diff --git a/TTS/encoder/models/resnet.py b/TTS/encoder/models/resnet.py
index 84e9967f..e75ab6c4 100644
--- a/TTS/encoder/models/resnet.py
+++ b/TTS/encoder/models/resnet.py
@@ -1,7 +1,7 @@
 import torch
 from torch import nn
 
-# from TTS.utils.audio import TorchSTFT
+# from TTS.utils.audio.torch_transforms import TorchSTFT
 from TTS.encoder.models.base_encoder import BaseEncoder
 
 
diff --git a/TTS/tts/layers/losses.py b/TTS/tts/layers/losses.py
index 5130ac0b..b9a03af1 100644
--- a/TTS/tts/layers/losses.py
+++ b/TTS/tts/layers/losses.py
@@ -8,7 +8,7 @@ from torch.nn import functional
 
 from TTS.tts.utils.helpers import sequence_mask
 from TTS.tts.utils.ssim import SSIMLoss as _SSIMLoss
-from TTS.utils.audio import TorchSTFT
+from TTS.utils.audio.torch_transforms import TorchSTFT
 
 
 # pylint: disable=abstract-method
diff --git a/TTS/utils/audio/__init__.py b/TTS/utils/audio/__init__.py
new file mode 100644
index 00000000..f18f2219
--- /dev/null
+++ b/TTS/utils/audio/__init__.py
@@ -0,0 +1 @@
+from TTS.utils.audio.processor import AudioProcessor
diff --git a/TTS/utils/audio/numpy_transforms.py b/TTS/utils/audio/numpy_transforms.py
new file mode 100644
index 00000000..f6f03855
--- /dev/null
+++ b/TTS/utils/audio/numpy_transforms.py
@@ -0,0 +1,425 @@
+from typing import Tuple
+
+import librosa
+import numpy as np
+import pyworld as pw
+import scipy
+import soundfile as sf
+
+# For using kwargs
+# pylint: disable=unused-argument
+
+
+def build_mel_basis(
+    *,
+    sample_rate: int = None,
+    fft_size: int = None,
+    num_mels: int = None,
+    mel_fmax: int = None,
+    mel_fmin: int = None,
+    **kwargs,
+) -> np.ndarray:
+    """Build melspectrogram basis.
+
+    Returns:
+        np.ndarray: melspectrogram basis.
+    """
+    if mel_fmax is not None:
+        assert mel_fmax <= sample_rate // 2
+        assert mel_fmax - mel_fmin > 0
+    return librosa.filters.mel(sr=sample_rate, n_fft=fft_size, n_mels=num_mels, fmin=mel_fmin, fmax=mel_fmax)
+
+
+def millisec_to_length(
+    *, frame_length_ms: int = None, frame_shift_ms: int = None, sample_rate: int = None, **kwargs
+) -> Tuple[int, int]:
+    """Compute hop and window length from milliseconds.
+
+    Returns:
+        Tuple[int, int]: hop length and window length for STFT.
+    """
+    factor = frame_length_ms / frame_shift_ms
+    assert (factor).is_integer(), " [!] frame_shift_ms should divide frame_length_ms"
+    win_length = int(frame_length_ms / 1000.0 * sample_rate)
+    hop_length = int(win_length / float(factor))
+    return win_length, hop_length
+
+
+def _log(x, base):
+    if base == 10:
+        return np.log10(x)
+    return np.log(x)
+
+
+def _exp(x, base):
+    if base == 10:
+        return np.power(10, x)
+    return np.exp(x)
+
+
+def amp_to_db(*, x: np.ndarray = None, gain: float = 1, base: int = 10, **kwargs) -> np.ndarray:
+    """Convert amplitude values to decibels.
+
+    Args:
+        x (np.ndarray): Amplitude spectrogram.
+        gain (float): Gain factor. Defaults to 1.
+        base (int): Logarithm base. Defaults to 10.
+
+    Returns:
+        np.ndarray: Decibels spectrogram.
+    """
+    assert (x < 0).sum() == 0, " [!] Input values must be non-negative."
+    return gain * _log(np.maximum(1e-8, x), base)
+
+
+# pylint: disable=no-self-use
+def db_to_amp(*, x: np.ndarray = None, gain: float = 1, base: int = 10, **kwargs) -> np.ndarray:
+    """Convert decibels spectrogram to amplitude spectrogram.
+
+    Args:
+        x (np.ndarray): Decibels spectrogram.
+        gain (float): Gain factor. Defaults to 1.
+        base (int): Logarithm base. Defaults to 10.
+
+    Returns:
+        np.ndarray: Amplitude spectrogram.
+    """
+    return _exp(x / gain, base)
+
+
+def preemphasis(*, x: np.ndarray, coef: float = 0.97, **kwargs) -> np.ndarray:
+    """Apply pre-emphasis to the audio signal. Useful to reduce the correlation between neighbouring signal values.
+
+    Args:
+        x (np.ndarray): Audio signal.
+
+    Raises:
+        RuntimeError: Preemphasis coeff is set to 0.
+
+    Returns:
+        np.ndarray: Decorrelated audio signal.
+    """
+    if coef == 0:
+        raise RuntimeError(" [!] Preemphasis is set 0.0.")
+    return scipy.signal.lfilter([1, -coef], [1], x)
+
+
+def deemphasis(*, x: np.ndarray = None, coef: float = 0.97, **kwargs) -> np.ndarray:
+    """Reverse pre-emphasis."""
+    if coef == 0:
+        raise RuntimeError(" [!] Preemphasis is set 0.0.")
+    return scipy.signal.lfilter([1], [1, -coef], x)
+
+
+def spec_to_mel(*, spec: np.ndarray, mel_basis: np.ndarray = None, **kwargs) -> np.ndarray:
+    """Convert a full scale linear spectrogram output of a network to a melspectrogram.
+
+    Args:
+        spec (np.ndarray): Normalized full scale linear spectrogram.
+
+    Shapes:
+        - spec: :math:`[C, T]`
+
+    Returns:
+        np.ndarray: Normalized melspectrogram.
+    """
+    return np.dot(mel_basis, spec)
+
+
+def mel_to_spec(*, mel: np.ndarray = None, mel_basis: np.ndarray = None, **kwargs) -> np.ndarray:
+    """Convert a melspectrogram to full scale spectrogram."""
+    assert (mel < 0).sum() == 0, " [!] Input values must be non-negative."
+    inv_mel_basis = np.linalg.pinv(mel_basis)
+    return np.maximum(1e-10, np.dot(inv_mel_basis, mel))
+
+
+def wav_to_spec(*, wav: np.ndarray = None, **kwargs) -> np.ndarray:
+    """Compute a spectrogram from a waveform.
+
+    Args:
+        wav (np.ndarray): Waveform. Shape :math:`[T_wav,]`
+
+    Returns:
+        np.ndarray: Spectrogram. Shape :math:`[C, T_spec]`. :math:`T_spec == T_wav / hop_length`
+    """
+    D = stft(y=wav, **kwargs)
+    S = np.abs(D)
+    return S.astype(np.float32)
+
+
+def wav_to_mel(*, wav: np.ndarray = None, mel_basis=None, **kwargs) -> np.ndarray:
+    """Compute a melspectrogram from a waveform."""
+    D = stft(y=wav, **kwargs)
+    S = spec_to_mel(spec=np.abs(D), mel_basis=mel_basis, **kwargs)
+    return S.astype(np.float32)
+
+
+def spec_to_wav(*, spec: np.ndarray, power: float = 1.5, **kwargs) -> np.ndarray:
+    """Convert a spectrogram to a waveform using Griffi-Lim vocoder."""
+    S = spec.copy()
+    return griffin_lim(spec=S**power, **kwargs)
+
+
+def mel_to_wav(*, mel: np.ndarray = None, power: float = 1.5, **kwargs) -> np.ndarray:
+    """Convert a melspectrogram to a waveform using Griffi-Lim vocoder."""
+    S = mel.copy()
+    S = mel_to_spec(mel=S, mel_basis=kwargs["mel_basis"])  # Convert back to linear
+    return griffin_lim(spec=S**power, **kwargs)
+
+
+### STFT and ISTFT ###
+def stft(
+    *,
+    y: np.ndarray = None,
+    fft_size: int = None,
+    hop_length: int = None,
+    win_length: int = None,
+    pad_mode: str = "reflect",
+    window: str = "hann",
+    center: bool = True,
+    **kwargs,
+) -> np.ndarray:
+    """Librosa STFT wrapper.
+
+    Check http://librosa.org/doc/main/generated/librosa.stft.html argument details.
+
+    Returns:
+        np.ndarray: Complex number array.
+    """
+    return librosa.stft(
+        y=y,
+        n_fft=fft_size,
+        hop_length=hop_length,
+        win_length=win_length,
+        pad_mode=pad_mode,
+        window=window,
+        center=center,
+    )
+
+
+def istft(
+    *,
+    y: np.ndarray = None,
+    fft_size: int = None,
+    hop_length: int = None,
+    win_length: int = None,
+    window: str = "hann",
+    center: bool = True,
+    **kwargs,
+) -> np.ndarray:
+    """Librosa iSTFT wrapper.
+
+    Check http://librosa.org/doc/main/generated/librosa.istft.html argument details.
+
+    Returns:
+        np.ndarray: Complex number array.
+    """
+    return librosa.istft(y, hop_length=hop_length, win_length=win_length, center=center, window=window)
+
+
+def griffin_lim(*, spec: np.ndarray = None, num_iter=60, **kwargs) -> np.ndarray:
+    angles = np.exp(2j * np.pi * np.random.rand(*spec.shape))
+    S_complex = np.abs(spec).astype(np.complex)
+    y = istft(y=S_complex * angles, **kwargs)
+    if not np.isfinite(y).all():
+        print(" [!] Waveform is not finite everywhere. Skipping the GL.")
+        return np.array([0.0])
+    for _ in range(num_iter):
+        angles = np.exp(1j * np.angle(stft(y=y, **kwargs)))
+        y = istft(y=S_complex * angles, **kwargs)
+    return y
+
+
+def compute_stft_paddings(
+    *, x: np.ndarray = None, hop_length: int = None, pad_two_sides: bool = False, **kwargs
+) -> Tuple[int, int]:
+    """Compute paddings used by Librosa's STFT. Compute right padding (final frame) or both sides padding
+    (first and final frames)"""
+    pad = (x.shape[0] // hop_length + 1) * hop_length - x.shape[0]
+    if not pad_two_sides:
+        return 0, pad
+    return pad // 2, pad // 2 + pad % 2
+
+
+def compute_f0(
+    *, x: np.ndarray = None, pitch_fmax: float = None, hop_length: int = None, sample_rate: int = None, **kwargs
+) -> np.ndarray:
+    """Compute pitch (f0) of a waveform using the same parameters used for computing melspectrogram.
+
+    Args:
+        x (np.ndarray): Waveform. Shape :math:`[T_wav,]`
+
+    Returns:
+        np.ndarray: Pitch. Shape :math:`[T_pitch,]`. :math:`T_pitch == T_wav / hop_length`
+
+    Examples:
+        >>> WAV_FILE = filename = librosa.util.example_audio_file()
+        >>> from TTS.config import BaseAudioConfig
+        >>> from TTS.utils.audio.processor import AudioProcessor        >>> conf = BaseAudioConfig(pitch_fmax=8000)
+        >>> ap = AudioProcessor(**conf)
+        >>> wav = ap.load_wav(WAV_FILE, sr=22050)[:5 * 22050]
+        >>> pitch = ap.compute_f0(wav)
+    """
+    assert pitch_fmax is not None, " [!] Set `pitch_fmax` before caling `compute_f0`."
+
+    f0, t = pw.dio(
+        x.astype(np.double),
+        fs=sample_rate,
+        f0_ceil=pitch_fmax,
+        frame_period=1000 * hop_length / sample_rate,
+    )
+    f0 = pw.stonemask(x.astype(np.double), f0, t, sample_rate)
+    return f0
+
+
+### Audio Processing ###
+def find_endpoint(
+    *,
+    wav: np.ndarray = None,
+    trim_db: float = -40,
+    sample_rate: int = None,
+    min_silence_sec=0.8,
+    gain: float = None,
+    base: int = None,
+    **kwargs,
+) -> int:
+    """Find the last point without silence at the end of a audio signal.
+
+    Args:
+        wav (np.ndarray): Audio signal.
+        threshold_db (int, optional): Silence threshold in decibels. Defaults to -40.
+        min_silence_sec (float, optional): Ignore silences that are shorter then this in secs. Defaults to 0.8.
+        gian (float, optional): Gain to be used to convert trim_db to trim_amp. Defaults to None.
+        base (int, optional): Base of the logarithm used to convert trim_db to trim_amp. Defaults to 10.
+
+    Returns:
+        int: Last point without silence.
+    """
+    window_length = int(sample_rate * min_silence_sec)
+    hop_length = int(window_length / 4)
+    threshold = db_to_amp(x=-trim_db, gain=gain, base=base)
+    for x in range(hop_length, len(wav) - window_length, hop_length):
+        if np.max(wav[x : x + window_length]) < threshold:
+            return x + hop_length
+    return len(wav)
+
+
+def trim_silence(
+    *,
+    wav: np.ndarray = None,
+    sample_rate: int = None,
+    trim_db: float = None,
+    win_length: int = None,
+    hop_length: int = None,
+    **kwargs,
+) -> np.ndarray:
+    """Trim silent parts with a threshold and 0.01 sec margin"""
+    margin = int(sample_rate * 0.01)
+    wav = wav[margin:-margin]
+    return librosa.effects.trim(wav, top_db=trim_db, frame_length=win_length, hop_length=hop_length)[0]
+
+
+def volume_norm(*, x: np.ndarray = None, coef: float = 0.95, **kwargs) -> np.ndarray:
+    """Normalize the volume of an audio signal.
+
+    Args:
+        x (np.ndarray): Raw waveform.
+        coef (float): Coefficient to rescale the maximum value. Defaults to 0.95.
+
+    Returns:
+        np.ndarray: Volume normalized waveform.
+    """
+    return x / abs(x).max() * coef
+
+
+def rms_norm(*, wav: np.ndarray = None, db_level: float = -27.0, **kwargs) -> np.ndarray:
+    r = 10 ** (db_level / 20)
+    a = np.sqrt((len(wav) * (r**2)) / np.sum(wav**2))
+    return wav * a
+
+
+def rms_volume_norm(*, x: np.ndarray, db_level: float = -27.0, **kwargs) -> np.ndarray:
+    """Normalize the volume based on RMS of the signal.
+
+    Args:
+        x (np.ndarray): Raw waveform.
+        db_level (float): Target dB level in RMS. Defaults to -27.0.
+
+    Returns:
+        np.ndarray: RMS normalized waveform.
+    """
+    assert -99 <= db_level <= 0, " [!] db_level should be between -99 and 0"
+    wav = rms_norm(wav=x, db_level=db_level)
+    return wav
+
+
+def load_wav(*, filename: str, sample_rate: int = None, resample: bool = False, **kwargs) -> np.ndarray:
+    """Read a wav file using Librosa and optionally resample, silence trim, volume normalize.
+
+    Resampling slows down loading the file significantly. Therefore it is recommended to resample the file before.
+
+    Args:
+        filename (str): Path to the wav file.
+        sr (int, optional): Sampling rate for resampling. Defaults to None.
+        resample (bool, optional): Resample the audio file when loading. Slows down the I/O time. Defaults to False.
+
+    Returns:
+        np.ndarray: Loaded waveform.
+    """
+    if resample:
+        # loading with resampling. It is significantly slower.
+        x, _ = librosa.load(filename, sr=sample_rate)
+    else:
+        # SF is faster than librosa for loading files
+        x, _ = sf.read(filename)
+    return x
+
+
+def save_wav(*, wav: np.ndarray, path: str, sample_rate: int = None, **kwargs) -> None:
+    """Save float waveform to a file using Scipy.
+
+    Args:
+        wav (np.ndarray): Waveform with float values in range [-1, 1] to save.
+        path (str): Path to a output file.
+        sr (int, optional): Sampling rate used for saving to the file. Defaults to None.
+    """
+    wav_norm = wav * (32767 / max(0.01, np.max(np.abs(wav))))
+    scipy.io.wavfile.write(path, sample_rate, wav_norm.astype(np.int16))
+
+
+def mulaw_encode(*, wav: np.ndarray, mulaw_qc: int, **kwargs) -> np.ndarray:
+    mu = 2**mulaw_qc - 1
+    signal = np.sign(wav) * np.log(1 + mu * np.abs(wav)) / np.log(1.0 + mu)
+    signal = (signal + 1) / 2 * mu + 0.5
+    return np.floor(
+        signal,
+    )
+
+
+def mulaw_decode(*, wav, mulaw_qc: int, **kwargs) -> np.ndarray:
+    """Recovers waveform from quantized values."""
+    mu = 2**mulaw_qc - 1
+    x = np.sign(wav) / mu * ((1 + mu) ** np.abs(wav) - 1)
+    return x
+
+
+def encode_16bits(*, x: np.ndarray, **kwargs) -> np.ndarray:
+    return np.clip(x * 2**15, -(2**15), 2**15 - 1).astype(np.int16)
+
+
+def quantize(*, x: np.ndarray, quantize_bits: int, **kwargs) -> np.ndarray:
+    """Quantize a waveform to a given number of bits.
+
+    Args:
+        x (np.ndarray): Waveform to quantize. Must be normalized into the range `[-1, 1]`.
+        quantize_bits (int): Number of quantization bits.
+
+    Returns:
+        np.ndarray: Quantized waveform.
+    """
+    return (x + 1.0) * (2**quantize_bits - 1) / 2
+
+
+def dequantize(*, x, quantize_bits, **kwargs) -> np.ndarray:
+    """Dequantize a waveform from the given number of bits."""
+    return 2 * x / (2**quantize_bits - 1) - 1
diff --git a/TTS/utils/audio.py b/TTS/utils/audio/processor.py
similarity index 84%
rename from TTS/utils/audio.py
rename to TTS/utils/audio/processor.py
index fc9d1942..5a63b444 100644
--- a/TTS/utils/audio.py
+++ b/TTS/utils/audio/processor.py
@@ -6,179 +6,14 @@ import pyworld as pw
 import scipy.io.wavfile
 import scipy.signal
 import soundfile as sf
-import torch
-from torch import nn
 
 from TTS.tts.utils.helpers import StandardScaler
 
-
-class TorchSTFT(nn.Module):  # pylint: disable=abstract-method
-    """Some of the audio processing funtions using Torch for faster batch processing.
-
-    TODO: Merge this with audio.py
-
-    Args:
-
-        n_fft (int):
-            FFT window size for STFT.
-
-        hop_length (int):
-            number of frames between STFT columns.
-
-        win_length (int, optional):
-            STFT window length.
-
-        pad_wav (bool, optional):
-            If True pad the audio with (n_fft - hop_length) / 2). Defaults to False.
-
-        window (str, optional):
-            The name of a function to create a window tensor that is applied/multiplied to each frame/window. Defaults to "hann_window"
-
-        sample_rate (int, optional):
-            target audio sampling rate. Defaults to None.
-
-        mel_fmin (int, optional):
-            minimum filter frequency for computing melspectrograms. Defaults to None.
-
-        mel_fmax (int, optional):
-            maximum filter frequency for computing melspectrograms. Defaults to None.
-
-        n_mels (int, optional):
-            number of melspectrogram dimensions. Defaults to None.
-
-        use_mel (bool, optional):
-            If True compute the melspectrograms otherwise. Defaults to False.
-
-        do_amp_to_db_linear (bool, optional):
-            enable/disable amplitude to dB conversion of linear spectrograms. Defaults to False.
-
-        spec_gain (float, optional):
-            gain applied when converting amplitude to DB. Defaults to 1.0.
-
-        power (float, optional):
-            Exponent for the magnitude spectrogram, e.g., 1 for energy, 2 for power, etc.  Defaults to None.
-
-        use_htk (bool, optional):
-            Use HTK formula in mel filter instead of Slaney.
-
-        mel_norm (None, 'slaney', or number, optional):
-            If 'slaney', divide the triangular mel weights by the width of the mel band
-            (area normalization).
-
-            If numeric, use `librosa.util.normalize` to normalize each filter by to unit l_p norm.
-            See `librosa.util.normalize` for a full description of supported norm values
-            (including `+-np.inf`).
-
-            Otherwise, leave all the triangles aiming for a peak value of 1.0. Defaults to "slaney".
-    """
-
-    def __init__(
-        self,
-        n_fft,
-        hop_length,
-        win_length,
-        pad_wav=False,
-        window="hann_window",
-        sample_rate=None,
-        mel_fmin=0,
-        mel_fmax=None,
-        n_mels=80,
-        use_mel=False,
-        do_amp_to_db=False,
-        spec_gain=1.0,
-        power=None,
-        use_htk=False,
-        mel_norm="slaney",
-    ):
-        super().__init__()
-        self.n_fft = n_fft
-        self.hop_length = hop_length
-        self.win_length = win_length
-        self.pad_wav = pad_wav
-        self.sample_rate = sample_rate
-        self.mel_fmin = mel_fmin
-        self.mel_fmax = mel_fmax
-        self.n_mels = n_mels
-        self.use_mel = use_mel
-        self.do_amp_to_db = do_amp_to_db
-        self.spec_gain = spec_gain
-        self.power = power
-        self.use_htk = use_htk
-        self.mel_norm = mel_norm
-        self.window = nn.Parameter(getattr(torch, window)(win_length), requires_grad=False)
-        self.mel_basis = None
-        if use_mel:
-            self._build_mel_basis()
-
-    def __call__(self, x):
-        """Compute spectrogram frames by torch based stft.
-
-        Args:
-            x (Tensor): input waveform
-
-        Returns:
-            Tensor: spectrogram frames.
-
-        Shapes:
-            x: [B x T] or [:math:`[B, 1, T]`]
-        """
-        if x.ndim == 2:
-            x = x.unsqueeze(1)
-        if self.pad_wav:
-            padding = int((self.n_fft - self.hop_length) / 2)
-            x = torch.nn.functional.pad(x, (padding, padding), mode="reflect")
-        # B x D x T x 2
-        o = torch.stft(
-            x.squeeze(1),
-            self.n_fft,
-            self.hop_length,
-            self.win_length,
-            self.window,
-            center=True,
-            pad_mode="reflect",  # compatible with audio.py
-            normalized=False,
-            onesided=True,
-            return_complex=False,
-        )
-        M = o[:, :, :, 0]
-        P = o[:, :, :, 1]
-        S = torch.sqrt(torch.clamp(M**2 + P**2, min=1e-8))
-
-        if self.power is not None:
-            S = S**self.power
-
-        if self.use_mel:
-            S = torch.matmul(self.mel_basis.to(x), S)
-        if self.do_amp_to_db:
-            S = self._amp_to_db(S, spec_gain=self.spec_gain)
-        return S
-
-    def _build_mel_basis(self):
-        mel_basis = librosa.filters.mel(
-            self.sample_rate,
-            self.n_fft,
-            n_mels=self.n_mels,
-            fmin=self.mel_fmin,
-            fmax=self.mel_fmax,
-            htk=self.use_htk,
-            norm=self.mel_norm,
-        )
-        self.mel_basis = torch.from_numpy(mel_basis).float()
-
-    @staticmethod
-    def _amp_to_db(x, spec_gain=1.0):
-        return torch.log(torch.clamp(x, min=1e-5) * spec_gain)
-
-    @staticmethod
-    def _db_to_amp(x, spec_gain=1.0):
-        return torch.exp(x) / spec_gain
-
-
 # pylint: disable=too-many-public-methods
-class AudioProcessor(object):
-    """Audio Processor for TTS used by all the data pipelines.
 
-    TODO: Make this a dataclass to replace `BaseAudioConfig`.
+
+class AudioProcessor(object):
+    """Audio Processor for TTS.
 
     Note:
         All the class arguments are set to default values to enable a flexible initialization
diff --git a/TTS/utils/audio/torch_transforms.py b/TTS/utils/audio/torch_transforms.py
new file mode 100644
index 00000000..d4523ad0
--- /dev/null
+++ b/TTS/utils/audio/torch_transforms.py
@@ -0,0 +1,163 @@
+import librosa
+import torch
+from torch import nn
+
+
+class TorchSTFT(nn.Module):  # pylint: disable=abstract-method
+    """Some of the audio processing funtions using Torch for faster batch processing.
+
+    Args:
+
+        n_fft (int):
+            FFT window size for STFT.
+
+        hop_length (int):
+            number of frames between STFT columns.
+
+        win_length (int, optional):
+            STFT window length.
+
+        pad_wav (bool, optional):
+            If True pad the audio with (n_fft - hop_length) / 2). Defaults to False.
+
+        window (str, optional):
+            The name of a function to create a window tensor that is applied/multiplied to each frame/window. Defaults to "hann_window"
+
+        sample_rate (int, optional):
+            target audio sampling rate. Defaults to None.
+
+        mel_fmin (int, optional):
+            minimum filter frequency for computing melspectrograms. Defaults to None.
+
+        mel_fmax (int, optional):
+            maximum filter frequency for computing melspectrograms. Defaults to None.
+
+        n_mels (int, optional):
+            number of melspectrogram dimensions. Defaults to None.
+
+        use_mel (bool, optional):
+            If True compute the melspectrograms otherwise. Defaults to False.
+
+        do_amp_to_db_linear (bool, optional):
+            enable/disable amplitude to dB conversion of linear spectrograms. Defaults to False.
+
+        spec_gain (float, optional):
+            gain applied when converting amplitude to DB. Defaults to 1.0.
+
+        power (float, optional):
+            Exponent for the magnitude spectrogram, e.g., 1 for energy, 2 for power, etc.  Defaults to None.
+
+        use_htk (bool, optional):
+            Use HTK formula in mel filter instead of Slaney.
+
+        mel_norm (None, 'slaney', or number, optional):
+            If 'slaney', divide the triangular mel weights by the width of the mel band
+            (area normalization).
+
+            If numeric, use `librosa.util.normalize` to normalize each filter by to unit l_p norm.
+            See `librosa.util.normalize` for a full description of supported norm values
+            (including `+-np.inf`).
+
+            Otherwise, leave all the triangles aiming for a peak value of 1.0. Defaults to "slaney".
+    """
+
+    def __init__(
+        self,
+        n_fft,
+        hop_length,
+        win_length,
+        pad_wav=False,
+        window="hann_window",
+        sample_rate=None,
+        mel_fmin=0,
+        mel_fmax=None,
+        n_mels=80,
+        use_mel=False,
+        do_amp_to_db=False,
+        spec_gain=1.0,
+        power=None,
+        use_htk=False,
+        mel_norm="slaney",
+    ):
+        super().__init__()
+        self.n_fft = n_fft
+        self.hop_length = hop_length
+        self.win_length = win_length
+        self.pad_wav = pad_wav
+        self.sample_rate = sample_rate
+        self.mel_fmin = mel_fmin
+        self.mel_fmax = mel_fmax
+        self.n_mels = n_mels
+        self.use_mel = use_mel
+        self.do_amp_to_db = do_amp_to_db
+        self.spec_gain = spec_gain
+        self.power = power
+        self.use_htk = use_htk
+        self.mel_norm = mel_norm
+        self.window = nn.Parameter(getattr(torch, window)(win_length), requires_grad=False)
+        self.mel_basis = None
+        if use_mel:
+            self._build_mel_basis()
+
+    def __call__(self, x):
+        """Compute spectrogram frames by torch based stft.
+
+        Args:
+            x (Tensor): input waveform
+
+        Returns:
+            Tensor: spectrogram frames.
+
+        Shapes:
+            x: [B x T] or [:math:`[B, 1, T]`]
+        """
+        if x.ndim == 2:
+            x = x.unsqueeze(1)
+        if self.pad_wav:
+            padding = int((self.n_fft - self.hop_length) / 2)
+            x = torch.nn.functional.pad(x, (padding, padding), mode="reflect")
+        # B x D x T x 2
+        o = torch.stft(
+            x.squeeze(1),
+            self.n_fft,
+            self.hop_length,
+            self.win_length,
+            self.window,
+            center=True,
+            pad_mode="reflect",  # compatible with audio.py
+            normalized=False,
+            onesided=True,
+            return_complex=False,
+        )
+        M = o[:, :, :, 0]
+        P = o[:, :, :, 1]
+        S = torch.sqrt(torch.clamp(M**2 + P**2, min=1e-8))
+
+        if self.power is not None:
+            S = S**self.power
+
+        if self.use_mel:
+            S = torch.matmul(self.mel_basis.to(x), S)
+        if self.do_amp_to_db:
+            S = self._amp_to_db(S, spec_gain=self.spec_gain)
+        return S
+
+    def _build_mel_basis(self):
+        mel_basis = librosa.filters.mel(
+            self.sample_rate,
+            self.n_fft,
+            n_mels=self.n_mels,
+            fmin=self.mel_fmin,
+            fmax=self.mel_fmax,
+            htk=self.use_htk,
+            norm=self.mel_norm,
+        )
+        self.mel_basis = torch.from_numpy(mel_basis).float()
+
+    @staticmethod
+    def _amp_to_db(x, spec_gain=1.0):
+        return torch.log(torch.clamp(x, min=1e-5) * spec_gain)
+
+    @staticmethod
+    def _db_to_amp(x, spec_gain=1.0):
+        return torch.exp(x) / spec_gain
diff --git a/TTS/vocoder/layers/losses.py b/TTS/vocoder/layers/losses.py
index 848e292b..befc43cc 100644
--- a/TTS/vocoder/layers/losses.py
+++ b/TTS/vocoder/layers/losses.py
@@ -4,7 +4,7 @@ import torch
 from torch import nn
 from torch.nn import functional as F
 
-from TTS.utils.audio import TorchSTFT
+from TTS.utils.audio.torch_transforms import TorchSTFT
 from TTS.vocoder.utils.distribution import discretized_mix_logistic_loss, gaussian_loss
 
 #################################
diff --git a/TTS/vocoder/models/univnet_discriminator.py b/TTS/vocoder/models/univnet_discriminator.py
index d6b0e5d5..34e2d1c2 100644
--- a/TTS/vocoder/models/univnet_discriminator.py
+++ b/TTS/vocoder/models/univnet_discriminator.py
@@ -3,7 +3,7 @@ import torch.nn.functional as F
 from torch import nn
 from torch.nn.utils import spectral_norm, weight_norm
 
-from TTS.utils.audio import TorchSTFT
+from TTS.utils.audio.torch_transforms import TorchSTFT
 from TTS.vocoder.models.hifigan_discriminator import MultiPeriodDiscriminator
 
 LRELU_SLOPE = 0.1
diff --git a/tests/aux_tests/test_audio_processor.py b/tests/aux_tests/test_audio_processor.py
index 56611692..d01aeffa 100644
--- a/tests/aux_tests/test_audio_processor.py
+++ b/tests/aux_tests/test_audio_processor.py
@@ -3,7 +3,7 @@ import unittest
 
 from tests import get_tests_input_path, get_tests_output_path, get_tests_path
 from TTS.config import BaseAudioConfig
-from TTS.utils.audio import AudioProcessor
+from TTS.utils.audio.processor import AudioProcessor
 
 TESTS_PATH = get_tests_path()
 OUT_PATH = os.path.join(get_tests_output_path(), "audio_tests")
diff --git a/tests/aux_tests/test_numpy_transforms.py b/tests/aux_tests/test_numpy_transforms.py
new file mode 100644
index 00000000..0c1836b9
--- /dev/null
+++ b/tests/aux_tests/test_numpy_transforms.py
@@ -0,0 +1,105 @@
+import math
+import os
+import unittest
+from dataclasses import dataclass
+
+import librosa
+import numpy as np
+from coqpit import Coqpit
+
+from tests import get_tests_input_path, get_tests_output_path, get_tests_path
+from TTS.utils.audio import numpy_transforms as np_transforms
+
+TESTS_PATH = get_tests_path()
+OUT_PATH = os.path.join(get_tests_output_path(), "audio_tests")
+WAV_FILE = os.path.join(get_tests_input_path(), "example_1.wav")
+
+os.makedirs(OUT_PATH, exist_ok=True)
+
+
+# pylint: disable=no-self-use
+
+
+class TestNumpyTransforms(unittest.TestCase):
+    def setUp(self) -> None:
+        @dataclass
+        class AudioConfig(Coqpit):
+            sample_rate: int = 22050
+            fft_size: int = 1024
+            num_mels: int = 256
+            mel_fmax: int = 1800
+            mel_fmin: int = 0
+            hop_length: int = 256
+            win_length: int = 1024
+            pitch_fmax: int = 450
+            trim_db: int = -1
+            min_silence_sec: float = 0.01
+            gain: float = 1.0
+            base: float = 10.0
+
+        self.config = AudioConfig()
+        self.sample_wav, _ = librosa.load(WAV_FILE, sr=self.config.sample_rate)
+
+    def test_build_mel_basis(self):
+        """Check if the mel basis is correctly built"""
+        print(" > Testing mel basis building.")
+        mel_basis = np_transforms.build_mel_basis(**self.config)
+        self.assertEqual(mel_basis.shape, (self.config.num_mels, self.config.fft_size // 2 + 1))
+
+    def test_millisec_to_length(self):
+        """Check if the conversion from milliseconds to length is correct"""
+        print(" > Testing millisec to length conversion.")
+        win_len, hop_len = np_transforms.millisec_to_length(
+            frame_length_ms=1000, frame_shift_ms=12.5, sample_rate=self.config.sample_rate
+        )
+        self.assertEqual(hop_len, int(12.5 / 1000.0 * self.config.sample_rate))
+        self.assertEqual(win_len, self.config.sample_rate)
+
+    def test_amplitude_db_conversion(self):
+        di = np.random.rand(11)
+        o1 = np_transforms.amp_to_db(x=di, gain=1.0, base=10)
+        o2 = np_transforms.db_to_amp(x=o1, gain=1.0, base=10)
+        np.testing.assert_almost_equal(di, o2, decimal=5)
+
+    def test_preemphasis_deemphasis(self):
+        di = np.random.rand(11)
+        o1 = np_transforms.preemphasis(x=di, coeff=0.95)
+        o2 = np_transforms.deemphasis(x=o1, coeff=0.95)
+        np.testing.assert_almost_equal(di, o2, decimal=5)
+
+    def test_spec_to_mel(self):
+        mel_basis = np_transforms.build_mel_basis(**self.config)
+        spec = np.random.rand(self.config.fft_size // 2 + 1, 20)  # [C, T]
+        mel = np_transforms.spec_to_mel(spec=spec, mel_basis=mel_basis)
+        self.assertEqual(mel.shape, (self.config.num_mels, 20))
+
+    def mel_to_spec(self):
+        mel_basis = np_transforms.build_mel_basis(**self.config)
+        mel = np.random.rand(self.config.num_mels, 20)  # [C, T]
+        spec = np_transforms.mel_to_spec(mel=mel, mel_basis=mel_basis)
+        self.assertEqual(spec.shape, (self.config.fft_size // 2 + 1, 20))
+
+    def test_wav_to_spec(self):
+        spec = np_transforms.wav_to_spec(wav=self.sample_wav, **self.config)
+        self.assertEqual(
+            spec.shape, (self.config.fft_size // 2 + 1, math.ceil(self.sample_wav.shape[0] / self.config.hop_length))
+        )
+
+    def test_wav_to_mel(self):
+        mel_basis = np_transforms.build_mel_basis(**self.config)
+        mel = np_transforms.wav_to_mel(wav=self.sample_wav, mel_basis=mel_basis, **self.config)
+        self.assertEqual(
+            mel.shape, (self.config.num_mels, math.ceil(self.sample_wav.shape[0] / self.config.hop_length))
+        )
+
+    def test_compute_f0(self):
+        pitch = np_transforms.compute_f0(x=self.sample_wav, **self.config)
+        mel_basis = np_transforms.build_mel_basis(**self.config)
+        mel = np_transforms.wav_to_mel(wav=self.sample_wav, mel_basis=mel_basis, **self.config)
+        assert pitch.shape[0] == mel.shape[1]
+
+    def test_load_wav(self):
+        wav = np_transforms.load_wav(filename=WAV_FILE, resample=False, sample_rate=22050)
+        wav_resample = np_transforms.load_wav(filename=WAV_FILE, resample=True, sample_rate=16000)
+        self.assertEqual(wav.shape, (self.sample_wav.shape[0],))
+        self.assertNotEqual(wav_resample.shape, (self.sample_wav.shape[0],))