import librosa import numpy as np import av from io import BytesIO import ffmpeg import os import sys import random from lib.infer.infer_libs.csvutil import CSVutil #import csv platform_stft_mapping = { 'linux': 'stftpitchshift', 'darwin': 'stftpitchshift', 'win32': 'stftpitchshift.exe', } stft = platform_stft_mapping.get(sys.platform) def wav2(i, o, format): inp = av.open(i, 'rb') if format == "m4a": format = "mp4" out = av.open(o, 'wb', format=format) if format == "ogg": format = "libvorbis" if format == "mp4": format = "aac" ostream = out.add_stream(format) for frame in inp.decode(audio=0): for p in ostream.encode(frame): out.mux(p) for p in ostream.encode(None): out.mux(p) out.close() inp.close() def audio2(i, o, format, sr): inp = av.open(i, 'rb') out = av.open(o, 'wb', format=format) if format == "ogg": format = "libvorbis" if format == "f32le": format = "pcm_f32le" ostream = out.add_stream(format, channels=1) ostream.sample_rate = sr for frame in inp.decode(audio=0): for p in ostream.encode(frame): out.mux(p) out.close() inp.close() def load_audion(file, sr): try: file = ( file.strip(" ").strip('"').strip("\n").strip('"').strip(" ") ) # 防止小白拷路径头尾带了空格和"和回车 with open(file, "rb") as f: with BytesIO() as out: audio2(f, out, "f32le", sr) return np.frombuffer(out.getvalue(), np.float32).flatten() except AttributeError: audio = file[1] / 32768.0 if len(audio.shape) == 2: audio = np.mean(audio, -1) return librosa.resample(audio, orig_sr=file[0], target_sr=16000) except Exception as e: raise RuntimeError(f"Failed to load audio: {e}") def load_audio(file, sr, DoFormant=False, Quefrency=1.0, Timbre=1.0): converted = False DoFormant, Quefrency, Timbre = CSVutil("lib/csvdb/formanting.csv", "r", "formanting") DoFormant, Quefrency, Timbre = bool(DoFormant), float(Quefrency), float(Timbre) try: file = file.strip(" ").strip('"').strip("\n").strip('"').strip(" ") if not file.endswith(".wav"): converted = True # Conversión de formato usando ffmpeg converting = ( ffmpeg.input(file, threads=0) .output(f"{file}.wav") .run(cmd=["ffmpeg", "-nostdin"], capture_stdout=True, capture_stderr=True) ) file = f"{file}.wav" print(f" · File converted to Wav format: {file}\n") if DoFormant == False: # Procesamiento de formantes usando stftpitchshift command = ( f'{stft} -i "{file}" -q "{Quefrency}" ' f'-t "{Timbre}" -o "{file}FORMANTED.wav"' ) os.system(command) file = f"{file}FORMANTED.wav" print(f" · Formanted {file}!\n") with open(file, "rb") as f: with BytesIO() as out: audio2(f, out, "f32le", sr) audio_data = np.frombuffer(out.getvalue(), np.float32).flatten() if converted: try: os.remove(file) except Exception as e: pass; print(f"Couldn't remove converted type of file due to {e}") converted = False return audio_data except AttributeError: audio = file[1] / 32768.0 if len(audio.shape) == 2: audio = np.mean(audio, -1) return librosa.resample(audio, orig_sr=file[0], target_sr=16000) except Exception as e: raise RuntimeError(f"Failed to load audio: {e}") def check_audio_duration(file): try: file = file.strip(" ").strip('"').strip("\n").strip('"').strip(" ") probe = ffmpeg.probe(file) duration = float(probe['streams'][0]['duration']) if duration < 0.76: print( f"Audio file, {file.split('/')[-1]}, under ~0.76s detected - file is too short. Target at least 1-2s for best results." ) return False return True except Exception as e: raise RuntimeError(f"Failed to check audio duration: {e}")