Improving the robustness of CELP-like speech decoders using late-arrival packets information : application to G.729 standard in VoIP

L'utilisation de la voix sur Internet est une nouvelle tendance dans Ie secteur des télécommunications et de la réseautique. La paquetisation des données et de la voix est réalisée en utilisant Ie protocole Internet (IP). Plusieurs codecs existent pour convertir la voix codée en paquets. La voix codée est paquetisée et transmise sur Internet. À la réception, certains paquets sont soit perdus, endommages ou arrivent en retard. Ceci est cause par des contraintes telles que Ie délai («jitter»), la congestion et les erreurs de réseau. Ces contraintes dégradent la qualité de la voix. Puisque la transmission de la voix est en temps réel, Ie récepteur ne peut pas demander la retransmission de paquets perdus ou endommages car ceci va causer plus de délai. Au lieu de cela, des méthodes de récupération des paquets perdus (« concealment ») s'appliquent soit à l'émetteur soit au récepteur pour remplacer les paquets perdus ou endommages. Ce projet vise à implémenter une méthode innovatrice pour améliorer Ie temps de convergence suite a la perte de paquets au récepteur d'une application de Voix sur IP. La méthode a déjà été intégrée dans un codeur large-bande (AMR-WB) et a significativement amélioré la qualité de la voix en présence de <<jitter » dans Ie temps d'arrivée des trames au décodeur. Dans ce projet, la même méthode sera intégrée dans un codeur a bande étroite (ITU-T G.729) qui est largement utilise dans les applications de voix sur IP. Le codeur ITU-T G.729 défini des standards pour coder et décoder la voix a 8 kb/s en utilisant 1'algorithme CS-CELP (Conjugate Stmcture Algebraic Code-Excited Linear Prediction).Abstract: Voice over Internet applications is the new trend in telecommunications and networking industry today. Packetizing data/voice is done using the Internet protocol (IP). Various codecs exist to convert the raw voice data into packets. The coded and packetized speech is transmitted over the Internet. At the receiving end some packets are either lost, damaged or arrive late. This is due to constraints such as network delay (fitter), network congestion and network errors. These constraints degrade the quality of speech. Since voice transmission is in real-time, the receiver can not request the retransmission of lost or damaged packets as this will cause more delay. Instead, concealment methods are applied either at the transmitter side (coder-based) or at the receiver side (decoder-based) to replace these lost or late-arrival packets. This work attempts to implement a novel method for improving the recovery time of concealed speech The method has already been integrated in a wideband speech coder (AMR-WB) and significantly improved the quality of speech in the presence of jitter in the arrival time of speech frames at the decoder. In this work, the same method will be integrated in a narrowband speech coder (ITU-T G.729) that is widely used in VoIP applications. The ITUT G.729 coder defines the standards for coding and decoding speech at 8 kb/s using Conjugate Structure Algebraic Code-Excited Linear Prediction (CS-CELP) Algorithm

Design of a digital compression technique for shuttle television

The determination of the performance and hardware complexity of data compression algorithms applicable to color television signals, were studied to assess the feasibility of digital compression techniques for shuttle communications applications. For return link communications, it is shown that a nonadaptive two dimensional DPCM technique compresses the bandwidth of field-sequential color TV to about 13 MBPS and requires less than 60 watts of secondary power. For forward link communications, a facsimile coding technique is recommended which provides high resolution slow scan television on a 144 KBPS channel. The onboard decoder requires about 19 watts of secondary power

Study and simulation of low rate video coding schemes

The semiannual report is included. Topics covered include communication, information science, data compression, remote sensing, color mapped images, robust coding scheme for packet video, recursively indexed differential pulse code modulation, image compression technique for use on token ring networks, and joint source/channel coder design

Motion compensated interpolation for subband coding of moving images

Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1994.Includes bibliographical references (leaves 108-119).by Mark Daniel Polomski.M.S

Audio Processing and Loudness Estimation Algorithms with iOS Simulations

abstract: The processing power and storage capacity of portable devices have improved considerably over the past decade. This has motivated the implementation of sophisticated audio and other signal processing algorithms on such mobile devices. Of particular interest in this thesis is audio/speech processing based on perceptual criteria. Specifically, estimation of parameters from human auditory models, such as auditory patterns and loudness, involves computationally intensive operations which can strain device resources. Hence, strategies for implementing computationally efficient human auditory models for loudness estimation have been studied in this thesis. Existing algorithms for reducing computations in auditory pattern and loudness estimation have been examined and improved algorithms have been proposed to overcome limitations of these methods. In addition, real-time applications such as perceptual loudness estimation and loudness equalization using auditory models have also been implemented. A software implementation of loudness estimation on iOS devices is also reported in this thesis. In addition to the loudness estimation algorithms and software, in this thesis project we also created new illustrations of speech and audio processing concepts for research and education. As a result, a new suite of speech/audio DSP functions was developed and integrated as part of the award-winning educational iOS App 'iJDSP." These functions are described in detail in this thesis. Several enhancements in the architecture of the application have also been introduced for providing the supporting framework for speech/audio processing. Frame-by-frame processing and visualization functionalities have been developed to facilitate speech/audio processing. In addition, facilities for easy sound recording, processing and audio rendering have also been developed to provide students, practitioners and researchers with an enriched DSP simulation tool. Simulations and assessments have been also developed for use in classes and training of practitioners and students.Dissertation/ThesisM.S. Electrical Engineering 201

Uplink data measurement and analysis for 5G eCPRI radio unit

Abstract. The new 5G mobile network generation aims to enhance the performance of the cellular network in almost every possible aspect, offering higher data rates, lower latencies, and massive number of network connections. Arguably the most important change from LTE are the new RU-BBU split options for 5G promoted by 3GPP and other organizations. Another big conceptual shift introduced with 5G is the open RAN concept, pushed forward by organizations such as the O-RAN alliance. O-RAN aims to standardize the interfaces between different RAN elements in a way that promotes vendor interoperability and lowers the entry barrier for new equipment suppliers. Moreover, the 7-2x split option standardized by O-RAN has risen as the most important option within the different low layer split options. As the fronthaul interface, O-RAN has selected the packet-based eCPRI protocol, which has been designed to be more flexible and dynamic in terms of transport network and data-rates compared to its predecessor CPRI. Due to being a new interface, tools to analyse data from this interface are lacking. In this thesis, a new, Python-based data analysis tool for UL eCPRI data was created for data quality validation purposes from any O-RAN 7-2x functional split based 5G eCPRI radio unit. The main goal for this was to provide concrete KPIs from captured data, including timing offset, signal power level and error vector magnitude. The tool produces visual and text-based outputs that can be used in both manual and automated testing. The tool has enhanced eCPRI UL datapath testing in radio unit integration teams by providing actual quality metrics and enabling test automation.Uplink datamittaukset ja -analyysi 5G eCPRI radiolla. Tiivistelmä. Uusi 5G mobiiliverkkogeneraatio tuo mukanaan parannuksia lähes kaikkiin mobiiliverkon ominaisuuksiin, tarjoten nopeamman datasiirron, pienemmät viiveet ja valtavat laiteverkostot. Luultavasti tärkein muutos LTE teknologiasta ovat 3GPP:n ja muiden organisaatioiden ehdottamat uudet radion ja systeemimoduulin väliset funktionaaliset jakovaihtoehdot. Toinen huomattava muutos 5G:ssä on O-RAN:in ajama avoimen RAN:in konsepti, jonka tarkoituksena on standardisoida verkkolaitteiden väliset rajapinnat niin, että RAN voidaan rakentaa eri valmistajien laitteista, laskien uusien laitevalmistajien kynnystä astua verkkolaitemarkkinoille. O-RAN:n standardisoima 7-2x funktionaalinen jako on noussut tärkeimmäksi alemman tason jakovaihtoehdoista. Fronthaul rajapinnan protokollaksi O-RAN on valinnut pakettitiedonsiirtoon perustuvan eCPRI:n, joka on suunniteltu dynaamisemmaksi ja joustavammaksi datanopeuksien ja lähetysverkon suhteen kuin edeltävä CPRI protokolla. Uutena protokollana, eCPRI rajapinnalle soveltuvia data-analyysityökaluja ei ole juurikaan saatavilla. Tässä työssä luotiin uusi pythonpohjainen data-analyysityökalu UL suunnan eCPRI datalle, jotta datan laatu voidaan määrittää millä tahansa O-RAN 7-2x funktionaaliseen jakoon perustuvalla 5G eCPRI radiolla. Työkalun päätarkoitus on analysoida ja kuvata datan laatua laskemalla datan ajoitusoffsettia, tehotasoa, sekä EVM:ää. Työkalu tuottaa tulokset visuaalisena ja tekstipohjaisena, jotta analyysia voidaan tehdä niin manuaalisessa kuin automaattisessa testauksessa. Työkalun käyttöönotto on tehostanut UL suunnan dataputken testausta radio-integrointitiimeissä, tarjoten datan laatua kuvaavaa metriikkaa sekä mahdollistaen testauksen automatisoinnin

FEC for efficient video transmission over CDMA

Video is one of the most important and also one of the most challenging types of traffic on communication networks. One of the challenges arises because communication networks can insert errors into video, and also compressed video is fragile in the presence of errors; that is a single error can propagate over a large portion of the video. This thesis describes how to protect the content of video during transmission by employing error correcting codes. It deals with the concept of transmitting video packets with redundancy embedded in it for attempting error recovery and improving the quality at the receiver side. The effect of bit errors on the overall video quality has been studied. This thesis throws light on the importance of each bit in a video frame to the overall received quality. The effect of the important bits has been monitored by performing a “bit killing” operation, wherein the critical bits are flipped. We have adopted BCH codes to improve the quality of video. Two types of FEC schemes have been implemented. In the first FEC scheme, data is not prioritized and various segments in the video stream are given equal importance. BCH (63, 51, 2) is used for this case. In the second scheme, a stronger FEC is applied to the first frame (the first I frame) as the important data is found in it. Hence BCH (63, 24, 7) is used for the first frame and BCH (63, 57, 1) is used for the rest inter frames. The main aim of the algorithm is to strengthen the video stream against most possible error cases and also reduce the percentage of the overhead bits arising because of BCH coding. The strength of the BCH is also varied to study the effect of signal-to-noise ratio on the BER. All discussions are furnished with simulation results. The primary target is the transmission of video through CDMA channels

Handling Massive N-Gram Datasets Efficiently

This paper deals with the two fundamental problems concerning the handling of large n-gram language models: indexing, that is compressing the n-gram strings and associated satellite data without compromising their retrieval speed; and estimation, that is computing the probability distribution of the strings from a large textual source. Regarding the problem of indexing, we describe compressed, exact and lossless data structures that achieve, at the same time, high space reductions and no time degradation with respect to state-of-the-art solutions and related software packages. In particular, we present a compressed trie data structure in which each word following a context of fixed length k, i.e., its preceding k words, is encoded as an integer whose value is proportional to the number of words that follow such context. Since the number of words following a given context is typically very small in natural languages, we lower the space of representation to compression levels that were never achieved before. Despite the significant savings in space, our technique introduces a negligible penalty at query time. Regarding the problem of estimation, we present a novel algorithm for estimating modified Kneser-Ney language models, that have emerged as the de-facto choice for language modeling in both academia and industry, thanks to their relatively low perplexity performance. Estimating such models from large textual sources poses the challenge of devising algorithms that make a parsimonious use of the disk. The state-of-the-art algorithm uses three sorting steps in external memory: we show an improved construction that requires only one sorting step thanks to exploiting the properties of the extracted n-gram strings. With an extensive experimental analysis performed on billions of n-grams, we show an average improvement of 4.5X on the total running time of the state-of-the-art approach.Comment: Published in ACM Transactions on Information Systems (TOIS), February 2019, Article No: 2