import io from typing import Union import numpy as np from modules.Enhancer.ResembleEnhance import apply_audio_enhance as _apply_audio_enhance from modules.devices import devices from modules.synthesize_audio import synthesize_audio from modules.utils.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): return _apply_audio_enhance(audio_data, sr, enable_denoise, enable_enhance) @torch.inference_mode() @spaces.GPU def synthesize_ssml( ssml: str, batch_size=4, enable_enhance=False, enable_denoise=False, eos: str = "[uv_break]", spliter_thr: int = 100, ): 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, eos=eos, spliter_thr=spliter_thr ) 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, spliter_thr: int = 100, eos: str = "[uv_break]", ): 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, end_of_sentence=eos, spliter_threshold=spliter_thr, ) 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