forked from mrq/tortoise-tts
01b783fc02
- 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?
174 lines
5.4 KiB
Python
174 lines
5.4 KiB
Python
import os
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from glob import glob
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import torch
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import torchaudio
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import numpy as np
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from scipy.io.wavfile import read
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from tortoise.utils.stft import STFT
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def load_wav_to_torch(full_path):
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sampling_rate, data = read(full_path)
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if data.dtype == np.int32:
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norm_fix = 2 ** 31
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elif data.dtype == np.int16:
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norm_fix = 2 ** 15
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elif data.dtype == np.float16 or data.dtype == np.float32:
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norm_fix = 1.
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else:
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raise NotImplemented(f"Provided data dtype not supported: {data.dtype}")
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return (torch.FloatTensor(data.astype(np.float32)) / norm_fix, sampling_rate)
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def load_audio(audiopath, sampling_rate):
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if audiopath[-4:] == '.wav':
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audio, lsr = load_wav_to_torch(audiopath)
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elif audiopath[-4:] == '.mp3':
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# https://github.com/neonbjb/pyfastmp3decoder - Definitely worth it.
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from pyfastmp3decoder.mp3decoder import load_mp3
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audio, lsr = load_mp3(audiopath, sampling_rate)
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audio = torch.FloatTensor(audio)
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# Remove any channel data.
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if len(audio.shape) > 1:
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if audio.shape[0] < 5:
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audio = audio[0]
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else:
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assert audio.shape[1] < 5
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audio = audio[:, 0]
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if lsr != sampling_rate:
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audio = torchaudio.functional.resample(audio, lsr, sampling_rate)
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# 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.
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# '2' is arbitrarily chosen since it seems like audio will often "overdrive" the [-1,1] bounds.
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if torch.any(audio > 2) or not torch.any(audio < 0):
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print(f"Error with {audiopath}. Max={audio.max()} min={audio.min()}")
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audio.clip_(-1, 1)
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return audio.unsqueeze(0)
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TACOTRON_MEL_MAX = 2.3143386840820312
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TACOTRON_MEL_MIN = -11.512925148010254
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def denormalize_tacotron_mel(norm_mel):
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return ((norm_mel+1)/2)*(TACOTRON_MEL_MAX-TACOTRON_MEL_MIN)+TACOTRON_MEL_MIN
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def normalize_tacotron_mel(mel):
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return 2 * ((mel - TACOTRON_MEL_MIN) / (TACOTRON_MEL_MAX - TACOTRON_MEL_MIN)) - 1
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def dynamic_range_compression(x, C=1, clip_val=1e-5):
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"""
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PARAMS
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------
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C: compression factor
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"""
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return torch.log(torch.clamp(x, min=clip_val) * C)
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def dynamic_range_decompression(x, C=1):
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"""
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PARAMS
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------
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C: compression factor used to compress
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"""
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return torch.exp(x) / C
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def get_voices():
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subs = os.listdir('voices')
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voices = {}
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for sub in subs:
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subj = os.path.join('voices', sub)
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if os.path.isdir(subj):
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voices[sub] = list(glob(f'{subj}/*.wav')) + list(glob(f'{subj}/*.mp3'))
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return voices
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def load_voice(voice):
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voices = get_voices()
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paths = voices[voice]
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if len(paths) == 1 and paths[0].endswith('.pth'):
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return None, torch.load(paths[0])
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else:
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conds = []
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for cond_path in paths:
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c = load_audio(cond_path, 22050)
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conds.append(c)
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return conds, None
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def load_voices(voices):
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latents = []
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clips = []
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for voice in voices:
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latent, clip = load_voice(voice)
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if latent is None:
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assert len(latents) == 0, "Can only combine raw audio voices or latent voices, not both. Do it yourself if you want this."
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clips.extend(clip)
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elif voice is None:
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assert len(voices) == 0, "Can only combine raw audio voices or latent voices, not both. Do it yourself if you want this."
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latents.append(latent)
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if len(latents) == 0:
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return clips
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else:
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latents = torch.stack(latents, dim=0)
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return latents.mean(dim=0)
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class TacotronSTFT(torch.nn.Module):
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def __init__(self, filter_length=1024, hop_length=256, win_length=1024,
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n_mel_channels=80, sampling_rate=22050, mel_fmin=0.0,
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mel_fmax=8000.0):
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super(TacotronSTFT, self).__init__()
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self.n_mel_channels = n_mel_channels
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self.sampling_rate = sampling_rate
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self.stft_fn = STFT(filter_length, hop_length, win_length)
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from librosa.filters import mel as librosa_mel_fn
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mel_basis = librosa_mel_fn(
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sampling_rate, filter_length, n_mel_channels, mel_fmin, mel_fmax)
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mel_basis = torch.from_numpy(mel_basis).float()
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self.register_buffer('mel_basis', mel_basis)
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def spectral_normalize(self, magnitudes):
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output = dynamic_range_compression(magnitudes)
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return output
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def spectral_de_normalize(self, magnitudes):
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output = dynamic_range_decompression(magnitudes)
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return output
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def mel_spectrogram(self, y):
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"""Computes mel-spectrograms from a batch of waves
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PARAMS
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------
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y: Variable(torch.FloatTensor) with shape (B, T) in range [-1, 1]
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RETURNS
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-------
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mel_output: torch.FloatTensor of shape (B, n_mel_channels, T)
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"""
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assert(torch.min(y.data) >= -10)
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assert(torch.max(y.data) <= 10)
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y = torch.clip(y, min=-1, max=1)
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magnitudes, phases = self.stft_fn.transform(y)
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magnitudes = magnitudes.data
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mel_output = torch.matmul(self.mel_basis, magnitudes)
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mel_output = self.spectral_normalize(mel_output)
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return mel_output
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def wav_to_univnet_mel(wav, do_normalization=False):
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stft = TacotronSTFT(1024, 256, 1024, 100, 24000, 0, 12000)
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stft = stft.cuda()
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mel = stft.mel_spectrogram(wav)
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if do_normalization:
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mel = normalize_tacotron_mel(mel)
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return mel |