.. _ioformats: 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 `_. .. note:: 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 `soundfile` directly. 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 :func:`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'))`: .. code-block:: python :linenos: import librosa import soundfile as sf # Get example audio file filename = librosa.ex('trumpet') data, samplerate = sf.read(filename, dtype='float32') data = data.T data_22k = librosa.resample(data, samplerate, 22050) Blockwise Reading ----------------- 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. :func:`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: .. code-block:: python :linenos: import librosa sr = librosa.get_samplerate('/path/to/file.wav') # Set the frame parameters to be equivalent to the librosa defaults # in the file's native sampling rate frame_length = (2048 * sr) // 22050 hop_length = (512 * sr) // 22050 # Stream the data, working on 128 frames at a time stream = librosa.stream('path/to/file.wav', block_length=128, frame_length=frame_length, hop_length=hop_length) chromas = [] for y in stream: chroma_block = librosa.feature.chroma_stft(y=y, sr=sr, n_fft=frame_length, hop_length=hop_length, center=False) 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``. Each fragment ``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 :func:`librosa.stream`. 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 :func:`librosa.load` and :func:`librosa.stream`, provided that the underlying codec is supported by `soundfile`.) E.g.: read files from zip compressed archives: .. code-block:: python :linenos: import zipfile as zf import soundfile as sf import io with zf.ZipFile('test.zip') as myzip: with myzip.open('stereo_file.wav') as myfile: tmp = io.BytesIO(myfile.read()) data, samplerate = sf.read(tmp) Download and read from URL: .. code-block:: python :linenos: import soundfile as sf import io from six.moves.urllib.request import urlopen url = "https://raw.githubusercontent.com/librosa/librosa/master/tests/data/test1_44100.wav" data, 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: .. code-block:: python :linenos: import numpy as np import soundfile as sf rate = 44100 data = np.random.uniform(-1, 1, size=(rate * 10, 2)) # Write out audio as 24bit PCM WAV sf.write('stereo_file.wav', data, samplerate, subtype='PCM_24') # Write out audio as 24bit Flac sf.write('stereo_file.flac', data, samplerate, format='flac', subtype='PCM_24') # Write out audio as 16bit OGG sf.write('stereo_file.ogg', data, samplerate, format='ogg', subtype='vorbis')