Speech Recognition on an FPGA Using Discrete and Continuous Hidden Markov Models

Speech recognition is a computationally demanding task, particularly the stage which uses Viterbi decoding for converting pre-processed speech data into words or sub-word units. Any device that can reduce the load on, for example, a PC’s processor, is advantageous. Hence we present FPGA implementations of the decoder based alternately on discrete and continuous hidden Markov models (HMMs) representing monophones, and demonstrate that the discrete version can process speech nearly 5,000 times real time, using just 12% of the slices of a Xilinx Virtex XCV1000, but with a lower recognition rate than the continuous implementation, which is 75 times faster than real time, and occupies 45% of the same device

Speech recognition on an FPGA using Discrete and Continuous Hidden Markov models

Abstract. Speech recognition is a computationally demanding task, particularly the stage which uses Viterbi decoding for converting pre-processed speech data into words or sub-word units. Any device that can reduce the load on, for example, a PC’s processor, is advantageous. Hence we present FPGA implementations of the decoder based alternately on discrete and continuous hidden Markov models (HMMs) representing monophones, and demonstrate that the discrete version can process speech nearly 5,000 times real time, using just 12 % of the slices of a Xilinx Virtex XCV1000, but with a lower recognition rate than the continuous implementation, which is 75 times faster than real time, and occupies 45 % of the same device.

A Vectorized Processing Algorithm for Continuous Speech Recognition and Associated FPGA-Based Architecture

This work analyzes Continuous Automatic Speech Recognition (CSR) and in contrast to prior work, it shows that the CSR algorithms can be specified in a highly parallel form. Through use of the MATLAB software package, the parallelism is exploited to create a compact, vectorized algorithm that is able to execute the CSR task. After an in-depth analysis of the SPHINX 3 Large Vocabulary Continuous Speech Recognition (LVCSR) engine the major functional units were redesigned in the MATLAB environment, taking special effort to flatten the algorithms and restructure the data to allow for matrix-based computations. Performing this conversion resulted in reducing the original 14,000 lines of C++ code into less then 200 lines of highly-vectorized operations, substantially increasing the potential Instruction Line Parallelism of the system. Using this vector model as a baseline, a custom hardware system was then created that is capable of performing the speech recognition task in real-time on a Xilinx Virtex-4 FPGA device. Through the creation independent hardware engines for each stage of the speech recognition process, the throughput of each is maximized by customizing the logic to the specific task. Further, a unique architecture was designed that allows for the creation of a static data path throughout the hardware, effectively removing the need for complex bus arbitration in the system. By making using of shared memory resources and applying a token passing scheme to the system, both the data movement within the design as well as the amount of active data are continually minimized during run-time. These results provide a novel method for perform speech recognition in both hardware and software, helping to further the development of systems capable of recognizing human speech

Speech Recognition

Chapters in the first part of the book cover all the essential speech processing techniques for building robust, automatic speech recognition systems: the representation for speech signals and the methods for speech-features extraction, acoustic and language modeling, efficient algorithms for searching the hypothesis space, and multimodal approaches to speech recognition. The last part of the book is devoted to other speech processing applications that can use the information from automatic speech recognition for speaker identification and tracking, for prosody modeling in emotion-detection systems and in other speech processing applications that are able to operate in real-world environments, like mobile communication services and smart homes