import math import os os.environ["LRU_CACHE_CAPACITY"] = "3" import random import paddle import numpy as np import librosa from librosa.util import normalize from librosa.filters import mel as librosa_mel_fn from scipy.io.wavfile import read import soundfile as sf def load_wav_to_torch(full_path, target_sr=None, return_empty_on_exception=False): sampling_rate = None try: data, sampling_rate = sf.read(full_path, always_2d=True)# than soundfile. except Exception as ex: print(f"¶ÁÈ¡'{full_path}'ʧ°Ü\nÒì³££º") print(ex) if return_empty_on_exception: return [], sampling_rate or target_sr or 32000 else: raise Exception(ex) if len(data.shape) > 1: data = data[:, 0] assert len(data) > 2# check duration of audio file is > 2 samples (because otherwise the slice operation was on the wrong dimension) if np.issubdtype(data.dtype, np.integer): # if audio data is type int max_mag = -np.iinfo(data.dtype).min # maximum magnitude = min possible value of intXX else: # if audio data is type fp32 max_mag = max(np.amax(data), -np.amin(data)) max_mag = (2**31)+1 if max_mag > (2**15) else ((2**15)+1 if max_mag > 1.01 else 1.0) # data should be either 16-bit INT, 32-bit INT or [-1 to 1] float32 data = paddle.to_tensor(data.astype(np.float32),dtype = 'float32') / max_mag if (paddle.isinf(data) | paddle.isnan(data)).any() and return_empty_on_exception:# resample will crash with inf/NaN inputs. return_empty_on_exception will return empty arr instead of except return [], sampling_rate or target_sr or 32000 if target_sr is not None and sampling_rate != target_sr: data = paddle.to_tensor(librosa.core.resample(data.numpy(), orig_sr=sampling_rate, target_sr=target_sr)) sampling_rate = target_sr return data, sampling_rate def dynamic_range_compression(x, C=1, clip_val=1e-5): return np.log(np.clip(x, a_min=clip_val, a_max=None) * C) def dynamic_range_decompression(x, C=1): return np.exp(x) / C def dynamic_range_compression_torch(x, C=1, clip_val=1e-5): return paddle.log(paddle.clip(x, min=clip_val) * C) def dynamic_range_decompression_torch(x, C=1): return paddle.exp(x) / C class STFT(): def __init__(self, sr=22050, n_mels=80, n_fft=1024, win_size=1024, hop_length=256, fmin=20, fmax=11025, clip_val=1e-5): self.target_sr = sr self.n_mels = n_mels self.n_fft = n_fft self.win_size = win_size self.hop_length = hop_length self.fmin = fmin self.fmax = fmax self.clip_val = clip_val self.mel_basis = {} self.hann_window = {} def get_mel(self, y, center=False): sampling_rate = self.target_sr n_mels = self.n_mels n_fft = self.n_fft win_size = self.win_size hop_length = self.hop_length fmin = self.fmin fmax = self.fmax clip_val = self.clip_val if paddle.min(y) < -1.: print('min value is ', paddle.min(y)) if paddle.max(y) > 1.: print('max value is ', paddle.max(y)) if fmax not in self.mel_basis: mel = librosa_mel_fn(sr=sampling_rate, n_fft=n_fft, n_mels=n_mels, fmin=fmin, fmax=fmax) self.mel_basis[str(fmax)+'_'+str(y.device)] = paddle.to_tensor(mel,dtype = 'float32') self.hann_window[str(y.place)] = paddle.audio.functional.get_window('hann',self.win_size) y = paddle.nn.functional.pad(y.unsqueeze(1), (int((n_fft-hop_length)/2), int((n_fft-hop_length)/2)), mode='reflect') y = y.squeeze(1) spec = paddle.signal.stft(y, n_fft, hop_length=hop_length, win_length=win_size, window=self.hann_window[str(y.device)], center=center, pad_mode='reflect', normalized=False, onesided=True) # print(111,spec) spec = paddle.sqrt(spec.pow(2).sum(-1)+(1e-9)) # print(222,spec) spec = paddle.matmul(self.mel_basis[str(fmax)+'_'+str(y.device)], spec) # print(333,spec) spec = dynamic_range_compression_torch(spec, clip_val=clip_val) # print(444,spec) return spec def __call__(self, audiopath): audio, sr = load_wav_to_torch(audiopath, target_sr=self.target_sr) spect = self.get_mel(audio.unsqueeze(0)).squeeze(0) return spect stft = STFT()