You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
PaddleSpeech/audio/audiotools/core/loudness.py

388 lines
13 KiB

# MIT License, Copyright (c) 2023-Present, Descript.
# Copyright (c) 2025 PaddlePaddle Authors. All Rights Reserved.
#
# Modified from audiotools(https://github.com/descriptinc/audiotools/blob/master/audiotools/core/loudness.py)
import copy
import math
import typing
import numpy as np
import paddle
import paddle.nn.functional as F
import scipy
from . import _julius
def _unfold1d(x, kernel_size, stride):
# https://github.com/PaddlePaddle/Paddle/pull/70102
"""1D only unfolding similar to the one from Paddlepaddle.
Given an _input tensor of size `[*, T]` this will return
a tensor `[*, F, K]` with `K` the kernel size, and `F` the number
of frames. The i-th frame is a view onto `i * stride: i * stride + kernel_size`.
This will automatically pad the _input to cover at least once all entries in `_input`.
Args:
_input (Tensor): tensor for which to return the frames.
kernel_size (int): size of each frame.
stride (int): stride between each frame.
Shape:
- Inputs: `_input` is `[*, T]`
- Output: `[*, F, kernel_size]` with `F = 1 + ceil((T - kernel_size) / stride)`
"""
if 3 != x.dim():
raise NotImplementedError
N, C, length = x.shape
x = x.reshape([N * C, 1, length])
n_frames = math.ceil((max(length, kernel_size) - kernel_size) / stride) + 1
tgt_length = (n_frames - 1) * stride + kernel_size
x = F.pad(x, (0, tgt_length - length), data_format="NCL")
x = x.unsqueeze(-1)
unfolded = paddle.nn.functional.unfold(
x,
kernel_sizes=[kernel_size, 1],
strides=[stride, 1], )
unfolded = unfolded.transpose([0, 2, 1])
unfolded = unfolded.reshape([N, C, *unfolded.shape[1:]])
return unfolded
class Meter(paddle.nn.Layer):
"""Tensorized version of pyloudnorm.Meter. Works with batched audio tensors.
Parameters
----------
rate : int
Sample rate of audio.
filter_class : str, optional
Class of weighting filter used.
K-weighting' (default), 'Fenton/Lee 1'
'Fenton/Lee 2', 'Dash et al.'
by default "K-weighting"
block_size : float, optional
Gating block size in seconds, by default 0.400
zeros : int, optional
Number of zeros to use in FIR approximation of
IIR filters, by default 512
use_fir : bool, optional
Whether to use FIR approximation or exact IIR formulation.
If computing on GPU, ``use_fir=True`` will be used, as its
much faster, by default False
"""
def __init__(
self,
rate: int,
filter_class: str="K-weighting",
block_size: float=0.400,
zeros: int=512,
use_fir: bool=False, ):
super().__init__()
self.rate = rate
self.filter_class = filter_class
self.block_size = block_size
self.use_fir = use_fir
G = paddle.to_tensor(
np.array([1.0, 1.0, 1.0, 1.41, 1.41]), stop_gradient=True)
self.register_buffer("G", G)
# Compute impulse responses so that filtering is fast via
# a convolution at runtime, on GPU, unlike lfilter.
impulse = np.zeros((zeros, ))
impulse[..., 0] = 1.0
firs = np.zeros((len(self._filters), 1, zeros))
# passband_gain = torch.zeros(len(self._filters))
passband_gain = paddle.zeros([len(self._filters)], dtype="float32")
for i, (_, filter_stage) in enumerate(self._filters.items()):
firs[i] = scipy.signal.lfilter(filter_stage.b, filter_stage.a,
impulse)
passband_gain[i] = filter_stage.passband_gain
firs = paddle.to_tensor(
firs[..., ::-1].copy(), dtype="float32", stop_gradient=True)
self.register_buffer("firs", firs)
self.register_buffer("passband_gain", passband_gain)
def apply_filter_gpu(self, data: paddle.Tensor):
"""Performs FIR approximation of loudness computation.
Parameters
----------
data : paddle.Tensor
Audio data of shape (nb, nch, nt).
Returns
-------
paddle.Tensor
Filtered audio data.
"""
# Data is of shape (nb, nch, nt)
# Reshape to (nb*nch, 1, nt)
nb, nt, nch = data.shape
data = data.transpose([0, 2, 1])
data = data.reshape([nb * nch, 1, nt])
# Apply padding
pad_length = self.firs.shape[-1]
# Apply filtering in sequence
for i in range(self.firs.shape[0]):
data = F.pad(data, (pad_length, pad_length), data_format="NCL")
data = _julius.fft_conv1d(data, self.firs[i, None, ...])
data = self.passband_gain[i] * data
data = data[..., 1:nt + 1]
data = data.transpose([0, 2, 1])
data = data[:, :nt, :]
return data
@staticmethod
def scipy_lfilter(waveform, a_coeffs, b_coeffs, clamp: bool=True):
# 使用 scipy.signal.lfilter 进行滤波(处理三维数据)
output = np.zeros_like(waveform)
for batch_idx in range(waveform.shape[0]):
for channel_idx in range(waveform.shape[2]):
output[batch_idx, :, channel_idx] = scipy.signal.lfilter(
b_coeffs, a_coeffs, waveform[batch_idx, :, channel_idx])
return output
def apply_filter_cpu(self, data: paddle.Tensor):
"""Performs IIR formulation of loudness computation.
Parameters
----------
data : paddle.Tensor
Audio data of shape (nb, nch, nt).
Returns
-------
paddle.Tensor
Filtered audio data.
"""
_data = data.cpu().numpy().copy()
for _, filter_stage in self._filters.items():
passband_gain = filter_stage.passband_gain
a_coeffs = filter_stage.a
b_coeffs = filter_stage.b
filtered = self.scipy_lfilter(_data, a_coeffs, b_coeffs)
_data[:] = passband_gain * filtered
data = paddle.to_tensor(_data)
return data
def apply_filter(self, data: paddle.Tensor):
"""Applies filter on either CPU or GPU, depending
on if the audio is on GPU or is on CPU, or if
``self.use_fir`` is True.
Parameters
----------
data : paddle.Tensor
Audio data of shape (nb, nch, nt).
Returns
-------
paddle.Tensor
Filtered audio data.
"""
# if data.place.is_gpu_place() or self.use_fir:
# data = self.apply_filter_gpu(data)
# else:
# data = self.apply_filter_cpu(data)
data = self.apply_filter_cpu(data)
return data
def forward(self, data: paddle.Tensor):
"""Computes integrated loudness of data.
Parameters
----------
data : paddle.Tensor
Audio data of shape (nb, nch, nt).
Returns
-------
paddle.Tensor
Filtered audio data.
"""
return self.integrated_loudness(data)
def _unfold(self, input_data):
T_g = self.block_size
overlap = 0.75 # overlap of 75% of the block duration
step = 1.0 - overlap # step size by percentage
kernel_size = int(T_g * self.rate)
stride = int(T_g * self.rate * step)
unfolded = _unfold1d(
input_data.transpose([0, 2, 1]), kernel_size, stride)
unfolded = unfolded.transpose([0, 1, 3, 2])
return unfolded
def integrated_loudness(self, data: paddle.Tensor):
"""Computes integrated loudness of data.
Parameters
----------
data : paddle.Tensor
Audio data of shape (nb, nch, nt).
Returns
-------
paddle.Tensor
Filtered audio data.
"""
if not paddle.is_tensor(data):
data = paddle.to_tensor(data, dtype="float32")
else:
data = data.astype("float32")
input_data = data.clone()
# Data always has a batch and channel dimension.
# Is of shape (nb, nt, nch)
if input_data.ndim < 2:
input_data = input_data.unsqueeze(-1)
if input_data.ndim < 3:
input_data = input_data.unsqueeze(0)
nb, nt, nch = input_data.shape
# Apply frequency weighting filters - account
# for the acoustic respose of the head and auditory system
input_data = self.apply_filter(input_data)
G = self.G # channel gains
T_g = self.block_size # 400 ms gating block standard
Gamma_a = -70.0 # -70 LKFS = absolute loudness threshold
unfolded = self._unfold(input_data)
z = (1.0 / (T_g * self.rate)) * unfolded.square().sum(2)
l = -0.691 + 10.0 * paddle.log10(
(G[None, :nch, None] * z).sum(1, keepdim=True))
l = l.expand_as(z)
# find gating block indices above absolute threshold
z_avg_gated = z
z_avg_gated[l <= Gamma_a] = 0
masked = l > Gamma_a
z_avg_gated = z_avg_gated.sum(2) / masked.sum(2).astype("float32")
# calculate the relative threshold value (see eq. 6)
Gamma_r = -0.691 + 10.0 * paddle.log10(
(z_avg_gated * G[None, :nch]).sum(-1)) - 10.0
Gamma_r = Gamma_r[:, None, None]
Gamma_r = Gamma_r.expand([nb, nch, l.shape[-1]])
# find gating block indices above relative and absolute thresholds (end of eq. 7)
z_avg_gated = z
z_avg_gated[l <= Gamma_a] = 0
z_avg_gated[l <= Gamma_r] = 0
masked = (l > Gamma_a) * (l > Gamma_r)
z_avg_gated = z_avg_gated.sum(2) / (masked.sum(2) + 10e-6)
# TODO Currently, paddle has a segmentation fault bug in this section of the code
# z_avg_gated = paddle.nan_to_num(z_avg_gated)
# z_avg_gated = paddle.where(
# paddle.isnan(z_avg_gated),
# paddle.zeros_like(z_avg_gated), z_avg_gated)
z_avg_gated[z_avg_gated == float("inf")] = float(
np.finfo(np.float32).max)
z_avg_gated[z_avg_gated == -float("inf")] = float(
np.finfo(np.float32).min)
LUFS = -0.691 + 10.0 * paddle.log10(
(G[None, :nch] * z_avg_gated).sum(1))
return LUFS.astype("float32")
@property
def filter_class(self):
return self._filter_class
@filter_class.setter
def filter_class(self, value):
from pyloudnorm import Meter
meter = Meter(self.rate)
meter.filter_class = value
self._filter_class = value
self._filters = meter._filters
class LoudnessMixin:
_loudness = None
MIN_LOUDNESS = -70
"""Minimum loudness possible."""
def loudness(self,
filter_class: str="K-weighting",
block_size: float=0.400,
**kwargs):
"""Calculates loudness using an implementation of ITU-R BS.1770-4.
Allows control over gating block size and frequency weighting filters for
additional control. Measure the integrated gated loudness of a signal.
API is derived from PyLoudnorm, but this implementation is ported to PyTorch
and is tensorized across batches. When on GPU, an FIR approximation of the IIR
filters is used to compute loudness for speed.
Uses the weighting filters and block size defined by the meter
the integrated loudness is measured based upon the gating algorithm
defined in the ITU-R BS.1770-4 specification.
Parameters
----------
filter_class : str, optional
Class of weighting filter used.
K-weighting' (default), 'Fenton/Lee 1'
'Fenton/Lee 2', 'Dash et al.'
by default "K-weighting"
block_size : float, optional
Gating block size in seconds, by default 0.400
kwargs : dict, optional
Keyword arguments to :py:func:`audiotools.core.loudness.Meter`.
Returns
-------
paddle.Tensor
Loudness of audio data.
"""
if self._loudness is not None:
return self._loudness # .to(self.device)
original_length = self.signal_length
if self.signal_duration < 0.5:
pad_len = int((0.5 - self.signal_duration) * self.sample_rate)
self.zero_pad(0, pad_len)
# create BS.1770 meter
meter = Meter(
self.sample_rate,
filter_class=filter_class,
block_size=block_size,
**kwargs)
# meter = meter.to(self.device)
# measure loudness
loudness = meter.integrated_loudness(
self.audio_data.transpose([0, 2, 1]))
self.truncate_samples(original_length)
min_loudness = paddle.ones_like(loudness) * self.MIN_LOUDNESS
self._loudness = paddle.maximum(loudness, min_loudness)
return self._loudness # .to(self.device)