A new method to determine the frequency response of enclosures using masked tones

Existing methods for equalization of enclosures are traditionally carried out in an empty room due to the annoying effect of the signals employed (pink noise, 1/3 octave tones, etc.) over the audience. Under these conditions, the frequency response and the subsequent equalization are determined from results that do not take into account the presence of people nor any other change that may occur in the room while the musical event takes place. In addition, this kind of equalization cannot be objectively readjusted during the event. In this contribution, a method to determine the frequency response of enclosures in an imperceptible way for the audience is presented. It involves including a test signal within a musical track that, thanks to the behaviour of human hearing, can be masked and successfully recovered using reference microphones. Hence, it is possible to obtain the room transfer function in the presence of public and perform the equalization adjustments while the music track is being played. The results agree completely with traditional methods, the great advantage being the possibility of performing the equalization at any moment during the musical event and in a transparent fashion with respect to the audience.     

Robustness of spatial average equalization: a statistical reverberation model approach

Traditionally, multiple listener room equalization is performed to improve sound quality at all listeners, during audio playback, in a multiple listener environment (e.g., movie theaters, automobiles, etc.). A typical way of doing multiple listener equalization is through spatial averaging, where the room responses are averaged spatially between positions and an inverse equalization filter is found from the spatially averaged result. However, the equalization performance, will be affected if there is a mismatch between the position of the microphones (which are used for measuring the room responses for designing the equalization filter) and the actual center of listener head position (during playback). In this paper, we will present results on the effects of microphone-listener mismatch on spatial average equalization performance. The results indicate that, for the analyzed rectangular configuration, the region of effective equalization depends on (i) the distance of a listener from the source, (ii) the amount of mismatch between the responses, and (iii) the frequency of the audio signal. We also present some convergence analysis to interpret the results.     

Acoustic model for robustness analysis of optimal multipoint room equalization

In this paper, an acoustic model for the robustness analysis of optimal multipoint room equalization is proposed. The optimal multipoint equalization aims to have the optimal performance in a least-squares sense for all measured points. The model can be used for theoretical robustness estimation depending on the critical design parameters such as the number of measurement points, the distance between measurements, or the frequency before applying real equalization system. The analysis results show that it is important to set the appropriate number of measurement points and the distances between measurement points to ensure the enlarged equalization region at a specific frequency.     

A multichannel and multiple position adaptive room response equalizer in warped domain: Real-time implementation and performance evaluation

Room response equalization systems are used for improving the listening experience in cinema theatres, home theatres, car hi-fi systems. In this paper, an adaptive multichannel and multiple position room response equalization system and its real-time implementation are described. An adaptive and accurate estimation of the room responses is provided introducing a normalized least mean square optimization approach with a variable step-size, and taking advantage of an interchannel coherence reduction technique based on the missing fundamental phenomenon. Then, the equalizer is designed in warp frequency domain for improving equalization in the low frequency region, reducing the computational cost of the design procedure, and deriving an algorithm capable of working in real time. Indeed, a real-time implementation of the proposed adaptive equalizer has been obtained on NU-Tech framework and has been used in order to provide a deep objective and subjective evaluation of the equalization system. The results of these evaluations illustrate the effectiveness of the proposed approach, also in comparison with other techniques of the state of the art.     

Equalization filter design for downfiring flat television speakers

Many loudspeakers in newer flat television systems are mounted on the bottom of the television, with their diaphragms facing downward, so as to be hidden inside the TV frame. This kind of loudspeaker installation, called downfiring, induces relatively large reflections from the walls of a room. The increased reflections inevitably change the impulse response of the loudspeakers, which leads to a perceptible distortion in sound quality. In this study, an equalization procedure to resolve the distortion due to downfiring loudspeakers is presented. The change in the early-arriving sound from the room reflections was analyzed, and the specific experimental environment was set up to measure the early reflections for designing a single-channel equalization filter. The inverse technique for the single-input multiple-output (SIMO) system was applied such that both the magnitude and phase responses from downfiring loudspeakers can be equalized at multiple listener positions. We also investigated the possible equalization range in space and time, for which the single-channel filter can work effectively. The performance of the filter designed for the equalization range was then demonstrated by experiments. The experiments were performed using a flat television mock-up installed in a reflective environment. For the performance evaluation of the proposed method, we employed two different measures that can represent the magnitude and phase distortions: the mean squared error of the magnitude and an energy decay curve. The experimental results demonstrate that the single-channel filter can reduce the early reflections from a downfiring loudspeaker by a considerable amount.     

Tunable dual-wavelength erbium-doped fiber ring laser covering both C-band and L-band for high-speed communications

A simple, continuously tunable dual-wavelength erbium-doped fiber ring laser (TDEDFL) structure for applications in high-speed communication systems is proposed and experimentally demonstrated. The dual-wavelength tuning range is 58 nm covering both the C-band and L-band from 1547 to 1605 nm. We can not only obtain a 45% improvement over previously reported tuning ranges, but also tune the wavelength of each lasing output independently. The power equalization of the dual-wavelength outputs is less than 1.5 dB. We obtain extremely stable power variation and wavelength fluctuation at room temperature. Using this fiber laser, a 10-Gb/s data transmission over a 25-km single-mode fiber (SMF) can be made available with a power penalty of 0.5 dB is demonstrated with this laser.     

A study of intersymbol interference and transmission medium instability for an optical fibre system

A simple model is used to evaluate the power penalty accruing from nearest neighbour Intersymbol Interference (ISI) in a digital optical communications system using an integrate- and -dump decision process. The results compare closely with those obtained by Personick in a frequency domain analysis. An analysis of the effects of drift in the transmission bandwidth in the cable and of imperfect equalization indicates a simple design criterion for a fixed equalizer link.     

Spatially extended sound equalization in rectangular rooms.

The results of a theoretical study on global sound equalization in rectangular rooms at low frequencies are presented. The zone where sound equalization can be obtained is a continuous three-dimensional region that occupies almost the complete volume of the room. It is proved that the equalization of broadband signals can be achieved by the simulation of a traveling plane wave using FIR filters. The optimal solution has been calculated following the traditional least-squares approximation, where a modeling delay has been applied to minimize reverberation. An advantage of the method is that the sound field can be estimated with sensors placed in the limits of the equalization zone. As a consequence, a free space for the listeners can be obtained.     

Robustness of multichannel equalization in an acoustic reverberant environment

A theoretical framework is established, for the robustness of multichannel sound equalization in reverberant environments. Using results from statistical room acoustics, a closed-form expression is derived that predicts the degradation in performance of an equalization system as the sound source moves from its nominal position inside the enclosure. The presented analysis also provides means of identifying the performance bounds that can be expected when using such a system in an actual room. Using extensive computer simulations, the effect of physical parameters such as the relative positions of the source and the receivers, as well as effects of different design parameters are investigated. Based on the conditions imposed by these parameters, it is shown that, depending on the array geometry and the exact form of the equalizers, slight performance gains can be expected as the number of receivers is increased.     

Unified a Priori Error Estimate and a Posteriori Error Estimate of CIP-FEM for Elliptic Equations

This paper is devoted to a unified a priori and a posteriori error analysis of CIP-FEM (continuous interior penalty finite element method) for second-order elliptic problems. Compared with the classic a priori error analysis in literature, our technique can easily apply for any type regularity assumption on the exact solution, especially for the case of lower $H^{1+s}$ weak regularity under consideration, where 0 ≤$s$≤ 1/2. Because of the penalty term used in the CIP-FEM, Galerkin orthogonality is lost and Céa Lemma for conforming finite element methods can not be applied immediately when 0≤$s$≤1/2. To overcome this difficulty, our main idea is introducing an auxiliary $C^1$ finite element space in the analysis of the penalty term. The same tool is also utilized in the explicit a posteriori error analysis of CIP-FEM.     

Experimental implementation of a low-frequency global sound equalization method based on free field propagation

An experimental implementation of a global sound equalization method in a rectangular room using active control is described in this paper. The main purpose of the work has been to provide experimental evidence that sound can be equalized in a continuous three-dimensional region, the listening zone, which occupies a considerable part of the complete volume of the room. The equalization method, based on the simulation of a progressive plane wave, was implemented in a room with inner dimensions of 2.70 m × 2.74 m × 2.40 m. With this method, the sound was reproduced by a matrix of 4 × 5 loudspeakers in one of the walls. After traveling through the room, the sound wave was absorbed on the opposite wall, which had a similar arrangement of loudspeakers, by means of active control. A set of 40 digital FIR filters was used to modify the original input signal before it was fed to the loudspeakers, one filter for each transducer. The optimal arrangement of the loudspeakers and the maximum frequency that can be equalized is analyzed theoretically in this paper. The presented experimental results show that sound equalization was possible from 10 Hz to approximately 425 Hz in the listening zone. A flat frequency response with deviations within ±5 decibels from the desired value was achieved. A higher demanding performance with deviations within ±1.5 decibels from a flat frequency response was attained in the interval between 20 Hz and 280 Hz. At the same time, the impulse response was quite well approximated to a delayed delta function in the listening zone. Examples of the spatial distribution of the sound field are also shown.     

Adjustable EDFA gain equalization filter for DWDM channels based on a single LPG excited by flexural acoustic waves

A fast adjustable gain equalization filter for dense wavelength division multiplexing (DWDM) system is reported. The method is based on a single long period fiber grating (LPG) which is excited by means of flexural acoustic waves. The equalization of a typical erbium doped fiber amplifier (EDFA) gain spectrum with a gain flatness of <0.3 dB over a 32 nm bandwidth is demonstrated. The filter adjustment is obtained by choosing different acoustic loads applied to the acousto-optic modulator, which presents a switching time of ～17 μs. A maximum power penalty of 0.84 dB, relatively to the back-to-back signal, was achieved.     

Frequency Downconverting Fiber Grating-Based Beamformers for High-Capacity Phased Arrays

A technique is presented to remove the beat noise limitation in multibeam beam formers using a simple all-optical microwave frequency downconversion technique prior to performing the true-time delay equalization in the optical domain. The frequency conversion concept enables a significant increase in beam-number capac ity to be achieved due to the elimination of beat noise limits, and also effectively removes the power penalty due to chromatic dispersion limitations of the chirped grating units in the beamformer. The Bragg grating requirements for the frequency converting beamforming network are analyzed and show that tanh-profile apodized gratings can meet the isolation, reflectivity, and narrow bandwidth requirements. For an X-band phased array, more than a twofold increase in beam capacity is shown through the use of the frequency conversion technique with the grating-based beamformer, and the resulting beamformer has the minimum number of optical interconnects with true-time delay operation.     

Frequency Downconverting Fiber Grating-Based Beamformers for High-Capacity Phased Arrays

A technique is presented to remove the beat noise limitation in multibeam beam formers using a simple all-optical microwave frequency downconversion technique prior to performing the true-time delay equalization in the optical domain. The frequency conversion concept enables a significant increase in beam-number capac ity to be achieved due to the elimination of beat noise limits, and also effectively removes the power penalty due to chromatic dispersion limitations of the chirped grating units in the beamformer. The Bragg grating requirements for the frequency converting beamforming network are analyzed and show that tanh-profile apodized gratings can meet the isolation, reflectivity, and narrow bandwidth requirements. For an X-band phased array, more than a twofold increase in beam capacity is shown through the use of the frequency conversion technique with the grating-based beamformer, and the resulting beamformer has the minimum number of optical interconnects with true-time delay operation.     

Large Signal Analysis of FM-AM Conversion in Dispersive Optical Fibers for PCM Systems Including Second Order Dispersion

By using large signal analysis for dispersive optical fiber, the FM-AM conversion with respect to binary intensity modulated PCM systems including second order dispersion term is discussed. The modified expression for power penalty has been derived and its impact on laser linewidth and bit rate has been investigated. For power penalty less than 0.5 dB, the plots between bit rate and transmission distance are plotted. It is seen that the transmission distance increases with decrease in linewidth over significant bit rates. The transmission distance with first order dispersion term for 300 MHz linewidth is approximately 800km. With proper first order dispersion compensation, i.e., with second order dispersion only, the transmission distance can be enhanced to 10 8 km for this linewidth. The linewidth requirements for systems with different bit rates and transmission distances are also calculated and discussed. Further, it is seen that by including the second-order dispersion term, the bit rate and transmission distance decreases. For higher linewidths, this decrease in bit rate and transmission distance is very less and vice versa. For 300 MHz linewidth, the decrease in transmission distance is just 30 km, and for 30 MHz linewidth, the decrease is approximately 600 km over significant bit rates.     

Toward broadband electroacoustic resonators through optimized feedback control strategies

This paper presents a methodology for the design of broadband electroacoustic resonators for low-frequency room equalization. An electroacoustic resonator denotes a loudspeaker used as a membrane resonator, the acoustic impedance of which can be modified through proportional feedback control, to match a target impedance. However, such impedance matching only occurs over a limited bandwidth around resonance, which can limit its use for the low-frequency equalization of rooms, requiring an effective control at least up to the Schroeder frequency. Previous experiments have shown that impedance matching can be achieved over a range of a few octaves using a simple proportional control law. But there is still a limit to the feedback gain, beyond which the feedback-controlled loudspeaker becomes non-dissipative. This paper evaluates the benefits of using PID control and phase compensation techniques to improve the overall performance of the electroacoustic resonator. More specifically, it is shown that some adverse effects due to high-order dynamics in the moving-coil transducer can be mitigated. The corresponding control settings are also identified with equivalent electroacoustic resonator parameters, allowing a straightforward design of the controller. Experimental results using PID control and phase compensation are finally compared in terms of sound absorption performances. As a conclusion the overall performances of electroacoustic resonators for damping the modal resonances inside a duct are presented, along with general discussions on practical implementation and the extension to actual room modes damping.     

Multistage gain-flattened hybrid optical amplifier at reduced wavelength spacing

In this paper, two stage hybrid optical amplifier (HOA) composed of a single erbium doped fiber amplifier and Raman amplifier is proposed for dense wavelength division multiplexed (DWDM) system and investigate the impact of reduced channel spacing. The performance has been evaluated in the term of gain, gain flatness and noise figure. Also, using gain equalization technique, hybrid optical amplifier that has a gain flatness of 3 dB, and a noise figure of less than 7.4 dB is observed.     

基于平台直方图的红外图像自适应增强算法

针对红外图像的特点, 提出了一种基于平台直方图均衡化的自适应红外图像增强算法. 该算法通过自适应地选择平台阈值, 对红外图像进行增强处理, 克服了采用一般直方图均衡化增强红外图像的缺点, 同时算法的运算量远远小于其他平台直方图均衡化算法, 便于实时实现. 理论分析和仿真结果均表明, 该算法对红外图像具有很好的增强效果, 可较好的抑制背景的增强, 突出目标.     

Low power penalty tunable slow light using vertical-cavity surface-emitting laser amplifier

A tunable slow light of 2.5-Gb/s pseudo-random binary sequence signal using a 1550-nm vertical-cavity surface-emitting laser (VCSEL) is experimentally demonstrated. The influences of the bias current and the gain saturation on the slow light are investigated. With bias current increasing, tunable optical group delay up to 98 ps is obtained at room temperature. Demonstration of the time delay between 16 and 24 ps by signal intensity change is reported. Under an appropriate bias current, by tuning the input signal to track the peak gain wavelength of the VCSEL, slow light of a power penalty as low as 1 dB is achieved. With such a low power penalty, the VCSEL has a great potential application as a compact optical buffer.     

A momentum exchange-based immersed boundary-lattice Boltzmann method for simulating incompressible viscous flows

A momentum exchange-based immersed boundary-lattice Boltzmann method is presented in this Letter for simulating incompressible viscous flows. This method combines the good features of the lattice Boltzmann method (LBM) and the immersed boundary method (IBM) by using two unrelated computational meshes, an Eulerian mesh for the flow domain and a Lagrangian mesh for the solid boundaries in the flow. In this method, the non-slip boundary condition is enforced by introducing a forcing term into the lattice Boltzmann equation (LBE). Unlike the conventional IBM using the penalty method with a user-defined parameter or the direct forcing scheme based on the Navier–Stokes (NS) equations, the forcing term is simply calculated by the momentum exchange of the boundary particle density distribution functions, which are interpolated by the Lagrangian polynomials from the underlying Eulerian mesh. Numerical examples show that the present method can provide very accurate numerical results.