Refactor audio-style models into the audio folder

codes/data/__init__.py

@@ -1,5 +1,4 @@
 """create dataset and dataloader"""
-import logging
 import torch
 import torch.utils.data
 from munch import munchify
@@ -64,7 +63,7 @@ def create_dataset(dataset_opt, return_collate=False):
     elif mode == 'nv_tacotron':
         from data.audio.nv_tacotron_dataset import TextWavLoader as D
         from data.audio.nv_tacotron_dataset import TextMelCollate as C
-        from models.tacotron2.hparams import create_hparams
+        from models.audio.tts.tacotron2 import create_hparams
         default_params = create_hparams()
         default_params.update(dataset_opt)
         dataset_opt = munchify(default_params)
@@ -72,13 +71,13 @@ def create_dataset(dataset_opt, return_collate=False):
             collate = C()
     elif mode == 'paired_voice_audio':
         from data.audio.paired_voice_audio_dataset import TextWavLoader as D
-        from models.tacotron2.hparams import create_hparams
+        from models.audio.tts.tacotron2 import create_hparams
         default_params = create_hparams()
         default_params.update(dataset_opt)
         dataset_opt = munchify(default_params)
     elif mode == 'fast_paired_voice_audio':
         from data.audio.fast_paired_dataset import FastPairedVoiceDataset as D
-        from models.tacotron2.hparams import create_hparams
+        from models.audio.tts.tacotron2 import create_hparams
         default_params = create_hparams()
         default_params.update(dataset_opt)
         dataset_opt = munchify(default_params)

codes/data/audio/gpt_tts_dataset.py

@@ -6,9 +6,9 @@ import torch.utils.data
 from torch import LongTensor
 from tqdm import tqdm
 
-from models.tacotron2.taco_utils import load_filepaths_and_text
+from models.audio.tts.tacotron2 import load_filepaths_and_text
-from models.tacotron2.text import symbols
+from models.audio.tts.tacotron2 import symbols
-from models.tacotron2.text import text_to_sequence
+from models.audio.tts.tacotron2 import text_to_sequence
 
 
 class GptTtsDataset(torch.utils.data.Dataset):

codes/data/audio/grand_conjoined_dataset.py

@@ -1,7 +1,4 @@
 import os
-import os
-import random
-import shutil
 
 import torch
 import torch.nn.functional as F
@@ -9,19 +6,15 @@ import torch.utils.data
 import torchaudio
 from munch import munchify
 from tqdm import tqdm
-from transformers import GPT2TokenizerFast
 
-from data.audio.unsupervised_audio_dataset import load_audio, UnsupervisedAudioDataset
+from data.audio.unsupervised_audio_dataset import UnsupervisedAudioDataset
 from data.text.hf_datasets_wrapper import HfDataset
-from data.util import find_files_of_type, is_audio_file
-from models.tacotron2.taco_utils import load_filepaths_and_text
-from models.tacotron2.text import text_to_sequence
 from utils.util import opt_get
 
 
 def build_paired_voice_dataset(args):
     from data.audio.paired_voice_audio_dataset import TextWavLoader as D
-    from models.tacotron2.hparams import create_hparams
+    from models.audio.tts.tacotron2 import create_hparams
     default_params = create_hparams()
     default_params.update(args)
     dataset_opt = munchify(default_params)

codes/data/audio/nv_tacotron_dataset.py

@@ -1,5 +1,4 @@
 import os
-import os
 import random
 
 import torch
@@ -10,8 +9,8 @@ from tqdm import tqdm
 
 from data.audio.unsupervised_audio_dataset import load_audio
 from data.util import find_files_of_type, is_audio_file
-from models.tacotron2.taco_utils import load_filepaths_and_text
+from models.audio.tts.tacotron2 import load_filepaths_and_text
-from models.tacotron2.text import text_to_sequence
+from models.audio.tts.tacotron2 import text_to_sequence
 from utils.util import opt_get
 
 

codes/data/audio/paired_voice_audio_dataset.py

@@ -1,5 +1,4 @@
 import os
-import os
 import random
 import sys
 
@@ -10,8 +9,8 @@ import torchaudio
 from tqdm import tqdm
 
 from data.audio.unsupervised_audio_dataset import load_audio, load_similar_clips
-from models.tacotron2.taco_utils import load_filepaths_and_text
+from models.audio.tts.tacotron2 import load_filepaths_and_text
-from models.tacotron2.text import text_to_sequence, sequence_to_text
+from models.audio.tts.tacotron2 import text_to_sequence, sequence_to_text
 from utils.util import opt_get
 
 

codes/data/audio/unsupervised_audio_dataset.py

@@ -1,8 +1,6 @@
 import os
-import pathlib
 import random
 import sys
-from warnings import warn
 
 import torch
 import torch.utils.data
@@ -11,9 +9,8 @@ import torchaudio
 from audio2numpy import open_audio
 from tqdm import tqdm
 
-from data.audio.wav_aug import WavAugmentor
+from data.util import find_files_of_type, is_audio_file, load_paths_from_cache
-from data.util import find_files_of_type, is_wav_file, is_audio_file, load_paths_from_cache
+from models.audio.tts.tacotron2 import load_wav_to_torch
-from models.tacotron2.taco_utils import load_wav_to_torch
 from utils.util import opt_get
 
 
@@ -189,7 +186,7 @@ if __name__ == '__main__':
         'extra_samples': 4,
         'resample_clip': True,
     }
-    from data import create_dataset, create_dataloader, util
+    from data import create_dataset, create_dataloader
 
     ds = create_dataset(params)
     dl = create_dataloader(ds, params)

codes/data/audio/voice_tokenizer.py

@@ -1,16 +1,14 @@
 import re
 
-import datasets
 import torch
 from tokenizers import Tokenizer
 from tokenizers.models import BPE
 from tokenizers.pre_tokenizers import Whitespace
-from tokenizers.processors import ByteLevel
 from tokenizers.trainers import BpeTrainer
 
 from data.audio.paired_voice_audio_dataset import load_mozilla_cv, load_voxpopuli, load_tsv
-from models.tacotron2.taco_utils import load_filepaths_and_text
+from models.audio.tts.tacotron2 import load_filepaths_and_text
-from models.tacotron2.text.cleaners import english_cleaners
+from models.audio.tts.tacotron2.text.cleaners import english_cleaners
 
 
 def remove_extraneous_punctuation(word):

codes/data/audio/wav_aug.py

@@ -3,7 +3,7 @@ import random
 import torch
 import torchaudio.sox_effects
 
-from models.tacotron2.taco_utils import load_wav_to_torch
+from models.audio.tts.tacotron2 import load_wav_to_torch
 
 
 # Returns random double on [l,h] as a string

codes/models/audio/tts/ctc_code_generator.py

@@ -3,11 +3,11 @@ import json
 import torch
 import torch.nn as nn
 import torch.nn.functional as F
-from x_transformers import Encoder, XTransformer, TransformerWrapper
+from x_transformers import Encoder, TransformerWrapper
 
-from models.gpt_voice.unet_diffusion_tts6 import CheckpointedLayer
+from models.audio.tts.unet_diffusion_tts6 import CheckpointedLayer
-from models.gpt_voice.unified_voice2 import ConditioningEncoder
+from models.audio.tts.unified_voice2 import ConditioningEncoder
-from models.tacotron2.text.cleaners import english_cleaners
+from models.audio.tts.tacotron2.text.cleaners import english_cleaners
 from trainer.networks import register_model
 from utils.util import opt_get
 

codes/models/audio/tts/ctc_code_generator2.py

@@ -4,17 +4,11 @@ import json
 import torch
 import torch.nn as nn
 import torch.nn.functional as F
-from torch.nn import CrossEntropyLoss
+from transformers import T5Config, T5ForConditionalGeneration
-from transformers import T5Config, T5Model, T5PreTrainedModel, T5ForConditionalGeneration
-from transformers.file_utils import replace_return_docstrings
-from transformers.modeling_outputs import Seq2SeqLMOutput, BaseModelOutput
-from transformers.utils.model_parallel_utils import get_device_map, assert_device_map
-from x_transformers import Encoder, XTransformer
 
-from models.gpt_voice.transformer_builders import null_position_embeddings
+from models.audio.tts.transformer_builders import null_position_embeddings
-from models.gpt_voice.unet_diffusion_tts6 import CheckpointedLayer
+from models.audio.tts.unified_voice2 import ConditioningEncoder
-from models.gpt_voice.unified_voice2 import ConditioningEncoder
+from models.audio.tts.tacotron2.text.cleaners import english_cleaners
-from models.tacotron2.text.cleaners import english_cleaners
 from trainer.networks import register_model
 from utils.util import opt_get
 
@@ -146,7 +140,6 @@ def inf():
     model.load_state_dict(sd)
     raw_batch = torch.load('raw_batch.pth')
     with torch.no_grad():
-        from data.audio.unsupervised_audio_dataset import load_audio
         from scripts.audio.gen.speech_synthesis_utils import wav_to_mel
         ref_mel = torch.cat([wav_to_mel(raw_batch['conditioning'][0])[:, :, :256],
                                wav_to_mel(raw_batch['conditioning'][0])[:, :, :256]], dim=0).unsqueeze(0)

codes/models/audio/tts/tacotron2/hparams.py

@@ -1,5 +1,5 @@
 #import tensorflow as tf
-from models.tacotron2.text import symbols
+from models.audio.tts.tacotron2.text import symbols
 
 
 def create_hparams(hparams_string=None, verbose=False):

codes/models/audio/tts/tacotron2/layers.py

@@ -1,8 +1,8 @@
 import torch
 from librosa.filters import mel as librosa_mel_fn
-from models.tacotron2.audio_processing import dynamic_range_compression
+from models.audio.tts.tacotron2.audio_processing import dynamic_range_compression
-from models.tacotron2.audio_processing import dynamic_range_decompression
+from models.audio.tts.tacotron2.audio_processing import dynamic_range_decompression
-from models.tacotron2.stft import STFT
+from models.audio.tts.tacotron2.stft import STFT
 
 
 class LinearNorm(torch.nn.Module):

codes/models/audio/tts/tacotron2/stft.py

@@ -36,7 +36,7 @@ import torch.nn.functional as F
 from torch.autograd import Variable
 from scipy.signal import get_window
 from librosa.util import pad_center, tiny
-from models.tacotron2.audio_processing import window_sumsquare
+from models.audio.tts.tacotron2.audio_processing import window_sumsquare
 
 
 class STFT(torch.nn.Module):

codes/models/audio/tts/tacotron2/tacotron2.py

@@ -4,11 +4,10 @@ from munch import munchify
 from torch.autograd import Variable
 from torch import nn
 from torch.nn import functional as F
-from models.tacotron2.layers import ConvNorm, LinearNorm
+from models.audio.tts.tacotron2.layers import ConvNorm, LinearNorm
-from models.tacotron2.hparams import create_hparams
+from models.audio.tts.tacotron2.hparams import create_hparams
 from trainer.networks import register_model
-from models.tacotron2.taco_utils import get_mask_from_lengths
+from models.audio.tts.tacotron2.taco_utils import get_mask_from_lengths
-from utils.util import opt_get, checkpoint
 
 
 class LocationLayer(nn.Module):

codes/models/audio/tts/tacotron2/text/__init__.py

@@ -3,8 +3,8 @@ import re
 
 import torch
 
-from models.tacotron2.text import cleaners
+from models.audio.tts.tacotron2.text import cleaners
-from models.tacotron2.text.symbols import symbols
+from models.audio.tts.tacotron2.text.symbols import symbols
 
 
 # Mappings from symbol to numeric ID and vice versa:

codes/models/audio/tts/tacotron2/text/symbols.py

@@ -4,7 +4,7 @@
 Defines the set of symbols used in text input to the model.
 
 The default is a set of ASCII characters that works well for English or text that has been run through Unidecode. For other data, you can modify _characters. See TRAINING_DATA.md for details. '''
-from models.tacotron2.text import cmudict
+from models.audio.tts.tacotron2.text import cmudict
 
 _pad        = '_'
 _punctuation = '!\'(),.:;? '

codes/models/audio/tts/tacotron2/wave_tacotron.py

@@ -3,16 +3,16 @@ import torch
 from munch import munchify
 from torch.autograd import Variable
 from torch import nn
-from torch.nn import functional as F, Flatten
+from torch.nn import functional as F
 
 from models.arch_util import ConvGnSilu
 from models.diffusion.unet_diffusion import UNetModel, AttentionPool2d
-from models.tacotron2.layers import ConvNorm, LinearNorm
+from models.audio.tts.tacotron2.layers import LinearNorm
-from models.tacotron2.hparams import create_hparams
+from models.audio.tts.tacotron2.hparams import create_hparams
-from models.tacotron2.tacotron2 import Prenet, Attention, Encoder
+from models.audio.tts.tacotron2.tacotron2 import Attention, Encoder
 from trainer.networks import register_model
-from models.tacotron2.taco_utils import get_mask_from_lengths
+from models.audio.tts.tacotron2.taco_utils import get_mask_from_lengths
-from utils.util import opt_get, checkpoint
+from utils.util import checkpoint
 
 
 

codes/models/audio/tts/unet_diffusion_tts5.py

@@ -5,12 +5,11 @@ import torch
 import torch.nn as nn
 import torch.nn.functional as F
 from torch import autocast
-from x_transformers.x_transformers import AbsolutePositionalEmbedding, AttentionLayers
 
 from models.diffusion.nn import timestep_embedding, normalization, zero_module, conv_nd, linear
 from models.diffusion.unet_diffusion import AttentionBlock, TimestepEmbedSequential, \
     Downsample, Upsample, TimestepBlock
-from models.gpt_voice.mini_encoder import AudioMiniEncoder
+from models.audio.tts.mini_encoder import AudioMiniEncoder
 from scripts.audio.gen.use_diffuse_tts import ceil_multiple
 from trainer.networks import register_model
 from utils.util import checkpoint

codes/models/audio/tts/unet_diffusion_tts6.py

@@ -1,17 +1,15 @@
 import functools
 import random
-from collections import OrderedDict
 
 import torch
 import torch.nn as nn
 import torch.nn.functional as F
 from torch import autocast
-from x_transformers.x_transformers import AbsolutePositionalEmbedding, AttentionLayers
 
 from models.diffusion.nn import timestep_embedding, normalization, zero_module, conv_nd, linear
 from models.diffusion.unet_diffusion import AttentionBlock, TimestepEmbedSequential, \
     Downsample, Upsample, TimestepBlock
-from models.gpt_voice.mini_encoder import AudioMiniEncoder
+from models.audio.tts.mini_encoder import AudioMiniEncoder
 from scripts.audio.gen.use_diffuse_tts import ceil_multiple
 from trainer.networks import register_model
 from utils.util import checkpoint

codes/models/audio/tts/unet_diffusion_tts7.py

@@ -1,17 +1,15 @@
 import functools
 import random
-from collections import OrderedDict
 
 import torch
 import torch.nn as nn
 import torch.nn.functional as F
 from torch import autocast
-from x_transformers.x_transformers import AbsolutePositionalEmbedding, AttentionLayers, CrossAttender
 
 from models.diffusion.nn import timestep_embedding, normalization, zero_module, conv_nd, linear
 from models.diffusion.unet_diffusion import AttentionBlock, TimestepEmbedSequential, \
     Downsample, Upsample, TimestepBlock
-from models.gpt_voice.mini_encoder import AudioMiniEncoder
+from models.audio.tts.mini_encoder import AudioMiniEncoder
 from scripts.audio.gen.use_diffuse_tts import ceil_multiple
 from trainer.networks import register_model
 from utils.util import checkpoint

codes/models/audio/tts/unet_diffusion_tts8.py

@@ -1,20 +1,17 @@
 import functools
 import random
-from collections import OrderedDict
 
 import torch
 import torch.nn as nn
 import torch.nn.functional as F
 from torch import autocast
-from x_transformers.x_transformers import AbsolutePositionalEmbedding, AttentionLayers, CrossAttender
 
 from models.diffusion.nn import timestep_embedding, normalization, zero_module, conv_nd, linear
-from models.diffusion.unet_diffusion import AttentionBlock, TimestepEmbedSequential, \
+from models.diffusion.unet_diffusion import TimestepEmbedSequential, \
-    Downsample, Upsample, TimestepBlock
+    Downsample, Upsample
-from models.gpt_voice.mini_encoder import AudioMiniEncoder
+from models.audio.tts.mini_encoder import AudioMiniEncoder
 from scripts.audio.gen.use_diffuse_tts import ceil_multiple
 from trainer.networks import register_model
-from utils.util import checkpoint
 from x_transformers import Encoder, ContinuousTransformerWrapper
 
 

codes/models/audio/tts/unet_diffusion_tts9.py

@@ -1,6 +1,4 @@
-import functools
 import random
-from collections import OrderedDict
 
 import torch
 import torch.nn as nn
@@ -11,11 +9,11 @@ from x_transformers import Encoder
 from models.diffusion.nn import timestep_embedding, normalization, zero_module, conv_nd, linear
 from models.diffusion.unet_diffusion import AttentionBlock, TimestepEmbedSequential, \
     Downsample, Upsample, TimestepBlock
-from models.gpt_voice.mini_encoder import AudioMiniEncoder
+from models.audio.tts.mini_encoder import AudioMiniEncoder
-from models.gpt_voice.unet_diffusion_tts7 import CheckpointedXTransformerEncoder
+from models.audio.tts.unet_diffusion_tts7 import CheckpointedXTransformerEncoder
 from scripts.audio.gen.use_diffuse_tts import ceil_multiple
 from trainer.networks import register_model
-from utils.util import checkpoint, opt_get
+from utils.util import checkpoint
 
 
 def is_latent(t):

codes/models/audio/tts/unet_diffusion_vocoder_with_ref.py

@@ -1,13 +1,11 @@
-from models.diffusion.fp16_util import convert_module_to_f32, convert_module_to_f16
 from models.diffusion.nn import timestep_embedding, normalization, zero_module, conv_nd, linear
-from models.diffusion.unet_diffusion import AttentionPool2d, AttentionBlock, ResBlock, TimestepEmbedSequential, \
+from models.diffusion.unet_diffusion import AttentionBlock, ResBlock, TimestepEmbedSequential, \
     Downsample, Upsample
 import torch
 import torch.nn as nn
 
-from models.gpt_voice.mini_encoder import AudioMiniEncoder, EmbeddingCombiner
+from models.audio.tts.mini_encoder import AudioMiniEncoder
 from trainer.networks import register_model
-from utils.util import get_mask_from_lengths
 
 
 class DiscreteSpectrogramConditioningBlock(nn.Module):

codes/models/audio/tts/unified_voice2.py

@@ -1,15 +1,12 @@
-import functools
-
 import torch
 import torch.nn as nn
 import torch.nn.functional as F
-from transformers import GPT2Model, GPT2Config, GPT2PreTrainedModel
+from transformers import GPT2Config, GPT2PreTrainedModel
 from transformers.modeling_outputs import CausalLMOutputWithCrossAttentions
 from transformers.utils.model_parallel_utils import get_device_map, assert_device_map
 
 from models.arch_util import AttentionBlock
-from models.gpt_voice.transformer_builders import build_hf_gpt_transformer
+from models.audio.tts.transformer_builders import build_hf_gpt_transformer
-from models.tacotron2.text import symbols
 from trainer.networks import register_model
 from utils.util import opt_get
 

codes/models/audio/tts/voice_voice_clip.py

@@ -2,11 +2,9 @@ import random
 import torch
 import torch.nn as nn
 import torch.nn.functional as F
-from einops import rearrange
 from torch import einsum
 
-from models.gpt_voice.mini_encoder import AudioMiniEncoder
+from models.audio.tts.mini_encoder import AudioMiniEncoder
-from models.lucidrains.dalle.transformer import Transformer
 from trainer.injectors.spec_augment import spec_augment
 from trainer.networks import register_model
 from utils.util import opt_get

codes/models/audio/tts/w2v_matcher.py

@@ -5,7 +5,7 @@ import torch.nn as nn
 import torch.nn.functional as F
 from x_transformers import Encoder, Decoder, ContinuousTransformerWrapper
 
-from models.gpt_voice.mini_encoder import AudioMiniEncoder
+from models.audio.tts.mini_encoder import AudioMiniEncoder
 from trainer.networks import register_model
 
 
@@ -135,6 +135,31 @@ class Wav2VecMatcher(nn.Module):
 
         return ce_loss, mse_loss
 
+    def find_matching_w2v_logit(self, key, w2v_logit_iterable):
+        pass
+
+    def sample(self, text_tokens, conditioning_clip, w2v_logit_iterable, audio_clip_iterable):
+        text_emb = self.text_embedding(text_tokens)
+        cond_emb = self.conditioning_encoder(conditioning_clip)
+        enc_inputs = torch.cat([cond_emb.unsqueeze(1), text_emb], dim=1)
+        dec_context = self.encoder(enc_inputs)
+        dec_inputs = self.decoder_start_embedding
+        count = 0
+        while count < 400:
+            dec_out = self.decoder(dec_inputs, context=dec_context)
+
+            # Check if that was EOS.
+            l2 = F.mse_loss(dec_out[:,-1], self.decoder_stop_embedding)
+            if l2 < .1:  # TODO: fix threshold.
+                break
+
+            # Find a matching w2v logit from the given iterable.
+            matching_logit_index = self.find_matching_w2v_logit(dec_out[:,-1], w2v_logit_iterable)
+            matching_logit = w2v_logit_iterable[matching_logit_index]
+            dec_inputs = torch.cat([dec_inputs, self.w2v_value_encoder(matching_logit).unsqueeze(1)], dim=1)
+            produced_audio = torch.cat([produced_audio, audio_clip_iterable[matching_logit_index]], dim=-1)
+        return produced_audio
+
 
 @register_model
 def register_w2v_matcher(opt_net, opt):

codes/models/audio/vocoders/waveglow/denoiser.py

@@ -1,6 +1,6 @@
 import sys
 
-from models.tacotron2.stft import STFT
+from models.audio.tts.tacotron2.stft import STFT
 
 sys.path.append('tacotron2')
 import torch

codes/models/clip/text_cond_clip.py

@@ -1,10 +1,9 @@
 import torch
 import torch.nn as nn
 import torch.nn.functional as F
-from einops import rearrange
 from torch import einsum
 
-from models.gpt_voice.unified_voice2 import ConditioningEncoder
+from models.audio.tts.unified_voice2 import ConditioningEncoder
 from models.lucidrains.dalle.transformer import Transformer
 from trainer.networks import register_model
 from utils.util import opt_get

codes/models/spleeter/estimator.py

@@ -1,137 +0,0 @@
-import math
-
-import torch
-import torch.nn.functional as F
-from torch import nn
-from torch import istft
-
-from .unet import UNet
-from .util import tf2pytorch
-
-
-def load_ckpt(model, ckpt):
-    state_dict = model.state_dict()
-    for k, v in ckpt.items():
-        if k in state_dict:
-            target_shape = state_dict[k].shape
-            assert target_shape == v.shape
-            state_dict.update({k: torch.from_numpy(v)})
-        else:
-            print('Ignore ', k)
-
-    model.load_state_dict(state_dict)
-    return model
-
-
-def pad_and_partition(tensor, T):
-    """
-    pads zero and partition tensor into segments of length T
-
-    Args:
-        tensor(Tensor): BxCxFxL
-
-    Returns:
-        tensor of size (B*[L/T] x C x F x T)
-    """
-    old_size = tensor.size(3)
-    new_size = math.ceil(old_size/T) * T
-    tensor = F.pad(tensor, [0, new_size - old_size])
-    [b, c, t, f] = tensor.shape
-    split = new_size // T
-    return torch.cat(torch.split(tensor, T, dim=3), dim=0)
-
-
-class Estimator(nn.Module):
-    def __init__(self, num_instrumments, checkpoint_path):
-        super(Estimator, self).__init__()
-
-        # stft config
-        self.F = 1024
-        self.T = 512
-        self.win_length = 4096
-        self.hop_length = 1024
-        self.win = torch.hann_window(self.win_length)
-
-        ckpts = tf2pytorch(checkpoint_path, num_instrumments)
-
-        # filter
-        self.instruments = nn.ModuleList()
-        for i in range(num_instrumments):
-            print('Loading model for instrumment {}'.format(i))
-            net = UNet(2)
-            ckpt = ckpts[i]
-            net = load_ckpt(net, ckpt)
-            net.eval()  # change mode to eval
-            self.instruments.append(net)
-
-    def compute_stft(self, wav):
-        """
-        Computes stft feature from wav
-
-        Args:
-            wav (Tensor): B x L
-        """
-
-        stft = torch.stft(
-            wav, self.win_length, hop_length=self.hop_length, window=self.win.to(wav.device))
-
-        # only keep freqs smaller than self.F
-        stft = stft[:, :self.F, :, :]
-        real = stft[:, :, :, 0]
-        im = stft[:, :, :, 1]
-        mag = torch.sqrt(real ** 2 + im ** 2)
-
-        return stft, mag
-
-    def inverse_stft(self, stft):
-        """Inverses stft to wave form"""
-
-        pad = self.win_length // 2 + 1 - stft.size(1)
-        stft = F.pad(stft, (0, 0, 0, 0, 0, pad))
-        wav = istft(stft, self.win_length, hop_length=self.hop_length,
-                    window=self.win.to(stft.device))
-        return wav.detach()
-
-    def separate(self, wav):
-        """
-        Separates stereo wav into different tracks corresponding to different instruments
-
-        Args:
-            wav (tensor): B x L
-        """
-
-        # stft - B X F x L x 2
-        # stft_mag - B X F x L
-        stft, stft_mag = self.compute_stft(wav)
-
-        L = stft.size(2)
-
-        stft_mag = stft_mag.unsqueeze(1).repeat(1,2,1,1)  # B x 2 x F x T
-        stft_mag = pad_and_partition(stft_mag, self.T)  # B x 2 x F x T
-        stft_mag = stft_mag.transpose(2, 3)  # B x 2 x T x F
-
-        # compute instruments' mask
-        masks = []
-        for net in self.instruments:
-            mask = net(stft_mag)
-            masks.append(mask)
-
-        # compute denominator
-        mask_sum = sum([m ** 2 for m in masks])
-        mask_sum += 1e-10
-
-        wavs = []
-        for mask in masks:
-            mask = (mask ** 2 + 1e-10/2)/(mask_sum)
-            mask = mask.transpose(2, 3)  # B x 2 X F x T
-
-            mask = torch.cat(
-                torch.split(mask, 1, dim=0), dim=3)
-
-            mask = mask[:,0,:,:L].unsqueeze(-1)  # 2 x F x L x 1
-            stft_masked = stft * mask
-            wav_masked = self.inverse_stft(stft_masked)
-
-            wavs.append(wav_masked)
-
-        return wavs

codes/models/spleeter/separator.py

@@ -1,32 +0,0 @@
-import torch
-import torch.nn.functional as F
-
-from models.spleeter.estimator import Estimator
-
-
-class Separator:
-    def __init__(self, model_path, input_sr=44100, device='cuda'):
-        self.model = Estimator(2, model_path).to(device)
-        self.device = device
-        self.input_sr = input_sr
-
-    def separate(self, npwav, normalize=False):
-        if not isinstance(npwav, torch.Tensor):
-            assert len(npwav.shape) == 1
-            wav = torch.tensor(npwav, device=self.device)
-            wav = wav.view(1,-1)
-        else:
-            assert len(npwav.shape) == 2  # Input should be BxL
-            wav = npwav.to(self.device)
-
-        if normalize:
-            wav = wav / (wav.max() + 1e-8)
-
-        # Spleeter expects audio input to be 44.1kHz.
-        wav = F.interpolate(wav.unsqueeze(1), mode='nearest', scale_factor=44100/self.input_sr).squeeze(1)
-        res = self.model.separate(wav)
-        res = [F.interpolate(r.unsqueeze(1), mode='nearest', scale_factor=self.input_sr/44100)[:,0] for r in res]
-        return {
-            'vocals': res[0].cpu().numpy(),
-            'accompaniment': res[1].cpu().numpy()
-        }

codes/models/spleeter/unet.py

@@ -1,80 +0,0 @@
-import torch
-from torch import nn
-
-
-def down_block(in_filters, out_filters):
-    return nn.Conv2d(in_filters, out_filters, kernel_size=5,
-                     stride=2, padding=2,
-                     ), nn.Sequential(
-        nn.BatchNorm2d(out_filters, track_running_stats=True, eps=1e-3, momentum=0.01),
-        nn.LeakyReLU(0.2)
-    )
-
-
-def up_block(in_filters, out_filters, dropout=False):
-    layers = [
-        nn.ConvTranspose2d(in_filters, out_filters, kernel_size=5,
-                           stride=2, padding=2, output_padding=1
-                           ),
-        nn.ReLU(),
-        nn.BatchNorm2d(out_filters, track_running_stats=True, eps=1e-3, momentum=0.01)
-    ]
-    if dropout:
-        layers.append(nn.Dropout(0.5))
-
-    return nn.Sequential(*layers)
-
-
-class UNet(nn.Module):
-    def __init__(self, in_channels=2):
-        super(UNet, self).__init__()
-        self.down1_conv, self.down1_act = down_block(in_channels, 16)
-        self.down2_conv, self.down2_act = down_block(16, 32)
-        self.down3_conv, self.down3_act = down_block(32, 64)
-        self.down4_conv, self.down4_act = down_block(64, 128)
-        self.down5_conv, self.down5_act = down_block(128, 256)
-        self.down6_conv, self.down6_act = down_block(256, 512)
-
-        self.up1 = up_block(512, 256, dropout=True)
-        self.up2 = up_block(512, 128, dropout=True)
-        self.up3 = up_block(256, 64, dropout=True)
-        self.up4 = up_block(128, 32)
-        self.up5 = up_block(64, 16)
-        self.up6 = up_block(32, 1)
-        self.up7 = nn.Sequential(
-            nn.Conv2d(1, 2, kernel_size=4, dilation=2, padding=3),
-            nn.Sigmoid()
-        )
-
-    def forward(self, x):
-        d1_conv = self.down1_conv(x)
-        d1 = self.down1_act(d1_conv)
-
-        d2_conv = self.down2_conv(d1)
-        d2 = self.down2_act(d2_conv)
-
-        d3_conv = self.down3_conv(d2)
-        d3 = self.down3_act(d3_conv)
-
-        d4_conv = self.down4_conv(d3)
-        d4 = self.down4_act(d4_conv)
-
-        d5_conv = self.down5_conv(d4)
-        d5 = self.down5_act(d5_conv)
-
-        d6_conv = self.down6_conv(d5)
-        d6 = self.down6_act(d6_conv)
-
-        u1 = self.up1(d6)
-        u2 = self.up2(torch.cat([d5_conv, u1], axis=1))
-        u3 = self.up3(torch.cat([d4_conv, u2], axis=1))
-        u4 = self.up4(torch.cat([d3_conv, u3], axis=1))
-        u5 = self.up5(torch.cat([d2_conv, u4], axis=1))
-        u6 = self.up6(torch.cat([d1_conv, u5], axis=1))
-        u7 = self.up7(u6)
-        return u7 * x
-
-
-if __name__ == '__main__':
-    net = UNet(14)
-    print(net(torch.rand(1, 14, 20, 48)).shape)

codes/models/spleeter/util.py

@@ -1,91 +0,0 @@
-import numpy as np
-import tensorflow as tf
-
-from .unet import UNet
-
-
-def tf2pytorch(checkpoint_path, num_instrumments):
-    tf_vars = {}
-    init_vars = tf.train.list_variables(checkpoint_path)
-    # print(init_vars)
-    for name, shape in init_vars:
-        try:
-            # print('Loading TF Weight {} with shape {}'.format(name, shape))
-            data = tf.train.load_variable(checkpoint_path, name)
-            tf_vars[name] = data
-        except Exception as e:
-            print('Load error')
-    conv_idx = 0
-    tconv_idx = 0
-    bn_idx = 0
-    outputs = []
-    for i in range(num_instrumments):
-        output = {}
-        outputs.append(output)
-
-        for j in range(1,7):
-            if conv_idx == 0:
-                conv_suffix = ""
-            else:
-                conv_suffix = "_" + str(conv_idx)
-
-            if bn_idx == 0:
-                bn_suffix = ""
-            else:
-                bn_suffix = "_" + str(bn_idx)
-
-            output['down{}_conv.weight'.format(j)] = np.transpose(
-                tf_vars["conv2d{}/kernel".format(conv_suffix)], (3, 2, 0, 1))
-            # print('conv dtype: ',output['down{}.0.weight'.format(j)].dtype)
-            output['down{}_conv.bias'.format(
-                j)] = tf_vars["conv2d{}/bias".format(conv_suffix)]
-
-            output['down{}_act.0.weight'.format(
-                j)] = tf_vars["batch_normalization{}/gamma".format(bn_suffix)]
-            output['down{}_act.0.bias'.format(
-                j)] = tf_vars["batch_normalization{}/beta".format(bn_suffix)]
-            output['down{}_act.0.running_mean'.format(
-                j)] = tf_vars['batch_normalization{}/moving_mean'.format(bn_suffix)]
-            output['down{}_act.0.running_var'.format(
-                j)] = tf_vars['batch_normalization{}/moving_variance'.format(bn_suffix)]
-
-            conv_idx += 1
-            bn_idx += 1
-
-        # up blocks
-        for j in range(1, 7):
-            if tconv_idx == 0:
-                tconv_suffix = ""
-            else:
-                tconv_suffix = "_" + str(tconv_idx)
-
-            if bn_idx == 0:
-                bn_suffix = ""
-            else:
-                bn_suffix= "_" + str(bn_idx)
-
-            output['up{}.0.weight'.format(j)] = np.transpose(
-                tf_vars["conv2d_transpose{}/kernel".format(tconv_suffix)], (3,2,0, 1))
-            output['up{}.0.bias'.format(
-                j)] = tf_vars["conv2d_transpose{}/bias".format(tconv_suffix)]
-            output['up{}.2.weight'.format(
-                j)] = tf_vars["batch_normalization{}/gamma".format(bn_suffix)]
-            output['up{}.2.bias'.format(
-                j)] = tf_vars["batch_normalization{}/beta".format(bn_suffix)]
-            output['up{}.2.running_mean'.format(
-                j)] = tf_vars['batch_normalization{}/moving_mean'.format(bn_suffix)]
-            output['up{}.2.running_var'.format(
-                j)] = tf_vars['batch_normalization{}/moving_variance'.format(bn_suffix)]
-            tconv_idx += 1
-            bn_idx += 1
-
-        if conv_idx == 0:
-            suffix = ""
-        else:
-            suffix = "_" + str(conv_idx)
-        output['up7.0.weight'] = np.transpose(
-            tf_vars['conv2d{}/kernel'.format(suffix)], (3, 2, 0, 1))
-        output['up7.0.bias'] = tf_vars['conv2d{}/bias'.format(suffix)]
-        conv_idx += 1
-
-    return outputs

codes/scripts/audio/test_audio_segmentor.py

@@ -4,20 +4,17 @@ import random
 import argparse
 
 import audio2numpy
-import torchvision
 from munch import munchify
 
 import utils
 import utils.options as option
 import utils.util as util
 from data.audio.nv_tacotron_dataset import save_mel_buffer_to_file
-from models.tacotron2 import hparams
+from models.audio.tts.tacotron2 import hparams
-from models.tacotron2.layers import TacotronSTFT
+from models.audio.tts.tacotron2 import TacotronSTFT
-from models.tacotron2.text import sequence_to_text
+from models.audio.tts.tacotron2 import sequence_to_text
 from scripts.audio.use_vocoder import Vocoder
 from trainer.ExtensibleTrainer import ExtensibleTrainer
-from data import create_dataset, create_dataloader
-from tqdm import tqdm
 import torch
 import numpy as np
 from scipy.io import wavfile

codes/scripts/audio/test_audio_similarity.py

@@ -1,12 +1,11 @@
 import os
 
 import torch
-import torch.nn as nn
 import torch.nn.functional as F
 
 from data.util import is_wav_file, find_files_of_type
-from models.audio_resnet import resnet34, resnet50
+from models.audio_resnet import resnet50
-from models.tacotron2.taco_utils import load_wav_to_torch
+from models.audio.tts.tacotron2 import load_wav_to_torch
 from scripts.byol.byol_extract_wrapped_model import extract_byol_model_from_state_dict
 
 if __name__ == '__main__':

codes/scripts/audio/test_audio_speech_recognition.py

@@ -3,12 +3,10 @@ import logging
 import random
 import argparse
 
-import torchvision
-
 import utils
 import utils.options as option
 import utils.util as util
-from models.tacotron2.text import sequence_to_text
+from models.audio.tts.tacotron2 import sequence_to_text
 from trainer.ExtensibleTrainer import ExtensibleTrainer
 from data import create_dataset, create_dataloader
 from tqdm import tqdm

codes/scripts/diffusion/diffusion_noise_surfer.py

@@ -2,11 +2,8 @@ import os
 import os.path as osp
 import logging
 import random
-import time
 import argparse
-from collections import OrderedDict
 
-import numpy
 from PIL import Image
 from scipy.io import wavfile
 from torchvision.transforms import ToTensor
@@ -15,10 +12,7 @@ import utils
 import utils.options as option
 import utils.util as util
 from data.audio.unsupervised_audio_dataset import load_audio
-from models.tacotron2.taco_utils import load_wav_to_torch
 from trainer.ExtensibleTrainer import ExtensibleTrainer
-from data import create_dataset, create_dataloader
-from tqdm import tqdm
 import torch
 import numpy as np
 

codes/scripts/hugging_face_hub_upload.py

@@ -1,10 +1,3 @@
-import json
-
-import torch
-
-from models.asr.w2v_wrapper import Wav2VecWrapper
-from models.tacotron2.text import tacotron_symbol_mapping
-
 if __name__ == '__main__':
     """
     Utility script for uploading model weights to the HF hub

codes/trainer/eval/audio_diffusion_fid.py

@@ -2,12 +2,10 @@ import os
 import os.path as osp
 import torch
 import torchaudio
-import torchvision
-from pytorch_fid import fid_score
 from pytorch_fid.fid_score import calculate_frechet_distance
 from torch import distributed
 from tqdm import tqdm
-from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
+from transformers import Wav2Vec2ForCTC
 import torch.nn.functional as F
 import numpy as np
 
@@ -15,7 +13,7 @@ import trainer.eval.evaluator as evaluator
 from data.audio.paired_voice_audio_dataset import load_tsv_aligned_codes
 from data.audio.unsupervised_audio_dataset import load_audio
 from models.clip.mel_text_clip import MelTextCLIP
-from models.tacotron2.text import sequence_to_text, text_to_sequence
+from models.audio.tts.tacotron2 import text_to_sequence
 from scripts.audio.gen.speech_synthesis_utils import load_discrete_vocoder_diffuser, wav_to_mel, load_speech_dvae, \
     convert_mel_to_codes
 from utils.util import ceil_multiple, opt_get
@@ -116,6 +114,27 @@ class AudioDiffusionFid(evaluator.Evaluator):
                                                   'unaligned_input': torch.tensor(text_codes, device=audio.device).unsqueeze(0)})
         return gen, real_resampled, sample_rate
 
+
+    def perform_diffusion_tts9_from_codes(self, audio, codes, text, sample_rate=5500):
+        mel = wav_to_mel(audio)
+        mel_codes = convert_mel_to_codes(self.dvae, mel)
+        text_codes = text_to_sequence(text)
+        real_resampled = torchaudio.functional.resample(audio, 22050, sample_rate).unsqueeze(0)
+
+        output_size = real_resampled.shape[-1]
+        aligned_codes_compression_factor = output_size // mel_codes.shape[-1]
+        padded_size = ceil_multiple(output_size, 2048)
+        padding_added = padded_size - output_size
+        padding_needed_for_codes = padding_added // aligned_codes_compression_factor
+        if padding_needed_for_codes > 0:
+            mel_codes = F.pad(mel_codes, (0, padding_needed_for_codes))
+        output_shape = (1, 1, padded_size)
+        gen = self.diffuser.p_sample_loop(self.model, output_shape,
+                                    model_kwargs={'tokens': mel_codes,
+                                                  'conditioning_input': audio.unsqueeze(0),
+                                                  'unaligned_input': torch.tensor(text_codes, device=audio.device).unsqueeze(0)})
+        return gen, real_resampled, sample_rate
+
     def load_projector(self):
         """
         Builds the CLIP model used to project speech into a latent. This model has fixed parameters and a fixed loading

codes/trainer/eval/eval_wer.py

@@ -4,12 +4,12 @@ from datasets import load_metric
 
 import torch
 from tqdm import tqdm
-from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
+from transformers import Wav2Vec2Processor
 
 import trainer.eval.evaluator as evaluator
 from data import create_dataset, create_dataloader
-from models.asr.w2v_wrapper import only_letters, Wav2VecWrapper
+from models.audio.asr.w2v_wrapper import only_letters, Wav2VecWrapper
-from models.tacotron2.text import sequence_to_text, tacotron_symbols
+from models.audio.tts.tacotron2 import sequence_to_text, tacotron_symbols
 from pyctcdecode import build_ctcdecoder
 
 # Librispeech:

codes/trainer/eval/mel_evaluator.py

@@ -4,7 +4,7 @@ import trainer.eval.evaluator as evaluator
 
 from data import create_dataset
 from data.audio.nv_tacotron_dataset import TextMelCollate
-from models.tacotron2.loss import Tacotron2LossRaw
+from models.audio.tts.tacotron2 import Tacotron2LossRaw
 from torch.utils.data import DataLoader
 from tqdm import tqdm
 

codes/trainer/injectors/audio_injectors.py

@@ -1,7 +1,6 @@
 import random
 
 import torch
-import torch.nn as nn
 import torch.nn.functional as F
 import torchaudio
 
@@ -12,7 +11,7 @@ from utils.util import opt_get, load_model_from_config
 class MelSpectrogramInjector(Injector):
     def __init__(self, opt, env):
         super().__init__(opt, env)
-        from models.tacotron2.layers import TacotronSTFT
+        from models.audio.tts.tacotron2 import TacotronSTFT
         # These are the default tacotron values for the MEL spectrogram.
         filter_length = opt_get(opt, ['filter_length'], 1024)
         hop_length = opt_get(opt, ['hop_length'], 256)

codes/trainer/injectors/spec_augment.py

@@ -83,7 +83,7 @@ class CombineMelInjector(Injector):
         self.text_lengths = opt['text_lengths_key']
         self.output_audio_key = opt['output_audio_key']
         self.output_text_key = opt['output_text_key']
-        from models.tacotron2.text import symbols
+        from models.audio.tts.tacotron2 import symbols
         self.text_separator = len(symbols)+1  # Probably need to allow this to be set by user.
 
     def forward(self, state):

codes/trainer/losses.py

@@ -59,7 +59,7 @@ def create_loss(opt_loss, env):
         from models.switched_conv.mixture_of_experts import SwitchTransformersLoadBalancingLoss
         return SwitchTransformersLoadBalancingLoss(opt_loss, env)
     elif type == 'nv_tacotron2_loss':
-        from models.tacotron2.loss import Tacotron2Loss
+        from models.audio.tts.tacotron2 import Tacotron2Loss
         return Tacotron2Loss(opt_loss, env)
     else:
         raise NotImplementedError