#encoding = utf8 import librosa import librosa.filters import numpy as np from scipy import signal from scipy.io import wavfile from hparams import hparams as hp import soundfile as sf from IPython.display import Audio def load_wav(path, sr): return librosa.core.load(path, sr=sr)[0] def save_wav(wav, path, sr): wav *= 32767 / max(0.01, np.max(np.abs(wav))) # proposed by @dsmiller wavfile.write(path, sr, wav.astype(np.int16)) def save_wavenet_wav(wav, path, sr): librosa.output.write_wav(path, wav, sr=sr) def preemphasis(wav, k, preemphasize=True): if preemphasize: return signal.lfilter([1, -k], [1], wav) return wav def inv_preemphasis(wav, k, inv_preemphasize=True): if inv_preemphasize: return signal.lfilter([1], [1, -k], wav) return wav def get_hop_size(): hop_size = hp.hop_size if hop_size is None: assert hp.frame_shift_ms is not None hop_size = int(hp.frame_shift_ms / 1000 * hp.sample_rate) return hop_size def linearspectrogram(wav): D = _stft(preemphasis(wav, hp.preemphasis, hp.preemphasize)) S = _amp_to_db(np.abs(D)) - hp.ref_level_db if hp.signal_normalization: return _normalize(S) return S def melspectrogram(wav): D = _stft(preemphasis(wav, hp.preemphasis, hp.preemphasize)) S = _amp_to_db(_linear_to_mel(np.abs(D))) - hp.ref_level_db if hp.signal_normalization: return _normalize(S) return S def _lws_processor(): import lws return lws.lws(hp.n_fft, get_hop_size(), fftsize=hp.win_size, mode="speech") def _stft(y): if hp.use_lws: return _lws_processor(hp).stft(y).T else: return librosa.stft(y=y, n_fft=hp.n_fft, hop_length=get_hop_size(), win_length=hp.win_size) ########################################################## # Those are only correct when using lws!!! (This was messing with Wavenet quality for a long time!) def num_frames(length, fsize, fshift): """Compute number of time frames of spectrogram """ pad = (fsize - fshift) if length % fshift == 0: M = (length + pad * 2 - fsize) // fshift + 1 else: M = (length + pad * 2 - fsize) // fshift + 2 return M def pad_lr(x, fsize, fshift): """Compute left and right padding """ M = num_frames(len(x), fsize, fshift) pad = (fsize - fshift) T = len(x) + 2 * pad r = (M - 1) * fshift + fsize - T return pad, pad + r ########################################################## # Librosa correct padding def librosa_pad_lr(x, fsize, fshift): return 0, (x.shape[0] // fshift + 1) * fshift - x.shape[0] # Conversions _mel_basis = None def _linear_to_mel(spectogram): global _mel_basis if _mel_basis is None: _mel_basis = _build_mel_basis() return np.dot(_mel_basis, spectogram) def _build_mel_basis(): assert hp.fmax <= hp.sample_rate // 2 return librosa.filters.mel(sr=hp.sample_rate, n_fft=hp.n_fft, n_mels=hp.num_mels, fmin=hp.fmin, fmax=hp.fmax) def _amp_to_db(x): min_level = np.exp(hp.min_level_db / 20 * np.log(10)) return 20 * np.log10(np.maximum(min_level, x)) def _db_to_amp(x): return np.power(10.0, (x) * 0.05) def _normalize(S): if hp.allow_clipping_in_normalization: if hp.symmetric_mels: return np.clip((2 * hp.max_abs_value) * ((S - hp.min_level_db) / (-hp.min_level_db)) - hp.max_abs_value, -hp.max_abs_value, hp.max_abs_value) else: return np.clip(hp.max_abs_value * ((S - hp.min_level_db) / (-hp.min_level_db)), 0, hp.max_abs_value) assert S.max() <= 0 and S.min() - hp.min_level_db >= 0 if hp.symmetric_mels: return (2 * hp.max_abs_value) * ((S - hp.min_level_db) / (-hp.min_level_db)) - hp.max_abs_value else: return hp.max_abs_value * ((S - hp.min_level_db) / (-hp.min_level_db)) def _denormalize(D): if hp.allow_clipping_in_normalization: if hp.symmetric_mels: return (((np.clip(D, -hp.max_abs_value, hp.max_abs_value) + hp.max_abs_value) * -hp.min_level_db / (2 * hp.max_abs_value)) + hp.min_level_db) else: return ((np.clip(D, 0, hp.max_abs_value) * -hp.min_level_db / hp.max_abs_value) + hp.min_level_db) if hp.symmetric_mels: return (((D + hp.max_abs_value) * -hp.min_level_db / (2 * hp.max_abs_value)) + hp.min_level_db) else: return ((D * -hp.min_level_db / hp.max_abs_value) + hp.min_level_db) def load_audio(file_path, sr=16000): """加载音频文件并返回音频数据和采样率""" wav, sr = librosa.load(file_path, sr=sr) return wav, sr def split_audio(wav, sr, chunk_duration): """将音频按指定时长切割""" # 计算每个片段包含的采样点数量 chunk_size = int(chunk_duration * sr) num_chunks = int(np.ceil(len(wav) / chunk_size)) audio_chunks = [] for i in range(num_chunks): start_idx = i * chunk_size end_idx = min((i + 1) * chunk_size, len(wav)) chunk = wav[start_idx:end_idx] audio_chunks.append(chunk) return audio_chunks def save_chunks(chunks, sr, output_folder, base_filename="chunk"): """保存切割的音频块""" for idx, chunk in enumerate(chunks): output_path = f"{output_folder}/{base_filename}_{idx}.wav" sf.write(output_path, chunk, sr) print(f"Saved {output_path}") def play_audio_chunk(chunk, sr): """播放指定音频块""" return Audio(chunk, rate=sr)