MeloTTS / python /melotts.py
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Initial this repo, and update the axmodel of AX650 platform
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import os
os.environ["HF_ENDPOINT"] = "https://hf-mirror.com"
import numpy as np
import soundfile
import onnxruntime as ort
import axengine as axe
import argparse
import time
from split_utils import split_sentence
from text import cleaned_text_to_sequence
from text.cleaner import clean_text
from symbols import LANG_TO_SYMBOL_MAP
import re
def intersperse(lst, item):
result = [item] * (len(lst) * 2 + 1)
result[1::2] = lst
return result
def get_text_for_tts_infer(text, language_str, symbol_to_id=None):
norm_text, phone, tone, word2ph = clean_text(text, language_str)
phone, tone, language = cleaned_text_to_sequence(phone, tone, language_str, symbol_to_id)
phone = intersperse(phone, 0)
tone = intersperse(tone, 0)
language = intersperse(language, 0)
phone = np.array(phone, dtype=np.int32)
tone = np.array(tone, dtype=np.int32)
language = np.array(language, dtype=np.int32)
word2ph = np.array(word2ph, dtype=np.int32) * 2
word2ph[0] += 1
return phone, tone, language, norm_text, word2ph
def split_sentences_into_pieces(text, language, quiet=False):
texts = split_sentence(text, language_str=language)
if not quiet:
print(" > Text split to sentences.")
print('\n'.join(texts))
print(" > ===========================")
return texts
def get_args():
parser = argparse.ArgumentParser(
prog="melotts",
description="Run TTS on input sentence"
)
parser.add_argument("--sentence", "-s", type=str, required=False, default="爱芯元智半导体股份有限公司,致力于打造世界领先的人工智能感知与边缘计算芯片。服务智慧城市、智能驾驶、机器人的海量普惠的应用")
parser.add_argument("--wav", "-w", type=str, required=False, default="output.wav")
parser.add_argument("--encoder", "-e", type=str, required=False, default=None)
parser.add_argument("--decoder", "-d", type=str, required=False, default=None)
parser.add_argument("--dec_len", type=int, default=128)
parser.add_argument("--sample_rate", "-sr", type=int, required=False, default=44100)
parser.add_argument("--speed", type=float, required=False, default=0.8)
parser.add_argument("--language", "-l", type=str,
choices=["ZH", "ZH_MIX_EN", "JP", "EN", 'KR', "ES", "SP","FR"], required=False, default="ZH_MIX_EN")
return parser.parse_args()
def audio_numpy_concat(segment_data_list, sr, speed=1.):
audio_segments = []
for segment_data in segment_data_list:
audio_segments += segment_data.reshape(-1).tolist()
audio_segments += [0] * int((sr * 0.05) / speed)
audio_segments = np.array(audio_segments).astype(np.float32)
return audio_segments
def merge_sub_audio(sub_audio_list, pad_size, audio_len):
# Average pad part
if pad_size > 0:
for i in range(len(sub_audio_list) - 1):
sub_audio_list[i][-pad_size:] += sub_audio_list[i+1][:pad_size]
sub_audio_list[i][-pad_size:] /= 2
if i > 0:
sub_audio_list[i] = sub_audio_list[i][pad_size:]
sub_audio = np.concatenate(sub_audio_list, axis=-1)
return sub_audio[:audio_len]
# 计算每个词的发音时长
def calc_word2pronoun(word2ph, pronoun_lens):
indice = [0]
for ph in word2ph[:-1]:
indice.append(indice[-1] + ph)
word2pronoun = []
for i, ph in zip(indice, word2ph):
word2pronoun.append(np.sum(pronoun_lens[i : i + ph]))
return word2pronoun
# 生成有overlap的slice,slice索引是对于zp的
def generate_slices(word2pronoun, dec_len):
pn_start, pn_end = 0, 0
zp_start, zp_end = 0, 0
zp_len = 0
pn_slices = []
zp_slices = []
while pn_end < len(word2pronoun):
# 前一个slice长度大于2 且 加上现在这个字没有超过dec_len,则往前overlap两个字
if pn_end - pn_start > 2 and np.sum(word2pronoun[pn_end - 2 : pn_end + 1]) <= dec_len:
zp_len = np.sum(word2pronoun[pn_end - 2 : pn_end])
zp_start = zp_end - zp_len
pn_start = pn_end - 2
else:
zp_len = 0
zp_start = zp_end
pn_start = pn_end
while pn_end < len(word2pronoun) and zp_len + word2pronoun[pn_end] <= dec_len:
zp_len += word2pronoun[pn_end]
pn_end += 1
zp_end = zp_start + zp_len
pn_slices.append(slice(pn_start, pn_end))
zp_slices.append(slice(zp_start, zp_end))
return pn_slices, zp_slices
def main():
args = get_args()
sentence = args.sentence
sample_rate = args.sample_rate
enc_model = args.encoder # default="../models/encoder.onnx"
dec_model = args.decoder # default="../models/decoder.axmodel"
language = args.language # default: ZH_MIX_EN
dec_len = args.dec_len # default: 128
if language == "ZH":
language = "ZH_MIX_EN"
if enc_model is None:
if "ZH" in language:
enc_model = "../models/encoder-zh.onnx"
else:
enc_model = f"../models/encoder-{language.lower()}.onnx"
assert os.path.exists(enc_model), f"Encoder model ({enc_model}) not exist!"
if dec_model is None:
if "ZH" in language:
dec_model = "../models/decoder-zh.axmodel"
else:
dec_model = f"../models/decoder-{language.lower()}.axmodel"
assert os.path.exists(dec_model), f"Decoder model ({dec_model}) not exist!"
print(f"sentence: {sentence}")
print(f"sample_rate: {sample_rate}")
print(f"encoder: {enc_model}")
print(f"decoder: {dec_model}")
print(f"language: {language}")
_symbol_to_id = {s: i for i, s in enumerate(LANG_TO_SYMBOL_MAP[language])}
# Split sentence
start = time.time()
sens = split_sentences_into_pieces(sentence, language, quiet=False)
print(f"split_sentences_into_pieces take {1000 * (time.time() - start)}ms")
# Load models
start = time.time()
sess_enc = ort.InferenceSession(enc_model, providers=["CPUExecutionProvider"], sess_options=ort.SessionOptions())
sess_dec = axe.InferenceSession(dec_model)
print(f"load models take {1000 * (time.time() - start)}ms")
# Load static input
g = np.fromfile(f"../models/g-{language.lower()}.bin", dtype=np.float32).reshape(1, 256, 1)
# Final wav
audio_list = []
# Iterate over splitted sentences
for n, se in enumerate(sens):
if language in ['EN', 'ZH_MIX_EN']:
se = re.sub(r'([a-z])([A-Z])', r'\1 \2', se)
print(f"\nSentence[{n}]: {se}")
# Convert sentence to phones and tones
phones, tones, lang_ids, norm_text, word2ph = get_text_for_tts_infer(se, language, symbol_to_id=_symbol_to_id)
start = time.time()
# Run encoder
z_p, pronoun_lens, audio_len = sess_enc.run(None, input_feed={
'phone': phones, 'g': g,
'tone': tones, 'language': lang_ids,
'noise_scale': np.array([0], dtype=np.float32),
'length_scale': np.array([1.0 / args.speed], dtype=np.float32),
'noise_scale_w': np.array([0], dtype=np.float32),
'sdp_ratio': np.array([0], dtype=np.float32)})
print(f"encoder run take {1000 * (time.time() - start):.2f}ms")
# 计算每个词的发音长度
word2pronoun = calc_word2pronoun(word2ph, pronoun_lens)
# 生成word2pronoun和zp的切片
pn_slices, zp_slices = generate_slices(word2pronoun, dec_len)
audio_len = audio_len[0]
sub_audio_list = []
for i, (ps, zs) in enumerate(zip(pn_slices, zp_slices)):
zp_slice = z_p[..., zs]
# Padding前zp的长度
sub_dec_len = zp_slice.shape[-1]
# Padding前输出音频的长度
sub_audio_len = 512 * sub_dec_len
# Padding到dec_len
if zp_slice.shape[-1] < dec_len:
zp_slice = np.concatenate((zp_slice, np.zeros((*zp_slice.shape[:-1], dec_len - zp_slice.shape[-1]), dtype=np.float32)), axis=-1)
start = time.time()
audio = sess_dec.run(None, input_feed={"z_p": zp_slice,
"g": g
})[0].flatten()
# 处理overlap
audio_start = 0
if len(sub_audio_list) > 0:
if pn_slices[i - 1].stop > ps.start:
# 去掉第一个字
audio_start = 512 * word2pronoun[ps.start]
audio_end = sub_audio_len
if i < len(pn_slices) - 1:
if ps.stop > pn_slices[i + 1].start:
# 去掉最后一个字
audio_end = sub_audio_len - 512 * word2pronoun[ps.stop - 1]
audio = audio[audio_start:audio_end]
print(f"Decode slice[{i}]: decoder run take {1000 * (time.time() - start):.2f}ms")
sub_audio_list.append(audio)
sub_audio = merge_sub_audio(sub_audio_list, 0, audio_len)
audio_list.append(sub_audio)
audio = audio_numpy_concat(audio_list, sr=sample_rate, speed=args.speed)
soundfile.write(args.wav, audio, sample_rate)
print(f"Save to {args.wav}")
if __name__ == "__main__":
main()