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Upsampling

Upsampling is a technique in digital signal processing used to increase the sampling rate of a discrete-time signal, typically by an integer factor L \geq 2, through the insertion of L-1 zeros between consecutive original samples, followed by low-pass filtering to interpolate the new samples and mitigate imaging artifacts in the frequency domain.^[1]^[2] This process, also known as expansion or interpolation in multirate systems, compresses the signal's spectrum by the factor L while replicating it L times around the unit circle in the z-domain, necessitating the anti-imaging filter with a cutoff at \pi/L and gain L to preserve the original baseband content.^[1] In practice, upsampling without subsequent filtering results in a zero-stuffed signal prone to high-frequency distortions, whereas full interpolation yields a smoother, higher-rate approximation of the continuous-time signal.^[2] Beyond one-dimensional signals, upsampling extends to two-dimensional domains like digital image processing, where it enlarges an image's pixel array by inserting zeros into the grid and applying spatial filters—such as bilinear or bicubic interpolation—to estimate intermediate pixel values, thereby enhancing resolution for tasks like image zooming or pyramid reconstruction.^[3] In machine learning contexts, upsampling addresses dataset imbalances by artificially increasing instances of underrepresented classes through replication or synthetic generation, which helps prevent biased model predictions and boosts classification performance on skewed data distributions.^[4] Key applications of upsampling span audio rate conversion to match playback devices, telecommunications for efficient signal modulation and demodulation, and computer vision for super-resolution imaging, underscoring its role in enabling flexible, high-fidelity digital representations across engineering disciplines.^[1]^[3]

Fundamentals

Definition and Purpose

Upsampling is the process of increasing the sampling rate of a discrete-time signal, where the sampling rate F_s represents the number of samples acquired per unit time, typically measured in hertz (Hz). This conversion elevates the signal from its original rate F to a higher rate F' > F, achieved by an integer or rational factor L > 1, yielding a denser, higher-resolution approximation of the original continuous-time signal.^[5] The technique serves several key purposes in signal processing, including improving signal smoothness via finer temporal resolution for enhanced interpolation and data analysis; mitigating aliasing artifacts in downstream operations by broadening the available frequency spectrum; and enabling compatibility with systems requiring elevated rates, such as in digital audio where low-rate inputs like 44.1 kHz (common for compact discs) are upconverted to 48 kHz for high-fidelity storage, mixing, or playback.^[5]^[6] Historically, upsampling emerged in the 1970s amid growing interest in efficient multirate digital signal processing, with foundational advancements by Ronald E. Crochiere and Lawrence R. Rabiner in their 1975 exploration of novel rate-conversion methods for telecommunications and speech applications. Their 1981 comprehensive tutorial on interpolation and decimation further established upsampling as a core tool for practical DSP implementations.^[7]^[5] This section assumes familiarity with basic discrete-time signal concepts, such as sequences representing sampled continuous phenomena.

Mathematical Representation

In digital signal processing, upsampling by an integer factor L transforms a discrete-time signal x into an expanded signal y at a higher sampling rate. The time-domain input-output relationship is defined as

y = \begin{cases} x\left[\frac{m}{L}\right] & \text{if } m \text{ is a multiple of } L, \\ 0 & \text{otherwise}. \end{cases}

This can also be expressed using the Kronecker delta function \delta[\cdot] as

y = \sum_{k=-\infty}^{\infty} x \, \delta[m - kL],

where \delta = 1 if m = 0 and $0 otherwise.^[8]^[9] In the frequency domain, the discrete-time Fourier transform (DTFT) of the upsampled signal Y(e^{j\omega}) relates to the original signal's DTFT X(e^{j\omega}) by

Y(e^{j\omega}) = X(e^{j L \omega}),

which compresses the spectrum of X(e^{j\omega}) by a factor of L and introduces L periodic repetitions (spectral images) within the principal period [-\pi, \pi].^[8]^[2] These spectral images, centered at multiples of the original normalized sampling frequency scaled by $1/L (i.e., imaging frequencies at \omega = 2\pi k / L for integer k), represent distortions that can cause aliasing or overlap if not addressed, necessitating subsequent low-pass filtering to retain only the baseband spectrum.^[8] For non-integer (rational) upsampling by a factor L/M where L and M are coprime integers, the process generalizes to a cascade of upsampling by L followed by downsampling by M, with appropriate filtering to mitigate imaging and aliasing effects, though the detailed model combines the above operations without a single closed-form time-domain expression here.^[9]

Integer Factor Upsampling

Zero-Insertion Process

The zero-insertion process serves as the foundational step in integer factor upsampling within digital signal processing, where an integer number L-1 of zero-valued samples are inserted between each pair of consecutive samples from the original discrete-time signal x. This produces an expanded sequence y, formally defined such that y = x when m = nL for integer n, and y = 0 otherwise, with m = nL + k for k = 0, 1, ..., L-1.^[10] The procedure systematically spaces the original samples farther apart in the time index while filling the intervening positions with zeros, thereby increasing the length of the sequence by a factor of L without altering the values of the original samples.^[1] In the time domain, this insertion effectively expands the signal's temporal resolution by reducing the implied sampling period from T_s = 1/F_s (where F_s is the original sampling frequency) to T_s' = 1/(L F_s), which corresponds to a new sampling frequency of L F_s. The introduction of zeros preserves the sum of the squared sample values equal to that of the original signal but dilutes the average power by a factor of 1/L due to the increased sequence length; this dilution is compensated by the gain of L in the subsequent low-pass interpolation filter to restore the original average power.^[9] From a spectral perspective, the zero-insertion operation compresses the baseband spectrum of the original signal by a factor of L, while simultaneously generating L-1 unwanted image bands that replicate the compressed baseband at intervals corresponding to multiples of the original Nyquist frequency. The magnitude response of the discrete-time Fourier transform (DTFT) for the upsampled signal in the baseband region is expressed as

|Y(e^{j \omega})| = |X(e^{j L \omega})| \quad \text{for} \quad |\omega| < \frac{\pi}{L},

where X(e^{j \omega}) denotes the DTFT of the original signal x.^[1] These images arise due to the periodic nature of the DTFT and the stretching effect of the insertion, effectively repeating the spectrum L times within the new normalized frequency range of -\pi to \pi. A key limitation of the zero-insertion process is the introduction of these high-frequency image artifacts, which overlap with and distort the desired baseband signal unless mitigated, leading to aliasing-like corruption in the reconstructed output. For instance, with L=2, the original spectrum is compressed along the frequency axis, and a single image band appears centered at the new Nyquist frequency \pi (in normalized units), effectively doubling the number of spectral replicas and amplifying unwanted high-frequency components.^[1] This procedural step, while straightforward to implement, thus requires careful consideration of its spectral implications to avoid signal degradation in applications such as audio processing or communications systems.

Low-Pass Interpolation Filtering

In integer factor upsampling, the low-pass interpolation filtering stage follows the zero-insertion process to mitigate the spectral images introduced by inserting zeros between original samples, thereby reconstructing a smooth, bandlimited signal at the higher sampling rate.^[9] This filter is designed as an ideal low-pass with a normalized cutoff frequency of \pi / L, where L is the integer upsampling factor, ensuring that the baseband signal (up to the original Nyquist frequency) passes unattenuated while attenuating the replicated images beyond this cutoff. Additionally, the filter incorporates an ideal gain of L to compensate for the amplitude dilution caused by the zero insertions, restoring the overall signal power to match the original.^[9] The interpolation mechanism operates by estimating signal values at the newly inserted sample points through a process equivalent to sinc interpolation of the original signal, leveraging the Nyquist-Shannon sampling theorem to recover the underlying continuous-time waveform without introducing distortion in the passband. In practice, the filter's impulse response h approximates the ideal sinc function, h = L \cdot \frac{\sin(\pi m / L)}{\pi m / L}, which provides the theoretical basis for perfect reconstruction assuming the original signal is bandlimited.^[9] The basic implementation involves convolving the upsampled signal y—which consists of the original samples separated by L-1 zeros—with the filter's impulse response h, yielding the interpolated output z = \sum_{k=-\infty}^{\infty} y h[m - k]. This discrete convolution effectively smooths the abrupt transitions from the zero-inserted sequence, producing a higher-resolution signal suitable for subsequent processing or output.^[9] For an upsampling factor of L=2, assuming an original sampling frequency F_s, the filter attenuates frequencies above F_s / 2 (the original Nyquist frequency, corresponding to the normalized cutoff \pi / 2 at the new rate of $2F_s), while the new Nyquist limit becomes F_s, thereby preserving the original content up to F_s / 2 without imaging artifacts.^[9]

Interpolation Filter Design

Ideal Low-Pass Filter Characteristics

The ideal low-pass filter for distortion-free upsampling by an integer factor L exhibits a rectangular magnitude frequency response |H(e^{j\omega})| = L in the passband for |\omega| < \pi/L, and |H(e^{j\omega})| = 0 in the stopband for \pi/L < |\omega| \leq \pi.^[11]^[12] This design preserves the original signal's spectrum below the cutoff while attenuating the spectral images introduced by zero-insertion upsampling, with the passband gain of L compensating for the $1/L spectral scaling effect of the upsampling process.^[13] The filter also maintains a linear phase response across the passband to ensure no additional phase distortion in the interpolated output.^[14] The impulse response of this ideal filter is an infinite-length sinc function, given by

h = \begin{cases} 1 & n = 0 \\ \frac{\sin(\pi n / L)}{\pi n / L} & n \neq 0 \end{cases},

which derives from the inverse discrete-time Fourier transform (DTFT) of the rectangular frequency response.^[15] This form enables exact interpolation at the new sampling points by weighting contributions from all input samples according to their temporal distance. The transition band of the ideal filter has zero width, sharply demarcating the passband edge at \omega = \pi/L, which corresponds to the Nyquist frequency of the original sampling rate F_s when normalized to the upsampled rate L F_s.^[11] This precise cutoff aligns with the requirement to retain frequencies up to F_s/2 without allowing imaging artifacts to fold back into the baseband. In theory, this filter achieves perfect bandlimited interpolation, reconstructing the continuous-time signal exactly at the higher sampling rate as per the Shannon sampling theorem, provided the original signal is bandlimited to below F_s/2.^[16]^[17] However, its infinite impulse response support renders it impractical for digital implementation, necessitating approximations in real systems. In the context of integer factor upsampling, this ideal filter follows zero-insertion to yield seamless signal expansion.^[18]

Practical Design Techniques

Practical FIR interpolation filters approximate the ideal low-pass response by truncating the infinite sinc impulse response to a finite length, introducing trade-offs between aliasing attenuation, computational cost, and group delay.^[19] Common design techniques include windowing, frequency sampling, and optimal approximation methods, each balancing performance imperfections inherent in finite implementations.^[20] Windowing methods apply a tapering function to the truncated sinc sequence to suppress sidelobes and reduce Gibbs phenomenon ringing in the frequency domain. For example, the Hamming window provides moderate sidelobe attenuation of about -43 dB, while the Kaiser window offers adjustable performance via its shape parameter β, which controls the trade-off between mainlobe width and sidelobe levels; β ≈ 0.1102(A - 8.7) for stopband attenuation A > 50 dB.^[19] With normalized cutoff frequency f_c = 1/L, the windowed impulse response is given by

h = L \frac{\sin(\pi f_c n)}{\pi n} \cdot w, \quad n = -M/2, \dots, M/2,

where w is the window function; this approach is straightforward but suboptimal in error minimization.^[21] The frequency sampling method designs the filter by specifying the desired frequency response at evenly spaced points and computing the inverse discrete Fourier transform (IDFT) to obtain the impulse response coefficients. This technique is particularly suitable for FFT-based implementations and arbitrary responses, as it interpolates the frequency samples linearly.^[20] However, it can introduce interpolation errors if the sampling is coarse, often requiring windowing post-IDFT to mitigate time-domain aliasing.^[19] Optimal filter designs, such as equiripple approximations via the Parks-McClellan algorithm (also known as the Remez exchange method), minimize the maximum weighted error across passband and stopband using an iterative Chebyshev approximation. This yields filters with equal ripple in both bands for a given order, achieving better attenuation (e.g., 60 dB stopband) with fewer taps than windowed designs.^[22] Least-squares methods, in contrast, minimize integrated squared error, prioritizing stopband energy reduction over peak ripple.^[19] In practice, filter lengths of 20 to 100 taps are common, providing aliasing attenuation around 60 dB while introducing linear phase delay of (N-1)/2 samples; longer filters improve sharpness but increase latency and computation. A rough estimate for the required FIR order is N \approx 4 / \Delta f, where \Delta f is the normalized transition bandwidth, though more precise formulas like Kaiser's N \approx (A - 8)/(2.285 \Delta \omega) (with \Delta \omega = 2\pi \Delta f) account for desired attenuation A in dB.^[19] These trade-offs ensure realizable filters for upsampling applications without excessive imaging artifacts.^[20]

Fractional Factor Upsampling

Rational Resampling Overview

Rational resampling extends integer-factor upsampling to achieve rate changes by non-integer rational factors L/M, where L and M are positive integers with L > M > 1, typically coprime to minimize computational redundancy.^[23] This method effectively combines upsampling by L and downsampling by M, enabling arbitrary increases in sampling rate while preserving signal integrity, such as converting audio from 44.1 kHz to 48 kHz using L = 160, M = 147.^[24] Unlike pure integer upsampling, which only expands the rate by an integer L, rational resampling incorporates decimation to fine-tune the output rate precisely.^[25] The process begins with upsampling the input signal x(n) by inserting L-1 zeros between each sample, expanding the rate to L times the original and introducing spectral images.^[23] A low-pass filter H(z) is then applied with a cutoff frequency of \pi / \max(L, M) to eliminate these images and prevent aliasing during the subsequent downsampling, ensuring the passband covers the original signal's spectrum up to \pi M / L in the new rate.^[23] Finally, decimation by M is performed by retaining every M-th sample from the filtered output, yielding the resampled signal at rate (L/M) times the original. The filter's passband gain is set to L to compensate for the amplitude dilution caused by zero insertion, preserving the signal's amplitude level.^[25] A practical example is 3/2 upsampling to convert a 48 kHz signal to 72 kHz: two zeros are inserted per sample (L=3), followed by low-pass filtering with cutoff \pi/3 to attenuate frequencies above approximately 24 kHz in the intermediate 144 kHz rate, and then every second sample is kept (M=2).^[23] This approach maintains the signal's bandwidth while achieving the desired rate increase without introducing significant distortion.^[25]

Polyphase Filter Implementation

The polyphase filter structure decomposes an interpolation filter h into L parallel subfilters h_k = h[mL + k] for k = 0, 1, \dots, L-1, where each subfilter operates at the original low sampling rate, enabling efficient processing by skipping the inserted zeros from the upsampling expansion.^[26] This decomposition allows the system to avoid unnecessary computations on zero-valued samples, transforming the serial high-rate filtering into parallel low-rate branches that collectively generate the upsampled output.^[26] In upsampling by an integer factor L, the Noble identity permits commuting the low-pass filter with the expander (upsampler), shifting the filtering operation to before the rate expansion and thereby reducing the computational load by a factor of L compared to direct convolution at the high rate.^[26] The commutator model illustrates this efficiency, where a rotating commutator routes each input sample sequentially to one of the L subfilters, with each subfilter contributing to the output at the elevated rate, effectively implementing the interpolation without processing zeros.^[26] For rational upsampling by L/M, the polyphase implementation combines an upsampling by L with a subsequent downsampling by M, using a bank of M polyphase subfilters derived from the overall interpolation filter, paired with M upsamplers and L downsamplers in a commutated structure that operates primarily at the output rate.^[26] This results in a computational efficiency comparable to filtering at the input sampling rate, with approximately the filter length in multiplications per input sample, as the time-varying filter coefficients g_m(n) = h(nL - m) (modulo L) are selected cyclically to match the rational ratio.^[26] The polyphase approach offers significant advantages in hardware, particularly in digital signal processors (DSPs), by enabling parallel subfilter execution that minimizes memory access and power consumption while maximizing throughput.^[27] For example, with L=4, the decomposition into four subfilters reduces computations to 25% of the direct method, yielding 75% savings in multiplications relative to full-rate convolution.^[26]

Advanced Topics

Multirate Signal Processing Context

Multirate signal processing encompasses techniques for altering the sampling rate of digital signals within a system, where upsampling serves as a fundamental operation alongside decimation to enable efficient handling of varying data rates. In such systems, upsampling expands the sampling rate by an integer factor L through zero insertion followed by low-pass filtering, while decimation reduces it by an integer factor M via anti-aliasing low-pass filtering prior to downsampling; this duality allows for rational rate conversions

[L/M](/page/L&M)

by cascading the operations, thereby avoiding the computational burden of processing at the highest uniform rate throughout the entire system.^[26] The primary efficiency gain arises from performing signal manipulations at intermediate lower rates, which minimizes the number of arithmetic operations compared to single-rate processing, as quantified in early analyses showing reductions in complexity proportional to the rate change factors.^[26] The interplay between upsampling and downsampling is central to achieving arbitrary rational resampling ratios, where a cascade of upsampling by L and subsequent downsampling by M (or vice versa) facilitates conversion from an input rate f_s to (L/M) f_s, provided the filters are designed to suppress imaging spectra from upsampling and aliasing artifacts from downsampling. Anti-imaging filters, typically low-pass with cutoff at \pi/L, follow upsampling to eliminate replicas introduced by zero insertion, while anti-aliasing filters with cutoff \pi/M precede decimation to prevent spectral folding; this structured pairing ensures minimal distortion in the resampled signal, with noble identities in multirate theory guaranteeing perfect reconstruction under ideal conditions.^[26] Such cascades are foundational for applications requiring precise rate matching, like audio or communications, where improper filtering can introduce quantifiable errors exceeding 40 dB in signal-to-noise ratio without correction.^[26] In filter banks, upsampling plays a key role in the synthesis stage of subband coding schemes, where critically sampled analysis banks downsample signals into frequency subbands for efficient compression, and upsampling in the synthesis bank reconstructs the original by interleaving expanded subband signals before summation filtering to mitigate aliasing and imaging. This multirate architecture underpins wavelet transforms, as in the discrete wavelet transform (DWT), where two-channel filter banks with upsampling by 2 in synthesis enable multiresolution decomposition and perfect reconstruction using quadrature mirror filters (QMFs) or biorthogonal designs, achieving critically sampled multiresolution representations.^[28] Seminal developments in maximally decimated filter banks demonstrate that upsampling ensures aliasing cancellation when analysis and synthesis filters satisfy power complementary conditions, preserving the signal's energy across scales. Modern extensions of multirate processing integrate upsampling with software-defined radio (SDR) architectures, where it interpolates low-rate baseband signals to higher intermediate frequencies for digital upconversion to carrier bands, enabling agile modulation schemes like quadrature amplitude modulation (QAM) without analog hardware reconfiguration. In SDR, polyphase implementations of upsampling reduce latency and power by factors up to 10 relative to single-rate alternatives, supporting real-time processing in resource-constrained platforms.^[29] Additionally, multirate techniques enhance DFT/FFT-based spectrum analysis by allowing resampled signals to align with FFT bin resolutions, minimizing spectral leakage in non-uniform rate scenarios, as evidenced in fault detection applications where fractional decimation prior to FFT improves frequency resolution over fixed-rate methods.^[30]

Computational Efficiency Considerations

Direct upsampling by an integer factor L involves inserting L-1 zeros between each input sample, followed by low-pass filtering, resulting in a computational complexity of O(N L) operations per block of N input samples, as the filter must process the expanded sequence including the inserted zeros.^[31] In contrast, polyphase implementations decompose the filter into L subfilters, avoiding explicit zero insertion and reducing the complexity to O(N) operations per block by processing only the non-zero input samples through the relevant subfilter branches.^[32] For an upsampling factor of L=4, this yields a floating-point operations (FLOPs) comparison of approximately $4N for direct methods versus N for polyphase, assuming a fixed filter order and focusing on the rate conversion overhead. Optimizations in hardware and software further enhance efficiency. Single instruction, multiple data (SIMD) instructions, such as those in modern CPUs, enable parallel computation across polyphase subfilters, accelerating the convolution process for real-time applications.^[11] Field-programmable gate arrays (FPGAs) are particularly suited for polyphase upsampling due to their reconfigurability, allowing pipelined implementations that achieve high throughput with reduced resource utilization compared to software equivalents.^[33] In mobile digital signal processors (DSPs), polyphase structures minimize power consumption by limiting multiplications to essential computations, often reducing energy use by up to 30% relative to direct filtering in battery-constrained environments.^[34] Key trade-offs arise in filter design and operation. Higher-order filters improve interpolation quality by providing sharper frequency responses but introduce greater latency, with group delay approximately N/2 samples for linear-phase FIR filters of order N, impacting real-time performance in applications like audio processing.^[35] Adaptive upsampling techniques, which dynamically adjust the rate based on variable input characteristics, enhance efficiency by avoiding fixed oversampling, though they require additional overhead for rate estimation.^[36] Recent advancements address these challenges through specialized acceleration. Graphics processing units (GPUs) enable parallel execution of polyphase filters, achieving up to 10x speedup over CPU implementations for high-throughput upsampling in signal processing pipelines since the early 2010s.^[37] Machine learning-based interpolation, such as neural upsampling networks introduced since 2018, offers adaptive, data-driven alternatives to traditional methods, reducing artifacts in non-stationary signals while maintaining computational feasibility on modern hardware.^[38]

References

[1]
Upsampling and Downsampling | Spectral Audio Signal Processing
To upsample by the integer factor , we simply insert zeros between and for all . In other words, the upsampler implements the stretch operator defined in §2.3. ...
[2]
Upsampling - EEE 5502: Foundations of Digital Signal Processing
The basic upsampling operation is upsampling by 2. Upsampling may refer to two similar operations: upsampling without interpolation or upsampling with ...
[3]
[PDF] ECE 468: Digital Image Processing Lecture 18
Nov 4, 2011 · Upsampling/Downsampling. • Upsampling = Inserting zeros. • Downsampling = Discarding pixels f2↑(x, y) = f(x/2,y/2) , x, y are even. 0. ,. o.w..Missing: overview | Show results with:overview
[4]
What is upsampling? - IBM
Upsampling increases the number of data samples in a dataset. In doing so, it aims to correct imbalanced data and thereby improve model performance.Overview · Why use upsampling?
[5]
None
Summary of each segment:
[6]
An efficient algorithm for sample rate conversion from CD to DAT
... digital audio tape (DAT). This method involves upsampling the input signal by two, and then passing the interpolated signal through a fractional delay ...
[7]
New concepts in multirate digital processing of signals - ADS
New concepts in multirate digital processing of signals. Crochiere, R. E.; ;; Rabiner, L. R.. Abstract. Some new results on techniques for implementing digital ...
[8]
[PDF] Upsampling and Downsampling - Stanford CCRMA
Jun 2, 2020 · For the DFT, we have the Stretch Theorem (Repeat. Theorem) which relates upsampling (“stretch”) to spectral copies (“images”) in the DFT ...
[9]
[PDF] Selected Advanced Topics in Digital Signal Processing
Aug 27, 2020 · by Oppenheim and Schafer or Proakis and Manolakis. Nevertheless, I ... upsampling a signal by a factor of 4, delaying by a single ...<|control11|><|separator|>
[10]
[PDF] Digital Signal Processing: Principles, Algorithms & Applications (3rd ...
Page 1. 77un/ Edition. DIGITAL. SIGNAL. PROCESSING. Principles, Algorithms, m l Applications. John G. Proakis. Dimitris G. Manolakis. Page 2. Digital Signal.
[11]
dsp.FIRInterpolator - Perform polyphase FIR interpolation - MATLAB
The designed filter corresponds to a lowpass with a cutoff at π/ L in radial frequency units. firinterp = dsp.FIRInterpolator( L , "Auto" ) returns an FIR ...
[12]
Interpolator Design: Get the Stopbands Right - Neil Robertson
Jul 6, 2023 · Normally, the up-sampler has a gain of L to achieve interpolator output level equal to input level. ... All you have to do is a low pass filter ...
[13]
Interpolation filter gain - Signal Processing Stack Exchange
Aug 24, 2016 · If somebody could clarify why everywhere is written that the interpolation filter should have gain L where L is upsampling factor. What this ...Output gain when upsampling and downsamplingUpsampling - What purpose does the interpolation filter have?More results from dsp.stackexchange.com
[14]
The Ideal Lowpass Filter
An ideal lowpass may be characterized by a gain of 1 for all frequencies below some cut-off frequency $ f_c$ in Hz, and a gain of 0 for all higher frequencies.
[15]
comp.dsp | Stability of IIR Filters| page 3 - DSPRelated.com
Dec 5, 2015 · Here's a proof: The ideal interpolation filter for upsampling at an integer ratio L is given by h[n] = sin(pi*n/L) / (pi*n/L) = sinc(n/L). ... h ...
[16]
https://people.math.harvard.edu/~ctm/home/text/others/shannon/entropy/entropy.pdf
[17]
https://eng.libretexts.org/Bookshelves/Electrical_Engineering/Signal_Processing_and_Modeling/Signals_and_Systems_(Baraniuk_et_al.)/10%3A_Sampling_and_Reconstruction/10.04%3A_Perfect_Reconstruction
[18]
[PDF] 17 Interpolation - MIT OpenCourseWare
When the reconstruction filter is an ideal low- pass filter, the interpolating function is a sinc function. This is often referred to as bandlimited ...
[19]
FIR Digital Filter Design | Spectral Audio Signal Processing
In addition, there is a function kaiserord for estimating the parameters of a Kaiser window which will achieve the desired filter specifications. Bandpass ...
[20]
[PDF] Mixed-Signal and DSP Design Techniques, Digital Filters
FIR Filter Design Using the Frequency Sampling Method. This method is ... DIGITAL FILTERS. 6.18. Figure 6.21. FIR Filter Design Using the Parks-McClellan Program.<|control11|><|separator|>
[21]
https://www.ni.com/docs/en-US/bundle/labwindows-cvi/page/advancedanalysisconcepts/lvac_design_fir_windowing.html
[22]
firpm - Parks-McClellan optimal FIR filter design - MATLAB
Design a lowpass filter with a 1500 Hz passband cutoff frequency and 2000 Hz stopband cutoff frequency. Specify a sampling frequency of 8000 Hz. Require a ...Missing: practical windowing
[23]
[PDF] Digital Sampling Rate Conversion: Principles and Implementation
The resampling of a signal involves the conversion from the initial sampling rate to a new and different one. This is often necessary in practical ...
[24]
[PDF] Digital Audio Resampling Home Page - Stanford CCRMA
This provides sampling-rate conversion by any rational factor L/M. The conversion requires a digital lowpass filter whose cutoff frequency depends on max{L, M}.
[25]
Design of Decimators and Interpolators - MATLAB & Simulink
When upsampling by a rate of N , apply a lowpass filter after upsampling, this filter is known as an anti-imaging filter. The filter removes the spectral images ...
[26]
https://ieeexplore.ieee.org/document/1456237
[27]
https://ieeexplore.ieee.org/document/7566718
[28]
https://waveletsandsubbandcoding.org/Repository/VetterliKovacevic95_Manuscript.pdf
[29]
Multirate Signal Processing for Software Radio Architectures
Optimizing the sample rate while processing signals provides many advantages to digital systems.
[30]
Multirate Signal Processing to Improve FFT-Based Analysis for ...
Aug 6, 2025 · This paper introduces multirate signal processing techniques that improve the FFT-based methods by reducing spectral leakage with fractional ...
[31]
[PDF] Upsampling and Downsampling • Polyphase Fil - Stanford CCRMA
Jun 2, 2020 · For the DFT, we have the Stretch Theorem (Repeat. Theorem) which relates upsampling (“stretch”) to spectral copies (“images”) in the DFT ...
[32]
The Polyphase Implementation of Interpolation Filters in Digital ...
Dec 6, 2017 · This article discusses an efficient implementation of the interpolation filters called the polyphase implementation.
[33]
https://ieeexplore.ieee.org/document/10039914
[34]
[PDF] Review of Polyphase Filtering Technique in Signal Processing
The result can be computationally efficient fully on-board algorithm enabling in-place and real time processing which avoids up/down-link data transfers and ...
[35]
Computing the Group Delay of a Filter - Rick Lyons - DSPRelated.com
Nov 19, 2008 · Starting with a filter's N-sample h(n) impulse response, performing two N-point DFT's and an N-sample complex division, we can compute the filter's group delay ...Missing: latency upsampling
[36]
Adaptive rate filtering a computationally efficient signal processing ...
The idea is to adapt the sampling frequency and the filter order by following the input signal local characteristics. In this context, efficient solutions are ...Missing: upsampling variable
[37]
Implementation of a high-throughput low-latency polyphase ...
Sep 15, 2014 · We propose a novel GPU-based polyphase channelizer architecture that delivers high throughput. This architecture has advantages of providing reduced complexity.Missing: upsampling | Show results with:upsampling
[38]
[PDF] arXiv:2010.14356v2 [cs.SD] 9 Feb 2021
Feb 9, 2021 · ABSTRACT. A number of recent advances in neural audio synthesis rely on up- sampling layers, which can introduce undesired artifacts.