目录

音频重采样

作者Caroline ChenMoto Hira

本教程介绍如何使用 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

制备

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

要将音频波形从一个频率重新采样到另一个频率,可以使用 。 precomputes 并缓存用于重采样的内核, while 会动态计算它,因此 using 将导致在重新采样时加速 使用相同参数的多个波形(参见 基准测试 部分)。transforms.Resamplefunctional.resampletorchaudio.transforms.Resample

两种重采样方法都使用带限 sinc 要计算的插值 信号值。实现涉及 卷积,因此我们可以利用 GPU / 多线程 性能改进。

注意

在多个子进程中使用重采样时,例如数据加载 使用多个工作进程,您的应用程序可能会创建更多 线程,您的系统无法有效处理。 在这种情况下,设置可能会有所帮助。torch.set_num_threads(1)

因为有限数量的样本只能代表有限数量的 频率、重新采样不会产生完美的结果,并且 of 参数可用于控制其质量和计算 速度。我们通过对数 正弦扫频,这是一个在 频率随时间的变化。

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(对数刻度),y 轴频率 绘制波形,颜色强度为振幅。

sample_rate = 48000
waveform = get_sine_sweep(sample_rate)

plot_sweep(waveform, sample_rate, title="Original Waveform")
Audio(waveform.numpy()[0], rate=sample_rate)
原始波形(采样率:48000 Hz)


现在我们对其进行重新采样 (downsample)。

我们看到,在重采样波形的频谱图中,有一个 伪影,这在原始波形中不存在。 这种效果称为锯齿。此页面有 解释它是如何发生的,以及为什么它看起来像一个反射。

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)
重新采样的波形(采样率: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,因此该参数用于控制 用于对插值进行窗口化的 filter。它也被称为 自插值通过 在每个时间单位上为零。使用较大的滤波器更清晰、更精确,但计算量更大 贵。lowpass_filter_widthlowpass_filter_width

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(采样率: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(采样率:32000 Hz)

滚降

该参数表示为奈奎斯特频率的分数 frequency,即给定的 有限采样率。 确定低通滤波器截止和 控制混叠的程度,当频率 高于奈奎斯特频率的 Nyquist 频率被映射到较低的频率。下滚降 因此,将减少锯齿的数量,但它也会减少 一些更高的频率。rolloffrolloff

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

Window 函数

默认情况下,的 resample 使用 Hann 窗口过滤器,即 加权余弦函数。它还支持 Kaiser 窗口、 ,这是一个近乎最优的窗口函数,它包含一个额外的参数,该参数允许设计 filter 和 impulse 的宽度。这可以使用 parameter 进行控制。torchaudiobetaresampling_method

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 默认(采样率: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(采样率:32000 Hz)

与 librosa 的比较

torchaudio的 resample 函数可用于生成类似于 Librosa (Resampy) 的 Kaiser 窗口重新采样,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) (采样率: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) (采样率: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)(采样率: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)(采样率: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 两对采样率。我们展示了性能影响 、窗口类型和采样率可以 有。此外,我们还提供了与 的比较,并使用它们的相应参数 在。lowpass_filter_widthlibrosakaiser_bestkaiser_fasttorchaudio

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")

下采样 (48 -> 44.1 kHz)

df = benchmark(48_000, 44_100)
plot(df)
音频重采样教程
                 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

下采样 (16 -> 8 kHz)

df = benchmark(16_000, 8_000)
plot(df)
音频重采样教程
                 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

上采样 (44.1 -> 48 kHz)

df = benchmark(44_100, 48_000)
plot(df)
音频重采样教程
                 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

上采样 (8 -> 16 kHz)

df = benchmark(8_000, 16_000)
plot(df)
音频重采样教程
                 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:

  • 较大的结果会产生较大的重采样核, 因此增加了内核计算的计算时间 和卷积lowpass_filter_width

  • using 会导致计算时间比 default 长,因为计算中间 窗口值sinc_interp_kaisersinc_interp_hann

  • 采样率和重采样率之间的 GCD 较大 在允许更小的内核和更快的内核计算的简化中。

脚本总运行时间:(0 分 3.361 秒)

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