Download A Low-Latency Quasi-Linear-Phase Octave Graphic Equalizer This paper proposes a low-latency quasi-linear-phase octave graphic equalizer. The structure is derived from a recent linearphase graphic equalizer based on interpolated finite impulse response (IFIR) filters. The proposed system reduces the total latency of the previous equalizer by implementing a hybrid structure. An infinite impulse response (IIR) shelving filter is used in the structure to implement the first band of the equalizer, whereas the rest of the band filters are realized with the linear-phase FIR structure. The introduction of the IIR filter causes a nonlinear phase response in the low frequencies, but the total latency is reduced by 50% in comparison to the linear-phase equalizer. The proposed graphic equalizer is useful in real-time audio processing, where only little latency is tolerated.
Download Real-Time System for Sound Enhancement in Noisy Environment The noise can affect the listening experience in many real-life situations involving loudspeakers as a playback device. A solution to reduce the effect of the noise is to employ headphones, but they can be annoying and not allowed on some occasions. In this context, a system for improving the audio perception and the intelligibility of sounds in a domestic noisy environment is introduced and a real-time implementation is proposed. The system comprises three main blocks: a noise estimation procedure based on an adaptive algorithm, an auditory spectral masking algorithm that estimates the music threshold capable of masking the noise source, and an FFT equalizer that is used to apply the estimated level. It has been developed on an embedded DSP board considering one microphone for the ambient noise analysis and two vibrating sound transducers for sound reproduction. Several experiments on simulated and real-world scenarios have been realized to prove the effectiveness of the proposed approach.
Download Real-Time Implementation of a Linear-Phase Octave Graphic Equalizer This paper proposes a real-time implementation of a linear-phase octave graphic equalizer (GEQ), previously introduced by the same authors. The structure of the GEQ is based on interpolated finite impulse response (IFIR) filters and is derived from a single prototype FIR filter. The low computational cost and small latency make the presented GEQ suitable for real-time applications. In this work, the GEQ has been implemented as a plugin of a specific software, used for real-time tests. The performance of the equalizer has been evaluated through subjective tests, comparing it with a filterbank equalizer. For the tests, four standard equalization curves have been chosen. The experimental results show promising outcomes. The result is an accurate real-time-capable linear-phase GEQ with a reasonable latency.
Download A Modified Algorithm for a Loudspeaker Line Array Multi-Lobe Control The creation of personal sound zones is an effective solution
for delivering personalized auditory experiences in shared spaces.
Their applications span various domains, including in-car entertainment, home and office environments, and healthcare functions.
This paper presents a novel approach for the creation of personal
sound zones using a modified algorithm for multi-lobe control in
loudspeaker line array. The proposed method integrates a pressurematching beamforming algorithm with an innovative technique for
reducing side lobes, enhancing the precision and isolation of sound
zones.
The system was evaluated through simulations and experimental tests conducted in a semi-anechoic environment and a
large listening room. Results demonstrate the effectiveness of the
method in creating two separate sound zones.
Download A Non-Uniform Subband Implementation of an Active Noise Control System for Snoring Reduction The snoring noise can be extremely annoying and can negatively
affect people’s social lives. To reduce this problem, active noise
control (ANC) systems can be adopted for snoring cancellation.
Recently, adaptive subband systems have been developed to improve the convergence rate and reduce the computational complexity of the ANC algorithm. Several structures have been proposed
with different approaches. This paper proposes a non-uniform subband adaptive filtering (SAF) structure to improve a feedforward
active noise control algorithm. The non-uniform band distribution
allows for a higher frequency resolution of the lower frequencies,
where the snoring noise is most concentrated. Several experiments
have been carried out to evaluate the proposed system in comparison with a reference ANC system which uses a uniform approach.