KAVUAKA: a low-power application-specific processor architecture for digital hearing aids

The power consumption of digital hearing aids is very restricted due to their small physical size and the available hardware resources for signal processing are limited. However, there is a demand for more processing performance to make future hearing aids more useful and smarter. Future hearing aids should be able to detect, localize, and recognize target speakers in complex acoustic environments to further improve the speech intelligibility of the individual hearing aid user. Computationally intensive algorithms are required for this task. To maintain acceptable battery life, the hearing aid processing architecture must be highly optimized for extremely low-power consumption and high processing performance.The integration of application-specific instruction-set processors (ASIPs) into hearing aids enables a wide range of architectural customizations to meet the stringent power consumption and performance requirements. In this thesis, the application-specific hearing aid processor KAVUAKA is presented, which is customized and optimized with state-of-the-art hearing aid algorithms such as speaker localization, noise reduction, beamforming algorithms, and speech recognition. Specialized and application-specific instructions are designed and added to the baseline instruction set architecture (ISA). Among the major contributions are a multiply-accumulate (MAC) unit for real- and complex-valued numbers, architectures for power reduction during register accesses, co-processors and a low-latency audio interface. With the proposed MAC architecture, the KAVUAKA processor requires 16 % less cycles for the computation of a 128-point fast Fourier transform (FFT) compared to related programmable digital signal processors. The power consumption during register file accesses is decreased by 6 %to 17 % with isolation and by-pass techniques. The hardware-induced audio latency is 34 %lower compared to related audio interfaces for frame size of 64 samples.The final hearing aid system-on-chip (SoC) with four KAVUAKA processor cores and ten co-processors is integrated as an application-specific integrated circuit (ASIC) using a 40 nm low-power technology. The die size is 3.6 mm2. Each of the processors and co-processors contains individual customizations and hardware features with a varying datapath width between 24-bit to 64-bit. The core area of the 64-bit processor configuration is 0.134 mm2. The processors are organized in two clusters that share memory, an audio interface, co-processors and serial interfaces. The average power consumption at a clock speed of 10 MHz is 2.4 mW for SoC and 0.6 mW for the 64-bit processor.Case studies with four reference hearing aid algorithms are used to present and evaluate the proposed hardware architectures and optimizations. The program code for each processor and co-processor is generated and optimized with evolutionary algorithms for operation merging,instruction scheduling and register allocation. The KAVUAKA processor architecture is com-pared to related processor architectures in terms of processing performance, average power consumption, and silicon area requirements

Homogeneous and heterogeneous MPSoC architectures with network-on-chip connectivity for low-power and real-time multimedia signal processing

Two multiprocessor system-on-chip (MPSoC) architectures are proposed and compared in the paper with reference to audio and video processing applications. One architecture exploits a homogeneous topology; it consists of 8 identical tiles, each made of a 32-bit RISC core enhanced by a 64-bit DSP coprocessor with local memory. The other MPSoC architecture exploits a heterogeneous-tile topology with on-chip distributed memory resources; the tiles act as application specific processors supporting a different class of algorithms. In both architectures, the multiple tiles are interconnected by a network-on-chip (NoC) infrastructure, through network interfaces and routers, which allows parallel operations of the multiple tiles. The functional performances and the implementation complexity of the NoC-based MPSoC architectures are assessed by synthesis results in submicron CMOS technology. Among the large set of supported algorithms, two case studies are considered: the real-time implementation of an H.264/MPEG AVC video codec and of a low-distortion digital audio amplifier. The heterogeneous architecture ensures a higher power efficiency and a smaller area occupation and is more suited for low-power multimedia processing, such as in mobile devices. The homogeneous scheme allows for a higher flexibility and easier system scalability and is more suited for general-purpose DSP tasks in power-supplied devices

System-on-chip Computing and Interconnection Architectures for Telecommunications and Signal Processing

This dissertation proposes novel architectures and design techniques targeting SoC building blocks for telecommunications and signal processing applications. Hardware implementation of Low-Density Parity-Check decoders is approached at both the algorithmic and the architecture level. Low-Density Parity-Check codes are a promising coding scheme for future communication standards due to their outstanding error correction performance. This work proposes a methodology for analyzing effects of finite precision arithmetic on error correction performance and hardware complexity. The methodology is throughout employed for co-designing the decoder. First, a low-complexity check node based on the P-output decoding principle is designed and characterized on a CMOS standard-cells library. Results demonstrate implementation loss below 0.2 dB down to BER of 10^{-8} and a saving in complexity up to 59% with respect to other works in recent literature. High-throughput and low-latency issues are addressed with modified single-phase decoding schedules. A new "memory-aware" schedule is proposed requiring down to 20% of memory with respect to the traditional two-phase flooding decoding. Additionally, throughput is doubled and logic complexity reduced of 12%. These advantages are traded-off with error correction performance, thus making the solution attractive only for long codes, as those adopted in the DVB-S2 standard. The "layered decoding" principle is extended to those codes not specifically conceived for this technique. Proposed architectures exhibit complexity savings in the order of 40% for both area and power consumption figures, while implementation loss is smaller than 0.05 dB. Most modern communication standards employ Orthogonal Frequency Division Multiplexing as part of their physical layer. The core of OFDM is the Fast Fourier Transform and its inverse in charge of symbols (de)modulation. Requirements on throughput and energy efficiency call for FFT hardware implementation, while ubiquity of FFT suggests the design of parametric, re-configurable and re-usable IP hardware macrocells. In this context, this thesis describes an FFT/IFFT core compiler particularly suited for implementation of OFDM communication systems. The tool employs an accuracy-driven configuration engine which automatically profiles the internal arithmetic and generates a core with minimum operands bit-width and thus minimum circuit complexity. The engine performs a closed-loop optimization over three different internal arithmetic models (fixed-point, block floating-point and convergent block floating-point) using the numerical accuracy budget given by the user as a reference point. The flexibility and re-usability of the proposed macrocell are illustrated through several case studies which encompass all current state-of-the-art OFDM communications standards (WLAN, WMAN, xDSL, DVB-T/H, DAB and UWB). Implementations results are presented for two deep sub-micron standard-cells libraries (65 and 90 nm) and commercially available FPGA devices. Compared with other FFT core compilers, the proposed environment produces macrocells with lower circuit complexity and same system level performance (throughput, transform size and numerical accuracy). The final part of this dissertation focuses on the Network-on-Chip design paradigm whose goal is building scalable communication infrastructures connecting hundreds of core. A low-complexity link architecture for mesochronous on-chip communication is discussed. The link enables skew constraint looseness in the clock tree synthesis, frequency speed-up, power consumption reduction and faster back-end turnarounds. The proposed architecture reaches a maximum clock frequency of 1 GHz on 65 nm low-leakage CMOS standard-cells library. In a complex test case with a full-blown NoC infrastructure, the link overhead is only 3% of chip area and 0.5% of leakage power consumption. Finally, a new methodology, named metacoding, is proposed. Metacoding generates correct-by-construction technology independent RTL codebases for NoC building blocks. The RTL coding phase is abstracted and modeled with an Object Oriented framework, integrated within a commercial tool for IP packaging (Synopsys CoreTools suite). Compared with traditional coding styles based on pre-processor directives, metacoding produces 65% smaller codebases and reduces the configurations to verify up to three orders of magnitude

Application-specific instruction set processor for speech recognition.

Cheung Man Ting.Thesis (M.Phil.)--Chinese University of Hong Kong, 2005.Includes bibliographical references (leaves 69-71).Abstracts in English and Chinese.Chapter 1 --- Introduction --- p.1Chapter 1.1 --- The Emergence of ASIP --- p.1Chapter 1.1.1 --- Related Work --- p.3Chapter 1.2 --- Motivation --- p.6Chapter 1.3 --- ASIP Design Methodologies --- p.7Chapter 1.4 --- Fundamentals of Speech Recognition --- p.8Chapter 1.5 --- Thesis outline --- p.10Chapter 2 --- Automatic Speech Recognition --- p.11Chapter 2.1 --- Overview of ASR system --- p.11Chapter 2.2 --- Theory of Front-end Feature Extraction --- p.12Chapter 2.3 --- Theory of HMM-based Speech Recognition --- p.14Chapter 2.3.1 --- Hidden Markov Model (HMM) --- p.14Chapter 2.3.2 --- The Typical Structure of the HMM --- p.14Chapter 2.3.3 --- Discrete HMMs and Continuous HMMs --- p.15Chapter 2.3.4 --- The Three Basic Problems for HMMs --- p.17Chapter 2.3.5 --- Probability Evaluation --- p.18Chapter 2.4 --- The Viterbi Search Engine --- p.19Chapter 2.5 --- Isolated Word Recognition (IWR) --- p.22Chapter 3 --- Design of ASIP Platform --- p.24Chapter 3.1 --- Instruction Fetch --- p.25Chapter 3.2 --- Instruction Decode --- p.26Chapter 3.3 --- Datapath --- p.29Chapter 3.4 --- Register File Systems --- p.30Chapter 3.4.1 --- Memory Hierarchy --- p.30Chapter 3.4.2 --- Register File Organization --- p.31Chapter 3.4.3 --- Special Registers --- p.34Chapter 3.4.4 --- Address Generation --- p.34Chapter 3.4.5 --- Load and Store --- p.36Chapter 4 --- Implementation of Speech Recognition on ASIP --- p.37Chapter 4.1 --- Hardware Architecture Exploration --- p.37Chapter 4.1.1 --- Floating Point and Fixed Point --- p.37Chapter 4.1.2 --- Multiplication and Accumulation --- p.38Chapter 4.1.3 --- Pipelining --- p.41Chapter 4.1.4 --- Memory Architecture --- p.43Chapter 4.1.5 --- Saturation Logic --- p.44Chapter 4.1.6 --- Specialized Addressing Modes --- p.44Chapter 4.1.7 --- Repetitive Operation --- p.47Chapter 4.2 --- Software Algorithm Implementation --- p.49Chapter 4.2.1 --- Implementation Using Base Instruction Set --- p.49Chapter 4.2.2 --- Implementation Using Refined Instruction Set --- p.54Chapter 5 --- Simulation Results --- p.56Chapter 6 --- Conclusions and Future Work --- p.60Appendices --- p.62Chapter A --- Base Instruction Set --- p.62Chapter B --- Special Registers --- p.65Chapter C --- Chip Microphotograph of ASIP --- p.67Chapter D --- The Testing Board of ASIP --- p.68Bibliography --- p.6

Increasing the efficacy of automated instruction set extension

The use of Instruction Set Extension (ISE) in customising embedded processors for a specific application has been studied extensively in recent years. The addition of a set of complex arithmetic instructions to a baseline core has proven to be a cost-effective means of meeting design performance requirements. This thesis proposes and evaluates a reconfigurable ISE implementation called “Configurable Flow Accelerators” (CFAs), a number of refinements to an existing Automated ISE (AISE) algorithm called “ISEGEN”, and the effects of source form on AISE. The CFA is demonstrated repeatedly to be a cost-effective design for ISE implementation. A temporal partitioning algorithm called “staggering” is proposed and demonstrated on average to reduce the area of CFA implementation by 37% for only an 8% reduction in acceleration. This thesis then turns to concerns within the ISEGEN AISE algorithm. A methodology for finding a good static heuristic weighting vector for ISEGEN is proposed and demonstrated. Up to 100% of merit is shown to be lost or gained through the choice of vector. ISEGEN early-termination is introduced and shown to improve the runtime of the algorithm by up to 7.26x, and 5.82x on average. An extension to the ISEGEN heuristic to account for pipelining is proposed and evaluated, increasing acceleration by up to an additional 1.5x. An energyaware heuristic is added to ISEGEN, which reduces the energy used by a CFA implementation of a set of ISEs by an average of 1.6x, up to 3.6x. This result directly contradicts the frequently espoused notion that “bigger is better” in ISE. The last stretch of work in this thesis is concerned with source-level transformation: the effect of changing the representation of the application on the quality of the combined hardwaresoftware solution. A methodology for combined exploration of source transformation and ISE is presented, and demonstrated to improve the acceleration of the result by an average of 35% versus ISE alone. Floating point is demonstrated to perform worse than fixed point, for all design concerns and applications studied here, regardless of ISEs employed