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In this paper, we propose a bandwidth-scalable stereo audio coding method based on a layered structure. The proposed stereo coding method encodes super-wideband (SWB) stereo signals and is able to decode either wideband (WB) stereo signals or SWB stereo signals, depending on the network congestion. The performance of the proposed stereo coding method is then compared with that of a conventional stereo coding method that separately decodes WB or SWB stereo signals, in terms of subjective quality, algorithmic delay, and computational complexity. Experimental results show that when stereo audio signals sampled at a rate of 32kHz are compressed to 64kbit/s, the proposed method provides significantly better audio quality with a 64-sample shorter algorithmic delay, and comparable computational complexity.

This article provides a compact overview of the history, technology, and performance of MPEG Surround. The technology of MPEG Surround is based on the spatial audio coding (SAC) principle: In the encoder, a mono- or stereophonic down- mix is generated from the multichannel input signal, and additional parametric side information is extracted to guide the subsequent up-mix procedure in the decoder.With Moving Pictures Expert Group (MPEG) Surround, a data-rate efficient coding scheme for high-quality multichannel sound using novel parametric coding techniques has been standardized.

This paper investigates the problem of retrieving karaoke music using query-by-singing techniques. Unlike regular CD music, where the stereo sound involves two audio channels that usually sound the same, karaoke music encompasses two distinct channels in each track: one is a mixture of the lead vocals and background accompaniment, and the other consists of accompaniment only. Although the two audio channels are distinct, the accompaniments in the two channels often resemble each other. We exploit this characteristic to: i) infer the background accompaniment for the lead vocals from the accompaniment-only channel, so that the main melody underlying the lead vocals can be extracted more effectively, and ii) detect phrase onsets based on the Bayesian information criterion (BIC) to predict the onset points of a song where a user's sung query may begin, so that the similarity between the melodies of the query and the song can be examined more efficiently. To further refine extraction of the main melody, we propose correcting potential errors in the estimated sung notes by exploiting a composition characteristic of popular songs whereby the sung notes within a verse or chorus section usually vary no more than two octaves. In addition, to facilitate an efficient and accurate search of a large music database, we employ multiple-pass dynamic time warping (DTW) combined with multiple-level data abstraction (MLDA) to compare the similarities of melodies. The results of experiments conducted on a karaoke database comprised of 1071 popular songs demonstrate the feasibility of query-by-singing retrieval for karaoke music.

In this paper, a model of human sound localisation is described, and its prediction is compared to the results of listening tests. The model takes binaural signals as the input, processing them in a series of signal processing modules, which simulate the peripheral, binaural and the central stages of spatial hearing. In particular, the central processor of the model considers the excitation-inhibition (EI) cell activity patterns as the internal representation of available cues, and the source location estimates are obtained by using a simple pattern-matching procedure. In the listening tests, stereophonic images were presented to the listener's front, where the stimulus was either broadband or 1/3 octave band noise at 7 centre frequencies from 0.5 kHz to 6 kHz. The subjective responses compared well to the model prediction across frequency except for some cases where the image location was overestimated. Also, the prediction for the localisation of broadband phantom images agreed well with the test results, where the model prediction was integrated across frequency according to a tentatively suggested weighting function. Although the neuroscientific background is weak for the model, the good agreement with the subjective responses suggests that the model is worth investigating further.

Source separation and up-mixing in real commercial music recordings is a challenging problem. In the last few years, some algorithms have provided interesting results, but the problem remains unsolved. In this paper we describe a method for separating the sources present in a two channel mixture based on the panning coefficients used in the stereo mixdown. The sources are separated by estimating time-frequency masks using the multilevel extension of the Otsu thresholding algorithm used in image segmentation. A refinement step is also carried out for extraction and reassignment of inter-source residuals. Examples of application and performance evaluation are also discussed.

Loudspeaker reproduction systems are subject to a compromise between spatial realism and cost. By simulating loudspeaker reproduction on headphones, the resulting spatial realism is limited accordingly, despite the virtually unlimited spatial imaging capabilities of binaural audio rendering technology. More particularly, phantom imaging as often used for stereo audio material intended for loudspeaker reproduction is subject to various restrictions in terms of loudspeaker positioning in simulated space. As a consequence, phantom imaging should preferably be avoided when simulating virtual loudspeakers over headphones, especially if head tracking is incorporated or if a wide sound stage is desired. A novel method is described to extract phantom sound sources from stereo audio content and convert these to sound sources in a virtual listening environment.

This article presents a review on vision-aided systems and proposes an approach for visual rehabilitation using stereo vision technology. The proposed system utilizes stereo vision, image processing methodology, and a sonification procedure to support blind mobilization. The developed system includes wearable computer, stereo cameras as vision sensor, and stereo earphones, all molded in a helmet. The image of the scene in front of the visually handicapped is captured by the vision sensors. The captured images are processed to enhance the important features in the scene in front for mobilization assistance. The image processing is designed as a model of human vision by identifying the obstacles and their depth information. The processed image is mapped onto musical stereo sound for the blind's understanding of the scene in front. The developed method has been tested in the indoor and outdoor environments and the proposed image processing methodology is found to be effective for object identification.

This paper describes a heterogeneous multi-core processor (HMCP) architecture that integrates general-purpose processors (CPUs) and accelerators (ACCs) to achieve exceptional performance as well as low-power consumption for the SoCs of embedded systems. The memory architectures of CPUs and ACCs were unified to improve programming and compiling efficiency. Advanced audio codec-low complexity (AAC-LC) stereo audio encoding was parallelized on a heterogeneous multi-core having homogeneous processor cores and dynamically reconfigurable processor (DRP) ACC cores in a preliminary evaluation of the HMCP architecture. The performance evaluation revealed that 54times AAC encoding was achieved on the chip with two CPUs at 600 MHz and two DRPs at 300 MHz, which achieved encoding of an entire CD within 1- 2 min.

The system analysis of the MIMO optimal feedback quantization used to produce switching signals is proposed here. The switching signals are then fed into the switch mode power amplifier to drive actuator array. The architecture of the power amplifier used in this paper allows the sharing of switching elements among actuators and is different from the traditional one-to-one method in actuating a large number of devices. To actuate all of the devices at the same time, the MIMO optimal feedback quantization generates control signals optimally basing on minimizing the weighted measure of quantization error. The discussion on bounding the system states under the zero initial conditions is made. The result tells that the stability can be achieved by limiting the maximum amplitude of inputs when the MIMO system has stable zeros. To simplify the computational complexity of quantization, a sub-optimal method for actuating three-actuator system is mentioned. A design example of 2-input 2-output system which is applied to the class-D stereo audio amplifier (dual actuators) is addressed. The control performance and cross-talk behavior are investigated.

A portable ultra wide band radio direction finder has been constructed based on the principle of stereophonic direction recognition of sound by human ears. The instrument consists of two log periodic antennas having identical electrical properties. They are positioned in a plane, preferably parallel to the ground. The directivities of the antennas are aligned at slightly different directions with respect to each other. Their output powers are compared at the source frequency for the respective polarizations. The back lobes of the antennas have been reduced by symmetrically positioning two metallic plates (reflectors) behind the antennas. The antennas, reflectors and a compass are mounted over a video camera stand such that they could be manually positioned in the azimuth. Since the antennas are of identical make, ideally the radiation pattern of either antenna- reflector combination should behave as a flipped image of the other set. For any particular polarization and frequency, the outputs from the antennas are compared with each other. The antenna assembly is rotated between 0 and 360 ◦ in the azimuth until the power difference gets minimized. This position relates to the direction of the source and is indicated on the compass, provided there exits a single radio source at that frequency.

This study introduces a method for detection of ventricular depolarization activity and the transfer of this activity into an audible stereo audio signal. Heart potentials are measured by an ultra-high-frequency high-dynamic-range electrocardiograph (UHF-ECG) with a 25-kHz sampling rate. Averaged and prolonged UHF amplitude envelopes of V1–3 and V4–6 leads at a frequency range of 500–1000 Hz are used as a modulating function for two carrier audio frequencies. The right speaker makes it possible to listen to the depolarization of the septum and right ventricle (V1–3) and the left speaker the left ventricle lateral wall (V4–6). In the healthy heart, both speakers can be heard simultaneously. A delayed L or R speaker represents the dyssynchronous electrical activation of the ventricles. Examples of the normal heart, right bundle branch block and left bundle branch block can be heard at www.medisig.com/uhfecg.

The problem of side signal frame loss concealment for Mid/Side M/S stereo audio is addressed in this paper. The proposed level ratio based inter and intra channel LRIIC prediction method is designed to overcome the main challenge due to the diversified stereophonic nature of audio signals. To identify the stereophonic nature, we employ the time varying level ratio between mid also called monophonic signal and side signal. In the first phase, one Winer filter is designed from the monophonic signal and side signal of previous frame. And the other one is designed from the side signal of previous two frames. The two filters allow reconstruction of the current loss frame with inter and intra channel prediction respectively. In the second phase, available current frame of monophonic signal is used as the input of inter-channel Winer filter. While long-term prediction filter is employed to find the periodic components of previous frames as the input of intra-channel Winer filter. Finally, a level ratio based linear combination of inter and intra channel prediction output is employed to get the current lost side signal. Objective and subjective evaluation results for the proposed LRIIC approach, in comparison with existing techniques, all incorporated within a 3GPP AMR-WB+ decoder, provide evidence for gains across a variety of stereophonic signals.

Event-driven DSPs have the advantage of activity-dependent power consumption [1], and event-driven neural networks have shown superior power efficiency in real-time recognition tasks [2]. A bio-inspired silicon cochlea [3] functionally transforms sound input into multi-frequency-channel asynchronous event output, and hence is the natural candidate for the audio sensing frontend of event-driven signal processing systems like [1] and [2]. High-quality event encoding can be implemented as level-crossing (LC) ADCs, but the circuits are area- and power-inefficient [1]. Asynchronous delta modulation, the original form of LC sampling, on the other hand can be compactly realized even in small pixels of vision sensors [4]. Traditional audio processing employs digital FFTs and BPFs after signal acquisition by high-precision ADCs. However, it has been shown in [5] that for classification tasks like voice activity detection (VAD), good accuracy can still be attained when filtering is performed using low-power analog BPFs. This paper presents a 0.5V 55µW 64×2-channel binaural silicon cochlea aiming for ultra-low-power IoE applications like event-driven VAD, sound source localization, speaker identification and primitive speech recognition. The source-follower-based BPF and the asynchronous delta modulator (ADM) with adaptive self-oscillating comparison for event encoding are highlighted for the advancement of the system power efficiency.

In this paper, we present an approach to estimate mobile sound source trajectory. An artificially created sound source signal is used in this work. The main aim of this paper is to estimate the mobile object trajectory via sound processing methods. The performance of generalized cross correlation techniques is compared with that of noise reduction filters for the success of trajectory estimation. The azimuth angle between the sound source and receiver is calculated during the whole movement. The parameter of Interaural Time Difference (ITD) is utilized for determining azimuth angle. The success of estimated delay is compared with different types of Generalized Cross Correlation (GCC) algorithms. In this study, an approach for sound localization and trajectory estimation on 2D space is proposed. Besides, different types of pre-filter method are tried for removing the noise and signal smoothing of recorded sound signals. Some basic parameters of sound localization process are also explained. Moreover, the calculation error of average azimuth angle is compared with different GCC and pre-filtered methods. To conclude, it is observed that estimation of location and trajectory information of a mobile object from a stereo sound recording is realized successfully.

Recent studies encourage the development of sensorially-enriched media to enhance the user experience by stimulating senses other than sight and hearing. Sensory effects as odor, wind, vibration and light effects, as well as an enhanced audio quality, have been found to favour media enjoyment and to have a positive influence on the sense of Presence and on the perceived quality, relevance and reality of a multimedia experience. In particular, sports is among the genres that could benefit the most from these solutions. Several works have demonstrated also the technical feasibility of implementing and deploying end-to-end solutions integrating sensory effects into a legacy system. Thus, multi-sensorial media emerges as a mean to deliver a new form of immersive experiences to the mass market in a non-disruptive manner. However, many questions remain concerning issues as the sensory effects that can better complement a given audiovisual content or the best way in which to integrate and combine them to enhance the user experience of a target audience segment. The work presented in this paper aims to gain insight into the impact of binaural audio and sensory (light and olfactory) effects on the sports media experience, both at the overall level (average effect) and as a function of users' characteristics (heterogeneous effects). To this aim, we conducted an experimental study exploring the influence of these immersive elements on the quality and Presence dimensions of the media experience. Along the quality dimension, we look for possible variations on the quality scores assigned to the overall media experience and to the media components content, image, audio and sensory effects. The potential impact on Presence is analyzed in terms of Spatial Presence and Engagement. The users' characteristics considered encompass specific personal affective, cognitive and behavioral attributes. At the overall level we found that participants preferred binaural audio over standard stereo audio and that the presence of sensory effects increased significantly the level of Spatial Presence. Several heterogeneous effects were also revealed as a result of our experimental manipulations. Whereas binaural audio was found to have a generalized impact on the majority of the quality and Presence measures considered, the effects of sensory effects concentrate mainly on the Presence dimension. Personal characteristics explained most of the variation in the dependent variables, being individuals' preferences in relation to the content, knowledge of involved technologies, tendency to emotional involvement and conscientiousness among the user variables with the most generalized influence. In particular, the former two features seem to present a conflict in the allocation of attentional resources towards the media content versus the technical features of the system, respectively. Additionally, football fans' experience seems to be modulated by emotional processes whereas for not fans cognitive processes ---and in particular those related to quality judgment--- prevail.

International audience; We propose a projection-based method for the unmixing of multi-channel audio signals into their different constituent spatial objects. Here, spatial objects are modelled using a unified framework which handles both point sources and diffuse sources. We then propose a novel methodology to estimate and take advantage of the spatial dependencies of an object. Where previous research has processed the original multichannel mixtures directly and has been principally focused on the use of inter-channel covariance structures, here we instead process projections of the multichannel signal on many different spatial directions. These linear combinations consist of observations where some spatial objects are cancelled or enhanced. We then propose an algorithm which takes these projections as the observations, discarding dependencies between them. Since each one contains global information regarding all channels of the original multichannel mixture, this provides an effective means of learning the parameters of the original audio, while avoiding the need for joint-processing of all the channels. We further show how to recover the separated spatial objects and demonstrate the use of the technique on stereophonic music signals.

This paper presents a new demixing method that separates each source from a stereo sound mixture. Under the W-Disjoint Orthogonal assumption in DUET(Degenerate Unmixing Estimation Technique) algorithm. The proposed method is mainly processed in time-frequency domain by using windowed-fourier transform. In this paper there are two main contributions: a weighted mask by panning index distances and a binary mask by comparing each channel value. The former has tender demixing characteristic, and the latter has stronger demixing characteristic. In experimental results, we will show that both masks produce more robust demixing than the existing demixing methods do.

This paper presents a novel stereophonic acoustic echo canceling scheme without preprocessing. To accurately estimate echo path keeping the high level of performance in echo erasing, this scheme uses two filters, of which one filter is utilized as a guideline which does not erases echo but helps updating of the other filter, which actually erases echo. In addition, we propose a new filter dividing technique to apply to the filter divide scheme, and utilize this as the guideline. Numerical examples demonstrate that the proposed scheme improves the convergence behavior compared to conventional methods both in system mismatch (i.e., normalized coefficients error) and Echo Return Loss Enhancement (ERLE).

In this comprehensive study, algorithms for upmixing, downmixing, and joint up/downmixing are examined and compared. Five upmixing algorithms based on signal decorrelation and reverberation are employed to convert two-channel stereo signals to jive-channel signals. For downmixing, methods ranging from mixing with simple gain adjustment to more sophisticated head related transfer function (HRTF) filtering and crosstalk cancellation system (CCS) are utilized to downmix the center channel and the surround channels into the available two frontal loudspeakers. For situations where only two-channel content and loudspeakers are available, a number of up/down mixing schemes are used to simulate a virtual surround environment. Emphasis of comparison is placed on two consumer electronic products: a 5.1 home theater system and a dual-loudspeaker MP3 handset. The effect of loudspeaker spacing on rendering performance is examined. Listening tests are conducted to compare the processing methods in terms of three levels of subjective indices. The results are processed by using the Multi-Analysis Of VAriance (MANOVA) to justify the statistical significance, followed by a multiple regression analysis to correlate the auditory preference with various timbral and spatial attributes.

The main objective of this work is to develop an electronic travel aid to assist the blinds for obstacle identification in their navigation. This navigation assistance for visually impaired (NAVI) system presented in this paper consists of a single board processing system (SBPS), a vision sensor mounted headgear and a pair of stereo earphones. The image environment in front of the blind is captured by the vision sensor. The image is processed by a new real time image processing scheme using fuzzy clustering algorithms. The processed image is mapped onto a specially structured stereo acoustic patterns and transferred to the stereo earphones in the system. Blind individuals were trained with NAVI system and tested for obstacle identification. Suggestions from the blind volunteers regarding pleasantness and discrimination of sound pattern were also incorporated in the prototype. The proposed processing methodology is found to be effective for object identification and for producing stereo sound patterns in the NAVI system.

We propose ‘Ubiquitous Acoustic Spaces,’ where each sound source can emit some address information and automatically enable us to access its related cyber space, using mobile terminals. This will be used for a new type of wireless advertisement for mobile phones using sound media. In order to realize this, we have proposed a nearly lossless audio watermarking technique, utilizing two-channel stereo audio characteristics. This enables contactless asynchronous detection of embedded watermarks through loudspeaker and microphone devices even by using cell phones. We extend the embedding frequency band width and improve the extraction precision. Thus we enable watermark extraction from both stereo channels. Possible embedding data rate can be extended to 80 bps, and almost 100% data can be extracted in case of using PC or PDA devices as terminals. Moreover, we can carry out the watermark extraction even from cell phone loudspeakers or through public telephone networks. From our experiments, almost 100% extraction of 20 bps embedded data could be achieved with these phone devices. In this paper, we describe briefly our newly proposed dual monaural watermark embedding algorithms, and present a prototype system of ubiquitous acoustic spaces using cell phones, based on our experimental results of extraction precision on several capture signal conditions including experiments using cell phones or public telephone networks. Copyright © 2007 Institute of Electrical Engineers of Japan. Published by John Wiley & Sons, Inc.

A wearable Obstacle Stereo Feedback (OSF) System for the Blind people based on 3D space obstacle detection is presented to assist the navigation. The OSF system embedded with a depth sensor to perceive the in-front 3D spatial information in the form of point clouds. We implemented the downsampling Random Sample Consensus (RANSAC) algorithm to process the perceived point cloud, and detect the obstacles in front of the user. Finally, Head-Related Transfer Functions (HRTF) are applied to create the virtual stereo sound which represents the obstacles according to its coordinate in the 3D space. The experiment shows that OSF system can detect the obstacle in the indoor environment effectively and provides a feasible auditory perception to indicate the in-front safety zone for the blind user.

We proposed a novel technology called Multiple Beam-forming Area Sound Enhancement (MUBASE) to pick up only sound in a particular area. We also proposed a technology called Stereophonic Area Sound Reproduction (SASR) to reproduce area sound with newly stereophonic effect. In this paper we present a system that users can feel high realistic sound as if they are in a remote place. The system consists of MUBASE and SASR, and operates in conjunction with images of the remote place.

TouchNoise is a multitouch noise modulation interface designed for musical live performance. It allows the direct and indirect manipulation of sound particles in the stereophonic frequency spectrum. In order to increase TouchNoise's playability we conducted a comprehensive interface revision retaining only its core interaction concept. New interaction techniques and gestures for radial menus, effect range settings, and frequency band effects are introduced. The revision paved the way for a series of new functionalities, such as flocking, flow fields, and MIDI connectivity, making TouchNoise a fully-fledged, powerful interface for creative work with noise. This paper introduces the new TouchNoise interface and functionalities through a discussion of the revision process and derives interaction principles and design recommendations for musical multitouch interfaces in general.

This paper proposes an effective method to use a Distributed Mode Loudspeaker (DML). DMLs are loudspeakers with strong diffusion properties. In this paper, we proposed a sound reproduction system with a DML. The sound field generated by the proposed reproduction system is measured by the index of diffusivity. As a result, it is concluded that the sound field generated by the proposed reproduction system has a higher diffusivity when compared with a conventional reproduction system. The proposed system also demonstrates strong results with respect to the influence of the chamber's acoustic boundary on sound quality.

Slepi in slabovidni ljudje imajo veliko težav pri opravljanju vsakdanjih opravil. Tehnologija lahko pomaga pri reševanju problemov slepih. Še posebej velik problem predstavlja navigacija v nenadzorovanih okoljih, za kar obstaja veliko tehnoloških pripomočkov. Tehnologija je že toliko napredovala, da je velika procesorska moč na voljo ljudem po relativno nizki ceni. Novi algoritmi za rekonstrukcijo okolja omogočajo učinkovito zaznavanje ovir v realnem času. S pomočjo takšne tehnologije se robotika vzpenja, lahko pa jo uporabimo tudi za pomoč slepim. V tem delu predstavljamo sistem, ki na podlagi sekvence slik iz kamere razbere oddaljenost videnih predmetov. To stori s pomočjo direktnega monokularnega algoritma SLAM. Glede na zaznane ovire v vidnem polju kamere s predlagano metodo izračuna najbolj obetavno pot za uporabnika. Nato uporabniku svetuje, kam naj se premakne, da se ta ne zaleti. Uporabnik od sistema prejme zvočno informacijo prek stereo slušalk. V smeri, ki si jo sistem izbere, se oglaša zvok, katerega uporabnik triangulira in mu sledi. Testi so pokazali, da se uporabnik s sistemom v ovire zaleti šest-krat redkeje kot brez sistema, kar nakazuje na velik potencial našega pristopa. Accomplishing many of the every-day activities present a significant challenge to visually impaired people. One of such challenges is navigation in an unsupervised environment, which might be well addressed with technological solutions. The recent technological advances have made computational devices with significant processing power a widely affordable commodity. In parallel, the advances in computer-vision-based systems allow scene reconstruction in real time paving the way for realistic systems for obstacle detection. As the field of robotics is quickly advancing with the help of such technologies, we can use it to help visually impaired people. In this work we propose a computer-vision-based system for navigation of visually-impaired in unsupervised environment. The system is composed of environment analysis sub-system and the navigation sub-system. A direct monocular SLAM is used to estimate a model of the user's environment. A new algorithm for analysis of this model and detection of potential obstacles is proposed. Based on the obstacle detection result, the system selects the most appropriate direction of motion to minimize the probability of collision. This information is provided to the user by a sound-based interface -- a stereo headset. The direction of motion is provided to the user by placing sources of sounds in the virtual three-dimensional space that user perceives through the stereo headset. The system was evaluated in a real environment. The results have shown that the subjects wearing the system were six times less likely to collide with the obstacles in their environment than the subjects not wearing the system. Albeit basic, the experiments indicate a great potential of the proposed system for navigation in realistic environments.

In this paper, an approach has been made to authenticate an audio song in a collection of a similar one. Generating a unique signature with the help of another short length audio signal that will not be directly embedded into the original song but its encoded lower magnitude values will embed into the original song as a prime identity. Encoding hidden audio into lower magnitude values followed by embedding lower magnitude values into specified locations of magnitude values of original song in a predefined way that is not affected by its audible quality carries a secret authenticating message for quickly identifying it from similar signals. A comparative study has been made with similar existing techniques and experimental results are supported with mathematical formula based on Microsoft WAVE (“.wav”) stereo sound file.

Knowledge on how a number of loudspeakers are positioned relative to a listening position can be used to enhance the listening experience. Usually, these loudspeaker positions are estimated using calibration signals, either audible or psycho-acoustically hidden inside the desired audio signal. In this paper, we propose to use the desired audio signal instead. Specifically, we treat the case of estimating the distance between two loudspeakers playing back a stereo music or speech signal. In this connection, we develop a real-time maximum likelihood estimator and demonstrate that it has a variance in the millimetre range in a real environment for even a modest sampling frequency.

Acoustic echo cancellers are used in teleconferencing systems in order to reduce undesired echoes due to coupling between microphones and loudspeakers. Stereophonic systems provide more realistic experience than single-channel systems, since listeners have spatial information that helps to identify the speaker position. Assuming this scenario, a suitable choice for the system parameters becomes essential to improve the audio reproduction quality. Error-driven optimization strategies are usually used to obtain an optimal system configuration but there is no relationship with the quality desired by the user. In this paper, an interactive evolutionary algorithm is adopted for a stereophonic acoustic echo cancellation system in order to meet subjective specifications in the optimization stage. In this way, the optimal system configuration is derived according to a user-driven approach in order to satisfy the quality requirements demanded by users availing such stereophonic systems. Experimental results prove the effectiveness of the proposed interactive architecture according to both objective and subjective measures.

This paper presents two FM receiver architectures and three digital signal processing algorithms for enhancing the output audio quality of FM broadcast receivers. The two receiver architectures differ only in the front-end processing for estimating the carrier-to-noise ratio and noise floors of the stereo audio signals. The shared back-end processing consists of three algorithms to suppress the noise in the audio signal, to detect and cancel noise pulses (static), and to conceal the degrading effects of fast fading, respectively. Together these FM enhancement techniques achieve about 20 to 35 dB improvements in SNR and stereo separation over a wide range of RF signal strength spanning nearly 30 dB. Perceptually, the audio quality improvement is large and obvious when the received FM signal is weak.

We propose RippleTouch, a low resolution haptic interface that is capable of providing haptic stimulation to multiple areas of the body via a single point of contact actuator. Concept is based on the low frequency acoustic wave propagation properties of the human body. By stimulating the body with different amplitude modulated frequencies at a single contact point, we were able to dissipate the wave energy in a particular region of the body, creating a haptic stimulation without direct contact. The RippleTouch system was implemented on a regular chair, in which, four base range speakers were mounted underneath the seat and driven by a simple stereo audio interface. The system was evaluated to investigate the effect of frequency characteristics of the amplitude modulation system. Results demonstrate that we can effectively create haptic sensations at different parts of the body with a single contact point (i.e. chair surface). We believe RippleTouch concept would serve as a scalable solution for providing full-body haptic feedback in variety of situations including entertainment, communication, public spaces and vehicular applications.

The audio delay is becoming an important factor in audio streaming over short-range wireless network. In this study, we propose an efficient two-level delay control method, called frame sequence adaptation and audio sampling frequency compensation, for achieving stable audio delay with a small variation. To prove the effectiveness of our scheme, we implemented and evaluated the scheme on a Bluetooth network. Experimental results show that our scheme can control audio delay robustly and remove phase shift problem in multi-channel stereophonic audio broadcasting as well.

This paper presents a method to control the stereo stage width to provide additional functionality to a stereophonic sound-field recording/reproduction system that uses two microphones and two loudspeakers. The proposed method interpolates or extrapolates the time-frequency representations of the left and right stereo signals according to the desired scaling magnification. A mathematical analysis is introduced that clarifies the scaling ability of the proposed method. Listening tests using a stereo signal consisting of a mixture of two speech signals with different time-lags are performed. The results indicate that the two stereo signals yield almost the same aural impression when the scaling magnification is under two and the following two requirements are satisfied. 1) The time-lags in the stereo signal are scaled strictly before mixing. 2) The time-lags are scaled approximately by the proposed method after mixing. The listening tests also clarified the superiority of the proposed method in terms of the sound quality with respect to conventional methods

A computerized experimental set-up for second derivative Langmuir probe measurements is presented. As a signal generator for the differentiating signal a computer stereo sound card is used. The probe signal is detected using a selective amplifier - the lock-in amplifier combination, whose output is measured by an ADC card inside a computer. Special drivers working under Microsoft Windows98 Second Edition were created in Visual C++ 6.0 to control the operation of the probe system.

Testing multimodal applications with visually impaired children requires specific testing methods and procedures. In this article, we used a SensAble Phantom device to produce haptic feedback together with stereo sound and visual feedback. We also experimented with low-cost haptic devices, such as force-feedback game controllers and tactile-feedback devices. Here we discuss how to conduct usability testing for multimodal applications with visually impaired children

Technology used in Digital TV has the potential to enhance the viewing experience for millions of hard of hearing people. The Clean Audio project commissioned by the Independent Television Commission (ITC), and continued by Ofcom, looks at methods by which the extra information contained in 5.1 surround sound broadcasts may be used to improve the intelligibility and enjoyment of television audio for hard of hearing viewers and shows that audio processing can effectively turn a digital TV set top box into an assistive device to make digital TV more accessible. Listening tests were carried out which showed benefits in clarity and in perceived overall sound quality for hard of hearing participants by altering levels of centre and left and right channels. Further testing has shown average improvements in intelligibility of up to 9.4% by using surround sound equipment with a discrete central loudspeaker compared to stereophonic reproduction.

MPEG's most recent effort to progress the state of the art is the MPEG Surround work item. It provides an efficient method for coding multichannel sound via the transmission of a compressed stereophonic (or even monophonic) audio program plus a low-rate side-information channel. Benefits of this approach include backward compatibility with pervasive stereo playback systems while permitting next-generation players to reconstruct high-quality multichannel sound.

We present an object identification methodology applied in a navigation assistance for visually impaired (NAVI) system. The NAVI has a single board processing system (SBPS), a digital video camera mounted headgear, and a pair of stereo earphones. The captured image from the camera is processed by the SBPS to generate a specially structured stereo sound suitable for vision impaired people in understanding the presence of objects/obstacles in front of them. The image processing stage is designed to identify the objects in the captured image. Edge detection and edge-linking procedures are applied in the processing of image. A concept of object preference is included in the image processing scheme and this concept is realized using a fuzzy-rule base. The blind users are trained with the stereo sound produced by NAVI for achieving a collision-free autonomous navigation.

In the stereo audio DAC here presented, the tradeoff between area-power consumption-SNR/sub out/-dynamic range is optimized for the case of a 96-dB audio system. Using a single-opamp switched-capacitor structure for the reconstruction filter, a hybrid FIR/IIR transfer function allows to reject out-of-band noise. This circuit solution strongly reduces area and power consumption. In a 0.13-/spl mu/m CMOS technology, the stereo DAC achieves a 97-dB dynamic range and a 39-dB SNR/sub out/ with a 0.22-mm/sup 2/ area and 7.25-mW power consumption per channel.

The fast affine projection (FAP) algorithm (Gay and Tavathia, Proceedings of the IEEE International Conference on Acoustic, Speech and Signal Processing, 1995, 3023) is known to outperform the NLMS with a slight increase in complexity, but it involves the fast calculation of the inverse of a covariance matrix of the input data that could undermine the performance of the algorithm. The block subband adaptive algorithm in (Courville and Duhamel, IEEE Trans. Signal Processing 46(9) (1998) 2359) has also illustrated significant improvement in performance over the NLMS and other frequency domain adaptive algorithms. However, it is known that block processing algorithms have lower tracking capabilities than the their sample-by-sample counterparts. In this paper, we present a sample-by-sample version of the algorithm in (Courville and Duhamel, IEEE Trans. Signal Processing 46(9) (1998) 2359) and develop a low complexity implementation of this algorithm. As a sample-by-sample algorithm, it avoids the reduced tracking capability of block algorithms. Because it does not use matrix inversion, it avoids the numerical problems of FAP algorithms. We will show that the new sample-by-sample algorithm approximates the affine projection algorithm and possesses a similar property in reducing coefficient bias that appears in monophonic and stereophonic teleconferencing when the receiving room impulse responses are undermodeled. The new fast sample-by-sample algorithm is extended for stereo acoustic echo cancellation. Simulations of echo cancellations in actual rooms are presented to verify our findings.

We investigate stereophonic acoustic echo cancellation in which solutions for the system can be nonunique and propose the use of selective-tap adaptive filters to address this problem. The main concept is to employ tap selection to optimize jointly for minimum interchannel coherence and maximum L/sub 2/-norm of the subselected tap-input vectors. The exclusive maximum (XM) tap-selection approach is proposed and applied to normalized least-mean squares (NLMS) and recursive least-squares (RLS) algorithm. We propose an approach for solving the nonuniqueness problem employing XM tap selection in combination with a nonlinear preprocessor. Simulation results show a significant improvement in convergence rate compared with existing techniques.

The effect of head rotation on median plane sound source (or image) localization is studied. It is suggested that, at low frequency, the change of interaural time difference (ITD) caused by head rotation supplies information for determining sound source direction in the median plane. Based on the suggestion, the summed sound image localization equations for multiple loudspeakers arranged in the median plane are derived. Especially, for a pair of loudspeakers arranged front-back symmetrically in the median plane, the localization equations are similar to that of stereophonic sound in horizontal plane. A sound image localization experiment was carried out to prove the theoretical analysis. The results of this paper are not only available to virtual spatial auditory, but also supply a quantitative validation of the hypothesis that head rotation is a cue for sound source localization in the median plane at low frequency.

Multichannel audio can immerse a group of listeners in a seamless aural environment. Previously, we proposed a system capable of synthesizing the multiple channels of a virtual multichannel recording from a smaller set of reference recordings. This problem was termed multichannel audio resynthesis and the application was to reduce the excessive transmission requirements of multichannel audio. In this paper, we address the more general problem of multichannel audio synthesis, i.e., how to completely synthesize a multichannel audio recording from a specific stereophonic or monophonic recording, which would significantly enhance the recording's acoustic impression. We approach this problem by extending the model employed for the resynthesis problem. This is accomplished by adapting the resynthesis conversion parameters to the statistical properties of the recording that we wish to enhance. This parameter adaptation is similar to the task adaptation employed in speech recognition, when a specific model is applied to a different environment (speaker, language or channel). One particular approach to this problem is shown here to be quite advantageous toward solving the multichannel audio synthesis problem as well.

Cinema advertising offers a relatively less cluttered environment for advertisers to present their message to a captive audience. However, little is known about its effectiveness in countries such as Hong Kong (a city that is relatively underdeveloped in terms of cinema adspend). Building on the work of Ewing et al. (2001) and Dunnett and Hoek (1996), insights into perceptions of cinema advertising in Hong Kong were obtained from a survey of 150 interviewees. As opposed to previous studies, which utilised delayed recall, this study interviewed audience members immediately after they had viewed a particular movie. Results showed that cinema advertising exposure and recall rates were significantly related to various demographic variables, especially gender and age. Furthermore, the level of recall was found to be correlated with various situational stimuli in the cinema, such as the larger-than-life screen, Dolby stereo sound, the silent environment, comfortable seats and audience members’ expectations to f...

This paper proposes a novel broad-band noise reduction technique in stereophonic situations. The method utilises a combined non-recursive multi-stage sliding subspace projection and adaptive signal enhancement. Simulation results based upon real stereophonic speech contaminated by broad-band noise components show that the proposed method yields a performance improvement in the speech enhancement quality evaluated by both the cepstral distance and segmental gain in SNR objective measurements, as well as by listening, in comparison with the conventional nonlinear spectral subtraction approach.

One of the main difficulties experienced by the blind people is navigation without collision. It is well known that blind individuals are sensitive to sound. This paper presents a real time scheme in providing vision substitution to visually handicapped people. A prototype system had been constructed. It has a laptop computer, a headgear with a digital video camera and a set of stereo earphones interconnected. The images of the environment are captured by digital video camera. The critically designed image processing software processes the image to locate the object and its boundaries. The image processing software is designed to incorporate some of human vision properties using artificial intelligent techniques. The processed image is mapped onto structured stereo sound patterns, so that the blind can understand the environment around him through the set of stereo earphone. The prototype equipment is used to train blind volunteers with the acoustic patterns. Suggestions from the blind volunteers regarding pleasantness and discrimination of sound patterns were also incorporated in the system. The prototype equipment is tested by blinds for their successful indoor collision free motion and navigation within a building.

Binaural Cue Coding (BCC) is a method for multichannel spatial rendering based on one down-mixed audio channel and BCC side information. The BCC side information has a low data rate and it is derived from the multichannel encoder input signal. A natural application of BCC is multichannel audio data rate reduction since only a single down-mixed audio channel needs to be transmitted. An alternative BCC scheme for efficient joint transmission of independent source signals supports flexible spatial rendering at the decoder. This paper (Part I) discusses the most relevant binaural perception phenomena exploited by BCC. Based on that, it presents a psychoacoustically motivated approach for designing a BCC analyzer and synthesizer. This leads to a reference implementation for analysis and synthesis of stereophonic audio signals based on a Cochlear Filter Bank. BCC synthesizer implementations based on the FFT are presented as low-complexity alternatives. A subjective audio quality assessment of these implementations shows the robust performance of BCC for critical speech and audio material. Moreover, the results suggest that the performance given by the reference synthesizer is not significantly compromised when using a low-complexity FFT-based synthesizer. The companion paper (Part II) generalizes BCC analysis and synthesis for multichannel audio and proposes complete BCC schemes including quantization and coding. Part II also describes an alternative BCC scheme with flexible rendering capability at the decoder and proposes several applications for both BCC schemes.

Binaural Cue Coding (BCC) is a method for multichannel spatial rendering based on one down-mixed audio channel and side information. The companion paper (Part I) covers the psychoacoustic fundamentals of this method and outlines principles for the design of BCC schemes. The BCC analysis and synthesis methods of Part I are motivated and presented in the framework of stereophonic audio coding. This paper, Part II, generalizes the basic BCC schemes presented in Part I. It includes BCC for multichannel signals and employs an enhanced set of perceptual spatial cues for BCC synthesis. A scheme for multichannel audio coding is presented. Moreover, a modified scheme is derived that allows flexible rendering of the spatial image at the receiver supporting dynamic control. All aspects of complete BCC encoder and decoder implementations are discussed, such as down-mixing of the input signals, low complexity estimation of the spatial cues, and quantization and coding of the side information. Application examples are given and the performance of the coder implementations are evaluated and discussed based on subjective listening test results.