E-PUR: An Energy-Efficient Processing Unit for Recurrent Neural Networks

Recurrent Neural Networks (RNNs) are a key technology for emerging applications such as automatic speech recognition, machine translation or image description. Long Short Term Memory (LSTM) networks are the most successful RNN implementation, as they can learn long term dependencies to achieve high accuracy. Unfortunately, the recurrent nature of LSTM networks significantly constrains the amount of parallelism and, hence, multicore CPUs and many-core GPUs exhibit poor efficiency for RNN inference. In this paper, we present E-PUR, an energy-efficient processing unit tailored to the requirements of LSTM computation. The main goal of E-PUR is to support large recurrent neural networks for low-power mobile devices. E-PUR provides an efficient hardware implementation of LSTM networks that is flexible to support diverse applications. One of its main novelties is a technique that we call Maximizing Weight Locality (MWL), which improves the temporal locality of the memory accesses for fetching the synaptic weights, reducing the memory requirements by a large extent. Our experimental results show that E-PUR achieves real-time performance for different LSTM networks, while reducing energy consumption by orders of magnitude with respect to general-purpose processors and GPUs, and it requires a very small chip area. Compared to a modern mobile SoC, an NVIDIA Tegra X1, E-PUR provides an average energy reduction of 92x

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

Combining representations for improved sketch recognition

Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2009.Cataloged from PDF version of thesis.Includes bibliographical references (p. 89-96).Sketching is a common means of conveying, representing, and preserving information, and it has become a subject of research as a method for human-computer interaction, specifically in the area of computer-aided design. Digitally collected sketches contain both spatial and temporal information; additionally, they may contain a conceptual structure of shapes and sub shapes. These multiple aspects suggest several ways of representing sketches, each with advantages and disadvantages for recognition. Most existing sketch recognitions systems are based on a single representation and do not use all available information. We propose combining several representations and systems as a way to improve recognition accuracy. This thesis presents two methods for combining recognition systems. The first improves recognition by improving segmentation, while the second seeks to predict how well systems will recognize a given domain or symbol and combine their outputs accordingly. We show that combining several recognition systems based on different representations can improve the accuracy of existing recognition methods.by Sonya J. Cates.Ph.D

Multimodal feature extraction and fusion for audio-visual speech recognition

Multimodal signal processing analyzes a physical phenomenon through several types of measures, or modalities. This leads to the extraction of higher-quality and more reliable information than that obtained from single-modality signals. The advantage is two-fold. First, as the modalities are usually complementary, the end-result of multimodal processing is more informative than for each of the modalities individually, which represents the first advantage. This is true in all application domains: human-machine interaction, multimodal identification or multimodal image processing. The second advantage is that, as modalities are not always reliable, it is possible, when one modality becomes corrupted, to extract the missing information from the other one. There are two essential challenges in multimodal signal processing. First, the features used from each modality need to be as relevant and as few as possible. The fact that multimodal systems have to process more than just one modality means that they can run into errors caused by the curse of dimensionality much more easily than mono-modal ones. The curse of dimensionality is a term used essentially to say that the number of equally-distributed samples required to cover a region of space grows exponentially with the dimensionality of the space. This has important implications in the classification domain, since accurate models can only be obtained if an adequate number of samples is available, and obviously this required number of samples grows with the dimensionality of the features. Dimensionality reduction is thus a necessary step in any application dealing with complex signals, and this is achieved through selection, transforms or the combination of the two. The second essential challenge is multimodal integration. Since the signals involved do not necessarily have the same data rate, range or even dimensionality, combining information coming from such different sources is not straightforward. This can be done at different levels, starting from the basic signal level by combining the signals themselves, if they are compatible, up to the highest decision level, where only the individual decisions taken based on the signals are combined. Ideally, the fusion method should allow temporal variations in the relative importance of the two streams, to account for possible changes in their quality. However, this can only be done with methods operating at a high decision level. The aim of this thesis is to offer solutions to both these challenges, in the context of audio-visual speech recognition and speaker localization. Both these applications are from the field of human-machine interaction. Audio-visual speech recognition aims to improve the accuracy of speech recognizers by augmenting the audio with information extracted from the video, more particularly, the movement of the speaker's lips. This works well especially when the audio is corrupted, leading in this case to significant gains in accuracy. Speaker localization means detecting who is the active speaker in a audio-video sequence containing several persons, something that is useful for videoconferencing and the automated annotation of meetings. These two applications are the context in which we present our solutions to both feature selection and multimodal integration. First, we show how informative features can be extracted from the visual modality, using an information-theoretic framework which gives us a quantitative measure of the relevance of individual features. We also prove that reducing redundancy between these features is important for avoiding the curse of dimensionality and improving recognition results. The methods that we present are novel in the field of audio-visual speech recognition and we found that their use leads to significant improvements compared to the state of the art. Second, we present a method of multimodal fusion at the level of intermediate decisions using a weight for each of the streams. The weights are adaptive, changing according to the estimated reliability of each stream. This makes the system tolerant to changes in the quality of either stream, and even to the temporary interruption of one of the streams. The reliability estimate is based on the entropy of the posterior probability distributions of each stream at the intermediate decision level. Our results are superior to those obtained with a state of the art method based on maximizing the same posteriors. Moreover, we analyze the effect of a constraint typically imposed on stream weights in the literature, the constraint that they should sum to one. Our results show that removing this constraint can lead to improvements in recognition accuracy. Finally, we develop a method for audio-visual speaker localization, based on the correlation between audio energy and the movement of the speaker's lips. Our method is based on a joint probability model of the audio and video which is used to build a likelihood map showing the likely positions of the speaker's mouth. We show that our novel method performs better than a similar method from the literature. In conclusion, we analyze two different challenges of multimodal signal processing for two audio-visual problems, and offer innovative approaches for solving them

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.Postprint (published version

Speech recognition in parallel

Concomitantly with recent advances in speech coding, recognition and production, parallel computer systems are now commonplace delivenng raw computing power measured in hundreds of MIPS and Megaflops. It seems inevitable that within the next decade or so, gigaflop parallel processors will be achievable at modest cost. Indeed, gigaflops per cubic foot is now becoming a standard of measure for parallel computers