import collections import sys import librosa import numpy import random import time import torchaudio from scipy.io.wavfile import read MU = 1800 def read_wave_to_frames_withbgm(path, bgmpath, sr=16000, save_sr=44100, frame_duration=10): orig_sr, orig_wav = read(path) if orig_wav.dtype == numpy.int16: orig_wav = orig_wav / 32768. if len(orig_wav.shape) > 1: orig_wav = numpy.mean(orig_wav, -1) wav = librosa.resample(orig_wav, orig_sr=orig_sr, target_sr=sr, res_type='polyphase') wav = (wav * 2**15).astype(numpy.int16) wav_bytes = wav.tobytes() frames = frame_generator(frame_duration, wav_bytes, sr) if save_sr != orig_sr: vocal_wav = librosa.resample(orig_wav, orig_sr=orig_sr, target_sr=sr, res_type='polyphase') else: vocal_wav = orig_wav orig_sr, bgm_wav = read(bgmpath) if bgm_wav.dtype == numpy.int16: bgm_wav = bgm_wav / 32768. if len(bgm_wav.shape) > 1: bgm_wav = numpy.mean(bgm_wav, -1) return frames, wav, vocal_wav, bgm_wav def read_wave_to_frames(path, sr=16000, frame_duration=10): #start_time = time.time() #wav, orig_sr = librosa.load(path, sr=None, mono=True) orig_sr, wav = read(path) if wav.dtype == numpy.int16: wav = wav / 32768. if len(wav.shape) > 1: wav = numpy.mean(wav, -1) #print("load", time.time() - start_time) #start_time = time.time() wav = librosa.resample(wav, orig_sr=orig_sr, target_sr=sr, res_type='polyphase') #wav = librosa.resample(wav, orig_sr=orig_sr, target_sr=sr, res_type='soxr_qq') #wav, orig_sr = torchaudio.load(path) #wav = torchaudio.functional.resample(wav, orig_sr, sr) #wav = wav.numpy() #print("resample", time.time() - start_time) wav = (wav * 2**15).astype(numpy.int16) wav_bytes = wav.tobytes() frames = frame_generator(frame_duration, wav_bytes, sr) return frames, wav class Frame(object): """Represents a "frame" of audio data.""" def __init__(self, bytes, timestamp, duration): self.bytes = bytes self.timestamp = timestamp self.duration = duration def frame_generator(frame_duration_ms, audio, sample_rate): """Generates audio frames from PCM audio data. Takes the desired frame duration in milliseconds, the PCM data, and the sample rate. Yields Frames of the requested duration. """ n = int(sample_rate * (frame_duration_ms / 1000.0) * 2) offset = 0 timestamp = 0.0 duration = (float(n) / sample_rate) / 2.0 while offset + n < len(audio): yield Frame(audio[offset:offset + n], timestamp, duration) timestamp += duration offset += n def vad_generator(frames, sr, vad): vad_info = [] for frame in frames: vad_info.append(vad.is_speech(frame.bytes, sr)) return vad_info def vad_collector(sample_rate, frame_duration_ms, padding_duration_ms, vad, frames): """Filters out non-voiced audio frames. Given a webrtcvad.Vad and a source of audio frames, yields only the voiced audio. Uses a padded, sliding window algorithm over the audio frames. When more than 90% of the frames in the window are voiced (as reported by the VAD), the collector triggers and begins yielding audio frames. Then the collector waits until 90% of the frames in the window are unvoiced to detrigger. The window is padded at the front and back to provide a small amount of silence or the beginnings/endings of speech around the voiced frames. Arguments: sample_rate - The audio sample rate, in Hz. frame_duration_ms - The frame duration in milliseconds. padding_duration_ms - The amount to pad the window, in milliseconds. vad - An instance of webrtcvad.Vad. frames - a source of audio frames (sequence or generator). Returns: A generator that yields PCM audio data. """ num_padding_frames = int(padding_duration_ms / frame_duration_ms) # We use a deque for our sliding window/ring buffer. ring_buffer = collections.deque(maxlen=num_padding_frames) # We have two states: TRIGGERED and NOTTRIGGERED. We start in the # NOTTRIGGERED state. triggered = False voiced_frames = [] for frame in frames: is_speech = vad.is_speech(frame.bytes, sample_rate) sys.stdout.write('1' if is_speech else '0') if not triggered: ring_buffer.append((frame, is_speech)) num_voiced = len([f for f, speech in ring_buffer if speech]) # If we're NOTTRIGGERED and more than 90% of the frames in # the ring buffer are voiced frames, then enter the # TRIGGERED state. if num_voiced > 0.9 * ring_buffer.maxlen: triggered = True sys.stdout.write('+(%s)' % (ring_buffer[0][0].timestamp,)) # We want to yield all the audio we see from now until # we are NOTTRIGGERED, but we have to start with the # audio that's already in the ring buffer. for f, s in ring_buffer: voiced_frames.append(f) ring_buffer.clear() else: # We're in the TRIGGERED state, so collect the audio data # and add it to the ring buffer. voiced_frames.append(frame) ring_buffer.append((frame, is_speech)) num_unvoiced = len([f for f, speech in ring_buffer if not speech]) # If more than 90% of the frames in the ring buffer are # unvoiced, then enter NOTTRIGGERED and yield whatever # audio we've collected. if num_unvoiced > 0.9 * ring_buffer.maxlen: sys.stdout.write('-(%s)' % (frame.timestamp + frame.duration)) triggered = False yield b''.join([f.bytes for f in voiced_frames]) ring_buffer.clear() voiced_frames = [] if triggered: sys.stdout.write('-(%s)' % (frame.timestamp + frame.duration)) sys.stdout.write('\n') # If we have any leftover voiced audio when we run out of input, # yield it. if voiced_frames: yield b''.join([f.bytes for f in voiced_frames]) class ActivateInfo: def __init__(self, active, duration, start_pos, end_pos, keep=True): self.active = active self.duration = duration self.start_pos = start_pos self.end_pos = end_pos self.keep = keep def __add__(self, x): return x + self.duration def __repr__(self) -> str: return f"{self.active} {self.start_pos}, {self.end_pos}" class SegmentInfo: def __init__(self, type="raw", duration=0, start_pos=0, end_pos=0, frame_duration=10): self.type = type self.duration = duration self.start_pos = start_pos self.end_pos = end_pos self.frame_duration = frame_duration def get_wav_seg(self, wav: numpy.array, sr: int, frame_duration: int=None): fd = frame_duration if frame_duration is not None else self.frame_duration sample_pre_frame = fd*sr/1000 if self.type == "pad": return numpy.zeros((int(sample_pre_frame*self.duration), ), dtype=numpy.int16) return wav[int(self.start_pos*sample_pre_frame):int((self.end_pos*sample_pre_frame))] def __repr__(self) -> str: if self.type == "raw": text = f"{self.start_pos*self.frame_duration}:{self.end_pos*self.frame_duration}" else: text = f"[{self.duration*self.frame_duration}]" return text def get_sil_segments(active_info: ActivateInfo, sil_frame: int, attach_pos: str="mid") -> list: if active_info.duration >= sil_frame: if attach_pos == "tail": seg = [SegmentInfo(start_pos=active_info.start_pos, end_pos=active_info.start_pos+sil_frame)] elif attach_pos == "head": seg = [SegmentInfo(start_pos=active_info.end_pos-sil_frame, end_pos=active_info.end_pos)] elif attach_pos == "mid": seg = [ SegmentInfo(start_pos=active_info.start_pos, end_pos=active_info.start_pos+sil_frame // 2-1), SegmentInfo(start_pos=active_info.end_pos-sil_frame // 2+1, end_pos=active_info.end_pos), ] else: raise NotImplementedError else: if attach_pos == "tail": seg = [ SegmentInfo(start_pos=active_info.start_pos, end_pos=active_info.end_pos), SegmentInfo(type="pad", duration=sil_frame-active_info.duration), ] elif attach_pos == "head": seg = [ SegmentInfo(type="pad", duration=sil_frame-active_info.duration), SegmentInfo(start_pos=active_info.start_pos, end_pos=active_info.end_pos), ] elif attach_pos == "mid": seg = [ SegmentInfo(start_pos=active_info.start_pos, end_pos=active_info.end_pos), ] else: raise NotImplementedError return seg def merge_segment(segment: list) -> list: new_segment = [] last_s = None for s in segment: s: SegmentInfo if s.type == "pad": if last_s is not None: new_segment.append(last_s) last_s = None new_segment.append(s) continue if last_s is None: last_s = s else: if last_s.end_pos+1 == s.start_pos: last_s.end_pos = s.end_pos else: new_segment.append(last_s) last_s = s if last_s is not None: new_segment.append(last_s) return new_segment def random_frame(min_frame, max_frame): #return random.randint(min_frame, max_frame) #mu = (max_frame + max_frame + min_frame) / 3 mu = MU #sigma = (max_frame - mu) / 3 sigma = (mu - min_frame) / 3 length = random.gauss(mu, sigma) length = int(min(max(length, min_frame), max_frame)) #print(length) return length def cut_points_generator( vad_info, min_active_frame=20, sil_frame=50, sil_mid_frame=100, cut_min_frame=8 * 100, cut_max_frame=20 * 100, is_random_min_frame=False, ): curr_min_frame = cut_min_frame last_active_frame = 0 is_last_active = False for i, is_curr_active in enumerate(vad_info): if is_curr_active and not is_last_active: last_active_frame = i elif not is_curr_active and is_last_active and i - last_active_frame <= min_active_frame: for j in range(last_active_frame, i): vad_info[j] = False is_last_active = is_curr_active start_pos = 0 end_pos = 0 duration = 0 is_active = vad_info[0] activate_info = [] for pos, vi in enumerate(vad_info): if is_active == vi: duration += 1 else: activate_info.append(ActivateInfo(is_active, duration, start_pos, pos-1)) is_active = vi start_pos = pos duration = 1 activate_info.append(ActivateInfo(is_active, duration, start_pos, end_pos)) # print(activate_info) segment_info = [] curr_segment = [] curr_segment_duration = 0 max_active_block = len(activate_info) # 需要说明的是,active_info中必然是voice和unvoice交替的。 for i in range(max_active_block): curr_ai = activate_info[i] # print("start", curr_segment_duration, curr_ai.duration) if curr_ai.active: # 当分片中的第一个段是voice时,往前添加静音 if curr_segment_duration == 0: if i == 0: curr_segment.append(SegmentInfo("pad", sil_frame)) else: sil_seg = activate_info[i-1] raw_sil_duration = min(sil_frame, sil_seg.duration // 2) end_pos = sil_seg.end_pos curr_segment = get_sil_segments( ActivateInfo( True, duration=raw_sil_duration, start_pos=sil_seg.end_pos-raw_sil_duration, end_pos=sil_seg.end_pos ), sil_frame=sil_frame, attach_pos="head" ) curr_segment_duration += sil_frame # 然后判断往分片添加该voice段之后的长度变化 next_duration = curr_segment_duration + curr_ai.duration curr_ai_seg = SegmentInfo(start_pos=curr_ai.start_pos, end_pos=curr_ai.end_pos) # print(next_duration) if next_duration > cut_max_frame: # 当添加该段后超出最大长度后,丢弃该分片中之前的段,仅保留当前段 # 这里有个隐含的条件:每个分片中如果包含超过一个voice段,那么其总和必然短于最短长度。而当前段长度不短于cut_max_frame-curr_min_frame if curr_ai.duration > curr_segment_duration: new_segment = get_sil_segments(activate_info[i-1], sil_frame, "head") new_segment.append(curr_ai_seg) if i < max_active_block - 1: new_segment.extend(get_sil_segments(activate_info[i+1], sil_frame, "tail")) else: new_segment.append(SegmentInfo(type="pad", duration=sil_frame)) # print("1", len(segment_info), curr_segment) segment_info.append(merge_segment(new_segment)) if is_random_min_frame: curr_min_frame = random_frame(cut_min_frame, cut_max_frame) curr_segment = [] curr_segment_duration = 0 else: # print("2", len(segment_info), curr_segment) if curr_segment_duration > 10 * 100: segment_info.append(merge_segment(curr_segment)) if is_random_min_frame: curr_min_frame = random_frame(cut_min_frame, cut_max_frame) curr_segment = get_sil_segments(activate_info[i-1], sil_frame, "head") curr_segment.append(curr_ai_seg) curr_segment_duration = sil_frame + curr_ai.duration elif next_duration > curr_min_frame: # 长度足够就添加尾部静音后保存该分片,开新分片 curr_segment.append(curr_ai_seg) if i < max_active_block - 1: # print(activate_info[i+1]) curr_segment.extend(get_sil_segments(activate_info[i+1], sil_frame, "tail")) else: curr_segment.append(SegmentInfo(type="pad", duration=sil_frame)) # print("3", len(segment_info), curr_segment) segment_info.append(merge_segment(curr_segment)) if is_random_min_frame: curr_min_frame = random_frame(cut_min_frame, cut_max_frame) curr_segment = [] curr_segment_duration = 0 else: # 不够就加上然后等待 curr_segment.append(curr_ai_seg) curr_segment_duration += curr_ai.duration else: # 处理静音 if curr_segment_duration == 0: raw_sil_duration = min(sil_frame, curr_ai.duration // 2) end_pos = curr_ai.end_pos curr_segment = get_sil_segments( ActivateInfo( True, duration=raw_sil_duration, start_pos=curr_ai.end_pos-raw_sil_duration, end_pos=curr_ai.end_pos ), sil_frame=sil_frame, attach_pos="head" ) curr_segment_duration += sil_frame else: # 对于出现的静音片段,剪切到sil_mid_frame长度内 #curr_segment.extend(get_sil_segments(curr_ai, sil_mid_frame, attach_pos="mid")) #curr_segment_duration += min(sil_mid_frame, curr_ai.duration) if curr_ai.duration > sil_mid_frame: curr_segment.extend(get_sil_segments(curr_ai, sil_frame, "tail")) segment_info.append(merge_segment(curr_segment)) if is_random_min_frame: curr_min_frame = random_frame(cut_min_frame, cut_max_frame) curr_segment = [] curr_segment_duration = 0 else: # 对于出现的静音片段,剪切到sil_mid_frame长度内 curr_segment.extend(get_sil_segments(curr_ai, sil_mid_frame+1, attach_pos="mid")) curr_segment_duration += min(sil_mid_frame, curr_ai.duration) # print(curr_segment_duration, curr_segment) if len(curr_segment) > 3 and curr_segment_duration > 7 * 100: if activate_info[-1].active: curr_segment.append(SegmentInfo(type="pad", duration=sil_frame)) segment_info.append(merge_segment(curr_segment)) return segment_info def cut_points_storage_generator(raw_vad_info, cut_points: list, frame_duration=10) -> list: raw_vad_content = " ".join(["1" if i else "0" for i in raw_vad_info]) content = [] for cut_point in cut_points: line = [] for s in cut_point: s.frame_duration = frame_duration line.append(str(s)) content.append("|".join(line)) return raw_vad_content, "\n".join(content) def wavs_generator(raw_wav: numpy.array, cut_points: list, filename: str, sr: int, frame_duration: int) -> list: wavs = [] for idx, cp in enumerate(cut_points): clip = numpy.concatenate( [s.get_wav_seg(raw_wav, sr, frame_duration) for s in cp], axis=0 ) wavs.append((clip, f"{filename}_{idx}_{int(clip.shape[0]/sr*1000)}.wav")) return wavs