tortoise-tts/tortoise/utils/audio.py
James Betker 01b783fc02 Add support for extracting and feeding conditioning latents directly into the model
- Adds a new script and API endpoints for doing this
- Reworks autoregressive and diffusion models so that the conditioning is computed separately (which will actually provide a mild performance boost)
- Updates README

This is untested. Need to do the following manual tests (and someday write unit tests for this behemoth before
it becomes a problem..)
1) Does get_conditioning_latents.py work?
2) Can I feed those latents back into the model by creating a new voice?
3) Can I still mix and match voices (both with conditioning latents and normal voices) with read.py?
2022-05-01 17:25:18 -06:00

174 lines
5.4 KiB
Python

import os
from glob import glob
import torch
import torchaudio
import numpy as np
from scipy.io.wavfile import read
from tortoise.utils.stft import STFT
def load_wav_to_torch(full_path):
sampling_rate, data = read(full_path)
if data.dtype == np.int32:
norm_fix = 2 ** 31
elif data.dtype == np.int16:
norm_fix = 2 ** 15
elif data.dtype == np.float16 or data.dtype == np.float32:
norm_fix = 1.
else:
raise NotImplemented(f"Provided data dtype not supported: {data.dtype}")
return (torch.FloatTensor(data.astype(np.float32)) / norm_fix, sampling_rate)
def load_audio(audiopath, sampling_rate):
if audiopath[-4:] == '.wav':
audio, lsr = load_wav_to_torch(audiopath)
elif audiopath[-4:] == '.mp3':
# https://github.com/neonbjb/pyfastmp3decoder - Definitely worth it.
from pyfastmp3decoder.mp3decoder import load_mp3
audio, lsr = load_mp3(audiopath, sampling_rate)
audio = torch.FloatTensor(audio)
# Remove any channel data.
if len(audio.shape) > 1:
if audio.shape[0] < 5:
audio = audio[0]
else:
assert audio.shape[1] < 5
audio = audio[:, 0]
if lsr != sampling_rate:
audio = torchaudio.functional.resample(audio, lsr, sampling_rate)
# Check some assumptions about audio range. This should be automatically fixed in load_wav_to_torch, but might not be in some edge cases, where we should squawk.
# '2' is arbitrarily chosen since it seems like audio will often "overdrive" the [-1,1] bounds.
if torch.any(audio > 2) or not torch.any(audio < 0):
print(f"Error with {audiopath}. Max={audio.max()} min={audio.min()}")
audio.clip_(-1, 1)
return audio.unsqueeze(0)
TACOTRON_MEL_MAX = 2.3143386840820312
TACOTRON_MEL_MIN = -11.512925148010254
def denormalize_tacotron_mel(norm_mel):
return ((norm_mel+1)/2)*(TACOTRON_MEL_MAX-TACOTRON_MEL_MIN)+TACOTRON_MEL_MIN
def normalize_tacotron_mel(mel):
return 2 * ((mel - TACOTRON_MEL_MIN) / (TACOTRON_MEL_MAX - TACOTRON_MEL_MIN)) - 1
def dynamic_range_compression(x, C=1, clip_val=1e-5):
"""
PARAMS
------
C: compression factor
"""
return torch.log(torch.clamp(x, min=clip_val) * C)
def dynamic_range_decompression(x, C=1):
"""
PARAMS
------
C: compression factor used to compress
"""
return torch.exp(x) / C
def get_voices():
subs = os.listdir('voices')
voices = {}
for sub in subs:
subj = os.path.join('voices', sub)
if os.path.isdir(subj):
voices[sub] = list(glob(f'{subj}/*.wav')) + list(glob(f'{subj}/*.mp3'))
return voices
def load_voice(voice):
voices = get_voices()
paths = voices[voice]
if len(paths) == 1 and paths[0].endswith('.pth'):
return None, torch.load(paths[0])
else:
conds = []
for cond_path in paths:
c = load_audio(cond_path, 22050)
conds.append(c)
return conds, None
def load_voices(voices):
latents = []
clips = []
for voice in voices:
latent, clip = load_voice(voice)
if latent is None:
assert len(latents) == 0, "Can only combine raw audio voices or latent voices, not both. Do it yourself if you want this."
clips.extend(clip)
elif voice is None:
assert len(voices) == 0, "Can only combine raw audio voices or latent voices, not both. Do it yourself if you want this."
latents.append(latent)
if len(latents) == 0:
return clips
else:
latents = torch.stack(latents, dim=0)
return latents.mean(dim=0)
class TacotronSTFT(torch.nn.Module):
def __init__(self, filter_length=1024, hop_length=256, win_length=1024,
n_mel_channels=80, sampling_rate=22050, mel_fmin=0.0,
mel_fmax=8000.0):
super(TacotronSTFT, self).__init__()
self.n_mel_channels = n_mel_channels
self.sampling_rate = sampling_rate
self.stft_fn = STFT(filter_length, hop_length, win_length)
from librosa.filters import mel as librosa_mel_fn
mel_basis = librosa_mel_fn(
sampling_rate, filter_length, n_mel_channels, mel_fmin, mel_fmax)
mel_basis = torch.from_numpy(mel_basis).float()
self.register_buffer('mel_basis', mel_basis)
def spectral_normalize(self, magnitudes):
output = dynamic_range_compression(magnitudes)
return output
def spectral_de_normalize(self, magnitudes):
output = dynamic_range_decompression(magnitudes)
return output
def mel_spectrogram(self, y):
"""Computes mel-spectrograms from a batch of waves
PARAMS
------
y: Variable(torch.FloatTensor) with shape (B, T) in range [-1, 1]
RETURNS
-------
mel_output: torch.FloatTensor of shape (B, n_mel_channels, T)
"""
assert(torch.min(y.data) >= -10)
assert(torch.max(y.data) <= 10)
y = torch.clip(y, min=-1, max=1)
magnitudes, phases = self.stft_fn.transform(y)
magnitudes = magnitudes.data
mel_output = torch.matmul(self.mel_basis, magnitudes)
mel_output = self.spectral_normalize(mel_output)
return mel_output
def wav_to_univnet_mel(wav, do_normalization=False):
stft = TacotronSTFT(1024, 256, 1024, 100, 24000, 0, 12000)
stft = stft.cuda()
mel = stft.mel_spectrogram(wav)
if do_normalization:
mel = normalize_tacotron_mel(mel)
return mel