Showing papers on "Microphone published in 2004"

Plane-wave decomposition of the sound field on a sphere by spherical convolution

Abstract: Spherical microphone arrays have been recently studied for sound analysis and sound recordings, which have the advantage of spherical symmetry facilitating three-dimensional analysis. This paper complements the recent microphone array design studies by presenting a theoretical analysis of plane-wave decomposition given the sound pressure on a sphere. The analysis uses the spherical Fourier transform and the spherical convolution, where it is shown that the amplitudes of the incident plane waves can be calculated as a spherical convolution between the pressure on the sphere and another function which depends on frequency and the sphere radius. The spatial resolution of plane-wave decomposition given limited bandwidth in the spherical Fourier domain is formulated, and ways to improve the computation efficiency of plane-wave decomposition are introduced. The paper concludes with a simulation example of plane-wave decomposition.

Audio Signal Processing for Next-Generation Multimedia Communication Systems

Abstract: Preface. Contributing Authors. 1: Introduction Yiteng (Arden) Huang, J. Benesty. 1. Multimedia Communications. 2. Challenges and Opportunities. 3. Organization of the Book. I: Speech Acquisition and Enhancement. 2: Differential Microphone Arrays G.W. Elko. 1. Introduction. 2. Differential Microphone Arrays. 3. Array Directional Gain. 4. Optimal Arrays for Isotropic Fields. 5. Design Examples. 6. Sensitivity to Microphone Mismatch and Noise. 7. Conclusions. 3: Spherical Microphone Arrays for 3D Sound Recording J. Meyer, G.W. Elko. 1. Introduction. 2. Fundamental Concept. 3. The Eigenbeamformer. 4. Modal-Beamformer. 5. Robustness Measure. 6. Beampattern Design. 7. Measurements. 8. Summary. 9. Appendix A. 4: Subband Noise Reduction Methods for Speech Enhancement E.J. Diethorn. 1. Introduction. 2. Wiener Filtering. 3. Speech Enhancement by Short-Time Spectral Modification. 4. Averaging Techniques for Envelope Estimation. 5. Example Implementation. 6. Conclusion. II: Acoustic Echo Cancellation. 5: Adaptive Algorithms for MIMO Acoustic Echo Cancellation J. Benesty, T. Gansler, Yiteng (Arden) Huang, M. Rupp. 1. Introduction. 2. Normal Equations and Identification of a MIMO System. 3. The Classical and Factorized Multichannel RLS. 4. The Multichannel Fast RLS. 5. TheMultichannel LMS Algorithm. 6. The Multichannel APA. 7. The Multichannel Exponentiated Gradient Algorithm. 8. The Multichannel Frequency-domain Adaptive Algorithm. 9. Conclusions. 6: Double-talk Detectors for Acoustic Echo Cancellers T. Gansler, J. Benesty. 1. Introduction. 2. Basics of AEC and DTD. 3. Double-talk Detection Algorithms. 4. Comparison of DTDs by Means of the ROC. 5. Discussion. 7: The WinEC: A Real-Time Hands-Free Stereo Communication System T. Gansler, V. Fischer, E.J. Diethorn, J. Benesty. 1. Introduction. 2. System Description. 3. Algorithms of the Echo Canceller Module. 4. Residual Echo and Noise Suppression. 5. Simulations. 6. Real-Time Tests with Different Modes of Operation. 7. Discussion. III: Sound Source Tracking and Separation. 8: Time Delay Estimation Jingdong Chen, Yiteng (Arden) Huang, J. Benesty. 1. Introduction. 2. Signal Models. 3. Generalized Cross-Correlation Method. 4. The Multichannel Cross-Correlation Algorithm. 5. Adaptive Eigenvalue Decomposition Algorithm. 6. Adaptive Multichannel Time Delay Estimation. 7. Experiments. 8. Conclusions. 9: Source Localization Yiteng (Arden) Huang, J. Benesty, G.W. Elko. 1. Introduction. 2. Source Localization Problem. 3. Measurement Model and Cramer-Rao lower Bound for Source Localization. 4. Maximum Liklihood Estimator. 5. Least Squares Estimate. 6. Example

Method and apparatus for multi-sensory speech enhancement

Abstract: A method and system use an alternative sensor signal received from a sensor other than an air conduction microphone to estimate a clean speech value. The estimation uses either the alternative sensor signal alone, or in conjunction with the air conduction microphone signal. The clean speech value is estimated without using a model trained from noisy training data collected from an air conduction microphone. Under one embodiment, correction vectors are added to a vector formed from the alternative sensor signal in order to form a filter, which is applied to the air conductive microphone signal to produce the clean speech estimate. In other embodiments, the pitch of a speech signal is determined from the alternative sensor signal and is used to decompose an air conduction microphone signal. The decomposed signal is then used to determine a clean signal estimate.

Time-delay estimation via linear interpolation and cross correlation

Abstract: Time-delay estimation (TDE), which aims at measuring the relative time difference of arrival (TDOA) between different channels is a fundamental approach for identifying, localizing, and tracking radiating sources Recently, there has been a growing interest in the use of TDE based locator for applications such as automatic camera steering in a room conferencing environment where microphone sensors receive not only the direct-path signal, but also attenuated and delayed replicas of the source signal due to reflections from boundaries and objects in the room This multipath propagation effect introduces echoes and spectral distortions into the observation signal, termed as reverberation, which severely deteriorates a TDE algorithm in its performance This paper deals with the TDE problem with emphasis on combating reverberation using multiple microphone sensors The multichannel cross correlation coefficient (MCCC) is rederived here, in a new way, to connect it to the well-known linear interpolation technique Some interesting properties and bounds of the MCCC are discussed and a recursive algorithm is introduced so that the MCCC can be estimated and updated efficiently when new data snapshots are available We then apply the MCCC to the TDE problem The resulting new algorithm can be treated as a natural generalization of the generalized cross correlation (GCC) TDE method to the multichannel case It is shown that this new algorithm can take advantage of the redundancy provided by multiple microphone sensors to improve TDE against both reverberation and noise Experiments confirm that the relative time-delay estimation accuracy increases with the number of sensors

Multi-speaker audio system and automatic control method

Abstract: A sound produced at the location of a listener is captured by a microphone in each of a plurality of speaker devices. A sever apparatus receives an audio signal of the captured sound from all speaker devices, and calculates a distance difference between the distance of the location of the listener to the speaker device closest to the listener and the distance of the listener to each of the plurality of speaker devices. When one of the speaker devices emits a sound, the server apparatus receives an audio signal of the sound captured by and transmitted from each of the other speaker devices. The server apparatus calculates a speaker-to-speaker distance between the speaker device that has emitted the sound and each of the other speaker devices. The server apparatus calculates a layout configuration of the plurality of speaker devices based on the distance difference and the speaker-to-speaker distance.

Challenges and Recent Developments in Hearing Aids: Part I. Speech Understanding in Noise, Microphone Technologies and Noise Reduction Algorithms

Abstract: This review discusses the challenges in hearing aid design and fitting and the recent developments in advanced signal processing technologies to meet these challenges. The first part of the review discusses the basic concepts and the building blocks of digital signal processing algorithms, namely, the signal detection and analysis unit, the decision rules, and the time constants involved in the execution of the decision. In addition, mechanisms and the differences in the implementation of various strategies used to reduce the negative effects of noise are discussed. These technologies include the microphone technologies that take advantage of the spatial differences between speech and noise and the noise reduction algorithms that take advantage of the spectral difference and temporal separation between speech and noise. The specific technologies discussed in this paper include first-order directional microphones, adaptive directional microphones, second-order directional microphones, microphone matching algorithms, array microphones, multichannel adaptive noise reduction algorithms, and synchrony detection noise reduction algorithms. Verification data for these technologies, if available, are also summarized.

Augmented reality audio for mobile and wearable appliances

Abstract: The concept of augmented reality audio characterizes techniques where a real sound environment is extended with virtual auditory environments and communications scenarios. A framework is introduced for mobile augmented reality audio (MARA) based on a specific headset configuration where binaural microphone elements are integrated into stereo earphones. When microphone signals are routed directly to the earphones, a user is exposed to a pseudoacoustic representation of the real environment. Virtual sound events are then mixed with microphone signals to produce a hybrid, an augmented reality audio representation, for the user. An overview of related technology, literature, and application scenarios is provided. Listening test results with a prototype system show that the proposed system has interesting properties. For example, in some cases listeners found it very difficult to determine which sound sources in an augmented reality audio representation are real and which are virtual.

Speech enhancement based on the general transfer function GSC and postfiltering

Abstract: In speech enhancement applications microphone array postfiltering allows additional reduction of noise components at a beamformer output. Among microphone array structures the recently proposed general transfer function generalized sidelobe canceller (TF-GSC) has shown impressive noise reduction abilities in a directional noise field, while still maintaining low speech distortion. However, in a diffused noise field less significant noise reduction is obtainable. The performance is even further degraded when the noise signal is nonstationary. In this contribution we propose three postfiltering methods for improving the performance of microphone arrays. Two of which are based on single-channel speech enhancers and making use of recently proposed algorithms concatenated to the beamformer output. The third is a multichannel speech enhancer which exploits noise-only components constructed within the TF-GSC structure. This work concentrates on the assessment of the proposed postfiltering structures. An extensive experimental study, which consists of both objective and subjective evaluation in various noise fields, demonstrates the advantage of the multichannel postfiltering compared to the single-channel techniques.

MP3 player having a wireless earphone communication with a mobile

Abstract: The audio player of the present invention comprises: an audio playing module to decode audio compressed files, produce an audio signal transmitted to an earphone, and receive a voice signal from a microphone; and an attachment mechanism to combine a wireless earphone with the audio playing module, to accomplish an electrical connecting interface therebetween. Therefore, the audio playing module operates the wireless earphone via the electrical connecting interface and establishes two-way audio communication with a mobile phone. Another embodiment of this invention provides an audio player having a wireless transceiver, which can establish a wireless connectivity with a mobile phone and allow two-way audio communication, enabling the audio player to communicate with the mobile phone at any time in response to a ring indication received from the mobile phone.

Enhanced sound localization

Abstract: A new approach to sound localization, known as enhanced sound localization, is introduced, offering two major benefits over state-of-the-art algorithms. First, higher localization accuracy can be achieved compared to existing methods. Second, an estimate of the source orientation is obtained jointly, as a consequence of the proposed sound localization technique. The orientation estimates and improved localizations are a result of explicitly modeling the various factors that affect a microphone's level of access to different spatial positions and orientations in an acoustic environment. Three primary factors are accounted for, namely the source directivity, microphone directivity, and source-microphone distances. Using this model of the acoustic environment, several different enhanced sound localization algorithms are derived. Experiments are carried out in a real environment whose reverberation time is 0.1 seconds, with the average microphone SNR ranging between 10-20 dB. Using a 24-element microphone array, a weighted version of the SRP-PHAT algorithm is found to give an average localization error of 13.7 cm with 3.7% anomalies, compared to 14.7 cm and 7.8% anomalies with the standard SRP-PHAT technique.

Phase-based dual-microphone robust speech enhancement

Abstract: A dual-microphone speech-signal enhancement algorithm, utilizing phase-error based filters that depend only on the phase of the signals, is proposed. This algorithm involves obtaining time-varying, or alternatively, time-frequency (TF), phase-error filters based on prior knowledge regarding the time difference of arrival (TDOA) of the speech source of interest and the phases of the signals recorded by the microphones. It is shown that by masking the TF representation of the speech signals, the noise components are distorted beyond recognition while the speech source of interest maintains its perceptual quality. This is supported by digit recognition experiments which show a substantial recognition accuracy rate improvement over prior multimicrophone speech enhancement algorithms. For example, for a case with two speakers with a 0.1 s reverberation time, the phase-error based technique results in a 28.9% recognition rate gain over the single channel noisy signal, a gain of 22.0% over superdirective beamforming, and a gain of 8.5% over postfiltering.

Robust speech interface based on audio and video information fusion for humanoid HRP-2

Abstract: For cooperative work of robots and humans in the real world, a communicative function based on speech is indispensable for robots. To realize such a function in a noisy real environment, it is essential that robots be able to extract target speech spoken by humans from a mixture of sounds by their own resources. We have developed a method of detecting and extracting speech events based on the fusion of audio and video information. In this method, audio information (sound localization using a microphone array) and video information (human tracking using a camera) are fused by a Bayesian network to enable the detection of speech events. The information of detected speech events is then utilized in sound separation using adaptive beam forming. In this paper, some basic investigations for applying the above system to the humanoid robot HRP-2 are reported. Input devices, namely a microphone array and a camera, were mounted on the head of HRP-2, and acoustic characteristics for sound localization/separation performance were investigated. Also, the human tracking system was improved so that it can be used in a dynamic situation. Finally, overall performance of the system was tested via off-line experiments.

High-sensitivity piezoelectric microphones based on stacked cellular polymer films (L)

Abstract: Improvements of the sensitivity of piezoelectric microphones based on charged cellular polymer films are reported. The improvements are achieved by (1) an increase of the piezoelectric d33-coefficient of the cellular polypropylene films by pressure expansion and (2) stacking of the films. Microphones consisting of a single film of such material have sensitivities of about 2 mV/Pa at 1 kHz, independent of size, while for a microphone with five stacked films a sensitivity of 10.5 mV/Pa was measured. The equivalent noise level is about 37 dB(A) for the single-film transducer and 26 dB(A) for the stacked version. Advantages of these new piezoelectric transducers include their simple design, low cost, and small weight, as well as a large range of shapes and sizes possible.

Single microphone source separation using high resolution signal reconstruction

Abstract: We present a method for separating two speakers from a single microphone channel. The method exploits the fine structure of male and female speech and relies on a strong high frequency resolution model for the source signals. The algorithm is able to identify the correct combination of male and female speech that best explains an observation and is able to reconstruct the component signals, relying on prior knowledge to 'fill in' regions that are masked by the other speaker. The two speaker single microphone source separation problem is one of the most challenging source separation scenarios and few quantitative results have been reported in the literature. We provide a test set based on the Aurora 2 data set and report performance numbers on a portion of this set. We achieve results of 6.59 dB average increase in SNR for female speakers and 5.51 dB for male speakers.

Set-up method for array-type sound system

Abstract: There is disclosed a method for setting up a Sound Projector such that it is suitable for a variety of functions, including surround sound. The method allows a semi-automatic or automatic set-up to be accomplished whereby the Sound Projector emits test signals and these are received by one or more microphones in order to detect the position and angles of the major reflecting surfaces in the room. In a preferred embodiment, the room is scanned by a moving directional sound beam and the first reflection of said sound beam is detected at a microphone in order to determine the distance of the reflective surfaces from the Sound Projector for all or most possible angels of sound beams.

Predicting hearing aid microphone preference in everyday listening.

Abstract: Seventeen hearing-impaired adults were fit with omnidirectional/directional hearing aids, which they wore during a four-week trial. For each listening situation encountered in daily living during a total of seven days, participants selected the preferred microphone mode and described the listening situation in terms of five environmental variables, using a paper and pencil form. Results indicated that hearing-impaired adults typically spend the majority of their active listening time in situations with background noise present and surrounding the listener, and the signal source located in front and relatively near. Microphone preferences were fairly evenly distributed across listening situations but differed depending on the characteristics of the listening environment. The omnidirectional mode tended to be preferred in relatively quiet listening situations or, in the presence of background noise, when the signal source was relatively far away. The directional mode tended to be preferred when background noise was present and the signal source was located in front of and relatively near the listener. Results suggest that knowing only signal location and distance and whether background noise is present or absent, omnidirectional/directional hearing aids can be set in the preferred mode in most everyday listening situations. These findings have relevance for counseling patients when to set manually switchable omnidirectional/directional hearing aids in each microphone mode, as well as for the development of automatic algorithms for selecting omnidirectional versus directional microphone processing.

Differential Microphone Arrays

Abstract: Noise and reverberation can seriously degrade both microphone reception and loudspeaker transmission of audio signals in telecommunication systems. Directional loudspeakers and microphone arrays can be effective in combating these problems. This chapter covers the design and implementation of differential arrays that are by definition small compared to the acoustic wavelength. Differential arrays are therefore superdirectional arrays since their directivity is higher than that of a uniformly summed array with the same geometry. Aside from their small size, another beneficial feature of differential arrays is the inherent independence of their directional response as a function of frequency. Derivations are included for several optimal differential arrays that may be useful for teleconferencing and speech pickup in noisy and reverberant environments. Expressions and design details covering optimal multiple-order differential arrays are given. The results shown in this chapter should be useful in designing and selecting directional microphones for a variety of applications.

Tracking of multiple moving speakers with multiple microphone arrays

Abstract: In this paper we use multiple microphone arrays to fuse the location estimate from each microphone array which yields an improved estimate of the positions and velocities of multiple, simultaneously active, moving speakers based on time delay of arrivals (TDOAs). Our approach: 1) incorporates kinematic information of moving speakers by using an interacting multiple model (IMM) estimator for each speaker in order to constrain the evolution of the location measurements; 2) fuses the location estimates of the same speaker from multiple microphone arrays for better acoustical coverage of the sensed environment, and 3) directly accounts for the measurement origin uncertainty, i.e., which measurement comes from which speaker by using the probabilistic data association (PDA) technique with the IMM estimator. We demonstrate that a network of arrays combined with the estimation technique widely used in multisensor multitarget tracking area provides a consistent and coherent way to reduce the uncertainty and ambiguity of measurements. The effectiveness of our approach is illustrated by extensive simulation study on tracking a single moving speaker and two closely-spaced speakers with a crossing segment in their trajectories.

A two-step noise reduction technique

Abstract: The paper addresses the problem of single microphone speech enhancement in noisy environments Common short-time noise reduction techniques proposed in the art are expressed as a spectral gain depending on the a priori SNR In the well-known decision-directed approach, the a priori SNR depends on the speech spectrum estimation in the previous frame As a consequence, the gain function matches the previous frame rather than the current one which degrades the noise reduction performance We propose a new method, called the two-step noise reduction (TSNR) technique, which solves this problem while maintaining the benefits of the decision-directed approach This method is analyzed and results in voice communication and speech recognition contexts are given

Performance of directional microphones for hearing aids: real-world versus simulation.

Abstract: The purpose of this study was to assess the accuracy of clinical and laboratory measures of directional microphone benefit. Three methods of simulating a noisy restaurant listening situation ([1] a multimicrophone/multiloudspeaker simulation, the R-SPACE, [2] a single noise source behind the listener, and [3] a single noise source above the listener) were evaluated and compared to the "live" condition. Performance with three directional microphone systems differing in polar pattern (omnidirectional, supercardioid, and hypercardioid array) and directivity indices (0.34, 4.20, and 7.71) was assessed using a modified version of the Hearing in Noise Test (HINT). The evaluation revealed that the three microphones could be ordered with regard to the benefit obtained using any of the simulation techniques. However, the absolute performance obtained with each microphone type differed among simulations. Only the R-SPACE simulation yielded accurate estimates of the absolute performance of all three microphones in the live condition. Performance in the R-SPACE condition was not significantly different from performance in the "live restaurant" condition. Neither of the single noise source simulations provided accurate predictions of real-world (live) performance for all three microphones.

Measurements of directional properties of reverberant sound fields in rooms using a spherical microphone array

Abstract: The directional variation of sound at a point has been studied in three rooms, using a measurement system described previously [J. Acoust. Soc. Am. 112, 1980–1991 (2002)]. The system uses a pair of 32-element spherical microphone arrays to obtain directional impulse responses in each of 60 steering directions, with an angular resolution of 28°, covering all directions in the whole solid angle. Together, the array measurements span the frequency range from 300 to 3300 Hz. The angular distribution of incident sound energy is visualized on a three-dimensional plot, and quantified by computing the directional diffusion and the directional peak-to-average level difference (“anisotropy index”) of the sound field. The small-to-medium-sized rooms had reverberation times of 360, 400, and 600 ms. Measurements were made for several source and receiver locations in each, and were analyzed over several time ranges (full decay time of room, late time decay, 2-ms windows throughout the decay). All measured sound fields ...

Personal multi-information recorder

Abstract: A method for recording information on a device worn on a person includes capturing video information, capturing audio information and receiving a signal from a remote unit. The method further includes determining from the signal that the remote unit is a minimum distance from the device and in response commencing recording of the video information and the audio information in storage. In one alternative, the video information is captured via a video camera and audio information is captured via a microphone. In another alternative, the remote unit is installed in the person's vehicle.

Motion-Tracked Binaural Sound

Abstract: A new method is presented for capturing, recording, and reproducing spatial sound that provides a vivid sense of realism. The method generalizes binaural recording, preserving the information needed for dynamic head-motion cues. These dynamic cues greatly reduce the need for customization to the listener. During either capture or recording, the sound field in the vicinity of the head is sampled with a microphone array. During reproduction, a head tracker is used to determine the microphones that are closest to the positions of the listener's ears. Interpolation procedures are used to produce the headphone signals. The properties of different methods for interpolating the microphone signals are presented and analyzed.

Method and system for communication enhancement in a noisy environment

Abstract: The invention is directed to a method comprising receiving input signals emanating from at least two microphone arrays each comprising at least two microphones, processing the input signals of each microphone array by a beamformer to determine temporal and spatial information about the input signals of each microphone array.

Speech perception in noise: directional microphones versus frequency modulation (FM) systems.

Abstract: The major consequence of sensorineural hearing loss (SNHL) is communicative difficulty, especially with the addition of noise and/or reverberation. The purpose of this investigation was to compare two types of technologies that have been shown to improve the speech-perception performance of individuals with SNHL: directional microphones and frequency modulation (FM) systems. Forty-six adult subjects with slight to severe SNHL served as subjects. Speech perception was assessed using the Hearing in Noise Test (HINT) with correlated diffuse noise under five different listening conditions. Results revealed that speech perception was significantly better with the use of the FM system over that of any of the hearing aid conditions, even with the use of the directional microphone. Additionally, speech perception was significantly better with the use of two hearing aids used in conjunction with two FM receivers rather than with just one FM receiver. Directional microphone performance was significantly better than omnidirectional microphone performance. All aided listening conditions were significantly better than the unaided listening condition.

Determination of Absolute Contributions of Aircraft Noise Components using Fly-Over Array Measurements

Abstract: †For wind tunnel array measurements, the source power integration technique has proved to be a valuable technique to determine absolute source levels. This paper describes the extension of the source power integration technique to moving sources, and the application to fly-over array measurements on landing aircraft at Amsterdam Airport Schiphol. The technique is applied in combination with a modified version of the Delay-and-Sum beamforming technique, which includes microphone- and frequency-dependent weight factors. These weight factors are used to correct for microphone spatial density and to account for the effects of coherence loss. This beamforming technique works well in combination with the array design, which consists of a number of concentric rings of microphones, with increasing density towards the center. In this paper, it is demonstrated that the extended source power integration technique is able to determine absolute levels from fly-over array measurements, when it is used in combination with the special beamforming technique and the Schiphol array design. Thus, absolute quantification of difference source regions on an aircraft is feasible. Microphone auto-correlations must be included in the beamforming process in order to obtain correct integrated levels, although acoustic images look better when beamforming is done without auto-correlations. The source power integration technique is applied to a Boeing 737-400, and to an Airbus A340.

Battery charging device for a mobile apparatus using an ear-microphone jack

Abstract: A battery charging device using an ear-microphone jack of a mobile apparatus. The battery charging device comprises a battery; an ear-microphone socket having a microphone contact, an earphone contact, and a common ground contact; a modem chip having a microphone input port connected to the microphone contact or an internal microphone and an earphone output port connected to the earphone contact or an internal speaker; and a cutting-off element connected between a power supply contact and the battery, for providing a charging voltage provided via the power supply contact to the battery and preventing a current from flowing backward from the battery to the modem chip. A power supply device comprises a plug having terminals connectable to contacts of the ear-microphone socket; and a power source connector for connecting a power supply source to the plug.

Sound canceling systems and methods

Abstract: A system for sound cancellation includes a source microphone for detecting sound and a speaker for broadcasting a canceling sound with respect to a cancellation location. A computational module is in communication with the source microphone and the speaker. The computational module is configured to receive a signal from the source microphone, identify a cancellation signal using a predetermined adaptive filtering function responsive to acoustics of the cancellation location, and transmit a cancellation signal to the speaker.

Adjustable speaker systems and methods

Abstract: Systems and methods for optimizing speaker performance. The system (30) includes a self-contained speaker unit (32) that includes a speaker (56), an amplifier (54) coupled to the speaker (56) , and a processor (50) coupled to the amplifier (54). The processor (50) receives a first sound signal from a receiver (36) and a second sound signal from a microphone (34) , processes the first sound signal based on a plurality of parameters, outputs the processed sound signal to the speaker (56) , and generates a video signal based on the second sound signal. A wireless remote control (42) allows a user to manipulate the parameters. The processor (50) generates a test sound signal and outputs it to the receiver (36) . The receiver (36) processes the test sound signal and returns it to the processor (50) for output through the speaker (56). The video signal includes a graphical user interface having a frequency response graph of the second sound signal and an eight band equalizer.