librosa.core.load

librosa.core.load(path, sr=22050, mono=True, offset=0.0, duration=None, dtype=<class ‘numpy.float32’>, res_type=’kaiser_best’)[source]

Load an audio file as a floating point time series.

Audio will be automatically resampled to the given rate (default sr=22050).

To preserve the native sampling rate of the file, use sr=None.

Parameters:
path : string

path to the input file.

Any format supported by audioread will work.

sr : number > 0 [scalar]

target sampling rate

‘None’ uses the native sampling rate

mono : bool

convert signal to mono

offset : float

start reading after this time (in seconds)

duration : float

only load up to this much audio (in seconds)

dtype : numeric type

data type of y

res_type : str

resample type (see note)

Note

By default, this uses resampy’s high-quality mode (‘kaiser_best’).

To use a faster method, set res_type=’kaiser_fast’.

To use scipy.signal.resample, set res_type=’scipy’.

Returns:
y : np.ndarray [shape=(n,) or (2, n)]

audio time series

sr : number > 0 [scalar]

sampling rate of y

Examples

>>> # Load a wav file
>>> filename = librosa.util.example_audio_file()
>>> y, sr = librosa.load(filename)
>>> y
array([ -4.756e-06,  -6.020e-06, ...,  -1.040e-06,   0.000e+00], dtype=float32)
>>> sr
22050
>>> # Load a wav file and resample to 11 KHz
>>> filename = librosa.util.example_audio_file()
>>> y, sr = librosa.load(filename, sr=11025)
>>> y
array([ -2.077e-06,  -2.928e-06, ...,  -4.395e-06,   0.000e+00], dtype=float32)
>>> sr
11025
>>> # Load 5 seconds of a wav file, starting 15 seconds in
>>> filename = librosa.util.example_audio_file()
>>> y, sr = librosa.load(filename, offset=15.0, duration=5.0)
>>> y
array([ 0.069,  0.1  , ..., -0.101,  0.   ], dtype=float32)
>>> sr
22050