Advanced I/O Use Cases¶
This section covers advanced use cases for input and output which go beyond the I/O functionality currently provided by librosa.
Read specific formats¶
librosa uses soundfile and audioread for reading audio.
As of v0.7, librosa uses
soundfile by default, and falls back on
audioread only when dealing with codecs unsupported by
soundfile (notably, MP3, and some variants of WAV).
For a list of codecs supported by
soundfile, see the libsndfile documentation.
See installation instruction for PySoundFile here.
Librosa’s load function is meant for the common case where you want to load an entire (fragment of a) recording into memory, but some applications require more flexibility.
In these cases, we recommend using
Reading audio files using
soundfile is similar to the method in librosa. One important difference is that the read data is of shape (nb_samples, nb_channels) compared to (nb_channels, nb_samples) in
librosa.core.load. Also the signal is not resampled to 22050 Hz by default, hence it would need be transposed and resampled for further processing in librosa. The following example is equivalent to librosa.load(librosa.util.ex(‘trumpet’)):
1import librosa 2import soundfile as sf 3 4# Get example audio file 5filename = librosa.ex('trumpet') 6 7data, samplerate = sf.read(filename, dtype='float32') 8data = data.T 9data_22k = librosa.resample(data, samplerate, 22050)
For large audio signals it could be beneficial to not load the whole audio file
into memory. Librosa 0.7 introduced a streaming interface, which can be used to
work on short fragments of audio sequentially.
librosa.stream cuts an input
file into blocks of audio, which correspond to a given number of frames,
which can be iterated over as in the following example:
1import librosa 2 3sr = librosa.get_samplerate('/path/to/file.wav') 4 5# Set the frame parameters to be equivalent to the librosa defaults 6# in the file's native sampling rate 7frame_length = (2048 * sr) // 22050 8hop_length = (512 * sr) // 22050 9 10# Stream the data, working on 128 frames at a time 11stream = librosa.stream('path/to/file.wav', 12 block_length=128, 13 frame_length=frame_length, 14 hop_length=hop_length) 15 16chromas =  17for y in stream: 18 chroma_block = librosa.feature.chroma_stft(y=y, sr=sr, 19 n_fft=frame_length, 20 hop_length=hop_length, 21 center=False) 22 chromas.append(chromas)
In this example, each audio fragment
y will consist of 128 frames worth of samples,
or more specifically,
len(y) == frame_length + (block_length - 1) * hop_length.
y will overlap with the subsequent fragment by
frame_length - hop_length
samples, which ensures that stream processing will provide equivalent results to if the entire
sequence was processed in one step (assuming padding / centering is disabled).
For more details about the streaming interface, refer to
Read file-like objects¶
If you want to read audio from file-like objects (also called virtual files)
you can use
soundfile as well. (This will also work with
that the underlying codec is supported by
E.g.: read files from zip compressed archives:
1import zipfile as zf 2import soundfile as sf 3import io 4 5with zf.ZipFile('test.zip') as myzip: 6 with myzip.open('stereo_file.wav') as myfile: 7 tmp = io.BytesIO(myfile.read()) 8 data, samplerate = sf.read(tmp)
Download and read from URL:
1import soundfile as sf 2import io 3 4from six.moves.urllib.request import urlopen 5 6url = "https://raw.githubusercontent.com/librosa/librosa/master/tests/data/test1_44100.wav" 7 8data, samplerate = sf.read(io.BytesIO(urlopen(url).read()))
Write out audio files¶
PySoundFile provides output functionality that can be used directly with numpy array audio buffers:
1import numpy as np 2import soundfile as sf 3 4rate = 44100 5data = np.random.uniform(-1, 1, size=(rate * 10, 2)) 6 7# Write out audio as 24bit PCM WAV 8sf.write('stereo_file.wav', data, samplerate, subtype='PCM_24') 9 10# Write out audio as 24bit Flac 11sf.write('stereo_file.flac', data, samplerate, format='flac', subtype='PCM_24') 12 13# Write out audio as 16bit OGG 14sf.write('stereo_file.ogg', data, samplerate, format='ogg', subtype='vorbis')