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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}") |