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import os
import re
import traceback
import xml.etree.ElementTree as ET
from io import BytesIO
import librosa
import numpy as np
import soundfile as sf
from graiax import silkcoder
from scipy.signal import resample_poly
from contants import ModelType
from config import config, BASE_DIR
from logger import logger
from manager.observer import Observer
from utils.data_utils import check_is_none
from utils.sentence import sentence_split_and_markup, sentence_split, sentence_split_reading
class TTSManager(Observer):
def __init__(self, model_manager, **kwargs):
self.model_manager = model_manager
self.strength_dict = {"x-weak": 0.25, "weak": 0.5, "Medium": 0.75, "Strong": 1, "x-strong": 1.25}
self.logger = logger
self.infer_map = {
ModelType.VITS: self.vits_infer,
ModelType.W2V2_VITS: self.w2v2_vits_infer,
ModelType.HUBERT_VITS: self.hubert_vits_infer,
ModelType.BERT_VITS2: self.bert_vits2_infer,
ModelType.GPT_SOVITS: self.gpt_sovits_infer
}
self.speaker_lang = None
if getattr(config, "LANGUAGE_AUTOMATIC_DETECT", []) != []:
self.speaker_lang = getattr(config, "LANGUAGE_AUTOMATIC_DETECT")
@property
def sid2model(self):
return self.model_manager.sid2model
@property
def spk2model(self):
return self.model_manager.spk2model
@property
def voice_speakers(self):
return self.model_manager.voice_speakers
@property
def dimensional_emotion_model(self):
return self.model_manager.dimensional_emotion_model
def update(self, message, **kwargs):
if message == "model_loaded":
self._handle_model_loaded(kwargs['model_manager'])
elif message == "model_unloaded":
self._handle_model_unloaded(kwargs['model_manager'])
def _handle_model_loaded(self, model_manager):
self.model_manager = model_manager
def _handle_model_unloaded(self, model_manager):
self.model_manager = model_manager
def encode(self, sampling_rate, audio, format):
with BytesIO() as f:
if format.upper() == 'OGG':
sf.write(f, audio, sampling_rate, format="ogg")
return BytesIO(f.getvalue())
elif format.upper() == 'SILK':
sf.write(f, audio, sampling_rate, format="wav")
return BytesIO(silkcoder.encode(f))
elif format.upper() == 'MP3':
sf.write(f, audio, sampling_rate, format="mp3")
return BytesIO(f.getvalue())
elif format.upper() == 'WAV':
sf.write(f, audio, sampling_rate, format="wav")
return BytesIO(f.getvalue())
elif format.upper() == 'FLAC':
sf.write(f, audio, sampling_rate, format="flac")
return BytesIO(f.getvalue())
else:
raise ValueError(f"Unsupported format:{format}")
def convert_time_string(self, time_string):
time_value = float(re.findall(r'\d+\.?\d*', time_string)[0])
time_unit = re.findall(r'[a-zA-Z]+', time_string)[0].lower()
if time_unit.upper() == 'MS':
return time_value / 1000
elif time_unit.upper() == 'S':
return time_value
elif time_unit.upper() == 'MIN':
return time_value * 60
elif time_unit.upper() == 'H':
return time_value * 3600
elif time_unit.upper() == 'D':
return time_value * 24 * 3600 # 不会有人真写D吧?
else:
raise ValueError("Unsupported time unit: {}".format(time_unit))
def generate_audio_chunks(self, audio):
chunk_size = 4096
while True:
chunk = audio.read(chunk_size)
if not chunk:
break
yield chunk
def resample_audio(self, audio, orig_sr, target_sr):
if orig_sr == target_sr:
return audio
gcd = np.gcd(orig_sr, target_sr)
audio = resample_poly(audio, target_sr // gcd, orig_sr // gcd)
return audio
def get_model(self, model_type, id, speaker=None):
if speaker is not None:
return self.spk2model[model_type][speaker]["model"]
return self.sid2model[model_type][id]["model"]
def get_real_id(self, model_type, id, speaker=None):
if speaker is not None:
return self.spk2model[model_type][speaker]["real_id"]
return self.sid2model[model_type][id]["real_id"]
def get_model_id(self, model_type, id):
return self.sid2model[model_type][id]["model_id"]
def normalize(self, state):
int_keys = ["id", "segement_size", "emotion", "top_k"]
float_keys = ["noise", "noisew", "length", "sdp_ratio", "style_weight", "top_p", "temperature"]
for key in state:
if key in int_keys and state[key] is not None:
state[key] = int(state[key])
elif key in float_keys and state[key] is not None:
state[key] = float(state[key])
return state
def parse_ssml(self, ssml):
root = ET.fromstring(ssml)
format = root.attrib.get("format", "wav")
voice_tasks = []
brk_count = 0
strength_dict = {"x-weak": 0.25, "weak": 0.5, "Medium": 0.75, "Strong": 1, "x-strong": 1.25}
params = {ModelType.VITS: config.vits_config.asdict(),
ModelType.W2V2_VITS: config.w2v2_vits_config.asdict(),
ModelType.HUBERT_VITS: config.hubert_vits_config.asdict(),
ModelType.BERT_VITS2: config.bert_vits2_config.asdict(),
ModelType.GPT_SOVITS: config.gpt_sovits_config.asdict(),
}
for element in root.iter():
if element.tag == "voice":
# 不填写则默认从已加载的模型中选择
model_type = element.attrib.get("model_type", root.attrib.get("model_type", list(
self.model_manager.available_tts_model)[0]))
if model_type is None:
raise ValueError(f"None model_type was specified")
else:
model_type = model_type.upper()
# logging.debug(f"Default model:{list(self.model_manager.available_tts_model)[0]}")
# id = int(element.attrib.get("id", root.attrib.get("id", default_parameter.id)))
# lang = element.attrib.get("lang", root.attrib.get("lang", default_parameter.lang))
# length = float(
# element.attrib.get("length", root.attrib.get("length", default_parameter.length)))
# noise = float(
# element.attrib.get("noise", root.attrib.get("noise", default_parameter.noise)))
# noisew = float(
# element.attrib.get("noisew", root.attrib.get("noisew", default_parameter.noisew)))
# segment_size = int(element.attrib.get("segment_size", root.attrib.get("segment_size",
# config["default_parameter"][
# "segment_size"])))
# emotion = int(element.attrib.get("emotion", root.attrib.get("emotion", 0)))
# Bert-VITS2的参数
# sdp_ratio = int(element.attrib.get("sdp_ratio", root.attrib.get("sdp_ratio",
# config["default_parameter"][
# "sdp_ratio"])))
voice_element = ET.tostring(element, encoding='unicode')
pattern_voice = r'<voice.*?>(.*?)</voice>'
pattern_break = r'<break\s*?(.*?)\s*?/>'
matches_voice = re.findall(pattern_voice, voice_element)[0]
matches_break = re.split(pattern_break, matches_voice)
for match in matches_break:
strength = re.search(r'\s*strength\s*=\s*[\'\"](.*?)[\'\"]', match)
time = re.search(r'\s*time\s*=\s*[\'\"](.*?)[\'\"]', match)
# break标签 strength属性
if strength:
brk = strength_dict[strength.group(1)]
voice_tasks.append({"break": brk})
brk_count += 1
# break标签 time属性
elif time:
brk = self.convert_time_string(time.group(1))
voice_tasks.append({"break": brk})
brk_count += 1
# break标签 为空说明只写了break,默认停顿0.75s
elif match == "":
voice_tasks.append({"break": 0.75})
brk_count += 1
# voice标签中除了break剩下的就是文本
else:
task = {
"model_type": model_type,
"speaker_lang": self.speaker_lang,
"text": match,
}
try:
task.update(params.get(model_type)) # 默认参数
except Exception as e:
raise ValueError(f"Invalid model_type:{model_type}")
task.update(root.attrib)
task.update(element.attrib) # 所有参数都放进去,推理函数会选出需要的参数
task = self.normalize(task)
voice_tasks.append(task)
# 分段末尾停顿0.75s
voice_tasks.append({"break": 0.75})
elif element.tag == "break":
# brk_count大于0说明voice标签中有break
if brk_count > 0:
brk_count -= 1
continue
brk = strength_dict.get(element.attrib.get("strength"),
self.convert_time_string(element.attrib.get("time", "750ms")))
voice_tasks.append({"break": brk})
# for i in voice_tasks:
# self.logger.debug(i)
return voice_tasks, format
def process_ssml_infer_task(self, tasks, format):
audios = []
sampling_rates = []
last_sampling_rate = 22050
for task in tasks:
if task.get("break"):
audios.append(np.zeros(int(task.get("break") * last_sampling_rate), dtype=np.int16))
sampling_rates.append(last_sampling_rate)
else:
model_type_str = task.get("model_type").upper()
if model_type_str not in [ModelType.VITS, ModelType.W2V2_VITS, ModelType.BERT_VITS2,
ModelType.GPT_SOVITS]:
raise ValueError(f"Unsupported model type: {task.get('model_type')}")
model_type = ModelType(model_type_str)
model = self.get_model(model_type, task.get("id"))
sampling_rates.append(model.sampling_rate)
last_sampling_rate = model.sampling_rate
# self.logger.debug(model, model.sampling_rate, task)
audio = self.infer_map[model_type](task, encode=False)
audios.append(audio)
# 得到最高的采样率
target_sr = max(sampling_rates)
# 所有音频要与最高采样率保持一致
resampled_audios = [self.resample_audio(audio, sr, target_sr) for audio, sr in zip(audios, sampling_rates)]
audio = np.concatenate(resampled_audios, axis=0)
encoded_audio = self.encode(target_sr, audio, format)
return encoded_audio
def vits_infer(self, state, encode=True):
model = self.get_model(ModelType.VITS, state["id"])
if model.dynamic_loading:
model.load_model()
state["id"] = self.get_real_id(ModelType.VITS, state["id"]) # Change to real id
# 去除所有多余的空白字符
if state["text"] is not None:
state["text"] = re.sub(r'\s+', ' ', state["text"]).strip()
sampling_rate = model.sampling_rate
sentences_list = sentence_split_and_markup(
text=state["text"],
target_language=state["lang"],
segment_size=state["segment_size"],
speaker_lang=state["speaker_lang"],
)
# 停顿0.5s,避免语音分段合成再拼接后的连接突兀
brk = np.zeros(int(0.5 * sampling_rate), dtype=np.int16)
audios = []
sentences_num = len(sentences_list)
for i, sentence in enumerate(sentences_list):
sentence_audio = model.infer(sentence, state["id"], state["noise"], state["noisew"], state["length"])
audios.append(sentence_audio)
if i < sentences_num - 1:
audios.append(brk)
audio = np.concatenate(audios, axis=0)
if model.dynamic_loading:
model.release_model()
return self.encode(sampling_rate, audio, state["format"]) if encode else audio
def stream_vits_infer(self, state, fname=None):
model = self.get_model(ModelType.VITS, state["id"])
state["id"] = self.get_real_id(ModelType.VITS, state["id"])
# 去除所有多余的空白字符
if state["text"] is not None:
state["text"] = re.sub(r'\s+', ' ', state["text"]).strip()
sampling_rate = model.sampling_rate
sentences_list = sentence_split_and_markup(state["text"], state["lang"], state["segment_size"])
# 停顿0.5s,避免语音分段合成再拼接后的连接突兀
brk = np.zeros(int(0.5 * sampling_rate), dtype=np.int16)
sentences_num = len(sentences_list)
for i, sentence in enumerate(sentences_list):
sentence_audio = model.infer(sentence, state["id"], state["noise"], state["noisew"], state["length"])
audios = []
audios.append(sentence_audio)
if i < sentences_num - 1:
audios.append(brk)
audio = np.concatenate(audios, axis=0)
encoded_audio = self.encode(sampling_rate, audio, state["format"])
for encoded_audio_chunk in self.generate_audio_chunks(encoded_audio):
yield encoded_audio_chunk
def hubert_vits_infer(self, state, encode=True):
model = self.get_model(ModelType.HUBERT_VITS, state["id"])
state["id"] = self.get_real_id(ModelType.HUBERT_VITS, state["id"])
sampling_rate = model.sampling_rate
audio = model.infer(state["audio_path"], state["id"], state["noise"], state["noisew"], state["length"],
f0_scale=1)
return self.encode(sampling_rate, audio, state["format"]) if encode else audio
def w2v2_vits_infer(self, state, encode=True):
model = self.get_model(ModelType.W2V2_VITS, state["id"])
state["id"] = self.get_real_id(ModelType.W2V2_VITS, state["id"])
# 去除所有多余的空白字符
if state["text"] is not None:
state["text"] = re.sub(r'\s+', ' ', state["text"]).strip()
emotion = state["emotion_reference"] if state["emotion_reference"] is not None else state["emotion"]
sampling_rate = model.sampling_rate
sentences_list = sentence_split_and_markup(
text=state["text"],
target_language=state["lang"],
segment_size=state["segment_size"],
speaker_lang=model.lang
)
# 停顿0.5s,避免语音分段合成再拼接后的连接突兀
brk = np.zeros(int(0.5 * sampling_rate), dtype=np.int16)
audios = []
sentences_num = len(sentences_list)
for i, sentence in enumerate(sentences_list):
sentence_audio = model.infer(sentence, state["id"], state["noise"], state["noisew"], state["length"],
emotion)
audios.append(sentence_audio)
if i < sentences_num - 1:
audios.append(brk)
audio = np.concatenate(audios, axis=0)
return self.encode(sampling_rate, audio, state["format"]) if encode else audio
def vits_voice_conversion(self, state, encode=True):
original_model_id = int(self.get_model_id(ModelType.VITS, state["original_id"]))
target_model_id = int(self.get_model_id(ModelType.VITS, state["target_id"]))
if original_model_id != target_model_id:
raise ValueError(f"speakers are in diffrent VITS Model")
model = self.get_model(ModelType.VITS, state["original_id"])
state["original_id"] = int(self.get_real_id(ModelType.VITS, state["original_id"]))
state["target_id"] = int(self.get_real_id(ModelType.VITS, state["target_id"]))
sampling_rate = model.sampling_rate
audio = model.voice_conversion(state["audio_path"], state["original_id"], state["target_id"])
return self.encode(sampling_rate, audio, state["format"]) if encode else audio
def get_dimensional_emotion_npy(self, audio):
audio16000, sampling_rate = librosa.load(audio, sr=16000, mono=True)
emotion = self.dimensional_emotion_model(audio16000, sampling_rate)['hidden_states']
emotion_npy = BytesIO()
np.save(emotion_npy, emotion.squeeze(0))
emotion_npy.seek(0)
return emotion_npy
def bert_vits2_infer(self, state, encode=True):
model = self.get_model(model_type=ModelType.BERT_VITS2, id=state["id"], speaker=state["speaker"])
state["id"] = self.get_real_id(model_type=ModelType.BERT_VITS2, id=state["id"], speaker=state["speaker"])
# 去除所有多余的空白字符
if state["text"] is not None:
state["text"] = re.sub(r'\s+', ' ', state["text"]).strip()
sampling_rate = model.sampling_rate
sentences_list = sentence_split(state["text"], state["segment_size"])
if model.zh_bert_extra:
infer_func = model.infer
state["lang"] = ["zh"]
elif model.ja_bert_extra:
infer_func = model.infer
state["lang"] = ["ja"]
elif len(state["lang"]) == 1 and "auto" not in state["lang"]:
infer_func = model.infer
else:
infer_func = model.infer_multilang
audios = []
for sentences in sentences_list:
state["text"] = sentences
audio = infer_func(**state)
audios.append(audio)
audio = np.concatenate(audios)
return self.encode(sampling_rate, audio, state["format"]) if encode else audio
def stream_bert_vits2_infer(self, state, encode=True):
model = self.get_model(model_type=ModelType.BERT_VITS2, id=state["id"], speaker=state["speaker"])
state["id"] = self.get_real_id(model_type=ModelType.BERT_VITS2, id=state["id"], speaker=state["speaker"])
# 去除所有多余的空白字符
if state["text"] is not None:
state["text"] = re.sub(r'\s+', ' ', state["text"]).strip()
sampling_rate = model.sampling_rate
sentences_list = sentence_split(state["text"], state["segment_size"])
if len(state["lang"]) == 1 and "auto" not in state["lang"]:
infer_func = model.infer
else:
infer_func = model.infer_multilang
for sentences in sentences_list:
state["text"] = sentences
audio = infer_func(**state)
process_audio = self.encode(sampling_rate, audio, state["format"]) if encode else audio
for audio_chunk in self.generate_audio_chunks(process_audio):
yield audio_chunk
def _set_reference_audio(self, state):
# 检查参考音频
if check_is_none(state.get("reference_audio")): # 无参考音频
# 未选择预设
if check_is_none(state.get("preset")):
presets = config.gpt_sovits_config.presets
refer_preset = presets.get(next(iter(presets)))
else: # 已选择预设
refer_preset = config.gpt_sovits_config.presets.get(state.get("preset"))
if refer_preset is None:
raise ValueError(f"Error preset:{state.get('preset')}")
refer_wav_path = refer_preset.refer_wav_path
if check_is_none(refer_wav_path):
raise ValueError(f"The refer_wav_path:{refer_wav_path} in preset:{state.get('preset')} is None!")
refer_wav_path = os.path.join(BASE_DIR, config.system.data_path, refer_wav_path)
state["prompt_text"], state["prompt_lang"] = refer_preset.prompt_text, refer_preset.prompt_lang
# 将reference_audio换成指定预设里的参考音频
state["reference_audio"] = refer_wav_path
if check_is_none(state.get("prompt_lang")):
presets = config.gpt_sovits_config.presets
state["prompt_lang"] = presets.get(next(iter(presets)), "auto")
state["reference_audio"], state["reference_audio_sr"] = librosa.load(state["reference_audio"], sr=None,
dtype=np.float32)
state["reference_audio"] = state["reference_audio"].flatten()
return state
def gpt_sovits_infer(self, state, encode=True):
model = self.get_model(ModelType.GPT_SOVITS, state["id"])
state = self._set_reference_audio(state)
audio = next(model.infer(**state))
sampling_rate = model.sampling_rate
return self.encode(sampling_rate, audio, state["format"]) if encode else audio
def stream_gpt_sovits_infer(self, state, encode=True):
model = self.get_model(ModelType.GPT_SOVITS, state["id"])
state = self._set_reference_audio(state)
state["return_fragment"] = True
audio_generater = model.infer(**state)
for audio in audio_generater:
sampling_rate = model.sampling_rate
process_audio = self.encode(sampling_rate, audio, state["format"]) if encode else audio
for encoded_audio_chunk in self.generate_audio_chunks(process_audio):
yield encoded_audio_chunk
def reading(self, in_state, nr_state):
in_model = self.get_model(in_state["model_type"], in_state["id"])
nr_model = self.get_model(nr_state["model_type"], nr_state["id"])
infer_func = {ModelType.VITS: self.vits_infer,
ModelType.W2V2_VITS: self.w2v2_vits_infer,
ModelType.BERT_VITS2: self.bert_vits2_infer,
ModelType.GPT_SOVITS: self.gpt_sovits_infer
}
sentences_list = sentence_split_reading(in_state["text"])
audios = []
sampling_rates = []
for sentence, is_quote in sentences_list:
try:
if is_quote:
in_state["text"] = sentence
audio = infer_func[in_state["model_type"]](in_state, encode=False)
sampling_rates.append(in_model.sampling_rate)
else:
nr_state["text"] = sentence
audio = infer_func[nr_state["model_type"]](nr_state, encode=False)
sampling_rates.append(nr_model.sampling_rate)
audios.append(audio)
except Exception as e:
logging.error(traceback.print_exc())
logging.error(e)
# 得到最高的采样率
target_sr = max(sampling_rates)
# 所有音频要与最高采样率保持一致
resampled_audios = [self.resample_audio(audio, sr, target_sr) for audio, sr in
zip(audios, sampling_rates)]
audio = np.concatenate(resampled_audios, axis=0)
encoded_audio = self.encode(target_sr, audio, in_state["format"])
return encoded_audio
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