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# -*- coding: utf-8 -*-
"""
Subtractive synthesis
=====================
**Author**: `Moto Hira <moto@meta.com>`__
This tutorial is the continuation of
`Filter Design Tutorial <./filter_design_tutorial.html>`__.
This tutorial shows how to perform subtractive synthesis with TorchAudio's DSP functions.
Subtractive synthesis creates timbre by applying filters to source waveform.
.. warning::
This tutorial requires prototype DSP features, which are
available in nightly builds.
Please refer to https://pytorch.org/get-started/locally
for instructions for installing a nightly build.
"""
import torch
import torchaudio
print(torch.__version__)
print(torchaudio.__version__)
######################################################################
# Overview
# --------
#
#
try:
from torchaudio.prototype.functional import filter_waveform, frequency_impulse_response, sinc_impulse_response
except ModuleNotFoundError:
print(
"Failed to import prototype DSP features. "
"Please install torchaudio nightly builds. "
"Please refer to https://pytorch.org/get-started/locally "
"for instructions to install a nightly build."
)
raise
import matplotlib.pyplot as plt
from IPython.display import Audio
######################################################################
# Filtered Noise
# --------------
#
# Subtractive synthesis starts with a waveform and applies filters to
# some frequency components.
#
# For the first example of subtractive synthesis, we apply
# time-varying low pass filter to white noise.
#
# First, we create a white noise.
#
SAMPLE_RATE = 16_000
duration = 4
num_frames = int(duration * SAMPLE_RATE)
noise = torch.rand((num_frames,)) - 0.5
######################################################################
#
def plot_input():
fig, axes = plt.subplots(2, 1, sharex=True)
t = torch.linspace(0, duration, num_frames)
axes[0].plot(t, noise)
axes[0].grid(True)
axes[1].specgram(noise, Fs=SAMPLE_RATE)
Audio(noise, rate=SAMPLE_RATE)
plot_input()
######################################################################
# Windowed-sinc filter
# --------------------
#
######################################################################
#
# Sweeping cutoff frequency
# ~~~~~~~~~~~~~~~~~~~~~~~~~
#
# We use :py:func:`~torchaudio.prototype.functional.sinc_impulse_response` to
# create series of low pass filters, while changing the cut-off
# frequency from zero to Nyquist frequency.
#
num_filters = 64 * duration
window_size = 2049
f_cutoff = torch.linspace(0.0, 0.8, num_filters)
kernel = sinc_impulse_response(f_cutoff, window_size)
######################################################################
#
# To apply time-varying filter, we use
# :py:func:`~torchaudio.prototype.functional.filter_waveform`
#
filtered = filter_waveform(noise, kernel)
######################################################################
#
# Let's look at the spectrogram of the resulting audio and listen to it.
#
def plot_sinc_ir(waveform, cutoff, sample_rate, vol=0.2):
num_frames = waveform.size(0)
duration = num_frames / sample_rate
num_cutoff = cutoff.size(0)
nyquist = sample_rate / 2
_, axes = plt.subplots(2, 1, sharex=True)
t = torch.linspace(0, duration, num_frames)
axes[0].plot(t, waveform)
axes[0].grid(True)
axes[1].specgram(waveform, Fs=sample_rate, scale="dB")
t = torch.linspace(0, duration, num_cutoff)
axes[1].plot(t, cutoff * nyquist, color="gray", linewidth=0.8, label="Cutoff Frequency", linestyle="--")
axes[1].legend(loc="upper center")
axes[1].set_ylim([0, nyquist])
waveform /= waveform.abs().max()
return Audio(vol * waveform, rate=sample_rate, normalize=False)
######################################################################
#
plot_sinc_ir(filtered, f_cutoff, SAMPLE_RATE)
######################################################################
#
# Oscillating cutoff frequency
# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~
#
# By oscillating the cutoff frequency, we can emulate an effect of
# Low-frequency oscillation (LFO).
#
PI2 = torch.pi * 2
num_filters = 90 * duration
f_lfo = torch.linspace(0.9, 0.1, num_filters)
f_cutoff_osci = torch.linspace(0.01, 0.03, num_filters) * torch.sin(torch.cumsum(f_lfo, dim=0))
f_cutoff_base = torch.linspace(0.8, 0.03, num_filters) ** 1.7
f_cutoff = f_cutoff_base + f_cutoff_osci
######################################################################
#
kernel = sinc_impulse_response(f_cutoff, window_size)
filtered = filter_waveform(noise, kernel)
######################################################################
#
plot_sinc_ir(filtered, f_cutoff, SAMPLE_RATE)
######################################################################
#
# Wah-wah effects
# ~~~~~~~~~~~~~~~
#
# Wah-wah effects are applications of low-pass filter or band-pass filter.
# They change the cut-off freuqnecy or Q-factor quickly.
f_lfo = torch.linspace(0.15, 0.15, num_filters)
f_cutoff = 0.07 + 0.06 * torch.sin(torch.cumsum(f_lfo, dim=0))
######################################################################
#
kernel = sinc_impulse_response(f_cutoff, window_size)
filtered = filter_waveform(noise, kernel)
######################################################################
#
plot_sinc_ir(filtered, f_cutoff, SAMPLE_RATE)
######################################################################
# Arbitrary frequence response
# ----------------------------
#
# By using
# :py:func:`~torchaudio.prototype.functinal.frequency_impulse_response`,
# one can directly control the power distribution over frequency.
#
magnitudes = torch.sin(torch.linspace(0, 10, 64)) ** 4.0
kernel = frequency_impulse_response(magnitudes)
filtered = filter_waveform(noise, kernel.unsqueeze(0))
######################################################################
#
def plot_waveform(magnitudes, filtered, sample_rate):
nyquist = sample_rate / 2
num_samples = filtered.size(-1)
duration = num_samples / sample_rate
# Re-organize magnitudes for overlay
N = 10 # number of overlays
interval = torch.linspace(0.05, 0.95, N)
offsets = duration * interval
# Select N magnitudes for overlays
mags = torch.stack(
[magnitudes for _ in range(N)]
if magnitudes.ndim == 1
else [magnitudes[int(i * magnitudes.size(0))] for i in interval]
)
mag_x = offsets.unsqueeze(-1) + 0.1 * mags
mag_y = torch.linspace(0, nyquist, magnitudes.size(-1)).tile((N, 1))
_, ax = plt.subplots(1, 1, sharex=True)
ax.vlines(offsets, 0, nyquist, color="gray", linestyle="--", linewidth=0.8)
ax.plot(mag_x.T.numpy(), mag_y.T.numpy(), color="gray", linewidth=0.8)
ax.specgram(filtered, Fs=sample_rate)
return Audio(filtered, rate=sample_rate)
######################################################################
#
plot_waveform(magnitudes, filtered, SAMPLE_RATE)
######################################################################
#
# It is also possible to make a non-stationary filter.
magnitudes = torch.stack([torch.linspace(0.0, w, 1000) for w in torch.linspace(4.0, 40.0, 250)])
magnitudes = torch.sin(magnitudes) ** 4.0
######################################################################
#
kernel = frequency_impulse_response(magnitudes)
filtered = filter_waveform(noise, kernel)
######################################################################
#
plot_waveform(magnitudes, filtered, SAMPLE_RATE)
######################################################################
#
# Of course it is also possible to emulate simple low pass filter.
magnitudes = torch.concat([torch.ones((32,)), torch.zeros((32,))])
######################################################################
#
kernel = frequency_impulse_response(magnitudes)
filtered = filter_waveform(noise, kernel.unsqueeze(0))
######################################################################
#
plot_waveform(magnitudes, filtered, SAMPLE_RATE)
######################################################################
# References
# ----------
#
# - https://en.wikipedia.org/wiki/Additive_synthesis
# - https://computermusicresource.com/Simple.bell.tutorial.html
# - https://computermusicresource.com/Definitions/additive.synthesis.html
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