Browse Topic: Audio equipment

Items (220)
Verifying large alternate product code for an JA GV document - JA1124XX
Active Safety Systems Standards Committee
A cooperative flight test campaign between the US Army and NASA was performed. This test sought to characterize the acoustic emissions of a fully instrumented MD530F helicopter using a snapshot array and a phased array of microphones. The snapshot array of microphones aimed to provide even coverage across the surface of a hemisphere, providing an acoustic emission hemisphere in a single 'snapshot' of time. The phased array of microphones was designed to provide enough resolution to determine noise sources from each individual blade as well as perform source separation from main rotor and tail rotor emissions. Test conditions for the characterization effort were chosen using a traditional one-factor-at-a-time approach as well as three design of experiment approaches. Characterization conditions included constant speed level flight, descent, and ascent conditions. Transient maneuver conditions were also captured over the snapshot array. The vehicle instrumentation included measurements of pilot controls, optical sensors to measure blade azimuth locations, pitch link loads, along with strain gauges to measure structural loads, blades and fuselage. This report will provide an overview of the test, document the data acquired, and provide some initial results.
Stephenson, James, Pascioni, Kyle, Houston, Mary, Stutz, Colin, Preston, Martin
This paper describes a mathematical framework for determining the optimal sensor set location for adequately capturing the sound generated by rotors. The approach leverages the gappy-POD method proposed by Everson and Sirovich [J. Opt. Soc. Am., Vol. 12, 1995, pp. 1657-1664], which first identifies the various mode constituents that make up the first few rotor blade-pass frequency harmonics of the sound-field. The algorithm is developed using a covariance matrix for the POD problem comprising auto- and cross-spectral densities of spatially and temporally resolved sound waves captured by an array of microphones oriented parallel to the axis of a laboratory-scale hovering rotor. Three different forms of the technique are developed and compared. These comprise a homogeneous form and two heterogeneous forms; the heterogeneous forms are referred to as XX-topos and XX-chronos and depends on which term in the error minimization equation is assigned the gappy sensor set. A greedy algorithm is then employed to determine the optimal location of the limited sensor set. The findings are analyzed for different combinations of POD modes and blade-pass frequency harmonics of the sound generated by the hovering rotor.
Tinney, Charles, Valdez, John, Zhao-Dubuc, Irene
Design and Validation of Low-Cost Intensity Probe2019-01-14626/5/2019
Sound intensity measurement techniques that used a two-microphone configuration, were first developed in the late 1970s. Originally, the focus was on improving precision during testing or post-processing. However, with the advent of modern, sophisticated equipment, the focus has shifted to the apparatus. Availability of phase-matched microphones has made post-test correction obsolete as the microphones eliminate a majority of the errors before the data is even collected. This accuracy, however, comes at a cost, as phase-matched microphones are highly priced. This paper discusses employing the method of improving post-processing precision, using inexpensive, current equipment. The phase error of the system is corrected using a simple calibration technique and a handheld phase calibrator that is similar to the one used for amplitude calibration of microphones. The intensity probe and calibrator is manufactured using rapid prototyping and the executable software that goes with the probe is designed in NI LabVIEW. The entire setup uses inexpensive parts to lower the cost and modern software to compensate for the errors due to these parts. The design of the probe and the accompanying software will be discussed in the paper. Additionally, the accuracy of the probe will be compared to a commercially available sound intensity probe and the results will be discussed.
Gundre, Karan, Barnard, Andrew
This paper investigates the effect of pitch attitude on both performance and acoustics of a lift-offset coaxial rotor based on a first-principles and high-fidelity CFD/CSD loose coupling approach at 150 and 200 knots. The pitch attitudes selected for this research are -5° , 0° , and 5° . The CFD/CSD loose coupling simulations are carried out using the CREATETM -AV software Helios while the coaxial rotor acoustics is simulated using PSU-WOPWOP at eight microphones positioned below the lower rotor. A detailed aerodynamic analysis is performed at 150 knots. A total of six major aerodynamic interactions are identified: 1) hub-wake interaction, 2) self-BVI, 3) parallel rotor-to-rotor BVI, 4) blade-crossover events, 5) root-induced BVI, and 6) reversed-flow- edge-vortex interactions. The strength of these interactions is dependent on the vehicle pitch attitude. The cases with negative pitch attitude show significantly stronger impulsive pressure pulses, which is found to be induced by parallel rotor-to-rotor BVIs of the lower rotor. Moreover, the positive and zero pitch attitude cases tend to dominate the acoustic region on the starboard side of the coaxial rotor, whereas the negative pitch attitude case shows higher acoustic pressure peaks on the port side. Overall, the case with a positive pitch attitude shows significant improvement in rotor aerodynamic efficiency, rotor acoustics, and vehicle power performance at high speed. The hemispherical acoustic simulation results at 150 knots also show that the noise is less likely to propagate in the forward direction with a positive pitch attitude.
Jia, Zhongqi, Sharma, Kalki, Lee, Seongkyu, Brentner, Kenneth
The Tiltrotor Test Rig (TTR) was tested in the National Full-Scale Aerodynamics Complex (NFAC) 40- by 80-Foot Wind Tunnel from 2017 to 2018. The primary goal of the test was to understand the operational capabilities of the TTR, while also acquiring research data, including acoustic data. Four microphones were placed around the TTR: two on the wind tunnel floor and two on struts. Acoustic measurements of the TTR rotor were acquired to 1) understand the acoustic testing capabilities of the TTR in the NFAC 40- by 80-FootWind Tunnel, 2) compare to previous XV-15 rotor acoustic data acquired in the NFAC 80- by 120-Foot Wind Tunnel, and 3) provide data for future validation studies. A data quality study revealed that the NFAC 40- by 80-Foot Wind Tunnel is an adequate acoustic environment to test the TTR rotor. For a given thrust and advance ratio, a shaft angle sweep was performed and acoustic measurements were compared against 1996 and 1999 XV-15 data in the NFAC 80- by 120-Foot Wind Tunnel; differences between the three tests are discussed.
Schatzman, Natasha, Malpica, Carlos
Source Separations and Identification by Structural Holography2016-01-17996/15/2016
The source field reconstruction aims at identifying the excitation field measuring the response of the system. In Near-field Acoustic Holography, the response of the system (the radiated acoustic pressure) is measured on a hologram using a microphones array and the source field (the acoustic velocity field) is reconstructed with a back-propagation technique performed in the wave number domain. The objective of the present works is to use such a technique to reconstruct displacement field on the whole surface of a plate by measuring vibrations on a one-dimensional holograms. This task is much more difficult in the vibratory domain because of the complexity of the equation of motion of the structure. The method presented here and called "Structural Holography" is particularly interesting when a direct measurement of the velocity field is not possible. Moreover, Structural Holography decreases the number of measurements required to reconstruct the displacement field of the entire plate. This method permits to separate the sources in the case of multi-sources excitations by considering them as direct or back waves. It's possible to compute the structural intensity of one particular source without the contributions of others sources. The purpose of this paper is to introduce the Structural Holography method. The first part presents the theoretical background of the method. A numerical simulation of displacement fields generate by few sources for an infinite plate is presented in a second part. The structural intensity for each source is computed by removing the contribution of others source. Finally, some results are presented for a simply supported plate.
Thomas, Jean-Hugh, Chesnais, Corentin, Totaro, Nicolas, Guyader, Jean-Louis
In order to measure the noise footprint of a helicopter approach procedure, microphones are usually distributed over a large area. In this paper, however, a method is utilized where the measurement of the noise footprint is conducted in parts by distributing the microphones over a limited area and executing the same approach procedure, while the position of landing point is shifted. The flight tests are conducted with the EC135-ACT/FHS helicopter. Guidance to the pilots is provided by use of a "Tunnel-in-the-Sky" head down pilot display. Since the same flight procedure is executed multiple times it is possible to conduct a statistical analysis on the accuracy and reproducibility of the approach procedure in terms of position, velocity and generated noise. This statistical analysis shows that the geometric accuracy is in the order of 6 meters in the horizontal plane and 12 meters in the vertical plane. The variations in the measured sound exposure level from one approach to another are in the order of 3-7.5 dB(A). The approach procedure is also executed with excessive cyclic pilot control inputs in order to identify an upper limit of the extra noise generated by unsteady flight. However, these approaches are indistinguishable, in terms of the sound exposure level or sound pressure level, from the approaches with normal pilot control inputs.
Frederic, Willem, Gursky, Bianca
Investigating GSM Interference in Automotive Sound Systems Using State of the Art Electromagnetic Simulation2013-36-019810/7/2013
EMI (Electromagnetic Interference) is one of the major concerns today in the automotive industry. The main reason is that vehicles are using and depending more on electronic technology. The causes of electromagnetic interference problems are not only related to the ever-increasing number of embedded electronics systems in vehicles, but also to external electronic devices that are brought in to automobiles by drivers and passengers (e.g. cell phone, MP3 players, Bluetooth devices, portable video games). Even though these problems can cause serious issues on safety systems like the airbag, their symptoms are often noticed in a less harm way in the sound system. A very common EMI problem in automotive sound systems is a particular noise caused by devices that uses GSM (Global System for Mobile Communications) technology. Most of the cell phones and vehicle locators rely on GSM technology. In Brazil there is a national normative known as Contran 245, which when issued, will enforce the use of locators on all vehicles commercialized in Brazil. Hence, EMI problems caused by GSM technology will likely to be noticed more often in Brazil. GSM devices emit short duration RF (Radio Frequency) pulses at a rate of 217Hz. The interference pulses contain the fundamental frequency plus a large number of harmonics that overlaps the frequency range of listening, which goes up to 20kHz. This work presents a detailed study of the EMI of GSM devices in automotive sound systems using state of the art numerical simulation technology provided by ANSYS HFSS and ANSYS Designer tools. A system model comprising a vehicle with a complete sound system, wiring harness and GSM devices communicating to a base station GSM tower is used for this analysis. Measurement data is also presented showing a good match to the simulated results. Options to decrease the GSM interference and overcome this issue are also discussed.
Mologni, Juliano F., Siqueira, Cesareo L. R., Nogueira, Artur, Colin, Arnaud, Alves, Marco A. R., Kopp, Markus
Spherical Beamforming with Enhanced Dynamic Range2013-01-19775/13/2013
Beamforming with an array of microphones on a sphere is an attractive tool for doing noise source localization in cabin environments. In order to achieve acceptable angular resolution, the array must have some minimum diameter, implying that many microphones are needed to obtain low sidelobe level over the frequency range of interest. For electric cars there is an increased need to cover high frequencies. The present paper describes a method to significantly reduce the sidelobe level over a broad frequency range relative to Spherical Harmonics Beamforming (SHB). For each focus point, a set of Finite Impulse Response (FIR) filters are optimized to minimize the highest sidelobe in a Filter And Sum (FAS) beamformer, while maintaining sensitivity at the focus point and limiting the White Noise Gain (WNG). Similar approaches have been published, but a main issue in connection with applications for noise source mapping is the computational load in performing the optimization of FIR filters for each focus point. The paper describes a method to handle that challenge. Basically, filter coefficient are pre-optimized for a pre-defined 3D mesh and stored in a database. When focusing has to be performed at an arbitrary point in space, an interpolation is performed, typically on the filter vectors related to a set of neighboring mesh points. Results are presented for simulated and real measurements. These show that the optimized FAS method provides very significant improvement relative to SHB on sidelobe suppression and thus on the dynamic range available for source identification.
Hald, Jorgen
New Approaches for Low Voltage Audio System and Circuit Design2012-01-00374/16/2012
New vehicle systems that stop and restart the engine to avoid prolonged idling place new requirements on audio systems. In particular, there is a need for the audio system to continue to operate during the engine start sequence. In the past this was not an issue, as audio systems could be turned off during the engine start sequence. Among the ramifications for this “operate while starting” requirement is that the audio system must operate at a lower voltage, say 6 volts, instead of 10 volts. Audio system operation in this context means continuing to play music without interruption or spurious noises. The main focus of this presentation is on audio circuits, both line-level and speaker level. In the past, line level outputs could be fed by linear regulated power supplies that developed 8 to 10 volts. The signal levels sent from one module to another could thus be in the 3 volt RMS range. Given DSP integrated circuits with 3.3 or 5 volt power supplies, it was often necessary to “gain up” the DSP audio outputs to meet system requirements. Running line level input and output circuits on 3.3 volt power supplies implies that gaining up is no longer necessary, and in fact it may be necessary to gain down DSP audio outputs. The challenge is to perform the gain reduction without degrading signal to noise in the signal processing chain. Circuits that accomplish this are shown. A prototype amplifier that operates with output IC power supplies as low as 6 volts has been successfully demonstrated. The design and test results from the prototype amplifier are discussed.
Tillo, Joe, Tressler, Steve, Hanson, Jonathan
Model-Based Design Case Study: Low Cost Audio Head Unit2011-01-00524/12/2011
The use of model-based software development in automotive applications has increased in recent years. Current vehicles contain millions of lines of code, and millions of dollars are spent each year fixing software issues. Most new features are software controlled and many times include distributed functionality, resulting in increased vehicle software content and accelerated complexity. To handle rapid change, OEMs and suppliers must work together to accelerate software development and testing. As development processes adapt to meet this challenge, model-based design can provide a solution. Model-based design is a broad development approach that is applied to a variety of applications in various industries. This paper reviews a project using the MATLAB/Simulink/Stateflow environment to complete a functional model of a low cost radio. We approached the project independent of any specific software architecture, using only written specifications and current working units as a guide to functionality. The final result is a model that captures the functional behavior of the radio. While the model could be used for auto code-generation, this project focused on utilizing the radio model to prove-out the functional and HMI specifications. A discussion of our design approach, including architecture, testing, related HMI components, and lessons learned, are included in the paper.
Marx, Jason, Sun, Weiqian
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