Spaces:
Running
on
Zero
Running
on
Zero
File size: 14,119 Bytes
dd217c7 |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336 337 338 339 340 341 342 343 344 345 346 347 348 349 350 351 352 353 354 355 356 357 358 359 360 361 362 363 364 365 366 367 368 369 370 371 372 373 374 375 376 377 378 |
import re
import torch
import torchaudio
import numpy as np
import tempfile
from einops import rearrange
from vocos import Vocos
from pydub import AudioSegment, silence
from model import CFM, UNetT, DiT, MMDiT
from cached_path import cached_path
from model.utils import (
load_checkpoint,
get_tokenizer,
convert_char_to_pinyin,
save_spectrogram,
)
from transformers import pipeline
import soundfile as sf
import tomli
import argparse
import tqdm
from pathlib import Path
parser = argparse.ArgumentParser(
prog="python3 inference-cli.py",
description="Commandline interface for E2/F5 TTS with Advanced Batch Processing.",
epilog="Specify options above to override one or more settings from config.",
)
parser.add_argument(
"-c",
"--config",
help="Configuration file. Default=cli-config.toml",
default="inference-cli.toml",
)
parser.add_argument(
"-m",
"--model",
help="F5-TTS | E2-TTS",
)
parser.add_argument(
"-r",
"--ref_audio",
type=str,
help="Reference audio file < 15 seconds."
)
parser.add_argument(
"-s",
"--ref_text",
type=str,
default="666",
help="Subtitle for the reference audio."
)
parser.add_argument(
"-t",
"--gen_text",
type=str,
help="Text to generate.",
)
parser.add_argument(
"-o",
"--output_dir",
type=str,
help="Path to output folder..",
)
parser.add_argument(
"--remove_silence",
help="Remove silence.",
)
args = parser.parse_args()
config = tomli.load(open(args.config, "rb"))
ref_audio = args.ref_audio if args.ref_audio else config["ref_audio"]
ref_text = args.ref_text if args.ref_text != "666" else config["ref_text"]
gen_text = args.gen_text if args.gen_text else config["gen_text"]
output_dir = args.output_dir if args.output_dir else config["output_dir"]
model = args.model if args.model else config["model"]
remove_silence = args.remove_silence if args.remove_silence else config["remove_silence"]
wave_path = Path(output_dir)/"out.wav"
spectrogram_path = Path(output_dir)/"out.png"
SPLIT_WORDS = [
"but", "however", "nevertheless", "yet", "still",
"therefore", "thus", "hence", "consequently",
"moreover", "furthermore", "additionally",
"meanwhile", "alternatively", "otherwise",
"namely", "specifically", "for example", "such as",
"in fact", "indeed", "notably",
"in contrast", "on the other hand", "conversely",
"in conclusion", "to summarize", "finally"
]
device = (
"cuda"
if torch.cuda.is_available()
else "mps" if torch.backends.mps.is_available() else "cpu"
)
vocos = Vocos.from_pretrained("charactr/vocos-mel-24khz")
print(f"Using {device} device")
# --------------------- Settings -------------------- #
target_sample_rate = 24000
n_mel_channels = 100
hop_length = 256
target_rms = 0.1
nfe_step = 32 # 16, 32
cfg_strength = 2.0
ode_method = "euler"
sway_sampling_coef = -1.0
speed = 1.0
# fix_duration = 27 # None or float (duration in seconds)
fix_duration = None
def load_model(repo_name, exp_name, model_cls, model_cfg, ckpt_step):
ckpt_path = str(cached_path(f"hf://SWivid/{repo_name}/{exp_name}/model_{ckpt_step}.safetensors"))
# ckpt_path = f"ckpts/{exp_name}/model_{ckpt_step}.pt" # .pt | .safetensors
vocab_char_map, vocab_size = get_tokenizer("Emilia_ZH_EN", "pinyin")
model = CFM(
transformer=model_cls(
**model_cfg, text_num_embeds=vocab_size, mel_dim=n_mel_channels
),
mel_spec_kwargs=dict(
target_sample_rate=target_sample_rate,
n_mel_channels=n_mel_channels,
hop_length=hop_length,
),
odeint_kwargs=dict(
method=ode_method,
),
vocab_char_map=vocab_char_map,
).to(device)
model = load_checkpoint(model, ckpt_path, device, use_ema = True)
return model
# load models
F5TTS_model_cfg = dict(
dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4
)
E2TTS_model_cfg = dict(dim=1024, depth=24, heads=16, ff_mult=4)
def split_text_into_batches(text, max_chars=200, split_words=SPLIT_WORDS):
if len(text.encode('utf-8')) <= max_chars:
return [text]
if text[-1] not in ['。', '.', '!', '!', '?', '?']:
text += '.'
sentences = re.split('([。.!?!?])', text)
sentences = [''.join(i) for i in zip(sentences[0::2], sentences[1::2])]
batches = []
current_batch = ""
def split_by_words(text):
words = text.split()
current_word_part = ""
word_batches = []
for word in words:
if len(current_word_part.encode('utf-8')) + len(word.encode('utf-8')) + 1 <= max_chars:
current_word_part += word + ' '
else:
if current_word_part:
# Try to find a suitable split word
for split_word in split_words:
split_index = current_word_part.rfind(' ' + split_word + ' ')
if split_index != -1:
word_batches.append(current_word_part[:split_index].strip())
current_word_part = current_word_part[split_index:].strip() + ' '
break
else:
# If no suitable split word found, just append the current part
word_batches.append(current_word_part.strip())
current_word_part = ""
current_word_part += word + ' '
if current_word_part:
word_batches.append(current_word_part.strip())
return word_batches
for sentence in sentences:
if len(current_batch.encode('utf-8')) + len(sentence.encode('utf-8')) <= max_chars:
current_batch += sentence
else:
# If adding this sentence would exceed the limit
if current_batch:
batches.append(current_batch)
current_batch = ""
# If the sentence itself is longer than max_chars, split it
if len(sentence.encode('utf-8')) > max_chars:
# First, try to split by colon
colon_parts = sentence.split(':')
if len(colon_parts) > 1:
for part in colon_parts:
if len(part.encode('utf-8')) <= max_chars:
batches.append(part)
else:
# If colon part is still too long, split by comma
comma_parts = re.split('[,,]', part)
if len(comma_parts) > 1:
current_comma_part = ""
for comma_part in comma_parts:
if len(current_comma_part.encode('utf-8')) + len(comma_part.encode('utf-8')) <= max_chars:
current_comma_part += comma_part + ','
else:
if current_comma_part:
batches.append(current_comma_part.rstrip(','))
current_comma_part = comma_part + ','
if current_comma_part:
batches.append(current_comma_part.rstrip(','))
else:
# If no comma, split by words
batches.extend(split_by_words(part))
else:
# If no colon, split by comma
comma_parts = re.split('[,,]', sentence)
if len(comma_parts) > 1:
current_comma_part = ""
for comma_part in comma_parts:
if len(current_comma_part.encode('utf-8')) + len(comma_part.encode('utf-8')) <= max_chars:
current_comma_part += comma_part + ','
else:
if current_comma_part:
batches.append(current_comma_part.rstrip(','))
current_comma_part = comma_part + ','
if current_comma_part:
batches.append(current_comma_part.rstrip(','))
else:
# If no comma, split by words
batches.extend(split_by_words(sentence))
else:
current_batch = sentence
if current_batch:
batches.append(current_batch)
return batches
def infer_batch(ref_audio, ref_text, gen_text_batches, model, remove_silence):
if model == "F5-TTS":
ema_model = load_model(model, "F5TTS_Base", DiT, F5TTS_model_cfg, 1200000)
elif model == "E2-TTS":
ema_model = load_model(model, "E2TTS_Base", UNetT, E2TTS_model_cfg, 1200000)
audio, sr = ref_audio
if audio.shape[0] > 1:
audio = torch.mean(audio, dim=0, keepdim=True)
rms = torch.sqrt(torch.mean(torch.square(audio)))
if rms < target_rms:
audio = audio * target_rms / rms
if sr != target_sample_rate:
resampler = torchaudio.transforms.Resample(sr, target_sample_rate)
audio = resampler(audio)
audio = audio.to(device)
generated_waves = []
spectrograms = []
for i, gen_text in enumerate(tqdm.tqdm(gen_text_batches)):
# Prepare the text
if len(ref_text[-1].encode('utf-8')) == 1:
ref_text = ref_text + " "
text_list = [ref_text + gen_text]
final_text_list = convert_char_to_pinyin(text_list)
# Calculate duration
ref_audio_len = audio.shape[-1] // hop_length
zh_pause_punc = r"。,、;:?!"
ref_text_len = len(ref_text.encode('utf-8')) + 3 * len(re.findall(zh_pause_punc, ref_text))
gen_text_len = len(gen_text.encode('utf-8')) + 3 * len(re.findall(zh_pause_punc, gen_text))
duration = ref_audio_len + int(ref_audio_len / ref_text_len * gen_text_len / speed)
# inference
with torch.inference_mode():
generated, _ = ema_model.sample(
cond=audio,
text=final_text_list,
duration=duration,
steps=nfe_step,
cfg_strength=cfg_strength,
sway_sampling_coef=sway_sampling_coef,
)
generated = generated[:, ref_audio_len:, :]
generated_mel_spec = rearrange(generated, "1 n d -> 1 d n")
generated_wave = vocos.decode(generated_mel_spec.cpu())
if rms < target_rms:
generated_wave = generated_wave * rms / target_rms
# wav -> numpy
generated_wave = generated_wave.squeeze().cpu().numpy()
generated_waves.append(generated_wave)
spectrograms.append(generated_mel_spec[0].cpu().numpy())
# Combine all generated waves
final_wave = np.concatenate(generated_waves)
with open(wave_path, "wb") as f:
sf.write(f.name, final_wave, target_sample_rate)
# Remove silence
if remove_silence:
aseg = AudioSegment.from_file(f.name)
non_silent_segs = silence.split_on_silence(aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=500)
non_silent_wave = AudioSegment.silent(duration=0)
for non_silent_seg in non_silent_segs:
non_silent_wave += non_silent_seg
aseg = non_silent_wave
aseg.export(f.name, format="wav")
print(f.name)
# Create a combined spectrogram
combined_spectrogram = np.concatenate(spectrograms, axis=1)
save_spectrogram(combined_spectrogram, spectrogram_path)
print(spectrogram_path)
def infer(ref_audio_orig, ref_text, gen_text, model, remove_silence, custom_split_words):
if not custom_split_words.strip():
custom_words = [word.strip() for word in custom_split_words.split(',')]
global SPLIT_WORDS
SPLIT_WORDS = custom_words
print(gen_text)
print("Converting audio...")
with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f:
aseg = AudioSegment.from_file(ref_audio_orig)
non_silent_segs = silence.split_on_silence(aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=500)
non_silent_wave = AudioSegment.silent(duration=0)
for non_silent_seg in non_silent_segs:
non_silent_wave += non_silent_seg
aseg = non_silent_wave
audio_duration = len(aseg)
if audio_duration > 15000:
print("Audio is over 15s, clipping to only first 15s.")
aseg = aseg[:15000]
aseg.export(f.name, format="wav")
ref_audio = f.name
if not ref_text.strip():
print("No reference text provided, transcribing reference audio...")
pipe = pipeline(
"automatic-speech-recognition",
model="openai/whisper-large-v3-turbo",
torch_dtype=torch.float16,
device=device,
)
ref_text = pipe(
ref_audio,
chunk_length_s=30,
batch_size=128,
generate_kwargs={"task": "transcribe"},
return_timestamps=False,
)["text"].strip()
print("Finished transcription")
else:
print("Using custom reference text...")
# Split the input text into batches
audio, sr = torchaudio.load(ref_audio)
max_chars = int(len(ref_text.encode('utf-8')) / (audio.shape[-1] / sr) * (30 - audio.shape[-1] / sr))
gen_text_batches = split_text_into_batches(gen_text, max_chars=max_chars)
print('ref_text', ref_text)
for i, gen_text in enumerate(gen_text_batches):
print(f'gen_text {i}', gen_text)
print(f"Generating audio using {model} in {len(gen_text_batches)} batches, loading models...")
return infer_batch((audio, sr), ref_text, gen_text_batches, model, remove_silence)
infer(ref_audio, ref_text, gen_text, model, remove_silence, ",".join(SPLIT_WORDS)) |