The Acoustics Research Group of Brigham Young University analyzed Server room of one of the C7 Data Centers Colocation Lindon 5 Facility in 2010. Results show a signicant amount of sound power below 500Hz and EDT's between 0.25-3.20 seconds depending on location. For more information on C7 Data Centers, check - http://www.c7.com/data-center/
1) The document describes a proposed fault tolerant controller system (FTCS) strategy for temperature control of an air conditioning system. The FTCS is designed to be robust against noise, sensor failures, disturbances, and parameter variations while maintaining optimal performance.
2) A mathematical model is developed consisting of models for the heat exchanger and conditioning space. A neural network model is also developed to model the system dynamics.
3) The neural network is trained using a backpropagation algorithm to minimize the mean squared error between the network outputs and actual system outputs over time. Training results show the network can accurately model the system.
This document describes the design and implementation of an acoustic beamforming system using a microphone array and the TDS3230 DSK board. It discusses the theory of delay-and-sum beamforming and the design of the microphone array. Simulations were conducted in MATLAB to test source localization and spatial filtering capabilities. The system requirements and overall design are then described, including the hardware, DSP software, and GUI interface. Tests were planned to evaluate source localization accuracy and the ability to filter out noise sources located at different angles.
This thesis investigates using a reverberation chamber as a controlled environment for wireless device testing. The author characterizes the wireless channel in a reverberation chamber at NIST using time and frequency domain techniques. It is shown that by loading the chamber with absorbers, the delay spread can be changed to emulate channels with different coherence bandwidths. Stirrers are used to control Doppler spread and emulate fast and slow fading channels. The author estimates the Rician K-factor from measured envelope statistics using maximum likelihood estimation and compares results with a method using amplitude and phase. Bit error rate performance of modulated signals is evaluated in emulated narrowband and wideband fading channels. The author concludes reverberation chambers can accurately emulate real-
This document provides a summary of multitone testing techniques for electronic devices. It begins with an overview of how multitone testing allows acquiring measurement results from multiple frequencies simultaneously, unlike traditional single-tone and two-tone testing which require changing the input frequency each time. It then discusses some key problems that necessitate multitone testing such as reducing time and cost. Finally, it provides motivations for carrying out multitone testing research such as gaining experience in technical writing and learning newer testing methods used in manufacturing. The document contains figures illustrating multitone signals and how they are used to test devices.
Audio Equalization Using LMS Adaptive FilteringBob Minnich
This document describes research into using an adaptive filter with the LMS algorithm for audio equalization. It introduces audio equalization and the problem of frequency response variations between the source and listener. The proposed solution is to use an adaptive filter to adjust for these variations. It then provides details on adaptive filtering and the LMS algorithm. Finally, it describes MATLAB simulations conducted to test the approach, including using white noise as an input signal, simulating signal distortions, and accounting for room delay using cross-correlation.
This document describes a system identification project for a hard disk drive servosystem. The goal is to identify the high-order system model using two different methods: sine sweep and average ETFE. For the sine sweep method, increasing the input magnitude improves the estimated model accuracy. With an input magnitude of 1010 and increased frequency resolution, the estimated bode plot matches the true model well. For the average ETFE method, increasing the number of experiments from 50 to 2000 also improves the estimated model accuracy such that it closely matches the true bode plot.
Describe The Main Functions Of Each Layer In The Osi Model...Amanda Brady
Tone injection is a technique used to increase the constellation size of a signal constellation. It works by mapping each point in the original constellation to multiple equivalent points in an expanded constellation. This allows for embedding additional information by substituting points, improving spectral efficiency. However, it also increases implementation complexity and may degrade performance due to increased decision regions. Tone injection is useful for applications requiring high data rates within bandwidth constraints.
1) The document describes a proposed fault tolerant controller system (FTCS) strategy for temperature control of an air conditioning system. The FTCS is designed to be robust against noise, sensor failures, disturbances, and parameter variations while maintaining optimal performance.
2) A mathematical model is developed consisting of models for the heat exchanger and conditioning space. A neural network model is also developed to model the system dynamics.
3) The neural network is trained using a backpropagation algorithm to minimize the mean squared error between the network outputs and actual system outputs over time. Training results show the network can accurately model the system.
This document describes the design and implementation of an acoustic beamforming system using a microphone array and the TDS3230 DSK board. It discusses the theory of delay-and-sum beamforming and the design of the microphone array. Simulations were conducted in MATLAB to test source localization and spatial filtering capabilities. The system requirements and overall design are then described, including the hardware, DSP software, and GUI interface. Tests were planned to evaluate source localization accuracy and the ability to filter out noise sources located at different angles.
This thesis investigates using a reverberation chamber as a controlled environment for wireless device testing. The author characterizes the wireless channel in a reverberation chamber at NIST using time and frequency domain techniques. It is shown that by loading the chamber with absorbers, the delay spread can be changed to emulate channels with different coherence bandwidths. Stirrers are used to control Doppler spread and emulate fast and slow fading channels. The author estimates the Rician K-factor from measured envelope statistics using maximum likelihood estimation and compares results with a method using amplitude and phase. Bit error rate performance of modulated signals is evaluated in emulated narrowband and wideband fading channels. The author concludes reverberation chambers can accurately emulate real-
This document provides a summary of multitone testing techniques for electronic devices. It begins with an overview of how multitone testing allows acquiring measurement results from multiple frequencies simultaneously, unlike traditional single-tone and two-tone testing which require changing the input frequency each time. It then discusses some key problems that necessitate multitone testing such as reducing time and cost. Finally, it provides motivations for carrying out multitone testing research such as gaining experience in technical writing and learning newer testing methods used in manufacturing. The document contains figures illustrating multitone signals and how they are used to test devices.
Audio Equalization Using LMS Adaptive FilteringBob Minnich
This document describes research into using an adaptive filter with the LMS algorithm for audio equalization. It introduces audio equalization and the problem of frequency response variations between the source and listener. The proposed solution is to use an adaptive filter to adjust for these variations. It then provides details on adaptive filtering and the LMS algorithm. Finally, it describes MATLAB simulations conducted to test the approach, including using white noise as an input signal, simulating signal distortions, and accounting for room delay using cross-correlation.
This document describes a system identification project for a hard disk drive servosystem. The goal is to identify the high-order system model using two different methods: sine sweep and average ETFE. For the sine sweep method, increasing the input magnitude improves the estimated model accuracy. With an input magnitude of 1010 and increased frequency resolution, the estimated bode plot matches the true model well. For the average ETFE method, increasing the number of experiments from 50 to 2000 also improves the estimated model accuracy such that it closely matches the true bode plot.
Describe The Main Functions Of Each Layer In The Osi Model...Amanda Brady
Tone injection is a technique used to increase the constellation size of a signal constellation. It works by mapping each point in the original constellation to multiple equivalent points in an expanded constellation. This allows for embedding additional information by substituting points, improving spectral efficiency. However, it also increases implementation complexity and may degrade performance due to increased decision regions. Tone injection is useful for applications requiring high data rates within bandwidth constraints.
The Raytheon UMass-Lowell Research Institute (RURI) constructed an anechoic chamber and positioning system to test electronic devices from 8-12 GHz. Ray tracing simulations predicted the chamber would meet specifications with amplitude taper below 1 dB and phase taper below 22.5 degrees within the quiet zone. Measurements confirmed the quiet zone performance matched expectations. The completed anechoic chamber and positioner will allow RURI to fully characterize electronic devices from design to testing.
This document discusses using COMSOL to simulate acoustic scattering for an acoustic imaging system to track hand movements. COMSOL is used to calculate diffraction patterns from 1cm scattering centers representing fingers. The arrival times of the diffracted wavefront at receiver locations are input to a neural network to map the times to scattering center positions and reconstruct hand position with no moving parts. COMSOL helped determine system parameters like wavelength and detector placement. Over 2800 simulations were run to generate training data for the neural network, which can reconstruct positions to within 2% error.
The document discusses design for test (DFT) techniques. It explains that DFT aims to improve the testability of chip designs by adding mechanisms to control and observe internal nodes for manufacturing testing. This allows testing of each block or component on the chip to identify defective parts. Specifically, it discusses using scan chains to test combinational logic, and techniques like MBIST and boundary scan for testing memories and I/O, respectively. The goal of DFT is to effectively test designs at the component level to improve quality and yield.
Simulation and hardware implementation of Adaptive algorithms on tms320 c6713...Raj Kumar Thenua
Raj Kumar Thenua presented his dissertation on "Simulation and Hardware Implementation of NLMS algorithm on TMS320C6713 Digital Signal Processor". The presentation outlined the introduction to adaptive noise cancellation, various adaptive algorithms like LMS, NLMS and RLS. MATLAB simulation results were analyzed for tone signals comparing the performance of algorithms. The best performing NLMS algorithm was implemented on a TMS320C6713 DSP processor. Results for tone signals and ECG signals showed improvement in SNR. The dissertation concluded the real-time implementation enabled analysis of actual signals and provided better noise reduction than simulation.
This document summarizes a student laboratory project to create a neural network that can identify input signals. The neural network was programmed on a PSoC 5LP board to recognize either a sine wave or square wave. It takes a 128-point FFT of the input signal and feeds it through a three-layer neural network. A learning algorithm was implemented in C++ to adjust weights and minimize error over multiple runs. Testing showed the network could correctly identify sine and square waves by displaying the result on an LCD screen, though the sine wave output was imperfect due to limited learning iterations.
The Doppler EffectWhat is the Doppler effect, and why is it impo.docxcherry686017
The Doppler Effect
What is the Doppler effect, and why is it important to understand?
Sound
1. Describe what is meant by "sound." Explain how sound is created, transmitted, and sensed.
2. Set the source velocity (the Italian label reads Velocidad del emisor) to 0.0. Run the simulation (click the Empieza button). Calculate the frequency of the waves by counting the number of full waves that pass through a point in ONE second. You can press the Pausa and Continua buttons to step through the animation to pause and restart the wave motion.
3. The distance between numbered tick marks is 1 meter. Measure the wavelength using these tick marks. Use the wavelength and the frequency you calculated in number 2 to calculate the velocity of the wave.
The Doppler Effect in Sound
4. Now, set the source velocity (Velocidad del emisor) equal to 0.50. Run the simulation until the wave source (red rectangle) has moved close to the observer (blue rectangle). Calculate the new wavelength for the waves on each sideof the moving source? Count the tick marks in one full wave to make this calculation, knowing that each tick mark equals 0.2 meters.
5. Examine the motion of the waves. Has the frequency increased or decreased on each side of the source?
6. Use the equation x f = v, to calculate the frequency at a point on each side of the source. Remember that the velocity of the wave DOES NOT change (so use the velocity you calculated in #2). You will also use the wavelength you calculated for the wavelengths of the waves on each side of the source for #4.
7. Use the equations provided on page 2 of the Read section to calculate what the frequency actually should be on each side of the source (show your work below). Use this to see how accurate your answers in #6 were.
Electromagnetic Waves and Light
8. Summarize how electromagnetic waves are similar to acoustical (sound) waves. How are they different?
The Doppler Shift in Light
9. How is the Doppler shift used in astronomy? What is meant by the terms red-shift and blue-shift?
Summary (Homework)
10. Radar is a process that uses reflected electromagnetic waves in order to create an image of an object. Doppler radar (often used in weather) is used to tell the speed and direction clouds are moving. Explain how this might work. (Hint: Think about how radio waves might change when they reflect off of moving objects.)
11. Explain why the pitch of an object approaching an observer (such as a fire truck with its siren on) differs from the pitch as it moves away from the observer. Remember that pitch is the brain's interpretation of a sound's frequency.
12. Now answer the Focus Question. What is the Doppler effect, and why is it important to understand?
3. (10 pt) ASCII, Unicode, and EBCDIC are, of course, not the only numeric / character codes. The Sophomites from the planet Collegium use the rather strange code shown in the Figure below. T ...
DESIGN OF INTELLIGENT CONTROL SYSTEM USING ACOUSTIC PARAMETERS FOR GRINDING M...cscpconf
This paper utilizes acoustic parameters such as FS,NC, N, P, INC, FL, FH, W for acoustic signals S of different running conditions of a ballmill to deriveout the acoustic signatures and
hence control signals, which is to be used for designing the control systems of the mill. The parameters FS, NC, N, P, INC, FL, FH and W are represented by sample rate in Hz, number of
cepstral coefficients, length of frame in samples, number of filters in filter bank, frame increment, low end of the lowest filter, high end of highest filter and the window over which the analysis is to be performed respectively. The work establishes an
appropriate theoretical background that helps to predict dynamic breakage characteristics with respect to particle size
distribution of materials, adequately supported by experimental data. The signatures of different running conditions of grinding mill have been extracted from the captured signal in time frame
these have been used as feedback signal to monitor the grinding operation. Condenser based microphones have been used for capturing acoustic signals in time domain directly in
computers and stored for further analysis. Matlab R2010b has been used for different analysis of the experiment. On analyzing the signatures, it has been observed whether the fines are
produced progressively to attain the desired size range or the mill producing undesired products. Thus, the approach has been used in this paper has the ability to arrive in the stage of
optimum grinding by tuning parameters of the mill in real time, and also it can prevent the mill to enter into an erroneous state. Moreover, on study it has found that the present scheme can be
used more accurately in comparison to the earlier work of the author. This paper presents an implementation scheme to use acoustic signal as the control signal to regulate the operation of
a grinding mill.
Evaluation of a Multipoint Equalization System based on Impulse Responses Pro...a3labdsp
This document evaluates a multipoint equalization algorithm that combines fractional octave smoothing of impulse responses measured in multiple locations in a room or car. It investigates how the equalization performance is affected by varying parameters like the number of measurement positions and equalization zone size. The algorithm extracts a representative prototype response and inverse filter from the smoothed impulse responses. Tests show the proposed approach achieves better spectral deviation and equalization than a single-point method, and performance decreases but remains effective as the equalization zone is expanded to more distant positions.
This document provides a mid-term report on the design and analysis of a voltage controlled oscillator (VCO) for a master's project. It discusses completing the circuit diagram, symbol creation, and test circuit simulation in Cadence using 180nm technology. Simulation results so far show the VCO design is one third complete, with layout and performance analysis remaining. The report includes background on VCO metrics like frequency, tuning range, phase noise, and power consumption. It also reviews applications of VCOs in frequency translation and discusses challenges in designing low phase noise CMOS VCOs.
This document describes the Illinois Scan Architecture, a technique for reducing test costs for chips with scan designs. It works by dividing the main scan chain into multiple parallel internal chains, with a single scan input pin. This allows test vectors to be broadcast to all chains simultaneously, reducing test time and data volume by the number of chains with little impact on fault coverage. The document provides experimental data showing reductions in test vectors, cycles, and data volume for several ISCAS circuits using Illinois Scan. It also discusses techniques for further optimizing the technique, such as grouping chains intelligently to minimize the number of scan input pins needed.
A study on improving speaker diarization system = Nghiên cứu phương pháp cải ...Man_Ebook
The document is a master's thesis that explores improving a speaker diarization system by comparing X-Vectors and ECAPA-TDNN embeddings. It includes:
- An introduction outlining the research interest in evaluating these two embedding methods on Vietnamese data sets for speaker verification and diarization tasks.
- Details of the baseline system that uses X-Vector embeddings and the proposed system that replaces it with ECAPA-TDNN embeddings.
- Experiments conducted on private (IPCC_110000, VTR_1350) and public (ZALO_400) data sets to evaluate the systems, along with results showing the proposed system outperforms the baseline.
A study on improving speaker diarization system = Nghiên cứu phương pháp cải ...Man_Ebook
The document is a master's thesis submitted by Tung Lam Nguyen to the Hanoi University of Science and Technology. It studies improving speaker diarization systems by exploring the capabilities of ECAPA-TDNN embeddings versus X-Vector embeddings in a Vietnamese speaker diarization system. The thesis contains experiments evaluating both baseline and proposed systems on speaker verification and speaker diarization tasks using various private and public Vietnamese datasets. The results show that the proposed system using ECAPA-TDNN outperforms the baseline X-Vector system on all tasks and datasets.
This document describes the development of a real-time audio DSP device controlled by an ultrasonic range sensor. The device takes an audio input, applies an echo effect with feedback gain controlled by the range sensor, and outputs the processed audio. Key aspects include:
1) Hardware components like an ARM microcontroller, audio input/output, range sensor, and display are selected and interconnected.
2) Software is developed in C++ to initialize components, average sensor data, implement the echo effect DSP with range-controlled feedback gain, and display information.
3) The device is implemented on a breadboard, tested, and results like noise and distortion are analyzed to improve the design. Further modifications are also
Cognitive Radio Network simulation is available from v7 of NetSim. Cognitive Radio Networks
allows you to connect, if required, with Ethernet, Wireless LAN, IP Routing, TCP / UDP and
allows users to log packet and event traces.
A Low Latency Implementation of a Non Uniform Partitioned Overlap and Save Al...a3labdsp
FIR convolution is a widely used operation in digital signal processing field, especially for filtering operations in real time scenarios. In this context, low computationally demanding techniques for calculating convolutions with low input/output latency become essential, considering that the real time requirements are strictly related to the impulse response length. In this paper, a multithreading real time implementation of a Non Uniform Partitioned Overlap and Save algorithm is proposed with the aim of lowering the workload required in applications like reverberation, also exploiting the human ear sensitivity. Several results are reported in order to show the effectiveness of the proposed approach in terms of computational cost, taking into consideration different impulse responses and also introducing comparisons with existing techniques of the state of the art.
This document analyzes the acoustical quality of the Haas School of Business at UC Berkeley. It finds that 65% of occupants are dissatisfied with the acoustics, particularly in open offices. Sound level measurements were taken in various room types, with some rooms exceeding noise standards. Recommendations include installing sound masking systems in open offices to mask distracting noises while keeping background noise levels within standards. Office booths are also proposed to provide private spaces for phone calls but may not be practical or affordable. Upgrading to improved acoustic partitions is another option but may not significantly improve satisfaction given the cost.
This document summarizes a lab report on digital signal processing. The lab covered A/D converters, Fourier series, and sources of error in digital signals like quantization, clipping, and aliasing. It also looked at Nyquist plots. The purpose of the lab was to gain understanding of digital signal processing and how it is used to analyze signals that cannot be visually inspected. Key aspects covered include how analog signals are converted to digital, Fourier analysis to interpret signals, and sources of error introduced in digitization.
This document summarizes the design, fabrication, and testing of a rotating acoustic diffusor implemented in the NASA LaRC Structural Acoustic Loads and Transmission (SALT) facility reverberation room. A 10' x 6' planar diffusor panel was built from a steel frame covered with corrugated PVC to reduce measurement variance by modulating room modes during data acquisition. Tests were conducted with and without the diffusor rotating, showing reduced variance in noise reduction spectra. Additionally, finite element modeling was used to simulate the diffusor's ability to modulate room modes at different angles. The results indicate the rotating diffusor improves data quality and further investigation is warranted.
Food Grade PAO - Based Compressor LubricantOlivia Grey
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Airbase Industries Synthetic Piston Compressor Oil for reference - https://goo.gl/Hr3yWo
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Professional Mariners require a medical program that they can depend on to protect them throughout the world. Regardless of the type of vessel, Mariners require coverage that provides security, flexibility and benefits unique to today’s Marine Industry demands.
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The Raytheon UMass-Lowell Research Institute (RURI) constructed an anechoic chamber and positioning system to test electronic devices from 8-12 GHz. Ray tracing simulations predicted the chamber would meet specifications with amplitude taper below 1 dB and phase taper below 22.5 degrees within the quiet zone. Measurements confirmed the quiet zone performance matched expectations. The completed anechoic chamber and positioner will allow RURI to fully characterize electronic devices from design to testing.
This document discusses using COMSOL to simulate acoustic scattering for an acoustic imaging system to track hand movements. COMSOL is used to calculate diffraction patterns from 1cm scattering centers representing fingers. The arrival times of the diffracted wavefront at receiver locations are input to a neural network to map the times to scattering center positions and reconstruct hand position with no moving parts. COMSOL helped determine system parameters like wavelength and detector placement. Over 2800 simulations were run to generate training data for the neural network, which can reconstruct positions to within 2% error.
The document discusses design for test (DFT) techniques. It explains that DFT aims to improve the testability of chip designs by adding mechanisms to control and observe internal nodes for manufacturing testing. This allows testing of each block or component on the chip to identify defective parts. Specifically, it discusses using scan chains to test combinational logic, and techniques like MBIST and boundary scan for testing memories and I/O, respectively. The goal of DFT is to effectively test designs at the component level to improve quality and yield.
Simulation and hardware implementation of Adaptive algorithms on tms320 c6713...Raj Kumar Thenua
Raj Kumar Thenua presented his dissertation on "Simulation and Hardware Implementation of NLMS algorithm on TMS320C6713 Digital Signal Processor". The presentation outlined the introduction to adaptive noise cancellation, various adaptive algorithms like LMS, NLMS and RLS. MATLAB simulation results were analyzed for tone signals comparing the performance of algorithms. The best performing NLMS algorithm was implemented on a TMS320C6713 DSP processor. Results for tone signals and ECG signals showed improvement in SNR. The dissertation concluded the real-time implementation enabled analysis of actual signals and provided better noise reduction than simulation.
This document summarizes a student laboratory project to create a neural network that can identify input signals. The neural network was programmed on a PSoC 5LP board to recognize either a sine wave or square wave. It takes a 128-point FFT of the input signal and feeds it through a three-layer neural network. A learning algorithm was implemented in C++ to adjust weights and minimize error over multiple runs. Testing showed the network could correctly identify sine and square waves by displaying the result on an LCD screen, though the sine wave output was imperfect due to limited learning iterations.
The Doppler EffectWhat is the Doppler effect, and why is it impo.docxcherry686017
The Doppler Effect
What is the Doppler effect, and why is it important to understand?
Sound
1. Describe what is meant by "sound." Explain how sound is created, transmitted, and sensed.
2. Set the source velocity (the Italian label reads Velocidad del emisor) to 0.0. Run the simulation (click the Empieza button). Calculate the frequency of the waves by counting the number of full waves that pass through a point in ONE second. You can press the Pausa and Continua buttons to step through the animation to pause and restart the wave motion.
3. The distance between numbered tick marks is 1 meter. Measure the wavelength using these tick marks. Use the wavelength and the frequency you calculated in number 2 to calculate the velocity of the wave.
The Doppler Effect in Sound
4. Now, set the source velocity (Velocidad del emisor) equal to 0.50. Run the simulation until the wave source (red rectangle) has moved close to the observer (blue rectangle). Calculate the new wavelength for the waves on each sideof the moving source? Count the tick marks in one full wave to make this calculation, knowing that each tick mark equals 0.2 meters.
5. Examine the motion of the waves. Has the frequency increased or decreased on each side of the source?
6. Use the equation x f = v, to calculate the frequency at a point on each side of the source. Remember that the velocity of the wave DOES NOT change (so use the velocity you calculated in #2). You will also use the wavelength you calculated for the wavelengths of the waves on each side of the source for #4.
7. Use the equations provided on page 2 of the Read section to calculate what the frequency actually should be on each side of the source (show your work below). Use this to see how accurate your answers in #6 were.
Electromagnetic Waves and Light
8. Summarize how electromagnetic waves are similar to acoustical (sound) waves. How are they different?
The Doppler Shift in Light
9. How is the Doppler shift used in astronomy? What is meant by the terms red-shift and blue-shift?
Summary (Homework)
10. Radar is a process that uses reflected electromagnetic waves in order to create an image of an object. Doppler radar (often used in weather) is used to tell the speed and direction clouds are moving. Explain how this might work. (Hint: Think about how radio waves might change when they reflect off of moving objects.)
11. Explain why the pitch of an object approaching an observer (such as a fire truck with its siren on) differs from the pitch as it moves away from the observer. Remember that pitch is the brain's interpretation of a sound's frequency.
12. Now answer the Focus Question. What is the Doppler effect, and why is it important to understand?
3. (10 pt) ASCII, Unicode, and EBCDIC are, of course, not the only numeric / character codes. The Sophomites from the planet Collegium use the rather strange code shown in the Figure below. T ...
DESIGN OF INTELLIGENT CONTROL SYSTEM USING ACOUSTIC PARAMETERS FOR GRINDING M...cscpconf
This paper utilizes acoustic parameters such as FS,NC, N, P, INC, FL, FH, W for acoustic signals S of different running conditions of a ballmill to deriveout the acoustic signatures and
hence control signals, which is to be used for designing the control systems of the mill. The parameters FS, NC, N, P, INC, FL, FH and W are represented by sample rate in Hz, number of
cepstral coefficients, length of frame in samples, number of filters in filter bank, frame increment, low end of the lowest filter, high end of highest filter and the window over which the analysis is to be performed respectively. The work establishes an
appropriate theoretical background that helps to predict dynamic breakage characteristics with respect to particle size
distribution of materials, adequately supported by experimental data. The signatures of different running conditions of grinding mill have been extracted from the captured signal in time frame
these have been used as feedback signal to monitor the grinding operation. Condenser based microphones have been used for capturing acoustic signals in time domain directly in
computers and stored for further analysis. Matlab R2010b has been used for different analysis of the experiment. On analyzing the signatures, it has been observed whether the fines are
produced progressively to attain the desired size range or the mill producing undesired products. Thus, the approach has been used in this paper has the ability to arrive in the stage of
optimum grinding by tuning parameters of the mill in real time, and also it can prevent the mill to enter into an erroneous state. Moreover, on study it has found that the present scheme can be
used more accurately in comparison to the earlier work of the author. This paper presents an implementation scheme to use acoustic signal as the control signal to regulate the operation of
a grinding mill.
Evaluation of a Multipoint Equalization System based on Impulse Responses Pro...a3labdsp
This document evaluates a multipoint equalization algorithm that combines fractional octave smoothing of impulse responses measured in multiple locations in a room or car. It investigates how the equalization performance is affected by varying parameters like the number of measurement positions and equalization zone size. The algorithm extracts a representative prototype response and inverse filter from the smoothed impulse responses. Tests show the proposed approach achieves better spectral deviation and equalization than a single-point method, and performance decreases but remains effective as the equalization zone is expanded to more distant positions.
This document provides a mid-term report on the design and analysis of a voltage controlled oscillator (VCO) for a master's project. It discusses completing the circuit diagram, symbol creation, and test circuit simulation in Cadence using 180nm technology. Simulation results so far show the VCO design is one third complete, with layout and performance analysis remaining. The report includes background on VCO metrics like frequency, tuning range, phase noise, and power consumption. It also reviews applications of VCOs in frequency translation and discusses challenges in designing low phase noise CMOS VCOs.
This document describes the Illinois Scan Architecture, a technique for reducing test costs for chips with scan designs. It works by dividing the main scan chain into multiple parallel internal chains, with a single scan input pin. This allows test vectors to be broadcast to all chains simultaneously, reducing test time and data volume by the number of chains with little impact on fault coverage. The document provides experimental data showing reductions in test vectors, cycles, and data volume for several ISCAS circuits using Illinois Scan. It also discusses techniques for further optimizing the technique, such as grouping chains intelligently to minimize the number of scan input pins needed.
A study on improving speaker diarization system = Nghiên cứu phương pháp cải ...Man_Ebook
The document is a master's thesis that explores improving a speaker diarization system by comparing X-Vectors and ECAPA-TDNN embeddings. It includes:
- An introduction outlining the research interest in evaluating these two embedding methods on Vietnamese data sets for speaker verification and diarization tasks.
- Details of the baseline system that uses X-Vector embeddings and the proposed system that replaces it with ECAPA-TDNN embeddings.
- Experiments conducted on private (IPCC_110000, VTR_1350) and public (ZALO_400) data sets to evaluate the systems, along with results showing the proposed system outperforms the baseline.
A study on improving speaker diarization system = Nghiên cứu phương pháp cải ...Man_Ebook
The document is a master's thesis submitted by Tung Lam Nguyen to the Hanoi University of Science and Technology. It studies improving speaker diarization systems by exploring the capabilities of ECAPA-TDNN embeddings versus X-Vector embeddings in a Vietnamese speaker diarization system. The thesis contains experiments evaluating both baseline and proposed systems on speaker verification and speaker diarization tasks using various private and public Vietnamese datasets. The results show that the proposed system using ECAPA-TDNN outperforms the baseline X-Vector system on all tasks and datasets.
This document describes the development of a real-time audio DSP device controlled by an ultrasonic range sensor. The device takes an audio input, applies an echo effect with feedback gain controlled by the range sensor, and outputs the processed audio. Key aspects include:
1) Hardware components like an ARM microcontroller, audio input/output, range sensor, and display are selected and interconnected.
2) Software is developed in C++ to initialize components, average sensor data, implement the echo effect DSP with range-controlled feedback gain, and display information.
3) The device is implemented on a breadboard, tested, and results like noise and distortion are analyzed to improve the design. Further modifications are also
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allows users to log packet and event traces.
A Low Latency Implementation of a Non Uniform Partitioned Overlap and Save Al...a3labdsp
FIR convolution is a widely used operation in digital signal processing field, especially for filtering operations in real time scenarios. In this context, low computationally demanding techniques for calculating convolutions with low input/output latency become essential, considering that the real time requirements are strictly related to the impulse response length. In this paper, a multithreading real time implementation of a Non Uniform Partitioned Overlap and Save algorithm is proposed with the aim of lowering the workload required in applications like reverberation, also exploiting the human ear sensitivity. Several results are reported in order to show the effectiveness of the proposed approach in terms of computational cost, taking into consideration different impulse responses and also introducing comparisons with existing techniques of the state of the art.
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Acoustical Analysis of Active Control in Server Room of C7 Data Centers Colocation Facility
1. Brigham Young University
Acoustical Analysis of Active Control in
the Server Room of a C7 Data Centers
Colocation Facility
Feasibility Report
Advisor:
Dr. Scott Sommerfeldt
C7 Data Centers representative:
Mike Maughan
Group Leaders:
Jesse Daily
James Esplin
Zach Collins
Matthew Shaw
June 21, 2010
3. Feasibility Report
1 Executive Summary
C7 Data Centers provides colocation, virtualization and disaster recovery services and
wanted to explore the possibility of applying active noise control technology to reduce
ambient noise in the data center server room. Active noise control uses loud speakers and
microphones to observe the acoustical environment to emit an “inverse” sound signal to
cancel out the noise. Three methods were used to determine the feasibility of using active
noise control inside the data center server room: measured acoustical data, computational
modeling, and experimental testing.
For the measured data, four measurement locations were considered: inside the data center
entrance door, in front of a C7 Data Centers cage near the center of the room, in a “hot
row” aisle and inside a “cold row” aisle. Two different computational simulations were
performed, the first determined the effect of using a simplified “best-case” scenario model,
and the second used a more complex model. Finally, an experimental test was constructed
to determine if active noise control is feasible for the C7 Data Centers Lindon 5 server
room.
The culmination of our results indicates that active noise control over an extended region
is not possible in the data center room in general. This is due to the random nature of
the noise and the high modal density of the room itself. However, our results indicate that
active noise control may be feasible inside the cold row aisle, given that modest control
(2-4dB) was achieved in laboratory experiments.
2 Introduction
C7 Data Centers has been proactive in seeking out new technologies to make the data
center experience more enjoyable for their customers. Reducing ambient noise on the data
center floor isnt necessarily a need, but more of an interest to see whether something
minimally invasive could be done to counter-act the noise. The C7 Data Centers Lindon
2
4. 5 data center, as is typical of data centers, has little acoustic absorption. The common,
passive forms of acoustical absorption, such as foam or fiberglass panels were not employed
as they contain materials which are highly flammable and give off particulates that could
trigger the early warning fire detection systems.
Active noise control (ANC) was considered as it uses loud speakers and microphones to
observe the acoustical environment and then emits an “inverse” sound signal to attenuate
the noise. Since active control does not use fibrous materials, it may be the ideal solution
for reducing low frequency noise in the server room. To determine if active control is
feasible, acoustic data of the server room was used in concert with computational models
and experimental testing.
3 Methods
Three methods were used to determine the feasibility of using active control in a C7 Data
Centers server room: measured acoustical data, computational modeling, and experimen-
tal testing. Taking acoustical data of the server room yielded valuable information on the
acoustic characteristics of the room and of the noise emitted by the servers and HVAC.
The recorded noise was then used in computational models of the room. These model
simulations were used to estimate the effect that active noise control might have on elim-
inating unwanted sound. Finally, an experimental active control mockup was made to
determine if a physical solution was feasible and to validate or invalidate the computa-
tional model.
3.1 Data Collection
Experimental data of the server room at C7’s Lindon 5 facility was taken to measure the
acoustic characteristics of the room and of the noise emitted by the servers and HVAC.
Four measurement locations were considered (see Fig. 1): by the entrance, in front of a
C7 equipment cage, in a “hot row” and in a “cold row.”
Room Acoustic Measurements
All rooms reverberate sound. As sound reflects off walls and objects within the room, the
sound decays until it cannot be heard. This decay of sound is called the reverberation time
of a room. Reverberation time is found by emitting an impulsive sound (popping a balloon,
firing a starting pistol, etc.) and measuring the time it takes for the sound to decay 60dB
(decibels). This room acoustics measurement is called the T60 time. The Early Decay
3
5. Figure 1: Measurement locations inside the server room.
Time (EDT) is the T60 time calculated by curve fitting the decay curve between 0 and
-10dB.
Reverberation time (or Early Decay Time, EDT) measurements were taken at three loca-
tions (the entrance, the C7 cage and the “cold row”) to determine the acoustic character-
istics of the room. At the entrance to the server room, eight microphones and an energy
density probe were set up to measure the T60 time of the room. A twelve sided loudspeaker
(dodecahedron) was used to ensure that sound was emitted spherically. The dodecahedron
loudspeaker was placed 25’ away from the center of the energy density probe and driven
with a swept sine wave. The Electronic and Acoustic System Evaluation and Response
Analysis (EASERA) was used to measure the EDT for all eight microphones. The micro-
phones were then moved to the “cold row” and the EDT’s were measured again.
Table 1 shows the Early Decay Times (EDT) of the server room as a function of frequency.
The higher the early decay time, the more of a problem a particular frequency range is for
the room. In Table 1, the first two columns show the EDT for the entrance to the server
room with two different source positions (Entrance 1 and Entrance 2). The next two
columns show the remaining EDT data (the C7 Cage and the “Cold Row”).
4
6. Table 1: Early Decay Times (EDT), times in seconds.
Frequency Entrance 1 Entrance 2 C7 Cage Cold Row
File Reference L1-S1-R1 L1-S2-R1 L2-S3-R2 L4-S4-R3
100Hz 1.09 1.48 3.20 0.57
125Hz 1.16 2.28 1.34 0.49
160Hz 1.21 1.28 1.01 0.36
200Hz 1.50 2.25 1.34 0.49
250Hz 1.46 1.55 1.99 0.63
400Hz 2.26 2.06 1.54 0.32
500Hz 1.45 1.73 1.97 0.25
800Hz 1.73 2.23 1.32 0.33
1000Hz 2.06 1.91 1.53 0.37
2000Hz 1.74 1.86 1.37 0.37
4000Hz 1.22 1.10 0.99 0.40
8000Hz 0.54 0.59 0.47 0.31
250-2kHz 1.72 1.89 1.68 0.41
500-4kHz 1.64 1.73 1.50 0.38
An EDT of 3.20 seconds was measured for the 100Hz band by the C7 cage, which indicates
that 100Hz is a reverberant frequency for the C7 Data Centers server room. At 8000Hz,
all of the EDT’s are below 0.6 seconds, meaning that those frequencies are not highly
reverberant.
Noise in the Server Room
Noise is generally defined as unwanted sound. In C7 Data Centers server room, noise can
be defined as the sounds emitted by the many computer fans and HVAC in the server room.
These sources emit sound over a wide range of frequencies due to the turbulent air flow
produced by the fans. Since fans operate at discrete frequencies, certain frequencies have
more power than others. Eliminating these frequencies would greatly reduce the amount
of noise present in the server room.
Measurements of the noise in the server room were taken at three locations. These locations
were chosen for being high foot traffic areas for both technicians and visitors to the data
center. The first location was in front of a C7 equipment cage in the main aisle of the
server room. The second location was in a “hot row”, an aisle between cabinet rows where
the hot air coming out of the servers is directed. Lastly, noise was also recorded in a
“cold row” (see figure 3), an enclosed aisle between cabinet rows that contains the cold air
and directs it into the server cabinets. Figure 2 shows the sound power recorded at the
5
7. Figure 2: Power spectra of unfiltered C7 Data Center server
room noise
Figure 3: Measurements taken
from inside the “cold row”
entrance to C7’s Lindon 5 facility server room. These noise data were later used as input
for computational models and experimental testing.
3.2 Computational Simulations
The server room can be computationally modeled to determine the feasibility of using active
noise control. This model can be used not only to determine the feasibility of active noise
control but also to determine ideal locations and configurations for speaker installation.
The model will consist of ‘primary sources’ emitting the recorded noise taken in the C7
Data Centers Lindon 5 server room and ‘control sources’ trying to actively control the
noise.
Governing Equations
The computational model is based on fundamental acoustic principles. The server noise
was approximated by several point sources emitting fan noise. The wave equation models
the emission of sound pressure waves with a point source that can be defined as
6
8. 2
ˆp −
1
c2
∂2 ˆp
∂t2
= −Q0(t)δ(r − r0), (1)
where Q0(t) is the volume velocity of the point source. The pressure ˆp is the pressure field
in the volume and can be characterized as the sumed response of all the point sources.
ˆp =
Nmax
N=1
qN ψN (2)
The variable qn represents all n point sources that must be solved in the matrix,
(k2
0 − k2
1)C11 + D11 D12 · · ·
D21 (k2
0 − k2
2)C22 + D22 · · ·
...
...
...
·
q1
q2
...
=
− ˆQ0ψ∗
1(r0)
− ˆQ0ψ∗
2(r0)
...
or
A · Q = B, (3)
where ki is the acoustic wave number for each source and
Cmn =
V
ψ∗
mψnd3
x = Λmnδmn (4)
Dmn =
s
(β − βI
)ψ∗
mψnda (5)
Referring back to Eq. (2), the variable ψn represents the eigenfunction solutions in Carte-
sian coordinates which are defined as,
ψN = cos
mπ
Lx
x cos
nπ
Ly
y cos
lπ
Lz
z . (6)
The variables Lx, Ly, Lz are the dimensions of the room.
In Eq. (3), the A matrix can become very large. As the A matrix grows, solving for
the qN ’s becomes increasingly difficult. To simplify the solution process, the off-diagonal
terms in the A matrix can be deleted, effectively decoupling the A matrix and making the
matrix far easier to solve. This code has been used previously with success for smaller
rooms.
7
9. Figure 4: Attenuation of the noise in the “cold row” using the root-sum-squared technique.
Computational Model Results
Two different computational simulations were performed. The first was to determine the
effect of using a “best-case” scenario. The second model created was a more reasonable
control method that used a more complex model.
The first simulation was performed to estimate the maximum achievable potential energy
reduction in the “cold rows” of the server room. The dimensions of the simulated “cold
row” were 10.4×3.4×2.4m. Fifteen thousand modes were used, and control was simulated
every 20Hz from 100Hz to 1kHz. The disturbance sound field was simulated by a uniform
random distribution of 100 sources throughout the “cold row”, each with random phase
and amplitude. Seven control sources were used, the first at the center of the room. The
other six were 1m from the center of the room in the shape of a cube. The control field for
each case was simulated and recorded.
The optimum controller was found for each speaker by minimizing the sum of squared
pressures at 36 simulated locations, uniformly distributed around the center of the “cold
row.” The covered region extended from the center approximately 1m each direction in
length, 0.55m each direction in height, and 0.6m each direction in width. The dB atten-
uation spectrum achieved using control speakers one-at-a-time was recorded. The seven
attenuation spectra were combined using a root-sum-squared averaging technique. The
resulting attenuation achieved over the frequency range studied is shown in Fig. 4.
8
10. Figure 5: Attenuation of the noise in the “cold row” using global potential energy.
As can be seen, control effects in the range of 2-6dB are achievable below 200Hz. As
frequency increases, attenuation remains steady at 1dB. This level of control required 7
simulated speakers and 36 simulated microphones. While this is infeasible in real life, it
does give some idea of the limits of achievable attenuation.
Next, the “cold row” was modeled again by controlling the global potential energy through-
out the entire “cold row” (see Fig. 5).
Global levels drop by 1-3dB below 200Hz, and about 0.5dB above 200Hz. While this
indicates that large-area control is infeasible, it also shows that spillover problems are not
likely very serious.
For the second computational model, a diffuse sound field was created in a room with
slightly absorbent walls. All four previously mentioned locations within the data center
were considered, each at a number of different frequencies. Table 2 shows the frequencies
tested at each location:
For locations 1-3, the size of the room was 28 × 31.7 × 4.1m. The size of the room at
location 4 was 10.4 × 3.3 × 2.4m.
For each measurement location and frequency, a variety of different control methods were
attempted. The number of primary noise sources varied between 1 and 3 while the sec-
ondary control sources were varied between 1 and 6 sources.
For each control method, the attenuated pressure field was found at three orthogonal
9
11. Table 2: Frequencies tested at each location.
Entrance C7 cage Hot row Cold row
356Hz 360Hz 305Hz 360Hz
500Hz 457Hz 455Hz 457Hz
540Hz 540Hz
588Hz
Figure 6: 360Hz, 1 primary source, 1 control source in the x, y and z planes.
planes cutting through the sensor. The attenuated field was found by minimizing the
squared pressure at the sensors and by minimizing the energy density at the sensors (see
Figs. 6 - 8, positive values indicate attenuation). The circles in the figures indicate error
sensor locations.
By controlling squared pressure, the field directly around the sensors is attenuated dra-
matically, but there is not a large region of control. When energy density is minimized at
the sensors, there is more of a global attenuation, but there are still many locations where
the sound level is boosted.
Preliminary runs of the model were not encouraging. After 80 hours of computation, none
of the models had produced any results. This was due to the fact that the solver was
unable to converge and solve the large matrices in the code. To eliminate this problem,
the code was changed to uncouple the point sources. This uncoupling allowed the model
to solve for most frequencies (lower frequencies were more likely to converge).
10
12. Figure 7: 457Hz, 2 primary sources, 2 control sources in the x, y and z planes.
Figure 8: 540Hz, 3 primary sources, 6 control sources in the x, y and z planes.
3.3 Experimental Testing
Methods
An experimental test was constructed to determine if active noise control was feasible for
C7 Data Centers Lindon 5 server room. There are two possible locations for employing
ANC in C7’s Lindon 5 server room: the server room in general, and the cold rows. Ideally,
the experimental ANC would be conducted in rooms of similar dimensions and room rever-
beration times for both the server room and the cold rows. However, rooms with similar
characteristics without extraneous noise were not available at BYU. For simplicity, normal
classes and labs were used to test ANC. While these experimental results were not identical
11
13. to those that would be produced in the actual locations, the results yielded valuable insight
into whether or not ANC was possible.
The first step in determining if ANC is possible was to try to control a simple sine wave.
As a sine wave only has a single frequency, it is the easiest signal to control. If the
ANC failed to control a sine wave then there was little possibility of controlling a more
complex signal such as white noise. The experiment was conducted in U186C in the Eyring
Science Center at Brigham Young University (room dimensions ≈ 4m x 4m x 4m). The
experimental setup consisted of two Mackie loudspeakers. The first loudspeaker acted as
the source, while the second loudspeaker was used as the control. A two-dimensional, four
channel energy density sensor (a sensor that measures both pressure and velocity instead
of just pressure) was used as the error sensor. A 1
4 microphone was used to monitor the
effectiveness of ANC. The microphone was moved away from the reference energy density
sensor to measure how ANC performed as a function of distance.
Results
A sine wave was first controlled to ensure the ANC system was functioning correctly. Two
frequencies were controlled: 100Hz and 250Hz, and the resulting pressure was monitored
at distances of 12.7cm, 25.4cm, 50.8cm, and 121.92cm from the error sensor. The results
can be seen in Table 3. For the 100Hz tone at 12.7cm, active noise control attenuates the
fundamental tone nearly 30dB, and the third harmonic by 20dB. The 100Hz tone at 25.4cm
was attenuated 10dB for the fundamental but was boosted 10dB for the third harmonic.
The overall sound pressure level of each of these tests can be seen in Figs. 9 and 10. For
the 100Hz tone, ANC was able to attenuate the overall signal out to a distance of about
40cm (see Fig. 9). However, for the 250Hz signal, ANC was only able to attenuate the
source signal out to a distance of 20cm before the signal is actually boosted by as much as
4dB (see Fig. 10). These results are typical of what can be expected for ANC in a room
of similar dimensions.
12
14. Figure 9: Overall sound pressure level for 100Hz sine wave as a function of distance.
Figure 10: Overall sound pressure level for 250Hz sine wave as a function of distance.
The noise from the server room was then output as the source to be controlled. The noise
was controlled at distances of 1cm, 12.7cm, 25.4cm, 50.8cm, and 121.92cm. The results
can be seen in Fig. 11. It can be seen that for all distances, ANC was able to attenuate
the main tone at 40Hz by several dB. Above 100Hz ANC contributes very little, neither
attenuating nor boosting the signal. These results were then processed to visualize the
13
15. Table 3: ANC results for sine waves at 100Hz and 250Hz. Results measured at 12.7cm,
25.4cm, 50.8cm, and 121.92cm.
100Hz 250Hz
12.7cm26.4cm50.8cm121.92cm
overall sound pressure level (see Fig. 12). The ANC results for the noise from the server
room were promising: 2-4dB of attenuation was achieved out to a distance of 120cm (see
Fig. 11). Three comments need to be made about these results.
1. The experimental setup used only a single source to output the recorded noise from
14
16. the server room. In reality, there will be many sources of noise in the server room
and in the “cold rows.”
2. The experimental setup used only one control source to implement ANC in the room.
Since this was just an experimental mockup, only one control source was attempted.
If ANC were implemented in the server room, multiple control sources could be used
to perform ANC.
3. The aspect ratio of the test room was approximately one. This means that the modal
density in the room increases rapidly with the room dimensions. Rooms having high
modal densities are more difficult to control. The “cold rows” have a much higher
aspect ratio (long and thin), meaning that they will have fewer modes present in the
room.
4 Conclusion
4.1 Summary
Three aspects of engineering were considered in characterizing the C7 Data Centers Lindon
5 server room noise: taking experimental data, running computational simulations, and
mockup testing. The first step was taking the acoustic data present at the server room.
These data were crucial in determining what frequencies contained the most energy and
were integral for use in the computational models and the experimental mockups. Data
were taken in 4 locations in the C7 Data Centers Lindon 5 server room, including data
that were taken in the “cold rows.”
Computational simulations were performed to model the use of active noise control in the
server room. The models used were originally designed by Buye Xu for modeling modal
analysis in an enclosed 3-D space. This model was used to simulate both the “cold rows”
and the entire server room. Initially, the solvers in the code struggled to converge. The
codes were modified to decouple all the point sources, which makes the simulations less real-
istic, but simplified the matrices generated during the simulation. Multiple frequencies were
tested with multiple control sources. Results were not encouraging. Computational results
show that almost no control is possible in the server room or in the “cold rows.”
Finally, an experimental mockup was set up to attempt active noise control in a controlled
laboratory setting. A sine wave was output as a trial signal to test the active noise control
system. Once a sine wave was controlled in a laboratory setting, the recorded noise from the
C7 Data Centers Lindon 5 server room was output through a loudspeaker. This recorded
signal was then controlled in the lab using ANC with some attenuation achieved.
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17. Figure 11: ANC results for server room noise. Results measured at 1cm, 12.7cm, 25.4cm,
50.8cm, and 121.92cm.
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18. Figure 12: Overall sound pressure level for server room noise as a function of distance.
4.2 Conclusions
Data were taken at C7 Data centers Lindon 5 facility. These were used to calculate the
early decay times of the main server room and the “cold rows.” Results show a significant
amount of sound power below 500Hz and EDT’s between 0.25-3.20 seconds depending on
location. These data were also used in both the computational models and the experimental
testing of active noise control.
While useful, the computational model had some severe drawbacks. The rooms input into
this model were on the order of a few meters to several dozen meters. The code was run
on BYU’s supercomputer, Mary Lou, but after 80 hours of computation time, none of
the coupled codes were able to run to completion. Only after decoupling the code were
a few of the codes able to run to completion for a few select frequencies. The results of
these simulations indicate that ANC is not effective. However, these results are not in full
agreement with the results of the experimental testing done in the lab.
The experimental testing done using the measurements from the C7 Data Centers Lindon
5 server room indicate that active noise control is able to attenuate the noise by 2-4dB
and may therefore be a viable option. These results were verified by first controlling a
simple sine wave. Given the difficulties encountered with the computational code, the
experimental testing is considered to yield the more reliable results.
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19. 4.3 Recommendation
The initial plan for implementing active noise control in the C7 Data Centers Lindon 5
facility was at two locations, in the server room, and in the “cold rows.” The cumulation
of our results indicates that active noise control will not be effective in the main area of
the server room (this includes the Entrance and the C7 Cage area). This is due to the
random nature of the noise and the high modal density present in the main room. However
our results indicate that active noise control may be possible in the “cold rows.” Since
active noise control was possible in the laboratory experiments, it can be said that similar
control within the “cold rows” may be feasible, but more elaborate experiments should be
performed to ensure that active noise control can yield sufficient acoustic control.
5 Contact Information
Dr. Scott Sommerfeldt 1
Dean, College of Physical and Mathematical Sciences
phone: 801.422.2205
email: scott sommerfeldt@byu.edu
Jesse Daily
Graduate student
phone: 925.321.4815
email: dai01001@gmail.com
Mike Maughan
C7 Data Centers
email: mikem@c7dc.com
Contributing Students: Zach Collins, Jesse Daily, James Esplin, Jarom Giraud, David
Krueger, Dan Manwill, Matt Shaw, Brad Solomon, Dan Tengelsen, Alan Wall, Buye
Xu
1
Corresponding author
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