
1 Introduction
Fourier transforms are the backbone of signal processing and engineering, with profound impli-
cations to nearly every field of scientific computing and technology. This is primarily due to the
discovery of the well-known Fast Fourier Transform (FFT) algorithm [CT65], which is ubiquitous
in engineering applications, from image and audio processing to fast integer multiplication and op-
timization. The classic FFT algorithm of [CT65] computes the Discrete Fourier Transform (DFT)
of a length-nvector x, where both the time and frequency domains are assumed to be discrete. This
algorithm takes O(n) samples in the time domain, and constructs bx= DFT(x) in O(nlog(n)) time.
The discrete setting of DFT limits its applicability in two main aspects: The first one is that many
real-world signals are continuous (analog) by nature; Secondly, many real-world applications (such
as image processing) involve signals which are sparse in the frequency domain (i.e., kbxk0=kn)
[ITU92,Wat94,Rab02]. This feature underlies the compressed sensing paradigm [CRT06], which
leverages sparsity to obtain sublinear algorithms for signal reconstruction, with time and sample
complexity depending only on the sparsity k. Unfortunately, the continuous case cannot simply be
reduced to the discrete case via standard discretization (i.e., using a sliding-window function), as
it “smears out” the frequencies and blows up the sparsity, which motivates a more direct approach
for the continuous problem [PS15].
The study of Fourier-sparse signals dates back to the work of Prony in 1795 [dP95], who studied
the problem of exact recovery of the “ground-truth” signal xin the vanilla noiseless setting. By
contrast, the realistic setting of reconstruction from noisy-samples [PS15] is a different ballgame,
and exact recovery is generally impossible [Moi15]. In the Fourier Interpolation problem, the
ground-truth signal
x∗(t) =
k
X
j=1
vje2πifjt, vj∈C, fj∈[−F, F ]∀j∈[k],
is a k-Fourier-sparse signal with bandlimit F. Given noisy access to the ground truth x(t) = x∗(t)+
g(t) in limited time duration t∈[0, T ] (which means that we need to recover x∗(t) by taking samples
from x(t)), the goal is to reconstruct a e
k-Fourier-sparse signal y(t) (i.e., y(t) = Pe
k
j=1 evje2πi
e
fjtfor
some evj∈C,e
fj∈[−F, F ] for all j∈[e
k]) such that
ky(t)−x∗(t)k2
T≤c(kgk2
T+δkx∗(t)k2
T)
holds for some c=O(1), where the T-norm of any function f:R→Cis defined as
kf(t)k2
T:= 1
TZT
0|f(t)|2dt.
We note that it is not necessary for y(t)’s frequencies and magnitudes ( e
fj,evj) being close to the
ground-truth signal x∗(t)’s frequencies and magnitudes (fj0, vj0).
Prior to this work, the state-of-the-art algorithm for the Fourier interpolation problem was given
by [CKPS16], which achieves e
O(k51) sample complexity, e
O(k10ω+40) running time, e
O(k10) output
sparsity, and c≥2000 approximation ratio. In [SSWZ22], the approximation ratio was improved to
≈1 + √2, but the sample complexity remained large, and runtime remained slow. For calibration,
we note that o(k4) sample complexity for Fourier interpolation is not known to be achievable even
with exponential decoding time. In this work, we focus on improving the efficiency of [CKPS16]’s
algorithm across all aspects: (i) runtime, (ii) sample complexity, and (iii) output-sparsity. Our
main result is:
1