inference/infer_tool.py

import hashlib
import io
import json
import logging
import os
import time
from pathlib import Path
from inference import slicer

import librosa
import numpy as np
# import onnxruntime
import parselmouth
import soundfile
import torch
import hashlib
import io
import json
import logging
import os
import time
from pathlib import Path
from inference import slicer

import librosa
import numpy as np
# import onnxruntime
import parselmouth
import soundfile
import torch
import torchaudio

import cluster
from hubert import hubert_model
import utils
from models import SynthesizerTrn

logging.getLogger('matplotlib').setLevel(logging.WARNING)


def read_temp(file_name):
    if not os.path.exists(file_name):
        with open(file_name, "w") as f:
            f.write(json.dumps({"info": "temp_dict"}))
        return {}
    else:
        try:
            with open(file_name, "r") as f:
                data = f.read()
            data_dict = json.loads(data)
            if os.path.getsize(file_name) > 50 * 1024 * 1024:
                f_name = file_name.replace("\\", "/").split("/")[-1]
                print(f"clean {f_name}")
                for wav_hash in list(data_dict.keys()):
                    if int(time.time()) - int(data_dict[wav_hash]["time"]) > 14 * 24 * 3600:
                        del data_dict[wav_hash]
        except Exception as e:
            print(e)
            print(f"{file_name} error,auto rebuild file")
            data_dict = {"info": "temp_dict"}
        return data_dict


def write_temp(file_name, data):
    with open(file_name, "w") as f:
        f.write(json.dumps(data))


def timeit(func):
    def run(*args, **kwargs):
        t = time.time()
        res = func(*args, **kwargs)
        print('executing \'%s\' costed %.3fs' % (func.__name__, time.time() - t))
        return res

    return run


def format_wav(audio_path):
    if Path(audio_path).suffix == '.wav':
        return
    raw_audio, raw_sample_rate = librosa.load(audio_path, mono=True, sr=None)
    soundfile.write(Path(audio_path).with_suffix(".wav"), raw_audio, raw_sample_rate)


def get_end_file(dir_path, end):
    file_lists = []
    for root, dirs, files in os.walk(dir_path):
        files = [f for f in files if f[0] != '.']
        dirs[:] = [d for d in dirs if d[0] != '.']
        for f_file in files:
            if f_file.endswith(end):
                file_lists.append(os.path.join(root, f_file).replace("\\", "/"))
    return file_lists


def get_md5(content):
    return hashlib.new("md5", content).hexdigest()

def fill_a_to_b(a, b):
    if len(a) < len(b):
        for _ in range(0, len(b) - len(a)):
            a.append(a[0])

def mkdir(paths: list):
    for path in paths:
        if not os.path.exists(path):
            os.mkdir(path)

def pad_array(arr, target_length):
    current_length = arr.shape[0]
    if current_length >= target_length:
        return arr
    else:
        pad_width = target_length - current_length
        pad_left = pad_width // 2
        pad_right = pad_width - pad_left
        padded_arr = np.pad(arr, (pad_left, pad_right), 'constant', constant_values=(0, 0))
        return padded_arr
    
def split_list_by_n(list_collection, n, pre=0):
    for i in range(0, len(list_collection), n):
        yield list_collection[i-pre if i-pre>=0 else i: i + n]


class F0FilterException(Exception):
    pass

class Svc(object):
    def __init__(self, net_g_path, config_path,
                 device=None,
                 cluster_model_path="logs/44k/kmeans_10000.pt",
                 nsf_hifigan_enhance = False
                 ):
        self.net_g_path = net_g_path
        if device is None:
            self.dev = torch.device("cuda" if torch.cuda.is_available() else "cpu")
        else:
            self.dev = torch.device(device)
        self.net_g_ms = None
        self.hps_ms = utils.get_hparams_from_file(config_path)
        self.target_sample = self.hps_ms.data.sampling_rate
        self.hop_size = self.hps_ms.data.hop_length
        self.spk2id = self.hps_ms.spk
        self.nsf_hifigan_enhance = nsf_hifigan_enhance
        # 加载hubert
        self.hubert_model = utils.get_hubert_model().to(self.dev)
        self.load_model()
        if os.path.exists(cluster_model_path):
            self.cluster_model = cluster.get_cluster_model(cluster_model_path)
        if self.nsf_hifigan_enhance:
            from modules.enhancer import Enhancer
            self.enhancer = Enhancer('nsf-hifigan', 'pretrain/nsf_hifigan/model',device=self.dev)

    def load_model(self):
        # 获取模型配置
        self.net_g_ms = SynthesizerTrn(
            self.hps_ms.data.filter_length // 2 + 1,
            self.hps_ms.train.segment_size // self.hps_ms.data.hop_length,
            **self.hps_ms.model)
        _ = utils.load_checkpoint(self.net_g_path, self.net_g_ms, None)
        if "half" in self.net_g_path and torch.cuda.is_available():
            _ = self.net_g_ms.half().eval().to(self.dev)
        else:
            _ = self.net_g_ms.eval().to(self.dev)



    def get_unit_f0(self, in_path, tran, cluster_infer_ratio, speaker, f0_filter ,F0_mean_pooling):

        wav, sr = librosa.load(in_path, sr=self.target_sample)

        if F0_mean_pooling == True:
            f0, uv = utils.compute_f0_uv_torchcrepe(torch.FloatTensor(wav), sampling_rate=self.target_sample, hop_length=self.hop_size,device=self.dev)
            if f0_filter and sum(f0) == 0:
                raise F0FilterException("未检测到人声")
            f0 = torch.FloatTensor(list(f0))
            uv = torch.FloatTensor(list(uv))
        if F0_mean_pooling == False:
            f0 = utils.compute_f0_parselmouth(wav, sampling_rate=self.target_sample, hop_length=self.hop_size)
            if f0_filter and sum(f0) == 0:
                raise F0FilterException("未检测到人声")
            f0, uv = utils.interpolate_f0(f0)
            f0 = torch.FloatTensor(f0)
            uv = torch.FloatTensor(uv)

        f0 = f0 * 2 ** (tran / 12)
        f0 = f0.unsqueeze(0).to(self.dev)
        uv = uv.unsqueeze(0).to(self.dev)

        wav16k = librosa.resample(wav, orig_sr=self.target_sample, target_sr=16000)
        wav16k = torch.from_numpy(wav16k).to(self.dev)
        c = utils.get_hubert_content(self.hubert_model, wav_16k_tensor=wav16k)
        c = utils.repeat_expand_2d(c.squeeze(0), f0.shape[1])

        if cluster_infer_ratio !=0:
            cluster_c = cluster.get_cluster_center_result(self.cluster_model, c.cpu().numpy().T, speaker).T
            cluster_c = torch.FloatTensor(cluster_c).to(self.dev)
            c = cluster_infer_ratio * cluster_c + (1 - cluster_infer_ratio) * c

        c = c.unsqueeze(0)
        return c, f0, uv

    def infer(self, speaker, tran, raw_path,
              cluster_infer_ratio=0,
              auto_predict_f0=False,
              noice_scale=0.4,
              f0_filter=False,
              F0_mean_pooling=False,
              enhancer_adaptive_key = 0
              ):

        speaker_id = self.spk2id.__dict__.get(speaker)
        if not speaker_id and type(speaker) is int:
            if len(self.spk2id.__dict__) >= speaker:
                speaker_id = speaker
        sid = torch.LongTensor([int(speaker_id)]).to(self.dev).unsqueeze(0)
        c, f0, uv = self.get_unit_f0(raw_path, tran, cluster_infer_ratio, speaker, f0_filter,F0_mean_pooling)
        if "half" in self.net_g_path and torch.cuda.is_available():
            c = c.half()
        with torch.no_grad():
            start = time.time()
            audio = self.net_g_ms.infer(c, f0=f0, g=sid, uv=uv, predict_f0=auto_predict_f0, noice_scale=noice_scale)[0,0].data.float()
            if self.nsf_hifigan_enhance:
                audio, _ = self.enhancer.enhance(
                                                                        audio[None,:], 
                                                                        self.target_sample, 
                                                                        f0[:,:,None], 
                                                                        self.hps_ms.data.hop_length, 
                                                                        adaptive_key = enhancer_adaptive_key)
            use_time = time.time() - start
            print("vits use time:{}".format(use_time))
        return audio, audio.shape[-1]

    def clear_empty(self):
        # 清理显存
        torch.cuda.empty_cache()

    def slice_inference(self,
                        raw_audio_path,
                        spk,
                        tran,
                        slice_db,
                        cluster_infer_ratio,
                        auto_predict_f0,
                        noice_scale,
                        pad_seconds=0.5,
                        clip_seconds=0,
                        lg_num=0,
                        lgr_num =0.75,
                        F0_mean_pooling = False,
                        enhancer_adaptive_key = 0
                        ):
        wav_path = raw_audio_path
        chunks = slicer.cut(wav_path, db_thresh=slice_db)
        audio_data, audio_sr = slicer.chunks2audio(wav_path, chunks)
        per_size = int(clip_seconds*audio_sr)
        lg_size = int(lg_num*audio_sr)
        lg_size_r = int(lg_size*lgr_num)
        lg_size_c_l = (lg_size-lg_size_r)//2
        lg_size_c_r = lg_size-lg_size_r-lg_size_c_l
        lg = np.linspace(0,1,lg_size_r) if lg_size!=0 else 0
        
        audio = []
        for (slice_tag, data) in audio_data:
            print(f'#=====segment start, {round(len(data) / audio_sr, 3)}s======')
            # padd
            length = int(np.ceil(len(data) / audio_sr * self.target_sample))
            if slice_tag:
                print('jump empty segment')
                _audio = np.zeros(length)
                audio.extend(list(pad_array(_audio, length)))
                continue
            if per_size != 0:
                datas = split_list_by_n(data, per_size,lg_size)
            else:
                datas = [data]
            for k,dat in enumerate(datas):
                per_length = int(np.ceil(len(dat) / audio_sr * self.target_sample)) if clip_seconds!=0 else length
                if clip_seconds!=0: print(f'###=====segment clip start, {round(len(dat) / audio_sr, 3)}s======')
                # padd
                pad_len = int(audio_sr * pad_seconds)
                dat = np.concatenate([np.zeros([pad_len]), dat, np.zeros([pad_len])])
                raw_path = io.BytesIO()
                soundfile.write(raw_path, dat, audio_sr, format="wav")
                raw_path.seek(0)
                out_audio, out_sr = self.infer(spk, tran, raw_path,
                                                    cluster_infer_ratio=cluster_infer_ratio,
                                                    auto_predict_f0=auto_predict_f0,
                                                    noice_scale=noice_scale,
                                                    F0_mean_pooling = F0_mean_pooling,
                                                    enhancer_adaptive_key = enhancer_adaptive_key
                                                    )
                _audio = out_audio.cpu().numpy()
                pad_len = int(self.target_sample * pad_seconds)
                _audio = _audio[pad_len:-pad_len]
                _audio = pad_array(_audio, per_length)
                if lg_size!=0 and k!=0:
                    lg1 = audio[-(lg_size_r+lg_size_c_r):-lg_size_c_r] if lgr_num != 1 else audio[-lg_size:]
                    lg2 = _audio[lg_size_c_l:lg_size_c_l+lg_size_r]  if lgr_num != 1 else _audio[0:lg_size]
                    lg_pre = lg1*(1-lg)+lg2*lg
                    audio = audio[0:-(lg_size_r+lg_size_c_r)] if lgr_num != 1 else audio[0:-lg_size]
                    audio.extend(lg_pre)
                    _audio = _audio[lg_size_c_l+lg_size_r:] if lgr_num != 1 else _audio[lg_size:]
                audio.extend(list(_audio))
        return np.array(audio)

class RealTimeVC:
    def __init__(self):
        self.last_chunk = None
        self.last_o = None
        self.chunk_len = 16000  # 区块长度
        self.pre_len = 3840  # 交叉淡化长度，640的倍数

    """输入输出都是1维numpy 音频波形数组"""

    def process(self, svc_model, speaker_id, f_pitch_change, input_wav_path,
                cluster_infer_ratio=0,
                auto_predict_f0=False,
                noice_scale=0.4,
                f0_filter=False):

        import maad
        audio, sr = torchaudio.load(input_wav_path)
        audio = audio.cpu().numpy()[0]
        temp_wav = io.BytesIO()
        if self.last_chunk is None:
            input_wav_path.seek(0)

            audio, sr = svc_model.infer(speaker_id, f_pitch_change, input_wav_path,
                                        cluster_infer_ratio=cluster_infer_ratio,
                                        auto_predict_f0=auto_predict_f0,
                                        noice_scale=noice_scale,
                                        f0_filter=f0_filter)

            audio = audio.cpu().numpy()
            self.last_chunk = audio[-self.pre_len:]
            self.last_o = audio
            return audio[-self.chunk_len:]
        else:
            audio = np.concatenate([self.last_chunk, audio])
            soundfile.write(temp_wav, audio, sr, format="wav")
            temp_wav.seek(0)

            audio, sr = svc_model.infer(speaker_id, f_pitch_change, temp_wav,
                                        cluster_infer_ratio=cluster_infer_ratio,
                                        auto_predict_f0=auto_predict_f0,
                                        noice_scale=noice_scale,
                                        f0_filter=f0_filter)

            audio = audio.cpu().numpy()
            ret = maad.util.crossfade(self.last_o, audio, self.pre_len)
            self.last_chunk = audio[-self.pre_len:]
            self.last_o = audio
            return ret[self.chunk_len:2 * self.chunk_len]