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import os | |
import librosa | |
import re | |
import numpy as np | |
import torch | |
import xml.etree.ElementTree as ET | |
import config | |
import soundfile as sf | |
from io import BytesIO | |
from graiax import silkcoder | |
from logger import logger | |
from contants import ModelType | |
from scipy.signal import resample_poly | |
# torch.set_num_threads(1) # 设置torch线程为1 | |
class TTS: | |
def __init__(self, voice_obj, voice_speakers, **kwargs): | |
self._voice_obj = voice_obj | |
self._voice_speakers = voice_speakers | |
self._strength_dict = {"x-weak": 0.25, "weak": 0.5, "Medium": 0.75, "Strong": 1, "x-strong": 1.25} | |
self._speakers_count = sum([len(self._voice_speakers[i]) for i in self._voice_speakers]) | |
self._vits_speakers_count = len(self._voice_speakers[ModelType.VITS.value]) | |
self._hubert_speakers_count = len(self._voice_speakers[ModelType.HUBERT_VITS.value]) | |
self._w2v2_speakers_count = len(self._voice_speakers[ModelType.W2V2_VITS.value]) | |
self._w2v2_emotion_count = kwargs.get("w2v2_emotion_count", 0) | |
self._bert_vits2_speakers_count = len(self._voice_speakers[ModelType.BERT_VITS2.value]) | |
self.dem = None | |
# Initialization information | |
self.logger = logger | |
self.logger.info(f"torch:{torch.__version__} cuda_available:{torch.cuda.is_available()}") | |
self.logger.info(f'device:{kwargs.get("device")} device.type:{kwargs.get("device").type}') | |
if getattr(config, "DIMENSIONAL_EMOTION_MODEL", None) != None: | |
try: | |
import audonnx | |
root = os.path.dirname(config.DIMENSIONAL_EMOTION_MODEL) | |
model_file = config.DIMENSIONAL_EMOTION_MODEL | |
self.dem = audonnx.load(root=root, model_file=model_file) | |
except Exception as e: | |
self.logger.warning(f"Load DIMENSIONAL_EMOTION_MODEL failed {e}") | |
if self._vits_speakers_count != 0: self.logger.info(f"[{ModelType.VITS.value}] {self._vits_speakers_count} speakers") | |
if self._hubert_speakers_count != 0: self.logger.info(f"[{ModelType.HUBERT_VITS.value}] {self._hubert_speakers_count} speakers") | |
if self._w2v2_speakers_count != 0: self.logger.info(f"[{ModelType.W2V2_VITS.value}] {self._w2v2_speakers_count} speakers") | |
if self._bert_vits2_speakers_count != 0: self.logger.info( | |
f"[{ModelType.BERT_VITS2.value}] {self._bert_vits2_speakers_count} speakers") | |
self.logger.info(f"{self._speakers_count} speakers in total.") | |
if self._speakers_count == 0: | |
self.logger.warning(f"No model was loaded.") | |
def voice_speakers(self): | |
return self._voice_speakers | |
def speakers_count(self): | |
return self._speakers_count | |
def vits_speakers_count(self): | |
return self._vits_speakers_count | |
def hubert_speakers_count(self): | |
return self._hubert_speakers_count | |
def w2v2_speakers_count(self): | |
return self._w2v2_speakers_count | |
def w2v2_emotion_count(self): | |
return self._w2v2_emotion_count | |
def bert_vits2_speakers_count(self): | |
return self._bert_vits2_speakers_count | |
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 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} | |
for element in root.iter(): | |
if element.tag == "voice": | |
id = int(element.attrib.get("id", root.attrib.get("id", config.ID))) | |
lang = element.attrib.get("lang", root.attrib.get("lang", config.LANG)) | |
length = float(element.attrib.get("length", root.attrib.get("length", config.LENGTH))) | |
noise = float(element.attrib.get("noise", root.attrib.get("noise", config.NOISE))) | |
noisew = float(element.attrib.get("noisew", root.attrib.get("noisew", config.NOISEW))) | |
max = int(element.attrib.get("max", root.attrib.get("max", "0"))) | |
# 不填写默认就是vits | |
model_type = element.attrib.get("model_type", root.attrib.get("model_type", "vits")) | |
# w2v2-vits/emotion-vits才有emotion | |
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.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: | |
voice_tasks.append({"id": id, | |
"text": match, | |
"lang": lang, | |
"length": length, | |
"noise": noise, | |
"noisew": noisew, | |
"max": max, | |
"model_type": model_type, | |
"emotion": emotion, | |
"sdp_ratio": sdp_ratio | |
}) | |
# 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") * 22050), 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.value, ModelType.W2V2_VITS.value, ModelType.BERT_VITS2.value]: | |
raise ValueError(f"Unsupported model type: {task.get('model_type')}") | |
model_type = ModelType(model_type_str) | |
voice_obj = self._voice_obj[model_type][task.get("id")][1] | |
real_id = self._voice_obj[model_type][task.get("id")][0] | |
task["id"] = real_id | |
sampling_rates.append(voice_obj.sampling_rate) | |
last_sampling_rate = voice_obj.sampling_rate | |
audio = voice_obj.get_audio(task) | |
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, task): | |
format = task.get("format", "wav") | |
voice_obj = self._voice_obj[ModelType.VITS][task.get("id")][1] | |
real_id = self._voice_obj[ModelType.VITS][task.get("id")][0] | |
task["id"] = real_id # Change to real id | |
sampling_rate = voice_obj.sampling_rate | |
audio = voice_obj.get_audio(task, auto_break=True) | |
encoded_audio = self.encode(sampling_rate, audio, format) | |
return encoded_audio | |
def stream_vits_infer(self, task, fname=None): | |
format = task.get("format", "wav") | |
voice_obj = self._voice_obj[ModelType.VITS][task.get("id")][1] | |
task["id"] = self._voice_obj[ModelType.VITS][task.get("id")][0] | |
sampling_rate = voice_obj.sampling_rate | |
genertator = voice_obj.get_stream_audio(task, auto_break=True) | |
# audio = BytesIO() | |
for chunk in genertator: | |
encoded_audio = self.encode(sampling_rate, chunk, format) | |
for encoded_audio_chunk in self.generate_audio_chunks(encoded_audio): | |
yield encoded_audio_chunk | |
# if getattr(config, "SAVE_AUDIO", False): | |
# audio.write(encoded_audio.getvalue()) | |
# if getattr(config, "SAVE_AUDIO", False): | |
# path = f"{config.CACHE_PATH}/{fname}" | |
# utils.save_audio(audio.getvalue(), path) | |
def hubert_vits_infer(self, task): | |
format = task.get("format", "wav") | |
voice_obj = self._voice_obj[ModelType.HUBERT_VITS][task.get("id")][1] | |
task["id"] = self._voice_obj[ModelType.HUBERT_VITS][task.get("id")][0] | |
sampling_rate = voice_obj.sampling_rate | |
audio = voice_obj.get_audio(task) | |
encoded_audio = self.encode(sampling_rate, audio, format) | |
return encoded_audio | |
def w2v2_vits_infer(self, task): | |
format = task.get("format", "wav") | |
voice_obj = self._voice_obj[ModelType.W2V2_VITS][task.get("id")][1] | |
task["id"] = self._voice_obj[ModelType.W2V2_VITS][task.get("id")][0] | |
sampling_rate = voice_obj.sampling_rate | |
audio = voice_obj.get_audio(task, auto_break=True) | |
encoded_audio = self.encode(sampling_rate, audio, format) | |
return encoded_audio | |
def vits_voice_conversion(self, task): | |
original_id = task.get("original_id") | |
target_id = task.get("target_id") | |
format = task.get("format") | |
original_id_obj = int(self._voice_obj[ModelType.VITS][original_id][2]) | |
target_id_obj = int(self._voice_obj[ModelType.VITS][target_id][2]) | |
if original_id_obj != target_id_obj: | |
raise ValueError(f"speakers are in diffrent VITS Model") | |
task["original_id"] = int(self._voice_obj[ModelType.VITS][original_id][0]) | |
task["target_id"] = int(self._voice_obj[ModelType.VITS][target_id][0]) | |
voice_obj = self._voice_obj[ModelType.VITS][original_id][1] | |
sampling_rate = voice_obj.sampling_rate | |
audio = voice_obj.voice_conversion(task) | |
encoded_audio = self.encode(sampling_rate, audio, format) | |
return encoded_audio | |
def get_dimensional_emotion_npy(self, audio): | |
if self.dem is None: | |
raise ValueError(f"Please configure DIMENSIONAL_EMOTION_MODEL path in config.py") | |
audio16000, sampling_rate = librosa.load(audio, sr=16000, mono=True) | |
emotion = self.dem(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, task): | |
format = task.get("format", "wav") | |
voice_obj = self._voice_obj[ModelType.BERT_VITS2][task.get("id")][1] | |
task["id"] = self._voice_obj[ModelType.BERT_VITS2][task.get("id")][0] | |
sampling_rate = voice_obj.sampling_rate | |
audio = voice_obj.get_audio(task, auto_break=True) | |
encoded_audio = self.encode(sampling_rate, audio, format) | |
return encoded_audio | |