Development and Evaluation of a Real-Time Framework for a Portable Assistive Hearing Device

Testing and verification of digital hearing aid devices, and the embedded software and algorithms can prove to be a challenging task especially taking into account time-to-market considerations. This thesis describes a PC based, real-time, highly configurable framework for the evaluation of audio algorithms. Implementation of audio processing algorithms on such a platform can provide hearing aid designers and manufacturers the ability to test new and existing processing techniques and collect data about their performance in real-life situations, and without the need to develop a prototype device. The platform is based on the Eurotech Catalyst development kit and the Fedora Linux OS, and it utilizes the JACK audio engine to facilitate reliable real-time performance Additionally, we demonstrate the capabilities of this platform by implementing an audio processing chain targeted at improving speech intelligibility for people suffering from auditory neuropathy. Evaluation is performed for both noisy and noise-free environments. Subjective evaluation of the results, using normal hearing listeners and an auditory neuropathy simulator, demonstrates improvement in some conditions

Algorithms and architectures for the multirate additive synthesis of musical tones

In classical Additive Synthesis (AS), the output signal is the sum of a large number of independently controllable sinusoidal partials. The advantages of AS for music synthesis are well known as is the high computational cost. This thesis is concerned with the computational optimisation of AS by multirate DSP techniques. In note-based music synthesis, the expected bounds of the frequency trajectory of each partial in a finite lifecycle tone determine critical time-invariant partial-specific sample rates which are lower than the conventional rate (in excess of 40kHz) resulting in computational savings. Scheduling and interpolation (to suppress quantisation noise) for many sample rates is required, leading to the concept of Multirate Additive Synthesis (MAS) where these overheads are minimised by synthesis filterbanks which quantise the set of available sample rates. Alternative AS optimisations are also appraised. It is shown that a hierarchical interpretation of the QMF filterbank preserves AS generality and permits efficient context-specific adaptation of computation to required note dynamics. Practical QMF implementation and the modifications necessary for MAS are discussed. QMF transition widths can be logically excluded from the MAS paradigm, at a cost. Therefore a novel filterbank is evaluated where transition widths are physically excluded. Benchmarking of a hypothetical orchestral synthesis application provides a tentative quantitative analysis of the performance improvement of MAS over AS. The mapping of MAS into VLSI is opened by a review of sine computation techniques. Then the functional specification and high-level design of a conceptual MAS Coprocessor (MASC) is developed which functions with high autonomy in a loosely-coupled master- slave configuration with a Host CPU which executes filterbanks in software. Standard hardware optimisation techniques are used, such as pipelining, based upon the principle of an application-specific memory hierarchy which maximises MASC throughput

Fairness in a data center

Existing data centers utilize several networking technologies in order to handle the performance requirements of different workloads. Maintaining diverse networking technologies increases complexity and is not cost effective. This results in the current trend to converge all traffic into a single networking fabric. Ethernet is both cost-effective and ubiquitous, and as such it has been chosen as the technology of choice for the converged fabric. However, traditional Ethernet does not satisfy the needs of all traffic workloads, for the most part, due to its lossy nature and, therefore, has to be enhanced to allow for full convergence. The resulting technology, Data Center Bridging (DCB), is a new set of standards defined by the IEEE to make Ethernet lossless even in the presence of congestion. As with any new networking technology, it is critical to analyze how the different protocols within DCB interact with each other as well as how each protocol interacts with existing technologies in other layers of the protocol stack. This dissertation presents two novel schemes that address critical issues in DCB networks: fairness with respect to packet lengths and fairness with respect to flow control and bandwidth utilization. The Deficit Round Robin with Adaptive Weight Control (DRR-AWC) algorithm actively monitors the incoming streams and adjusts the scheduling weights of the outbound port. The algorithm was implemented on a real DCB switch and shown to increase fairness for traffic consisting of mixed-length packets. Targeted Priority-based Flow Control (TPFC) provides a hop-by-hop flow control mechanism that restricts the flow of aggressor streams while allowing victim streams to continue unimpeded. Two variants of the targeting mechanism within TPFC are presented and their performance evaluated through simulation

Low delay video coding

Analogue wireless cameras have been employed for decades, however they have not become an universal solution due to their difficulties of set up and use. The main problem is the link robustness which mainly depends on the requirement of a line-of-sight view between transmitter and receiver, a working condition not always possible. Despite the use of tracking antenna system such as the Portable Intelligent Tracking Antenna (PITA [1]), if strong multipath fading occurs (e.g. obstacles between transmitter and receiver) the picture rapidly falls apart. Digital wireless cameras based on Orthogonal Frequency Division Multiplexing (OFDM) modulation schemes give a valid solution for the above problem. OFDM offers strong multipath protection due to the insertion of the guard interval; in particular, the OFDM-based DVB-T standard has proven to offer excellent performance for the broadcasting of multimedia streams with bit rates over 10 Mbps in difficult terrestrial propagation channels, for fixed and portable applications. However, in typical conditions, the latency needed to compress/decompress a digital video signal at Standard Definition (SD) resolution is of the order of 15 frames, which corresponds to ≃ 0.5 sec. This delay introduces a serious problem when wireless and wired cameras have to be interfaced. Cabled cameras do not use compression, because the cable which directly links transmitter and receiver does not impose restrictive bandwidth constraints. Therefore, the only latency that affects a cable cameras link system is the on cable propagation delay, almost not significant, when switching between wired and wireless cameras, the residual latency makes it impossible to achieve the audio-video synchronization, with consequent disagreeable effects. A way to solve this problem is to provide a low delay digital processing scheme based on a video coding algorithm which avoids massive intermediate data storage. The analysis of the last MPEG based coding standards puts in evidence a series of problems which limits the real performance of a low delay MPEG coding system. The first effort of this work is to study the MPEG standard to understand its limit from both the coding delay and implementation complexity points of views. This thesis also investigates an alternative solution based on HERMES codec, a proprietary algorithm which is described implemented and evaluated. HERMES achieves better results than MPEG in terms of latency and implementation complexity, at the price of higher compression ratios, which means high output bit rates. The use of HERMES codec together with an enhanced OFDM system [2] leads to a competitive solution for wireless digital professional video applications

An accurate analysis for guaranteed performance of multiprocessor streaming applications

Already for more than a decade, consumer electronic devices have been available for entertainment, educational, or telecommunication tasks based on multimedia streaming applications, i.e., applications that process streams of audio and video samples in digital form. Multimedia capabilities are expected to become more and more commonplace in portable devices. This leads to challenges with respect to cost efficiency and quality. This thesis contributes models and analysis techniques for improving the cost efficiency, and therefore also the quality, of multimedia devices. Portable consumer electronic devices should feature flexible functionality on the one hand and low power consumption on the other hand. Those two requirements are conflicting. Therefore, we focus on a class of hardware that represents a good trade-off between those two requirements, namely on domain-specific multiprocessor systems-on-chip (MP-SoC). Our research work contributes to dynamic (i.e., run-time) optimization of MP-SoC system metrics. The central question in this area is how to ensure that real-time constraints are satisfied and the metric of interest such as perceived multimedia quality or power consumption is optimized. In these cases, we speak of quality-of-service (QoS) and power management, respectively. In this thesis, we pursue real-time constraint satisfaction that is guaranteed by the system by construction and proven mainly based on analytical reasoning. That approach is often taken in real-time systems to ensure reliable performance. Therefore the performance analysis has to be conservative, i.e. it has to use pessimistic assumptions on the unknown conditions that can negatively influence the system performance. We adopt this hypothesis as the foundation of this work. Therefore, the subject of this thesis is the analysis of guaranteed performance for multimedia applications running on multiprocessors. It is very important to note that our conservative approach is essentially different from considering only the worst-case state of the system. Unlike the worst-case approach, our approach is dynamic, i.e. it makes use of run-time characteristics of the input data and the environment of the application. The main purpose of our performance analysis method is to guide the run-time optimization. Typically, a resource or quality manager predicts the execution time, i.e., the time it takes the system to process a certain number of input data samples. When the execution times get smaller, due to dependency of the execution time on the input data, the manager can switch the control parameter for the metric of interest such that the metric improves but the system gets slower. For power optimization, that means switching to a low-power mode. If execution times grow, the manager can set parameters so that the system gets faster. For QoS management, for example, the application can be switched to a different quality mode with some degradation in perceived quality. The real-time constraints are then never violated and the metrics of interest are kept as good as possible. Unfortunately, maintaining system metrics such as power and quality at the optimal level contradicts with our main requirement, i.e., providing performance guarantees, because for this one has to give up some quality or power consumption. Therefore, the performance analysis approach developed in this thesis is not only conservative, but also accurate, so that the optimization of the metric of interest does not suffer too much from conservativity. This is not trivial to realize when two factors are combined: parallel execution on multiple processors and dynamic variation of the data-dependent execution delays. We achieve the goal of conservative and accurate performance estimation for an important class of multiprocessor platforms and multimedia applications. Our performance analysis technique is realizable in practice in QoS or power management setups. We consider a generic MP-SoC platform that runs a dynamic set of applications, each application possibly using multiple processors. We assume that the applications are independent, although it is possible to relax this requirement in the future. To support real-time constraints, we require that the platform can provide guaranteed computation, communication and memory budgets for applications. Following important trends in system-on-chip communication, we support both global buses and networks-on-chip. We represent every application as a homogeneous synchronous dataflow (HSDF) graph, where the application tasks are modeled as graph nodes, called actors. We allow dynamic datadependent actor execution delays, which makes HSDF graphs very useful to express modern streaming applications. Our reason to consider HSDF graphs is that they provide a good basic foundation for analytical performance estimation. In this setup, this thesis provides three major contributions: 1. Given an application mapped to an MP-SoC platform, given the performance guarantees for the individual computation units (the processors) and the communication unit (the network-on-chip), and given constant actor execution delays, we derive the throughput and the execution time of the system as a whole. 2. Given a mapped application and platform performance guarantees as in the previous item, we extend our approach for constant actor execution delays to dynamic datadependent actor delays. 3. We propose a global implementation trajectory that starts from the application specification and goes through design-time and run-time phases. It uses an extension of the HSDF model of computation to reflect the design decisions made along the trajectory. We present our model and trajectory not only to put the first two contributions into the right context, but also to present our vision on different parts of the trajectory, to make a complete and consistent story. Our first contribution uses the idea of so-called IPC (inter-processor communication) graphs known from the literature, whereby a single model of computation (i.e., HSDF graphs) are used to model not only the computation units, but also the communication unit (the global bus or the network-on-chip) and the FIFO (first-in-first-out) buffers that form a ‘glue’ between the computation and communication units. We were the first to propose HSDF graph structures for modeling bounded FIFO buffers and guaranteed throughput network connections for the network-on-chip communication in MP-SoCs. As a result, our HSDF models enable the formalization of the on-chip FIFO buffer capacity minimization problem under a throughput constraint as a graph-theoretic problem. Using HSDF graphs to formalize that problem helps to find the performance bottlenecks in a given solution to this problem and to improve this solution. To demonstrate this, we use the JPEG decoder application case study. Also, we show that, assuming constant – worst-case for the given JPEG image – actor delays, we can predict execution times of JPEG decoding on two processors with an accuracy of 21%. Our second contribution is based on an extension of the scenario approach. This approach is based on the observation that the dynamic behavior of an application is typically composed of a limited number of sub-behaviors, i.e., scenarios, that have similar resource requirements, i.e., similar actor execution delays in the context of this thesis. The previous work on scenarios treats only single-processor applications or multiprocessor applications that do not exploit all the flexibility of the HSDF model of computation. We develop new scenario-based techniques in the context of HSDF graphs, to derive the timing overlap between different scenarios, which is very important to achieve good accuracy for general HSDF graphs executing on multiprocessors. We exploit this idea in an application case study – the MPEG-4 arbitrarily-shaped video decoder, and demonstrate execution time prediction with an average accuracy of 11%. To the best of our knowledge, for the given setup, no other existing performance technique can provide a comparable accuracy and at the same time performance guarantees

An efficient phonation-driven control system using laryngeal bioimpedance and machine learning

The extraction and conversion of human voice information are crucial in several applications across multiple subject areas such as medicine, music technology and human-computer interaction. The presented research employs the variation of laryngeal bioimpedance, measured during phonation, for extracting and processing voice information. Compared to sound recordings and microphones, bioimpedance readings deliver a much simpler signal, allowing fast and computationally non-taxing processing. In the first stage of this research, a novel system for measuring laryngeal bioimpedance was designed and built. The circuit design was implemented with a multiplexed sensor system based on multiple electrode pairs to allow self-calibration of the sensors and increase usability and applicability. In the following stage, the resulting device was used to generate a novel dataset of laryngeal bioimpedance measurements for the distinction of speech and singing. This was then used in the training and deployment of an Artificial Neural Network using the Mel Frequency Cepstrum Coefficients of the recorded bioimpedance measurements. A real-time system for converting voice into digital control messages was developed and presented as the third stage of this research. The system was implemented using the MIDI protocol for using voice to control hardware and software electronic instruments. The thesis then concludes with the integration of the complete system. The conducted research results in a self-calibrating device for the measurement of laryngeal bioimpedance which delivers an fast and efficacious real-time voice-to-MIDI conversion. In addition, the creation of a unique dataset for the distinction of singing and speech allowed the deployment of real-time classification system. Collectively, the proposed system improves applicability and usability of laryngeal bioimpedance and expands the existing knowledge in the distinction of speech and singing

Real-Time Polyphonic Octave Doubling for the Guitar

This thesis studies digital signal processing solutions for enriching live guitar sound by way of mixing-in octave-doubled versions of the chords and melodies performed on the instrument in real-time. Following a review of techniques applicable for real-time polyphonic octave doubling, four candidate solutions are proposed, amongst which two novel methods: ERB-PS2 and ERB-SSM2. Performance of said candidates is compared to that of three state of the art effect pedal offerings of the market. In particular, an evaluation of the added roughness and transient alterations introduced by each solution in the output sound is conducted. The ERB-PS2 method, which consists in doubling the instantaneous phases of the sub-bands signals extracted with a constant-ERB-bandwidth non-decimated IIR filter bank, is found to provide the best overall performance amongst the candidates. This novel solution provides greatly reduced latency compared to the baseline pedals, with comparable, and in some case improved, sound quality