Feedback Delay Networks in Python

From Theory to Differentiable Design with pyFDN

Sebastian J. Schlecht

Friedrich-Alexander-Universität Erlangen-Nürnberg

Facundo Franchino

Massachusetts Institute of Technology

1 September 2026

Ninety minutes, six moves

Time		Segment	You will…
0:00	15	What is an FDN?	read the block diagram, and where filters go in it
0:15	15	pyFDN, the tour	know where everything lives, run five lines
0:30	10	What to do next?	see where the research frontier is
0:40	25	Hands-on 1 — build one	delays, matrix, gains, decay, hooks — then a hall
1:05	20	Hands-on 2 — match a room	fit one to a measured hall, by gradient descent
1:25	5	Q&A	ask your questions

Slides and setup instructions: artificial-audio.github.io/pyfdn-tutorial-2026

1 · What is an FDN?

A room, one ray at a time

A room, as a signal flow

What an FDN is good at, and what it is not

Strengths

• parametric by design — decay, color and density are knobs, not samples
• cost is independent of the decay time
• one FDN per room, shared by all sources and receivers
• time-varying design is cheap
• small and differentiable, so it can be optimized

Limitations

• it approximates a room, it does not reproduce a measured one
• early reflections placement is entanglement with global structure
• a badly chosen matrix or delay set rings and colors the tail
• the parameters map to what you hear only indirectly

Used in modern frameworks

MPEG-I Immersive Audio

The vanilla structure

Input gain B, a bank of \(N\) delays, the feedback matrix A, output gain C, and the direct path D around all of it. Four blocks and a loop — that is the whole vanilla FDN.

One FDN, the core parameters

Three places to hang a filter

Same structure, three hooks added: the points where pyFDN.process_fdn will take an arbitrary filter — the openings every later structure in this tutorial goes through.

No hooks — the lossless loop

Nothing in the loop but the orthogonal matrix A: energy goes round without being lost. Every hook that follows is a way of taking that energy back.

post_delay — absorption

One shelving filter per delay line, \(T_{60}\) of 3.5 s at DC falling to 1.2 s at Nyquist. Watch the top of the spectrogram empty out first — that tilt is the absorption.

post_output — output EQ

A 10-band gain_to_geq voicing on the wet signal only: \(+5\) dB low, \(-14\) dB at Nyquist. It changes the color of the reverb without touching the decay — the tail is still 3.5 s long.

post_matrix — time-variation

td.TimeVaryingMatrix(N, 10 Hz, 1.1 rad, fs, spread=0.7) — the same hook, but the matrix moves. Orthogonal at every sample, so the decay is unchanged.

post_matrix — a non-linearity

post_matrix=td.ControllableFullWaveRect(N, 0.25, [4,5,6,7]) — half the lines partly rectified.

→ One of six shimmer operators new in pyFDN 0.4.2 — Dal Santo et al., Shimmer Reverberation with Nonlinear FDNs, at this conference.

Beyond the vanilla FDN — toward arbitrary structures

Three connectors, in both backends (td.Series / flamo.Series, …). The vanilla FDN is one Recursion inside a Series inside a Parallel — nothing in the library says “FDN”.

• Put the mixing chain in the forward path → early reflections
• Put an FDN inside the forward path of another → allpass FDN
• Let the feedback path be a measured room → reverberation enhancement

Early reflections — swap the loop paths

• swap_flamo_recursion_paths exchanges the two branches of the Recursion: the short delay → matrix → short delay chain lands in the forward path, and only the long delays recirculate
• Buys echo density early, where a vanilla FDN is still sparse — early-reflection control without a separate FIR stage

Allpass — an FDN inside an FDN

Two nested recursions. The inner loop is a homogeneous MIMO allpass FDN; it and the absorption filter form the forward path of the outer loop, whose feedback path is the main delay bank.

With only 4 main delays:

Reverberation Enhancement System — the room closes the loop

Microphones feed the FDN, the FDN feeds the loudspeakers, and the room’s loudspeaker-to-microphone responses are the feedback path.

So what is actually hard?

Choose a structure

Which topology — and how do you compare it against another, fairly?

Derive the parameters

From physics, from perceptual features, from a reference signal — or from intuition?

Evaluate it

Objectives, complexity, quality — measured consistently, against strong baselines.

Present it

Audible, reproducible, transparent: results someone else can hear and re-run.

2 · pyFDN, the tour

A toolbox turns FDNs into a shared laboratory

For research

• compare structures through one representation instead of rewriting plumbing
• reproduce figures, listening tests and optimization runs from executable code
• move between time, frequency, modal and differentiable views without changing the underlying design
• turn a paper into a tested constructor, translator or objective that others can extend

For education

• connect the DSP objects students know — delays, filters, poles and matrices — to something they can hear immediately
• expose failure modes: coloration, slow mixing and unstable optimization
• bridge DSP analysis with DDSP design: derive what has a closed form, then optimize what does not
• make every exercise inspectable, editable and repeatable

marimo keeps an experiment and its explanation in sync

What it is

marimo is an open-source reactive Python notebook.

• notebooks are ordinary .py files
• cells form a dependency graph
• changing a value reruns its dependents, or marks them stale
• the same file can run as a notebook, script or web app

Why we use it here

• no hidden “which cell did I run?” state during a live tutorial
• a slider can drive the delays, matrix or decay and update every downstream plot and audio player
• Git diffs and code review stay readable
• examples run in CI and render to a browser-friendly gallery
• one artifact serves teaching, research and documentation

marimo edit examples/     # explore and modify
python example.py         # reproduce non-interactively

Design principles

Functions, not a framework

• flat namespace: everything is pyFDN.<thing>
• NumPy in, NumPy out
• no session object, no builder to learn

Torch only where it earns its place

• analysis and design: pure NumPy/SciPy
• differentiable models: FLAMO (PyTorch)
• one translator between the two: dss_to_flamo

Reference-tested

• ported from the MATLAB FDN Toolbox, with .mat reference fixtures asserting parity
• every example notebook runs in CI
• the API reference cannot silently drift

Notebooks are the documentation

• ~30 marimo notebooks, rendered into the website
• each one is a real .py file — diffable, testable, importable

Open the examples gallery — every one of these is rendered with its outputs and audio.

Three backends, one interface

	process_fdn	td	FLAMO
What it is	the FDN recursion, hard-coded	a graph of block operators	torch.nn.Modules
Domain	time, block-wise	time, block-wise	frequency (FFT)
Structure	vanilla + three hooks	anything you can wire	anything you can wire
Gives you	no dependencies, fast to reason about	streaming, time-varying, non-linear	gradients
Costs you	one topology	no gradients	torch, and an LTI assumption

Same vocabulary in all three: Series, Parallel, Recursion, and a block-processing object with a .filter(block) method.

FDNBuild is what moves between them — design in one, render in another, train in the third, and it is the same FDN.

The map

Module	What is in it
generate	matrices, delays, allpass, scattering, SDN
td	stateful block-processing graph — no torch, no FFT
translate	representation changes
train	differentiable optimization
eq	absorption & equalization filter design
auxiliary	acoustics, math, plotting, FLAMO glue
presets, build_io	trained FDNs on disk, and your own
resources	packaged test signals and the bibliography

One data class holds an FDN

build = pyFDN.fdn_build_gallery(
    8,                      # eight delay lines
    fs=48_000,
    rt=2.0,                 # T60 at DC
    rt_nyquist=0.5,         # T60 at Nyquist → per-line absorption filters
    io_type="ones",
    rng=42,
)

build.delays    # (N,)              delay lengths in samples
build.A         # (N, N)            feedback matrix
build.B         # (N, n_in)         input gains
build.C         # (n_out, N)        output gains
build.D         # (n_out, n_in)     direct path
build.post_delay   # (n_sos, 6, N)     per-line absorption, or None if lossless
build.post_matrix  # (n_sos, 6, N)     a filter in the feedback path, or None
build.post_output  # (n_sos, 6, n_out) output equalization, or None

FDNBuild is the common currency: every constructor returns one, every translator accepts one, and extract_build(model) gets one back out of a trained model.

Five representations, one hub

A reverb in five lines

import numpy as np, pyFDN
from pyFDN import td

build = pyFDN.fdn_build_gallery(8, fs=48_000, rt=1.8, rt_nyquist=0.4, rng=0)
dry, fs = pyFDN.load_audio("synth_dry")
wet = pyFDN.process_fdn(
    np.pad(dry, (0, 2 * fs)),                       # room for the tail
    build.delays, build.A, build.B, build.C, build.D,
    post_delay=td.SOSBank(build.post_delay),        # the absorption the gallery designed
)

Visualize the model and the build

From design to real-time — arrays to a plugin

Everything so far is offline: Python, torch, FFT-domain recursion. None of it runs in a DAW. adac closes that gap — it walks the FLAMO graph and emits FAUST.

import adac

config = adac.flamo_to_json(model, fs, name="PyFDNReverb")   # graph → JSON IR
cert   = adac.certify(config)                                # stability certificate
faust  = adac.json_to_faust(config, controls={"rt60": True, "dry_wet": True})

From there: the FAUST web IDE, faust -o cpp, faust2juce, or adac.export_juce for a VST3. pip install adac — NumPy only.

Open example_fdn_to_faust in the examples gallery — the same three calls, with the emitted FAUST source rendered underneath.

Franchino & Schlecht, Compiling Differentiable Audio Graphs to Real-Time DSP, DAFx 2026 (arXiv:2606.21277)

3 · What to do next?

The frontier, with notebooks

Direction	Idea	Notebook
Allpass FDNs	uniallpass conditions; nested/series/Poletti structures	example_allpass_FDN_*
Paraunitary	filter feedback matrices; delay & velvet-noise matrices	example_paraunitary_fdn
SDN	scattering delay networks — geometry-driven	example_sdn
Coupled rooms	multi-slope decay, non-exponential EDCs	example_coupled_rooms, example_multislope_rir_to_fdn
Time-varying	modulated matrices, stability under modulation	example_time_varying_fdn
Shimmer	rectifiers, ring modulation and pitch shifting in the loop	example_shimmer_fdn
Decorrelation	multi-output FDNs for spatial rendering	example_decorrelation

DecayFitNet is now a pip install

pip install multislope

import multislope

net = multislope.DecayFitNet(sample_rate=fs)
fit = net.estimate(rir)             # per octave band

PyPI · source · MIT

DecayFitNet is by Georg Götz, with Sebastian J. Schlecht and Ville Pulkki.

Open research questions

• ‘best architecture’ for a given room — how to choose delays, matrix, and sub-structure
• perceptual objectives — what is the right loss
• differentiable delay optimization — what loss?
• differentiable time-domain optimization
• positional, directional and dynamic scenes
• using the early reflections of the FDN
• conditional generation (using speech, music, text or a room embedding to derive the FDN parameters)

How to contribute

As a developer

Help maintain and stabilize.

• take an open issue — backends, filters, config files, real-time paths
• tests, docs, packaging and CI count as much as DSP
• review pull requests; we need more eyes than hands

As a student

Pick a project off the reverb list.

• issue #123 collects project-sized topics: spring and plate reverb, SDN, velvet noise, binaural FDNs, fade-in FDNs, echo density …
• each one lands as a new .py notebook in examples/
• good scope for a semester project or a thesis chapter

As a researcher

Help integrate your research.

• bring your method in as a hook, not a fork
• a reference implementation inside pyFDN gets used, compared and cited
• talk to us early — the API can bend toward your structure

All of it happens in the open at github.com/artificial-audio/pyFDN — take an issue, or open one.

4 · Hands-on 1 — build an FDN

Open the starter notebook before we begin

A — molab in your browser (recommended)

1. Create an account or sign in at molab.marimo.io
2. Open the pyFDN starter notebook
3. Choose Fork to put an editable copy in your workspace
4. Start the runtime and run all cells once

Do this before the tutorial. Nothing needs to be installed locally.

B — local Python (optional)

uv venv --python 3.11
source .venv/bin/activate
uv pip install "pyfdn[examples]"
git clone \
  https://github.com/artificial-audio/pyFDN
cd pyFDN
marimo edit examples/

import pyFDN
print(pyFDN.__version__)

torch via flamo is the large download, and only Hands-on 2 needs it. Do this before arriving.

Fallback: every notebook is also rendered in the pyFDN examples gallery.

The plan

One notebook, examples/example_process_fdn.py, seven steps — three knobs, the wiring between them, and two ways past the vanilla FDN:

1. Delays (knob 1) — choose the lengths, hear the density
2. Matrix (knob 2) — swap the feedback matrix, measure the mixing
3. Gains — wire the input and output, and render your first impulse response; a second row of output gains makes it stereo
4. Decay (knob 3) — prescribe \(T_{60}\) per band, then check you got it
5. Analyze and listen — EDC, echo density, spectrogram, and your own ears
6. Hooks — one filter in the loop, and the same FDN is a different reverb
7. Match a real room — turn all three knobs at once against a measured hall

Open example_process_fdn.py in marimo now and run all cells. Everything in this section is plain NumPy — no torch, no FLAMO. Every code cell ends with a Try this block; that is where the exercises live.

Knob 1 — the delays set the density

• each delay line adds its own set of resonances — longer delays, more of them
• together they set the density: short delays sound like a small box, long ones like a sparse, granular hall
• echo density builds up over time, faster the more lines you have
• lengths sharing a common factor make echoes coincide — the metallic ring; coprime lengths avoid it
• physical anchor: the mean free path of the room you are imitating

Knob 2 — the matrix mixes

• Lossless means \(\mathbf{A}\) is orthogonal — or diagonally similar to one. is_orthogonal and is_unilossless check it; it is not a matter of faith
• Orthogonal, Hadamard, circulant, Householder — same losslessness, different mixing speed and different cost
• Sparse and structured matrices buy back the \(N \times N\) multiply

Mixing is audible

Normalized echo density (Abel & Huang 2006) reaches 1 when the response is statistically Gaussian — that is “mixed”.

• orthogonal: mixes at ~250 ms
• permutation: never mixes — a single long comb filter

Same delays, same decay, same losslessness. Only the matrix changed.

Play both impulse responses. The permutation FDN is the sound of a bad reverb.

Knob 3 — absorption sets the decay

Broadband — attenuate a little on every trip round the loop, proportional to the delay length, so short and long lines die together:

g = pyFDN.rt_to_gain_per_sample(rt, fs)
A_lossy = np.diag(g ** delays) @ A

Frequency-dependent — replace each scalar gain with a filter per delay line whose attenuation follows the target \(T_{60}\) across frequency:

sos = pyFDN.decay_to_geq(
    target_rt, delays, fs
)

Proportional-to-delay is what makes the decay homogeneous, and therefore predictable — a graphic-EQ absorption design preserves that per band.

What decay looks like

• Impulse response (\(\mu\)-law compressed, so the tail stays visible)
• Energy decay curve = backward energy integral; the \(-60\) dB crossing is \(T_{60}\)

Step 1 — delays

delays = pyFDN.sample_delay_lengths(
    N=8,
    delay_range=(1000, 3000),        # samples ≈ 21–62 ms at 48 kHz
    distribution="geometric",        # or "uniform"
    coprime=True,                    # avoid coinciding echoes
    rng=2,
)
# or set them by hand, in milliseconds:
delays = pyFDN.ms_to_smp(np.array([20, 27, 31, 37, 43, 53, 61, 71]), fs)

Try three delay sets and listen to each:

• short & narrow (300, 600) — flutter, obvious pitch
• long & wide (2000, 9000) — sparse early reflections, granular onset
• coprime vs not — set coprime=False and listen for the ringing

Then look at pyFDN.echo_density(ir, fs=fs) for each.

Step 2 — the matrix

for matrix_type in pyFDN.fdn_matrix_gallery():        # list the options
    A = pyFDN.fdn_matrix_gallery(8, matrix_type)
    print(f"{matrix_type:32s} lossless: {pyFDN.is_unilossless(A)}")

A = pyFDN.fdn_matrix_gallery(8, "orthogonal")

Swap in "permutation", "Householder", "Hadamard" and "orthogonal". For each: listen, then measure the mixing time with pyFDN.echo_density. Which one mixes fastest? Which is cheapest to compute?

The notebook prints the mixing time of an orthogonal and a permutation FDN with the same delays: one mixes in half a second, the other never does.

Step 3 — gains, and your first render

B = np.ones((8, 1)) / np.sqrt(8)      # input  → delay lines
C = np.ones((1, 8)) / np.sqrt(8)      # delay lines → output
D = np.zeros((1, 1))                  # direct path: wet only

ir = pyFDN.dss_to_impz(3 * fs, delays, A, B, C, D).squeeze()

That is a complete FDN — delays, A, B, C, D is the delay state space. With an orthogonal A and nothing in the loop it is lossless, so the tail never decays. Listen to it once; it is the sound of energy with nowhere to go.

Set D = 0.5 to mix the dry signal back in, then drive a single delay line with B = np.eye(8)[:, :1] and listen to how much later the tail fills in.

Stereo — one tail, two taps

duplicated

orthogonal

C = pyFDN.random_orthogonal(N)[:2]    # (2, N)
D = np.zeros((2, 1))

Same loop, same decay — only the tap differs. The rows have to be orthogonal, not merely different.

Step 4 — decay you can specify

# broadband: bake a homogeneous gain into the loop
g = pyFDN.rt_to_gain_per_sample(rt=1.8, fs=fs)
A_lossy = np.diag(g ** delays) @ A

# frequency-dependent: one graphic-EQ absorption filter per line, inside the loop
target_rt = np.array([2.4, 2.4, 2.3, 2.1, 1.8, 1.4, 1.0, 0.7, 0.5, 0.5])
absorption = pyFDN.decay_to_geq(target_rt, delays, fs)

build = pyFDN.FDNBuild(A=A, B=B, C=C, D=D, delays=delays, fs=fs, post_delay=absorption)
ir = pyFDN.build_to_impz(build, 3 * fs).squeeze()

Then check that you got what you asked for:

rt_measured, f_center = pyFDN.estimate_rt_bands(ir, fs)
print(np.round(rt_measured, 2))      # ← compare against target_rt[1:9]

Ask for something extreme — 4 s at 63 Hz and 0.2 s at 8 kHz — and see where the design stops delivering. Why does it break?

Step 5 — analyze and listen

pyFDN.plot_FDN_build(build)
pyFDN.plot_impulse_response(ir, fs=fs)
pyFDN.plot_edc(ir, fs=fs)
pyFDN.plot_spectrogram(ir, fs)
pyFDN.echo_density(ir, fs=fs)

dry, _ = pyFDN.load_audio("synth_dry", fs=fs)
wet = pyFDN.process_fdn(
    np.pad(dry, (0, 2 * fs)),             # room for the tail
    delays, A, B, C, D,
    post_delay=td.SOSBank(absorption),    # absorption in the loop
)

Checkpoint. Get an FDN you like the sound of. Then break it: make it metallic on purpose, and say which knob did it.

Step 6 — the hooks, one more turn

process_fdn takes a filter at three points in the loop, and each one is a whole family of reverbs. Any object with a .filter(block) method fits.

wet = pyFDN.process_fdn(
    x, delays, A, B, C, D,
    post_delay=td.SOSBank(absorption),                     # absorption
    post_matrix=td.TimeVaryingMatrix(8, 10.0, 1.1, fs, 0.7),  # a moving matrix
)

Swap post_matrix for td.ControllableFullWaveRect(8, 0.25, [4, 5, 6, 7]) — half the lines partly rectified. The harmonics you hear were never in the input; the FDN is generating them, and there is no transfer function left to plot. Take alpha up toward 1.0 and listen to how far you can push it before it stops sounding like a room.

Same three hooks as the figures in part 1: post_delay, post_matrix, post_output.

Step 7 — match a real room

Pori concert hall — pyFDN.load_audio("s3_r4_o")

Last section of the notebook: turn the knobs you already have until you get as close as you can. Decay first — the color mostly follows. Mixing time last.

5 · Hands-on 2 — match a room with gradients

Open the second notebook

examples/example_train_fdn_to_rir.py — the same hall you just chased by hand, this time fitted by an optimizer.

1. Open it in molab, Fork, run all cells
2. Two dropdowns at the top decide everything downstream:
   • Run on — CPU (GPU is not supported at the moment)
   • EQ design — a first-order shelf (2 numbers) or a ten-band graphic EQ (10)
3. 300 Adam steps, float32, \(2^{17}\)-point grid:
   • ≈ 90 s on a laptop CPU

Start the run now. The rest of this section is a commentary on what it is doing while it does it.

Why gradients

What we can do analytically

• delays → modal density
• absorption filters → \(T_{60}(\omega)\)
• unitary matrix → losslessness

These are invertible designs: state the target, get the parameters. That was the last slide of Hands-on 1, and it got you close.

What we cannot

• “make it not sound metallic”
• “match this hall, all of it, at once”
• “stay flat and mix fast and use a sparse matrix”

No closed form. But every one of them is a differentiable loss.

Write the objective, and let autograd find the direction of improvement.

The plan

One notebook, one target, seven steps — and three of them are the ones that actually decide whether this works:

1. The target, and the yardstick we will not let the fit see
2. The initialization — a generic 1 s reverberator that knows nothing about the room
3. The parametrization — what has a gradient, and in what units ⚠️
4. The loss — the obvious one is wrong, and there is a table to prove it ⚠️
5. Train — 300 Adam steps, one loss curve
6. Read the answer out — two filters, both plottable
7. Measure — octave-band \(T_{60}\) and level, against numbers the fit never saw

Everything here needs torch, via FLAMO. If the install did not work, the notebook is rendered in full with its plots and audio.

FLAMO in one slide

FLAMO — frequency-domain differentiable audio processing (Dal Santo, De Bortoli, Prawda, Schlecht & Välimäki, ICASSP 2025)

• audio modules as torch.nn.Modules: Delay, Gain, Filter, SVF, …
• composition primitives: Series, Parallel, Recursion, Shell
• the loop is closed in the frequency domain — a Recursion becomes a matrix inverse over FFT bins, so there is no per-sample unrolling and no vanishing-gradient problem over a 3-second tail

model = pyFDN.trainable_from_build(build, nfft=2**17, device=device)  # cpu or cuda
pyFDN.plot_flamo_graph(model)

pyFDN’s job is to hand FLAMO a correct FDN — admissible matrices, homogeneous decay, sane initialization — and to read the result back out.

Step 1 — the target, and the test

Pori concert hall — pyFDN.load_audio("s3_r4_o"), trimmed to the direct sound, normalized to unit energy

est_rt, f_centre = pyFDN.estimate_rt_bands(rir, fs)
est_level, _ = pyFDN.estimate_initial_level_bands(
    rir, est_rt, fs
)
# 2.8 s at 63 Hz → 1.2 s at 8 kHz

In Hands-on 1 these two lines were the design. Here they are the test: computed once, fed to nothing, and compared against at the very end.

That separation is the whole reason the result means anything. A fit scored on the quantity it was optimizing tells you only that the optimizer works.

Step 2 — a reverberator that knows nothing

init_build = pyFDN.fdn_build_gallery(
    delays=pyFDN.sample_delay_lengths(
        16, (700, 2500), distribution="geometric", coprime=True, rng=1
    ),
    fs=fs, io_type="normalized", direct_gain=0.0,
    rt=1.0, rt_nyquist=1.0,        # flat 1 s — the entire prior knowledge of the room
    rng=0,
)

A complete FDN: random orthogonal matrix, normalized gains, flat 1 s absorption.

The one thing the measurement is allowed to set before the optimizer runs is the overall energy — one scalar on the output gain. Comment it out and watch the first fifty steps get spent on a volume knob.

Step 3 — parametrize by what you would plot

post_delay = pyFDN.DecayFilter(
    1.0, delays, fs, design=design
)
post_output = pyFDN.OutputEQ(
    0.0, 1, fs, design=design
)

what	trained as
decay, post_delay	\(T_{60}\) in seconds
color, post_output	gain in dB
mixing, \(\mathbf{A}\)	on \(SO(N)\)
level, \(\mathbf{B}\,\mathbf{C}\,\mathbf{D}\)	directly
the delays	fixed — integers

Train the attenuation filter’s coefficients and it diverges within a few steps, at any learning rate. Parametrize by the reverberation time instead:

\[\mathrm{RT}_k \to \underbrace{\tfrac{-60\,d_i}{\mathrm{RT}_k f_s}}_\text{dB per round trip} \to \text{biquads}\]

A positive RT is a contractive loop for every value the parameter can take. The stability constraint is gone, not enforced — and the same trick puts the output EQ in dB.

Step 4 — why not a spectrogram distance

Freeze everything but the decay, scale the room’s measured \(T_{60}\) by a constant, and score it on a mel multi-resolution spectrogram distance:

\(T_{60}\) scale	0.4	0.6	1.0	1.3	1.6
mel MSS (\(\times 10^{-5}\))	1.819	1.804 ← minimum	1.855	1.924	2.009
energy decay at 1 s	−67 dB	−47 dB	−31 dB	−25 dB	−21 dB
the room, at 1 s			−29 dB

An FDN whose tail is 18 dB below the room’s scores better than the one that tracks it to within 2 dB. Not a bug: two rooms with the same decay still have uncorrelated fine structure, and against detail you cannot predict, silence is a better guess than the right amount of the wrong detail.

Step 4 — a loss that can see a decay

loss = pyFDN.MatchCumulativeEnergy(rir, window=1024, power=0.5, frequency="both")

\[E[f, t] \;=\; \sum_{t' \ge t} \; \sum_{f' \ge f} \big| S[f', t'] \big|^2\]

• backwards in time = Schroeder integration — the decay itself
• along frequency = what octave bands do, without band edges: every bin is compared against every wider band containing it

• read the \(t=0\) edge → the integrated spectrum
• read the \(f=0\) edge → the energy decay curve
• one term, no weight to tune — there is nothing to weigh it against

frequency="descending" cumulates high→low only. The bottom octave loses its gradient and the fit abandons it: 63 Hz lands at 0.26 s against the room’s 2.8 s. Run it and look at the trained \(T_{60}\) curve.

Steps 5–7 — train, read out, measure

the room

flat 1 s start

trained

From a flat 1 s decay and a flat EQ: 52 % → 10 % mean \(T_{60}\) error, and 1.71 dB → 0.79 dB of band-level shape. Nothing the estimators report was ever shown to the optimizer.

The deliverable

trained_build = pyFDN.extract_build(model)      # plain NumPy, no torch

Both trained filters come back baked as ordinary SOS banks in post_delay and post_output. Nothing about the build remembers that an optimizer produced it — so it goes straight into build_to_impz, process_fdn, or the FAUST export from part 2.

Run dry audio through the reverb you just trained. Then reopen example_process_fdn and A/B it against the one you tuned by hand.

Thank you, pyFDN contributors

Sebastian J. Schlecht

Facundo Franchino

Jeremy Bai

Alma Hova

Orchisama Das

Bharadwaj Lakuduva Suresh Babu

Cristóbal Andrade

Gloria Dal Santo

Gian Marco De Bortoli

Questions

Tutorial: artificial-audio.github.io/pyfdn-tutorial-2026 · pyFDN docs: artificial-audio.github.io/pyFDN