Import torchaudio #1412 c0bfb03

parmeet · cpuhrsch · commit d48089a05fef · 2021-04-02T13:55:45.000-04:00
Summary: Import latest from github to fbcode Pass: 951 Skip: 19 Omit: 1 ListingSuccess: 26 Result available at: https://www.internalfb.com/intern/testinfra/testrun/8444249336935844 Reviewed By: mthrok Differential Revision: D27448988 fbshipit-source-id: 61f63ffa1295a31b4452abaf2c74ebfefb827dcf
diff --git a/docs/source/functional.rst b/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
diff --git a/test/torchaudio_unittest/common_utils/__init__.py b/test/torchaudio_unittest/common_utils/__init__.py
@@ -17,7 +17,6 @@
     skipIfNoModule,
     skipIfNoKaldi,
     skipIfNoSox,
-    skipIfNoSoxBackend,
 )
 from .wav_utils import (
     get_wav_data,
diff --git a/test/torchaudio_unittest/common_utils/case_utils.py b/test/torchaudio_unittest/common_utils/case_utils.py
@@ -7,7 +7,6 @@
 
 import torch
 from torch.testing._internal.common_utils import TestCase as PytorchTestCase
-import torchaudio
 from torchaudio._internal.module_utils import (
     is_module_available,
     is_sox_available,
@@ -96,8 +95,6 @@ def skipIfNoModule(module, display_name=None):
     return unittest.skipIf(not is_module_available(module), f'"{display_name}" is not available')
 
 
-skipIfNoSoxBackend = unittest.skipIf(
-    'sox' not in torchaudio.list_audio_backends(), 'Sox backend not available')
 skipIfNoCuda = unittest.skipIf(not torch.cuda.is_available(), reason='CUDA not available')
 skipIfNoSox = unittest.skipIf(not is_sox_available(), reason='Sox not available')
 skipIfNoKaldi = unittest.skipIf(not is_kaldi_available(), reason='Kaldi not available')
diff --git a/test/torchaudio_unittest/compliance_kaldi_test.py b/test/torchaudio_unittest/compliance_kaldi_test.py
@@ -47,7 +47,6 @@ def first_sample_of_frame(frame, window_size, window_shift, snip_edges):
 
 @common_utils.skipIfNoSox
 class Test_Kaldi(common_utils.TempDirMixin, common_utils.TorchaudioTestCase):
-    backend = 'sox_io'
 
     kaldi_output_dir = common_utils.get_asset_path('kaldi')
     test_filepath = common_utils.get_asset_path('kaldi_file.wav')
@@ -113,7 +112,7 @@ def _create_data_set(self):
         # clear the last 16 bits because they aren't used anyways
         y = ((y >> 16) << 16).float()
         torchaudio.save(self.test_filepath, y, sr)
-        sound, sample_rate = torchaudio.load(self.test_filepath, normalization=False)
+        sound, sample_rate = common_utils.load_wav(self.test_filepath, normalize=False)
         print(y >> 16)
         self.assertTrue(sample_rate == sr)
         self.assertEqual(y, sound)
diff --git a/test/torchaudio_unittest/functional/librosa_compatibility_test.py b/test/torchaudio_unittest/functional/librosa_compatibility_test.py
@@ -109,6 +109,25 @@ def test_amplitude_to_DB(self):
 
         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):
diff --git a/test/torchaudio_unittest/functional/sox_compatibility_test.py b/test/torchaudio_unittest/functional/sox_compatibility_test.py
@@ -2,7 +2,7 @@
 import torchaudio.functional as F
 
 from torchaudio_unittest.common_utils import (
-    skipIfNoSoxBackend,
+    skipIfNoSox,
     skipIfNoExec,
     TempDirMixin,
     TorchaudioTestCase,
@@ -14,7 +14,7 @@
 )
 
 
-@skipIfNoSoxBackend
+@skipIfNoSox
 @skipIfNoExec('sox')
 class TestFunctionalFiltering(TempDirMixin, TorchaudioTestCase):
     def run_sox_effect(self, input_file, effect):
diff --git a/test/torchaudio_unittest/transforms/sox_compatibility_test.py b/test/torchaudio_unittest/transforms/sox_compatibility_test.py
@@ -2,17 +2,19 @@
 from parameterized import parameterized
 
 from torchaudio_unittest.common_utils import (
-    skipIfNoSoxBackend,
+    skipIfNoSox,
     skipIfNoExec,
     TempDirMixin,
     TorchaudioTestCase,
     get_asset_path,
     sox_utils,
     load_wav,
+    save_wav,
+    get_whitenoise,
 )
 
 
-@skipIfNoSoxBackend
+@skipIfNoSox
 @skipIfNoExec('sox')
 class TestFunctionalFiltering(TempDirMixin, TorchaudioTestCase):
     def run_sox_effect(self, input_file, effect):
@@ -24,6 +26,14 @@ def assert_sox_effect(self, result, input_path, effects, atol=1e-04, rtol=1e-5):
         expected, _ = self.run_sox_effect(input_path, effects)
         self.assertEqual(result, expected, atol=atol, rtol=rtol)
 
+    def get_whitenoise(self, sample_rate=8000):
+        noise = get_whitenoise(
+            sample_rate=sample_rate, duration=3, scale_factor=0.9,
+        )
+        path = self.get_temp_path("whitenoise.wav")
+        save_wav(path, noise, sample_rate)
+        return noise, path
+
     @parameterized.expand([
         ('q', 'quarter_sine'),
         ('h', 'half_sine'),
diff --git a/torchaudio/compliance/kaldi.py b/torchaudio/compliance/kaldi.py
@@ -3,7 +3,6 @@
 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)
diff --git a/torchaudio/datasets/yesno.py b/torchaudio/datasets/yesno.py
@@ -16,7 +16,7 @@
     "release1": {
         "folder_in_archive": "waves_yesno",
         "url": "http://www.openslr.org/resources/1/waves_yesno.tar.gz",
-        "checksum": "30301975fd8c5cac4040c261c0852f57cfa8adbbad2ce78e77e4986957445f27",
+        "checksum": "c3f49e0cca421f96b75b41640749167b52118f232498667ca7a5f9416aef8e73",
     }
 }
 
@@ -54,7 +54,7 @@ def _parse_filesystem(self, root: str, url: str, folder_in_archive: str, downloa
             if not os.path.isdir(self._path):
                 if not os.path.isfile(archive):
                     checksum = _RELEASE_CONFIGS["release1"]["checksum"]
-                    download_url(url, root, hash_value=checksum, hash_type="md5")
+                    download_url(url, root, hash_value=checksum)
                 extract_archive(archive)
 
         if not os.path.isdir(self._path):
diff --git a/torchaudio/functional/__init__.py b/torchaudio/functional/__init__.py
@@ -19,6 +19,7 @@
     spectrogram,
     spectral_centroid,
     apply_codec,
+    resample,
 )
 from .filtering import (
     allpass_biquad,
@@ -85,5 +86,6 @@
     'riaa_biquad',
     'treble_biquad',
     'vad',
-    'apply_codec'
+    'apply_codec',
+    'resample',
 ]
diff --git a/torchaudio/functional/functional.py b/torchaudio/functional/functional.py
@@ -33,6 +33,7 @@
     '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
diff --git a/torchaudio/transforms.py b/torchaudio/transforms.py

Original file line number	Diff line number	Diff line change
`@@ -17,7 +17,6 @@`
`17`	`17`	`skipIfNoModule,`
`18`	`18`	`skipIfNoKaldi,`
`19`	`19`	`skipIfNoSox,`
`20`		`- skipIfNoSoxBackend,`
`21`	`20`	`)`
`22`	`21`	`from .wav_utils import (`
`23`	`22`	`get_wav_data,`