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