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New Publications are available now online for this publication.
Please follow the links to view the publication.Comparative study of error control coding in underwater acoustic channel
http://dl-live.theiet.org/content/conferences/10.1049/ic.2011.0079
High data-transmission rate and reliable communication is a challenge for Underwater Acoustic Channel (UAC). The UAC is frequency-dependent subjected to multipath, and low speed of sound propagation. In the presence of existing noise and interference, for effective communication selection of proper Codec (Coding Decoding) Technique and Modem (modulation and demodulation) are essential. In this paper we have discussed the existing three different codec (algebraic, trellis and iterative) technique and advantage of basic digital modulation (PSK, QPSK, FSK) used in UAC communication.Design and ASIC implementation of 2-d DWT IDWT
http://dl-live.theiet.org/content/conferences/10.1049/ic.2011.0063
In this paper, high-efficient lifting-based architectures for the 5/3 discrete wavelet transform (DWT) are proposed. Filter coefficients of the biorthogonal 5/3 wavelet low-pass filter are quantized before implementation in the high-speed computation hardware. In the proposed architecture, all multiplications are performed using less shifts and additions. The digital signal represented in time-scale obtained by using digital filtering techniques is known as Discrete Wavelet Transform. Here the signal to be analyzed is passed through filters with different cutoff frequencies at different scales. The DWT is computed by successive lowpass and highpass filtering of the discrete time-domain signal. The efficient architecture had been chosen for lifting scheme based DWT/IDWT process and modeled in Verilog with synthesis point of view. To meet the standards of quality, ICs should be thoroughly tested, where the necessity of suitable DFT scans arises. The estimated dynamic power consumption is 2.81mW and leakage power is 20.370uW.A novel 8×8 transform method applied in video coding
http://dl-live.theiet.org/content/conferences/10.1049/cp.2011.0846
Transform Coding has been playing an important role in video coding and increasingly becomes a research focus especially in the current popular standards such as H.264/AVC, AVS and HEVC. It is important to select an excellent transform method as transform module has a direct impact on the efficiency of video codec. This paper proposes a new 8×8 transform method as well as its integer approximation applied in video coding. Experiments show that it achieves a higher performance.The design of RS (255,239) encoder based on ADSL
http://dl-live.theiet.org/content/conferences/10.1049/cp.2011.0726
The design of RS (255,239) encoder based on ADSL system GF (2 8) is studied, the core encoder multiplier unit and limited domain constant realization of hardware are presented in the paper. Because the coding process adopts 16 dedicated constant multiplier units, with variable comparison multiplier unit using before, which greatly simplified the hardware structure of multiplier unit, saved the hardware area, and improved the speed of multiplier unit. In addition, the performance of RS (255,239) coding is validated using the MATLAB programming language.Common TTCN-3 codec to reduce test engineering cost
http://dl-live.theiet.org/content/conferences/10.1049/cp.2010.0781
TTCN-3 is an abstract language for specification of Abstract Test Suites. Coding of TTCN-3 variable into physically transmittable messages and decoding of bit strings into their TTCN-3 representation is an external and specialized component, called CoDec. CoDec development, either implicitly or explicitly, must be included in any TTCN-3 testing activity. Experience showed that there is a high cost associated with CoDec development and maintenance. This paper presents a methodological approach to minimize the complexity of CoDec development. The approach is method and also been developed to a tool. It enables the developer to complete the codec work by just write an XML file. Currently it only supports few types of protocol, but it Provide a framework that can expand itself to support more and more protocol data format.Packet loss concealment for speech transmission based on compressed sensing
http://dl-live.theiet.org/content/conferences/10.1049/cp.2009.1957
Providing effective real-time speech transmission with acceptable quality for VoIP applications and mobile and wireless speech communication is a very important and challenging task. In these communication networks, the bad situations with heavy packet loss happen from time to time. In this paper, a packet loss concealment scheme based on the compressed sensing technology is proposed. It only uses the available speech samples to recover the lost ones by mean of compressed sensing, so it is independent of speech codec algorithms; therefore it can be used in any speech communication systems in receiver side. Evaluation results have shown that the performance of it is still good even when packet loss rate is 40% or higher, it is also good for both waveform and parameter based speech coding applications.Enhanced uplink scheduling algorithm based on frame duration for VoIP over M-WiMAX
http://dl-live.theiet.org/content/conferences/10.1049/cp.2009.2007
In order to solve the uplink resource wastes during silence period and optimize the system VoIP capacity, this paper proposes an enhanced uplink scheduling algorithm that can successfully support various DTX VoIP CODECs over MWiMAX based on the CODEC frame duration. Efficient schemes is proposed to implement the uplink resource interval and allocation procedure, by adaptation of uplink resource intervals of BS according to the reserved bit in the generic MAC header of IEEE802.16e, which is utilized to indicate voice activity of each voice user. Performance comparisons between the conventional algorithms (UGS, rtPS and ertPS) and the proposed algorithm are given. Further, numerical validations are given to prove the efficiency of the proposed algorithm over enhancement of resource utilization, total throughput, increment of system capacity, and number of voice users supported.Event based video coding architecture
http://dl-live.theiet.org/content/conferences/10.1049/cp_20080421
In this work, scalable video codec (SVC) has been used for security surveillance video. The main event in the video is motion of an object. A strategy has been proposed to find out the different motion levels (events) in the video. The level of motion in a group of picture (GOP) is used to assign different scalability features to the GOP. The architecture of SVC has been modified to provide this support. The model of the modified SVC architecture is presented in detail. The improved system handles fewer amounts of data for processing and storage yet conveys all the important information related to surveillance video. The implementation and experimental results on the surveillance video has been presented. Results show that the proposed system efficiently detects motion and adapts to the scalability level accordingly.Design technique of Viterbi decoder in satellite communication
http://dl-live.theiet.org/content/conferences/10.1049/cp_20070108
By analyzing the well-known Viterbi algorithm, a Viterbi decoder especially used in satellite communication system is presented. In this paper, the design technique of each part has been discussed. The decoder adopts 4 butterfly units in part- parallel calculation structure with concise interconnection, which is implemented with Xilinx Virtex-II 2V1000- 5BGS575 FPGA (field programmable gate array). The design delays 54 clock periods and costs 501 slices.Complexity scalable 3D video coding based on mixed transform technique
http://dl-live.theiet.org/content/conferences/10.1049/cp_20060530
Mobile devices, performing video coding and streaming over wireless communication networks are of limited battery power. To prolong the operational lifetime of these devices a complexity scalable embedded video coding scheme is desired. We present a framework for the systematic analysis of the computational complexity of the 3D video encoding scheme in terms of average processor cycles. A mixed transform based 3D video codec is proposed with no motion estimation/compensation. A parametric approach to control the computational complexity of this 3D video coding scheme by defining a set of complexity control parameters. We generate a wide range of PSNR-rate-complexity operating points for different sequences, by modifying the complexity control parameters.End-to-end delivery strategies for H.264 video streaming over 3G wireless networks
http://dl-live.theiet.org/content/conferences/10.1049/cp_20061498
This paper address the important issues of delivery strategies for H.264 video transmission over 3G wireless networks. By taking the time-varying wireless channel condition and H.264 video streaming structure characteristics into account, we implement JSCC (joint source and channel codec) based on measuring packer loss-ratio. First, we present an end-to-end transport architecture for H.264 video streaming over 3G wireless networks. Second, we analyze error resilience tools of H.264 video streaming which adapt to the time-varying wireless channel condition. Third, we propose a strategies which dynamically estimate the time-varying wireless network condition by measuring packets loss-ratio and throughout. In the end, furthermore, we present simplified packing strategy in order to enhancing robust of end-to-end delivery H.264 video streaming. By integrated strategies, we can improve the end-to-end quality of services. The simulation results show our delivery strategies can significantly improve the robustness of end-to-end delivery H.264 video streaming. (6 pages)An HSBE-LPC low bit wideband speech coding algorithm
http://dl-live.theiet.org/content/conferences/10.1049/cp_20061450
This paper addressed the design, implementation and evaluation of a low bit-rate wideband speech coding algorithm HSBE-LPC based on harmonic sinusoidal speech model (HSSM) and linear predictive coding technique. An improved HSSM was presented, where an improved sub-harmonic-harmonic high precision pitch algorithm to detect pitch, exacting frequency offsets and the SEEVOC to get envelope. The voiced possibility was iteratively computed combined with the least square method of sinusoidal speech model (SSM) technique on the basis of the obtained pitch and envelope. A split vector quantization method with dynamic time warping to search codebook is used to encode LPC parameters obtained from amplitudes. The algorithm calculated LPC in frequency domain and picked up harmonic offsets to ensure high quality of re-construct speech. Simulation results and the mean opinion score (MOS) indicate that the proposed algorithm can reduce speech coding rate with high accuracy for speech parameter analysis, easy on-line realization and satisfied synthesis speech quality. (4 pages)The optimization and real-time implementation of G.729A codec on TMS320C5510
http://dl-live.theiet.org/content/conferences/10.1049/cp_20061449
Three layer optimizations in algorithm, DSP programming and compiling respectively for G.729A codec and real-time implementation on TMS320C5510 DSPs is described. A new algorithm is proposed for calculating line spectrum pair parameters with higher accuracy and less executing time consumption. The outstanding solutions of DSP programming and compiling optimizations for real-time implementation are also introduced. The optimal codes with different techniques are implemented and tested on TMS320C5510 DSPs. The test results of the optimal codes with DSP/BIOS programming show that the encoding and decoding are correct, real-time proceeding and computing delay is much smaller than the required minimum delay, the whole performance of G.729A codec is much improved. (4 pages)Development of broadcast technologies for mobile TV
http://dl-live.theiet.org/content/conferences/10.1049/ic_20050006
This presentation discussed the issues, technologies, and spectrum implications of broadcast technologies for mobile TV. The presentation aims to address such broadcasting dilemmas as limited battery capacity, limited processing power, and competing demands for antenna/silicon space. It also gives the advantages and disadvantages of the following technologies: DVB-H, DMB, and FLO. (16 pages)A novel MB selection method for adaptive intra refresh in H.264/AVC video coding standard
http://dl-live.theiet.org/content/conferences/10.1049/cp_20050110
In this paper, we investigate the H.264/AVC codec from two angles. The first is complexity of the reference H.264/AVC encoder. The second is error resilience update of the encoder using adaptive intra refresh (AIR). We propose a novel intra macroblock (MB) selection method to improve the error robustness of the H.264 codec. The proposed algorithm outperforms the one in the reference H.264/AVC codec without any increase in computational complexity.Perceptual quality of H.264/AVC deblocking filter
http://dl-live.theiet.org/content/conferences/10.1049/cp_20050116
The H.264/AVC standard defines an optional in-loop deblocking filter. The effect of this filter on subjective video quality is investigated. Filter settings preferred by users are recorded for a group of 82 users across a range of video sequences and coded bitrates. The results indicate two clear groupings of user preferences for low- and medium-activity sequences. There is no clear user preference when the sequence contains high motion and activity. The implications of these results for performance optimisation of H264/AVC codecs are discussed.Energy optimization for mobile MPEG-4 video decoder
http://dl-live.theiet.org/content/conferences/10.1049/cp_20051484
Most compiler optimization techniques concern most about speed. In this paper, we present two high-level power/energy optimization methods for ARM-based battery-powered embedded multimedia systems, e.g. mobile phones, pocket PCs, personal multimedia systems, etc. The experiments using MPEG-4 simple profile level 0 (SP@L0) video decoder on ARM920T with two QCIF video sequences 15 fps, 24 kbps show that the proposed techniques can complement the existing speed-oriented ones to achieve lower energy/power consumption up to 13% relative to all ARM C++ optimization levels despite the 16-KB instruction and 16-KB data caches of ARM 920T core. (6 pages)SMPTE D-16 an MPEG-4 video
http://dl-live.theiet.org/content/conferences/10.1049/ic_20040574
This paper is a powerpoint presentation of the provisions in SMPTE D-16 that pertain to MPEG-4 video. The presentation gives a detailed analysis of the coding process in MPEG-4 as well as the development of video codecs. The MPEG-4 has two encoding modes in discrete cosine transform and has two profiles, namely the core studio profile and simple studio profile. Comparisons were also presented regarding the two coding methods.SMPTE VC-1 [Microsoft Windows media]
http://dl-live.theiet.org/content/conferences/10.1049/ic_20040564
The paper is a powerpoint presentation on the Microsoft implementation of SMPTE VC-1. This paper emphasizes that Windows Media Audio and Video 9 Series/VC-1 codec can be used in a wide range of bit rates and resolutions, any CE device, any transport stream and any optical disc format. It includes a subject comparison between the WMV-9, MPEG and DVD formats.Scalable video requirements for surveillance applications
http://dl-live.theiet.org/content/conferences/10.1049/ic_20040091
An example of a distributed video surveillance system is presented and the requirements on the video codec for such a system are developed. An overview of scalable video coding technologies is then provided which concludes that a wavelet based video codec offers many advantages for a distributed video surveillance system. In addition, wavelet based coding technologies offer significant error resilience properties, making them particularly robust for use on mobile networks.A comparison of LSF quantization techniques for 3G mobile communications
http://dl-live.theiet.org/content/conferences/10.1049/cp_20040613
This paper uses the 3<sup xmlns="http://pub2web.metastore.ingenta.com/ns/">rd</sup> Generation (3G) adaptive multi rate (AMR) speech coder to evaluate the performance of four weighting functions (WF) for use in the quantisation of line spectral frequency (LSF) values. The LSF are a transformation of the speech synthesis filter coefficients. Each WF performance is analysed in conjunction with both split vector and split matrix LSF quantisation techniques. A spectral distortion (SD) measure using the quantised and unquantised LSF is generated to determine the best WF for each quantiser. The results show that the adaptive WF (AWF) performs best in the split vector case and the 3G AMR WF performs best in the split matrix case.Reconfigurable software based communication - video services in reconfigurable mobile devices
http://dl-live.theiet.org/content/conferences/10.1049/cp_20040709
To accommodate and facilitate the flexibility required to deliver future multimedia based services in heterogeneous/composite radio environments, communications systems will need to be increasingly software based. The paper describes an implementation of the reconfiguration management architecture (RMA) as facilitator for runtime reconfiguration of video codecs within the framework of a reconfigurable software based communication system (RSS). The paper describes the testbed implementation and highlights the considerations that are needed for implementation of portable video codecs.Codec mode selection in EDGE AMR channels
http://dl-live.theiet.org/content/conferences/10.1049/cp_20030265
We examine the dynamic behaviour of adaptive multi rate (AMR) traffic channels in an EDGE system. The goal of AMR is to provide optimum speech quality for any channel condition. A secure decision concerning the optimum codec mode is as important as a quick reaction to channel quality changes. It is shown that these competing demands cannot both be matched in all cases with the ETSI proposal (see ETSI TSG GERAN 09, Technical Specification 5.5.0, 2002). An extended algorithm for a better decision process is introduced.Macroblock skip-mode prediction for complexity control of video encoders
http://dl-live.theiet.org/content/conferences/10.1049/cp_20030473
We propose a macroblock skip-mode prediction algorithm to reduce the computational effort of video encoders. The algorithm classifies each macroblock as "skipped" or "not skipped" by estimating the energy of low-frequency quantized coefficients prior to coding, making it possible to significantly reduce computation by not coding these skipped macroblocks. Results show that the algorithm can achieve substantial computational savings with only a small degradation in rate-distortion performance.Amending the syntax of the MPEG-4 Simple Scalable Profile to use error resilience tools
http://dl-live.theiet.org/content/conferences/10.1049/cp_20030472
This paper describes an amendment to the ISO/IEC 14496-2 standard commonly known as MPEG-4 video which incorporates error resilience tools in the Simple Scalable Profile of the MPEG-4 video codec, such that scalable MPEG-4 is made suitable for deployment in a mobile communication environment. This is known as the Error Resilient Simple Scalable Profile (ER-SSP). When it was defined in 2000, the syntax of the Simple Scalable Profile (SSP) prohibited the use of the error resiliency tools which were available in the base layer of the MPEG-4 codec, and which are essential for use of MPEG-4 in a mobile communication environment. This paper reports on the rationale behind the newly defined Error Resilient Simple Scalable Profile within the MPEG standard, describes the syntax for the new profile, and explains the suitability of incorporating the error resiliency tools of the base layer into the enhancement layer. It is shown that minor modifications are required in the Header Extension to synchronize the decoding process between the two layers. Hence it is shown that base layer error resilience tools are equally applicable to the enhancement layer with nominal syntax changes.Convolutional decoding for reconfigurable mobile systems
http://dl-live.theiet.org/content/conferences/10.1049/cp_20010060
The advent of enabling technologies for reconfigurable logic processing requires that investigation into new design methodologies is made before they can be used effectively. This paper presents the design of an FPGA configuration for the decoding of convolutional codes. These codes provide resilience in noisy transmission conditions for many different digital communication systems. The decoder can be reconfigured to decode any convolutional code up to constraint length 9 and any rate to a minimum of 1/6. This investigative design reveals methods on how to implement parametable algorithms using configurable logic.Packet loss resilient, scalable audio compression and streaming for IP networks
http://dl-live.theiet.org/content/conferences/10.1049/cp_20010024
Current popular Internet audio streaming solutions impose a division between source coding (provided, for example, by MPEG Layer III-MP3) and channel coding, which is accomplished in the server, typically by means of packet retransmission. We present a novel joint source and channel coder which provides packet loss recovery and continuous bit rate scalability. These functionalities are well suited to streaming audio over 3<sup xmlns="http://pub2web.metastore.ingenta.com/ns/">rd</sup> and future generation wireless broadband networks.Error coding strategies for MELP vocoder in wireless and ATM environments
http://dl-live.theiet.org/content/conferences/10.1049/ic_20000046
The U.S. Government has developed a new military standard (U.S. MilStd3005) 2400 bps voice coding algorithm known as MELP (mixed excitation linear prediction). This vocoder has quite acceptable voice quality and intelligibility under benign error channel conditions. However, when it is subjected to high error rates (due to noise and fading) as may be experienced in tactical wireless operations, amelioration techniques can be utilized which take advantage of the underlying inter-frame residual redundancy statistics of the specific MELP parameters. In wireless ATM system implementations, the vocoder error correction and amelioration technique must deal with the potential loss of ATM cells (in addition to the traditional RF impairments). This paper describes simulation experiments conducted on the MELP vocoding algorithm in both scenarios and notes that optimization strategies are highly dependent on both the error coding overhead allowed and the expected nature of the bitstream error and loss statistics. (33 pages)Performance of voice coders under military conditions
http://dl-live.theiet.org/content/conferences/10.1049/ic_20000044
Discusses voice coding in military situations. Most vocoder systems are specified and sold on the basis of good quality speech being available at the input terminal. The exception being the use of a hands-free microphone in an automobile. Military vehicles including armoured vehicles and helicopters can produce very high acoustic fields with components in the critical band which includes the fundamental larynx pitch. Modern approaches to warfighting put emphasis on coalition operations and interoperability is the name of the game. Although some international standards for military voice coding exist, it is highly likely that a tandem circuit between two allies will involve the transcoding of speech between two differing coding schemes. The advent of lower rate voice coding in civil systems also brings the problem of military systems interoperating with civil ones. (7 pages)Design and integration of voice Codec cell on a Bluetooth<sup xmlns="http://pub2web.metastore.ingenta.com/ns/">TM</sup> baseband IC
http://dl-live.theiet.org/content/conferences/10.1049/ic_20000518
The MT1020A device from MITEL Semiconductor is a highly integrated single chip Bluetooth baseband processor targeted at low power applications. This paper describes the design of a voice CODEC with the constraints that it be low power and work in the noisy environment of a large digital IC. These constraints are typical in the system-on-a-chip era, particularly for communications applications. (7 pages)Real-time video communications over GPRS
http://dl-live.theiet.org/content/conferences/10.1049/cp_20000085
One of the stated aims of 3<sup xmlns="http://pub2web.metastore.ingenta.com/ns/">rd</sup> generation mobile communications systems is to provide enhanced multimedia services to the user. Such services will include an integration of video, high-quality audio and data services. This paper describes a real-time mobile video communications system which employs ETSI's GPRS mobile packet system as a mobile access medium. Subjective performance tests are carried out for the error-resilient MPEG-4 video decoder using a real-time GPRS radio access emulator.High performance reconfigurable digital signal processing for the time varying channel
http://dl-live.theiet.org/content/conferences/10.1049/cp_20000184
In the time varying channel there is a requirement for the communication system to vary its function to adapt to the changing transmission conditions. The need for greater transmission capacity puts increasing demand on the processing requirements of such systems. This paper describes the concept of reconfigurable digital processing and highlights some of the problems encountered while designing such systems. An example of a reconfigurable channel coding algorithm is presented which illustrates the difficulty of designing reconfigurable systems particularly for adaptable communications systems such as those used for HF communication. This example illustrates the use of reconfigurable systems based around Golay and Hamming codes which are commonly used in HF systems. A design methodology is presented with software and hardware tools to enable the reconfigurable processing designer to easily design a reconfigurable digital processing system for communication in a time varying environment.Reliable vocoder data transmission over fading HF channels
http://dl-live.theiet.org/content/conferences/10.1049/cp_20000141
The quality of shortwave channels may vary considerably due to the time-specific changes of the transmission conditions. So the correct transmission of a continuous data stream, e.g. digitized speech signals output by a vocoder cannot be guaranteed under adverse conditions even by means of complex error correction measures or costly high transmitter power. This results in the loss of vocoder data-the transmission of speech information is partly or fully interrupted which impairs intelligibility. For this reason we propose a voice transmission system as a solution to the problem of correct voice data transmission. It is a method that offers quasi-real-time behaviour for channels that are subject to no interference or only slight interference, and that offers complete and correct data transmission with reduced real-time behaviour (mailbox operation) in the presence of strong or long-term interference and fading. To reach intelligibility at the transmitter end, the vocoder data are first divided into intelligible blocks, e.g. into coherent words between two speech pauses and stored in a buffer. The blocks are then transferred to the lower layers of the ISO/OSI reference model where an ARQ protocol is employed which leads to correct data transmission. At the receiver end, the blocks can then be listened to individually or as a whole and, if necessary, repeated once or several times (mailbox operation). This method was tested over a radio link of 800 km between Rostock University and Rohde & Schwarz Munich laboratory. 1200 bps vocoders from Comsat Co. and XK 2100 transceivers with packet radio protocol (PRP) RSX.25 from Rohde & Schwarz were used.Test Card ‘M’ - do you get the picture?
http://dl-live.theiet.org/content/conferences/10.1049/ic_19990414
Problems in digital TV are already appearing. There is an absolute and immediate need to devise systems of test that allow the consumer products of today to continue to work in the future, as the full features of the DVB and Digital Television Group specifications are rolled out. Tests with test card `M' have confirmed that receivers and set top boxes do not yet fully comply with these specifications. The major role played by test card `M' in identifying and helping to correct these deficiencies is presented in this paper. Digital television systems are extremely complex, making them very difficult to test. Test card `M' is a suite of specially crafted MPEG bitstreams which provides the solution for repeatable and structured testing without the need for sophisticated test equipment to evaluate the results and trace faults. This paper describes the contents of test card `M' and our experiences of using it with the new generation of digital terrestrial receivers. By testing the decoder designs at the early stages, broadcasters and manufacturers can be assured that all the advanced features of digital broadcasting will be available when the need arises. The experience gained in test card `M' development and application will be invaluable for testing of future digital TV systems in the UK and elsewhere. (5 pages)Turbo4: a high bit-rate chip for turbo code encoding and decoding
http://dl-live.theiet.org/content/conferences/10.1049/ic_19990784
This paper deals with an experimental IC developed for encoding and decoding turbo codes. The chip includes an encoder and a decoding module which performs one iteration of the decoding process. All the necessary interleaving memories and delay-lines are included in the circuit. The encoder is made up of a parallel concatenation of 2 convolutional encoders (constraint length=5) separated by an interleaver (64×32 matrix). The decoder uses the SOVA technique and a dedicated module achieves the synchronization task as well as a supervision function. Very high level performances can be achieved in 5 iterations: with a QPSK modulation, a BER of 2.E-8 is obtained with Eb/No=2 dB. The turbo4 chip can work in continuous mode up to 54 Mbits/s useful data throughput and is well suited for data flow applications such as video broadcasting. The IC is designed in a 0.25 μm CMOS technology and its core size is less than 8 mm<sup xmlns="http://pub2web.metastore.ingenta.com/ns/">2</sup>. (5 pages)Resilient video compression using absolute value coding
http://dl-live.theiet.org/content/conferences/10.1049/cp_19990391
In this paper we consider the problem of temporal channel error propagation, within a motion compensated video codec. To combat this problem we propose a new coding strategy, based on the coding of absolute values, rather than differential values. The proposed technique is flexible enough to be useful with a wide variety of coding methods, including DCT, wavelet coding and matching pursuits. Results are presented which demonstrate that the proposed technique allows much quicker recovery from errors than other existing techniques such as periodic replenishment. The method has the additional advantage, that the important active regions of the image are recovered more quickly than stationary background areas.Wavelet-based image sequence coding for transmission over HF-channels
http://dl-live.theiet.org/content/conferences/10.1049/cp_19990387
The authors propose a new combined source-channel coding framework. They introduce a new source codec, which follows a multiresolution concept facilitating a decoding at variable spatial resolutions. This property can be used for a source-encoder controlled UEP. Dependent upon the importance of the actual part of image information, the source encoder tells the channel encoder which grade of error protection is to apply. Furthermore, using this strategy it is possible to adapt the protection strength to the channel state. The new channel codec was designed for digital channels with memory.Audio compression using wavelets
http://dl-live.theiet.org/content/conferences/10.1049/ic_19980819
This paper outlines the work on audio compression using wavelets which has been carried out at King's College London. It describes the MPEG-audio coding standard, upon which the wavelet codec is based, the wavelets and their use in audio compression, and some improvements which have been made to the codec. (7 pages)Low bit-rate frequency extension coding
http://dl-live.theiet.org/content/conferences/10.1049/ic_19980820
There are now a number of applications, most notably streamed Internet audio, which require audio and speech to be encoded at a low bit rate, typically 16 kbit/sec or below. To achieve an acceptable quality, the original signal is normally low-pass filtered to somewhere between 4 and 5.5 kHz before encoding. Rather than discard the upper frequency band completely, we propose encoding just the noise component of it using about 500 bits/sec. This greatly enhances contemporary music and close-microphone speech, but has little effect on classical music. The process can be used to enhance any audio or speech codec, knowing only its encoding/decoding delay. (5 pages)K ring compression codec
http://dl-live.theiet.org/content/conferences/10.1049/ic_19980830
The background to a DSP algorithm which will revolutionise digital audio storage and transmission is presented. The algorithm is for the k ring compression codec (k codec) which has been in development for a number of years. This kodec is unique among compression codecs as it performs data absorption into a fixed size digital memory. It works with all types of data including samples, effects algorithms, sequence data as well as digital audio data streams. The algorithm is not lossy and is suitable for compressing a complete sampler memory dump so as to fit on the floppy disk included as standard, and still have space for plenty more audio data. The kodec is not limited to audio storage, and so will find many applications in the multimedia market, video recorders, computers and communications. Further development is underway for distributing the Internet possibly using a piggyback signal on standard atomic clock time signals. Mass storage media connected via SCSI is set to become redundant, reducing the cost of sampler technology to the price of CD players when first released. The sampler is the ideal music source device for digital music. The interactivity of sampler technology will bring new possibilities to the home music scene for remixing in your living room. (2 pages)A software platform for multiway audio distribution over the Internet
http://dl-live.theiet.org/content/conferences/10.1049/ic_19980821
The Robust Audio Tool (RAT) allows users to achieve real-time multiway communication over the Internet. It was initially intended for use in multiway conferences, but is being used as an Internet audio broadcast application, by radio stations in the US and elsewhere. RAT can also be used in a point-to-point manner, and as a transcoder between networks of differing capabilities, e.g. for mobile access to the Internet. The emphasis of work in RAT has been on maximising the audio quality despite inherent problems of packet transport, processor scheduling and audio capabilities of the end system. The important features of RAT, in comparison to other Internet audio tools, is that it is able to support multirate processing, has no restrictions on audio frame duration, and supports multi-channel audio, and both fixed and variable size audio frames. We discuss methods of real-time multimedia delivery, and identify issues of particular importance for music transmission over the Internet. For music coding researchers interested in using RAT to exploit their research, we present an overview of the architecture of the RAT and specifically focus on codec integration. Finally, we present some off-line performance measurements of a public domain MPEG1 music codec that has been integrated into the RAT, and illustrate the Internet performance in terms of packet loss, and variable transit delays. (6 pages)ATM based video and audio codecs
http://dl-live.theiet.org/content/conferences/10.1049/ic_19970614
This paper presents the ATM-related technical features of the range of ATM based video and audio hardware codecs manufactured by FORE Systems. These devices are stand-alone and attach directly to the ATM network and are thus completely decoupled from any particular hardware and operating systems platform. The devices are managed from a remote workstation or PC. Video and audio sourced from the hardware codecs can be displayed on workstations and PCs. There are separate unidirectional codecs: the AVA-300 for encode, the ATV-300 for decode. Proxy signalling enables the construction of simple, cheap devices which still support full network signalling. The use of native ATM AALS transport provides high performance and low latency data transfer. ATM multicast and QoS support allow video and audio to co-exist with data applications. Motion-JPEG provides TV-quality video at medium data rates. M-JPEG combined with variable-Q allows maximum use to be made to be made of the available bandwidth. Finally, the CellChain reduces costs by allowing multiple devices to share a single switch port. (6 pages)Study of wavelet decompositions for image/video compression by software codecs
http://dl-live.theiet.org/content/conferences/10.1049/cp_19970854
Although other workers have examined the quality/compression tradeoff for various wavelet approaches, as far as we are aware, no adequate study has been made of the tradeoff between fidelity, processing requirement, and bandwidth. This is crucial to the design of software-only codecs dependent on utilising the main system processing resource. This paper describes on-going work aimed at defining an optimum architecture for such components, and presents results of a systematic study of the standard and non-standard decomposition approaches in conjunction with the Haar wavelet transformation For all images the non-standard decomposition is markedly superior to the standard decomposition achieving over four times the compression for similar quality of reproduction. The algorithm for the non-standard decomposition and the standard decomposition are both of outstandingly low computational complexity with the non-standard being slightly quicker. The results show that the non-standard decomposition in conjunction with the Haar wavelet is an excellent choice as the algorithmic base for a software-only codec.Video surveillance using low bandwidth, high compression systems
http://dl-live.theiet.org/content/conferences/10.1049/cp_19970415
The paper discusses recent developments in video coding and compression that have been undertaken at Bath University and in collaboration with industrial partners in the UK and EU. Conventional CCTV security systems are analysed and the requirements for digital video compression outlined. Having summarised the main video coding systems currently in use, the paper describes specific features in the Bath Scaleable Video codec (BSV). The results of employing this codec in a security application are shown and the principal benefits are summarised.Subjective testing of MPEG-2 NBC multichannel audio coding
http://dl-live.theiet.org/content/conferences/10.1049/cp_19971316
This paper presents an overview of the preparations for, and the results of, the subjective listening tests on the new MPEG-2 non-backwards compatible (NBC) multichannel audio coding algorithm carried out by the BBC and NHK during September and October 1996. The codecs tested were: MPEG-2 NBC at 256 kbit/s, MPEG-2 NBC at 320 kbit/s, MPEG-2 NBC low-complexity at 320 kbit/s and the 1995 version of MPEG-2 Layer II at 640 kbit/s operating in a backwards compatible mode. The results showed good performance for all of the codecs. The MPEG-2 NBC codec at 320 kbit/s generally performed better than the other codecs and, although not quite transparent for a few test excerpts under these rigorous conditions, it passed the EBU criterion for “indistinguishable quality”. The MPEG-2 NBC low complexity version at 320 kbit/s was, by a small margin, not quite as good as that of MPEG-2 NBC at 320 kbit/s.Wavelets versus conventional filters for low bit rate audio coding
http://dl-live.theiet.org/content/conferences/10.1049/cp_19971290
Wavelet filtering is a promising tool for use in audio signal compression. What is still lacking, however, is a thorough understanding of wavelet filters performance relative to the more sophisticated examples of conventional filters. So, as we seek to apply wavelet filters to low bit rate audio coding, attention must be focused not only to the bit-rate/signal-quality trade-off, but also the complexity and processing delay should not be underemphasised. The results presented in this paper attempt to clarify these issues. To assess the coding gain of wavelet and conventional filters, various codec models have been designed and implemented based on a wavelet packet algorithm, an auditory perception model and entropy noiseless coding. The wavelet packet based coding approach is compared to the MPEG-audio international standard in terms of objective and subjective measurements and is shown to be superior to MPEG-audio layer I and competitive with layer II.NewsDepot: integrating real time and file audio contributions
http://dl-live.theiet.org/content/conferences/10.1049/cp_19971265
The advance of digital technologies for gathering, editing, programming and distribution of news, sports and current affairs at radio stations can no longer be stopped. Curiously enough separate systems have been created for the handling of “live” audio contributions and for the handling of file transfer. At the most there is a primitive connection between the two. It would certainly simplify life if “live” and “file” contributions could be interchanged in the “real time” mode. At this moment this is not the case. The article describes an integrated solution in which both forms cannot only be used in parallel, but also mixed. This solution is being brought to market under the name NewsDepot. The NewsDepot is a gateway for contributions coming from “outside” to either the normal “file” orientated broadcast automation system or the real time audio codecs in the studio.A human vision system model for objective picture quality measurements
http://dl-live.theiet.org/content/conferences/10.1049/cp_19971319
It is often useful to know the perceptual impact of distortions introduced at critical points in the production, distribution and display of video. Direct measurement using human observers is possible for some applications, but this is a time consuming operation that must be performed under carefully controlled conditions. A faster and more easily standardized approach is to use a vision model that provides accurate estimates of the visibility of differences between original and distorted image sequences. A model that fulfils this need is the Sarnoff just-noticeable difference (JND) model. This model is based on known physiological and psychophysical principles of human visual discrimination performance, rather than on particular classes of image distortions, and is therefore more robust across the often unpredictable range of distortions that can occur in modern digital video. The operation and general structure of the JND model are described, and its performance in a range of video applications is discussed.Aspects of error resilience for block based video coders in multimedia communications
http://dl-live.theiet.org/content/conferences/10.1049/ic_19961331
There has been a gradually increasing need for using video in multimedia communication services. As there is a big tendency to move towards a third-generation universal telecommunication environment, the necessity to carry video information on mobile networks increases. Since mobile environments naturally impose a high level of hostility against carried intelligence, error resilience techniques must be formulated and applied on video coders to render their bit streams more immune to deteriorated channel conditions. The authors discuss bit rate variability in block based video coders, the effects of channel errors on an H.263 video bit stream, and error resilience improvement. (6 pages)Inmarsat aeronautical public correspondence services: satellite facsimile and voice-band data for air travellers
http://dl-live.theiet.org/content/conferences/10.1049/cp_19960434
Rapid growth of aeronautical voice traffic and the determination to keep up with the needs and desires of air travellers sets the challenging scenario that Inmarsat is facing in providing aeronautical communications today and in the future. By means of two advanced, digital codecs for voice/facsimile/data transmission, Inmarsat can support a wide variety of aeronautical traffic and services for any type of aircraft, ranging from inter-continental to light corporate jets.