def spectrogram(samples, fft_length=256, sample_rate=2, hop_length=128):
"""
Compute the spectrogram for a real signal.
The parameters follow the naming convention of
matplotlib.mlab.specgram
Args:
samples (1D array): input audio signal
fft_length (int): number of elements in fft window
sample_rate (scalar): sample rate
hop_length (int): hop length (relative offset between neighboring
fft windows).
Returns:
x (2D array): spectrogram [frequency x time]
freq (1D array): frequency of each row in x
Note:
This is a truncating computation e.g. if fft_length=10,
hop_length=5 and the signal has 23 elements, then the
last 3 elements will be truncated.
"""
assert not np.iscomplexobj(samples), "Must not pass in complex numbers"
window = np.hanning(fft_length)[:, None]
window_norm = np.sum(window ** 2)
# The scaling below follows the convention of
# matplotlib.mlab.specgram which is the same as
# matlabs specgram.
scale = window_norm * sample_rate
trunc = (len(samples) - fft_length) % hop_length
x = samples[:len(samples) - trunc]
# "stride trick" reshape to include overlap
nshape = (fft_length, (len(x) - fft_length) // hop_length + 1)
nstrides = (x.strides[0], x.strides[0] * hop_length)
x = as_strided(x, shape=nshape, strides=nstrides)
# window stride sanity check
assert np.all(x[:, 1] == samples[hop_length:(hop_length + fft_length)])
# broadcast window, compute fft over columns and square mod
# This function computes the one-dimensional n-point discrete Fourier Transform (DFT) of a real-valued array by means of an efficient algorithm called the Fast Fourier Transform (FFT).
x = np.fft.rfft(x * window, axis=0)
x = np.absolute(x) ** 2
# scale, 2.0 for everything except dc and fft_length/2
x[1:-1, :] *= (2.0 / scale)
x[(0, -1), :] /= scale
freqs = float(sample_rate) / fft_length * np.arange(x.shape[0])
return x, freqs
评论列表
文章目录
