Power Consumption of Digital Hearing Aid Computations Using Customized Numerical Representations

We investigate the impact of numerical representation on the power consumption of digital hearing aids. A fundamental building block, a non-linear amplifier, is implemented using traditional 16-bit linear or customized 9-bit logarithmic and 10-bit floating point numerical representations. An individual channel of a multi-channel hearing aid is constructed, targeting both FPGA and ASIC deployment options. Using signal transition counts in the post-synthesis simulation to model power consumption, we compare the relative power consumption of the non-linear amplifiers, a full hearing aid channel, and the complete hearing aid signal processing for these three numerical representations. Our results show that for the non-linear amplifier, the logarithmic and floating-point representations provide significant savings over a traditional linear representation. However, since the total power consumption is dominated by the FIR filters, the total power saving is on the order of the filters

Reducing Power Consumption Using Customized Numerical Representations in Digital Hearing Aids

This thesis examines the effects of changing the numerical representation of audio signals in digital hearing aids to minimize power consumption. Within the hearing aid design a majority of the power used is consumed in the many finite impulse response filters. The main processing involved in these filters is a multiply-accumulate function. We examine the power consumption of 12 different multiply-accumulate units that use the following numerical representations: a 16-bit linear representation, a 9-bit logarithmic representation, and 10 different floating-point rep-representations ranging from 9 to 13 bits. A selection of the multiply-accumulators are simulated using a continuous-circuit simulator. The power estimates from this are compared with signal transition counts from a discrete event simulator to quantify the relationship between transition counts and power consumption. This relationship is then used to examine other numerical representations

Toward a Robust Sparse Data Representation for Wireless Sensor Networks

Compressive sensing has been successfully used for optimized operations in wireless sensor networks. However, raw data collected by sensors may be neither originally sparse nor easily transformed into a sparse data representation. This paper addresses the problem of transforming source data collected by sensor nodes into a sparse representation with a few nonzero elements. Our contributions that address three major issues include: 1) an effective method that extracts population sparsity of the data, 2) a sparsity ratio guarantee scheme, and 3) a customized learning algorithm of the sparsifying dictionary. We introduce an unsupervised neural network to extract an intrinsic sparse coding of the data. The sparse codes are generated at the activation of the hidden layer using a sparsity nomination constraint and a shrinking mechanism. Our analysis using real data samples shows that the proposed method outperforms conventional sparsity-inducing methods.Comment: 8 page

Energy-Efficient Recurrent Neural Network Accelerators for Real-Time Inference

Over the past decade, Deep Learning (DL) and Deep Neural Network (DNN) have gone through a rapid development. They are now vastly applied to various applications and have profoundly changed the life of hu- man beings. As an essential element of DNN, Recurrent Neural Networks (RNN) are helpful in processing time-sequential data and are widely used in applications such as speech recognition and machine translation. RNNs are difficult to compute because of their massive arithmetic operations and large memory footprint. RNN inference workloads used to be executed on conventional general-purpose processors including Central Processing Units (CPU) and Graphics Processing Units (GPU); however, they have un- necessary hardware blocks for RNN computation such as branch predictor, caching system, making them not optimal for RNN processing. To accelerate RNN computations and outperform the performance of conventional processors, previous work focused on optimization methods on both software and hardware. On the software side, previous works mainly used model compression to reduce the memory footprint and the arithmetic operations of RNNs. On the hardware side, previous works also designed domain-specific hardware accelerators based on Field Pro- grammable Gate Arrays (FPGA) or Application Specific Integrated Circuits (ASIC) with customized hardware pipelines optimized for efficient pro- cessing of RNNs. By following this software-hardware co-design strategy, previous works achieved at least 10X speedup over conventional processors. Many previous works focused on achieving high throughput with a large batch of input streams. However, in real-time applications, such as gaming Artificial Intellegence (AI), dynamical system control, low latency is more critical. Moreover, there is a trend of offloading neural network workloads to edge devices to provide a better user experience and privacy protection. Edge devices, such as mobile phones and wearable devices, are usually resource-constrained with a tight power budget. They require RNN hard- ware that is more energy-efficient to realize both low-latency inference and long battery life. Brain neurons have sparsity in both the spatial domain and time domain. Inspired by this human nature, previous work mainly explored model compression to induce spatial sparsity in RNNs. The delta network algorithm alternatively induces temporal sparsity in RNNs and can save over 10X arithmetic operations in RNNs proven by previous works. In this work, we have proposed customized hardware accelerators to exploit temporal sparsity in Gated Recurrent Unit (GRU)-RNNs and Long Short-Term Memory (LSTM)-RNNs to achieve energy-efficient real-time RNN inference. First, we have proposed DeltaRNN, the first-ever RNN accelerator to exploit temporal sparsity in GRU-RNNs. DeltaRNN has achieved 1.2 TOp/s effective throughput with a batch size of 1, which is 15X higher than its related works. Second, we have designed EdgeDRNN to accelerate GRU-RNN edge inference. Compared to DeltaRNN, EdgeDRNN does not rely on on-chip memory to store RNN weights and focuses on reducing off-chip Dynamic Random Access Memory (DRAM) data traffic using a more scalable architecture. EdgeDRNN have realized real-time inference of large GRU-RNNs with submillisecond latency and only 2.3 W wall plug power consumption, achieving 4X higher energy efficiency than commercial edge AI platforms like NVIDIA Jetson Nano. Third, we have used DeltaRNN to realize the first-ever continuous speech recognition sys- tem with the Dynamic Audio Sensor (DAS) as the front-end. The DAS is a neuromorphic event-driven sensor that produces a stream of asyn- chronous events instead of audio data sampled at a fixed sample rate. We have also showcased how an RNN accelerator can be integrated with an event-driven sensor on the same chip to realize ultra-low-power Keyword Spotting (KWS) on the extreme edge. Fourth, we have used EdgeDRNN to control a powered robotic prosthesis using an RNN controller to replace a conventional proportional–derivative (PD) controller. EdgeDRNN has achieved 21 μs latency of running the RNN controller and could maintain stable control of the prosthesis. We have used DeltaRNN and EdgeDRNN to solve these problems to prove their value in solving real-world problems. Finally, we have applied the delta network algorithm on LSTM-RNNs and have combined it with a customized structured pruning method, called Column-Balanced Targeted Dropout (CBTD), to induce spatio-temporal sparsity in LSTM-RNNs. Then, we have proposed another FPGA-based accelerator called Spartus, the first RNN accelerator that exploits spatio- temporal sparsity. Spartus achieved 9.4 TOp/s effective throughput with a batch size of 1, the highest among present FPGA-based RNN accelerators with a power budget around 10 W. Spartus can complete the inference of an LSTM layer having 5 million parameters within 1 μs

A Survey on Design Methodologies for Accelerating Deep Learning on Heterogeneous Architectures

In recent years, the field of Deep Learning has seen many disruptive and impactful advancements. Given the increasing complexity of deep neural networks, the need for efficient hardware accelerators has become more and more pressing to design heterogeneous HPC platforms. The design of Deep Learning accelerators requires a multidisciplinary approach, combining expertise from several areas, spanning from computer architecture to approximate computing, computational models, and machine learning algorithms. Several methodologies and tools have been proposed to design accelerators for Deep Learning, including hardware-software co-design approaches, high-level synthesis methods, specific customized compilers, and methodologies for design space exploration, modeling, and simulation. These methodologies aim to maximize the exploitable parallelism and minimize data movement to achieve high performance and energy efficiency. This survey provides a holistic review of the most influential design methodologies and EDA tools proposed in recent years to implement Deep Learning accelerators, offering the reader a wide perspective in this rapidly evolving field. In particular, this work complements the previous survey proposed by the same authors in [203], which focuses on Deep Learning hardware accelerators for heterogeneous HPC platforms

Ultra low-power, high-performance accelerator for speech recognition

Automatic Speech Recognition (ASR) is undoubtedly one of the most important and interesting applications in the cutting-edge era of Deep-learning deployment, especially in the mobile segment. Fast and accurate ASR comes at a high energy cost, requiring huge memory storage and computational power, which is not affordable for the tiny power budget of mobile devices. Hardware acceleration can reduce power consumption of ASR systems as well as reducing its memory pressure, while delivering high-performance. In this thesis, we present a customized accelerator for large-vocabulary, speaker-independent, continuous speech recognition. A state-of-the-art ASR system consists of two major components: acoustic-scoring using DNN and speech-graph decoding using Viterbi search. As the first step, we focus on the Viterbi search algorithm, that represents the main bottleneck in the ASR system. The accelerator includes some innovative techniques to improve the memory subsystem, which is the main bottleneck for performance and power, such as a prefetching scheme and a novel bandwidth saving technique tailored to the needs of ASR. Furthermore, as the speech graph is vast taking more than 1-Gigabyte memory space, we propose to change its representation by partitioning it into several sub-graphs and perform an on-the-fly composition during the Viterbi run-time. This approach together with some simple yet efficient compression techniques result in 31x memory footprint reduction, providing 155x real-time speedup and orders of magnitude power and energy saving compared to CPUs and GPUs. In the next step, we propose a novel hardware-based ASR system that effectively integrates a DNN accelerator for the pruned/quantized models with the Viterbi accelerator. We show that, when either pruning or quantizing the DNN model used for acoustic scoring, ASR accuracy is maintained but the execution time of the ASR system is increased by 33%. Although pruning and quantization improves the efficiency of the DNN, they result in a huge increase of activity in the Viterbi search since the output scores of the pruned model are less reliable. In order to avoid the aforementioned increase in Viterbi search workload, our system loosely selects the N-best hypotheses at every time step, exploring only the N most likely paths. Our final solution manages to efficiently combine both DNN and Viterbi accelerators using all their optimizations, delivering 222x real-time ASR with a small power budget of 1.26 Watt, small memory footprint of 41 MB, and a peak memory bandwidth of 381 MB/s, being amenable for low-power mobile platforms.Los sistemas de reconocimiento automático del habla (ASR por sus siglas en inglés, Automatic Speech Recognition) son sin lugar a dudas una de las aplicaciones más relevantes en el área emergente de aprendizaje profundo (Deep Learning), specialmente en el segmento de los dispositivos móviles. Realizar el reconocimiento del habla de forma rápida y precisa tiene un elevado coste en energía, requiere de gran capacidad de memoria y de cómputo, lo cual no es deseable en sistemas móviles que tienen severas restricciones de consumo energético y disipación de potencia. El uso de arquitecturas específicas en forma de aceleradores hardware permite reducir el consumo energético de los sistemas de reconocimiento del habla, al tiempo que mejora el rendimiento y reduce la presión en el sistema de memoria. En esta tesis presentamos un acelerador específicamente diseñado para sistemas de reconocimiento del habla de gran vocabulario, independientes del orador y que funcionan en tiempo real. Un sistema de reconocimiento del habla estado del arte consiste principalmente en dos componentes: el modelo acústico basado en una red neuronal profunda (DNN, Deep Neural Network) y la búsqueda de Viterbi basada en un grafo que representa el lenguaje. Como primer objetivo nos centramos en la búsqueda de Viterbi, ya que representa el principal cuello de botella en los sistemas ASR. El acelerador para el algoritmo de Viterbi incluye técnicas innovadoras para mejorar el sistema de memoria, que es el mayor cuello de botella en rendimiento y energía, incluyendo técnicas de pre-búsqueda y una nueva técnica de ahorro de ancho de banda a memoria principal específicamente diseñada para sistemas ASR. Además, como el grafo que representa el lenguaje requiere de gran capacidad de almacenamiento en memoria (más de 1 GB), proponemos cambiar su representación y dividirlo en distintos grafos que se componen en tiempo de ejecución durante la búsqueda de Viterbi. De esta forma conseguimos reducir el almacenamiento en memoria principal en un factor de 31x, alcanzar un rendimiento 155 veces superior a tiempo real y reducir el consumo energético y la disipación de potencia en varios órdenes de magnitud comparado con las CPUs y las GPUs. En el siguiente paso, proponemos un novedoso sistema hardware para reconocimiento del habla que integra de forma efectiva un acelerador para DNNs podadas y cuantizadas con el acelerador de Viterbi. Nuestros resultados muestran que podar y/o cuantizar el DNN para el modelo acústico permite mantener la precisión pero causa un incremento en el tiempo de ejecución del sistema completo de hasta el 33%. Aunque podar/cuantizar mejora la eficiencia del DNN, éstas técnicas producen un gran incremento en la carga de trabajo de la búsqueda de Viterbi ya que las probabilidades calculadas por el DNN son menos fiables, es decir, se reduce la confianza en las predicciones del modelo acústico. Con el fin de evitar un incremento inaceptable en la carga de trabajo de la búsqueda de Viterbi, nuestro sistema restringe la búsqueda a las N hipótesis más probables en cada paso de la búsqueda. Nuestra solución permite combinar de forma efectiva un acelerador de DNNs con un acelerador de Viterbi incluyendo todas las optimizaciones de poda/cuantización. Nuestro resultados experimentales muestran que dicho sistema alcanza un rendimiento 222 veces superior a tiempo real con una disipación de potencia de 1.26 vatios, unos requisitos de memoria modestos de 41 MB y un uso de ancho de banda a memoria principal de, como máximo, 381 MB/s, ofreciendo una solución adecuada para dispositivos móviles

CREW: Computation reuse and efficient weight storage for hardware-accelerated MLPs and RNNs

Deep Neural Networks (DNNs) have achieved tremendous success for cognitive applications. The core operation in a DNN is the dot product between quantized inputs and weights. Prior works exploit the weight/input repetition that arises due to quantization to avoid redundant computations in Convolutional Neural Networks (CNNs). However, in this paper we show that their effectiveness is severely limited when applied to FullyConnected (FC) layers, which are commonly used in state-of-the-art DNNs, as it is the case of modern Recurrent Neural Networks (RNNs) and Transformer models. To improve energy-efficiency of FC computation we present CREW, a hardware accelerator that implements Computation Reuse and an Efficient Weight Storage mechanism to exploit the large number of repeated weights in FC layers. CREW first performs the multiplications of the unique weights by their respective inputs and stores the results in an on-chip buffer. The storage requirements are modest due to the small number of unique weights and the relatively small size of the input compared to convolutional layers. Next, CREW computes each output by fetching and adding its required products. To this end, each weight is replaced offline by an index in the buffer of unique products. Indices are typically smaller than the quantized weights, since the number of unique weights for each input tends to be much lower than the range of quantized weights, which reduces storage and memory bandwidth requirements. Overall, CREW greatly reduces the number of multiplications and provides significant savings in model memory footprint and memory bandwidth usage. We evaluate CREW on a diverse set of modern DNNs. On average, CREW provides 2.61x speedup and 2.42x energy savings over a TPU-like accelerator. Compared to UCNN, a state-of-art computation reuse technique, CREW achieves 2.10x speedup and 2.08x energy savings on average.This work has been supported by the CoCoUnit ERC Advanced Grant of the EU’s Horizon 2020 program (grant No 833057), the Spanish State Research Agency (MCIN/AEI) under grant PID2020-113172RB-I00, the ICREA Academia program, and the Spanish Ministry of Education under grant FPU15/02294.Peer ReviewedPostprint (published version

Deep Learning for Mobile Multimedia: A Survey

Deep Learning (DL) has become a crucial technology for multimedia computing. It offers a powerful instrument to automatically produce high-level abstractions of complex multimedia data, which can be exploited in a number of applications, including object detection and recognition, speech-to- text, media retrieval, multimodal data analysis, and so on. The availability of affordable large-scale parallel processing architectures, and the sharing of effective open-source codes implementing the basic learning algorithms, caused a rapid diffusion of DL methodologies, bringing a number of new technologies and applications that outperform, in most cases, traditional machine learning technologies. In recent years, the possibility of implementing DL technologies on mobile devices has attracted significant attention. Thanks to this technology, portable devices may become smart objects capable of learning and acting. The path toward these exciting future scenarios, however, entangles a number of important research challenges. DL architectures and algorithms are hardly adapted to the storage and computation resources of a mobile device. Therefore, there is a need for new generations of mobile processors and chipsets, small footprint learning and inference algorithms, new models of collaborative and distributed processing, and a number of other fundamental building blocks. This survey reports the state of the art in this exciting research area, looking back to the evolution of neural networks, and arriving to the most recent results in terms of methodologies, technologies, and applications for mobile environments