Robust sound localization in 0.18 /spl mu/m CMOS

This paper presents a hardware implementation of a sound localization algorithm that localizes a single sound source by using the information gathered by two separated microphones. This is achieved through estimating the time delay of arrival (TDOA) of sound at the two microphones. We have used a TDOA algorithm known as the "phase transform" to minimize the effects of reverberations and noise from the environment. Simplifications to the chosen TDOA algorithm were made in order to replace complex operations, such as the cosine function, with less expensive ones, such as iterative additions. The custom digital signal processor implementing this algorithm was designed in a 0.18-/spl mu/m CMOS process and tested successfully. The test chip is capable of localizing the direction of a sound source within 2.2/spl deg/ of accuracy, utilizing approximately 30 mW of power and 6.25 mm/sup 2/ of silicon area.     

话筒在电视制作中的选择和使用

电视节目制作中声音质量的好坏，直接影响电视节目的质量。要保证电视节目声音的质量。必须正确地选择和使用话筒。讨论常用话筒的选择、使用以及影响声音质量和解决方法等问题。     

Exploiting bounds optimization for the semi-formal verification of analog circuits

This paper proposes a semi-formal methodology for modeling and verification of analog circuits behavioral properties using multivariate optimization techniques. Analog circuit differential models are automatically extracted and their qualitative behavior is computed for interval-valued parameters, inputs and initial conditions. The method has the advantage of guaranteeing the rough enclosure of any possible dynamical behavior of analog circuits. The circuit behavioral properties are then verified on the generated transient response bounds. Experimental results show that the resulting state variable envelopes can be effectively employed for a sound verification of analog circuit properties, in an acceptable run-time.     

基于MSP430的声音定位系统的设计

声源定位在当今日益智能化的社会中有着诸多方面的应用,特别是在智能控制,军事等领域。文中设计的系统以MSP430F449单片机为控制核心,步进电机驱动的小车作为移动声源载体,设计两个间距固定的麦克风作为接收设备,测量声源发声到麦克风接收到声音的时间差数据,声源通过无线模块接收到数据从而计算出声源的坐标,然后控制声源小车运行到指定坐标实现定位控制。经测试,本系统能精确的控制小车到达指定坐标点,系统工作稳定。     

An investigation of time delay estimation in room acoustic environment using magnitude ratio

This paper investigates the relation between the nonstationary sound source and the frequency domain magnitude ratio of two microphones based on short-term frequency analysis. The fluctuation level of nonstationary sound sources is modeled by the exponent of polynomials from the concept of moving pole model. According to this model, the sufficient condition for utilizing the fluctuation level and magnitude ratio to estimate the time delay between two microphones is suggested. Simulation results are presented to show the performance of the suggested method.     

Narrowband auscultatory blood pressure measurement

Auscultatory blood pressure measurement uses the presence and absence of acoustic pulses generated by an artery (i.e., Korotkoff sound), detected with a stethoscope or a sensitive microphone, to noninvasively estimate systolic and diastolic pressures. Unfortunately, in high noise situations, such as ambulatory environments or when the patient moves moderately, the current auscultatory blood pressure method is unreliable, if at all possible. Empirical evidence suggests that the pulse beneath an artery occlusion travels relatively slow compared with the speed of sound. By placing two microphones along the bicep muscle near the brachial artery under the occlusion cuff, a similar blood pressure pulse appears in the two microphones with a relative time delay. The acoustic noise, on the other hand, appears in both microphones simultaneously. The contribution of this paper is to utilize this phenomenon by filtering the microphone waveforms to create spatially narrowband information signals. With a narrowband signal, the microphone signal phasing information is adequate for distinguishing between acoustic noise and the blood pressure pulse. By choosing the microphone spacing correctly, subtraction of the two signals will enhance the information signal and cancel the noise signal. The general spacing problem is also presented.     

A 1.0-mW, 71-dB SNDR,fourth-order ΣΔ interface circuit for MEMS microphones

In this paper an integrated interface circuit for condenser MEMS microphones is presented. It consists of an input buffer followed by a multi-bit (12-levels), analog, second-order ΣΔ modulator and a fully-digital, single-bit, fourth-order ΣΔ modulator, thus providing a single-bit output signal with fourth order noise shaping, compatible with standard audio chipsets. The circuit, supplied with 3.3 V, exhibits a current consumption of 215 μA for the analog part and 95 μA for the digital part. The measured signal-to-noise and distortion ratio (SNDR) is 71 dB, with an input signal amplitude as large as −1.8 dB with respect to full-scale, obtained thanks to the use of a feed-forward architecture in the analog ΣΔ modulator, which relaxes the voltage swing requirements of the operational amplifiers. The test chip, fabricated in a 0.35-μm CMOS process, occupies an area of 3 mm², including pads.     

基于DSP二维声源定向系统设计

利用ADSP BF533 DSP处理器设计了一种二维声源定向系统。系统基于声波到达时间差技术,采用相位匹配算法,对两个传声器采集的声音信号进行分析。通过算法仿真验证了算法的可行性和准确性,并将算法在DSP上实现。     

节目制作中话筒的合理选用

声音的拾取要靠话筒,为了各种收音工作的需要,话筒产品种类繁多,所以节目制作中应综合考虑声源、环境等因素,合理选择与使用话筒.     

A low-power correlation-derivative CMOS VLSI circuit for bearing estimation

We present a CMOS integrated circuit (IC) for bearing estimation in the low-audio range that performs a correlation derivative approach in a 0.35-/spl mu/m technology. The IC calculates the bearing angle of a sound source with a mean variance of one degree in a 360/spl deg/ range using four microphones: one pair is used to produce the indication and the other to define the quadrant. An adaptive algorithm decides which pair to use depending on the direction of the incoming signal, in such a way to obtain the best estimate. The IC contains two blocks with 104 stages each. Every stage has a delay unit, a block to reduce the clock speed, and a 10-bit UP/DN counter. The IC measures 2 mm by 2.4 mm, and dissipates 600 /spl mu/W at 3.3 V and 200 kHz. It is purely digital and uses a one-bit quantization of the input signals.     

A comparative study of sound localization algorithms for energy aware sensor network nodes

Sound localization using energy-aware hardware for sensor networks nodes is a problem with many applications in surveillance and security. In this paper, we evaluate four algorithms for sound localization using signals recorded in a natural environment with an array of commercial off-the-shelf microelectromechanical systems microphones and a specially designed compact acoustic enclosure. We evaluate performance of the algorithms and their hardware complexity which relates directly to energy consumption.     

A Monaural Cue Sound Localizer

We describe the first single microphone sound localization system and its inspiration from theories of human monaural sound localization. Reflections and diffractions caused by the external ear (pinna) allow humans to estimate sound source elevations using only one ear. Our single microphone localization model relies on a specially shaped reflecting structure that serves the role of the pinna. Specially designed analog VLSI circuitry uses echo-time processing to localize the sound. A CMOS integrated circuit has been designed, fabricated, and successfully demonstrated on actual sounds.     

Analog circuit optimization using adjoint network based sensitivity analysis

Performance optimization as per the desired specifications is a major requirement of analog and mixed signal circuit design process. Rapid scaling of the semiconductor technology demands efficient optimization techniques with minimal manual efforts. In this paper, a gradient based method for analog circuit optimization using adjoint network based sensitivity analysis is presented. The sensitivity of circuit response with respect to the different parameters is computed by using analog circuit and its adjoint transformation. The proposed method is applied to optimize performance of a two stage operational amplifier (OpAmp). Subsequently, the OpAmp circuit is simulated using Cadence Virtuoso for optimized parameters and the results are validated with post fabrication measurement results.     

Implementation of a Microprocessor-Based Tactile Hearing Prosthesis

A microprocessor-based tactile vocoder is described that offers important advantages over its analog counterpart. Digital implementation provides more flexibility and precision. The system simulates, in real-time, the characteristics of a 16-channel tactile vocoder developed at Queen's University in Kingston, Ont., Canada and is intended for use with a linear array of 16 miniature vibrators. Design considerations of the circuit logic, program structure, and algorithms used to implement the vocoder and the performance characteristics such as the dynamic range and noise of the various algorithms are discussed. The overall system has a noise level less than ?60 dB re the maximum input and the bandwidth is 7.5 kHz. The system will first be used in the laboratory and in the classroom with two microphones and two tactile displays. Future plans call for the design to be adapted to lower power devices, when they become available, to achieve a single-display, wearable version that can be evaluated under natural conditions of acoustic signal and noise.     

基于噪声测量的模拟电路故障分析研究

模拟电路故障检测是在当前电子计算机技术不断完善的背景下,基于集成电路故障检测困难发展而来的新型电路检测模式。对于模拟电路来说,故障检测是一个难题,在此基础上,噪声检测在近年来开始逐渐兴起。为了进一步探究利用噪声进行模拟电路故障检测的方法,文中从模拟电路故障检测的背景展开论述,阐述了传统和现代两种不同的模拟电路故障检测方式,分析了利用噪声原理进行模拟电路故障检测的优势和特点,并且详细论述了当前利用RS运算符进行噪声故障检测的计算方法,并对利用噪声进行模拟电路故障检测的步骤做出了详细的阐述。     

模拟电路故障诊断的小波包预处理神经网络改进算法

提出了基于小波包预处理的神经网络模拟电路故障诊断方法的两种改进方法:最优小波包变换(OWPT)预处理和不完全小波包变换(IWPT)预处理BP神经网络算法。首先对模拟电路的响应信号用这两种方法进行预处理,然后计算预处理后信号各个频段上的归一化能量,把归一化的能量作为训练样本送给BP网络进行训练,有效减少了BP网络的输入节点和隐层节点的个数,从而减小了神经网络的规模,降低了计算的复杂度,加快了网络的训练和收敛速度。仿真实验表明此方法能够快速有效的对模拟电路的故障进行诊断和定位。     

基于听诊音频谱分析的数字听诊系统设计

为获得定量化的听诊结果,设计了一种基于听诊音频谱分析的数字听诊系统。该数字听诊系统由硅麦克风传感器接收人体器官发出的声音,经过运算放大、滤波和模数转换,将模拟信号转换为数字信号传送至专用集成电路芯片进行后续处理,再由终端设备输出详细、准确的听诊音频谱分析结果,达到帮助精确记录和科学分析听诊结果的目的。     

线性调频脉冲发声系统的设计

介绍了线性调频脉冲压缩原理及一种基于脉冲压缩技术的线性调频信号发声系统,该系统包括电路系统,计算机控制程序和发射装置.电路系统使用信号发生器件、单片机及模拟开关来产生周期性线性调频信号.计算机控制程序采用微软基础类库(MFC)窗口程序,通过串口控制单片机.使用压电声音发射器发声,并做了线性调频声音发射、接收实验,对接收的信号进行了脉冲压缩.     

The active control of sound

Active control of sound results from destructive interference between the sound field of an original acoustic source and that from a controllable array of `secondary' acoustic sources. For this destructive interference to occur over an appreciable region of space the sound field of the secondary sources must match that from the primary source in both time and space. The spatial matching requirement leads to an upper frequency of applicability of active control. Active control complements conventional passive methods of sound control, which do not work well at low frequencies. Practical feedforward controllers, using a multichannel generalisation of the well known LMS adaptive algorithm, have been developed, using as many as 16 loudspeakers and 32 microphones, and applied with considerable success in the control of low-frequency propeller noise inside aircraft and low-frequency engine noise inside cars. The authors describe such systems     

双传声器指向性噪声消除在手机中的应用

介绍了一种手机中使用的指向性双传声器噪声消除系统.系统采用两个灵敏度一致性较好的全指向性传声器,形成锥形指向性拾音波束,只接收120.范围以内的信号,大大减少拾音波束之外的噪声干扰;紧接着使用单通道语音增强算法,可进一步衰减采集到信号中的背景噪声.实验结果表明,该系统在保持了较小的语音失真的前提下,对于各种类型的背景噪...