Move everything into the tortoise/ subdirectory

For eventual packaging.
remotes/1710189933836426429/master
James Betker 2022-05-01 16:24:24 +07:00
parent 9c35b73a1f
commit 23a3d5d00b
23 changed files with 26 additions and 267 deletions

@ -1,38 +0,0 @@
import os
import torchaudio
from api import TextToSpeech
from utils.audio import load_audio
if __name__ == '__main__':
fname = 'Y:\\clips\\books2\\subset512-oco.tsv'
stop_after = 128
outpath_base = 'D:\\tmp\\tortoise-tts-eval\\audiobooks'
outpath_real = 'D:\\tmp\\tortoise-tts-eval\\real'
os.makedirs(outpath_real, exist_ok=True)
with open(fname, 'r', encoding='utf-8') as f:
lines = [l.strip().split('\t') for l in f.readlines()]
tts = TextToSpeech()
for k in range(3):
outpath = f'{outpath_base}_{k}'
os.makedirs(outpath, exist_ok=True)
recorder = open(os.path.join(outpath, 'transcript.tsv'), 'w', encoding='utf-8')
for e, line in enumerate(lines):
if e >= stop_after:
break
transcript = line[0]
path = os.path.join(os.path.dirname(fname), line[1])
cond_audio = load_audio(path, 22050)
torchaudio.save(os.path.join(outpath_real, os.path.basename(line[1])), cond_audio, 22050)
sample = tts.tts_with_preset(transcript, [cond_audio, cond_audio], preset='standard')
down = torchaudio.functional.resample(sample, 24000, 22050)
fout_path = os.path.join(outpath, os.path.basename(line[1]))
torchaudio.save(fout_path, down.squeeze(0), 22050)
recorder.write(f'{transcript}\t{fout_path}\n')
recorder.flush()
recorder.close()

@ -1,133 +0,0 @@
import torch
import torch.nn as nn
import torch.nn.functional as F
from torch import einsum
from torch.utils.checkpoint import checkpoint
from models.arch_util import AttentionBlock
from models.xtransformers import ContinuousTransformerWrapper, Encoder
def exists(val):
return val is not None
def masked_mean(t, mask):
t = t.masked_fill(~mask, 0.)
return t.sum(dim = 1) / mask.sum(dim = 1)
class CollapsingTransformer(nn.Module):
def __init__(self, model_dim, output_dims, heads, dropout, depth, mask_percentage=0, **encoder_kwargs):
super().__init__()
self.transformer = ContinuousTransformerWrapper(
max_seq_len=-1,
use_pos_emb=False,
attn_layers=Encoder(
dim=model_dim,
depth=depth,
heads=heads,
ff_dropout=dropout,
ff_mult=1,
attn_dropout=dropout,
use_rmsnorm=True,
ff_glu=True,
rotary_pos_emb=True,
**encoder_kwargs,
))
self.pre_combiner = nn.Sequential(nn.Conv1d(model_dim, output_dims, 1),
AttentionBlock(output_dims, num_heads=heads, do_checkpoint=False),
nn.Conv1d(output_dims, output_dims, 1))
self.mask_percentage = mask_percentage
def forward(self, x, **transformer_kwargs):
h = self.transformer(x, **transformer_kwargs)
h = h.permute(0,2,1)
h = checkpoint(self.pre_combiner, h).permute(0,2,1)
if self.training:
mask = torch.rand_like(h.float()) > self.mask_percentage
else:
mask = torch.ones_like(h.float()).bool()
return masked_mean(h, mask)
class ConvFormatEmbedding(nn.Module):
def __init__(self, *args, **kwargs):
super().__init__()
self.emb = nn.Embedding(*args, **kwargs)
def forward(self, x):
y = self.emb(x)
return y.permute(0,2,1)
class CVVP(nn.Module):
def __init__(
self,
model_dim=512,
transformer_heads=8,
dropout=.1,
conditioning_enc_depth=8,
cond_mask_percentage=0,
mel_channels=80,
mel_codes=None,
speech_enc_depth=8,
speech_mask_percentage=0,
latent_multiplier=1,
):
super().__init__()
latent_dim = latent_multiplier*model_dim
self.temperature = nn.Parameter(torch.tensor(1.))
self.cond_emb = nn.Sequential(nn.Conv1d(mel_channels, model_dim//2, kernel_size=5, stride=2, padding=2),
nn.Conv1d(model_dim//2, model_dim, kernel_size=3, stride=2, padding=1))
self.conditioning_transformer = CollapsingTransformer(model_dim, model_dim, transformer_heads, dropout, conditioning_enc_depth, cond_mask_percentage)
self.to_conditioning_latent = nn.Linear(latent_dim, latent_dim, bias=False)
if mel_codes is None:
self.speech_emb = nn.Conv1d(mel_channels, model_dim, kernel_size=5, padding=2)
else:
self.speech_emb = ConvFormatEmbedding(mel_codes, model_dim)
self.speech_transformer = CollapsingTransformer(model_dim, latent_dim, transformer_heads, dropout, speech_enc_depth, speech_mask_percentage)
self.to_speech_latent = nn.Linear(latent_dim, latent_dim, bias=False)
def get_grad_norm_parameter_groups(self):
return {
'conditioning': list(self.conditioning_transformer.parameters()),
'speech': list(self.speech_transformer.parameters()),
}
def forward(
self,
mel_cond,
mel_input,
return_loss=False
):
cond_emb = self.cond_emb(mel_cond).permute(0,2,1)
enc_cond = self.conditioning_transformer(cond_emb)
cond_latents = self.to_conditioning_latent(enc_cond)
speech_emb = self.speech_emb(mel_input).permute(0,2,1)
enc_speech = self.speech_transformer(speech_emb)
speech_latents = self.to_speech_latent(enc_speech)
cond_latents, speech_latents = map(lambda t: F.normalize(t, p=2, dim=-1), (cond_latents, speech_latents))
temp = self.temperature.exp()
if not return_loss:
sim = einsum('n d, n d -> n', cond_latents, speech_latents) * temp
return sim
sim = einsum('i d, j d -> i j', cond_latents, speech_latents) * temp
labels = torch.arange(cond_latents.shape[0], device=mel_input.device)
loss = (F.cross_entropy(sim, labels) + F.cross_entropy(sim.t(), labels)) / 2
return loss
if __name__ == '__main__':
clvp = CVVP()
clvp(torch.randn(2,80,100),
torch.randn(2,80,95),
return_loss=True)

@ -1,65 +0,0 @@
import os
from random import shuffle
import torchaudio
from api import TextToSpeech
from utils.audio import load_audio
def permutations(args):
res = []
k = next(iter(args.keys()))
vals = args[k]
del args[k]
if not args:
return [{k: v} for v in vals]
lower = permutations(args)
for v in vals:
for l in lower:
lc = l.copy()
lc[k] = v
res.append(lc)
return res
if __name__ == '__main__':
fname = 'Y:\\clips\\books2\\subset512-oco.tsv'
stop_after = 512
outpath_base = 'D:\\tmp\\tortoise-tts-eval\\sweep-2'
outpath_real = 'D:\\tmp\\tortoise-tts-eval\\real'
arg_ranges = {
'top_p': [.8,1],
'temperature': [.8,.9,1],
'diffusion_temperature': [.8,1],
'cond_free_k': [1,2,5,10],
}
cfgs = permutations(arg_ranges)
shuffle(cfgs)
for cfg in cfgs:
cfg_desc = '_'.join([f'{k}-{v}' for k,v in cfg.items()])
outpath = os.path.join(outpath_base, f'{cfg_desc}')
os.makedirs(outpath, exist_ok=True)
os.makedirs(outpath_real, exist_ok=True)
with open(fname, 'r', encoding='utf-8') as f:
lines = [l.strip().split('\t') for l in f.readlines()]
recorder = open(os.path.join(outpath, 'transcript.tsv'), 'w', encoding='utf-8')
tts = TextToSpeech()
for e, line in enumerate(lines):
if e >= stop_after:
break
transcript = line[0]
path = os.path.join(os.path.dirname(fname), line[1])
cond_audio = load_audio(path, 22050)
torchaudio.save(os.path.join(outpath_real, os.path.basename(line[1])), cond_audio, 22050)
sample = tts.tts(transcript, [cond_audio, cond_audio], num_autoregressive_samples=32, repetition_penalty=2.0,
k=1, diffusion_iterations=32, length_penalty=1.0, **cfg)
down = torchaudio.functional.resample(sample, 24000, 22050)
fout_path = os.path.join(outpath, os.path.basename(line[1]))
torchaudio.save(fout_path, down.squeeze(0), 22050)
recorder.write(f'{transcript}\t{fout_path}\n')
recorder.flush()
recorder.close()

@ -1,4 +1,3 @@
import argparse
import os
import random
from urllib import request
@ -8,19 +7,18 @@ import torch.nn.functional as F
import progressbar
import torchaudio
from models.classifier import AudioMiniEncoderWithClassifierHead
from models.cvvp import CVVP
from models.diffusion_decoder import DiffusionTts
from models.autoregressive import UnifiedVoice
from tortoise.models.classifier import AudioMiniEncoderWithClassifierHead
from tortoise.models.cvvp import CVVP
from tortoise.models.diffusion_decoder import DiffusionTts
from tortoise.models.autoregressive import UnifiedVoice
from tqdm import tqdm
from models.arch_util import TorchMelSpectrogram
from models.clvp import CLVP
from models.vocoder import UnivNetGenerator
from utils.audio import load_audio, wav_to_univnet_mel, denormalize_tacotron_mel
from utils.diffusion import SpacedDiffusion, space_timesteps, get_named_beta_schedule
from utils.tokenizer import VoiceBpeTokenizer, lev_distance
from tortoise.models.arch_util import TorchMelSpectrogram
from tortoise.models.clvp import CLVP
from tortoise.models.vocoder import UnivNetGenerator
from tortoise.utils.audio import wav_to_univnet_mel, denormalize_tacotron_mel
from tortoise.utils.diffusion import SpacedDiffusion, space_timesteps, get_named_beta_schedule
from tortoise.utils.tokenizer import VoiceBpeTokenizer
pbar = None

@ -4,7 +4,7 @@ import os
import torchaudio
from api import TextToSpeech
from utils.audio import load_audio, get_voices
from tortoise.utils.audio import load_audio, get_voices
if __name__ == '__main__':
parser = argparse.ArgumentParser()

@ -1,7 +1,7 @@
import argparse
from api import classify_audio_clip
from utils.audio import load_audio
from tortoise.utils.audio import load_audio
if __name__ == '__main__':
parser = argparse.ArgumentParser()

@ -5,7 +5,7 @@ import torch
import torch.nn as nn
import torch.nn.functional as F
import torchaudio
from models.xtransformers import ContinuousTransformerWrapper, RelativePositionBias
from tortoise.models.xtransformers import ContinuousTransformerWrapper, RelativePositionBias
def zero_module(module):

@ -6,8 +6,8 @@ import torch.nn.functional as F
from transformers import GPT2Config, GPT2PreTrainedModel, LogitsProcessorList
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 utils.typical_sampling import TypicalLogitsWarper
from tortoise.models.arch_util import AttentionBlock
from tortoise.utils.typical_sampling import TypicalLogitsWarper
def null_position_embeddings(range, dim):

@ -1,9 +1,8 @@
import torch
import torch.nn as nn
import torch.nn.functional as F
from torch.utils.checkpoint import checkpoint
from models.arch_util import Upsample, Downsample, normalization, zero_module, AttentionBlock
from tortoise.models.arch_util import Upsample, Downsample, normalization, zero_module, AttentionBlock
class ResBlock(nn.Module):

@ -3,9 +3,9 @@ import torch.nn as nn
import torch.nn.functional as F
from torch import einsum
from models.arch_util import CheckpointedXTransformerEncoder
from models.transformer import Transformer
from models.xtransformers import Encoder
from tortoise.models.arch_util import CheckpointedXTransformerEncoder
from tortoise.models.transformer import Transformer
from tortoise.models.xtransformers import Encoder
def exists(val):

@ -7,7 +7,7 @@ import torch.nn as nn
import torch.nn.functional as F
from torch import autocast
from models.arch_util import normalization, AttentionBlock
from tortoise.models.arch_util import normalization, AttentionBlock
def is_latent(t):

@ -2,12 +2,10 @@ import argparse
import os
import torch
import torch.nn.functional as F
import torchaudio
from api import TextToSpeech, format_conditioning
from utils.audio import load_audio, get_voices
from utils.tokenizer import VoiceBpeTokenizer
from api import TextToSpeech
from tortoise.utils.audio import load_audio, get_voices
def split_and_recombine_text(texts, desired_length=200, max_len=300):

@ -4,7 +4,7 @@ import os
if __name__ == '__main__':
result = "<html><head><title>These words were never spoken.</title></head><body><h1>Handpicked results</h1>"
for fv in os.listdir('results/favorites'):
for fv in os.listdir('../results/favorites'):
url = f'https://github.com/neonbjb/tortoise-tts/raw/main/results/favorites/{fv}'
result = result + f'<audio controls="" style="width: 600px;"><source src="{url}" type="audio/mp3"></audio><br>\n'
@ -30,7 +30,7 @@ if __name__ == '__main__':
line = line + f'<td><audio controls="" style="width: 150px;"><source src="{url}" type="audio/mp3"></audio></td>'
line = line + "</tr>"
lines.append(line)
for txt in os.listdir('results/various/'):
for txt in os.listdir('../results/various/'):
if 'desktop' in txt:
continue
line = f'<tr><td>{txt}</td>'
@ -42,7 +42,7 @@ if __name__ == '__main__':
result = result + '\n'.join(lines) + "</table>"
result = result + "<h1>Longform result for all voices:</h1>"
for lf in os.listdir('results/riding_hood'):
for lf in os.listdir('../results/riding_hood'):
url = f'https://github.com/neonbjb/tortoise-tts/raw/main/results/riding_hood/{lf}'
result = result + f'<audio controls="" style="width: 600px;"><source src="{url}" type="audio/mp3"></audio><br>\n'

@ -6,7 +6,7 @@ import torchaudio
import numpy as np
from scipy.io.wavfile import read
from utils.stft import STFT
from tortoise.utils.stft import STFT
def load_wav_to_torch(full_path):