Advanced OFDM systems for terrestrial multimedia links

Recently, there has been considerable discussion about new wireless technologies and standards able to achieve high data rates. Due to the recent advances of digital signal processing and Very Large Scale Integration (VLSI) technologies, the initial obstacles encountered for the implementation of Orthogonal Frequency Division Multiplexing (OFDM) modulation schemes, such as massive complex multiplications and high speed memory accesses, do not exist anymore. OFDM offers strong multipath protection due to the insertion of the guard interval; in particular, the OFDM-based DVB-T standard had proved to offer excellent performance for the broadcasting of multimedia streams with bitrates over ten megabits per second in difficult terrestrial propagation channels, for fixed and portable applications. Nevertheless, for mobile scenarios, improving the receiver design is not enough to achieve error-free transmission especially in presence of deep shadow and multipath fading and some modifications of the standard can be envisaged. To address long and medium range applications like live mobile wireless television production, some further modifications are required to adapt the modulated bandwidth and fully exploit channels up to 24MHz wide. For these reasons, an extended OFDM system is proposed that offers variable bandwidth, improved protection to shadow and multipath fading and enhanced robustness thanks to the insertion of deep time-interleaving coupled with a powerful turbo codes concatenated error correction scheme. The system parameters and the receiver architecture have been described in C++ and verified with extensive simulations. In particular, the study of the receiver algorithms was aimed to achieve the optimal tradeoff between performances and complexity. Moreover, the modulation/demodulation chain has been implemented in VHDL and a prototype system has been manufactured. Ongoing field trials are demonstrating the ability of the proposed system to successfully overcome the impairments due to mobile terrestrial channels, like multipath and shadow fading. For short range applications, Time-Division Multiplexing (TDM) is an efficient way to share the radio resource between multiple terminals. The main modulation parameters for a TDM system are discussed and it is shown that the 802.16a TDM OFDM physical layer fulfills the application requirements; some practical examples are given. A pre-distortion method is proposed that exploit the reciprocity of the radio channel to perform a partial channel inversion achieving improved performances with no modifications of existing receivers.

Efficient error correction solutions for OFDM based digital video

Mauro Lattuada

The robustness of the delivery of digital compressed video streams has always been one of the main topic of the standardization bodies to assure the Quality of the Service (QoS). The MPEG standard which is nowadays the reference for digital television defines how to format the various component parts of a multimedia program and how they are combined into a single transmission bit stream. Although the MPEG stream may be directly used over a wide variety of media, it may also be used over a communication network and it is already packetized into short data packets. The multiplexed stream can be transmitted over a variety of communication networks such as Radio Frequency links (UHF/VHF), Digital Satellite links, Cable TV Networks, Standard Terrestrial Communication Links (PDH, SDH) or Packet/Cell Links (ATM, IP, IPv6, Ethernet). In critical and error prone channels the robustness of the stream has to be granted by employing strong error correction schemes. Depending on the used media the transmission has its own coding scheme and modulation methods designed for the particular environment. For example, in the case of IP based links the transfer protocol accepts loss of data and the problem can be easily overcome by retransmission. Wireless simplex transmission cannot exploit the same approach. Wireless transmission always means variable channel conditions especially when one or both terminals are moving on the outfield. The channel fading evolution and the signal impairments increase the error rate and corrupt the data. In the recent past the standardization of the DVB-T shows the possibility to provide a robust service for digital video delivery. This standard provides superior picture quality with the opportunity to view pictures in standard format or wide screen (4/3 or 16/9) formats, services including subtitling, multiple audio tracks, interactive content, multimedia content where, for instance, programs may be linked to world wide web material. Beside the provided services and features the standard offers also a modulation and transmission scheme capable to cope with the impairments of the terrestrial channel and to provide a BER after forward-error correction below 10-12. Unfortunately, these performances are guaranteed only for fixed reception, since the standard was originally aimed to replace classical analog television broadcasting in the UHF band. This thesis addresses the study of enhanced error-correction schemes in the scope of a compact, flexible and robust high-throughput real time video transmission system able to cope with high mobility channels. The study of a transmission system and error correction scheme tailored for this kind of application is here presented. The system exploits the OFDM modulation technique. This modulation scheme has already proven to be very effective in hostile environments. A flexible modulation scheme is employed to obtain variable bandwidth transmission and enhanced performances especially for mobile applications. The main topic of this research is the study of an effective error correction scheme adapted to this application. While nowadays the digital television standards use a concatenated error correction scheme based on convolutional inner coder, in this work the proposed solution is based on a turbo inner coder. Turbo codes has already shown good performances compared to convolutional codes. These results are here applied to a concatenated scheme to obtain very low BER. A comparison of the performances of various turbo coding techniques under certain implementation constraints is done in this work.The second element of the error correction scheme is the channel interleaver used to average the channel fading in time and frequency. This mechanism is conceived to be adapted to the modulated bandwidth and is optimized for high throughput real time video. This study shows that the proposed system proves to be very effective in high mobile channels because of the deep averaging strategies in conjunction with the high efficiency turbo based concatenated error correction scheme.

EPFL2005

Architecture of a Real-Time Platform Independant GPS L1 Software Receiver

Personal digital assistants or mobile phones applications are not anymore restricted to multimedia or wireless communications, but have been extended to handle Global Positioning System (GPS) functionalities. Consequently, the growing market of GPS capable mobile devices is driving the interest of software receiver solutions as they provide several advantages with respect to traditional hardware implementations. First, they share the same system resources such as the processor, embedded memory and power with other system units, reducing both the size and the costs of their integration. Second, they can be easily reprogrammed – via a firmware update – for incorporating the latest developments, such as the exploitation of the future satellites signals or some improved multipath mitigation techniques. Finally, they offer a more flexible solution for rapid research and development as compared to conventional hardware receivers where the chip design is fixed and obtained after a long integration process. With the increasing performance of modern processors, it becomes now feasible to implement in software a multi-channel GPS receiver operating in real-time. However, a major problem with the software architecture is the large computing resources required for the digital signal processing. Former studies have demonstrated that a straightforward transposition of traditional hardware based architectures into software would lead to an amount of integer operations which is not suitable for today's fastest computers. From this observation, several strategies have been proposed in the literature in order to reduce the complexity of the receiver operations. The first one relies on the utilization of advanced microprocessor instructions set which provides the capability of processing vectors of data by operating on multiple integer values at the same time. This results in significant gains in execution speed, but also severely limits the portability of the code, since the operations are tied to specific processors architectures. Another alternative consists in exploiting the native bitwise representation of the signal. The data bits are stored in separate vectors on which logical parallel operations can be performed. The objective is to take advantage of the universality, high parallelism, and speed of the bitwise operations for which a single integer operation translates into a few simple parallel logical relations. However, the inherent drawback of the bitwise processing is the lack of flexibility as the complexity becomes bit-depth dependent. This thesis has been carried out in the framework of a two-year industrial project (2007-2009) in collaboration with U-blox AG in Thalwil. It aimed to the realization of a multi-channel, platform-independent real-time GPS L1 software receiver. The main challenge of this project consisted in providing real-time performances while keeping the portability of the code to make the receiver suitable for any type of software implementation. In that sense, new techniques and algorithms have been developed for optimizing the processing chain in order to lower the processor load. The main idea consists in regrouping data which share the same characteristics, and process them in batches instead of sequentially. This way, it becomes possible to progressively reduce the data throughput and consequently the amount of operations to perform. A completely new receiver architecture has been proposed and validated through the realization of a functional prototype, thus demonstrating the feasibility of the concept.

EPFL2010

A flexible framework for efficient design of advanced 3D audio systems in multimedia applications

Aleksandar Simeonov

The recent advance of computer technology pushed the computing power of the modern computer systems to unprecedented levels; modern processors moved from the conventional, scalar type of architecture to more sophisticated, parallel ones, combining fast processing speed clock with multiple processing units. This jump of the computing power made possible the real-time implementations of many 3D audio algorithms that was hard to imagine 10-15 years before; on the other hand, the permanent increase of customer demand for real-time manipulation of the multimedia content leads to the development and implementation of more complex, and respectively more computationally demanding algorithms. The goal to deliver such sophisticated solutions to the market in a short time is already a difficult task; to achieve efficient solutions in such short terms, squeezing the maximum of the computing power of the currently available technologies, is even more challenging. In my thesis, I propose a solution to this problem by designing and implementing a framework that allows the developer to quickly design, implement and test custom 3D audio algorithms in an efficient way, with minimum efforts to port the already available code to another platform. The proposed approach started from a preliminary technology review and analysis in terms of development tools to find out and evaluate the state-of-the-art from 3D audio point of view. However, the performed analysis revealed some important drawbacks of these solutions in terms of fast reconfigurability for the implementation of custom 3D audio algorithms, and their capability for a precise synchronization and management of multiple stream media applications. First, it is in some cases necessary to develop completely new code to achieve the desired results, due to a lack of flexibility in the algorithm configurations or to unavailable functionality. Secondly, if the integration of different solutions and APIs, running at different hardware and software levels is pursued, this causes some problems with the synchronization between them, since high level APIs often introduce high latencies. The result is in some cases the system instability with consequent degraded quality, or even loss, of audio content for live and networked scenarios. In the end, it was noticed that if the APIs for 3D sound are higher level and developer-friendly they normally bring with them high latencies, sometimes-rigid 3D model solutions or high portability costs. If on the other hand they are lower-level solutions conceived for platform-independent low latency development and true real-time performance, the development of existing and particularly new 3D models may result long if good and robust source code is not available. To improve the situation, a new approach has been conceived to develop an independent middle-to-low level DSP library, which can provide a mid-layer development tool to easily design and configure new 3D algorithms. This simplifies at the same time the implementation task by providing the main building blocks in the field of signal processing and media streams, leaving to the developer enough flexibility and reconfigurability to implement the 3D audio models in a short time with the desired precision and quality. An object-oriented, two-level hierarchical structure has been adopted for the library. As a root of the hierarchy, some common properties for all primitives have been defined (e.g. number of input/output channels, single sample and block processing functions, etc.) and assigned to an object, common for all the DSP primitives (Parent Node). The first level (Atomic Level) contains the DSP primitives that cannot be built by the others: multiplier, adder, multiply-add, delay line, etc. Ideally, this layer corresponds to an instruction set and memory structures of a virtual machine designed for signal processing. The second level (Basic DSP Level) contains generic DSP primitives, e.g. low-pass filter, all-pass, nested all-pass, double nested all-pass, generic FIR and IIR filters, geometric and trigonometric operators (on vectors), FDN matrices, etc. This level is build by primitives, defined in the Atomic Level. Finally, as a test application, a 3D Audio layer has been implemented containing 3D algorithms, compatible first with the standardized VRML and MPEG-4 geometrical and perceptual room models, and in addition with several other algorithms like: Schroeder reverberator, Dattorro network reverberator, Gardner reverberator, etc. This allowed the successful integration of the proposed 3D audio library in a MPEG-4 compatible player, which has been implemented within the scope of the CARROUSO multimedia project. CARROUSO addressed the problem of how to transfer sound fields, generated at a certain real or virtual space, to another usually remote located space. The activities in CARROUSO did concentrate on the development and implementation of key components for recording, encoding, transmission and rendering, including hardware, software and algorithms. The transmission and coding was based on MPEG-4 and digital video broadcasting (DVB) technology, while sound recording and reproduction was based on wave field synthesis (WFS) technology. In this context, the MPEG-4 player is essential part of the chain, displaying the video content of the virtual scene, and providing the interface between the decoded sound content and the WFS rendering system. In contrast to the state-of-the art multi-channel sound systems, the new system has an enlarged listening area and is more flexible in respect to the number and the configuration of loudspeakers. The innovative work in CARROUSO found later its logical continuation in the MAP project where the developed 3D API has been exploited to integrate custom or standardized 3D audio models into a professional sound production toolbox, running on mixed general-purpose/FPGA/multimedia-DSP technology to obtain true real-time performance. In addition to the possibility to monitor locally multi-channel output, this tool provides at the same time the functionality to generate/encode object-oriented audio content, compatible to MPEG-4 BIFS format, which turns it into a sophisticated MPEG-4 authoring tool. As a conclusion, it could be noted, that the four years research and work during the CARROUSO and MAP projects, have proven in practice that the proposed 3D audio development framework is not only flexible and easily configurable, but it can be also very efficient, and ported to different platforms without putting significant effort in this task.

EPFL2004