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forked from mrq/tortoise-tts

Add files required for hifigan, including autoregressive.py modification

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Jarod Mica 2023-11-26 21:51:28 -08:00
parent 95f679f4ba
commit 156bb5e7da
4 changed files with 2208 additions and 65 deletions

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import os
import random
import uuid
from time import time
from urllib import request
import torch
import torch.nn.functional as F
import progressbar
import torchaudio
import numpy as np
from tortoise.models.classifier import AudioMiniEncoderWithClassifierHead
from tortoise.models.diffusion_decoder import DiffusionTts
from tortoise.models.autoregressive import UnifiedVoice
from tqdm import tqdm
from tortoise.models.arch_util import TorchMelSpectrogram
from tortoise.models.clvp import CLVP
from tortoise.models.cvvp import CVVP
from tortoise.models.hifigan_decoder import HifiganGenerator
from tortoise.models.random_latent_generator import RandomLatentConverter
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
from tortoise.utils.wav2vec_alignment import Wav2VecAlignment
from contextlib import contextmanager
# from tortoise.models.stream_generator import init_stream_support
from huggingface_hub import hf_hub_download
from tortoise.utils.device import get_device, get_device_name, get_device_batch_size, print_stats, do_gc
pbar = None
# init_stream_support()
STOP_SIGNAL = False
DEFAULT_MODELS_DIR = os.path.join(os.path.expanduser('~'), '.cache', 'tortoise', 'models')
MODELS_DIR = os.environ.get('TORTOISE_MODELS_DIR', DEFAULT_MODELS_DIR)
MODELS = {
'autoregressive.pth': 'https://huggingface.co/Manmay/tortoise-tts/resolve/main/autoregressive.pth',
'classifier.pth': 'https://huggingface.co/Manmay/tortoise-tts/resolve/main/classifier.pth',
'rlg_auto.pth': 'https://huggingface.co/Manmay/tortoise-tts/resolve/main/rlg_auto.pth',
'hifidecoder.pth': 'https://huggingface.co/Manmay/tortoise-tts/resolve/main/hifidecoder.pth',
}
def download_models(specific_models=None):
"""
Call to download all the models that Tortoise uses.
"""
os.makedirs(MODELS_DIR, exist_ok=True)
def show_progress(block_num, block_size, total_size):
global pbar
if pbar is None:
pbar = progressbar.ProgressBar(maxval=total_size)
pbar.start()
downloaded = block_num * block_size
if downloaded < total_size:
pbar.update(downloaded)
else:
pbar.finish()
pbar = None
for model_name, url in MODELS.items():
if specific_models is not None and model_name not in specific_models:
continue
model_path = os.path.join(MODELS_DIR, model_name)
if os.path.exists(model_path):
continue
print(f'Downloading {model_name} from {url}...')
request.urlretrieve(url, model_path, show_progress)
print('Done.')
def get_model_path(model_name, models_dir=MODELS_DIR):
"""
Get path to given model, download it if it doesn't exist.
"""
if model_name not in MODELS:
raise ValueError(f'Model {model_name} not found in available models.')
model_path = os.path.join(models_dir, model_name)
if not os.path.exists(model_path) and models_dir == MODELS_DIR:
download_models([model_name])
# Add the logic to download models if not available
# model_path = hf_hub_download(repo_id="Manmay/tortoise-tts", filename=model_name, cache_dir=models_dir)
return model_path
def check_for_kill_signal():
global STOP_SIGNAL
if STOP_SIGNAL:
STOP_SIGNAL = False
raise Exception("Kill signal detected")
def pad_or_truncate(t, length):
"""
Utility function for forcing <t> to have the specified sequence length, whether by clipping it or padding it with 0s.
"""
if t.shape[-1] == length:
return t
elif t.shape[-1] < length:
return F.pad(t, (0, length-t.shape[-1]))
else:
return t[..., :length]
def load_discrete_vocoder_diffuser(trained_diffusion_steps=4000, desired_diffusion_steps=200, cond_free=True, cond_free_k=1):
"""
Helper function to load a GaussianDiffusion instance configured for use as a vocoder.
"""
return SpacedDiffusion(use_timesteps=space_timesteps(trained_diffusion_steps, [desired_diffusion_steps]), model_mean_type='epsilon',
model_var_type='learned_range', loss_type='mse', betas=get_named_beta_schedule('linear', trained_diffusion_steps),
conditioning_free=cond_free, conditioning_free_k=cond_free_k)
def format_conditioning(clip, cond_length=132300, device="cuda" if not torch.backends.mps.is_available() else 'mps'):
"""
Converts the given conditioning signal to a MEL spectrogram and clips it as expected by the models.
"""
gap = clip.shape[-1] - cond_length
if gap < 0:
clip = F.pad(clip, pad=(0, abs(gap)))
elif gap > 0:
rand_start = random.randint(0, gap)
clip = clip[:, rand_start:rand_start + cond_length]
mel_clip = TorchMelSpectrogram()(clip.unsqueeze(0)).squeeze(0)
return mel_clip.unsqueeze(0).to(device)
def fix_autoregressive_output(codes, stop_token, complain=True):
"""
This function performs some padding on coded audio that fixes a mismatch issue between what the diffusion model was
trained on and what the autoregressive code generator creates (which has no padding or end).
This is highly specific to the DVAE being used, so this particular coding will not necessarily work if used with
a different DVAE. This can be inferred by feeding a audio clip padded with lots of zeros on the end through the DVAE
and copying out the last few codes.
Failing to do this padding will produce speech with a harsh end that sounds like "BLAH" or similar.
"""
# Strip off the autoregressive stop token and add padding.
stop_token_indices = (codes == stop_token).nonzero()
if len(stop_token_indices) == 0:
if complain:
print("No stop tokens found in one of the generated voice clips. This typically means the spoken audio is "
"too long. In some cases, the output will still be good, though. Listen to it and if it is missing words, "
"try breaking up your input text.")
return codes
else:
codes[stop_token_indices] = 83
stm = stop_token_indices.min().item()
codes[stm:] = 83
if stm - 3 < codes.shape[0]:
codes[-3] = 45
codes[-2] = 45
codes[-1] = 248
return codes
def do_spectrogram_diffusion(diffusion_model, diffuser, latents, conditioning_latents, temperature=1, verbose=True):
"""
Uses the specified diffusion model to convert discrete codes into a spectrogram.
"""
with torch.no_grad():
output_seq_len = latents.shape[1] * 4 * 24000 // 22050 # This diffusion model converts from 22kHz spectrogram codes to a 24kHz spectrogram signal.
output_shape = (latents.shape[0], 100, output_seq_len)
precomputed_embeddings = diffusion_model.timestep_independent(latents, conditioning_latents, output_seq_len, False)
noise = torch.randn(output_shape, device=latents.device) * temperature
mel = diffuser.p_sample_loop(diffusion_model, output_shape, noise=noise,
model_kwargs={'precomputed_aligned_embeddings': precomputed_embeddings},
progress=verbose)
return denormalize_tacotron_mel(mel)[:,:,:output_seq_len]
def classify_audio_clip(clip):
"""
Returns whether or not Tortoises' classifier thinks the given clip came from Tortoise.
:param clip: torch tensor containing audio waveform data (get it from load_audio)
:return: True if the clip was classified as coming from Tortoise and false if it was classified as real.
"""
classifier = AudioMiniEncoderWithClassifierHead(2, spec_dim=1, embedding_dim=512, depth=5, downsample_factor=4,
resnet_blocks=2, attn_blocks=4, num_attn_heads=4, base_channels=32,
dropout=0, kernel_size=5, distribute_zero_label=False)
classifier.load_state_dict(torch.load(get_model_path('classifier.pth'), map_location=torch.device('cpu')))
clip = clip.cpu().unsqueeze(0)
results = F.softmax(classifier(clip), dim=-1)
return results[0][0]
def pick_best_batch_size_for_gpu():
"""
Tries to pick a batch size that will fit in your GPU. These sizes aren't guaranteed to work, but they should give
you a good shot.
"""
if torch.cuda.is_available():
_, available = torch.cuda.mem_get_info()
availableGb = available / (1024 ** 3)
if availableGb > 14:
return 16
elif availableGb > 10:
return 8
elif availableGb > 7:
return 4
if torch.backends.mps.is_available():
import psutil
available = psutil.virtual_memory().total
availableGb = available / (1024 ** 3)
if availableGb > 14:
return 16
elif availableGb > 10:
return 8
elif availableGb > 7:
return 4
return 1
# Taken from MRQ's api
@torch.inference_mode()
def format_conditioning(clip, cond_length=132300, device='cuda', sampling_rate=22050):
"""
Converts the given conditioning signal to a MEL spectrogram and clips it as expected by the models.
"""
gap = clip.shape[-1] - cond_length
if gap < 0:
clip = F.pad(clip, pad=(0, abs(gap)))
elif gap > 0:
rand_start = random.randint(0, gap)
clip = clip[:, rand_start:rand_start + cond_length]
mel_clip = TorchMelSpectrogram(sampling_rate=sampling_rate)(clip.unsqueeze(0)).squeeze(0)
mel_clip = mel_clip.unsqueeze(0)
return migrate_to_device(mel_clip, device)
# Taken from MRQ's api
def hash_file(path, algo="md5", buffer_size=0):
import hashlib
hash = None
if algo == "md5":
hash = hashlib.md5()
elif algo == "sha1":
hash = hashlib.sha1()
else:
raise Exception(f'Unknown hash algorithm specified: {algo}')
if not os.path.exists(path):
raise Exception(f'Path not found: {path}')
with open(path, 'rb') as f:
if buffer_size > 0:
while True:
data = f.read(buffer_size)
if not data:
break
hash.update(data)
else:
hash.update(f.read())
return "{0}".format(hash.hexdigest())
# Taken from MRQ's api
def migrate_to_device( t, device ):
if t is None:
return t
if not hasattr(t, 'device'):
t.device = device
t.manually_track_device = True
elif t.device == device:
return t
if hasattr(t, 'manually_track_device') and t.manually_track_device:
t.device = device
t = t.to(device)
do_gc()
return t
class TextToSpeech:
"""
Main entry point into Tortoise.
"""
def __init__(self, autoregressive_batch_size=None, models_dir=MODELS_DIR,
enable_redaction=True, kv_cache=False, use_deepspeed=False, half=False, device=None,
tokenizer_vocab_file=None, tokenizer_basic=False,
autoregressive_model_path=None, tokenizer_json=None,
minor_optimizations=True,
input_sample_rate=22050, output_sample_rate=24000,
):
"""
Constructor
:param autoregressive_batch_size: Specifies how many samples to generate per batch. Lower this if you are seeing
GPU OOM errors. Larger numbers generates slightly faster.
:param models_dir: Where model weights are stored. This should only be specified if you are providing your own
models, otherwise use the defaults.
:param enable_redaction: When true, text enclosed in brackets are automatically redacted from the spoken output
(but are still rendered by the model). This can be used for prompt engineering.
Default is true.
:param device: Device to use when running the model. If omitted, the device will be automatically chosen.
"""
self.use_deepspeed = use_deepspeed # Store deepspeed
self.use_kv_cache = kv_cache # Store KV cache
self.preloaded_tensors = minor_optimizations
self.input_sample_rate = input_sample_rate
self.output_sample_rate = output_sample_rate
self.models_dir = models_dir
self.autoregressive_batch_size = pick_best_batch_size_for_gpu() if autoregressive_batch_size is None else autoregressive_batch_size
self.enable_redaction = enable_redaction
self.device = torch.device('cuda' if torch.cuda.is_available() else'cpu')
if torch.backends.mps.is_available():
self.device = torch.device('mps')
if self.enable_redaction:
self.aligner = Wav2VecAlignment()
self.load_tokenizer_json(tokenizer_json)
self.half = half
if os.path.exists(f'{models_dir}/autoregressive.ptt'):
# Assume this is a traced directory.
self.autoregressive = torch.jit.load(f'{models_dir}/autoregressive.ptt')
else:
if not autoregressive_model_path or not os.path.exists(autoregressive_model_path):
autoregressive_model_path = get_model_path('autoregressive.pth', models_dir)
self.load_autoregressive_model(autoregressive_model_path)
# self.autoregressive = UnifiedVoice(max_mel_tokens=604, max_text_tokens=402, max_conditioning_inputs=2, layers=30,
# model_dim=1024,
# heads=16, number_text_tokens=255, start_text_token=255, checkpointing=False,
# train_solo_embeddings=False).to(self.device).eval()
# self.autoregressive.load_state_dict(torch.load(autoregressive_model_path, weights_only=True), strict=False)
# self.autoregressive.post_init_gpt2_config(use_deepspeed=use_deepspeed, kv_cache=kv_cache, half=self.half)
# self.autoregressive = migrate_to_device(self.autoregressive, self.device)
# print(f"Loaded autoregressive model")
self.hifi_decoder = HifiganGenerator(in_channels=1024, out_channels = 1, resblock_type = "1",
resblock_dilation_sizes = [[1, 3, 5], [1, 3, 5], [1, 3, 5]], resblock_kernel_sizes = [3, 7, 11],
upsample_kernel_sizes = [16, 16, 4, 4], upsample_initial_channel = 512, upsample_factors = [8, 8, 2, 2],
cond_channels=1024).to(self.device).eval()
hifi_model = torch.load(get_model_path('hifidecoder.pth'))
self.hifi_decoder.load_state_dict(hifi_model, strict=False)
self.hifi_decoder.to(self.device)
# Random latent generators (RLGs) are loaded lazily.
self.rlg_auto = None
# Taken from MRQ's api.py
def load_autoregressive_model(self, autoregressive_model_path, is_xtts=False):
if hasattr(self,"autoregressive_model_path") and os.path.samefile(self.autoregressive_model_path, autoregressive_model_path):
return
self.autoregressive_model_path = autoregressive_model_path if autoregressive_model_path and os.path.exists(autoregressive_model_path) else get_model_path('autoregressive.pth', self.models_dir)
new_hash = hash_file(self.autoregressive_model_path)
if hasattr(self,"autoregressive_model_hash") and self.autoregressive_model_hash == new_hash:
return
self.autoregressive_model_hash = new_hash
self.loading = True
print(f"Loading autoregressive model: {self.autoregressive_model_path}")
if hasattr(self, 'autoregressive'):
del self.autoregressive
# XTTS requires a different "dimensionality" for its autoregressive model
if new_hash == "e4ce21eae0043f7691d6a6c8540b74b8" or is_xtts:
dimensionality = {
"max_mel_tokens": 605,
"max_text_tokens": 402,
"max_prompt_tokens": 70,
"max_conditioning_inputs": 1,
"layers": 30,
"model_dim": 1024,
"heads": 16,
"number_text_tokens": 5023, # -1
"start_text_token": 261,
"stop_text_token": 0,
"number_mel_codes": 8194,
"start_mel_token": 8192,
"stop_mel_token": 8193,
}
else:
dimensionality = {
"max_mel_tokens": 604,
"max_text_tokens": 402,
"max_conditioning_inputs": 2,
"layers": 30,
"model_dim": 1024,
"heads": 16,
"number_text_tokens": 255,
"start_text_token": 255,
"checkpointing": False,
"train_solo_embeddings": False
}
self.autoregressive = UnifiedVoice(**dimensionality).cpu().eval()
self.autoregressive.load_state_dict(torch.load(self.autoregressive_model_path))
self.autoregressive.post_init_gpt2_config(use_deepspeed=self.use_deepspeed, kv_cache=self.use_kv_cache)
if self.preloaded_tensors:
self.autoregressive = migrate_to_device( self.autoregressive, self.device )
self.loading = False
print(f"Loaded autoregressive model")
# Taken from MRQ's modified api.py
def load_tokenizer_json(self, tokenizer_json):
if hasattr(self,"tokenizer_json") and os.path.samefile(self.tokenizer_json, tokenizer_json):
return
self.loading = True
self.tokenizer_json = tokenizer_json if tokenizer_json else os.path.join(os.path.dirname(os.path.realpath(__file__)), '../tortoise/data/tokenizer.json')
print("Loading tokenizer JSON:", self.tokenizer_json)
if hasattr(self, 'tokenizer'):
del self.tokenizer
self.tokenizer = VoiceBpeTokenizer(vocab_file=self.tokenizer_json)
self.loading = False
print(f"Loaded tokenizer")
def get_conditioning_latents(self, voice_samples, return_mels=False, verbose=False, slices=1, max_chunk_size=None, force_cpu=False, original_ar=False, original_diffusion=False):
"""
Transforms one or more voice_samples into a tuple (autoregressive_conditioning_latent, diffusion_conditioning_latent).
These are expressive learned latents that encode aspects of the provided clips like voice, intonation, and acoustic
properties.
:param voice_samples: List of 2 or more ~10 second reference clips, which should be torch tensors containing 22.05kHz waveform data.
"""
with torch.no_grad():
# computing conditional latents requires being done on the CPU if using DML because M$ still hasn't implemented some core functions
if get_device_name() == "dml":
force_cpu = True
device = torch.device('cpu') if force_cpu else self.device
if not isinstance(voice_samples, list):
voice_samples = [voice_samples]
resampler_22K = torchaudio.transforms.Resample(
self.input_sample_rate,
22050,
lowpass_filter_width=16,
rolloff=0.85,
resampling_method="kaiser_window",
beta=8.555504641634386,
).to(device)
resampler_24K = torchaudio.transforms.Resample(
self.input_sample_rate,
24000,
lowpass_filter_width=16,
rolloff=0.85,
resampling_method="kaiser_window",
beta=8.555504641634386,
).to(device)
voice_samples = [migrate_to_device(v, device) for v in voice_samples]
auto_conds = []
diffusion_conds = []
if original_ar:
samples = [resampler_22K(sample) for sample in voice_samples]
for sample in tqdm(samples, desc="Computing AR conditioning latents..."):
auto_conds.append(format_conditioning(sample, device=device, sampling_rate=self.input_sample_rate, cond_length=132300))
else:
samples = [resampler_22K(sample) for sample in voice_samples]
concat = torch.cat(samples, dim=-1)
chunk_size = concat.shape[-1]
if slices == 0:
slices = 1
elif max_chunk_size is not None and chunk_size > max_chunk_size:
slices = 1
while int(chunk_size / slices) > max_chunk_size:
slices = slices + 1
chunks = torch.chunk(concat, slices, dim=1)
chunk_size = chunks[0].shape[-1]
for chunk in tqdm(chunks, desc="Computing AR conditioning latents..."):
auto_conds.append(format_conditioning(chunk, device=device, sampling_rate=self.input_sample_rate, cond_length=chunk_size))
auto_conds = torch.stack(auto_conds, dim=1)
self.autoregressive = migrate_to_device( self.autoregressive, device )
auto_latent = self.autoregressive.get_conditioning(auto_conds)
self.autoregressive = migrate_to_device( self.autoregressive, self.device if self.preloaded_tensors else 'cpu' )
if return_mels:
return auto_latent, auto_conds, diffusion_conds
else:
return auto_latent
def get_random_conditioning_latents(self):
# Lazy-load the RLG models.
if self.rlg_auto is None:
self.rlg_auto = RandomLatentConverter(1024).eval()
self.rlg_auto.load_state_dict(torch.load(get_model_path('rlg_auto.pth', self.models_dir), map_location=torch.device('cpu')))
with torch.no_grad():
return self.rlg_auto(torch.tensor([0.0]))
# taken from here https://github.com/coqui-ai/TTS/blob/d21f15cc850788f9cdf93dac0321395138665287/TTS/tts/models/xtts.py#L666
def handle_chunks(self, wav_gen, wav_gen_prev, wav_overlap, overlap_len):
"""Handle chunk formatting in streaming mode"""
wav_chunk = wav_gen[:-overlap_len]
if wav_gen_prev is not None:
wav_chunk = wav_gen[(wav_gen_prev.shape[0] - overlap_len) : -overlap_len]
if wav_overlap is not None:
crossfade_wav = wav_chunk[:overlap_len]
crossfade_wav = crossfade_wav * torch.linspace(0.0, 1.0, overlap_len).to(crossfade_wav.device)
wav_chunk[:overlap_len] = wav_overlap * torch.linspace(1.0, 0.0, overlap_len).to(wav_overlap.device)
wav_chunk[:overlap_len] += crossfade_wav
wav_overlap = wav_gen[-overlap_len:]
wav_gen_prev = wav_gen
return wav_chunk, wav_gen_prev, wav_overlap
def tts_stream(self, text, voice_samples=None, conditioning_latents=None, k=1, verbose=True, use_deterministic_seed=None,
return_deterministic_state=False, overlap_wav_len=1024, stream_chunk_size=40,
# autoregressive generation parameters follow
num_autoregressive_samples=512, temperature=.8, length_penalty=1, repetition_penalty=2.0, top_p=.8, max_mel_tokens=500,
# CVVP parameters follow
cvvp_amount=.0,
# diffusion generation parameters follow
diffusion_iterations=100, cond_free=True, cond_free_k=2, diffusion_temperature=1.0,
**hf_generate_kwargs):
"""
Produces an audio clip of the given text being spoken with the given reference voice.
:param text: Text to be spoken.
:param voice_samples: List of 2 or more ~10 second reference clips which should be torch tensors containing 22.05kHz waveform data.
:param conditioning_latents: A tuple of (autoregressive_conditioning_latent, diffusion_conditioning_latent), which
can be provided in lieu of voice_samples. This is ignored unless voice_samples=None.
Conditioning latents can be retrieved via get_conditioning_latents().
:param k: The number of returned clips. The most likely (as determined by Tortoises' CLVP model) clips are returned.
:param verbose: Whether or not to print log messages indicating the progress of creating a clip. Default=true.
~~AUTOREGRESSIVE KNOBS~~
:param num_autoregressive_samples: Number of samples taken from the autoregressive model, all of which are filtered using CLVP.
As Tortoise is a probabilistic model, more samples means a higher probability of creating something "great".
:param temperature: The softmax temperature of the autoregressive model.
:param length_penalty: A length penalty applied to the autoregressive decoder. Higher settings causes the model to produce more terse outputs.
:param repetition_penalty: A penalty that prevents the autoregressive decoder from repeating itself during decoding. Can be used to reduce the incidence
of long silences or "uhhhhhhs", etc.
:param top_p: P value used in nucleus sampling. (0,1]. Lower values mean the decoder produces more "likely" (aka boring) outputs.
:param max_mel_tokens: Restricts the output length. (0,600] integer. Each unit is 1/20 of a second.
~~DIFFUSION KNOBS~~
:param diffusion_iterations: Number of diffusion steps to perform. [0,4000]. More steps means the network has more chances to iteratively refine
the output, which should theoretically mean a higher quality output. Generally a value above 250 is not noticeably better,
however.
:param cond_free: Whether or not to perform conditioning-free diffusion. Conditioning-free diffusion performs two forward passes for
each diffusion step: one with the outputs of the autoregressive model and one with no conditioning priors. The output
of the two is blended according to the cond_free_k value below. Conditioning-free diffusion is the real deal, and
dramatically improves realism.
:param cond_free_k: Knob that determines how to balance the conditioning free signal with the conditioning-present signal. [0,inf].
As cond_free_k increases, the output becomes dominated by the conditioning-free signal.
Formula is: output=cond_present_output*(cond_free_k+1)-cond_absenct_output*cond_free_k
:param diffusion_temperature: Controls the variance of the noise fed into the diffusion model. [0,1]. Values at 0
are the "mean" prediction of the diffusion network and will sound bland and smeared.
~~OTHER STUFF~~
:param hf_generate_kwargs: The huggingface Transformers generate API is used for the autoregressive transformer.
Extra keyword args fed to this function get forwarded directly to that API. Documentation
here: https://huggingface.co/docs/transformers/internal/generation_utils
:return: Generated audio clip(s) as a torch tensor. Shape 1,S if k=1 else, (k,1,S) where S is the sample length.
Sample rate is 24kHz.
"""
deterministic_seed = self.deterministic_state(seed=use_deterministic_seed)
text_tokens = torch.IntTensor(self.tokenizer.encode(text)).unsqueeze(0).to(self.device)
text_tokens = F.pad(text_tokens, (0, 1)) # This may not be necessary.
assert text_tokens.shape[-1] < 400, 'Too much text provided. Break the text up into separate segments and re-try inference.'
if voice_samples is not None:
auto_conditioning = self.get_conditioning_latents(voice_samples, return_mels=False)
elif conditioning_latents is not None:
latent_tuple = conditioning_latents
if len(latent_tuple) == 2:
auto_conditioning = conditioning_latents
else:
auto_conditioning, auto_conds, _ = conditioning_latents
else:
auto_conditioning = self.get_random_conditioning_latents()
auto_conditioning = migrate_to_device( auto_conditioning, self.device )
with torch.no_grad():
calm_token = 83 # This is the token for coding silence, which is fixed in place with "fix_autoregressive_output"
if verbose:
print("Generating autoregressive samples..")
with torch.autocast(
device_type="cuda" , dtype=torch.float16, enabled=self.half
):
fake_inputs = self.autoregressive.compute_embeddings(
auto_conditioning,
text_tokens,
)
gpt_generator = self.autoregressive.get_generator(
fake_inputs=fake_inputs,
top_k=50,
top_p=top_p,
temperature=temperature,
do_sample=True,
num_beams=1,
num_return_sequences=1,
length_penalty=float(length_penalty),
repetition_penalty=float(repetition_penalty),
output_attentions=False,
output_hidden_states=True,
**hf_generate_kwargs,
)
all_latents = []
codes_ = []
wav_gen_prev = None
wav_overlap = None
is_end = False
first_buffer = 60
while not is_end:
try:
with torch.autocast(
device_type="cuda", dtype=torch.float16, enabled=self.half
):
codes, latent = next(gpt_generator)
all_latents += [latent]
codes_ += [codes]
except StopIteration:
is_end = True
if is_end or (stream_chunk_size > 0 and len(codes_) >= max(stream_chunk_size, first_buffer)):
first_buffer = 0
gpt_latents = torch.cat(all_latents, dim=0)[None, :]
wav_gen = self.hifi_decoder.inference(gpt_latents.to(self.device), auto_conditioning)
wav_gen = wav_gen.squeeze()
wav_chunk, wav_gen_prev, wav_overlap = self.handle_chunks(
wav_gen.squeeze(), wav_gen_prev, wav_overlap, overlap_wav_len
)
codes_ = []
yield wav_chunk
def tts(self, text, voice_samples=None, conditioning_latents=None, k=1, verbose=True, use_deterministic_seed=None,
# autoregressive generation parameters follow
num_autoregressive_samples=512, temperature=.8, length_penalty=6, repetition_penalty=8.0,
top_p=.8, max_mel_tokens=500,
# CVVP parameters follow
cvvp_amount=.0,
**hf_generate_kwargs):
"""
Produces an audio clip of the given text being spoken with the given reference voice.
:param text: Text to be spoken.
:param voice_samples: List of 2 or more ~10 second reference clips which should be torch tensors containing 22.05kHz waveform data.
:param conditioning_latents: A tuple of (autoregressive_conditioning_latent, diffusion_conditioning_latent), which
can be provided in lieu of voice_samples. This is ignored unless voice_samples=None.
Conditioning latents can be retrieved via get_conditioning_latents().
:param k: The number of returned clips. The most likely (as determined by Tortoises' CLVP model) clips are returned.
:param verbose: Whether or not to print log messages indicating the progress of creating a clip. Default=true.
~~AUTOREGRESSIVE KNOBS~~
:param num_autoregressive_samples: Number of samples taken from the autoregressive model, all of which are filtered using CLVP.
As Tortoise is a probabilistic model, more samples means a higher probability of creating something "great".
:param temperature: The softmax temperature of the autoregressive model.
:param length_penalty: A length penalty applied to the autoregressive decoder. Higher settings causes the model to produce more terse outputs.
:param repetition_penalty: A penalty that prevents the autoregressive decoder from repeating itself during decoding. Can be used to reduce the incidence
of long silences or "uhhhhhhs", etc.
:param top_p: P value used in nucleus sampling. (0,1]. Lower values mean the decoder produces more "likely" (aka boring) outputs.
:param max_mel_tokens: Restricts the output length. (0,600] integer. Each unit is 1/20 of a second.
~~DIFFUSION KNOBS~~
:param diffusion_iterations: Number of diffusion steps to perform. [0,4000]. More steps means the network has more chances to iteratively refine
the output, which should theoretically mean a higher quality output. Generally a value above 250 is not noticeably better,
however.
:param cond_free: Whether or not to perform conditioning-free diffusion. Conditioning-free diffusion performs two forward passes for
each diffusion step: one with the outputs of the autoregressive model and one with no conditioning priors. The output
of the two is blended according to the cond_free_k value below. Conditioning-free diffusion is the real deal, and
dramatically improves realism.
:param cond_free_k: Knob that determines how to balance the conditioning free signal with the conditioning-present signal. [0,inf].
As cond_free_k increases, the output becomes dominated by the conditioning-free signal.
Formula is: output=cond_present_output*(cond_free_k+1)-cond_absenct_output*cond_free_k
:param diffusion_temperature: Controls the variance of the noise fed into the diffusion model. [0,1]. Values at 0
are the "mean" prediction of the diffusion network and will sound bland and smeared.
~~OTHER STUFF~~
:param hf_generate_kwargs: The huggingface Transformers generate API is used for the autoregressive transformer.
Extra keyword args fed to this function get forwarded directly to that API. Documentation
here: https://huggingface.co/docs/transformers/internal/generation_utils
:return: Generated audio clip(s) as a torch tensor. Shape 1,S if k=1 else, (k,1,S) where S is the sample length.
Sample rate is 24kHz.
"""
deterministic_seed = self.deterministic_state(seed=use_deterministic_seed)
text_tokens = torch.IntTensor(self.tokenizer.encode(text)).unsqueeze(0).to(self.device)
text_tokens = F.pad(text_tokens, (0, 1)) # This may not be necessary.
assert text_tokens.shape[-1] < 400, 'Too much text provided. Break the text up into separate segments and re-try inference.'
if voice_samples is not None:
auto_conditioning = self.get_conditioning_latents(voice_samples, return_mels=False)
elif conditioning_latents is not None:
auto_conditioning = conditioning_latents
else:
auto_conditioning = self.get_random_conditioning_latents()
auto_conditioning = migrate_to_device(auto_conditioning, self.device)
with torch.no_grad():
calm_token = 83 # This is the token for coding silence, which is fixed in place with "fix_autoregressive_output"
if verbose:
print("Generating autoregressive samples..")
with torch.autocast(
device_type="cuda" , dtype=torch.float16, enabled=self.half
):
# print("Autoregressive model device:", next(self.autoregressive.parameters()).device)
# print("Hifi Decoder model device:", next(self.hifi_decoder.parameters()).device)
codes = self.autoregressive.inference_speech(auto_conditioning, text_tokens,
top_k=50,
top_p=top_p,
temperature=temperature,
do_sample=True,
num_beams=1,
num_return_sequences=1,
length_penalty=float(length_penalty),
repetition_penalty=float(repetition_penalty),
output_attentions=False,
output_hidden_states=True,
**hf_generate_kwargs)
gpt_latents = self.autoregressive(auto_conditioning.repeat(k, 1), text_tokens.repeat(k, 1),
torch.tensor([text_tokens.shape[-1]], device=text_tokens.device), codes,
torch.tensor([codes.shape[-1]*self.autoregressive.mel_length_compression], device=text_tokens.device),
return_latent=True, clip_inputs=False)
if verbose:
print("generating audio..")
wav_gen = self.hifi_decoder.inference(gpt_latents.to(self.device), auto_conditioning)
return wav_gen
def deterministic_state(self, seed=None):
"""
Sets the random seeds that tortoise uses to the current time() and returns that seed so results can be
reproduced.
"""
seed = int(time()) if seed is None else seed
torch.manual_seed(seed)
random.seed(seed)
# Can't currently set this because of CUBLAS. TODO: potentially enable it if necessary.
# torch.use_deterministic_algorithms(True)
return seed

View File

@ -9,9 +9,6 @@ from transformers.utils.model_parallel_utils import get_device_map, assert_devic
from tortoise.models.arch_util import AttentionBlock
from tortoise.utils.typical_sampling import TypicalLogitsWarper
from tortoise.utils.device import get_device_count
import tortoise.utils.torch_intermediary as ml
def null_position_embeddings(range, dim):
return torch.zeros((range.shape[0], range.shape[1], dim), device=range.device)
@ -36,23 +33,22 @@ class ResBlock(nn.Module):
class GPT2InferenceModel(GPT2PreTrainedModel):
def __init__(self, config, gpt, text_pos_emb, embeddings, norm, linear, kv_cache):
def __init__(self, config, gpt, text_pos_emb, embeddings, norm, linear, kv_cache=False):
super().__init__(config)
self.transformer = gpt
self.text_pos_embedding = text_pos_emb
self.embeddings = embeddings
self.final_norm = norm
self.lm_head = nn.Sequential(norm, linear)
self.kv_cache = kv_cache
# Model parallel
self.model_parallel = False
self.device_map = None
self.cached_mel_emb = None
def parallelize(self, device_map=None):
self.device_map = (
get_device_map(len(self.transformer.h), range(get_device_count()))
get_device_map(len(self.transformer.h), range(max(1, torch.cuda.device_count())))
if device_map is None
else device_map
)
@ -67,22 +63,24 @@ class GPT2InferenceModel(GPT2PreTrainedModel):
self.lm_head = self.lm_head.to("cpu")
self.model_parallel = False
torch.cuda.empty_cache()
if torch.backends.mps.is_available():
torch.mps.empty_cache()
def get_output_embeddings(self):
return self.lm_head
def set_output_embeddings(self, new_embeddings):
self.lm_head = new_embeddings
def store_mel_emb(self, mel_emb):
self.cached_mel_emb = mel_emb
def prepare_inputs_for_generation(self, input_ids, past=None, **kwargs):
token_type_ids = kwargs.get("token_type_ids", None)
if not self.kv_cache: past = None
def prepare_inputs_for_generation(self, input_ids, past_key_values=None, **kwargs):
token_type_ids = kwargs.get("token_type_ids", None) # usually None
if not self.kv_cache:
past_key_values = None
# only last token for inputs_ids if past is defined in kwargs
if past:
if past_key_values:
input_ids = input_ids[:, -1].unsqueeze(-1)
if token_type_ids is not None:
token_type_ids = token_type_ids[:, -1].unsqueeze(-1)
@ -94,13 +92,13 @@ class GPT2InferenceModel(GPT2PreTrainedModel):
# create position_ids on the fly for batch generation
position_ids = attention_mask.long().cumsum(-1) - 1
position_ids.masked_fill_(attention_mask == 0, 1)
if past:
if past_key_values:
position_ids = position_ids[:, -1].unsqueeze(-1)
else:
position_ids = None
return {
"input_ids": input_ids,
"past_key_values": past,
"past_key_values": past_key_values,
"use_cache": kwargs.get("use_cache"),
"position_ids": position_ids,
"attention_mask": attention_mask,
@ -127,7 +125,9 @@ class GPT2InferenceModel(GPT2PreTrainedModel):
assert self.cached_mel_emb is not None
assert inputs_embeds is None # Not supported by this inference model.
assert labels is None # Training not supported by this inference model.
return_dict = return_dict if return_dict is not None else self.config.use_return_dict
return_dict = (
return_dict if return_dict is not None else self.config.use_return_dict
)
# Create embedding
mel_len = self.cached_mel_emb.shape[1]
@ -136,14 +136,17 @@ class GPT2InferenceModel(GPT2PreTrainedModel):
text_emb = self.embeddings(text_inputs)
text_emb = text_emb + self.text_pos_embedding(text_emb)
if self.cached_mel_emb.shape[0] != text_emb.shape[0]:
mel_emb = self.cached_mel_emb.repeat_interleave(text_emb.shape[0]//self.cached_mel_emb.shape[0], 0)
else:
mel_emb = self.cached_mel_emb.repeat_interleave(
text_emb.shape[0] // self.cached_mel_emb.shape[0], 0
)
else: # this outcome only occurs once per loop in most cases
mel_emb = self.cached_mel_emb
emb = torch.cat([mel_emb, text_emb], dim=1)
else:
emb = self.embeddings(input_ids)
emb = emb + self.text_pos_embedding.get_fixed_embedding(attention_mask.shape[1]-mel_len, attention_mask.device)
emb = emb + self.text_pos_embedding.get_fixed_embedding(
attention_mask.shape[1] - mel_len, attention_mask.device
)
transformer_outputs = self.transformer(
inputs_embeds=emb,
past_key_values=past_key_values,
@ -162,7 +165,10 @@ class GPT2InferenceModel(GPT2PreTrainedModel):
# Set device for model parallelism
if self.model_parallel:
torch.cuda.set_device(self.transformer.first_device)
if torch.backends.mps.is_available():
self.to(self.transformer.first_device)
else:
torch.cuda.set_device(self.transformer.first_device)
hidden_states = hidden_states.to(self.lm_head.weight.device)
lm_logits = self.lm_head(hidden_states)
@ -187,7 +193,10 @@ class GPT2InferenceModel(GPT2PreTrainedModel):
called. This is required to match :obj:`past_key_values` with the correct beam_idx at every generation step.
"""
return tuple(
tuple(past_state.index_select(0, beam_idx.to(past_state.device)) for past_state in layer_past)
tuple(
past_state.index_select(0, beam_idx.to(past_state.device))
for past_state in layer_past
)
for layer_past in past
)
@ -222,8 +231,7 @@ class ConditioningEncoder(nn.Module):
class LearnedPositionEmbeddings(nn.Module):
def __init__(self, seq_len, model_dim, init=.02):
super().__init__()
# ml.Embedding
self.emb = ml.Embedding(seq_len, model_dim)
self.emb = nn.Embedding(seq_len, model_dim)
# Initializing this way is standard for GPT-2
self.emb.weight.data.normal_(mean=0.0, std=init)
@ -232,7 +240,7 @@ class LearnedPositionEmbeddings(nn.Module):
return self.emb(torch.arange(0, sl, device=x.device))
def get_fixed_embedding(self, ind, dev):
return self.emb(torch.arange(0, ind, device=dev))[ind-1:ind]
return self.emb(torch.tensor([ind], device=dev)).unsqueeze(0)
def build_hf_gpt_transformer(layers, model_dim, heads, max_mel_seq_len, max_text_seq_len, checkpointing):
@ -283,9 +291,9 @@ class MelEncoder(nn.Module):
class UnifiedVoice(nn.Module):
def __init__(self, layers=8, model_dim=512, heads=8, max_text_tokens=120, max_prompt_tokens=2, max_mel_tokens=250, max_conditioning_inputs=1,
def __init__(self, layers=8, model_dim=512, heads=8, max_text_tokens=120, max_mel_tokens=250, max_conditioning_inputs=1,
mel_length_compression=1024, number_text_tokens=256,
start_text_token=None, stop_text_token=0, number_mel_codes=8194, start_mel_token=8192,
start_text_token=None, number_mel_codes=8194, start_mel_token=8192,
stop_mel_token=8193, train_solo_embeddings=False, use_mel_codes_as_input=True,
checkpointing=True, types=1):
"""
@ -295,7 +303,6 @@ class UnifiedVoice(nn.Module):
heads: Number of transformer heads. Must be divisible by model_dim. Recommend model_dim//64
max_text_tokens: Maximum number of text tokens that will be encountered by model.
max_mel_tokens: Maximum number of MEL tokens that will be encountered by model.
max_prompt_tokens: compat set to 2, 70 for XTTS
max_conditioning_inputs: Maximum number of conditioning inputs provided to the model. If (1), conditioning input can be of format (b,80,s), otherwise (b,n,80,s).
mel_length_compression: The factor between <number_input_samples> and <mel_tokens>. Used to compute MEL code padding given wav input length.
number_text_tokens:
@ -312,7 +319,7 @@ class UnifiedVoice(nn.Module):
self.number_text_tokens = number_text_tokens
self.start_text_token = number_text_tokens * types if start_text_token is None else start_text_token
self.stop_text_token = stop_text_token
self.stop_text_token = 0
self.number_mel_codes = number_mel_codes
self.start_mel_token = start_mel_token
self.stop_mel_token = stop_mel_token
@ -320,16 +327,13 @@ class UnifiedVoice(nn.Module):
self.heads = heads
self.max_mel_tokens = max_mel_tokens
self.max_text_tokens = max_text_tokens
self.max_prompt_tokens = max_prompt_tokens
self.model_dim = model_dim
self.max_conditioning_inputs = max_conditioning_inputs
self.mel_length_compression = mel_length_compression
self.conditioning_encoder = ConditioningEncoder(80, model_dim, num_attn_heads=heads)
# ml.Embedding
self.text_embedding = ml.Embedding(self.number_text_tokens*types+1, model_dim)
self.text_embedding = nn.Embedding(self.number_text_tokens*types+1, model_dim)
if use_mel_codes_as_input:
# ml.Embedding
self.mel_embedding = ml.Embedding(self.number_mel_codes, model_dim)
self.mel_embedding = nn.Embedding(self.number_mel_codes, model_dim)
else:
self.mel_embedding = MelEncoder(model_dim, resblocks_per_reduction=1)
self.gpt, self.mel_pos_embedding, self.text_pos_embedding, self.mel_layer_pos_embedding, self.text_layer_pos_embedding = \
@ -342,10 +346,8 @@ class UnifiedVoice(nn.Module):
self.text_solo_embedding = 0
self.final_norm = nn.LayerNorm(model_dim)
# nn.Linear
self.text_head = ml.Linear(model_dim, self.number_text_tokens*types+1)
# nn.Linear
self.mel_head = ml.Linear(model_dim, self.number_mel_codes)
self.text_head = nn.Linear(model_dim, self.number_text_tokens*types+1)
self.mel_head = nn.Linear(model_dim, self.number_mel_codes)
# Initialize the embeddings per the GPT-2 scheme
embeddings = [self.text_embedding]
@ -353,20 +355,35 @@ class UnifiedVoice(nn.Module):
embeddings.append(self.mel_embedding)
for module in embeddings:
module.weight.data.normal_(mean=0.0, std=.02)
def post_init_gpt2_config(self, use_deepspeed=False, kv_cache=False):
seq_length = self.max_mel_tokens + self.max_text_tokens + self.max_prompt_tokens
gpt_config = GPT2Config(vocab_size=self.max_mel_tokens,
n_positions=seq_length,
n_ctx=seq_length,
n_embd=self.model_dim,
n_layer=self.layers,
n_head=self.heads,
gradient_checkpointing=False,
use_cache=True)
self.inference_model = GPT2InferenceModel(gpt_config, self.gpt, self.mel_pos_embedding, self.mel_embedding, self.final_norm, self.mel_head, kv_cache=kv_cache)
#print(f'use_deepspeed autoregressive_debug {use_deepspeed}')
if use_deepspeed and torch.cuda.is_available():
def post_init_gpt2_config(self, use_deepspeed=False, kv_cache=False, half=False):
seq_length = self.max_mel_tokens + self.max_text_tokens + 2
gpt_config = GPT2Config(
vocab_size=self.max_mel_tokens,
n_positions=seq_length,
n_ctx=seq_length,
n_embd=self.model_dim,
n_layer=self.layers,
n_head=self.heads,
gradient_checkpointing=False,
use_cache=True,
)
self.inference_model = GPT2InferenceModel(
gpt_config,
self.gpt,
self.mel_pos_embedding,
self.mel_embedding,
self.final_norm,
self.mel_head,
kv_cache=kv_cache,
)
if use_deepspeed and half and torch.cuda.is_available():
import deepspeed
self.ds_engine = deepspeed.init_inference(model=self.inference_model,
mp_size=1,
replace_with_kernel_inject=True,
dtype=torch.float16)
self.inference_model = self.ds_engine.module.eval()
elif use_deepspeed and torch.cuda.is_available():
import deepspeed
self.ds_engine = deepspeed.init_inference(model=self.inference_model,
mp_size=1,
@ -375,9 +392,9 @@ class UnifiedVoice(nn.Module):
self.inference_model = self.ds_engine.module.eval()
else:
self.inference_model = self.inference_model.eval()
self.gpt.wte = self.mel_embedding
# self.inference_model = PrunedGPT2InferenceModel(gpt_config, self.gpt, self.mel_pos_embedding, self.mel_embedding, self.final_norm, self.mel_head)
self.gpt.wte = self.mel_embedding
def build_aligned_inputs_and_targets(self, input, start_token, stop_token):
inp = F.pad(input, (1,0), value=start_token)
tar = F.pad(input, (0,1), value=stop_token)
@ -493,16 +510,33 @@ class UnifiedVoice(nn.Module):
loss_text = F.cross_entropy(text_logits, text_targets.long())
loss_mel = F.cross_entropy(mel_logits, mel_targets.long())
return loss_text.mean(), loss_mel.mean(), mel_logits
def compute_embeddings(
self,
cond_latents,
text_inputs,
):
text_inputs = F.pad(text_inputs, (0, 1), value=self.stop_text_token)
text_inputs = F.pad(text_inputs, (1, 0), value=self.start_text_token)
emb = self.text_embedding(text_inputs) + self.text_pos_embedding(text_inputs)
conds = cond_latents.unsqueeze(1)
emb = torch.cat([conds, emb], dim=1)
self.inference_model.store_mel_emb(emb)
gpt_inputs = torch.full(
(
emb.shape[0],
emb.shape[1] + 1, # +1 for the start_mel_token
),
fill_value=1,
dtype=torch.long,
device=text_inputs.device,
)
gpt_inputs[:, -1] = self.start_mel_token
return gpt_inputs
def inference_speech(self, speech_conditioning_latent, text_inputs, input_tokens=None, num_return_sequences=1,
max_generate_length=None, typical_sampling=False, typical_mass=.9, **hf_generate_kwargs):
seq_length = self.max_mel_tokens + self.max_text_tokens + self.max_prompt_tokens
if not hasattr(self, 'inference_model'):
self.post_init_gpt2_config(kv_cache=self.kv_cache)
max_generate_length=None, typical_sampling=False, typical_mass=.9, **hf_generate_kwargs):
text_inputs = F.pad(text_inputs, (0, 1), value=self.stop_text_token)
text_inputs, text_targets = self.build_aligned_inputs_and_targets(text_inputs, self.start_text_token, self.stop_text_token)
text_inputs, _ = self.build_aligned_inputs_and_targets(text_inputs, self.start_text_token, self.stop_text_token)
text_emb = self.text_embedding(text_inputs) + self.text_pos_embedding(text_inputs)
conds = speech_conditioning_latent.unsqueeze(1)
@ -528,7 +562,16 @@ class UnifiedVoice(nn.Module):
num_return_sequences=num_return_sequences, **hf_generate_kwargs)
return gen[:, trunc_index:]
def get_generator(self, fake_inputs, **hf_generate_kwargs):
return self.inference_model.generate_stream(
fake_inputs,
bos_token_id=self.start_mel_token,
pad_token_id=self.stop_mel_token,
eos_token_id=self.stop_mel_token,
max_length=500,
do_stream=True,
**hf_generate_kwargs,
)
if __name__ == '__main__':
gpt = UnifiedVoice(model_dim=256, heads=4, train_solo_embeddings=True, use_mel_codes_as_input=True, max_conditioning_inputs=4)
l = gpt(torch.randn(2, 3, 80, 800),
@ -536,4 +579,4 @@ if __name__ == '__main__':
torch.tensor([32, 120]),
torch.randint(high=8192, size=(2,250)),
torch.tensor([250*256,195*256]))
gpt.text_forward(torch.randn(2,80,800), torch.randint(high=50, size=(2,80)), torch.tensor([32, 80]))
gpt.text_forward(torch.randn(2,80,800), torch.randint(high=50, size=(2,80)), torch.tensor([32, 80]))

View File

@ -0,0 +1,303 @@
# adopted from https://github.com/jik876/hifi-gan/blob/master/models.py
import torch
from torch import nn
from torch.nn import Conv1d, ConvTranspose1d
from torch.nn import functional as F
from torch.nn.utils import remove_weight_norm, weight_norm
LRELU_SLOPE = 0.1
def get_padding(k, d):
return int((k * d - d) / 2)
class ResBlock1(torch.nn.Module):
"""Residual Block Type 1. It has 3 convolutional layers in each convolutional block.
Network::
x -> lrelu -> conv1_1 -> conv1_2 -> conv1_3 -> z -> lrelu -> conv2_1 -> conv2_2 -> conv2_3 -> o -> + -> o
|--------------------------------------------------------------------------------------------------|
Args:
channels (int): number of hidden channels for the convolutional layers.
kernel_size (int): size of the convolution filter in each layer.
dilations (list): list of dilation value for each conv layer in a block.
"""
def __init__(self, channels, kernel_size=3, dilation=(1, 3, 5)):
super().__init__()
self.convs1 = nn.ModuleList(
[
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
1,
dilation=dilation[0],
padding=get_padding(kernel_size, dilation[0]),
)
),
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
1,
dilation=dilation[1],
padding=get_padding(kernel_size, dilation[1]),
)
),
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
1,
dilation=dilation[2],
padding=get_padding(kernel_size, dilation[2]),
)
),
]
)
self.convs2 = nn.ModuleList(
[
weight_norm(
Conv1d(channels, channels, kernel_size, 1, dilation=1, padding=get_padding(kernel_size, 1))
),
weight_norm(
Conv1d(channels, channels, kernel_size, 1, dilation=1, padding=get_padding(kernel_size, 1))
),
weight_norm(
Conv1d(channels, channels, kernel_size, 1, dilation=1, padding=get_padding(kernel_size, 1))
),
]
)
def forward(self, x):
"""
Args:
x (Tensor): input tensor.
Returns:
Tensor: output tensor.
Shapes:
x: [B, C, T]
"""
for c1, c2 in zip(self.convs1, self.convs2):
xt = F.leaky_relu(x, LRELU_SLOPE)
xt = c1(xt)
xt = F.leaky_relu(xt, LRELU_SLOPE)
xt = c2(xt)
x = xt + x
return x
def remove_weight_norm(self):
for l in self.convs1:
remove_weight_norm(l)
for l in self.convs2:
remove_weight_norm(l)
class ResBlock2(torch.nn.Module):
"""Residual Block Type 2. It has 1 convolutional layers in each convolutional block.
Network::
x -> lrelu -> conv1-> -> z -> lrelu -> conv2-> o -> + -> o
|---------------------------------------------------|
Args:
channels (int): number of hidden channels for the convolutional layers.
kernel_size (int): size of the convolution filter in each layer.
dilations (list): list of dilation value for each conv layer in a block.
"""
def __init__(self, channels, kernel_size=3, dilation=(1, 3)):
super().__init__()
self.convs = nn.ModuleList(
[
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
1,
dilation=dilation[0],
padding=get_padding(kernel_size, dilation[0]),
)
),
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
1,
dilation=dilation[1],
padding=get_padding(kernel_size, dilation[1]),
)
),
]
)
def forward(self, x):
for c in self.convs:
xt = F.leaky_relu(x, LRELU_SLOPE)
xt = c(xt)
x = xt + x
return x
def remove_weight_norm(self):
for l in self.convs:
remove_weight_norm(l)
class HifiganGenerator(torch.nn.Module):
def __init__(
self,
in_channels,
out_channels,
resblock_type,
resblock_dilation_sizes,
resblock_kernel_sizes,
upsample_kernel_sizes,
upsample_initial_channel,
upsample_factors,
inference_padding=5,
cond_channels=0,
conv_pre_weight_norm=True,
conv_post_weight_norm=True,
conv_post_bias=True,
):
r"""HiFiGAN Generator with Multi-Receptive Field Fusion (MRF)
Network:
x -> lrelu -> upsampling_layer -> resblock1_k1x1 -> z1 -> + -> z_sum / #resblocks -> lrelu -> conv_post_7x1 -> tanh -> o
.. -> zI ---|
resblockN_kNx1 -> zN ---'
Args:
in_channels (int): number of input tensor channels.
out_channels (int): number of output tensor channels.
resblock_type (str): type of the `ResBlock`. '1' or '2'.
resblock_dilation_sizes (List[List[int]]): list of dilation values in each layer of a `ResBlock`.
resblock_kernel_sizes (List[int]): list of kernel sizes for each `ResBlock`.
upsample_kernel_sizes (List[int]): list of kernel sizes for each transposed convolution.
upsample_initial_channel (int): number of channels for the first upsampling layer. This is divided by 2
for each consecutive upsampling layer.
upsample_factors (List[int]): upsampling factors (stride) for each upsampling layer.
inference_padding (int): constant padding applied to the input at inference time. Defaults to 5.
"""
super().__init__()
self.inference_padding = inference_padding
self.num_kernels = len(resblock_kernel_sizes)
self.num_upsamples = len(upsample_factors)
# initial upsampling layers
self.conv_pre = weight_norm(Conv1d(in_channels, upsample_initial_channel, 7, 1, padding=3))
resblock = ResBlock1 if resblock_type == "1" else ResBlock2
# upsampling layers
self.ups = nn.ModuleList()
for i, (u, k) in enumerate(zip(upsample_factors, upsample_kernel_sizes)):
self.ups.append(
weight_norm(
ConvTranspose1d(
upsample_initial_channel // (2**i),
upsample_initial_channel // (2 ** (i + 1)),
k,
u,
padding=(k - u) // 2,
)
)
)
# MRF blocks
self.resblocks = nn.ModuleList()
for i in range(len(self.ups)):
ch = upsample_initial_channel // (2 ** (i + 1))
for _, (k, d) in enumerate(zip(resblock_kernel_sizes, resblock_dilation_sizes)):
self.resblocks.append(resblock(ch, k, d))
# post convolution layer
self.conv_post = weight_norm(Conv1d(ch, out_channels, 7, 1, padding=3, bias=conv_post_bias))
if cond_channels > 0:
self.cond_layer = nn.Conv1d(cond_channels, upsample_initial_channel, 1)
if not conv_pre_weight_norm:
remove_weight_norm(self.conv_pre)
if not conv_post_weight_norm:
remove_weight_norm(self.conv_post)
self.device = torch.device('cuda' if torch.cuda.is_available() else'cpu')
if torch.backends.mps.is_available():
self.device = torch.device('mps')
def forward(self, x, g=None):
"""
Args:
x (Tensor): feature input tensor.
g (Tensor): global conditioning input tensor.
Returns:
Tensor: output waveform.
Shapes:
x: [B, C, T]
Tensor: [B, 1, T]
"""
o = self.conv_pre(x)
if hasattr(self, "cond_layer"):
o = o + self.cond_layer(g)
for i in range(self.num_upsamples):
o = F.leaky_relu(o, LRELU_SLOPE)
o = self.ups[i](o)
z_sum = None
for j in range(self.num_kernels):
if z_sum is None:
z_sum = self.resblocks[i * self.num_kernels + j](o)
else:
z_sum += self.resblocks[i * self.num_kernels + j](o)
o = z_sum / self.num_kernels
o = F.leaky_relu(o)
o = self.conv_post(o)
o = torch.tanh(o)
return o
@torch.no_grad()
def inference(self, c, g=None):
"""
Args:
x (Tensor): conditioning input tensor.
Returns:
Tensor: output waveform.
Shapes:
x: [B, C, T]
Tensor: [B, 1, T]
"""
# c = c.to(self.conv_pre.weight.device)
# c = torch.nn.functional.pad(c, (self.inference_padding, self.inference_padding), "replicate")
up_1 = torch.nn.functional.interpolate(
c.transpose(1,2),
scale_factor=[1024 / 256],
mode="linear",
)
up_2 = torch.nn.functional.interpolate(
up_1,
scale_factor=[24000 / 22050],
mode="linear",
)
g = g.unsqueeze(0)
return self.forward(up_2.to(self.device), g.transpose(1,2))
def remove_weight_norm(self):
print("Removing weight norm...")
for l in self.ups:
remove_weight_norm(l)
for l in self.resblocks:
l.remove_weight_norm()
remove_weight_norm(self.conv_pre)
remove_weight_norm(self.conv_post)

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