Energy-Efficient Neural Network Architectures

Emerging systems for artificial intelligence (AI) are expected to rely on deep neural networks (DNNs) to achieve high accuracy for a broad variety of applications, including computer vision, robotics, and speech recognition. Due to the rapid growth of network size and depth, however, DNNs typically result in high computational costs and introduce considerable power and performance overheads. Dedicated chip architectures that implement DNNs with high energy efficiency are essential for adding intelligence to interactive edge devices, enabling them to complete increasingly sophisticated tasks by extending battery lie. They are also vital for improving performance in cloud servers that support demanding AI computations. This dissertation focuses on architectures and circuit technologies for designing energy-efficient neural network accelerators. First, a deep-learning processor is presented for achieving ultra-low power operation. Using a heterogeneous architecture that includes a low-power always-on front-end and a selectively-enabled high-performance back-end, the processor dynamically adjusts computational resources at runtime to support conditional execution in neural networks and meet performance targets with increased energy efficiency. Featuring a reconfigurable datapath and a memory architecture optimized for energy efficiency, the processor supports multilevel dynamic activation of neural network segments, performing object detection tasks with 5.3x lower energy consumption in comparison with a static execution baseline. Fabricated in 40nm CMOS, the processor test-chip dissipates 0.23mW at 5.3 fps. It demonstrates energy scalability up to 28.6 TOPS/W and can be configured to run a variety of workloads, including severely power-constrained ones such as always-on monitoring in mobile applications. To further improve the energy efficiency of the proposed heterogeneous architecture, a new charge-recovery logic family, called zero-short-circuit current (ZSCC) logic, is proposed to decrease the power consumption of the always-on front-end. By relying on dedicated circuit topologies and a four-phase clocking scheme, ZSCC operates with significantly reduced short-circuit currents, realizing order-of-magnitude power savings at relatively low clock frequencies (in the order of a few MHz). The efficiency and applicability of ZSCC is demonstrated through an ANSI S1.11 1/3 octave filter bank chip for binaural hearing aids with two microphones per ear. Fabricated in a 65nm CMOS process, this charge-recovery chip consumes 13.8µW with a 1.75MHz clock frequency, achieving 9.7x power reduction per input in comparison with a 40nm monophonic single-input chip that represents the published state of the art. The ability of ZSCC to further increase the energy efficiency of the heterogeneous neural network architecture is demonstrated through the design and evaluation of a ZSCC-based front-end. Simulation results show 17x power reduction compared with a conventional static CMOS implementation of the same architecture.PHDElectrical and Computer EngineeringUniversity of Michigan, Horace H. Rackham School of Graduate Studieshttps://deepblue.lib.umich.edu/bitstream/2027.42/147614/1/hsiwu_1.pd

A voice activity detection algorithm with sub-band detection based on time-frequency characteristics of mandarin

Voice activity detection algorithms are widely used in the areas of voice compression, speech synthesis, speech recognition, speech enhancement, and etc. In this paper, an efficient voice activity detection algorithm with sub-band detection based on time-frequency characteristics of mandarin is proposed. The proposed sub-band detection consists of two parts: crosswise detection and lengthwise detection. Energy detection and pitch detection are in the range of considerations. For a better performance, double-threshold criterion is used to reduce the misjudgment rate of the detection. Performance evaluation is based on six noise environments with different SNRs. Experiment results indicate that the proposed algorithm can detect the area of voice effectively in non-stationary environment and low SNR environment and has the potential to progress

Characterization of Impulse Noise and Hazard Analysis of Impulse Noise Induced Hearing Loss using AHAAH Modeling

Millions of people across the world are suffering from noise induced hearing loss (NIHL), especially under working conditions of either continuous Gaussian or non-Gaussian noise that might affect human\u27s hearing function. Impulse noise is a typical non-Gaussian noise exposure in military and industry, and generates severe hearing loss problem. This study mainly focuses on characterization of impulse noise using digital signal analysis method and prediction of the auditory hazard of impulse noise induced hearing loss by the Auditory Hazard Assessment Algorithm for Humans (AHAAH) modeling. A digital noise exposure system has been developed to produce impulse noises with peak sound pressure level (SPL) up to 160 dB. The characterization of impulse noise generated by the system has been investigated and analyzed in both time and frequency domains. Furthermore, the effects of key parameters of impulse noise on auditory risk unit (ARU) are investigated using both simulated and experimental measured impulse noise signals in the AHAAH model. The results showed that the ARUs increased monotonically with the peak pressure (both P+ and P-) increasing. With increasing of the time duration, the ARUs increased first and then decreased, and the peak of ARUs appeared at about t = 0.2 ms (for both t+ and t-). In addition, the auditory hazard of experimental measured impulse noises signals demonstrated a monotonically increasing relationship between ARUs and system voltages

Town of Dixmont Maine Ordinances

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Ultra-low-power circuits and systems for wearable and implantable medical devices

Thesis (Ph. D.)--Massachusetts Institute of Technology, Department of Electrical Engineering and Computer Science, 2013.Cataloged from PDF version of thesis.Includes bibliographical references (pages 219-231).Advances in circuits, sensors, and energy storage elements have opened up many new possibilities in the health industry. In the area of wearable devices, the miniaturization of electronics has spurred the rapid development of wearable vital signs, activity, and fitness monitors. Maximizing the time between battery recharge places stringent requirements on power consumption by the device. For implantable devices, the situation is exacerbated by the fact that energy storage capacity is limited by volume constraints, and frequent battery replacement via surgery is undesirable. In this case, the design of energy-efficient circuits and systems becomes even more crucial. This thesis explores the design of energy-efficient circuits and systems for two medical applications. The first half of the thesis focuses on the design and implementation of an ultra-low-power, mixed-signal front-end for a wearable ECG monitor in a 0.18pm CMOS process. A mixed-signal architecture together with analog circuit optimizations enable ultra-low-voltage operation at 0.6V which provides power savings through voltage scaling, and ensures compatibility with state-of-the-art DSPs. The fully-integrated front-end consumes just 2.9[mu]W, which is two orders of magnitude lower than commercially available parts. The second half of this thesis focuses on ultra-low-power system design and energy-efficient neural stimulation for a proof-of-concept fully-implantable cochlear implant. First, implantable acoustic sensing is demonstrated by sensing the motion of a human cadaveric middle ear with a piezoelectric sensor. Second, alternate energy-efficient electrical stimulation waveforms are investigated to reduce neural stimulation power when compared to the conventional rectangular waveform. The energy-optimal waveform is analyzed using a computational nerve fiber model, and validated with in-vivo ECAP recordings in the auditory nerve of two cats and with psychophysical tests in two human cochlear implant users. Preliminary human subject testing shows that charge and energy savings of 20-30% and 15-35% respectively are possible with alternative waveforms. A system-on-chip comprising the sensor interface, reconfigurable sound processor, and arbitrary-waveform neural stimulator is implemented in a 0.18[mu]m high-voltage CMOS process to demonstrate the feasibility of this system. The sensor interface and sound processor consume just 12[mu]W of power, representing just 2% of the overall system power which is dominated by stimulation. As a result, the energy savings from using alternative stimulation waveforms transfer directly to the system.by Marcus Yip.Ph.D

Applications of loudness models in audio engineering

This thesis investigates the application of perceptual models to areas of audio engineering, with a particular focus on music production. The goal was to establish efficient and practical tools for the measurement and control of the perceived loudness of musical sounds. Two types of loudness model were investigated: the single-band model and the multiband excitation pattern (EP) model. The heuristic single-band devices were designed to be simple but sufficiently effective for real-world application, whereas the multiband procedures were developed to give a reasonable account of a large body of psychoacoustic findings according to a functional model of the peripheral hearing system. The research addresses the extent to which current models of loudness generalise to musical instruments, and whether can they be successfully employed in music applications. The domain-specific disparity between the two types of model was first tackled by reducing the computational load of state-of-the-art EP models to allow for fast but low-error auditory signal processing. Two elaborate hearing models were analysed and optimised using musical instruments and speech as test stimuli. It was shown that, after significantly reducing the complexity of both procedures, estimates of global loudness, such as peak loudness, as well as the intermediate auditory representations can be preserved with high accuracy. Based on the optimisations, two real-time applications were developed: a binaural loudness meter and an automatic multitrack mixer. This second system was designed to work independently of the loudness measurement procedure, and therefore supports both linear and nonlinear models. This allowed for a single mixing device to be assessed using different loudness metrics and this was demonstrated by evaluating three configurations through subjective assessment. Unexpectedly, when asked to rate both the overall quality of a mix and the degree to which instruments were equally loud, listeners preferred mixes generated using heuristic single-band models over those produced using a multiband procedure. A series of more systematic listening tests were conducted to further investigate this finding. Subjective loudness matches of musical instruments commonly found in western popular music were collected to evaluate the performance of five published models. The results were in accord with the application-based assessment, namely that current EP procedures do not generalise well when estimating the relative loudness of musical sounds which have marked differences in spectral content. Model specific issues were identified relating to the calculation of spectral loudness summation (SLS) and the method used to determine the global-loudness percept of time-varying musical sounds; associated refinements were proposed. It was shown that a new multiband loudness model with a heuristic loudness transformation yields superior performance over existing methods. This supports the idea that a revised model of SLS is needed, and therefore that modification to this stage in existing psychoacoustic procedures is an essential step towards the goal of achieving real-world deployment

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Mechanical systems readiness assessment and performance monitoring study

The problem of mechanical devices which lack the real-time readiness assessment and performance monitoring capability required for future space missions is studied. The results of a test program to establish the feasibility of implementing structure borne acoustics, a nondestructive test technique, are described. The program included the monitoring of operational acoustic signatures of five separate mechanical components, each possessing distinct sound characteristics. Acoustic signatures were established for normal operation of each component. Critical failure modes were then inserted into the test components, and faulted acoustic signatures obtained. Predominant features of the sound signature were related back to operational events occurring within the components both for normal and failure mode operations. All of these steps can be automated. The structure borne acoustics technique lends itself to reducing checkout time, simplifying maintenance procedures, and reducing manual involvement in the checkout, operation, maintenance, and fault diagnosis of mechanical systems