mirror of
https://github.com/2noise/ChatTTS.git
synced 2026-08-30 17:05:27 +08:00
2d293cafd1
Co-authored-by: github-actions[bot] <github-actions[bot]@users.noreply.github.com>
76 lines
2.3 KiB
Python
76 lines
2.3 KiB
Python
from io import BufferedWriter, BytesIO
|
|
from pathlib import Path
|
|
from typing import Dict
|
|
|
|
import av
|
|
from av.audio.resampler import AudioResampler
|
|
import numpy as np
|
|
|
|
|
|
video_format_dict: Dict[str, str] = {
|
|
"m4a": "mp4",
|
|
}
|
|
|
|
audio_format_dict: Dict[str, str] = {
|
|
"ogg": "libvorbis",
|
|
"mp4": "aac",
|
|
}
|
|
|
|
|
|
def wav2(i: BytesIO, o: BufferedWriter, format: str):
|
|
"""
|
|
https://github.com/fumiama/Retrieval-based-Voice-Conversion-WebUI/blob/412a9950a1e371a018c381d1bfb8579c4b0de329/infer/lib/audio.py#L20
|
|
"""
|
|
inp = av.open(i, "r")
|
|
format = video_format_dict.get(format, format)
|
|
out = av.open(o, "w", format=format)
|
|
format = audio_format_dict.get(format, format)
|
|
|
|
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 load_audio(file: str, sr: int) -> np.ndarray:
|
|
if not Path(file).exists():
|
|
raise FileNotFoundError(f"File not found: {file}")
|
|
|
|
try:
|
|
container = av.open(file)
|
|
resampler = AudioResampler(format="fltp", layout="mono", rate=sr)
|
|
|
|
# Estimated maximum total number of samples to pre-allocate the array
|
|
# AV stores length in microseconds by default
|
|
estimated_total_samples = int(container.duration * sr // 1_000_000)
|
|
decoded_audio = np.zeros(estimated_total_samples + 1, dtype=np.float32)
|
|
|
|
offset = 0
|
|
for frame in container.decode(audio=0):
|
|
frame.pts = None # Clear presentation timestamp to avoid resampling issues
|
|
resampled_frames = resampler.resample(frame)
|
|
for resampled_frame in resampled_frames:
|
|
frame_data = resampled_frame.to_ndarray()[0]
|
|
end_index = offset + len(frame_data)
|
|
|
|
# Check if decoded_audio has enough space, and resize if necessary
|
|
if end_index > decoded_audio.shape[0]:
|
|
decoded_audio = np.resize(decoded_audio, end_index + 1)
|
|
|
|
decoded_audio[offset:end_index] = frame_data
|
|
offset += len(frame_data)
|
|
|
|
# Truncate the array to the actual size
|
|
decoded_audio = decoded_audio[:offset]
|
|
except Exception as e:
|
|
raise RuntimeError(f"Failed to load audio: {e}")
|
|
|
|
return decoded_audio
|