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Zero
Running
on
Zero
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 | |
import codecs | |
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( | |
"-f", | |
"--gen_file", | |
type=str, | |
help="File with text to generate. Ignores --text", | |
) | |
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"] | |
gen_file = args.gen_file if args.gen_file else config["gen_file"] | |
if gen_file: | |
gen_text = codecs.open(gen_file, "r", "utf-8").read() | |
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)) |