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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
@torch.inference_mode()
@spaces.GPU
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)
@torch.inference_mode()
@spaces.GPU
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()
@spaces.GPU
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
@torch.inference_mode()
@spaces.GPU
def refine_text(text: str, prompt: str):
text = text_normalize(text)
return refiner.refine_text(text, prompt=prompt)
@torch.inference_mode()
@spaces.GPU
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
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