A Low-Complexity Linear-Phase Graphic Audio Equalizer Based on IFIR Filters

Bruschi, Valeria; Nobili, Stefano; Terenzi, Alessandro; Cecchi, Stefania

doi:10.1109/lsp.2021.3057228

Cited by 7 publications

(5 citation statements)

References 12 publications

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“…In addition, the audio quality can be improved by denoising (e.g., Wiener filter [18]), echo cancellation (i.e., inverse of our echo module), or equalizer (i.e., volume adjustment for each frequency [14]); however, since these methods cannot introduce new frequency components into the compromised audio, they are limited on mitigating spectrum reduction attacks.…”

Section: A Multipath Effect and Audio Quality Improvementmentioning

confidence: 99%

Compensating Removed Frequency Components: Thwarting Voice Spectrum Reduction Attacks

Wang,

Sun,

2024

Proceedings 2024 Network and Distributed System Security Symposium

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Automatic speech recognition (ASR) provides diverse audio-to-text services for humans to communicate with machines. However, recent research reveals ASR systems are vulnerable to various malicious audio attacks. In particular, by removing the non-essential frequency components, a new spectrum reduction attack can generate adversarial audios that can be perceived by humans but cannot be correctly interpreted by ASR systems. It raises a new challenge for content moderation solutions to detect harmful content in audio and video available on social media platforms. In this paper, we propose an acoustic compensation system named ACE to counter the spectrum reduction attacks over ASR systems. Our system design is based on two observations, namely, frequency component dependencies and perturbation sensitivity. First, since the Discrete Fourier Transform computation inevitably introduces spectral leakage and aliasing effects to the audio frequency spectrum, the frequency components with similar frequencies will have a high correlation. Thus, considering the intrinsic dependencies between neighboring frequency components, it is possible to recover more of the original audio by compensating for the removed components based on the remaining ones. Second, since the removed components in the spectrum reduction attacks can be regarded as an inverse of adversarial noise, the attack success rate will decrease when the adversarial audio is replayed in an overthe-air scenario. Hence, we can model the acoustic propagation process to add over-the-air perturbations into the attacked audio. We implement a prototype of ACE and the experiments show that ACE can effectively reduce up to 87.9% of ASR inference errors caused by spectrum reduction attacks. Furthermore, by analyzing the residual errors on real audio samples, we summarize six general types of ASR inference errors and investigate the error causes and potential mitigation solutions.

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Section: A Multipath Effect and Audio Quality Improvementmentioning

confidence: 99%

Compensating Removed Frequency Components: Thwarting Voice Spectrum Reduction Attacks

Wang,

Sun,

2024

Proceedings 2024 Network and Distributed System Security Symposium

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“…where f (A) is the pdf of A (see (12)), and erf (•) is the error function. Therefore, observing that P(A M ) = P(0 ≤ A ≤ 2 m+q ) − P(0 ≤ A ≤ 2 m ) and P(A H ) = P(0 ≤ A ≤ 2 n −1)−P(0 ≤ A ≤ 2 m+q ), we obtain the results shown in Table 3.…”

Section: Data Availability Statementmentioning

confidence: 99%

“…The huge number of multipliers, used for adaptation, increases the power consumption and demands specific techniques aimed to reduce area and power while preserving the quality of results [7][8][9][10][11]. Low-power designs are also required for audio applications [12] in which banks of filters realize operations as equalization and denoising.…”

Section: Introductionmentioning

confidence: 99%

Design of Generalized Enhanced Static Segment Multiplier with Minimum Mean Square Error for Uniform and Nonuniform Input Distributions

Meo¹,

Saggese²,

Strollo³

et al. 2023

Electronics

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In this paper, we analyze the performances of an Enhanced Static Segment Multiplier (ESSM) when the inputs have both uniform and non-uniform distribution. The enhanced segmentation divides the multiplicands into a lower, a middle, and an upper segment. While the middle segment is placed at the center of the inputs in other implementations, we seek the optimal position able to minimize the approximation error. To this aim, two design parameters are exploited: m, defining the size and the accuracy of the multiplier, and q, defining the position of the middle segment for further accuracy tuning. A hardware implementation is proposed for our generalized ESSM (gESSM), and an analytical model is described, able to find m and q which minimize the mean square approximation error. With uniform inputs, the error slightly improves by increasing q, whereas a large error decrease is observed by properly choosing q when the inputs are half-normal (with a NoEB up to 18.5 bits for a 16-bit multiplier). Implementation results in 28 nm CMOS technology are also satisfactory, with area and power reductions up to 71% and 83%. We report image and audio processing applications, showing that gESSM is a suitable candidate in applications with non-uniform inputs.

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“…Hergum [33] proposed an interesting FIR GEQ design, which is based on interpolated FIR (IFIR) filters [34]. Recently, a similar design was proposed for another band division [35]. An IFIR filter contains a cascade of two filters with the first producing a periodic frequency response and the second attenuating the unwanted repetitions [34].…”

Section: Introductionmentioning

confidence: 99%

“…IFIR filters achieve linear-phase and small ripple at a low computational cost. Although Hergum's design is clever and efficient, the audio frequencies are divided in equal bands, as is the case with the equalizer proposed in [35]. This is incompatible with standard graphic equalizers, which use a logarithmic band division [3].…”

Section: Introductionmentioning

confidence: 99%

Linear-Phase Octave Graphic Equalizer

Bruschi¹,

Välimäki²,

Liski³

et al. 2022

J. Audio Eng. Soc.

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A computationally efficient octave-band graphic equalizer having a linear-phase response is introduced. The linear-phase graphic equalizer is useful in audio applications in which phase distortion is not tolerated, such as in multichannel equalization, parallel processing, phase compatibility of audio equipment, and crossover network design. The structure is based on the interpolated finite impulse response (IFIR) philosophy. The proposed octave-band graphic equalizer uses one prototype low-pass filter, which is a half-band FIR filter designed using the window method. Stretched versions of the prototype filter and its complementary high-pass filter implement all ten band filters needed. The graphic equalizer is realized in the parallel form, in which the outputs of all band filters, scaled with their individual command gain, are added to compute the equalized output signal. The command gains can be used directly as filter band gains. The number of operations needed per sample is only slightly more than that needed for the graphic equalizer based on minimum-phase recursive filters. A comparison with other implementation approaches demonstrates that the proposed structure requires 99% fewer operations than a high-order FIR filter. The proposed filter uses 39% fewer operations per sample than the fast Fourier transform-based filtering method and causes over 78% less latency.

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A Low-Complexity Linear-Phase Graphic Audio Equalizer Based on IFIR Filters

Cited by 7 publications

References 12 publications

Compensating Removed Frequency Components: Thwarting Voice Spectrum Reduction Attacks

Compensating Removed Frequency Components: Thwarting Voice Spectrum Reduction Attacks

Design of Generalized Enhanced Static Segment Multiplier with Minimum Mean Square Error for Uniform and Nonuniform Input Distributions

Linear-Phase Octave Graphic Equalizer

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