Sound Quality Enhancement Research Articles

Machine learning accelerated MMC-based topology optimization for sound quality enhancement of serialized acoustic structures

Machine learning accelerated MMC-based topology optimization for sound quality enhancement of serialized acoustic structures

Exploring the impact of noise and degradations on heart sound classification models

The development of data-driven heart sound classification models has been an active area of research in recent years. To develop such data-driven models in the first place, heart sound signals need to be captured using a signal acquisition device. However, it is almost impossible to capture noise-free heart sound signals due to the presence of internal and external noises in most situations. Such noises and degradations in heart sound signals can potentially reduce the accuracy of data-driven classification models. Although different techniques have been proposed in the literature to address the noise issue, how and to what extent different noise and degradations in heart sound signals impact the accuracy of data-driven classification models remains unexplored. To answer this question, we produced a synthetic heart sound dataset including normal and abnormal heart sounds contaminated with a variety of noise and degradations. We used this dataset to investigate the impact of noise and degradation in heart sound recordings on the performance of different classification models. The results show different noises and degradations affect the performance of heart sound classification models to a different extent; some are more problematic for classification models, and others are less destructive. Comparing the findings of this study with the results of a survey we previously carried out with a group of clinicians shows noise and degradations that are more detrimental to classification models are also more disruptive to accurate auscultation. The findings of this study can be leveraged to develop targeted heart sound quality enhancement approaches — which adapt the type and aggressiveness of quality enhancement based on the characteristics of noise and degradation in heart sound signals.

A computationally efficient feedforward time–frequency-domain hybrid active sound profiling algorithm for vehicle interior noise

Active sound profiling (ASP) is gaining ever-increasing attention owing to its powerful ability to balance the sound pressure levels of certain frequency points or frequency bands of target noise for sound quality enhancement. Most of the current ASP algorithms, however, only focus on narrowband or broadband noise alone and the computational efficiency is very low. To alleviate this problem, a feedforward time–frequency-domain hybrid active sound profiling (TFD-HASP) algorithm is proposed in this paper. The algorithm consists of three subsystems, in which the narrowband ASP (NASP) subsystem utilizes the time-domain internal model FXLMS algorithm along with the local secondary path estimation approach, the broadband ASP (BASP) subsystem is based on the delayless frequency-domain internal model FXLMS algorithm, and the sinusoidal noise canceller (SNC) subsystem is designed to eliminate narrowband interference to the BASP subsystem. In this way, the presented TFD-HASP algorithm not only has good spectral reshaping capability for both narrowband and broadband noise components, but also requires notably low computational effort. A series of simulations are performed to evaluate the performance of the TFD-HASP algorithm. In addition, the comfortable sound quality of a real vehicle interior noise at high speed is designed and realized. Experimental results demonstrate that the proposed algorithm achieves satisfactory active sound profiling for noise with both narrowband and broadband characteristics.

The Impact of Vocal Boost Manipulations on Musical Sound Quality for Cochlear Implant Users.

To evaluate the impact of vocal boost manipulations on cochlear implant (CI) musical sound quality appraisals. An anonymous, online study was distributed to 33 CI users. Participants listened to auditory tokens and assessed the musical quality of acoustic stimuli with vocal boosting and attenuation using a validated sound quality rating scale. Four versions of real-world musical stimuli were created: a version with +9 dB vocal boost, a version with -9 dB vocal attenuation, a composite stimulus containing a 1,000 Hz low-pass filter and white noise ("anchor"), and an unaltered version ("hidden reference"). Subjects listened to all four versions and provided ratings based on a 100-point scale that reflected the perceived sound quality difference of the music clip relative to the reference excerpt. Vocal boost increased musical sound quality ratings relative to the reference clip (11.7; 95% CI, 1.62-21.8, p=0.016) and vocal attenuation decreased musical sound quality ratings relative to the reference clip (28.5; 95% CI, 18.64-38.44, p < 0.001). When comparing the non-musical training group and musical training group, there was a significant difference in musical sound quality rating scores for the vocal boost condition (21.2; 95% CI: 1.76-40.7, p=0.028). CI-mediated musical sound quality appraisals are impacted by vocal boost and attenuation. Musically trained CI users to report greater musical sound quality enhancement with a vocal boost with respect to CI users with no musical training background. Implementation of front-end vocal boost manipulations in music may improve sound quality and music appreciation among CI users. 2 (Individual cohort study) Laryngoscope, 133:938-947, 2023.

An efficient narrowband active sound profiling system with online secondary path modeling

Active noise equalization or active sound profiling techniques for sound quality enhancement tend to balance the sound pressure levels (SPLs) of certain frequency points or frequency bands of target noise to meet human perception requirements rather than simply minimizing the SPLs like active noise control (ANC) technique. Therefore, when the secondary path is time-varying, online secondary path modeling (OSPM) appears particularly important in the active sound quality enhancement systems. This paper presents a novel parallel-structure narrowband active sound profiling (NASP) system with OSPM. In the system, the OSPM subsystem uses the control signals to estimate low-order local secondary path models at the controlled frequencies instead of the usual high-order global model. This way may ensure the convergence speed and accuracy of modeling. Meanwhile, the main sound profiling subsystem employs the estimated local secondary path models to profile the targeted multi-tonal narrowband noise based on the internal model FXLMS algorithm. By using bandpass filter banks, each tonal component may be tuned relatively independently. In addition, the proposed system enjoys good profiling accuracy and stability since no additional random noise is introduced. Several simulations are provided to demonstrate the superiority of the proposed NASP system with OSPM in cancellation mode over the conventional NANC system with OSPM and evaluate the behavior of the proposed system in different time-varying scenarios. Finally, real-time experiments are also conducted to confirm the system’s ability to deal with sudden changes in the secondary path and its sound profiling capability.

Sound Quality Enhancement by using the Single Core Exciter in OLED Panel

Sound Quality Enhancement by using the Single Core Exciter in OLED Panel

57‐5: An Enhancement of Sound Quality by Using the Single Exciter in OLED Panel

OLED panel speaker technology was first developed in 2017 by LG Display. This technology is based on the principle that an OLED panel acts as a diaphragm of a speaker by attaching an oscillator called an exciter to the back of the OLED panel. The technology delivers direct, clear sound from the panel to viewers, maximizing the immersion experience by allowing the sound and video to be played back at the same location. However, due to the strong directivity, the tone of the panel tends to be somewhat diminished at the center of the panel, which is the main TV viewing angle. In this paper, we improved the sound quality by improving the angle of directivity by making 2in1 exciter 1core model.

Multiple target sound quality balance for hybrid electric powertrain noise

Abstract The integration of the electric motor to the powertrain in hybrid electric vehicles (HEVs) presents acoustic stimuli that elicit new perceptions. The large number of spectral components, as well as the wider bandwidth of this sort of noises, pose new challenges to current noise, vibration and harshness (NVH) approaches. This paper presents a framework for enhancing the sound quality (SQ) of the hybrid electric powertrain noise perceived inside the passenger compartment. Compared with current active sound quality control (ASQC) schemes, where the SQ improvement is just an effect of the control actions, the proposed technique features an optimization stage, which enables the NVH specialist to actively implement the amplitude balance of the tones that better fits into the auditory expectations. Since Loudness, Roughness, Sharpness and Tonality are the most relevant SQ metrics for interior HEV noise, they are used as performance metrics in the concurrent optimization analysis, which, eventually, drives the control design method. Thus, multichannel active sound profiling systems that feature cross-channel compensation schemes are guided by the multi-objective optimization stage, by means of optimal sets of amplitude gain factors that can be implemented at each single sensor location, while minimizing cross-channel effects that can either degrade the original SQ condition, or even hinder the implementation of independent SQ targets. The proposed framework is verified experimentally, with realistic stationary hybrid electric powertrain noise, showing SQ enhancement for multiple locations within a scaled vehicle mock-up. The results show total success rates in excess of 90%, which indicate that the proposed method is promising, not only for the improvement of the SQ of HEV noise, but also for a variety of periodic disturbances with similar features.

Application of array microphone measurement for the enhancement of sound quality by use of adaptive time reversal method

The purpose of this research is to develop an algorithm which can be used in the system of speech recognition. In the previous research, a target signal can be reconstructed by a signal separation technique which is based on time-reversal method (TRM) via self-focusing property. In this research, a multiple signal separation technique which is based on adaptive time-reversal method (ATRM) is developed. The advantage of ATRM is to reconstruct target signal and to cancel noise signals simultaneously if impulse response functions of environment between target source, noise sources, and array microphone are known. In the simulation, correlation coefficient and signal to noise ratio (SNR) are implemented as indicators to evaluate the performance of signal separation for TRM and ATRM in a reverberation room, and to investigate the differences between two methods. The comparison indicates that number of array microphone is significantly reduced from 14-microphones via TRM to 6-microphones via ATRM. Furthermore, the experiments of signal separation are conducted to a model of smart TV with multiple noise sources in the environment. Finally, a real time system of signal separation was developed. All the results indicate that the target of source was clearly reconstructed with enhancing of correlation coefficient and SNR by using ATRM.

DOCKING STATION HAVING STRUCTURE FOR SOUND AMPLIFICATION AND SOUND QUALITY ENHANCEMENT

A docking station for sound amplification and sound quality enhancement is provided. The docking station includes a support structure for holding a mobile terminal having an internal speaker to sustain the posture of the mobile terminal, and a body for supporting the support structure, and for physically contacting the speaker to increase the volume of sound output from the speaker. The body includes a collecting hole for contacting the speaker to collect sound waves, and a guide hole that extends from the collecting hole through the body to the outside along an extension direction, is divided into two branches within the body to guide the collected sound waves along different paths, and has a horn shape whose cross section increases along the extension direction. Hence, the docking station can increase the volume of audible sound and sound quality without separate supply of power.

Implementation of Music Equalization Simulink Model on DSK6713

Audio equalization is a technique which consists of boosting or cutting certain frequency components of a given signal for sound quality enhancement. Equalizer is an electronic device or type of software that increases and decreases the power of sound waves. The paper deals with the analysis of music signals to develop insight on its frequency bands and auditory perception, through a case study of audio equalization. The graphic equalizer is designed through a Simulink model and is implemented in TMS320C6713 DSP board. This project is developed in order to deliver an insight on how audio equalization can be implemented with Matlab / Simulink for educational purposes.

Spatial Audio Object Coding With Two-Step Coding Structure for Interactive Audio Service

An interactive audio service is a new conceptual audio service that provides the users with opportunities for a variety of experiences on the alternative and advanced audio services. In the interactive audio service, users can freely control various audio objects to make their own audio sounds. A spatial audio object coding (SAOC) is a useful technology that can support most parts of the interactive audio service with a relatively low bit-rate, but is very poor to perfect gain control of a certain audio object, i.e., the target audio object. In this paper, the SAOC with a two-step coding structure is proposed to efficiently handle the target audio object as well as the normal audio objects. A transform coded excitation (TCX) based residual coding scheme is presented in the context of the sound quality enhancement. From experimental results, it can be noted that the various audio objects can be successfully handled with respect to the bit-rate and the sound quality by using the proposed two-step coding structure SAOC.

Sound quality enhancement circuit for audio signals and audio amplifier circuit using the same

The present invention comprises a circuit having at least not less than three cascade connected phase shift circuits and wherein the phase characteristic of the circuit is determined in such a manner that the phase of the audio input signals is gradually delayed depending on logarithmic increase of frequencies in a range of audible frequencies and a phase delay in a range of 450 degree˜600 degree in total in the range of audible frequencies is caused.

Enhancement of sound quality for computer telephony systems

Techniques for enhancing the sound quality of computer telephony systems are provided. In general, digital signals including telephony sounds are converted from the time domain to the frequency domain. If noise is detected in the frequency domain conversion of the digital signals, the noise is removed utilizing a filter. The noise detection and filtering can be performed in software so that enhanced audio quality can be achieved at a relatively low cost and with flexibility for very diverse environments

Audio sound quality enhancement apparatus and method

An audio enhancement apparatus includes a thermally insulating container and a substrate mounted at least partially within the thermally insulating container. A heat source is provided within the thermally insulating container and proximate to the substrate. At least one audio semiconductor is mounted to the substrate, where the audio semiconductor is located within the thermally insulating container. The thermally insulating container, in conjunction with the heat source, maintains an above-ambient temperature for the audio semiconductor while at least partially insulating components located outside of the thermally insulating container from the heat source. Another embodiment of the audio enhancement apparatus includes a thermally conductive substrate with at least one heat source located proximate to the substrate and at least one audio semiconductor mounted to the substrate. A heat dissipation device is also mounted proximate to the substrate.

Speech Enhancement by Overweighting Gain with Nonlinear Structure in Wavelet Packet Transform

A sound quality enhancement method by overweighting gain of a nonlinear structure in a wavelet packet area is provided to restrain the generation of musical noise efficiently and ensure reliable intelligibility in an enhanced voice. A sound quality enhancement method by overweighting gain of a nonlinear structure in a wavelet packet area comprises the following steps of: generating a converting signal that a voice signal polluted by noise is converted by UWPT(Uniform Wavelet Packet Transform); calculating a relative size difference, which is an identifier for calculating a relative difference between the amount of noise existing in a sub band and the amount of a voice polluted by noise; calculating the overweighting gain of the nonlinear structure from the relative size difference; calculating a transformed time-varying gain function based on an LSL(Least-Squares Line) algorithm; and performing spectral subtraction using the transformed time-varying gain function.

Musical instrument sound quality enhancement device

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