Subjective Listening Tests Research Articles

A new a priori SNR estimator based on multiple linear regression technique for speech enhancement

We propose a new approach to estimate the a priori signal-to-noise ratio (SNR) based on a multiple linear regression (MLR) technique. In contrast to estimation of the a priori SNR employing the decision-directed (DD) method, which uses the estimated speech spectrum in previous frame, we propose to find the a priori SNR based on the MLR technique by incorporating regression parameters such as the ratio between the local energy of the noisy speech and its derived minimum along with the a posteriori SNR. In the experimental step, regression coefficients obtained using the MLR are assigned according to various noise types, for which we employ a real-time noise classification scheme based on a Gaussian mixture model (GMM). Evaluations using both objective speech quality measures and subjective listening tests under various ambient noise environments show that the performance of the proposed algorithm is better than that of the conventional methods.

Cepstrum-Based Bandwidth Extension for Super-Wideband Coders

This letter proposes a bandwidth extension (BWE) method using the cepstral envelope coding and duplication of quantized wideband (WB) signals by means of analysis -by-synthesis (AbS) for super-wideband (SWB) coders. In the proposed method, a high frequency band is generated by utilizing the quantized cepstral coefficients extracted from the envelope and the quantized modified discrete cosine transform (MDCT) shape of the wideband signal. The proposed method is compared with the latest G.718 SWB codec and the experimental results show that the proposed method outperforms the baseline codec both in subjective listening tests and objective performance measures.

Contextual Additive Structure for HMM-Based Speech Synthesis

This paper proposes a spectral modeling technique based on an additive structure of context dependencies for HMM-based speech synthesis. Contextual additive structure models can represent complicated dependencies between acoustic features and context labels using multiple decision trees. However, the computational complexity of the context clustering is too high for the full context labels of speech synthesis. To overcome this problem, this paper proposes two approaches; covariance parameter tying and a likelihood calculation algorithm using the matrix inversion lemma. Additive structure models can be applied to HMM-based speech synthesis using these techniques and speech quality would significantly be improved. Experimental results show that the proposed method outperforms the conventional one in subjective listening tests.

Pitch modification of speech signal using harmonic model with time-varying parameters

The paper presents a solution to the problem of pitch modification of speech. The problem occurs in different speech processing applications such as voice conversion, accent correction, hiding speaker's personality and other. Developed processing scheme for voiced part of speech is based on the harmonic model with nonstationary (time-varying) parameters. In order to improve frequency resolution of the model parameters are extracted using narrow-band filtering in warped time domain aligned to instantaneous pitch frequency. Using subjective listening tests it is shown that developed system provides high naturalness and intelligibility of reconstructed speech and can be applied to wideband and narrowband communication channels with various coding standards (including G.711 and GSM).

Body Conducted Speech Enhancement by Equalization and Signal Fusion

This paper studies body-conducted speech for noise robust speech processing purposes. As body-conducted speech is typically limited in bandwidth, signal processing is required to obtain a signal that is both high in quality and low in noise. We propose an algorithm that first equalizes the body-conducted speech using filters obtained from a pre-defined filter set and subsequently fuses this equalized signal with a noisy conventional microphone signal using an optimal clean speech amplitude and phase estimator. We evaluated the proposed equalization and fusion technique using a combination of a conventional close-talk and a throat microphone. Subjective listening tests show that the proposed method successfully fuses the speech quality of the conventional signal and the noise robustness of the throat microphone signal. The listening tests also indicate that the inclusion of the body-conducted signal can improve single-channel speech enhancement methods, while a calculated set of objective signal quality measures confirm these observations.

Audio Watermarking Systems - Design, Implementation and Evaluation of an Echo Hiding Scheme Using Subjective Tests and Common Distortions

Today's digital media have made the product (digital content) very flexible and diminished the cost of its distribution. However, it contributes on piracy explosion as digital content can be duplicated and re-distributed at virtually no cost. Watermarking technology appears in order to protect the intellectual property and fight the piracy. It consists on embedding data like copyright labels inside a data source without changing its perceptual quality. In audio domain, watermarking techniques rely on the imperfection of the human auditory system in order to embed data. In this paper, we completed a design based on echo hiding technique and implement it in MATLAB. The main idea of this method is to embed data into an original signal by introducing an echo with the appropriate delay. Subjective listening tests reveal that the watermarks are imperceptible. Fidelity tests show that quantity of distortion imposed by watermarks on a signal is small. Robustness tests against common signal processing reveal good responses. The watermark information is always detectable and recoverable.

In situ evaluation of surround sound system performance

Surround sound systems are produced with the intention of reproducing the spatial aspects of sound, such as localization and envelopment. As part of his work on Ambisonics, Gerzon developed two metrics, the velocity and energy localization vectors, which are intended to predict the localization performance of a system. These are used during the design process to optimize the decoder that supplies signals to the loudspeaker array. At best, subjective listening tests are conducted on the finished system, but no objective assessments of the spatial qualities are made to verify that the realized performance correlates the predictions. In the present work, binaural recordings were made of a 3-D 24-loudspeaker installation at Stanford’s Bing Studio. Test signals were used to acquire the binaural impulse response of each loudspeaker in the array and of Ambisonic reproduction using the loudspeaker array. The measurements were repeated at several locations within the hall. Subsequent analysis calculated the ITDs and ILDs for all cases. Initial results from the analysis of the ITDs and ILDs for the center listening position show ITDs, which correspond very closely to what is expected in natural hearing, and ILDs, which are similar to natural hearing.

Listening as your neighbors: Simulation and evaluation of building structural transmission

The enjoyment of music can be jeopardized by the knockings on the door and your furious neighbor shows up. Building structural transmission passes annoying sound to neighbors even when the audio is at low volume. In this paper, we implement a computational model for the simulation and evaluation of building structural transmission based on field measurements of structural transmission profiles and subjective listening tests. This model is intended to serve as a signal analysis tool that allows the audio engineers to evaluate the interference level using computer software and monitor system, so they can assess and mitigate these interferences without the need of conducting tedious subjective evaluation experiments in real acoustical environments. Our proposed transmission models are summarized from multiple acoustical measurements conducted in real room pairs to ensure its authenticity. Essentially we build “virtual” room pairs so we can simulate the interferences reaching the neighbors. We also implement a computational auditory model based on audio content analysis and subjective evaluation experiments to automatically evaluate the interference level caused by the transmitted sound. The proposed simulation and evaluation tool finds extensive applications in various areas of audio engineering such as audio content production, noise control, and audio system design.

Parametric Audio Coding With Exponentially Damped Sinusoids

Sinusoidal modeling is one of the most popular techniques for low bitrate audio coding. Usually, the sinusoidal parameters (amplitude, pulsation and phase of each sinusoidal component) are kept constant within a time segment. An alternative model, the so-called Exponentially-Damped Sinusoidal (EDS) model, includes an additional damping parameter for each sinusoidal component to better represent the signal characteristics. It was however never shown that the EDS model could be efficient for perceptual audio coding. To that aim, we propose in this paper an efficient analysis/synthesis framework with dynamic time-segmentation on transients and psychoacoustic modeling, and an asymptotically optimal entropy-constrained quantization method for the four sinusoid parameters (e.g., including damping). We then apply this coding technique to real audio excerpts for a given entropy target corresponding to a low bitrate (20 kbits/s), and compare this method with a classical sinusoidal coding scheme using a constant-amplitude sinusoidal model and the perceptually weighted Matching Pursuit algorithm. Subjective listening tests show that the EDS model is more efficient on audio samples with fast transient content, and similar to the classical model for more stationary audio samples.

DRA Audio Coding Standard

Chinas DRA audio coding standard is shown to be a barebone transform coder, utilizing transient-localized MDCT for improved pre-echo suppression and statistic allocation of codebooks for high entropy coding efficiency. A signal path of up to 24 bits is provided throughout the codec so that highest audio quality can be delivered if bit rate suffices. Results of five ITU-R BS.1116 compliant subjective listening tests are presented.

A digital waveguide-based approach for Clavinet modeling and synthesis

The Clavinet is an electromechanical musical instrument produced in the mid-twentieth century. As is the case for other vintage instruments, it is subject to aging and requires great effort to be maintained or restored. This paper reports analyses conducted on a Hohner Clavinet D6 and proposes a computational model to faithfully reproduce the Clavinet sound in real time, from tone generation to the emulation of the electronic components. The string excitation signal model is physically inspired and represents a cheap solution in terms of both computational resources and especially memory requirements (compared, e.g., to sample playback systems). Pickups and amplifier models have been implemented which enhance the natural character of the sound with respect to previous work. A model has been implemented on a real-time software platform, Pure Data, capable of a 10-voice polyphony with low latency on an embedded device. Finally, subjective listening tests conducted using the current model are compared to previous tests showing slightly improved results.

An adaptive post-filtering method producing an artificial Lombard-like effect for intelligibility enhancement of narrowband telephone speech

Post-filtering can be used in mobile communications to improve the quality and intelligibility of speech. Energy reallocation with a high-pass type filter has been shown to work effectively in improving the intelligibility of speech in difficult noise conditions. This paper introduces a post-filtering algorithm that adapts to the background noise level as well as to the fundamental frequency of the speaker and models the spectral effects observed in natural Lombard speech. The introduced method and another post-filtering technique were compared to unprocessed telephone speech in subjective listening tests in terms of intelligibility and quality. The results indicate that the proposed method outperforms the reference method in difficult noise conditions.

가상 음원 이미징을 위한 향상된 진폭 패닝 기법

In this paper, the problems of the conventional amplitude panning method for a stereophonic panning system are analyzed. We observed that the distortion showed a feedforward comb filter response. As a remedy to this distortion, we propose a stereophonic panning system using a feedback comb filter. The comb filter is designed to minimize the difference between interaural level difference(ILD) of the proposed system and that of HRTF because ILD is most salient cue for the perception of the source direction. The proposed system is configured to operate selectively for the frequency band related to the source direction. The performance of the proposed system is verified by subjective listening tests.

Development of Low Bit Rate Speech Encoder based on Vector Quantization and Compressive Sensing

Speech coding is a representation of a digitized speech signal using as few bits as possible, while maintaining reasonable level of speech quality. Due to growing need for bandwidth conservation in wireless communication, the research in speech coding has increased. Recently, Compressive Sensing (CS) is gaining a great interest because of its ability to recover original signals by taking only few measurements. CS is a new approach that goes against the common data acquisition methods. In this research, a new system of speech encoding system is developed using compressive sensing. Since CS performs well in sparse signals, different sparsifying transforms are analyzed and compared using Gini coefficient. The quality of the speech coder is evaluated using Perceptual Evaluation of Speech Quality (PESQ), Signal-to-Noise Ratio (SNR) and subjective listening tests. Results show that the speech coders have achieved a PESQ score of 3.16 at 4 kbps which is a good quality as confirmed by listening tests. Furthermore, the coder is also compared with Code Excited Linear Prediction (CELP) coder.

Signal-to-noise ratio adaptive post-filtering method for intelligibility enhancement of telephone speech

Post-filtering can be utilized to improve the quality and intelligibility of telephone speech. Previous studies have shown that energy reallocation with a high-pass type filter works effectively in improving the intelligibility of speech in difficult noise conditions. The present study introduces a signal-to-noise ratio adaptive post-filtering method that utilizes energy reallocation to transfer energy from the first formant to higher frequencies. The proposed method adapts to the level of the background noise so that, in favorable noise conditions, the post-filter has a flat frequency response and the effect of the post-filtering is increased as the level of the ambient noise increases. The performance of the proposed method is compared with a similar post-filtering algorithm and unprocessed speech in subjective listening tests which evaluate both intelligibility and listener preference. The results indicate that both of the post-filtering methods maintain the quality of speech in negligible noise conditions and are able to provide intelligibility improvement over unprocessed speech in adverse noise conditions. Furthermore, the proposed post-filtering algorithm performs better than the other post-filtering method under evaluation in moderate to difficult noise conditions, where intelligibility improvement is mostly required.

Estimation and quantization of ICC-dependent phase parameters for parametric stereo audio coding

Conventional parametric stereo (PS) audio coding employs inter-channel phase difference and overall phase difference as phase parameters. In this article, it is shown that those parameters cannot correctly represent the phase relationship between the stereo channels when inter-channel correlation (ICC) is less than one, which is common in practical situations. To solve this problem, we introduce new phase parameters, channel phase differences (CPDs), defined as the phase differences between the mono downmix and the stereo channels. Since CPDs have a descriptive relationship with ICC as well as inter-channel intensity difference, they are more relevant to represent the phase difference between the channels in practical situations. We also propose methods of synthesizing CPDs at the decoder. Through computer simulations and subjective listening tests, it is confirmed that the proposed methods produce significantly lower phase errors than conventional PS, and it can noticeably improve sound quality for stereo inputs with low ICCs.

Disentangling preference ratings of concert hall acoustics using subjective sensory profiles

Subjective evaluation of acoustics was studied by recording nine concert halls with a simulated symphony orchestra on a seat 12 m from the orchestra. The recorded music was spatially reproduced for subjective listening tests and individual vocabulary profiling. In addition, the preferences of the assessors and objective parameters were gathered. The results show that concert halls were discriminated using perceptual characteristics, such as Envelopment/Loudness, Reverberance, Bassiness, Proximity, Definition, and Clarity. With these perceptual dimensions the preference ratings can be explained. Seventeen assessors were divided into two groups based on their preferences. The first group preferred concert halls with relatively intimate sound, in which it is quite easy to hear individual instruments and melody lines. In contrast, the second group preferred a louder and more reverberant sound with good envelopment and strong bass. Even though all halls were recorded exactly at the same distance, the preference is best explained with subjective Proximity and with Bassiness, Envelopment, and Loudness to some extent. Neither the preferences nor the subjective ratings could be fully explained by objective parameters (ISO3382-1:2009), although some correlations were found.

Stereophonic hands-free communication system based on microphone array fixed beamforming: real-time implementation and evaluation

Abstract In this article, the authors propose an optimally designed fixed beamformer (BF) for stereophonic acoustic echo cancelation (SAEC) in real hands-free communication applications. Several contributions related to the combination of beamforming and echo cancelation have appeared in the literature so far, but, up to the authors’ knowledge, the idea of using optimal fixed BFs in a real-time SAEC system both for echo reduction and stereophonic audio rendering is first addressed in this contribution. The employment of such designed BFs allows positively addressing both issues, as the several simulated and real tests seem to confirm. In particular, the stereo-recording quality attainable through the proposed approach has been preliminarily evaluated by means of subjective listening tests. Moreover, the overall system robustness against microphone array imperfections and noise presence has been experimentally evaluated. This allowed the authors to implement a real hands-free communication system in which the usage of the proposed beamforming technique has proven its superiority with respect to the usual two-microphone one in terms of echo reduction, and guaranteeing a comparable spatial image. Moreover, the proposed framework requires a low computational cost increment with regard to the baseline approach, since only few extra filtering operations with short filters need to be executed. Nevertheless, according to the performed simulations, the BF-based SAEC configuration seems to not require the signal decorrelation module, resulting in an overall computational saving.

A Parametric Objective Quality Assessment Tool for Speech Signals Degraded by Acoustic Echo

This paper discusses the automatic quality assessment of echo-degraded speech in the context of teleconference systems. Subjective listening tests conducted over a carefully designed database of signals degraded by acoustic echo have been used to assess how this impairment is perceived and to determine which parameters have a significant impact on speech quality. The results have shown that, similarly to electric transmission line echo, acoustic echo is mainly influenced by echo delay and echo gain. Based on this observation, a mapping between these two parameters and the mean subjective score is devised. Moreover, a signal-based algorithm for the estimation of these parameters is described, and its performance is evaluated. The complete system comprising both the parameter estimators and the mapping function achieves a correlation of 94% between predicted and actual subjective scores, and can be employed as a non-intrusive monitoring tool for in-service quality evaluation of teleconference systems. Further validation indicates the operating range of the proposed quality assessment tool can be extended by proper retraining.

Optimization and evaluation of sigmoid function with a priori SNR estimate for real-time speech enhancement

In this paper, an a priori signal-to-noise ratio (SNR) estimator with a modified sigmoid gain function is proposed for real-time speech enhancement. The proposed sigmoid gain function has three parameters, which can be optimized such that they match conventional gain functions. In addition, the joint temporal dynamics between the SNR estimate and the spectral gain function is investigated to improve the performance of the speech enhancement scheme. As the widely-used decision-directed (DD) a priori SNR estimate has a well-known one-frame delay that leads to the degradation of speech quality, a modified a priori SNR estimator is proposed for the DD approach to overcome this delay. Evaluations are performed by utilizing the objective evaluation metric that measures the trade-off between the noise reduction, the speech distortion and the musical noise in the enhanced signal. The results are compared using the PESQ and the SNRseg measures as well as subjective listening tests. Simulation results show that the proposed gain function, which can flexibly model exponential distributions, is a potential alternative speech enhancement gain function.