Degradation Of Sound Quality Research Articles

Multichannel Dynamic Sound Rendering and Echo Suppression in a Room Using Wave Field Synthesis

With multichannel sound rendering systems, an immersive sound can be experienced by a broader audience. Rendering of a stationary source is widely studied in the literature. However, a dynamic sound, where the source’s position is continuously changing, provides improved source localization and a listener experience more immersed in the rendered sound. This paper investigates the multichannel sound rendering of a dynamic sound source using a wave field synthesis method in a reverberant shoe-box room model. Finite difference time domain is employed to visualize the reproduced sound field in the time domain. To address the challenge of room echo-induced degradation in sound quality, a method is presented in which the reflected sound is compensated by rendering the primary source’s images from the auxiliary actuators, enhancing SNR by approximately 2–3.5 dB.

Curvelet based robust improved sine adaptive filter for feedback cancellation in hearing aids

Acoustic feedback is a common and persistent issue in hearing aids, which can lead to limitations in the achievable amplification and significant degradation in sound quality, including howling artefacts. Recently, the maximum versoria criteria (MVC) algorithm has been developed to reduce the acoustic feedback in hearing aids in the presence of impulsive noise. However, the robust technique MVC has not addressed the feedback signal’s sparseness characteristic while developing an adaptive feedback canceller. This paper proposes a robust and sparsity-aware technique called curvelet-improved sine-based adaptive filtering (CISAF) algorithm for adaptive feedback cancellation (AFC) in hearing aids using the prediction error method (PEM) in the case of non-gaussian interference. The step factor is enhanced based on the logarithmic cosh function to speed up the algorithm’s convergence while guaranteeing a low steady-state error. This variable step size (VSS) is included with CISAF as CISAF-VSS technique. Computer simulations show that the suggested method achieves a good convergence rate, maintains high speech quality, keeps a high segmental SNR (segSNR) and short-time objective intelligibility (STOI) at steady-state.

Effects of Sound Quality on the Accuracy of Telephone Captions Produced by Automatic Speech Recognition: A Preliminary Investigation.

Automatic speech recognition (ASR) is commonly used to produce telephone captions to provide telecommunication access for individuals who are d/Deaf and hard of hearing (DHH). However, little is known about the effects of degraded telephone audio on the intelligibility of ASR captioning. This research note investigates the accuracy of telephone captions produced by ASR under degraded audio conditions. Packet loss, delay, and repetition are common sources of degradation in sound quality for telephone audio. Eleven sets of wideband filtered sentences were degraded by high and low levels of simulated packet loss, delay, and repetition. These sets, along with a clean set of sentences, were submitted to ASR, and the accuracy of the resulting output was evaluated using three metrics: a word recognition score, word error rate, and word information loss. The resulting pattern of data indicated the relative impact of each degraded condition on message intelligibility. The high and low packet loss conditions had the largest effect on message intelligibility. This finding was interpreted to indicate that packet loss can have a substantial impact on the accuracy of telephone captions produced with ASR. The results of this investigation point to a potential area of improvement in service quality that could have a substantial impact on telecommunication services for consumers who are DHH. Further research in this area is needed to provide additional information concerning the scope and impact of packet loss on the accuracy of telephone captioning produced by ASR. https://doi.org/10.23641/asha.21699557.

A mixed-order modeling approach for head-related transfer function in the spherical harmonic domain

Head-related transfer function (HRTF) contains all the relevant cues that humans rely on to perceive the direction of a sound source, and its modeling has been widely studied in the past years. Motivated by the different degree of fluctuations in HRTF magnitude spectrum across frequency, in this paper, a mixed-order modeling approach in the spherical harmonic domain is proposed in which HRTF magnitude spectrum with low fluctuation is modeled by the low-order spherical harmonics and that with high fluctuation is modeled by the high-order spherical harmonics. Furthermore, a data-driven optimization approach is presented to determine the transition frequency that separates HRTF magnitude spectrum into two ranges with different degrees of fluctuations. The effectiveness of the proposed mixed-order modeling approach is examined using FABIAN HRTF database through listening tests. Results show that this mixed-order modeling approach enables to reduce the order of spherical harmonics to 8 in the low frequencies and 18 in the high frequencies without perceptual degradation in sound quality and localization.

Sound Quality Effects of an Adaptive Nonlinear Frequency Compression Processor with Normal-Hearing and Hearing-Impaired Listeners.

Frequency lowering (FL) technology offers a means of improving audibility of high-frequency sounds. For some listeners, the benefit of such technology can be accompanied by a perceived degradation in sound quality, depending on the strength of the FL setting. The studies presented in this article investigate the effect of a new type of FL signal processing for hearing aids, adaptive nonlinear frequency compression (ANFC), on subjective speech quality. Listener ratings of sound quality were collected for speech stimuli processed with systematically varied fitting parameters. Study 1 included 40 normal-hearing (NH) adult and child listeners. Study 2 included 11 hearing-impaired (HI) adult and child listeners. HI listeners were fitted with laboratory-worn hearing aids for use during listening tasks. Speech quality ratings were assessed across test conditions consisting of various strengths of static nonlinear frequency compression (NFC) and ANFC speech. Test conditions included those that were fine-tuned on an individual basis per hearing aid fitting and conditions that were modified to intentionally alter the sound quality of the signal. Listeners rated speech quality using the MUlti Stimulus test with Hidden Reference and Anchor (MUSHRA) test paradigm. Ratings were analyzed for reliability and to compare results across conditions. Results show that interrater reliability is high for both studies, indicating that NH and HI listeners from both adult and child age groups can reliably complete the MUSHRA task. Results comparing sound quality ratings across experimental conditions suggest that both the NH and HI listener groups rate the stimuli intended to have poor sound quality (e.g., anchors and the strongest available parameter settings) as having below-average sound quality ratings. A different trend in the results is reported when considering the other experimental conditions across the listener groups in the studies. Speech quality ratings measured with NH listeners improve as the strength of ANFC decreases, with a range of bad to good ratings reported, on average. Speech quality ratings measured with HI listeners are similar and above-average for many of the experimental stimuli, including those with fine-tuned NFC and ANFC parameters. Overall, HI listeners provide similar sound quality ratings when comparing static and adaptive forms of frequency compression, especially when considering the individualized parameter settings. These findings suggest that a range in settings may result in above-average sound quality for adults and children with hearing impairment. Furthermore, the fitter should fine-tune FL parameters for each individual listener, regardless of type of FL technology.

Sound quality degradation of audio equipment by hum noise induced by non-audible high frequency electrical noise

In recent audio equipment, induction and inflow of non-audible high frequency (HF) electrical noise to audio equipment is increasing more and more due to implementation of personal computer / network connectivity and high-resolution capability in digital audio equipment. In this study, in order to investigate another causes of influence of non-audible HF electric noise on playback sound, we measured a loudspeaker output of an audio amplifier by the audio analyser with inputting single HF sine wave from a function generator to analog input of the audio amplifier. From measured results, it was found that the power of harmonics of 60 Hz (hum noise) on the loudspeaker output of some audio amplifiers increased when single HF sine wave was inputted. Tendency of the increasing of the hum noise component depended on the frequency of the inputted sine wave and model of audio amplifier. These results show the importance of consideration of HF characteristic and countermeasure to the HF noise also in circuit design and tuning of audio equipment.

Intra-gender statistical singing voice conversion with direct waveform modification using log-spectral differential

This paper presents a novel intra-gender statistical singing voice conversion (SVC) technique with direct waveform modification based on the log-spectrum differential (DIFFSVC) that can convert the voice timbre of a source singer into that of a target singer without vocoder-based waveform generation of the converted singing voice. SVC makes it possible to convert the singing voice characteristics of an arbitrary source singer into those of an arbitrary target singer by converting some of its acoustic features, such as F0, aperiodicity, and spectral features based on a statistical conversion function. However, the sound quality of the converted singing voice is typically degraded compared with that of a natural singing voice, owing to various factors, such as analysis and modeling errors in the vocoding process and over-smoothing of the converted feature trajectory. To alleviate sound quality degradation, we propose a statistical conversion process that directly modifies the signal in the waveform domain by estimating the difference in the spectra of the source and target singers’ singing voices. Additionally, we propose the following several techniques for the DIFFSVC method: 1) derivation of a differential Gaussian mixture model (DIFFGMM) from a conventional Gaussian mixture model (GMM) and 2) a parameter generation algorithm considering the global variance (GV). The experimental results demonstrate that the proposed DIFFSVC methods enable significant improvements in the sound quality of the converted singing voice, while preserving the conversion accuracy of the singer’s identity compared with conventional SVC.

A feedback active noise control for in-ear headphone with robust on-line secondary path modeling

The active noise control (ANC) technique can be used to improve the comfortness and quality of the music listening with an in-ear headphone. For a compact headphone design, the feedback ANC operated with the filtered-x LMS (FxLMS) algorithm is preferred. In practice, the error in the estimated secondary path can be problematic for the FxLMS algorithm, since it can lead the algorithm to an unstable state, and thus cause degradation of sound quality with the in-ear headphone. In this study, an adaptive residual music canceler (RMC) is proposed for enhancing the accuracy of the reference signal of the feedback ANC. Since RMC is designed to track the bias of the current secondary path estimate, the secondary path is continuously updated by combining the previous estimate of the secondary path with the current weight vector of RMC. In addition, variable step-size schemes are developed for both the control and secondary path estimation filters, which enable the ANC system to adapt quickly and robustly to the va...

Objective evaluation of sound quality for attacks on robust audio watermarking

Various attacks on robust audio watermarking have been proposed. Reversible signal processing attacks, such as sampling frequency conversion, degrade sound quality of the distributed watermarked audio (stego audio) and disturb extraction of hidden data so that copyright detection systems using automated crawling are invalidated. Reversible signal processing of the attack can recover sound quality of the degraded audio data. In order to prove validity and security of audio watermarking system, analysis of the presumed attacks or reversible signal processing on stego audio, is required. However, these attacks on audio signal also degrade sound quality of commercial music where such pieces of music are considered to be not suitable for appreciation. Therefore, degradation of sound quality induced by various attacks should be taken into account to decide if the intensity of the attacks are realistic. In this study, objective audio quality measurement (PEAQ) was applied to the audio signals including typical perceptual coding, MP3, tandem MP3, MPEG4AAC, and reversible signal processing of sampling frequency conversion, noise addition, frequency shift, bandpass filtering, and echo addition. The results indicate requirements for robustness and criteria of the attacks on high quality and robust audio watermarking technology.

Restoring clipped speech signal based on spectral transformation of each frequency band

In recent years, high-quality speech recording is requested for comfortably using a communication system. However, a clipping distortion caused by input exceeding the maximum range of amplifier is one of the problems with a sound quality degradation. Although linear prediction method has been conventionally proposed for restoring a clipped speech signal, it has a problem that the frequently clipped speech signal degrades the restoration performance by increasing the prediction errors. In this paper, we propose a method for restoring a clipped speech signal based on a spectral transformation of each frequency band. In this method, the spectral envelope of target speech signal in each frame is approximated to the spectral envelope of original speech signal to remove the influence of a clipping distortion. In particular, the spectral envelope in higher frequency domain including a static characteristic of the speaker is replaced with the spectral envelope of the unclipped speech signal prepared in advance. Then, the spectral envelope in lower frequency domain including a characteristic of phoneme is approximated with Gaussian Mixture Models. We carried out an evaluation experiment for sound quality of speech signal processed by the proposed method. As a result, we confirmed the effectiveness of the proposed method.

Clinical Evaluation of Signal-to-Noise Ratio–Based Noise Reduction in Nucleus® Cochlear Implant Recipients

The aim of this study was to investigate whether a real-time noise reduction algorithm provided speech perception benefit for Cochlear™ Nucleus® cochlear implant recipients in the laboratory. The noise reduction algorithm attenuated masker-dominated channels. It estimated the signal-to-noise ratio of each channel on a short-term basis from a single microphone input, using a recursive minimum statistics method. In this clinical evaluation, the algorithm was implemented in two programs (noise reduction programs 1 [NR1] and 2 [NR2]), which differed in their level of noise reduction. These programs used advanced combination encoder (ACE™) channel selection and were compared with ACE without noise reduction in 13 experienced cochlear implant subjects. An adaptive speech reception threshold (SRT) test provided the signal-to-noise ratio for 50% sentence intelligibility in three different types of noises: speech-weighted, cocktail party, and street-side city noise. In all three noise types, mean SRTs for both NR programs were significantly better than those for ACE. The greatest improvement occurred for speech-weighted noise; the SRT benefit over ACE was 1.77 dB for NR1 and 2.14 dB for NR2. There were no significant differences in speech perception scores between the two NR programs. Subjects reported no degradation in sound quality with the experimental programs. The noise reduction algorithm was successful in improving sentence perception in speech-weighted noise, as well as in more dynamic types of background noise. The algorithm is currently being trialed in a behind-the-ear processor for take-home use.

Predicting Preference Judgments of Individual Normal and Hearing-Impaired Listeners With Gaussian Processes

A probabilistic kernel approach to pairwise preference learning based on Gaussian processes is applied to predict preference judgments for sound quality degradation mechanisms that might be present in a hearing aid. Subjective sound quality comparisons for 14 normal-hearing and 18 hearing-impaired subjects were used for evaluating the predictive performance. Stimuli were sentences subjected to three kinds of distortion (additive noise, peak clipping, and center clipping) with eight levels of degradation for each distortion type. The kernel approach gives a significant improvement in preference predictions of hearing-impaired subjects by individualizing the learning process. A significant difference is shown between normal-hearing and hearing-impaired subjects, because of nonlinearities in the perception of hearing-impaired subjects. In particular, hearing-impaired subjects have significant nonlinear preference judgments when making pairwise comparisons between peak clipped sentences with different clipping thresholds. The probabilistic kernel approach is shown to be robust when generalizing over distortions and over subjects.

Pitch Conversion Method And Device For Converting A Pitch Of An Input Signal Into A Desired Pitch

In a pitch conversion method and device which can reduce data throughput while suppressing a degradation of sound quality due to a pitch conversion as much as possible, an input signal pitch pattern per predetermined processing unit and a target pitch pattern are inputted, and a degradation degree indicating how a waveform of the input signal degrades upon pitch conversion from the input signal pitch pattern to the target pitch pattern is calculated. Alternatively, a degradation degree corresponding to a voice state and a phonemic type of the input signal is extracted from a database in which all of combinations of voice states and phonemic types estimated are associated with the degradation degrees to be recorded. Then, a pitch converter which performs a pitch conversion with small data throughput and a pitch converter which performs a pitch conversion with large data throughput are switched over depending on the degradation degree.

Efficient FFT Algorithm for Psychoacoustic Model of the MPEG-4 AAC

This paper proposes an efficient FFT algorithm for the Psycho-Acoustic Model (PAM) of MPEG-4 AAC. The proposed algorithm synthesizes FFT coefficients using MDCT and MDST coefficients through circular convolution. The complexity of the MDCT and MDST coefficients is approximately half of the original FFT. We also design a new PAM based on the proposed FFT algorithm, which has 15% lower computational complexity than the original PAM without degradation of sound quality. Subjective as well as objective test results are presented to confirm the efficiency of the proposed FFT computation algorithm and the PAM.

Clipping Prevention Scheme for MPEG Surround

MPEG Surround has a potential clipping problem since its encoding is based on downmixing a multichannel signal. We propose a clipping prevention scheme for MPEG Surround, which is composed of modification and recovery processes of a downmix signal with recovery information conveyed in arbitrary downmix gains of an MPEG Surround bitstream. Experiments show that the proposed scheme effectively prevents sound quality degradation caused by clipping problems with negligible additional bit rates.

Mixing recorder, control method therefor, and computer-readable medium including program for implementing the control method

There is provided a mixing recorder which enables the user to readily produce music using overdubbing and/or other recording techniques while suppressing degradation of sound quality to the minimum and enables the user to easily find out his/her desired mixing result from a large number of mixing results obtained in the process of mixing. An audio signal is input, and a source file is read out from a memory card and an audio signal is reproduced based on the source file. The input audio signal and the reproduced audio signal are mixed into a mixed signal, which is then stored as a new source file in the memory card. The source file stored in advance in the memory card is backed up before the new source file is stored in the memory card, and in the back-up, the source file is automatically backed up by generating a file given a new name associated with a name of a source file to be backed up and having the same contents as the contents of the source file.

The Effect of Digital Phase Cancellation Feedback Reduction Systems on Amplified Sound Quality

The effect of feedback reduction (FBR) systems on sound quality recorded from two commercially available hearing aids was evaluated using paired comparison judgments by 16 participants with mild to severe sloping hearing loss. These comparisons were made with the FBR systems on and off without audible feedback and while attempting to control for differences in gain and clinical fitting factors. Wilcoxon signed rank test analyses showed that the participants were unable to differentiate between signals that had been recorded with the FBR systems on and off within the same hearing aid. However, significant between-instrument differences in sound quality were identified. The results support the activation of the FFT-phase cancellation FBR systems evaluated herein without concern for a noticeable degradation of sound quality.

Use of frequency transposition in a thin-tube open-ear fitting

The use of thin-tube open-ear fitting has become increasingly popular in the last few years. This type of fitting offers improvements in the wearers’ perception of their own voice, especially for people with a mild hearing loss or a highfrequency hearing loss. It also provides an instant fit, is easy to demonstrate, and improves the appearance of behind-the-ear (BTE) hearing aid to wearers. Both the acceptance rate and the return rate of these models are more favorable than the average for hearing aids overall. Although the openness of the fitting may reduce the potential improvement in signal-to-noise ratio (SNR) offered by a directional microphone, Kuk et al. have shown that an open-fit directional microphone provides an SNR improvement of 2-3 dB over the same hearing aid in its omnidirectional microphone mode. 1 On the other hand, thin-tube open-ear fittings may compromise the audibility of the high frequencies for some wearers. 2 In this paper, we will review some of these potential compromises and discuss why frequency transposition may be a solution for the limitations. We will also show that frequency transposition improves the perception of everyday sounds and may enhance consonant recognition for some people fitted with thin-tube openear devices. ISSUES WITH THIN-TUBE, OPEN-EAR FITTINGS

Audio decoding apparatus and method for band expansion with aliasing adjustment

An audio decoding apparatus decodes high frequency component signals using a band expander that generates multiple high frequency subband signals from low frequency subband signals divided into multiple subbands and transmitted high frequency encoded information. The apparatus is provided with an aliasing detector and an aliasing remover. The aliasing detector detects the degree of occurrence of aliasing components in the multiple high frequency subband signals generated by the band expander. The aliasing remover suppresses aliasing components in the high frequency subband signals by adjusting the gain used to generate the high frequency subband signals. Thus occurrence of aliasing can be suppressed and the resulting degradation in sound quality can be reduced, even when real-valued subband signals are used in order to reduce the number of operations.

Voice quality conversion using interactive evolution of prosodic control

Recent years have seen the birth of new markets using voice quality conversion technology for a variety of application fields including man-personal machine interfaces, addition of narration in multimedia-content editing, and computer games. Optimal parameters for voice quality conversion, however, are speaker dependent, and consequently, no clear-cut algorithm has existed in the past and parameter adjustment has usually been performed by an experienced designer on a trial and error basis. This paper proposes the application of evolutionary computation, a stochastic search technique based on organic evolution, to parameter adjustment for voice conversion, and reports on several experimental results applicable to the fitting of prosodic coefficients. Evolutionary computation is said to be “applicable to even cases where the properties of the target function are not well known,” and we decided to apply it considering that this feature might be effective in our study. Providing an explicit evaluative function for evolutionary computation, however, is difficult, and we here adopt an interactive-evolution system in which genetic manipulation is performed repeatedly while evaluating results based on human emotions. Evaluation experiments were performed on raw human speech recorded by a microphone and speech mechanically synthesized from text. It was found that the application of evolutionary computation could achieve voice conversion satisfying specific targets with relatively little degradation of sound quality and no impression of artificial processing in comparison to parameter adjustment based on designer experience or trial and error. This paper also shows that prosodic conversion coefficients determined by the interactive evolution technique, while exhibiting speaker dependency, is not text dependent.