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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
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