torchaudio 2.9.1__cp310-cp310-macosx_11_0_arm64.whl

torchaudio/functional/filtering.py

@@ -0,0 +1,1685 @@
+import math
+import warnings
+from typing import Optional
+
+import torch
+import torch.nn.functional as F
+from torch import Tensor
+
+from torchaudio._extension import _IS_TORCHAUDIO_EXT_AVAILABLE
+
+
+def _dB2Linear(x: float) -> float:
+    return math.exp(x * math.log(10) / 20.0)
+
+
+def _generate_wave_table(
+    wave_type: str,
+    data_type: str,
+    table_size: int,
+    min: float,
+    max: float,
+    phase: float,
+    device: torch.device,
+) -> Tensor:
+    r"""A helper function for phaser. Generates a table with given parameters.
+
+    Args:
+        wave_type (str): SINE or TRIANGULAR
+        data_type (str): desired data_type ( `INT` or `FLOAT` )
+        table_size (int): desired table size
+        min (float): desired min value
+        max (float): desired max value
+        phase (float): desired phase
+        device (torch.device): Torch device on which table must be generated
+    Returns:
+        Tensor: A 1D tensor with wave table values
+    """
+
+    phase_offset = int(phase / math.pi / 2 * table_size + 0.5)
+
+    t = torch.arange(table_size, device=device, dtype=torch.int32)
+
+    point = (t + phase_offset) % table_size
+
+    d = torch.zeros_like(point, device=device, dtype=torch.float64)
+
+    if wave_type == "SINE":
+        d = (torch.sin(point.to(torch.float64) / table_size * 2 * math.pi) + 1) / 2
+    elif wave_type == "TRIANGLE":
+        d = point.to(torch.float64) * 2 / table_size
+        value = torch.div(4 * point, table_size, rounding_mode="floor")
+        d[value == 0] = d[value == 0] + 0.5
+        d[value == 1] = 1.5 - d[value == 1]
+        d[value == 2] = 1.5 - d[value == 2]
+        d[value == 3] = d[value == 3] - 1.5
+
+    d = d * (max - min) + min
+
+    if data_type == "INT":
+        mask = d < 0
+        d[mask] = d[mask] - 0.5
+        d[~mask] = d[~mask] + 0.5
+        d = d.to(torch.int32)
+    elif data_type == "FLOAT":
+        d = d.to(torch.float32)
+
+    return d
+
+
+def allpass_biquad(waveform: Tensor, sample_rate: int, central_freq: float, Q: float = 0.707) -> Tensor:
+    r"""Design two-pole all-pass filter.  Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Args:
+        waveform(torch.Tensor): audio waveform of dimension of `(..., time)`
+        sample_rate (int): sampling rate of the waveform, e.g. 44100 (Hz)
+        central_freq (float or torch.Tensor): central frequency (in Hz)
+        Q (float or torch.Tensor, optional): https://en.wikipedia.org/wiki/Q_factor (Default: ``0.707``)
+
+    Returns:
+        Tensor: Waveform of dimension of `(..., time)`
+
+    Reference:
+        - http://sox.sourceforge.net/sox.html
+        - https://www.w3.org/2011/audio/audio-eq-cookbook.html#APF
+    """
+    dtype = waveform.dtype
+    device = waveform.device
+    central_freq = torch.as_tensor(central_freq, dtype=dtype, device=device)
+    Q = torch.as_tensor(Q, dtype=dtype, device=device)
+
+    w0 = 2 * math.pi * central_freq / sample_rate
+
+    alpha = torch.sin(w0) / 2 / Q
+
+    b0 = 1 - alpha
+    b1 = -2 * torch.cos(w0)
+    b2 = 1 + alpha
+    a0 = 1 + alpha
+    a1 = -2 * torch.cos(w0)
+    a2 = 1 - alpha
+    return biquad(waveform, b0, b1, b2, a0, a1, a2)
+
+
+def band_biquad(
+    waveform: Tensor,
+    sample_rate: int,
+    central_freq: float,
+    Q: float = 0.707,
+    noise: bool = False,
+) -> Tensor:
+    r"""Design two-pole band filter.  Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`
+        sample_rate (int): sampling rate of the waveform, e.g. 44100 (Hz)
+        central_freq (float or torch.Tensor): central frequency (in Hz)
+        Q (float or torch.Tensor, optional): https://en.wikipedia.org/wiki/Q_factor (Default: ``0.707``).
+        noise (bool, optional) : If ``True``, uses the alternate mode for un-pitched audio (e.g. percussion).
+            If ``False``, uses mode oriented to pitched audio, i.e. voice, singing,
+            or instrumental music (Default: ``False``).
+
+    Returns:
+        Tensor: Waveform of dimension of `(..., time)`
+
+    Reference:
+        - http://sox.sourceforge.net/sox.html
+        - https://www.w3.org/2011/audio/audio-eq-cookbook.html#APF
+    """
+    dtype = waveform.dtype
+    device = waveform.device
+    central_freq = torch.as_tensor(central_freq, dtype=dtype, device=device)
+    Q = torch.as_tensor(Q, dtype=dtype, device=device)
+
+    w0 = 2 * math.pi * central_freq / sample_rate
+    bw_Hz = central_freq / Q
+
+    a0 = 1.0
+    a2 = torch.exp(-2 * math.pi * bw_Hz / sample_rate)
+    a1 = -4 * a2 / (1 + a2) * torch.cos(w0)
+
+    b0 = torch.sqrt(1 - a1 * a1 / (4 * a2)) * (1 - a2)
+
+    if noise:
+        mult = torch.sqrt(((1 + a2) * (1 + a2) - a1 * a1) * (1 - a2) / (1 + a2)) / b0
+        b0 = mult * b0
+
+    b1 = 0.0
+    b2 = 0.0
+
+    return biquad(waveform, b0, b1, b2, a0, a1, a2)
+
+
+def bandpass_biquad(
+    waveform: Tensor,
+    sample_rate: int,
+    central_freq: float,
+    Q: float = 0.707,
+    const_skirt_gain: bool = False,
+) -> Tensor:
+    r"""Design two-pole band-pass filter.  Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`
+        sample_rate (int): sampling rate of the waveform, e.g. 44100 (Hz)
+        central_freq (float or torch.Tensor): central frequency (in Hz)
+        Q (float or torch.Tensor, optional): https://en.wikipedia.org/wiki/Q_factor (Default: ``0.707``)
+        const_skirt_gain (bool, optional) : If ``True``, uses a constant skirt gain (peak gain = Q).
+            If ``False``, uses a constant 0dB peak gain. (Default: ``False``)
+
+    Returns:
+        Tensor: Waveform of dimension of `(..., time)`
+
+    Reference:
+        - http://sox.sourceforge.net/sox.html
+        - https://www.w3.org/2011/audio/audio-eq-cookbook.html#APF
+    """
+    dtype = waveform.dtype
+    device = waveform.device
+    central_freq = torch.as_tensor(central_freq, dtype=dtype, device=device)
+    Q = torch.as_tensor(Q, dtype=dtype, device=device)
+
+    w0 = 2 * math.pi * central_freq / sample_rate
+    alpha = torch.sin(w0) / 2 / Q
+
+    temp = torch.sin(w0) / 2 if const_skirt_gain else alpha
+    b0 = temp
+    b1 = 0.0
+    b2 = -temp
+    a0 = 1 + alpha
+    a1 = -2 * torch.cos(w0)
+    a2 = 1 - alpha
+    return biquad(waveform, b0, b1, b2, a0, a1, a2)
+
+
+def bandreject_biquad(waveform: Tensor, sample_rate: int, central_freq: float, Q: float = 0.707) -> Tensor:
+    r"""Design two-pole band-reject filter.  Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`
+        sample_rate (int): sampling rate of the waveform, e.g. 44100 (Hz)
+        central_freq (float or torch.Tensor): central frequency (in Hz)
+        Q (float or torch.Tensor, optional): https://en.wikipedia.org/wiki/Q_factor (Default: ``0.707``)
+
+    Returns:
+        Tensor: Waveform of dimension of `(..., time)`
+
+    Reference:
+        - http://sox.sourceforge.net/sox.html
+        - https://www.w3.org/2011/audio/audio-eq-cookbook.html#APF
+    """
+    dtype = waveform.dtype
+    device = waveform.device
+    central_freq = torch.as_tensor(central_freq, dtype=dtype, device=device)
+    Q = torch.as_tensor(Q, dtype=dtype, device=device)
+
+    w0 = 2 * math.pi * central_freq / sample_rate
+    alpha = torch.sin(w0) / 2 / Q
+
+    b0 = 1.0
+    b1 = -2 * torch.cos(w0)
+    b2 = 1.0
+    a0 = 1 + alpha
+    a1 = -2 * torch.cos(w0)
+    a2 = 1 - alpha
+    return biquad(waveform, b0, b1, b2, a0, a1, a2)
+
+
+def bass_biquad(
+    waveform: Tensor,
+    sample_rate: int,
+    gain: float,
+    central_freq: float = 100,
+    Q: float = 0.707,
+) -> Tensor:
+    r"""Design a bass tone-control effect.  Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`
+        sample_rate (int): sampling rate of the waveform, e.g. 44100 (Hz)
+        gain (float or torch.Tensor): desired gain at the boost (or attenuation) in dB.
+        central_freq (float or torch.Tensor, optional): central frequency (in Hz). (Default: ``100``)
+        Q (float or torch.Tensor, optional): https://en.wikipedia.org/wiki/Q_factor (Default: ``0.707``).
+
+    Returns:
+        Tensor: Waveform of dimension of `(..., time)`
+
+    Reference:
+        - http://sox.sourceforge.net/sox.html
+        - https://www.w3.org/2011/audio/audio-eq-cookbook.html#APF
+    """
+    dtype = waveform.dtype
+    device = waveform.device
+    central_freq = torch.as_tensor(central_freq, dtype=dtype, device=device)
+    Q = torch.as_tensor(Q, dtype=dtype, device=device)
+    gain = torch.as_tensor(gain, dtype=dtype, device=device)
+
+    w0 = 2 * math.pi * central_freq / sample_rate
+    alpha = torch.sin(w0) / 2 / Q
+    A = torch.exp(gain / 40 * math.log(10))
+
+    temp1 = 2 * torch.sqrt(A) * alpha
+    temp2 = (A - 1) * torch.cos(w0)
+    temp3 = (A + 1) * torch.cos(w0)
+
+    b0 = A * ((A + 1) - temp2 + temp1)
+    b1 = 2 * A * ((A - 1) - temp3)
+    b2 = A * ((A + 1) - temp2 - temp1)
+    a0 = (A + 1) + temp2 + temp1
+    a1 = -2 * ((A - 1) + temp3)
+    a2 = (A + 1) + temp2 - temp1
+
+    return biquad(waveform, b0 / a0, b1 / a0, b2 / a0, a0 / a0, a1 / a0, a2 / a0)
+
+
+def biquad(waveform: Tensor, b0: float, b1: float, b2: float, a0: float, a1: float, a2: float) -> Tensor:
+    r"""Perform a biquad filter of input tensor.  Initial conditions set to 0.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`
+        b0 (float or torch.Tensor): numerator coefficient of current input, x[n]
+        b1 (float or torch.Tensor): numerator coefficient of input one time step ago x[n-1]
+        b2 (float or torch.Tensor): numerator coefficient of input two time steps ago x[n-2]
+        a0 (float or torch.Tensor): denominator coefficient of current output y[n], typically 1
+        a1 (float or torch.Tensor): denominator coefficient of current output y[n-1]
+        a2 (float or torch.Tensor): denominator coefficient of current output y[n-2]
+
+    Returns:
+        Tensor: Waveform with dimension of `(..., time)`
+
+    Reference:
+       - https://en.wikipedia.org/wiki/Digital_biquad_filter
+    """
+
+    device = waveform.device
+    dtype = waveform.dtype
+
+    b0 = torch.as_tensor(b0, dtype=dtype, device=device).view(1)
+    b1 = torch.as_tensor(b1, dtype=dtype, device=device).view(1)
+    b2 = torch.as_tensor(b2, dtype=dtype, device=device).view(1)
+    a0 = torch.as_tensor(a0, dtype=dtype, device=device).view(1)
+    a1 = torch.as_tensor(a1, dtype=dtype, device=device).view(1)
+    a2 = torch.as_tensor(a2, dtype=dtype, device=device).view(1)
+
+    output_waveform = lfilter(
+        waveform,
+        torch.cat([a0, a1, a2]),
+        torch.cat([b0, b1, b2]),
+    )
+    return output_waveform
+
+
+def contrast(waveform: Tensor, enhancement_amount: float = 75.0) -> Tensor:
+    r"""Apply contrast effect.  Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Comparable with compression, this effect modifies an audio signal to make it sound louder
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`
+        enhancement_amount (float, optional): controls the amount of the enhancement
+            Allowed range of values for enhancement_amount : 0-100
+            Note that enhancement_amount = 0 still gives a significant contrast enhancement
+
+    Returns:
+        Tensor: Waveform of dimension of `(..., time)`
+
+    Reference:
+        - http://sox.sourceforge.net/sox.html
+    """
+
+    if not 0 <= enhancement_amount <= 100:
+        raise ValueError("Allowed range of values for enhancement_amount : 0-100")
+
+    contrast = enhancement_amount / 750.0
+
+    temp1 = waveform * (math.pi / 2)
+    temp2 = contrast * torch.sin(temp1 * 4)
+    output_waveform = torch.sin(temp1 + temp2)
+
+    return output_waveform
+
+
+def dcshift(waveform: Tensor, shift: float, limiter_gain: Optional[float] = None) -> Tensor:
+    r"""Apply a DC shift to the audio. Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: TorchScript
+
+    This can be useful to remove a DC offset
+    (caused perhaps by a hardware problem in the recording chain) from the audio
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`
+        shift (float): indicates the amount to shift the audio
+            Allowed range of values for shift : -2.0 to +2.0
+        limiter_gain (float of None, optional): It is used only on peaks to prevent clipping
+            It should have a value much less than 1 (e.g. 0.05 or 0.02)
+
+    Returns:
+        Tensor: Waveform of dimension of `(..., time)`
+
+    Reference:
+        - http://sox.sourceforge.net/sox.html
+    """
+    output_waveform = waveform
+    limiter_threshold = 0.0
+
+    if limiter_gain is not None:
+        limiter_threshold = 1.0 - (abs(shift) - limiter_gain)
+
+    # Note:
+    # the following index-based update breaks auto-grad support
+    if limiter_gain is not None and shift > 0:
+        mask = waveform > limiter_threshold
+        temp = (waveform[mask] - limiter_threshold) * limiter_gain / (1 - limiter_threshold)
+        output_waveform[mask] = (temp + limiter_threshold + shift).clamp(max=limiter_threshold)
+        output_waveform[~mask] = (waveform[~mask] + shift).clamp(min=-1, max=1)
+    elif limiter_gain is not None and shift < 0:
+        mask = waveform < -limiter_threshold
+        temp = (waveform[mask] + limiter_threshold) * limiter_gain / (1 - limiter_threshold)
+        output_waveform[mask] = (temp - limiter_threshold + shift).clamp(min=-limiter_threshold)
+        output_waveform[~mask] = (waveform[~mask] + shift).clamp(min=-1, max=1)
+    else:
+        output_waveform = (waveform + shift).clamp(min=-1, max=1)
+
+    return output_waveform
+
+
+def deemph_biquad(waveform: Tensor, sample_rate: int) -> Tensor:
+    r"""Apply ISO 908 CD de-emphasis (shelving) IIR filter.  Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`
+        sample_rate (int): sampling rate of the waveform, Allowed sample rate ``44100`` or ``48000``
+
+    Returns:
+        Tensor: Waveform of dimension of `(..., time)`
+
+    Reference:
+        - http://sox.sourceforge.net/sox.html
+        - https://www.w3.org/2011/audio/audio-eq-cookbook.html#APF
+    """
+
+    if sample_rate == 44100:
+        central_freq = 5283
+        width_slope = 0.4845
+        gain = -9.477
+    elif sample_rate == 48000:
+        central_freq = 5356
+        width_slope = 0.479
+        gain = -9.62
+    else:
+        raise ValueError("Sample rate must be 44100 (audio-CD) or 48000 (DAT)")
+
+    w0 = 2 * math.pi * central_freq / sample_rate
+    A = math.exp(gain / 40.0 * math.log(10))
+    alpha = math.sin(w0) / 2 * math.sqrt((A + 1 / A) * (1 / width_slope - 1) + 2)
+
+    temp1 = 2 * math.sqrt(A) * alpha
+    temp2 = (A - 1) * math.cos(w0)
+    temp3 = (A + 1) * math.cos(w0)
+
+    b0 = A * ((A + 1) + temp2 + temp1)
+    b1 = -2 * A * ((A - 1) + temp3)
+    b2 = A * ((A + 1) + temp2 - temp1)
+    a0 = (A + 1) - temp2 + temp1
+    a1 = 2 * ((A - 1) - temp3)
+    a2 = (A + 1) - temp2 - temp1
+
+    return biquad(waveform, b0, b1, b2, a0, a1, a2)
+
+
+def _add_noise_shaping(dithered_waveform: Tensor, waveform: Tensor) -> Tensor:
+    r"""Noise shaping is calculated by error:
+    error[n] = dithered[n] - original[n]
+    noise_shaped_waveform[n] = dithered[n] + error[n-1]
+    """
+    wf_shape = waveform.size()
+    waveform = waveform.reshape(-1, wf_shape[-1])
+
+    dithered_shape = dithered_waveform.size()
+    dithered_waveform = dithered_waveform.reshape(-1, dithered_shape[-1])
+
+    error = dithered_waveform - waveform
+
+    # add error[n-1] to dithered_waveform[n], so offset the error by 1 index
+    zeros = torch.zeros(1, dtype=error.dtype, device=error.device)
+    for index in range(error.size()[0]):
+        err = error[index]
+        error_offset = torch.cat((zeros, err))
+        error[index] = error_offset[: waveform.size()[1]]
+
+    noise_shaped = dithered_waveform + error
+    return noise_shaped.reshape(dithered_shape[:-1] + noise_shaped.shape[-1:])
+
+
+def _apply_probability_distribution(waveform: Tensor, density_function: str = "TPDF") -> Tensor:
+    r"""Apply a probability distribution function on a waveform.
+
+    Triangular probability density function (TPDF) dither noise has a
+    triangular distribution; values in the center of the range have a higher
+    probability of occurring.
+
+    Rectangular probability density function (RPDF) dither noise has a
+    uniform distribution; any value in the specified range has the same
+    probability of occurring.
+
+    Gaussian probability density function (GPDF) has a normal distribution.
+    The relationship of probabilities of results follows a bell-shaped,
+    or Gaussian curve, typical of dither generated by analog sources.
+    Args:
+        waveform (Tensor): Tensor of audio of dimension (..., time)
+        density_function (str, optional): The density function of a
+           continuous random variable (Default: ``"TPDF"``)
+           Options: Triangular Probability Density Function - `TPDF`
+                    Rectangular Probability Density Function - `RPDF`
+                    Gaussian Probability Density Function - `GPDF`
+    Returns:
+        Tensor: waveform dithered with TPDF
+    """
+
+    # pack batch
+    shape = waveform.size()
+    waveform = waveform.reshape(-1, shape[-1])
+
+    channel_size = waveform.size()[0] - 1
+    time_size = waveform.size()[-1] - 1
+
+    random_channel = (
+        int(
+            torch.randint(
+                channel_size,
+                [
+                    1,
+                ],
+            ).item()
+        )
+        if channel_size > 0
+        else 0
+    )
+    random_time = (
+        int(
+            torch.randint(
+                time_size,
+                [
+                    1,
+                ],
+            ).item()
+        )
+        if time_size > 0
+        else 0
+    )
+
+    number_of_bits = 16
+    up_scaling = 2 ** (number_of_bits - 1) - 2
+    signal_scaled = waveform * up_scaling
+    down_scaling = 2 ** (number_of_bits - 1)
+
+    signal_scaled_dis = waveform
+    if density_function == "RPDF":
+        RPDF = waveform[random_channel][random_time] - 0.5
+
+        signal_scaled_dis = signal_scaled + RPDF
+    elif density_function == "GPDF":
+        # TODO Replace by distribution code once
+        # https://github.com/pytorch/pytorch/issues/29843 is resolved
+        # gaussian = torch.distributions.normal.Normal(torch.mean(waveform, -1), 1).sample()
+
+        num_rand_variables = 6
+
+        gaussian = waveform[random_channel][random_time]
+        for ws in num_rand_variables * [time_size]:
+            rand_chan = int(
+                torch.randint(
+                    channel_size,
+                    [
+                        1,
+                    ],
+                ).item()
+            )
+            gaussian += waveform[rand_chan][
+                int(
+                    torch.randint(
+                        ws,
+                        [
+                            1,
+                        ],
+                    ).item()
+                )
+            ]
+
+        signal_scaled_dis = signal_scaled + gaussian
+    else:
+        # dtype needed for https://github.com/pytorch/pytorch/issues/32358
+        TPDF = torch.bartlett_window(time_size + 1, dtype=signal_scaled.dtype, device=signal_scaled.device)
+        TPDF = TPDF.repeat((channel_size + 1), 1)
+        signal_scaled_dis = signal_scaled + TPDF
+
+    quantised_signal_scaled = torch.round(signal_scaled_dis)
+    quantised_signal = quantised_signal_scaled / down_scaling
+
+    # unpack batch
+    return quantised_signal.reshape(shape[:-1] + quantised_signal.shape[-1:])
+
+
+def dither(waveform: Tensor, density_function: str = "TPDF", noise_shaping: bool = False) -> Tensor:
+    r"""Apply dither
+
+    .. devices:: CPU CUDA
+
+    .. properties:: TorchScript
+
+    Dither increases the perceived dynamic range of audio stored at a
+    particular bit-depth by eliminating nonlinear truncation distortion
+    (i.e. adding minimally perceived noise to mask distortion caused by quantization).
+
+    Args:
+        waveform (Tensor): Tensor of audio of dimension (..., time)
+        density_function (str, optional):
+            The density function of a continuous random variable. One of
+            ``"TPDF"`` (Triangular Probability Density Function),
+            ``"RPDF"`` (Rectangular Probability Density Function) or
+            ``"GPDF"`` (Gaussian Probability Density Function) (Default: ``"TPDF"``).
+        noise_shaping (bool, optional): a filtering process that shapes the spectral
+            energy of quantisation error (Default: ``False``)
+
+    Returns:
+       Tensor: waveform dithered
+    """
+    dithered = _apply_probability_distribution(waveform, density_function=density_function)
+
+    if noise_shaping:
+        return _add_noise_shaping(dithered, waveform)
+    else:
+        return dithered
+
+
+def equalizer_biquad(
+    waveform: Tensor,
+    sample_rate: int,
+    center_freq: float,
+    gain: float,
+    Q: float = 0.707,
+) -> Tensor:
+    r"""Design biquad peaking equalizer filter and perform filtering.  Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`
+        sample_rate (int): sampling rate of the waveform, e.g. 44100 (Hz)
+        center_freq (float): filter's central frequency
+        gain (float or torch.Tensor): desired gain at the boost (or attenuation) in dB
+        Q (float or torch.Tensor, optional): https://en.wikipedia.org/wiki/Q_factor (Default: ``0.707``)
+
+    Returns:
+        Tensor: Waveform of dimension of `(..., time)`
+    """
+    dtype = waveform.dtype
+    device = waveform.device
+    center_freq = torch.as_tensor(center_freq, dtype=dtype, device=device)
+    Q = torch.as_tensor(Q, dtype=dtype, device=device)
+    gain = torch.as_tensor(gain, dtype=dtype, device=device)
+
+    w0 = 2 * math.pi * center_freq / sample_rate
+    A = torch.exp(gain / 40.0 * math.log(10))
+    alpha = torch.sin(w0) / 2 / Q
+
+    b0 = 1 + alpha * A
+    b1 = -2 * torch.cos(w0)
+    b2 = 1 - alpha * A
+    a0 = 1 + alpha / A
+    a1 = -2 * torch.cos(w0)
+    a2 = 1 - alpha / A
+    return biquad(waveform, b0, b1, b2, a0, a1, a2)
+
+
+def filtfilt(
+    waveform: Tensor,
+    a_coeffs: Tensor,
+    b_coeffs: Tensor,
+    clamp: bool = True,
+) -> Tensor:
+    r"""Apply an IIR filter forward and backward to a waveform.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Inspired by https://docs.scipy.org/doc/scipy/reference/generated/scipy.signal.filtfilt.html
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`.  Must be normalized to -1 to 1.
+        a_coeffs (Tensor): denominator coefficients of difference equation of dimension of either
+                                1D with shape `(num_order + 1)` or 2D with shape `(num_filters, num_order + 1)`.
+                                Lower delay coefficients are first, e.g. ``[a0, a1, a2, ...]``.
+                                Must be same size as b_coeffs (pad with 0's as necessary).
+        b_coeffs (Tensor): numerator coefficients of difference equation of dimension of either
+                                1D with shape `(num_order + 1)` or 2D with shape `(num_filters, num_order + 1)`.
+                                Lower delay coefficients are first, e.g. ``[b0, b1, b2, ...]``.
+                                Must be same size as a_coeffs (pad with 0's as necessary).
+        clamp (bool, optional): If ``True``, clamp the output signal to be in the range [-1, 1] (Default: ``True``)
+
+    Returns:
+        Tensor: Waveform with dimension of either `(..., num_filters, time)` if ``a_coeffs`` and ``b_coeffs``
+        are 2D Tensors, or `(..., time)` otherwise.
+    """
+    forward_filtered = lfilter(waveform, a_coeffs, b_coeffs, clamp=False, batching=True)
+    backward_filtered = lfilter(
+        forward_filtered.flip(-1),
+        a_coeffs,
+        b_coeffs,
+        clamp=clamp,
+        batching=True,
+    ).flip(-1)
+    return backward_filtered
+
+
+def flanger(
+    waveform: Tensor,
+    sample_rate: int,
+    delay: float = 0.0,
+    depth: float = 2.0,
+    regen: float = 0.0,
+    width: float = 71.0,
+    speed: float = 0.5,
+    phase: float = 25.0,
+    modulation: str = "sinusoidal",
+    interpolation: str = "linear",
+) -> Tensor:
+    r"""Apply a flanger effect to the audio. Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., channel, time)` .
+            Max 4 channels allowed
+        sample_rate (int): sampling rate of the waveform, e.g. 44100 (Hz)
+        delay (float, optional): desired delay in milliseconds(ms)
+            Allowed range of values are 0 to 30
+        depth (float, optional): desired delay depth in milliseconds(ms)
+            Allowed range of values are 0 to 10
+        regen (float, optional): desired regen(feedback gain) in dB
+            Allowed range of values are -95 to 95
+        width (float, optional):  desired width(delay gain) in dB
+            Allowed range of values are 0 to 100
+        speed (float, optional):  modulation speed in Hz
+            Allowed range of values are 0.1 to 10
+        phase (float, optional):  percentage phase-shift for multi-channel
+            Allowed range of values are 0 to 100
+        modulation (str, optional):  Use either "sinusoidal" or "triangular" modulation. (Default: ``sinusoidal``)
+        interpolation (str, optional): Use either "linear" or "quadratic" for delay-line interpolation.
+            (Default: ``linear``)
+
+    Returns:
+        Tensor: Waveform of dimension of `(..., channel, time)`
+
+    Reference:
+        - http://sox.sourceforge.net/sox.html
+
+        - Scott Lehman, `Effects Explained`_,
+
+    .. _Effects Explained:
+        https://web.archive.org/web/20051125072557/http://www.harmony-central.com/Effects/effects-explained.html
+    """
+
+    if modulation not in ("sinusoidal", "triangular"):
+        raise ValueError('Only "sinusoidal" or "triangular" modulation allowed')
+
+    if interpolation not in ("linear", "quadratic"):
+        raise ValueError('Only "linear" or "quadratic" interpolation allowed')
+
+    actual_shape = waveform.shape
+    device, dtype = waveform.device, waveform.dtype
+
+    if actual_shape[-2] > 4:
+        raise ValueError("Max 4 channels allowed")
+
+    # convert to 3D (batch, channels, time)
+    waveform = waveform.view(-1, actual_shape[-2], actual_shape[-1])
+
+    # Scaling
+    feedback_gain = regen / 100
+    delay_gain = width / 100
+    channel_phase = phase / 100
+    delay_min = delay / 1000
+    delay_depth = depth / 1000
+
+    n_channels = waveform.shape[-2]
+
+    if modulation == "sinusoidal":
+        wave_type = "SINE"
+    else:
+        wave_type = "TRIANGLE"
+
+    # Balance output:
+    in_gain = 1.0 / (1 + delay_gain)
+    delay_gain = delay_gain / (1 + delay_gain)
+
+    # Balance feedback loop:
+    delay_gain = delay_gain * (1 - abs(feedback_gain))
+
+    delay_buf_length = int((delay_min + delay_depth) * sample_rate + 0.5)
+    delay_buf_length = delay_buf_length + 2
+
+    delay_bufs = torch.zeros(waveform.shape[0], n_channels, delay_buf_length, dtype=dtype, device=device)
+    delay_last = torch.zeros(waveform.shape[0], n_channels, dtype=dtype, device=device)
+
+    lfo_length = int(sample_rate / speed)
+
+    table_min = math.floor(delay_min * sample_rate + 0.5)
+    table_max = delay_buf_length - 2.0
+
+    lfo = _generate_wave_table(
+        wave_type=wave_type,
+        data_type="FLOAT",
+        table_size=lfo_length,
+        min=float(table_min),
+        max=float(table_max),
+        phase=3 * math.pi / 2,
+        device=device,
+    )
+
+    output_waveform = torch.zeros_like(waveform, dtype=dtype, device=device)
+
+    delay_buf_pos = 0
+    lfo_pos = 0
+    channel_idxs = torch.arange(0, n_channels, device=device)
+
+    for i in range(waveform.shape[-1]):
+
+        delay_buf_pos = (delay_buf_pos + delay_buf_length - 1) % delay_buf_length
+
+        cur_channel_phase = (channel_idxs * lfo_length * channel_phase + 0.5).to(torch.int64)
+        delay_tensor = lfo[(lfo_pos + cur_channel_phase) % lfo_length]
+        frac_delay = torch.frac(delay_tensor)
+        delay_tensor = torch.floor(delay_tensor)
+
+        int_delay = delay_tensor.to(torch.int64)
+
+        temp = waveform[:, :, i]
+
+        delay_bufs[:, :, delay_buf_pos] = temp + delay_last * feedback_gain
+
+        delayed_0 = delay_bufs[:, channel_idxs, (delay_buf_pos + int_delay) % delay_buf_length]
+
+        int_delay = int_delay + 1
+
+        delayed_1 = delay_bufs[:, channel_idxs, (delay_buf_pos + int_delay) % delay_buf_length]
+
+        int_delay = int_delay + 1
+
+        if interpolation == "linear":
+            delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay
+        else:
+            delayed_2 = delay_bufs[:, channel_idxs, (delay_buf_pos + int_delay) % delay_buf_length]
+
+            int_delay = int_delay + 1
+
+            delayed_2 = delayed_2 - delayed_0
+            delayed_1 = delayed_1 - delayed_0
+            a = delayed_2 * 0.5 - delayed_1
+            b = delayed_1 * 2 - delayed_2 * 0.5
+
+            delayed = delayed_0 + (a * frac_delay + b) * frac_delay
+
+        delay_last = delayed
+        output_waveform[:, :, i] = waveform[:, :, i] * in_gain + delayed * delay_gain
+
+        lfo_pos = (lfo_pos + 1) % lfo_length
+
+    return output_waveform.clamp(min=-1, max=1).view(actual_shape)
+
+
+def gain(waveform: Tensor, gain_db: float = 1.0) -> Tensor:
+    r"""Apply amplification or attenuation to the whole waveform.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Args:
+       waveform (Tensor): Tensor of audio of dimension (..., time).
+       gain_db (float, optional) Gain adjustment in decibels (dB) (Default: ``1.0``).
+
+    Returns:
+       Tensor: the whole waveform amplified by gain_db.
+    """
+    if gain_db == 0:
+        return waveform
+
+    ratio = 10 ** (gain_db / 20)
+
+    return waveform * ratio
+
+
+def highpass_biquad(waveform: Tensor, sample_rate: int, cutoff_freq: float, Q: float = 0.707) -> Tensor:
+    r"""Design biquad highpass filter and perform filtering.  Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`
+        sample_rate (int): sampling rate of the waveform, e.g. 44100 (Hz)
+        cutoff_freq (float or torch.Tensor): filter cutoff frequency
+        Q (float or torch.Tensor, optional): https://en.wikipedia.org/wiki/Q_factor (Default: ``0.707``)
+
+    Returns:
+        Tensor: Waveform dimension of `(..., time)`
+    """
+    dtype = waveform.dtype
+    device = waveform.device
+    cutoff_freq = torch.as_tensor(cutoff_freq, dtype=dtype, device=device)
+    Q = torch.as_tensor(Q, dtype=dtype, device=device)
+
+    w0 = 2 * math.pi * cutoff_freq / sample_rate
+    alpha = torch.sin(w0) / 2.0 / Q
+
+    b0 = (1 + torch.cos(w0)) / 2
+    b1 = -1 - torch.cos(w0)
+    b2 = b0
+    a0 = 1 + alpha
+    a1 = -2 * torch.cos(w0)
+    a2 = 1 - alpha
+    return biquad(waveform, b0, b1, b2, a0, a1, a2)
+
+
+def _lfilter_core_generic_loop(input_signal_windows: Tensor, a_coeffs_flipped: Tensor, padded_output_waveform: Tensor):
+    n_order = a_coeffs_flipped.size(1)
+    a_coeffs_flipped = a_coeffs_flipped.unsqueeze(2)
+    for i_sample, o0 in enumerate(input_signal_windows.permute(2, 0, 1)):
+        windowed_output_signal = padded_output_waveform[:, :, i_sample : i_sample + n_order]
+        o0 -= (windowed_output_signal.transpose(0, 1) @ a_coeffs_flipped)[..., 0].t()
+        padded_output_waveform[:, :, i_sample + n_order - 1] = o0
+
+
+if _IS_TORCHAUDIO_EXT_AVAILABLE:
+    _lfilter_core_loop = torch.ops.torchaudio._lfilter_core_loop
+else:
+    _lfilter_core_loop = _lfilter_core_generic_loop
+
+
+class DifferentiableFIR(torch.autograd.Function):
+    @staticmethod
+    def forward(ctx, waveform, b_coeffs):
+        n_order = b_coeffs.size(1)
+        n_channel = b_coeffs.size(0)
+        b_coeff_flipped = b_coeffs.flip(1).contiguous()
+        padded_waveform = F.pad(waveform, (n_order - 1, 0))
+        output = F.conv1d(padded_waveform, b_coeff_flipped.unsqueeze(1), groups=n_channel)
+        ctx.save_for_backward(waveform, b_coeffs, output)
+        return output
+
+    @staticmethod
+    def backward(ctx, dy):
+        x, b_coeffs, y = ctx.saved_tensors
+        n_batch = x.size(0)
+        n_channel = x.size(1)
+        n_order = b_coeffs.size(1)
+        db = (
+            F.conv1d(
+                F.pad(x, (n_order - 1, 0)).view(1, n_batch * n_channel, -1),
+                dy.view(n_batch * n_channel, 1, -1),
+                groups=n_batch * n_channel,
+            )
+            .view(n_batch, n_channel, -1)
+            .sum(0)
+            .flip(1)
+            if b_coeffs.requires_grad
+            else None
+        )
+        dx = F.conv1d(F.pad(dy, (0, n_order - 1)), b_coeffs.unsqueeze(1), groups=n_channel) if x.requires_grad else None
+        return (dx, db)
+
+
+class DifferentiableIIR(torch.autograd.Function):
+    @staticmethod
+    def forward(ctx, waveform, a_coeffs_normalized):
+        n_batch, n_channel, n_sample = waveform.shape
+        n_order = a_coeffs_normalized.size(1)
+        n_sample_padded = n_sample + n_order - 1
+
+        a_coeff_flipped = a_coeffs_normalized.flip(1).contiguous()
+        padded_output_waveform = torch.zeros(
+            n_batch, n_channel, n_sample_padded, device=waveform.device, dtype=waveform.dtype
+        )
+        _lfilter_core_loop(waveform, a_coeff_flipped, padded_output_waveform)
+        output = padded_output_waveform[:, :, n_order - 1 :]
+        ctx.save_for_backward(waveform, a_coeffs_normalized, output)
+        return output
+
+    @staticmethod
+    def backward(ctx, dy):
+        x, a_coeffs_normalized, y = ctx.saved_tensors
+        n_channel = x.size(1)
+        n_order = a_coeffs_normalized.size(1)
+        tmp = DifferentiableIIR.apply(dy.flip(2).contiguous(), a_coeffs_normalized).flip(2)
+        dx = tmp if x.requires_grad else None
+        da = (
+            -(
+                tmp.transpose(0, 1).reshape(n_channel, 1, -1)
+                @ F.pad(y, (n_order - 1, 0)).unfold(2, n_order, 1).transpose(0, 1).reshape(n_channel, -1, n_order)
+            )
+            .squeeze(1)
+            .flip(1)
+            if a_coeffs_normalized.requires_grad
+            else None
+        )
+        return (dx, da)
+
+
+def _lfilter(waveform, a_coeffs, b_coeffs):
+    filtered_waveform = DifferentiableFIR.apply(waveform, b_coeffs / a_coeffs[:, 0:1])
+    return DifferentiableIIR.apply(filtered_waveform, a_coeffs / a_coeffs[:, 0:1])
+
+
+def lfilter(waveform: Tensor, a_coeffs: Tensor, b_coeffs: Tensor, clamp: bool = True, batching: bool = True) -> Tensor:
+    r"""Perform an IIR filter by evaluating difference equation, using differentiable implementation
+    developed separately by *Yu et al.* :cite:`ismir_YuF23` and *Forgione et al.* :cite:`forgione2021dynonet`.
+    The gradients of ``a_coeffs`` are computed based on a faster algorithm from :cite:`ycy2024diffapf`.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Note:
+        To avoid numerical problems, small filter order is preferred.
+        Using double precision could also minimize numerical precision errors.
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`.  Must be normalized to -1 to 1.
+        a_coeffs (Tensor): denominator coefficients of difference equation of dimension of either
+                                1D with shape `(num_order + 1)` or 2D with shape `(num_filters, num_order + 1)`.
+                                Lower delays coefficients are first, e.g. ``[a0, a1, a2, ...]``.
+                                Must be same size as b_coeffs (pad with 0's as necessary).
+        b_coeffs (Tensor): numerator coefficients of difference equation of dimension of either
+                                1D with shape `(num_order + 1)` or 2D with shape `(num_filters, num_order + 1)`.
+                                Lower delays coefficients are first, e.g. ``[b0, b1, b2, ...]``.
+                                Must be same size as a_coeffs (pad with 0's as necessary).
+        clamp (bool, optional): If ``True``, clamp the output signal to be in the range [-1, 1] (Default: ``True``)
+        batching (bool, optional): Effective only when coefficients are 2D. If ``True``, then waveform should be at
+                                    least 2D, and the size of second axis from last should equals to ``num_filters``.
+                                    The output can be expressed as ``output[..., i, :] = lfilter(waveform[..., i, :],
+                                    a_coeffs[i], b_coeffs[i], clamp=clamp, batching=False)``. (Default: ``True``)
+
+    Returns:
+        Tensor: Waveform with dimension of either `(..., num_filters, time)` if ``a_coeffs`` and ``b_coeffs``
+        are 2D Tensors, or `(..., time)` otherwise.
+    """
+    if a_coeffs.size() != b_coeffs.size():
+        raise ValueError(
+            "Expected coeffs to be the same size."
+            f"Found: a_coeffs size: {a_coeffs.size()}, b_coeffs size: {b_coeffs.size()}"
+        )
+    if a_coeffs.ndim > 2:
+        raise ValueError(f"Expected coeffs to have greater than 1 dimension. Found: {a_coeffs.ndim}")
+
+    if a_coeffs.ndim > 1:
+        if batching:
+            if waveform.ndim <= 0:
+                raise ValueError("Expected waveform to have a positive number of dimensions." f"Found: {waveform.ndim}")
+            if waveform.shape[-2] != a_coeffs.shape[0]:
+                raise ValueError(
+                    "Expected number of batches in waveform and coeffs to be the same."
+                    f"Found: coeffs batches: {a_coeffs.shape[0]}, waveform batches: {waveform.shape[-2]}"
+                )
+        else:
+            waveform = torch.stack([waveform] * a_coeffs.shape[0], -2)
+    else:
+        a_coeffs = a_coeffs.unsqueeze(0)
+        b_coeffs = b_coeffs.unsqueeze(0)
+
+    # pack batch
+    shape = waveform.size()
+    waveform = waveform.reshape(-1, a_coeffs.shape[0], shape[-1])
+    output = _lfilter(waveform, a_coeffs, b_coeffs)
+
+    if clamp:
+        output = torch.clamp(output, min=-1.0, max=1.0)
+
+    # unpack batch
+    output = output.reshape(shape[:-1] + output.shape[-1:])
+
+    return output
+
+
+def lowpass_biquad(waveform: Tensor, sample_rate: int, cutoff_freq: float, Q: float = 0.707) -> Tensor:
+    r"""Design biquad lowpass filter and perform filtering.  Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Args:
+        waveform (torch.Tensor): audio waveform of dimension of `(..., time)`
+        sample_rate (int): sampling rate of the waveform, e.g. 44100 (Hz)
+        cutoff_freq (float or torch.Tensor): filter cutoff frequency
+        Q (float or torch.Tensor, optional): https://en.wikipedia.org/wiki/Q_factor (Default: ``0.707``)
+
+    Returns:
+        Tensor: Waveform of dimension of `(..., time)`
+    """
+    dtype = waveform.dtype
+    device = waveform.device
+    cutoff_freq = torch.as_tensor(cutoff_freq, dtype=dtype, device=device)
+    Q = torch.as_tensor(Q, dtype=dtype, device=device)
+
+    w0 = 2 * math.pi * cutoff_freq / sample_rate
+    alpha = torch.sin(w0) / 2 / Q
+
+    b0 = (1 - torch.cos(w0)) / 2
+    b1 = 1 - torch.cos(w0)
+    b2 = b0
+    a0 = 1 + alpha
+    a1 = -2 * torch.cos(w0)
+    a2 = 1 - alpha
+    return biquad(waveform, b0, b1, b2, a0, a1, a2)
+
+
+def _overdrive_core_loop_generic(
+    waveform: Tensor, temp: Tensor, last_in: Tensor, last_out: Tensor, output_waveform: Tensor
+):
+    for i in range(waveform.shape[-1]):
+        last_out = temp[:, i] - last_in + 0.995 * last_out
+        last_in = temp[:, i]
+        output_waveform[:, i] = waveform[:, i] * 0.5 + last_out * 0.75
+
+
+if _IS_TORCHAUDIO_EXT_AVAILABLE:
+    _overdrive_core_loop_cpu = torch.ops.torchaudio._overdrive_core_loop
+else:
+    _overdrive_core_loop_cpu = _overdrive_core_loop_generic
+
+
+def overdrive(waveform: Tensor, gain: float = 20, colour: float = 20) -> Tensor:
+    r"""Apply a overdrive effect to the audio. Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    This effect applies a non linear distortion to the audio signal.
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`
+        gain (float, optional): desired gain at the boost (or attenuation) in dB
+            Allowed range of values are 0 to 100
+        colour (float, optional):  controls the amount of even harmonic content in the over-driven output
+            Allowed range of values are 0 to 100
+
+    Returns:
+        Tensor: Waveform of dimension of `(..., time)`
+
+    Reference:
+        - http://sox.sourceforge.net/sox.html
+    """
+    actual_shape = waveform.shape
+    device, dtype = waveform.device, waveform.dtype
+
+    # convert to 2D (..,time)
+    waveform = waveform.view(-1, actual_shape[-1])
+
+    gain = _dB2Linear(gain)
+    colour = colour / 200
+    last_in = torch.zeros(waveform.shape[:-1], dtype=dtype, device=device)
+    last_out = torch.zeros(waveform.shape[:-1], dtype=dtype, device=device)
+
+    temp = waveform * gain + colour
+
+    mask1 = temp < -1
+    temp[mask1] = torch.tensor(-2.0 / 3.0, dtype=dtype, device=device)
+    # Wrapping the constant with Tensor is required for Torchscript
+
+    mask2 = temp > 1
+    temp[mask2] = torch.tensor(2.0 / 3.0, dtype=dtype, device=device)
+
+    mask3 = ~mask1 & ~mask2
+    temp[mask3] = temp[mask3] - (temp[mask3] ** 3) * (1.0 / 3)
+
+    output_waveform = torch.zeros_like(waveform, dtype=dtype, device=device)
+
+    # Uses CPU optimized loop function if available for CPU device
+    if device == torch.device("cpu"):
+        _overdrive_core_loop_cpu(waveform, temp, last_in, last_out, output_waveform)
+    else:
+        _overdrive_core_loop_generic(waveform, temp, last_in, last_out, output_waveform)
+
+    return output_waveform.clamp(min=-1, max=1).view(actual_shape)
+
+
+def phaser(
+    waveform: Tensor,
+    sample_rate: int,
+    gain_in: float = 0.4,
+    gain_out: float = 0.74,
+    delay_ms: float = 3.0,
+    decay: float = 0.4,
+    mod_speed: float = 0.5,
+    sinusoidal: bool = True,
+) -> Tensor:
+    r"""Apply a phasing effect to the audio. Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`
+        sample_rate (int): sampling rate of the waveform, e.g. 44100 (Hz)
+        gain_in (float, optional): desired input gain at the boost (or attenuation) in dB
+            Allowed range of values are 0 to 1
+        gain_out (float, optional): desired output gain at the boost (or attenuation) in dB
+            Allowed range of values are 0 to 1e9
+        delay_ms (float, optional): desired delay in milliseconds
+            Allowed range of values are 0 to 5.0
+        decay (float, optional):  desired decay relative to gain-in
+            Allowed range of values are 0 to 0.99
+        mod_speed (float, optional):  modulation speed in Hz
+            Allowed range of values are 0.1 to 2
+        sinusoidal (bool, optional):  If ``True``, uses sinusoidal modulation (preferable for multiple instruments)
+            If ``False``, uses triangular modulation (gives single instruments a sharper phasing effect)
+            (Default: ``True``)
+
+    Returns:
+        Tensor: Waveform of dimension of `(..., time)`
+
+    Reference:
+        - http://sox.sourceforge.net/sox.html
+        - Scott Lehman, `Effects Explained`_.
+
+    .. _Effects Explained:
+        https://web.archive.org/web/20051125072557/http://www.harmony-central.com/Effects/effects-explained.html
+    """
+    actual_shape = waveform.shape
+    device, dtype = waveform.device, waveform.dtype
+
+    # convert to 2D (channels,time)
+    waveform = waveform.view(-1, actual_shape[-1])
+
+    delay_buf_len = int((delay_ms * 0.001 * sample_rate) + 0.5)
+    delay_buf = torch.zeros(waveform.shape[0], delay_buf_len, dtype=dtype, device=device)
+
+    mod_buf_len = int(sample_rate / mod_speed + 0.5)
+
+    if sinusoidal:
+        wave_type = "SINE"
+    else:
+        wave_type = "TRIANGLE"
+
+    mod_buf = _generate_wave_table(
+        wave_type=wave_type,
+        data_type="INT",
+        table_size=mod_buf_len,
+        min=1.0,
+        max=float(delay_buf_len),
+        phase=math.pi / 2,
+        device=device,
+    )
+
+    delay_pos = 0
+    mod_pos = 0
+
+    output_waveform_pre_gain_list = []
+    waveform = waveform * gain_in
+    delay_buf = delay_buf * decay
+    waveform_list = [waveform[:, i] for i in range(waveform.size(1))]
+    delay_buf_list = [delay_buf[:, i] for i in range(delay_buf.size(1))]
+    mod_buf_list = [mod_buf[i] for i in range(mod_buf.size(0))]
+
+    for i in range(waveform.shape[-1]):
+        idx = int((delay_pos + mod_buf_list[mod_pos]) % delay_buf_len)
+        mod_pos = (mod_pos + 1) % mod_buf_len
+        delay_pos = (delay_pos + 1) % delay_buf_len
+        temp = (waveform_list[i]) + (delay_buf_list[idx])
+        delay_buf_list[delay_pos] = temp * decay
+        output_waveform_pre_gain_list.append(temp)
+
+    output_waveform = torch.stack(output_waveform_pre_gain_list, dim=1).to(dtype=dtype, device=device)
+    output_waveform.mul_(gain_out)
+
+    return output_waveform.clamp(min=-1, max=1).view(actual_shape)
+
+
+def riaa_biquad(waveform: Tensor, sample_rate: int) -> Tensor:
+    r"""Apply RIAA vinyl playback equalization.  Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`
+        sample_rate (int): sampling rate of the waveform, e.g. 44100 (Hz).
+            Allowed sample rates in Hz : ``44100``,``48000``,``88200``,``96000``
+
+    Returns:
+        Tensor: Waveform of dimension of `(..., time)`
+
+    Reference:
+        - http://sox.sourceforge.net/sox.html
+        - https://www.w3.org/2011/audio/audio-eq-cookbook.html#APF
+    """
+
+    if sample_rate == 44100:
+        zeros = [-0.2014898, 0.9233820]
+        poles = [0.7083149, 0.9924091]
+
+    elif sample_rate == 48000:
+        zeros = [-0.1766069, 0.9321590]
+        poles = [0.7396325, 0.9931330]
+
+    elif sample_rate == 88200:
+        zeros = [-0.1168735, 0.9648312]
+        poles = [0.8590646, 0.9964002]
+
+    elif sample_rate == 96000:
+        zeros = [-0.1141486, 0.9676817]
+        poles = [0.8699137, 0.9966946]
+
+    else:
+        raise ValueError("Sample rate must be 44.1k, 48k, 88.2k, or 96k")
+
+    # polynomial coefficients with roots zeros[0] and zeros[1]
+    b0 = 1.0
+    b1 = -(zeros[0] + zeros[1])
+    b2 = zeros[0] * zeros[1]
+
+    # polynomial coefficients with roots poles[0] and poles[1]
+    a0 = 1.0
+    a1 = -(poles[0] + poles[1])
+    a2 = poles[0] * poles[1]
+
+    # Normalize to 0dB at 1kHz
+    y = 2 * math.pi * 1000 / sample_rate
+    b_re = b0 + b1 * math.cos(-y) + b2 * math.cos(-2 * y)
+    a_re = a0 + a1 * math.cos(-y) + a2 * math.cos(-2 * y)
+    b_im = b1 * math.sin(-y) + b2 * math.sin(-2 * y)
+    a_im = a1 * math.sin(-y) + a2 * math.sin(-2 * y)
+    g = 1 / math.sqrt((b_re**2 + b_im**2) / (a_re**2 + a_im**2))
+
+    b0 *= g
+    b1 *= g
+    b2 *= g
+
+    return biquad(waveform, b0, b1, b2, a0, a1, a2)
+
+
+def treble_biquad(
+    waveform: Tensor,
+    sample_rate: int,
+    gain: float,
+    central_freq: float = 3000,
+    Q: float = 0.707,
+) -> Tensor:
+    r"""Design a treble tone-control effect.  Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: Autograd TorchScript
+
+    Args:
+        waveform (Tensor): audio waveform of dimension of `(..., time)`
+        sample_rate (int): sampling rate of the waveform, e.g. 44100 (Hz)
+        gain (float or torch.Tensor): desired gain at the boost (or attenuation) in dB.
+        central_freq (float or torch.Tensor, optional): central frequency (in Hz). (Default: ``3000``)
+        Q (float or torch.Tensor, optional): https://en.wikipedia.org/wiki/Q_factor (Default: ``0.707``).
+
+    Returns:
+        Tensor: Waveform of dimension of `(..., time)`
+
+    Reference:
+        - http://sox.sourceforge.net/sox.html
+        - https://www.w3.org/2011/audio/audio-eq-cookbook.html#APF
+    """
+    dtype = waveform.dtype
+    device = waveform.device
+    central_freq = torch.as_tensor(central_freq, dtype=dtype, device=device)
+    Q = torch.as_tensor(Q, dtype=dtype, device=device)
+    gain = torch.as_tensor(gain, dtype=dtype, device=device)
+
+    w0 = 2 * math.pi * central_freq / sample_rate
+    alpha = torch.sin(w0) / 2 / Q
+    A = torch.exp(gain / 40 * math.log(10))
+
+    temp1 = 2 * torch.sqrt(A) * alpha
+    temp2 = (A - 1) * torch.cos(w0)
+    temp3 = (A + 1) * torch.cos(w0)
+
+    b0 = A * ((A + 1) + temp2 + temp1)
+    b1 = -2 * A * ((A - 1) + temp3)
+    b2 = A * ((A + 1) + temp2 - temp1)
+    a0 = (A + 1) - temp2 + temp1
+    a1 = 2 * ((A - 1) - temp3)
+    a2 = (A + 1) - temp2 - temp1
+
+    return biquad(waveform, b0, b1, b2, a0, a1, a2)
+
+
+def _measure(
+    measure_len_ws: int,
+    samples: Tensor,
+    spectrum: Tensor,
+    noise_spectrum: Tensor,
+    spectrum_window: Tensor,
+    spectrum_start: int,
+    spectrum_end: int,
+    cepstrum_window: Tensor,
+    cepstrum_start: int,
+    cepstrum_end: int,
+    noise_reduction_amount: float,
+    measure_smooth_time_mult: float,
+    noise_up_time_mult: Tensor,
+    noise_down_time_mult: Tensor,
+    boot_count: int,
+) -> float:
+    device = samples.device
+
+    if spectrum.size(-1) != noise_spectrum.size(-1):
+        raise ValueError(
+            "Expected spectrum size to match noise spectrum size in final dimension."
+            f"Found: spectrum size: {spectrum.size()}, noise_spectrum size: {noise_spectrum.size()}"
+        )
+
+    dft_len_ws = spectrum.size()[-1]
+
+    dftBuf = torch.zeros(dft_len_ws, device=device)
+
+    dftBuf[:measure_len_ws] = samples * spectrum_window[:measure_len_ws]
+
+    # lsx_safe_rdft((int)p->dft_len_ws, 1, c->dftBuf);
+    _dftBuf = torch.fft.rfft(dftBuf)
+
+    mult: float = boot_count / (1.0 + boot_count) if boot_count >= 0 else measure_smooth_time_mult
+
+    _d = _dftBuf[spectrum_start:spectrum_end].abs()
+    spectrum[spectrum_start:spectrum_end].mul_(mult).add_(_d * (1 - mult))
+    _d = spectrum[spectrum_start:spectrum_end] ** 2
+
+    _zeros = torch.zeros(spectrum_end - spectrum_start, device=device)
+    _mult = (
+        _zeros
+        if boot_count >= 0
+        else torch.where(
+            _d > noise_spectrum[spectrum_start:spectrum_end],
+            noise_up_time_mult,  # if
+            noise_down_time_mult,  # else,
+        )
+    )
+
+    noise_spectrum[spectrum_start:spectrum_end].mul_(_mult).add_(_d * (1 - _mult))
+    _d = torch.sqrt(
+        torch.max(
+            _zeros,
+            _d - noise_reduction_amount * noise_spectrum[spectrum_start:spectrum_end],
+        ),
+    )
+
+    _cepstrum_Buf: Tensor = torch.zeros(dft_len_ws >> 1, device=device)
+    _cepstrum_Buf[spectrum_start:spectrum_end] = _d * cepstrum_window
+    _cepstrum_Buf[spectrum_end : dft_len_ws >> 1].zero_()
+
+    # lsx_safe_rdft((int)p->dft_len_ws >> 1, 1, c->dftBuf);
+    _cepstrum_Buf = torch.fft.rfft(_cepstrum_Buf)
+
+    result: float = float(torch.sum(_cepstrum_Buf[cepstrum_start:cepstrum_end].abs().pow(2)))
+    result = math.log(result / (cepstrum_end - cepstrum_start)) if result > 0 else -math.inf
+    return max(0, 21 + result)
+
+
+def vad(
+    waveform: Tensor,
+    sample_rate: int,
+    trigger_level: float = 7.0,
+    trigger_time: float = 0.25,
+    search_time: float = 1.0,
+    allowed_gap: float = 0.25,
+    pre_trigger_time: float = 0.0,
+    # Fine-tuning parameters
+    boot_time: float = 0.35,
+    noise_up_time: float = 0.1,
+    noise_down_time: float = 0.01,
+    noise_reduction_amount: float = 1.35,
+    measure_freq: float = 20.0,
+    measure_duration: Optional[float] = None,
+    measure_smooth_time: float = 0.4,
+    hp_filter_freq: float = 50.0,
+    lp_filter_freq: float = 6000.0,
+    hp_lifter_freq: float = 150.0,
+    lp_lifter_freq: float = 2000.0,
+) -> Tensor:
+    r"""Voice Activity Detector. Similar to SoX implementation.
+
+    .. devices:: CPU CUDA
+
+    .. properties:: TorchScript
+
+    Attempts to trim silence and quiet background sounds from the ends of recordings of speech.
+    The algorithm currently uses a simple cepstral power measurement to detect voice,
+    so may be fooled by other things, especially music.
+
+    The effect can trim only from the front of the audio,
+    so in order to trim from the back, the reverse effect must also be used.
+
+    Args:
+        waveform (Tensor): Tensor of audio of dimension `(channels, time)` or `(time)`
+            Tensor of shape `(channels, time)` is treated as a multi-channel recording
+            of the same event and the resulting output will be trimmed to the earliest
+            voice activity in any channel.
+        sample_rate (int): Sample rate of audio signal.
+        trigger_level (float, optional): The measurement level used to trigger activity detection.
+            This may need to be cahnged depending on the noise level, signal level,
+            and other characteristics of the input audio. (Default: 7.0)
+        trigger_time (float, optional): The time constant (in seconds)
+            used to help ignore short bursts of sound. (Default: 0.25)
+        search_time (float, optional): The amount of audio (in seconds)
+            to search for quieter/shorter bursts of audio to include prior
+            to the detected trigger point. (Default: 1.0)
+        allowed_gap (float, optional): The allowed gap (in seconds) between
+            quieter/shorter bursts of audio to include prior
+            to the detected trigger point. (Default: 0.25)
+        pre_trigger_time (float, optional): The amount of audio (in seconds) to preserve
+            before the trigger point and any found quieter/shorter bursts. (Default: 0.0)
+        boot_time (float, optional) The algorithm (internally) uses adaptive noise
+            estimation/reduction in order to detect the start of the wanted audio.
+            This option sets the time for the initial noise estimate. (Default: 0.35)
+        noise_up_time (float, optional) Time constant used by the adaptive noise estimator
+            for when the noise level is increasing. (Default: 0.1)
+        noise_down_time (float, optional) Time constant used by the adaptive noise estimator
+            for when the noise level is decreasing. (Default: 0.01)
+        noise_reduction_amount (float, optional) Amount of noise reduction to use in
+            the detection algorithm (e.g. 0, 0.5, ...). (Default: 1.35)
+        measure_freq (float, optional) Frequency of the algorithm's
+            processing/measurements. (Default: 20.0)
+        measure_duration: (float, optional) Measurement duration.
+            (Default: Twice the measurement period; i.e. with overlap.)
+        measure_smooth_time (float, optional) Time constant used to smooth
+            spectral measurements. (Default: 0.4)
+        hp_filter_freq (float, optional) "Brick-wall" frequency of high-pass filter applied
+            at the input to the detector algorithm. (Default: 50.0)
+        lp_filter_freq (float, optional) "Brick-wall" frequency of low-pass filter applied
+            at the input to the detector algorithm. (Default: 6000.0)
+        hp_lifter_freq (float, optional) "Brick-wall" frequency of high-pass lifter used
+            in the detector algorithm. (Default: 150.0)
+        lp_lifter_freq (float, optional) "Brick-wall" frequency of low-pass lifter used
+            in the detector algorithm. (Default: 2000.0)
+
+    Returns:
+        Tensor: Tensor of audio of dimension `(..., time)`.
+
+    Reference:
+        - http://sox.sourceforge.net/sox.html
+    """
+    device = waveform.device
+
+    if waveform.ndim > 2:
+        warnings.warn(
+            "Expected input tensor dimension of 1 for single channel"
+            f" or 2 for multi-channel. Got {waveform.ndim} instead. "
+            "Batch semantics is not supported. "
+            "Please refer to https://github.com/pytorch/audio/issues/1348"
+            " and https://github.com/pytorch/audio/issues/1468."
+        )
+
+    measure_duration: float = 2.0 / measure_freq if measure_duration is None else measure_duration
+
+    measure_len_ws = int(sample_rate * measure_duration + 0.5)
+    measure_len_ns = measure_len_ws
+    # for (dft_len_ws = 16; dft_len_ws < measure_len_ws; dft_len_ws <<= 1);
+    dft_len_ws = 16
+    while dft_len_ws < measure_len_ws:
+        dft_len_ws *= 2
+
+    measure_period_ns = int(sample_rate / measure_freq + 0.5)
+    measures_len = math.ceil(search_time * measure_freq)
+    search_pre_trigger_len_ns = measures_len * measure_period_ns
+    gap_len = int(allowed_gap * measure_freq + 0.5)
+
+    fixed_pre_trigger_len_ns = int(pre_trigger_time * sample_rate + 0.5)
+    samplesLen_ns = fixed_pre_trigger_len_ns + search_pre_trigger_len_ns + measure_len_ns
+
+    spectrum_window = torch.zeros(measure_len_ws, device=device)
+    for i in range(measure_len_ws):
+        # sox.h:741 define SOX_SAMPLE_MIN (sox_sample_t)SOX_INT_MIN(32)
+        spectrum_window[i] = 2.0 / math.sqrt(float(measure_len_ws))
+    # lsx_apply_hann(spectrum_window, (int)measure_len_ws);
+    spectrum_window *= torch.hann_window(measure_len_ws, device=device, dtype=torch.float)
+
+    spectrum_start: int = int(hp_filter_freq / sample_rate * dft_len_ws + 0.5)
+    spectrum_start: int = max(spectrum_start, 1)
+    spectrum_end: int = int(lp_filter_freq / sample_rate * dft_len_ws + 0.5)
+    spectrum_end: int = min(spectrum_end, dft_len_ws // 2)
+
+    cepstrum_window = torch.zeros(spectrum_end - spectrum_start, device=device)
+    for i in range(spectrum_end - spectrum_start):
+        cepstrum_window[i] = 2.0 / math.sqrt(float(spectrum_end) - spectrum_start)
+    # lsx_apply_hann(cepstrum_window,(int)(spectrum_end - spectrum_start));
+    cepstrum_window *= torch.hann_window(spectrum_end - spectrum_start, device=device, dtype=torch.float)
+
+    cepstrum_start = math.ceil(sample_rate * 0.5 / lp_lifter_freq)
+    cepstrum_end = math.floor(sample_rate * 0.5 / hp_lifter_freq)
+    cepstrum_end = min(cepstrum_end, dft_len_ws // 4)
+
+    if cepstrum_end <= cepstrum_start:
+        raise ValueError(
+            "Expected cepstrum_start to be smaller than cepstrum_end."
+            f"Found: cepstrum_start: {cepstrum_start}, cepstrum_end: {cepstrum_end}."
+        )
+
+    noise_up_time_mult = torch.tensor(math.exp(-1.0 / (noise_up_time * measure_freq)), device=device)
+    noise_down_time_mult = torch.tensor(math.exp(-1.0 / (noise_down_time * measure_freq)), device=device)
+    measure_smooth_time_mult = math.exp(-1.0 / (measure_smooth_time * measure_freq))
+    trigger_meas_time_mult = math.exp(-1.0 / (trigger_time * measure_freq))
+
+    boot_count_max = int(boot_time * measure_freq - 0.5)
+    boot_count = measures_index = flushedLen_ns = 0
+
+    # pack batch
+    shape = waveform.size()
+    waveform = waveform.view(-1, shape[-1])
+
+    n_channels, ilen = waveform.size()
+
+    mean_meas = torch.zeros(n_channels, device=device)
+    spectrum = torch.zeros(n_channels, dft_len_ws, device=device)
+    noise_spectrum = torch.zeros(n_channels, dft_len_ws, device=device)
+    measures = torch.zeros(n_channels, measures_len, device=device)
+
+    has_triggered: bool = False
+    num_measures_to_flush: int = 0
+
+    pos = 0
+    for pos in range(measure_len_ns, ilen, measure_period_ns):
+        for i in range(n_channels):
+            meas: float = _measure(
+                measure_len_ws=measure_len_ws,
+                samples=waveform[i, pos - measure_len_ws : pos],
+                spectrum=spectrum[i],
+                noise_spectrum=noise_spectrum[i],
+                spectrum_window=spectrum_window,
+                spectrum_start=spectrum_start,
+                spectrum_end=spectrum_end,
+                cepstrum_window=cepstrum_window,
+                cepstrum_start=cepstrum_start,
+                cepstrum_end=cepstrum_end,
+                noise_reduction_amount=noise_reduction_amount,
+                measure_smooth_time_mult=measure_smooth_time_mult,
+                noise_up_time_mult=noise_up_time_mult,
+                noise_down_time_mult=noise_down_time_mult,
+                boot_count=boot_count,
+            )
+            measures[i, measures_index] = meas
+            mean_meas[i] = mean_meas[i] * trigger_meas_time_mult + meas * (1.0 - trigger_meas_time_mult)
+
+            has_triggered = has_triggered or (mean_meas[i] >= trigger_level)
+            if has_triggered:
+                n: int = measures_len
+                k: int = measures_index
+                jTrigger: int = n
+                jZero: int = n
+                j: int = 0
+
+                for j in range(n):
+                    if (measures[i, k] >= trigger_level) and (j <= jTrigger + gap_len):
+                        jZero = jTrigger = j
+                    elif (measures[i, k] == 0) and (jTrigger >= jZero):
+                        jZero = j
+                    k = (k + n - 1) % n
+                j = min(j, jZero)
+                # num_measures_to_flush = range_limit(j, num_measures_to_flush, n);
+                num_measures_to_flush = min(max(num_measures_to_flush, j), n)
+            # end if has_triggered
+        # end for channel
+        measures_index += 1
+        measures_index = measures_index % measures_len
+        if boot_count >= 0:
+            boot_count = -1 if boot_count == boot_count_max else boot_count + 1
+
+        if has_triggered:
+            flushedLen_ns = (measures_len - num_measures_to_flush) * measure_period_ns
+            break
+    # end for window
+    if not has_triggered and shape[-1] >= fixed_pre_trigger_len_ns:
+        return waveform[..., :fixed_pre_trigger_len_ns].view(shape[:-1] + torch.Size([fixed_pre_trigger_len_ns]))
+
+    res = waveform[:, max(pos - samplesLen_ns + flushedLen_ns, 0) :]
+    # unpack batch
+    return res.view(shape[:-1] + res.shape[-1:])