feat(tools): load_audio supports mean of stereo

tools/audio/av.py

@@ -1,8 +1,9 @@
 from io import BufferedWriter, BytesIO
 from pathlib import Path
-from typing import Dict
+from typing import Dict, Tuple, Optional, Union, List
 
 import av
+from av.audio.frame import AudioFrame
 from av.audio.resampler import AudioResampler
 import numpy as np
 
@@ -39,41 +40,66 @@ def wav2(i: BytesIO, o: BufferedWriter, format: str):
     inp.close()
 
 
-def load_audio(file: str, sr: int) -> np.ndarray:
+def load_audio(
+        file: Union[str, BytesIO, Path],
+        sr: Optional[int]=None,
+        format: Optional[str]=None,
+        mono=True
+    ) -> Union[np.ndarray, Tuple[np.ndarray, int]]:
     """
     https://github.com/fumiama/Retrieval-based-Voice-Conversion-WebUI/blob/412a9950a1e371a018c381d1bfb8579c4b0de329/infer/lib/audio.py#L39
     """
-
-    if not Path(file).exists():
+    if (isinstance(file, str) and not Path(file).exists()) or (isinstance(file, Path) and not file.exists()):
         raise FileNotFoundError(f"File not found: {file}")
+    rate = 0
+
+    container = av.open(file, format=format)
+    audio_stream = next(s for s in container.streams if s.type == "audio")
+    channels = 1 if audio_stream.layout == "mono" else 2
+    container.seek(0)
+    resampler = AudioResampler(format="fltp", layout=audio_stream.layout, rate=sr) if sr is not None else None
+
+    # 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) if sr is not None else 48000
+    decoded_audio = np.zeros(estimated_total_samples + 1 if channels == 1 else (channels, estimated_total_samples + 1), dtype=np.float32)
+
+    offset = 0
+
+    def process_packet(packet: List[AudioFrame]):
+        frames_data = []
+        rate = 0
+        for frame in packet:
+            frame.pts = None  # 清除时间戳，避免重新采样问题
+            resampled_frames = resampler.resample(frame) if resampler is not None else [frame]
+            for resampled_frame in resampled_frames:
+                frame_data = resampled_frame.to_ndarray()
+                rate = resampled_frame.rate
+                frames_data.append(frame_data)
+        return (rate, frames_data)
 
-    try:
-        container = av.open(file)
-        resampler = AudioResampler(format="fltp", layout="mono", rate=sr)
+    def frame_iter(container):
+        for p in container.demux(container.streams.audio[0]):
+            yield p.decode()
 
-        # 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)
+    for r, frames_data in map(process_packet, frame_iter(container)):
+        if not rate: rate = r
+        for frame_data in frames_data:
+            end_index = offset + len(frame_data[0])
 
-        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)
+            # 检查 decoded_audio 是否有足够的空间，并在必要时调整大小
+            if end_index > decoded_audio.shape[1]:
+                decoded_audio = np.resize(decoded_audio, (decoded_audio.shape[0], end_index*4))
 
-                # 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)
+            np.copyto(decoded_audio[..., offset:end_index], frame_data)
+            offset += len(frame_data[0])
 
-                decoded_audio[offset:end_index] = frame_data
-                offset += len(frame_data)
+    # Truncate the array to the actual size
+    decoded_audio = decoded_audio[..., :offset]
 
-        # Truncate the array to the actual size
-        decoded_audio = decoded_audio[:offset]
-    except Exception as e:
-        raise RuntimeError(f"Failed to load audio: {e}")
+    if mono and decoded_audio.shape[0] > 1:
+        decoded_audio = decoded_audio.mean(0)
 
-    return decoded_audio
+    if sr is not None:
+        return decoded_audio
+    return decoded_audio, rate