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音频重采样

作者: Caroline Chen, Moto Hira

本教程展示了如何使用 torchaudio 的重采样 API。

import torch
import torchaudio
import torchaudio.functional as F
import torchaudio.transforms as T

print(torch.__version__)
print(torchaudio.__version__)
2.5.0
2.5.0

准备

首先,我们导入模块并定义辅助函数。

import math
import timeit

import librosa
import matplotlib.colors as mcolors
import matplotlib.pyplot as plt
import pandas as pd
import resampy
from IPython.display import Audio

pd.set_option("display.max_rows", None)
pd.set_option("display.max_columns", None)

DEFAULT_OFFSET = 201


def _get_log_freq(sample_rate, max_sweep_rate, offset):
    """Get freqs evenly spaced out in log-scale, between [0, max_sweep_rate // 2]

    offset is used to avoid negative infinity `log(offset + x)`.

    """
    start, stop = math.log(offset), math.log(offset + max_sweep_rate // 2)
    return torch.exp(torch.linspace(start, stop, sample_rate, dtype=torch.double)) - offset


def _get_inverse_log_freq(freq, sample_rate, offset):
    """Find the time where the given frequency is given by _get_log_freq"""
    half = sample_rate // 2
    return sample_rate * (math.log(1 + freq / offset) / math.log(1 + half / offset))


def _get_freq_ticks(sample_rate, offset, f_max):
    # Given the original sample rate used for generating the sweep,
    # find the x-axis value where the log-scale major frequency values fall in
    times, freq = [], []
    for exp in range(2, 5):
        for v in range(1, 10):
            f = v * 10**exp
            if f < sample_rate // 2:
                t = _get_inverse_log_freq(f, sample_rate, offset) / sample_rate
                times.append(t)
                freq.append(f)
    t_max = _get_inverse_log_freq(f_max, sample_rate, offset) / sample_rate
    times.append(t_max)
    freq.append(f_max)
    return times, freq


def get_sine_sweep(sample_rate, offset=DEFAULT_OFFSET):
    max_sweep_rate = sample_rate
    freq = _get_log_freq(sample_rate, max_sweep_rate, offset)
    delta = 2 * math.pi * freq / sample_rate
    cummulative = torch.cumsum(delta, dim=0)
    signal = torch.sin(cummulative).unsqueeze(dim=0)
    return signal


def plot_sweep(
    waveform,
    sample_rate,
    title,
    max_sweep_rate=48000,
    offset=DEFAULT_OFFSET,
):
    x_ticks = [100, 500, 1000, 5000, 10000, 20000, max_sweep_rate // 2]
    y_ticks = [1000, 5000, 10000, 20000, sample_rate // 2]

    time, freq = _get_freq_ticks(max_sweep_rate, offset, sample_rate // 2)
    freq_x = [f if f in x_ticks and f <= max_sweep_rate // 2 else None for f in freq]
    freq_y = [f for f in freq if f in y_ticks and 1000 <= f <= sample_rate // 2]

    figure, axis = plt.subplots(1, 1)
    _, _, _, cax = axis.specgram(waveform[0].numpy(), Fs=sample_rate)
    plt.xticks(time, freq_x)
    plt.yticks(freq_y, freq_y)
    axis.set_xlabel("Original Signal Frequency (Hz, log scale)")
    axis.set_ylabel("Waveform Frequency (Hz)")
    axis.xaxis.grid(True, alpha=0.67)
    axis.yaxis.grid(True, alpha=0.67)
    figure.suptitle(f"{title} (sample rate: {sample_rate} Hz)")
    plt.colorbar(cax)

重采样概述

要将音频波形从一种频率重采样到另一种频率,可以使用 torchaudio.transforms.Resampletorchaudio.functional.resample()transforms.Resample 预先计算并缓存用于重采样的内核,而 functional.resample 则在运行时计算内核,因此当使用相同参数对多个波形进行重采样时,使用 torchaudio.transforms.Resample 会加快速度(参见基准测试部分)。

两种重采样方法都使用 带限 sinc 插值 来计算任意时间步长的信号值。该实现涉及卷积,因此我们可以利用 GPU/多线程来提高性能。

注意

当在多个子进程中使用重采样时,例如使用多个工作进程进行数据加载,您的应用程序可能会创建超出系统有效处理能力的线程数量。在这种情况下,设置 torch.set_num_threads(1) 可能会有所帮助。

由于有限数量的样本只能表示有限数量的频率,因此重采样不会产生完美的结果,可以使用各种参数来控制其质量和计算速度。我们通过对对数正弦扫描进行重采样来展示这些属性,对数正弦扫描是在时间内频率呈指数增长的正弦波。

下面的频谱图显示了信号的频率表示,其中 x 轴对应于原始波形的频率(对数刻度),y 轴对应于绘制波形的频率,颜色强度对应于幅度。

sample_rate = 48000
waveform = get_sine_sweep(sample_rate)

plot_sweep(waveform, sample_rate, title="Original Waveform")
Audio(waveform.numpy()[0], rate=sample_rate)
Original Waveform (sample rate: 48000 Hz)


现在我们对其进行重采样(降采样)。

我们看到,在重采样波形的频谱图中,出现了一个在原始波形中不存在的伪像。这种效应称为混叠。 此页面 解释了混叠是如何发生的,以及为什么它看起来像反射。

resample_rate = 32000
resampler = T.Resample(sample_rate, resample_rate, dtype=waveform.dtype)
resampled_waveform = resampler(waveform)

plot_sweep(resampled_waveform, resample_rate, title="Resampled Waveform")
Audio(resampled_waveform.numpy()[0], rate=resample_rate)
Resampled Waveform (sample rate: 32000 Hz)


使用参数控制重采样质量

低通滤波器宽度

由于用于插值的滤波器无限延伸,因此使用 lowpass_filter_width 参数来控制要用于对插值进行加窗的滤波器的宽度。它也称为零交叉次数,因为插值在每个时间单位处都穿过零。使用更大的 lowpass_filter_width 会提供更锐利、更精确的滤波器,但计算成本更高。

sample_rate = 48000
resample_rate = 32000

resampled_waveform = F.resample(waveform, sample_rate, resample_rate, lowpass_filter_width=6)
plot_sweep(resampled_waveform, resample_rate, title="lowpass_filter_width=6")
lowpass_filter_width=6 (sample rate: 32000 Hz)
resampled_waveform = F.resample(waveform, sample_rate, resample_rate, lowpass_filter_width=128)
plot_sweep(resampled_waveform, resample_rate, title="lowpass_filter_width=128")
lowpass_filter_width=128 (sample rate: 32000 Hz)

滚降

rolloff 参数用奈奎斯特频率的几分之一表示,奈奎斯特频率是给定有限采样率可表示的最大频率。rolloff 决定低通滤波器的截止频率,并控制混叠程度,混叠是指高于奈奎斯特频率的频率映射到较低频率时的现象。因此,较低的滚降会减少混叠量,但它也会降低一些较高频率。

sample_rate = 48000
resample_rate = 32000

resampled_waveform = F.resample(waveform, sample_rate, resample_rate, rolloff=0.99)
plot_sweep(resampled_waveform, resample_rate, title="rolloff=0.99")
rolloff=0.99 (sample rate: 32000 Hz)
resampled_waveform = F.resample(waveform, sample_rate, resample_rate, rolloff=0.8)
plot_sweep(resampled_waveform, resample_rate, title="rolloff=0.8")
rolloff=0.8 (sample rate: 32000 Hz)

窗函数

默认情况下,torchaudio 的重采样使用汉宁窗滤波器,汉宁窗滤波器是一种加权余弦函数。它还支持凯泽窗,凯泽窗是一种接近最佳的窗函数,它包含一个额外的 beta 参数,允许设计滤波器的平滑度和脉冲宽度。可以使用 resampling_method 参数来控制这一点。

sample_rate = 48000
resample_rate = 32000

resampled_waveform = F.resample(waveform, sample_rate, resample_rate, resampling_method="sinc_interp_hann")
plot_sweep(resampled_waveform, resample_rate, title="Hann Window Default")
Hann Window Default (sample rate: 32000 Hz)
resampled_waveform = F.resample(waveform, sample_rate, resample_rate, resampling_method="sinc_interp_kaiser")
plot_sweep(resampled_waveform, resample_rate, title="Kaiser Window Default")
Kaiser Window Default (sample rate: 32000 Hz)

与 librosa 的比较

torchaudio 的重采样函数可用于产生与 librosa (resampy) 的凯泽窗重采样结果类似的结果,但会带有一些噪声

sample_rate = 48000
resample_rate = 32000

kaiser_best

resampled_waveform = F.resample(
    waveform,
    sample_rate,
    resample_rate,
    lowpass_filter_width=64,
    rolloff=0.9475937167399596,
    resampling_method="sinc_interp_kaiser",
    beta=14.769656459379492,
)
plot_sweep(resampled_waveform, resample_rate, title="Kaiser Window Best (torchaudio)")
Kaiser Window Best (torchaudio) (sample rate: 32000 Hz)
librosa_resampled_waveform = torch.from_numpy(
    librosa.resample(waveform.squeeze().numpy(), orig_sr=sample_rate, target_sr=resample_rate, res_type="kaiser_best")
).unsqueeze(0)
plot_sweep(librosa_resampled_waveform, resample_rate, title="Kaiser Window Best (librosa)")
Kaiser Window Best (librosa) (sample rate: 32000 Hz)
mse = torch.square(resampled_waveform - librosa_resampled_waveform).mean().item()
print("torchaudio and librosa kaiser best MSE:", mse)
torchaudio and librosa kaiser best MSE: 2.0806901153660115e-06

kaiser_fast

resampled_waveform = F.resample(
    waveform,
    sample_rate,
    resample_rate,
    lowpass_filter_width=16,
    rolloff=0.85,
    resampling_method="sinc_interp_kaiser",
    beta=8.555504641634386,
)
plot_sweep(resampled_waveform, resample_rate, title="Kaiser Window Fast (torchaudio)")
Kaiser Window Fast (torchaudio) (sample rate: 32000 Hz)
librosa_resampled_waveform = torch.from_numpy(
    librosa.resample(waveform.squeeze().numpy(), orig_sr=sample_rate, target_sr=resample_rate, res_type="kaiser_fast")
).unsqueeze(0)
plot_sweep(librosa_resampled_waveform, resample_rate, title="Kaiser Window Fast (librosa)")
Kaiser Window Fast (librosa) (sample rate: 32000 Hz)
mse = torch.square(resampled_waveform - librosa_resampled_waveform).mean().item()
print("torchaudio and librosa kaiser fast MSE:", mse)
torchaudio and librosa kaiser fast MSE: 2.5200744248601437e-05

性能基准测试

以下是针对两对采样率之间的波形降采样和升采样的基准测试。我们展示了 lowpass_filter_width、窗类型和采样率可能带来的性能影响。此外,我们还使用 torchaudio 中相应的参数,与 librosakaiser_bestkaiser_fast 进行了比较。

print(f"torchaudio: {torchaudio.__version__}")
print(f"librosa: {librosa.__version__}")
print(f"resampy: {resampy.__version__}")
torchaudio: 2.5.0
librosa: 0.10.0
resampy: 0.2.2
def benchmark_resample_functional(
    waveform,
    sample_rate,
    resample_rate,
    lowpass_filter_width=6,
    rolloff=0.99,
    resampling_method="sinc_interp_hann",
    beta=None,
    iters=5,
):
    return (
        timeit.timeit(
            stmt="""
torchaudio.functional.resample(
    waveform,
    sample_rate,
    resample_rate,
    lowpass_filter_width=lowpass_filter_width,
    rolloff=rolloff,
    resampling_method=resampling_method,
    beta=beta,
)
        """,
            setup="import torchaudio",
            number=iters,
            globals=locals(),
        )
        * 1000
        / iters
    )
def benchmark_resample_transforms(
    waveform,
    sample_rate,
    resample_rate,
    lowpass_filter_width=6,
    rolloff=0.99,
    resampling_method="sinc_interp_hann",
    beta=None,
    iters=5,
):
    return (
        timeit.timeit(
            stmt="resampler(waveform)",
            setup="""
import torchaudio

resampler = torchaudio.transforms.Resample(
    sample_rate,
    resample_rate,
    lowpass_filter_width=lowpass_filter_width,
    rolloff=rolloff,
    resampling_method=resampling_method,
    dtype=waveform.dtype,
    beta=beta,
)
resampler.to(waveform.device)
        """,
            number=iters,
            globals=locals(),
        )
        * 1000
        / iters
    )
def benchmark_resample_librosa(
    waveform,
    sample_rate,
    resample_rate,
    res_type=None,
    iters=5,
):
    waveform_np = waveform.squeeze().numpy()
    return (
        timeit.timeit(
            stmt="""
librosa.resample(
    waveform_np,
    orig_sr=sample_rate,
    target_sr=resample_rate,
    res_type=res_type,
)
        """,
            setup="import librosa",
            number=iters,
            globals=locals(),
        )
        * 1000
        / iters
    )
def benchmark(sample_rate, resample_rate):
    times, rows = [], []
    waveform = get_sine_sweep(sample_rate).to(torch.float32)

    args = (waveform, sample_rate, resample_rate)

    # sinc 64 zero-crossings
    f_time = benchmark_resample_functional(*args, lowpass_filter_width=64)
    t_time = benchmark_resample_transforms(*args, lowpass_filter_width=64)
    times.append([None, f_time, t_time])
    rows.append("sinc (width 64)")

    # sinc 6 zero-crossings
    f_time = benchmark_resample_functional(*args, lowpass_filter_width=16)
    t_time = benchmark_resample_transforms(*args, lowpass_filter_width=16)
    times.append([None, f_time, t_time])
    rows.append("sinc (width 16)")

    # kaiser best
    kwargs = {
        "lowpass_filter_width": 64,
        "rolloff": 0.9475937167399596,
        "resampling_method": "sinc_interp_kaiser",
        "beta": 14.769656459379492,
    }
    lib_time = benchmark_resample_librosa(*args, res_type="kaiser_best")
    f_time = benchmark_resample_functional(*args, **kwargs)
    t_time = benchmark_resample_transforms(*args, **kwargs)
    times.append([lib_time, f_time, t_time])
    rows.append("kaiser_best")

    # kaiser fast
    kwargs = {
        "lowpass_filter_width": 16,
        "rolloff": 0.85,
        "resampling_method": "sinc_interp_kaiser",
        "beta": 8.555504641634386,
    }
    lib_time = benchmark_resample_librosa(*args, res_type="kaiser_fast")
    f_time = benchmark_resample_functional(*args, **kwargs)
    t_time = benchmark_resample_transforms(*args, **kwargs)
    times.append([lib_time, f_time, t_time])
    rows.append("kaiser_fast")

    df = pd.DataFrame(times, columns=["librosa", "functional", "transforms"], index=rows)
    return df
def plot(df):
    print(df.round(2))
    ax = df.plot(kind="bar")
    plt.ylabel("Time Elapsed [ms]")
    plt.xticks(rotation=0, fontsize=10)
    for cont, col, color in zip(ax.containers, df.columns, mcolors.TABLEAU_COLORS):
        label = ["N/A" if v != v else str(v) for v in df[col].round(2)]
        ax.bar_label(cont, labels=label, color=color, fontweight="bold", fontsize="x-small")

降采样 (48 -> 44.1 kHz)

df = benchmark(48_000, 44_100)
plot(df)
audio resampling tutorial
                 librosa  functional  transforms
sinc (width 64)      NaN        0.90        0.40
sinc (width 16)      NaN        0.72        0.35
kaiser_best        83.91        1.21        0.38
kaiser_fast         7.89        0.95        0.34

降采样 (16 -> 8 kHz)

df = benchmark(16_000, 8_000)
plot(df)
audio resampling tutorial
                 librosa  functional  transforms
sinc (width 64)      NaN        1.29        1.10
sinc (width 16)      NaN        0.54        0.37
kaiser_best        11.29        1.36        1.17
kaiser_fast         3.14        0.67        0.41

升采样 (44.1 -> 48 kHz)

df = benchmark(44_100, 48_000)
plot(df)
audio resampling tutorial
                 librosa  functional  transforms
sinc (width 64)      NaN        0.87        0.36
sinc (width 16)      NaN        0.70        0.34
kaiser_best        32.74        1.14        0.38
kaiser_fast         7.88        0.94        0.34

升采样 (8 -> 16 kHz)

df = benchmark(8_000, 16_000)
plot(df)
audio resampling tutorial
                 librosa  functional  transforms
sinc (width 64)      NaN        0.70        0.46
sinc (width 16)      NaN        0.38        0.22
kaiser_best        11.24        0.71        0.48
kaiser_fast         2.99        0.41        0.24

总结

详细说明结果

  • 更大的 lowpass_filter_width 会导致更大的重采样内核,因此会增加内核计算和卷积的计算时间

  • 使用 sinc_interp_kaiser 的计算时间比默认的 sinc_interp_hann 更长,因为计算中间窗值更复杂

  • 采样率和重采样率之间的最大公约数越大,简化程度就越高,从而可以产生更小的内核和更快的内核计算速度。

脚本的总运行时间:(0 分钟 3.361 秒)

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