Move resample to functional and add librosa comparison (#1402)

This PR additionally adds batching to kaldi compliance resample interface.

docs/source/functional.rst

@@ -55,7 +55,7 @@ apply_codec
 -----------
 
 .. autofunction:: apply_codec
-		  
+
 :hidden:`Complex Utility`
 ~~~~~~~~~~~~~~~~~~~~~~~~~
 
@@ -230,3 +230,8 @@ vad
 ---------------------------
 
 .. autofunction:: spectral_centroid
+
+:hidden:`resample`
+---------------------------
+
+.. autofunction:: resample

test/torchaudio_unittest/functional/librosa_compatibility_test.py

@@ -109,6 +109,25 @@ class TestFunctional(common_utils.TorchaudioTestCase):
 
         self.assertEqual(ta_out, lr_out, atol=5e-5, rtol=1e-5)
 
+    def test_resample(self):
+        input_path = common_utils.get_asset_path('sinewave.wav')
+        waveform, sample_rate = common_utils.load_wav(input_path)
+
+        upsample_rate = sample_rate * 2
+        downsample_rate = sample_rate // 2
+
+        ta_upsampled = F.resample(waveform, sample_rate, upsample_rate)
+        lr_upsampled = librosa.resample(waveform.squeeze(0).numpy(), sample_rate, upsample_rate)
+        lr_upsampled = torch.from_numpy(lr_upsampled).unsqueeze(0)
+
+        self.assertEqual(ta_upsampled, lr_upsampled, atol=1e-2, rtol=1e-5)
+
+        ta_downsampled = F.resample(waveform, sample_rate, downsample_rate)
+        lr_downsampled = librosa.resample(waveform.squeeze(0).numpy(), sample_rate, downsample_rate)
+        lr_downsampled = torch.from_numpy(lr_downsampled).unsqueeze(0)
+
+        self.assertEqual(ta_downsampled, lr_downsampled, atol=1e-2, rtol=1e-5)
+
 
 @unittest.skipIf(not LIBROSA_AVAILABLE, "Librosa not available")
 class TestPhaseVocoder(common_utils.TorchaudioTestCase):

torchaudio/compliance/kaldi.py

@@ -3,7 +3,6 @@ from typing import Tuple
 import math
 import torch
 from torch import Tensor
-from torch.nn import functional as F
 
 import torchaudio
 import torchaudio._internal.fft
@@ -753,71 +752,16 @@ def mfcc(
     return feature
 
 
-def _get_sinc_resample_kernel(orig_freq: int, new_freq: int, lowpass_filter_width: int,
-                              device: torch.device, dtype: torch.dtype):
-    assert lowpass_filter_width > 0
-    kernels = []
-    base_freq = min(orig_freq, new_freq)
-    # This will perform antialiasing filtering by removing the highest frequencies.
-    # At first I thought I only needed this when downsampling, but when upsampling
-    # you will get edge artifacts without this, as the edge is equivalent to zero padding,
-    # which will add high freq artifacts.
-    base_freq *= 0.99
-
-    # The key idea of the algorithm is that x(t) can be exactly reconstructed from x[i] (tensor)
-    # using the sinc interpolation formula:
-    #   x(t) = sum_i x[i] sinc(pi * orig_freq * (i / orig_freq - t))
-    # We can then sample the function x(t) with a different sample rate:
-    #    y[j] = x(j / new_freq)
-    # or,
-    #    y[j] = sum_i x[i] sinc(pi * orig_freq * (i / orig_freq - j / new_freq))
-
-    # We see here that y[j] is the convolution of x[i] with a specific filter, for which
-    # we take an FIR approximation, stopping when we see at least `lowpass_filter_width` zeros crossing.
-    # But y[j+1] is going to have a different set of weights and so on, until y[j + new_freq].
-    # Indeed:
-    # y[j + new_freq] = sum_i x[i] sinc(pi * orig_freq * ((i / orig_freq - (j + new_freq) / new_freq))
-    #                 = sum_i x[i] sinc(pi * orig_freq * ((i - orig_freq) / orig_freq - j / new_freq))
-    #                 = sum_i x[i + orig_freq] sinc(pi * orig_freq * (i / orig_freq - j / new_freq))
-    # so y[j+new_freq] uses the same filter as y[j], but on a shifted version of x by `orig_freq`.
-    # This will explain the F.conv1d after, with a stride of orig_freq.
-    width = math.ceil(lowpass_filter_width * orig_freq / base_freq)
-    # If orig_freq is still big after GCD reduction, most filters will be very unbalanced, i.e.,
-    # they will have a lot of almost zero values to the left or to the right...
-    # There is probably a way to evaluate those filters more efficiently, but this is kept for
-    # future work.
-    idx = torch.arange(-width, width + orig_freq, device=device, dtype=dtype)
-
-    for i in range(new_freq):
-        t = (-i / new_freq + idx / orig_freq) * base_freq
-        t = t.clamp_(-lowpass_filter_width, lowpass_filter_width)
-        t *= math.pi
-        # we do not use torch.hann_window here as we need to evaluate the window
-        # at specific positions, not over a regular grid.
-        window = torch.cos(t / lowpass_filter_width / 2)**2
-        kernel = torch.where(t == 0, torch.tensor(1.).to(t), torch.sin(t) / t)
-        kernel.mul_(window)
-        kernels.append(kernel)
-
-    scale = base_freq / orig_freq
-    return torch.stack(kernels).view(new_freq, 1, -1).mul_(scale), width
-
-
 def resample_waveform(waveform: Tensor,
                       orig_freq: float,
                       new_freq: float,
                       lowpass_filter_width: int = 6) -> Tensor:
-    r"""Resamples the waveform at the new frequency. This matches Kaldi's OfflineFeatureTpl ResampleWaveform
-    which uses a LinearResample (resample a signal at linearly spaced intervals to upsample/downsample
-    a signal). LinearResample (LR) means that the output signal is at linearly spaced intervals (i.e
-    the output signal has a frequency of ``new_freq``). It uses sinc/bandlimited interpolation to
-    upsample/downsample the signal.
+    r"""Resamples the waveform at the new frequency.
 
-    https://ccrma.stanford.edu/~jos/resample/Theory_Ideal_Bandlimited_Interpolation.html
-    https://github.com/kaldi-asr/kaldi/blob/master/src/feat/resample.h#L56
+    This is a wrapper around ``torchaudio.functional.resample``.
 
     Args:
-        waveform (Tensor): The input signal of size (c, n)
+        waveform (Tensor): The input signal of size (..., time)
         orig_freq (float): The original frequency of the signal
         new_freq (float): The desired frequency
         lowpass_filter_width (int, optional): Controls the sharpness of the filter, more == sharper
@@ -826,21 +770,4 @@ def resample_waveform(waveform: Tensor,
     Returns:
         Tensor: The waveform at the new frequency
     """
-    assert waveform.dim() == 2
-    assert orig_freq > 0.0 and new_freq > 0.0
-
-    orig_freq = int(orig_freq)
-    new_freq = int(new_freq)
-    gcd = math.gcd(orig_freq, new_freq)
-    orig_freq = orig_freq // gcd
-    new_freq = new_freq // gcd
-
-    kernel, width = _get_sinc_resample_kernel(orig_freq, new_freq, lowpass_filter_width,
-                                              waveform.device, waveform.dtype)
-
-    num_wavs, length = waveform.shape
-    waveform = F.pad(waveform, (width, width + orig_freq))
-    resampled = F.conv1d(waveform[:, None], kernel, stride=orig_freq)
-    resampled = resampled.transpose(1, 2).reshape(num_wavs, -1)
-    target_length = int(math.ceil(new_freq * length / orig_freq))
-    return resampled[..., :target_length]
+    return torchaudio.functional.resample(waveform, orig_freq, new_freq, lowpass_filter_width)

torchaudio/functional/__init__.py

@@ -19,6 +19,7 @@ from .functional import (
     spectrogram,
     spectral_centroid,
     apply_codec,
+    resample,
 )
 from .filtering import (
     allpass_biquad,
@@ -85,5 +86,6 @@ __all__ = [
     'riaa_biquad',
     'treble_biquad',
     'vad',
-    'apply_codec'
+    'apply_codec',
+    'resample',
 ]

torchaudio/functional/functional.py

@@ -33,6 +33,7 @@ __all__ = [
     'sliding_window_cmn',
     "spectral_centroid",
     "apply_codec",
+    "resample",
 ]
 
 
@@ -1209,3 +1210,105 @@ def compute_kaldi_pitch(
     )
     result = result.reshape(shape[:-1] + result.shape[-2:])
     return result
+
+
+def _get_sinc_resample_kernel(orig_freq: int, new_freq: int, lowpass_filter_width: int,
+                              device: torch.device, dtype: torch.dtype):
+    assert lowpass_filter_width > 0
+    kernels = []
+    base_freq = min(orig_freq, new_freq)
+    # This will perform antialiasing filtering by removing the highest frequencies.
+    # At first I thought I only needed this when downsampling, but when upsampling
+    # you will get edge artifacts without this, as the edge is equivalent to zero padding,
+    # which will add high freq artifacts.
+    base_freq *= 0.99
+
+    # The key idea of the algorithm is that x(t) can be exactly reconstructed from x[i] (tensor)
+    # using the sinc interpolation formula:
+    #   x(t) = sum_i x[i] sinc(pi * orig_freq * (i / orig_freq - t))
+    # We can then sample the function x(t) with a different sample rate:
+    #    y[j] = x(j / new_freq)
+    # or,
+    #    y[j] = sum_i x[i] sinc(pi * orig_freq * (i / orig_freq - j / new_freq))
+
+    # We see here that y[j] is the convolution of x[i] with a specific filter, for which
+    # we take an FIR approximation, stopping when we see at least `lowpass_filter_width` zeros crossing.
+    # But y[j+1] is going to have a different set of weights and so on, until y[j + new_freq].
+    # Indeed:
+    # y[j + new_freq] = sum_i x[i] sinc(pi * orig_freq * ((i / orig_freq - (j + new_freq) / new_freq))
+    #                 = sum_i x[i] sinc(pi * orig_freq * ((i - orig_freq) / orig_freq - j / new_freq))
+    #                 = sum_i x[i + orig_freq] sinc(pi * orig_freq * (i / orig_freq - j / new_freq))
+    # so y[j+new_freq] uses the same filter as y[j], but on a shifted version of x by `orig_freq`.
+    # This will explain the F.conv1d after, with a stride of orig_freq.
+    width = math.ceil(lowpass_filter_width * orig_freq / base_freq)
+    # If orig_freq is still big after GCD reduction, most filters will be very unbalanced, i.e.,
+    # they will have a lot of almost zero values to the left or to the right...
+    # There is probably a way to evaluate those filters more efficiently, but this is kept for
+    # future work.
+    idx = torch.arange(-width, width + orig_freq, device=device, dtype=dtype)
+
+    for i in range(new_freq):
+        t = (-i / new_freq + idx / orig_freq) * base_freq
+        t = t.clamp_(-lowpass_filter_width, lowpass_filter_width)
+        t *= math.pi
+        # we do not use torch.hann_window here as we need to evaluate the window
+        # at specific positions, not over a regular grid.
+        window = torch.cos(t / lowpass_filter_width / 2)**2
+        kernel = torch.where(t == 0, torch.tensor(1.).to(t), torch.sin(t) / t)
+        kernel.mul_(window)
+        kernels.append(kernel)
+
+    scale = base_freq / orig_freq
+    return torch.stack(kernels).view(new_freq, 1, -1).mul_(scale), width
+
+
+def resample(
+        waveform: Tensor,
+        orig_freq: float,
+        new_freq: float,
+        lowpass_filter_width: int = 6
+) -> Tensor:
+    r"""Resamples the waveform at the new frequency. This matches Kaldi's OfflineFeatureTpl ResampleWaveform
+    which uses a LinearResample (resample a signal at linearly spaced intervals to upsample/downsample
+    a signal). LinearResample (LR) means that the output signal is at linearly spaced intervals (i.e
+    the output signal has a frequency of ``new_freq``). It uses sinc/bandlimited interpolation to
+    upsample/downsample the signal.
+
+    https://ccrma.stanford.edu/~jos/resample/Theory_Ideal_Bandlimited_Interpolation.html
+    https://github.com/kaldi-asr/kaldi/blob/master/src/feat/resample.h#L56
+
+    Args:
+        waveform (Tensor): The input signal of dimension (..., time)
+        orig_freq (float): The original frequency of the signal
+        new_freq (float): The desired frequency
+        lowpass_filter_width (int, optional): Controls the sharpness of the filter, more == sharper
+            but less efficient. We suggest around 4 to 10 for normal use. (Default: ``6``)
+
+    Returns:
+        Tensor: The waveform at the new frequency of dimension (..., time).
+    """
+    # pack batch
+    shape = waveform.size()
+    waveform = waveform.view(-1, shape[-1])
+
+    assert orig_freq > 0.0 and new_freq > 0.0
+
+    orig_freq = int(orig_freq)
+    new_freq = int(new_freq)
+    gcd = math.gcd(orig_freq, new_freq)
+    orig_freq = orig_freq // gcd
+    new_freq = new_freq // gcd
+
+    kernel, width = _get_sinc_resample_kernel(orig_freq, new_freq, lowpass_filter_width,
+                                              waveform.device, waveform.dtype)
+
+    num_wavs, length = waveform.shape
+    waveform = torch.nn.functional.pad(waveform, (width, width + orig_freq))
+    resampled = torch.nn.functional.conv1d(waveform[:, None], kernel, stride=orig_freq)
+    resampled = resampled.transpose(1, 2).reshape(num_wavs, -1)
+    target_length = int(math.ceil(new_freq * length / orig_freq))
+    resampled = resampled[..., :target_length]
+
+    # unpack batch
+    resampled = resampled.view(shape[:-1] + resampled.shape[-1:])
+    return resampled

torchaudio/transforms.py

@@ -6,7 +6,6 @@ from typing import Callable, Optional
 import torch
 from torch import Tensor
 from torchaudio import functional as F
-from torchaudio.compliance import kaldi
 
 
 __all__ = [
@@ -649,17 +648,7 @@ class Resample(torch.nn.Module):
             Tensor: Output signal of dimension (..., time).
         """
         if self.resampling_method == 'sinc_interpolation':
-
-            # pack batch
-            shape = waveform.size()
-            waveform = waveform.view(-1, shape[-1])
-
-            waveform = kaldi.resample_waveform(waveform, self.orig_freq, self.new_freq)
-
-            # unpack batch
-            waveform = waveform.view(shape[:-1] + waveform.shape[-1:])
-
-            return waveform
+            return F.resample(waveform, self.orig_freq, self.new_freq)
 
         raise ValueError('Invalid resampling method: {}'.format(self.resampling_method))