dadk / audio.py
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"""We use the PyAV library to decode the audio: https://github.com/PyAV-Org/PyAV
The advantage of PyAV is that it bundles the FFmpeg libraries so there is no additional
system dependencies. FFmpeg does not need to be installed on the system.
However, the API is quite low-level so we need to manipulate audio frames directly.
"""
import gc
import io
import itertools
from typing import BinaryIO, Union
import av
import numpy as np
def decode_audio(
input_file: Union[str, BinaryIO],
sampling_rate: int = 16000,
split_stereo: bool = False,
):
"""Decodes the audio.
Args:
input_file: Path to the input file or a file-like object.
sampling_rate: Resample the audio to this sample rate.
split_stereo: Return separate left and right channels.
Returns:
A float32 Numpy array.
If `split_stereo` is enabled, the function returns a 2-tuple with the
separated left and right channels.
"""
resampler = av.audio.resampler.AudioResampler(
format="s16",
layout="mono" if not split_stereo else "stereo",
rate=sampling_rate,
)
raw_buffer = io.BytesIO()
dtype = None
with av.open(input_file, mode="r", metadata_errors="ignore") as container:
frames = container.decode(audio=0)
frames = _ignore_invalid_frames(frames)
frames = _group_frames(frames, 500000)
frames = _resample_frames(frames, resampler)
for frame in frames:
array = frame.to_ndarray()
dtype = array.dtype
raw_buffer.write(array)
# It appears that some objects related to the resampler are not freed
# unless the garbage collector is manually run.
del resampler
gc.collect()
audio = np.frombuffer(raw_buffer.getbuffer(), dtype=dtype)
# Convert s16 back to f32.
audio = audio.astype(np.float32) / 32768.0
if split_stereo:
left_channel = audio[0::2]
right_channel = audio[1::2]
return left_channel, right_channel
return audio
def _ignore_invalid_frames(frames):
iterator = iter(frames)
while True:
try:
yield next(iterator)
except StopIteration:
break
except av.error.InvalidDataError:
continue
def _group_frames(frames, num_samples=None):
fifo = av.audio.fifo.AudioFifo()
for frame in frames:
frame.pts = None # Ignore timestamp check.
fifo.write(frame)
if num_samples is not None and fifo.samples >= num_samples:
yield fifo.read()
if fifo.samples > 0:
yield fifo.read()
def _resample_frames(frames, resampler):
# Add None to flush the resampler.
for frame in itertools.chain(frames, [None]):
yield from resampler.resample(frame)
def pad_or_trim(array, length: int, *, axis: int = -1):
"""
Pad or trim the audio array to N_SAMPLES, as expected by the encoder.
"""
if array.shape[axis] > length:
array = array.take(indices=range(length), axis=axis)
if array.shape[axis] < length:
pad_widths = [(0, 0)] * array.ndim
pad_widths[axis] = (0, length - array.shape[axis])
array = np.pad(array, pad_widths)
return array