I'm aware of the following question: How to create a pydub AudioSegment using an numpy array?
My question is the right opposite. If I have a pydub AudioSegment how can I convert it to a numpy array?
I would like to use scipy filters and so on. It is not very clear to me what is the internal structure of the AudioSegment raw data.
Pydub has a facility for getting the audio data as an array of samples, it is an array.array
instance (not a numpy array) but you should be able to convert it to a numpy array relatively easily:
from pydub import AudioSegment
sound = AudioSegment.from_file("sound1.wav")
# this is an array
samples = sound.get_array_of_samples()
You may be able to create a numpy variant of the implementation though. That method is implemented pretty simply:
def get_array_of_samples(self):
"""
returns the raw_data as an array of samples
"""
return array.array(self.array_type, self._data)
Creating a new audio segment from a (modified?) array of samples is also possible:
new_sound = sound._spawn(samples)
The above is a little hacky, it was written for internal use within the AudioSegment class, but it mainly just figures out what type of audio data you're using (array of samples, list of samples, bytes, bytestring, etc). It's safe to use despite the underscore prefix.
You can get an array.array
from an AudioSegment
then convert it to a numpy.ndarray
:
from pydub import AudioSegment
import numpy as np
song = AudioSegment.from_mp3('song.mp3')
samples = song.get_array_of_samples()
samples = np.array(samples)
None of the existing answers is perfect, they miss reshaping and sample width. I have written this function that helps to convert the audio to the standard audio representation in np:
def pydub_to_np(audio: pydub.AudioSegment) -> (np.ndarray, int):
"""
Converts pydub audio segment into np.float32 of shape [duration_in_seconds*sample_rate, channels],
where each value is in range [-1.0, 1.0].
Returns tuple (audio_np_array, sample_rate).
"""
return np.array(audio.get_array_of_samples(), dtype=np.float32).reshape((-1, audio.channels)) / (
1 << (8 * audio.sample_width - 1)), audio.frame_rate
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