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on
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Running
on
Zero
import io | |
from typing import Union | |
import numpy as np | |
from modules.Enhancer.ResembleEnhance import load_enhancer | |
from modules.devices import devices | |
from modules.synthesize_audio import synthesize_audio | |
from modules.hf import spaces | |
from modules.webui import webui_config | |
import torch | |
from modules.ssml_parser.SSMLParser import create_ssml_parser, SSMLBreak, SSMLSegment | |
from modules.SynthesizeSegments import SynthesizeSegments, combine_audio_segments | |
from modules.speaker import speaker_mgr, Speaker | |
from modules.data import styles_mgr | |
from modules.api.utils import calc_spk_style | |
from modules.normalization import text_normalize | |
from modules import refiner | |
from modules.utils import audio | |
from modules.SentenceSplitter import SentenceSplitter | |
from pydub import AudioSegment | |
import torch.profiler | |
def get_speakers(): | |
return speaker_mgr.list_speakers() | |
def get_speaker_names() -> tuple[list[Speaker], list[str]]: | |
speakers = get_speakers() | |
def get_speaker_show_name(spk): | |
if spk.gender == "*" or spk.gender == "": | |
return spk.name | |
return f"{spk.gender} : {spk.name}" | |
speaker_names = [get_speaker_show_name(speaker) for speaker in speakers] | |
speaker_names.sort(key=lambda x: x.startswith("*") and "-1" or x) | |
return speakers, speaker_names | |
def get_styles(): | |
return styles_mgr.list_items() | |
def load_spk_info(file): | |
if file is None: | |
return "empty" | |
try: | |
spk: Speaker = Speaker.from_file(file) | |
infos = spk.to_json() | |
return f""" | |
- name: {infos.name} | |
- gender: {infos.gender} | |
- describe: {infos.describe} | |
""".strip() | |
except: | |
return "load failed" | |
def segments_length_limit( | |
segments: list[Union[SSMLBreak, SSMLSegment]], total_max: int | |
) -> list[Union[SSMLBreak, SSMLSegment]]: | |
ret_segments = [] | |
total_len = 0 | |
for seg in segments: | |
if isinstance(seg, SSMLBreak): | |
ret_segments.append(seg) | |
continue | |
total_len += len(seg["text"]) | |
if total_len > total_max: | |
break | |
ret_segments.append(seg) | |
return ret_segments | |
def apply_audio_enhance(audio_data, sr, enable_denoise, enable_enhance): | |
if not enable_denoise and not enable_enhance: | |
return audio_data, sr | |
device = devices.device | |
# NOTE: 这里很奇怪按道理得放到 device 上,但是 enhancer 做 chunk 的时候会报错...所以得 cpu() | |
tensor = torch.from_numpy(audio_data).float().squeeze().cpu() | |
enhancer = load_enhancer(device) | |
if enable_enhance or enable_denoise: | |
lambd = 0.9 if enable_denoise else 0.1 | |
tensor, sr = enhancer.enhance( | |
tensor, sr, tau=0.5, nfe=64, solver="rk4", lambd=lambd, device=device | |
) | |
audio_data = tensor.cpu().numpy() | |
return audio_data, int(sr) | |
def synthesize_ssml( | |
ssml: str, | |
batch_size=4, | |
enable_enhance=False, | |
enable_denoise=False, | |
): | |
try: | |
batch_size = int(batch_size) | |
except Exception: | |
batch_size = 8 | |
ssml = ssml.strip() | |
if ssml == "": | |
return None | |
parser = create_ssml_parser() | |
segments = parser.parse(ssml) | |
max_len = webui_config.ssml_max | |
segments = segments_length_limit(segments, max_len) | |
if len(segments) == 0: | |
return None | |
synthesize = SynthesizeSegments(batch_size=batch_size) | |
audio_segments = synthesize.synthesize_segments(segments) | |
combined_audio = combine_audio_segments(audio_segments) | |
sr = combined_audio.frame_rate | |
audio_data, sr = apply_audio_enhance( | |
audio.audiosegment_to_librosawav(combined_audio), | |
sr, | |
enable_denoise, | |
enable_enhance, | |
) | |
# NOTE: 这里必须要加,不然 gradio 没法解析成 mp3 格式 | |
audio_data = audio.audio_to_int16(audio_data) | |
return sr, audio_data | |
# @torch.inference_mode() | |
def tts_generate( | |
text, | |
temperature=0.3, | |
top_p=0.7, | |
top_k=20, | |
spk=-1, | |
infer_seed=-1, | |
use_decoder=True, | |
prompt1="", | |
prompt2="", | |
prefix="", | |
style="", | |
disable_normalize=False, | |
batch_size=4, | |
enable_enhance=False, | |
enable_denoise=False, | |
spk_file=None, | |
): | |
try: | |
batch_size = int(batch_size) | |
except Exception: | |
batch_size = 4 | |
max_len = webui_config.tts_max | |
text = text.strip()[0:max_len] | |
if text == "": | |
return None | |
if style == "*auto": | |
style = None | |
if isinstance(top_k, float): | |
top_k = int(top_k) | |
params = calc_spk_style(spk=spk, style=style) | |
spk = params.get("spk", spk) | |
infer_seed = infer_seed or params.get("seed", infer_seed) | |
temperature = temperature or params.get("temperature", temperature) | |
prefix = prefix or params.get("prefix", prefix) | |
prompt1 = prompt1 or params.get("prompt1", "") | |
prompt2 = prompt2 or params.get("prompt2", "") | |
infer_seed = np.clip(infer_seed, -1, 2**32 - 1, out=None, dtype=np.float64) | |
infer_seed = int(infer_seed) | |
if not disable_normalize: | |
text = text_normalize(text) | |
if spk_file: | |
spk = Speaker.from_file(spk_file) | |
sample_rate, audio_data = synthesize_audio( | |
text=text, | |
temperature=temperature, | |
top_P=top_p, | |
top_K=top_k, | |
spk=spk, | |
infer_seed=infer_seed, | |
use_decoder=use_decoder, | |
prompt1=prompt1, | |
prompt2=prompt2, | |
prefix=prefix, | |
batch_size=batch_size, | |
) | |
audio_data, sample_rate = apply_audio_enhance( | |
audio_data, sample_rate, enable_denoise, enable_enhance | |
) | |
# NOTE: 这里必须要加,不然 gradio 没法解析成 mp3 格式 | |
audio_data = audio.audio_to_int16(audio_data) | |
return sample_rate, audio_data | |
def refine_text(text: str, prompt: str): | |
text = text_normalize(text) | |
return refiner.refine_text(text, prompt=prompt) | |
def split_long_text(long_text_input): | |
spliter = SentenceSplitter(webui_config.spliter_threshold) | |
sentences = spliter.parse(long_text_input) | |
sentences = [text_normalize(s) for s in sentences] | |
data = [] | |
for i, text in enumerate(sentences): | |
data.append([i, text, len(text)]) | |
return data | |