diff --git a/codes/data/__init__.py b/codes/data/__init__.py
index aa74d6e6..eda1db10 100644
--- a/codes/data/__init__.py
+++ b/codes/data/__init__.py
@@ -81,6 +81,8 @@ def create_dataset(dataset_opt, return_collate=False):
         default_params.update(dataset_opt)
         dataset_opt = munchify(default_params)
         from data.audio.stop_prediction_dataset import StopPredictionDataset as D
+    elif mode == 'stop_prediction2':
+        from data.audio.stop_prediction_dataset_2 import StopPredictionDataset as D
     else:
         raise NotImplementedError('Dataset [{:s}] is not recognized.'.format(mode))
     dataset = D(dataset_opt)
diff --git a/codes/data/audio/stop_prediction_dataset_2.py b/codes/data/audio/stop_prediction_dataset_2.py
new file mode 100644
index 00000000..61c93a45
--- /dev/null
+++ b/codes/data/audio/stop_prediction_dataset_2.py
@@ -0,0 +1,92 @@
+import os
+import pathlib
+import random
+
+from munch import munchify
+from torch.utils.data import Dataset
+import torch
+from tqdm import tqdm
+
+from data.audio.nv_tacotron_dataset import save_mel_buffer_to_file
+from models.tacotron2 import hparams
+from models.tacotron2.layers import TacotronSTFT
+from models.tacotron2.taco_utils import load_wav_to_torch
+from utils.util import opt_get
+
+
+# A dataset that consumes the result from the script `produce_libri_stretched_dataset`, which itself is a combined
+# set of clips from the librivox corpus of equal length with the sentence alignment labeled.
+class StopPredictionDataset(Dataset):
+    def __init__(self, opt):
+        path = opt['path']
+        label_compaction = opt_get(opt, ['label_compaction'], 1)
+        hp = munchify(hparams.create_hparams())
+        cache_path = os.path.join(path, 'cache.pth')
+        if os.path.exists(cache_path):
+            self.files = torch.load(cache_path)
+        else:
+            print("Building cache..")
+            self.files = list(pathlib.Path(path).glob('*.wav'))
+            torch.save(self.files, cache_path)
+        self.sampling_rate = 22050  # Fixed since the underlying data is also fixed at this SR.
+        self.mel_length = 2000
+        self.stft = TacotronSTFT(
+            hp.filter_length, hp.hop_length, hp.win_length,
+            hp.n_mel_channels, hp.sampling_rate, hp.mel_fmin,
+            hp.mel_fmax)
+        self.label_compaction = label_compaction
+
+    def __getitem__(self, index):
+        audio, _ = load_wav_to_torch(self.files[index])
+        starts, ends = torch.load(str(self.files[index]).replace('.wav', '_se.pth'))
+
+        if audio.std() > 1:
+            print(f"Something is very wrong with the given audio. std_dev={audio.std()}. file={self.files[index]}")
+            return None
+        audio.clip_(-1, 1)
+        mels = self.stft.mel_spectrogram(audio.unsqueeze(0))[:, :, :self.mel_length].squeeze(0)
+
+        # Form labels.
+        labels_start = torch.zeros((2000 // self.label_compaction,), dtype=torch.long)
+        for s in starts:
+            # Mel compaction operates at a ratio of 1/256, the dataset also allows further compaction.
+            s = s // (256 * self.label_compaction)
+            if s >= 2000//self.label_compaction:
+                continue
+            labels_start[s] = 1
+        labels_end = torch.zeros((2000 // self.label_compaction,), dtype=torch.long)
+        for e in ends:
+            e = e // (256 * self.label_compaction)
+            if e >= 2000//self.label_compaction:
+                continue
+            labels_end[e] = 1
+
+        return {
+            'mels': mels,
+            'labels_start': labels_start,
+            'labels_end': labels_end,
+        }
+
+
+    def __len__(self):
+        return len(self.files)
+
+
+if __name__ == '__main__':
+    opt = {
+        'path': 'D:\\data\\audio\\libritts\\stop_dataset',
+        'label_compaction': 4,
+    }
+    ds = StopPredictionDataset(opt)
+    j = 0
+    for i in tqdm(range(100)):
+        b = ds[random.randint(0, len(ds))]
+        start_indices = torch.nonzero(b['labels_start']).squeeze(1)
+        end_indices = torch.nonzero(b['labels_end']).squeeze(1)
+        assert len(end_indices) <= len(start_indices)  # There should always be more START tokens then END tokens.
+        for i in range(len(end_indices)):
+            s = start_indices[i].item()*4
+            e = end_indices[i].item()*4
+            m = b['mels'][:, s:e]
+            save_mel_buffer_to_file(m, f'{j}.npy')
+            j += 1
\ No newline at end of file
diff --git a/codes/models/gpt_voice/gpt_audio_segmentor.py b/codes/models/gpt_voice/gpt_audio_segmentor.py
index e801c533..8664bea0 100644
--- a/codes/models/gpt_voice/gpt_audio_segmentor.py
+++ b/codes/models/gpt_voice/gpt_audio_segmentor.py
@@ -62,27 +62,26 @@ class GptSegmentor(nn.Module):
                                attn_dropout=.1, ff_dropout=.1, non_causal_sequence_partition=self.MAX_MEL_FRAMES)
 
         self.final_norm = nn.LayerNorm(model_dim)
+        self.start_head = nn.Linear(model_dim, 1)
         self.stop_head = nn.Linear(model_dim, 1)
 
-    def forward(self, mel_inputs, termination_points=None):
+    def forward(self, mel_inputs, start_labels=None, end_labels=None):
         mel_emb = self.mel_encoder(mel_inputs)
         mel_emb = mel_emb.permute(0,2,1).contiguous()
         mel_emb = mel_emb + self.mel_pos_embedding(torch.arange(mel_emb.shape[1], device=mel_emb.device))
 
         enc = self.gpt(mel_emb)
-        stop_logits = self.final_norm(enc)
-        stop_logits = self.stop_head(stop_logits)
-
-        if termination_points is not None:
-            # The MEL gets decimated to 1/4 the size by the encoder, so we need to do the same to the termination points.
-            termination_points = F.interpolate(termination_points.unsqueeze(1), size=mel_emb.shape[1], mode='area').squeeze()
-            termination_points = (termination_points > 0).float()
+        logits = self.final_norm(enc)
+        stop_logits = self.stop_head(logits)
+        start_logits = self.start_head(logits)
 
+        if start_labels is not None:
             # Compute loss
-            loss = F.binary_cross_entropy_with_logits(stop_logits.squeeze(-1), termination_points)
-            return loss.mean()
+            start_loss = F.binary_cross_entropy_with_logits(start_logits.squeeze(-1), start_labels.float())
+            end_loss = F.binary_cross_entropy_with_logits(stop_logits.squeeze(-1), end_labels.float())
+            return start_loss.mean(), end_loss.mean()
         else:
-            return stop_logits
+            return start_logits, stop_logits
 
 
 
diff --git a/codes/models/tacotron2/hparams.py b/codes/models/tacotron2/hparams.py
index f3ccc52f..eeae2649 100644
--- a/codes/models/tacotron2/hparams.py
+++ b/codes/models/tacotron2/hparams.py
@@ -36,7 +36,7 @@ def create_hparams(hparams_string=None, verbose=False):
         input_sample_rate=22050,  # When different from sampling_rate, dataset automatically interpolates to sampling_rate
         sampling_rate=22050,
         filter_length=1024,
-        hop_length=256,
+        hop_length=256,  # This means a MEL is 1/256th the equivalent audio.
         win_length=1024,
         n_mel_channels=80,
         mel_fmin=0.0,
diff --git a/codes/scripts/audio/preprocess_libritts.py b/codes/scripts/audio/librivox/preprocess_libritts.py
similarity index 100%
rename from codes/scripts/audio/preprocess_libritts.py
rename to codes/scripts/audio/librivox/preprocess_libritts.py
diff --git a/codes/scripts/audio/librivox/produce_libri_stretched_dataset.py b/codes/scripts/audio/librivox/produce_libri_stretched_dataset.py
new file mode 100644
index 00000000..cb56ba76
--- /dev/null
+++ b/codes/scripts/audio/librivox/produce_libri_stretched_dataset.py
@@ -0,0 +1,99 @@
+# Combines all libriTTS WAV->text mappings into a single file
+import os
+import random
+
+import audio2numpy
+import torch
+from scipy.io import wavfile
+from tqdm import tqdm
+
+from utils.audio_resampler import AudioResampler
+
+
+def secs_to_frames(secs, sr):
+    return int(secs*sr)
+
+
+def get_audio_clip(audio, sr, start, end):
+    start = secs_to_frames(start, sr)
+    end = secs_to_frames(end, sr)
+    assert end > start
+    if end >= audio.shape[0]:
+        return None
+    return audio[start:end]
+
+
+# Produces an audio clip that would produce a MEL spectrogram of length mel_length by parsing parsed_sentences starting
+# at starting_index and moving forwards until the full length is finished.
+# Returns:
+#  On failure, returns tuple: (end_index, None, [], [])
+#  On success: returns tuple: (end_index, clip, start_points, end_points)
+#    clip.shape = (<mel_length*256>,)
+#    start_points = list(ints) where each sentence in the clip starts
+#    end_points = list(ints) where each sentence in the clip ends
+def gather_clip(audio, parsed_sentences, starting_index, sr, mel_length):
+    audio_length = (mel_length * 256) / sr  # This is technically a hyperparameter, but I have no intent of changing the MEL hop length.
+    starts = []
+    ends = []
+    start, end = parsed_sentences[starting_index][4:6]
+    start = float(start)
+    end = float(end)
+    clipstart = max(start - random.random() * 2, 0)  # Offset start backwards by up to 2 seconds
+    clipend = start + audio_length
+    clip = get_audio_clip(audio, sr, clipstart, clipend)
+    if clip is not None:
+        # Fetch the start and endpoints that go along with this clip.
+        starts.append(secs_to_frames(start-clipstart, sr))
+        while end < clipend:
+            ends.append(secs_to_frames(end-clipstart, sr))
+            starting_index += 1
+            if starting_index >= len(parsed_sentences):
+                break
+            start, end = parsed_sentences[starting_index][4:6]
+            start = float(start)
+            end = float(end)
+            if start < clipend:
+                starts.append(secs_to_frames(start-clipstart, sr))
+
+    return starting_index+1, clip, starts, ends
+
+
+if __name__ == '__main__':
+    full_book_root = 'D:\\data\\audio\\libritts\\full_books\\mp3'
+    libri_root = 'D:\\data\\audio\\libritts\\test-clean'
+    desired_mel_length = 2000
+    desired_audio_sample_rate = 22050
+    output_dir = 'D:\\data\\audio\\libritts\\stop_dataset_eval'
+
+    os.makedirs(output_dir, exist_ok=True)
+    j = 0
+    readers = os.listdir(libri_root)
+    for it, reader_dir in enumerate(tqdm(readers)):
+        #if it <= 145:  # Hey idiot! If you change this, change j too!
+        #    continue
+        reader = os.path.join(libri_root, reader_dir)
+        if not os.path.isdir(reader):
+            continue
+        for chapter_dir in os.listdir(reader):
+            chapter = os.path.join(reader, chapter_dir)
+            if not os.path.isdir(chapter):
+                continue
+            id = f'{os.path.basename(reader)}_{os.path.basename(chapter)}'
+            book_file = os.path.join(chapter, f'{id}.book.tsv')
+            if not os.path.exists(book_file):
+                continue
+            with open(book_file, encoding='utf-8') as f:
+                full_chapter, sr = audio2numpy.open_audio(os.path.join(full_book_root, reader_dir, chapter_dir, f'{chapter_dir}.mp3'))
+                full_chapter = torch.tensor(full_chapter)
+                if len(full_chapter.shape) > 1:
+                    full_chapter = full_chapter[:, 0]  # Only use mono-audio.
+                resampler = AudioResampler(sr, desired_audio_sample_rate, dtype=torch.float)
+                full_chapter = resampler(full_chapter.unsqueeze(0)).squeeze(0)
+                parsed_sentences = [line.strip().split('\t') for line in f]
+                i = 0
+                while i < len(parsed_sentences):
+                    i, clip, ns, ne = gather_clip(full_chapter, parsed_sentences, i, desired_audio_sample_rate, desired_mel_length)
+                    if clip is not None:
+                        wavfile.write(os.path.join(output_dir, f'{j}.wav'), desired_audio_sample_rate, clip.cpu().numpy())
+                        torch.save((ns,ne), os.path.join(output_dir, f'{j}_se.pth'))
+                        j += 1
diff --git a/codes/scripts/audio/use_vocoder.py b/codes/scripts/audio/use_vocoder.py
index fecda155..f2270c2c 100644
--- a/codes/scripts/audio/use_vocoder.py
+++ b/codes/scripts/audio/use_vocoder.py
@@ -1,6 +1,9 @@
+import pathlib
+
 import numpy
 import torch
 from scipy.io import wavfile
+from tqdm import tqdm
 
 from models.waveglow.waveglow import WaveGlow
 
@@ -21,8 +24,12 @@ class Vocoder:
 
 
 if __name__ == '__main__':
-    inp = '3.npy'
-    mel = torch.tensor(numpy.load(inp)).to('cuda')
-    vocoder = Vocoder()
-    wav = vocoder.transform_mel_to_audio(mel)
-    wavfile.write(f'{inp}.wav', 22050, wav[0].cpu().numpy())
\ No newline at end of file
+    path = 'data/audio'
+    files = list(pathlib.Path(path).glob('*.npy'))
+
+    for inp in tqdm(files):
+        inp = str(inp)
+        mel = torch.tensor(numpy.load(inp)).to('cuda')
+        vocoder = Vocoder()
+        wav = vocoder.transform_mel_to_audio(mel)
+        wavfile.write(f'{inp}.wav', 22050, wav[0].cpu().numpy())
\ No newline at end of file
diff --git a/codes/train.py b/codes/train.py
index f129db54..7b2da295 100644
--- a/codes/train.py
+++ b/codes/train.py
@@ -282,7 +282,7 @@ class Trainer:
 
 if __name__ == '__main__':
     parser = argparse.ArgumentParser()
-    parser.add_argument('-opt', type=str, help='Path to option YAML file.', default='../options/train_lrdvae_audio_clips.yml')
+    parser.add_argument('-opt', type=str, help='Path to option YAML file.', default='../options/train_gpt_stop_libritts.yml')
     parser.add_argument('--launcher', choices=['none', 'pytorch'], default='none', help='job launcher')
     parser.add_argument('--local_rank', type=int, default=0)
     args = parser.parse_args()
diff --git a/codes/utils/audio_resampler.py b/codes/utils/audio_resampler.py
new file mode 100644
index 00000000..53b98f1e
--- /dev/null
+++ b/codes/utils/audio_resampler.py
@@ -0,0 +1,249 @@
+import torch
+import numpy as np
+from scipy import special
+
+# Courtesy of https://www.kaggle.com/smallyellowduck/fast-audio-resampling-layer-in-pytorch
+class AudioResampler(torch.nn.Module):
+    """
+    Efficiently resample audio signals
+    This module is much faster than resampling with librosa because it exploits pytorch's efficient conv1d operations
+    This module is also faster than the existing pytorch resample function in
+    https://github.com/pytorch/audio/blob/b6a61c3f7d0267c77f8626167cc1eda0335f2753/torchaudio/compliance/kaldi.py#L892
+
+    Based on
+    https://github.com/danpovey/filtering/blob/master/lilfilter/resampler.py
+    with improvements to include additional filter types and input parameters that align with the librosa api
+    """
+
+    def __init__(self,
+                 input_sr, output_sr, dtype,
+                 num_zeros=64, cutoff_ratio=0.95, filter='kaiser', beta=14.0):
+        super().__init__()  # init the base class
+        """
+        This creates an object that can apply a symmetric FIR filter
+        based on torch.nn.functional.conv1d.
+
+        Args:
+          input_sr:  The input sampling rate, AS AN INTEGER..
+              does not have to be the real sampling rate but should
+              have the correct ratio with output_sr.
+          output_sr:  The output sampling rate, AS AN INTEGER.
+              It is the ratio with the input sampling rate that is
+              important here.
+          dtype:  The torch dtype to use for computations (would be preferrable to 
+               set things up so passing the dtype isn't necessary)
+          num_zeros: The number of zeros per side in the (sinc*hanning-window)
+              filter function.  More is more accurate, but 64 is already
+              quite a lot. The kernel size is 2*num_zeros + 1.
+          cutoff_ratio: The filter rolloff point as a fraction of the
+             Nyquist frequency.
+          filter: one of ['kaiser', 'kaiser_best', 'kaiser_fast', 'hann']
+          beta: parameter for 'kaiser' filter
+
+        You can think of this algorithm as dividing up the signals
+        (input,output) into blocks where there are `input_sr` input
+        samples and `output_sr` output samples.  Then we treat it
+        using convolutional code, imagining there are `input_sr`
+        input channels and `output_sr` output channels per time step.
+
+        """
+        assert isinstance(input_sr, int) and isinstance(output_sr, int)
+        if input_sr == output_sr:
+            self.resample_type = 'trivial'
+            return
+
+        def gcd(a, b):
+            """ Return the greatest common divisor of a and b"""
+            assert isinstance(a, int) and isinstance(b, int)
+            if b == 0:
+                return a
+            else:
+                return gcd(b, a % b)
+
+        d = gcd(input_sr, output_sr)
+        input_sr, output_sr = input_sr // d, output_sr // d
+
+        assert dtype in [torch.float32, torch.float64]
+        assert num_zeros > 3  # a reasonable bare minimum
+        np_dtype = np.float32 if dtype == torch.float32 else np.float64
+
+        assert filter in ['hann', 'kaiser', 'kaiser_best', 'kaiser_fast']
+
+        if filter == 'kaiser_best':
+            num_zeros = 64
+            beta = 14.769656459379492
+            cutoff_ratio = 0.9475937167399596
+            filter = 'kaiser'
+        elif filter == 'kaiser_fast':
+            num_zeros = 16
+            beta = 8.555504641634386
+            cutoff_ratio = 0.85
+            filter = 'kaiser'
+
+        # Define one 'block' of samples `input_sr` input samples
+        # and `output_sr` output samples.  We can divide up
+        # the samples into these blocks and have the blocks be
+        # in correspondence.
+
+        # The sinc function will have, on average, `zeros_per_block`
+        # zeros per block.
+        zeros_per_block = min(input_sr, output_sr) * cutoff_ratio
+
+        # The convolutional kernel size will be n = (blocks_per_side*2 + 1),
+        # i.e. we add that many blocks on each side of the central block.  The
+        # window radius (defined as distance from center to edge)
+        # is `blocks_per_side` blocks.  This ensures that each sample in the
+        # central block can "see" all the samples in its window.
+        #
+        # Assuming the following division is not exact, adding 1
+        # will have the same effect as rounding up.
+        # blocks_per_side = 1 + int(num_zeros / zeros_per_block)
+        blocks_per_side = int(np.ceil(num_zeros / zeros_per_block))
+
+        kernel_width = 2 * blocks_per_side + 1
+
+        # We want the weights as used by torch's conv1d code; format is
+        #  (out_channels, in_channels, kernel_width)
+        # https://pytorch.org/docs/stable/nn.functional.html
+        weights = torch.tensor((output_sr, input_sr, kernel_width), dtype=dtype)
+
+        # Computations involving time will be in units of 1 block.  Actually this
+        # is the same as the `canonical` time axis since each block has input_sr
+        # input samples, so it would be one of whatever time unit we are using
+        window_radius_in_blocks = blocks_per_side
+
+        # The `times` below will end up being the args to the sinc function.
+        # For the shapes of the things below, look at the args to `view`.  The terms
+        # below will get expanded to shape (output_sr, input_sr, kernel_width) through
+        # broadcasting
+        # We want it so that, assuming input_sr == output_sr, along the diagonal of
+        # the central block we have t == 0.
+        # The signs of the output_sr and input_sr terms need to be opposite.  The
+        # sign that the kernel_width term needs to be will depend on whether it's
+        # convolution or correlation, and the logic is tricky.. I will just find
+        # which sign works.
+
+        times = (
+                np.arange(output_sr, dtype=np_dtype).reshape((output_sr, 1, 1)) / output_sr -
+                np.arange(input_sr, dtype=np_dtype).reshape((1, input_sr, 1)) / input_sr -
+                (np.arange(kernel_width, dtype=np_dtype).reshape((1, 1, kernel_width)) - blocks_per_side))
+
+        def hann_window(a):
+            """
+            hann_window returns the Hann window on [-1,1], which is zero
+            if a < -1 or a > 1, and otherwise 0.5 + 0.5 cos(a*pi).
+            This is applied elementwise to a, which should be a NumPy array.
+
+            The heaviside function returns (a > 0 ? 1 : 0).
+            """
+            return np.heaviside(1 - np.abs(a), 0.0) * (0.5 + 0.5 * np.cos(a * np.pi))
+
+        def kaiser_window(a, beta):
+            w = special.i0(beta * np.sqrt(np.clip(1 - ((a - 0.0) / 1.0) ** 2.0, 0.0, 1.0))) / special.i0(beta)
+            return np.heaviside(1 - np.abs(a), 0.0) * w
+
+        # The weights below are a sinc function times a Hann-window function.
+        #
+        # Multiplication by zeros_per_block normalizes the sinc function
+        # (to compensate for scaling on the x-axis), so that the integral is 1.
+        #
+        # Division by input_sr normalizes the input function. Think of the input
+        # as a stream of dirac deltas passing through a low pass filter:
+        # in order to have the same magnitude as the original input function,
+        # we need to divide by the number of those deltas per unit time.
+        if filter == 'hann':
+            weights = (np.sinc(times * zeros_per_block)
+                       * hann_window(times / window_radius_in_blocks)
+                       * zeros_per_block / input_sr)
+        else:
+            weights = (np.sinc(times * zeros_per_block)
+                       * kaiser_window(times / window_radius_in_blocks, beta)
+                       * zeros_per_block / input_sr)
+
+        self.input_sr = input_sr
+        self.output_sr = output_sr
+
+        # weights has dim (output_sr, input_sr, kernel_width).
+        # If output_sr == 1, we can fold the input_sr into the
+        # kernel_width (i.e. have just 1 input channel); this will make the
+        # convolution faster and avoid unnecessary reshaping.
+
+        assert weights.shape == (output_sr, input_sr, kernel_width)
+        if output_sr == 1:
+            self.resample_type = 'integer_downsample'
+            self.padding = input_sr * blocks_per_side
+            weights = torch.tensor(weights, dtype=dtype, requires_grad=False)
+            self.weights = weights.transpose(1, 2).contiguous().view(1, 1, input_sr * kernel_width)
+
+        elif input_sr == 1:
+            # In this case we'll be doing conv_transpose, so we want the same weights that
+            # we would have if we were *downsampling* by this factor-- i.e. as if input_sr,
+            # output_sr had been swapped.
+            self.resample_type = 'integer_upsample'
+            self.padding = output_sr * blocks_per_side
+            weights = torch.tensor(weights, dtype=dtype, requires_grad=False)
+            self.weights = weights.flip(2).transpose(0, 2).contiguous().view(1, 1, output_sr * kernel_width)
+        else:
+            self.resample_type = 'general'
+            self.reshaped = False
+            self.padding = blocks_per_side
+            self.weights = torch.tensor(weights, dtype=dtype, requires_grad=False)
+
+        self.weights = torch.nn.Parameter(self.weights, requires_grad=False)
+
+    @torch.no_grad()
+    def forward(self, data):
+        """
+        Resample the data
+
+        Args:
+         input: a torch.Tensor with the same dtype as was passed to the
+           constructor.
+         There must be 2 axes, interpreted as (minibatch_size, sequence_length)...
+         the minibatch_size may in practice be the number of channels.
+
+        Return:  Returns a torch.Tensor with the same dtype as the input, and
+         dimension (minibatch_size, (sequence_length//input_sr)*output_sr),
+         where input_sr and output_sr are the corresponding constructor args,
+         modified to remove any common factors.
+        """
+        if self.resample_type == 'trivial':
+            return data
+        elif self.resample_type == 'integer_downsample':
+            (minibatch_size, seq_len) = data.shape
+            # will be shape (minibatch_size, in_channels, seq_len) with in_channels == 1
+            data = data.unsqueeze(1)
+            data = torch.nn.functional.conv1d(data,
+                                              self.weights,
+                                              stride=self.input_sr,
+                                              padding=self.padding)
+            # shape will be (minibatch_size, out_channels = 1, seq_len);
+            # return as (minibatch_size, seq_len)
+            return data.squeeze(1)
+
+        elif self.resample_type == 'integer_upsample':
+            data = data.unsqueeze(1)
+            data = torch.nn.functional.conv_transpose1d(data,
+                                                        self.weights,
+                                                        stride=self.output_sr,
+                                                        padding=self.padding)
+
+            return data.squeeze(1)
+        else:
+            assert self.resample_type == 'general'
+            (minibatch_size, seq_len) = data.shape
+            num_blocks = seq_len // self.input_sr
+            if num_blocks == 0:
+                # TODO: pad with zeros.
+                raise RuntimeError("Signal is too short to resample")
+            # data = data[:, 0:(num_blocks*self.input_sr)]  # Truncate input
+            data = data[:, 0:(num_blocks * self.input_sr)].view(minibatch_size, num_blocks, self.input_sr)
+
+            # Torch's conv1d expects input data with shape (minibatch, in_channels, time_steps), so transpose
+            data = data.transpose(1, 2)
+
+            data = torch.nn.functional.conv1d(data, self.weights,
+                                              padding=self.padding)
+
+            assert data.shape == (minibatch_size, self.output_sr, num_blocks)
+            return data.transpose(1, 2).contiguous().view(minibatch_size, num_blocks * self.output_sr)
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