor, other types are casted to torch if possible model: preloaded .jit silero VAD model threshold: float (default - 0.5) Speech threshold. Silero VAD outputs speech probabilities for each audio chunk, probabilities ABOVE this value are considered as SPEECH. It is better to tune this parameter for each dataset separately, but "lazy" 0.5 is pretty good for most datasets. sampling_rate: int (default - 16000) Currently silero VAD models support 8000 and 16000 sample rates min_speech_duration_ms: int (default - 250 milliseconds) Final speech chunks shorter min_speech_duration_ms are thrown out max_speech_duration_s: int (default - inf) Maximum duration of speech chunks in seconds Chunks longer than max_speech_duration_s will be split at the timestamp of the last silence that lasts more than 100ms (if any), to prevent agressive cutting. Otherwise, they will be split aggressively just before max_speech_duration_s. min_silence_duration_ms: int (default - 100 milliseconds) In the end of each speech chunk wait for min_silence_duration_ms before separating it window_size_samples: int (default - 1536 samples) Audio chunks of window_size_samples size are fed to the silero VAD model. WARNING! Silero VAD models were trained using 512, 1024, 1536 samples for 16000 sample rate and 256, 512, 768 samples for 8000 sample rate. Values other than these may affect model perfomance!! speech_pad_ms: int (default - 30 milliseconds) Final speech chunks are padded by speech_pad_ms each side return_seconds: bool (default - False) whether return timestamps in seconds (default - samples) visualize_probs: bool (default - False) whether draw prob hist or not progress_tracking_callback: Callable[[float], None] (default - None) callback function taking progress in percents as an argument Returns ---------- speeches: list of dicts list containing ends and beginnings of speech chunks (samples or seconds based on return_seconds) """ if not torch.is_tensor(audio): try: audio = torch.Tensor(audio) except: raise TypeError("Audio cannot be casted to tensor. Cast it manually") if len(audio.shape) > 1: for i in range(len(audio.shape)): # trying to squeeze empty dimensions audio = audio.squeeze(0) if len(audio.shape) > 1: raise ValueError("More than one dimension in audio. Are you trying to process audio with 2 channels?") if sampling_rate > 16000 and (sampling_rate % 16000 == 0): step = sampling_rate // 16000 sampling_rate = 16000 audio = audio[::step] warnings.warn('Sampling rate is a multiply of 16000, casting to 16000 manually!') else: step = 1 if sampling_rate == 8000 and window_size_samples > 768: warnings.warn('window_size_samples is too big for 8000 sampling_rate! Better set window_size_samples to 256, 512 or 768 for 8000 sample rate!') if window_size_samples not in [256, 512, 768, 1024, 1536]: warnings.warn('Unusual window_size_samples! Supported window_size_samples:\n - [512, 1024, 1536] for 16000 sampling_rate\n - [256, 512, 768] for 8000 sampling_rate') model.reset_states() min_speech_samples = sampling_rate * min_speech_duration_ms / 1000 speech_pad_samples = sampling_rate * speech_pad_ms / 1000 max_speech_samples = sampling_rate * max_speech_duration_s - window_size_samples - 2 * speech_pad_samples min_silence_samples = sampling_rate * min_silence_duration_ms / 1000 min_silence_samples_at_max_speech = sampling_rate * 98 / 1000 audio_length_samples = len(audio) speech_probs = [] for current_start_sample in range(0, audio_length_samples, window_size_samples): chunk = audio[current_start_sample: current_start_sample + window_size_samples] if len(chunk) < window_size_samples: chunk = torch.nn.functional.pad(chunk, (0, int(window_size_samples - len(chunk)))) speech_prob = model(chunk, sampling_rate).item() speech_probs.append(speech_prob) # caculate progress and seng it to callback function progress = current_start_sample + window_size_samples if progress > audio_length_samples: progress = audio_length_samples progress_percent = (progress / audio_length_samples) * 100 if progress_tracking_callback: progress_tracking_callback(progress_percent) triggered = False speeches = [] current_speech = {} neg_threshold = threshold - 0.15 temp_end = 0 # to save potential segment end (and tolerate some silence) prev_end = next_start = 0 # to save potential segment limits in case of maximum segment size reached 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 = {} if next_start < prev_end: # previously reached silence (< neg_thres) and is still not speech (< thres) 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 if ((window_size_samples * i) - temp_end) > min_silence_samples_at_max_speech : # condition to avoid cutting in very short silence 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)) if return_seconds: for speech_dict in speeches: speech_dict['start'] = round(speech_dict['start'] / sampling_rate, 1) speech_dict['end'] = round(speech_dict['end'] / sampling_rate, 1) elif step > 1: for speech_dict in speeches: speech_dict['start'] *= step speech_dict['end'] *= step if visualize_probs: make_visualization(speech_probs, window_size_samples / sampling_rate) return speeches def get_number_ts(wav: torch.Tensor, model, model_stride=8, hop_length=160, sample_rate=16000): wav = torch.unsqueeze(wav, dim=0) perframe_logits = model(wav)[0] perframe_preds = torch.argmax(torch.softmax(perframe_logits, dim=1), dim=1).squeeze() # (1, num_frames_strided) extended_preds = [] for i in perframe_preds: extended_preds.extend([i.item()] * model_stride) # len(extended_preds) is *num_frames_real*; for each frame of audio we know if it has a number in it. triggered = False timings = [] cur_timing = {} for i, pred in enumerate(extended_preds): if pred == 1: if not triggered: cur_timing['start'] = int((i * hop_length) / (sample_rate / 1000)) triggered = True elif pred == 0: if triggered: cur_timing['end'] = int((i * hop_length) / (sample_rate / 1000)) timings.append(cur_timing) cur_timing = {} triggered = False if cur_timing: cur_timing['end'] = int(len(wav) / (sample_rate / 1000)) timings.append(cur_timing) return timings def get_language(wav: torch.Tensor, model): wav = torch.unsqueeze(wav, dim=0) lang_logits = model(wav)[2] lang_pred = torch.argmax(torch.softmax(lang_logits, dim=1), dim=1).item() # from 0 to len(languages) - 1 assert lang_pred < len(languages) return languages[lang_pred] def get_language_and_group(wav: torch.Tensor, model, lang_dict: dict, lang_group_dict: dict, top_n=1): wav = torch.unsqueeze(wav, dim=0) lang_logits, lang_group_logits = model(wav) softm = torch.softmax(lang_logits, dim=1).squeeze() softm_group = torch.softmax(lang_group_logits, dim=1).squeeze() srtd = torch.argsort(softm, descending=True) srtd_group = torch.argsort(softm_group, descending=True) outs = [] outs_group = [] for i in range(top_n): prob = round(softm[srtd[i]].item(), 2) prob_group = round(softm_group[srtd_group[i]].item(), 2) outs.append((lang_dict[str(srtd[i].item())], prob)) outs_group.append((lang_group_dict[str(srtd_group[i].item())], prob_group)) return outs, outs_group class VADIterator: def __init__(self, model, threshold: float = 0.5, sampling_rate: int = 16000, min_silence_duration_ms: int = 100, speech_pad_ms: int = 30 ): """ Class for stream imitation Parameters ---------- model: preloaded .jit silero VAD model threshold: float (default - 0.5) Speech threshold. Silero VAD outputs speech probabilities for each audio chunk, probabilities ABOVE this value are considered as SPEECH. It is better to tune this parameter for each dataset separately, but "lazy" 0.5 is pretty good for most datasets. sampling_rate: int (default - 16000) Currently silero VAD models support 8000 and 16000 sample rates min_silence_duration_ms: int (default - 100 milliseconds) In the end of each speech chunk wait for min_silence_duration_ms before separating it speech_pad_ms: int (default - 30 milliseconds) Final speech chunks are padded by speech_pad_ms each side """ self.model = model self.threshold = threshold self.sampling_rate = sampling_rate if sampling_rate not in [8000, 16000]: raise ValueError('VADIterator does not support sampling rates other than [8000, 16000]') self.min_silence_samples = sampling_rate * min_silence_duration_ms / 1000 self.speech_pad_samples = sampling_rate * speech_pad_ms / 1000 self.reset_states() def reset_states(self): self.model.reset_states() self.triggered = False self.temp_end = 0 self.current_sample = 0 def __call__(self, x, return_seconds=False): """ x: torch.Tensor audio chunk (see examples in repo) return_seconds: bool (default - False) whether return timestamps in seconds (default - samples) """ if not torch.is_tensor(x): try: x = torch.Tensor(x) except: raise TypeError("Audio cannot be casted to tensor. Cast it manually") window_size_samples = len(x[0]) if x.dim() == 2 else len(x) self.current_sample += window_size_samples speech_prob = self.model(x, self.sampling_rate).item() if (speech_prob >= self.threshold) and self.temp_end: self.temp_end = 0 if (speech_prob >= self.threshold) and not self.triggered: self.triggered = True speech_start = self.current_sample - self.speech_pad_samples return {'start': int(speech_start) if not return_seconds else round(speech_start / self.sampling_rate, 1)} if (speech_prob < self.threshold - 0.15) and self.triggered: if not self.temp_end: self.temp_end = self.current_sample if self.current_sample - self.temp_end < self.min_silence_samples: return None else: speech_end = self.temp_end + self.speech_pad_samples self.temp_end = 0 self.triggered = False return {'end': int(speech_end) if not return_seconds else round(speech_end / self.sampling_rate, 1)} return None def collect_chunks(tss: List[dict], wav: torch.Tensor): chunks = [] for i in tss: chunks.append(wav[i['start']: i['end']]) return torch.cat(chunks) def drop_chunks(tss: List[dict], wav: torch.Tensor): chunks = [] cur_start = 0 for i in tss: chunks.append((wav[cur_start: i['start']])) cur_start = i['end'] return torch.cat(chunks)