DWT-DCT-Based Data Hiding for Speech Bandwidth Extension

The limited narrowband frequency range, about 300-3400Hz, used in telephone network channels results in less intelligible and poor-quality telephony speech. To address this drawback, a novel robust speech bandwidth extension using Discrete Wavelet Transform- Discrete Cosine Transform Based Data Hiding (DWTDCTBDH) is proposed. In this technique, the missing speech information is embedded in the narrowband speech signal. The embedded missing speech information is recovered steadily at the receiver end to generate a wideband speech of considerably better quality. The robustness of the proposed method to quantization and channel noises is confirmed by the mean square error test. The enhancement in the quality of reconstructed wideband speech of the proposed method over conventional methods is reasserted by subjective listening and objective tests

Quality of media traffic over Lossy internet protocol networks: Measurement and improvement.

Voice over Internet Protocol (VoIP) is an active area of research in the world of communication. The high revenue made by the telecommunication companies is a motivation to develop solutions that transmit voice over other media rather than the traditional, circuit switching network. However, while IP networks can carry data traffic very well due to their besteffort nature, they are not designed to carry real-time applications such as voice. As such several degradations can happen to the speech signal before it reaches its destination. Therefore, it is important for legal, commercial, and technical reasons to measure the quality of VoIP applications accurately and non-intrusively. Several methods were proposed to measure the speech quality: some of these methods are subjective, others are intrusive-based while others are non-intrusive. One of the non-intrusive methods for measuring the speech quality is the E-model standardised by the International Telecommunication Union-Telecommunication Standardisation Sector (ITU-T). Although the E-model is a non-intrusive method for measuring the speech quality, but it depends on the time-consuming, expensive and hard to conduct subjective tests to calibrate its parameters, consequently it is applicable to a limited number of conditions and speech coders. Also, it is less accurate than the intrusive methods such as Perceptual Evaluation of Speech Quality (PESQ) because it does not consider the contents of the received signal. In this thesis an approach to extend the E-model based on PESQ is proposed. Using this method the E-model can be extended to new network conditions and applied to new speech coders without the need for the subjective tests. The modified E-model calibrated using PESQ is compared with the E-model calibrated using i ii subjective tests to prove its effectiveness. During the above extension the relation between quality estimation using the E-model and PESQ is investigated and a correction formula is proposed to correct the deviation in speech quality estimation. Another extension to the E-model to improve its accuracy in comparison with the PESQ looks into the content of the degraded signal and classifies packet loss into either Voiced or Unvoiced based on the received surrounding packets. The accuracy of the proposed method is evaluated by comparing the estimation of the new method that takes packet class into consideration with the measurement provided by PESQ as a more accurate, intrusive method for measuring the speech quality. The above two extensions for quality estimation of the E-model are combined to offer a method for estimating the quality of VoIP applications accurately, nonintrusively without the need for the time-consuming, expensive, and hard to conduct subjective tests. Finally, the applicability of the E-model or the modified E-model in measuring the quality of services in Service Oriented Computing (SOC) is illustrated

The Future of the Internet III

Presents survey results on technology experts' predictions on the Internet's social, political, and economic impact as of 2020, including its effects on integrity and tolerance, intellectual property law, and the division between personal and work lives

Media gateway utilizando um GPU

Mestrado em Engenharia de Computadores e Telemátic

Recent Advances in Signal Processing

The signal processing task is a very critical issue in the majority of new technological inventions and challenges in a variety of applications in both science and engineering fields. Classical signal processing techniques have largely worked with mathematical models that are linear, local, stationary, and Gaussian. They have always favored closed-form tractability over real-world accuracy. These constraints were imposed by the lack of powerful computing tools. During the last few decades, signal processing theories, developments, and applications have matured rapidly and now include tools from many areas of mathematics, computer science, physics, and engineering. This book is targeted primarily toward both students and researchers who want to be exposed to a wide variety of signal processing techniques and algorithms. It includes 27 chapters that can be categorized into five different areas depending on the application at hand. These five categories are ordered to address image processing, speech processing, communication systems, time-series analysis, and educational packages respectively. The book has the advantage of providing a collection of applications that are completely independent and self-contained; thus, the interested reader can choose any chapter and skip to another without losing continuity

Campus Telecommunications Systems: Managing Change

The purpose of this book is to provide a broadbased understanding of the rapidly changing environment of campus telecommunications. The anticipated audience for this material is the non-technical university administrator who may not have direct responsibility for telecommunications, but has a need to understand the general environment in which his telecommunications manager functions and the basic concepts of the technology. Five topic areas were selected that best cover the preponderance of issues. No attempt has been made to associate or closely coordinate materials from one chapter\u27s subject to that of any other. Each chapter generally stands alone. In total, however, the five chapters address the topics and issues that most often generate inquiries from university administrators outside the telecommunications department. Introduction 1 The Changing Telecommunications Environment 2 Telecommunications Technology and the Campus 3 Student Services 4 Financing a New Telecommunications System . 5 Selecting a Consultant Glossary Inde

Measurements in Perceptual Annoyance of Audio Coding Artifacts

Tässä diplomityössä tutkitaan matalan bittinopeuden puhe- ja audiokooderin USACin kehityksessä merkittäväksi koettujen koodausartifaktien psykoakustista ärsyttävyyttä. Tutkielmassa käsitellään neljää ilmiötä, jotka on eritelty alempana. Artifaktit mallinnettiin MATLAB(R)-ohjelmistolla ja niiden ärsyttävyyttä arvioitiin kuuntelukokein. Työn toimeksiantaja on saksalainen Fraunhofer-instituutti, joka tunnetaan muun muassa MP3-koodekin kehittäjänä. Audionkoodauksessa signaaleja käsitellään yleensä noin 20-50 millisekunnin pituisina kehyksinä, jolloin koodausartifaktit voivat vaihdella nopeastikin. Tämän ilmiön ärsyttävyyttä tutkittiin varioimalla kapeakaistaisen kohinan sekä yksittäisten harmonisten voimakkuutta eri nopeuksilla. Koetulosten perusteella keskinopea vaihtelu koetaan ärsyttävimmäksi. Harmoninen kaistanleveyden laajennus (harmonic bandwidth extension) on menetelmä, jolla voidaan luoda harmonisia komponentteja rajataajuuden yläpuolelle alkuperäistä spektriä venyttämällä. Näin audiosignaalin bittinopeutta voidaan laskea, kun ylimpiä harmonisia ei tarvitse koodata eksplisiittisesti, vaan ne voidaan generoida dekoodauksessa. Koska luotujen harmonisisten joukko on kuitenkin aina puutteellinen, saattaa syntyä vaikutelma ylimääräisestä sävelkorkeudesta (ghost pitch). Kuuntelukokeessa tutkittiin synteettisillä äänillä, miten tämän ilmiön voimakkuus riippuu äänen perustaajuudesta ja valitusta rajataajuudesta. Kuulon peittokäyrää voidaan approksimoida tehokkaasti spektrin verhokäyrällä, jota käyttäen itse signaalikehys voidaan siirtää perkeptuaaliseen alueeseen kvantisoitavaksi. Kvantisointikohinan peittymistä voidaan tehostaa säätämällä verhokäyrän pehmeyttä sen siirtofunktioon sijoitetulla vakiolla. Työssä esitetään ehdotus tämän parametrin arvoksi. Sopivasti muokattua verhokäyrää voidaan käyttää myös spektrin voimakkaiden osien vahvistamiseen ja heikkojen osien vaimentamiseen. Puhesignaaleilla huomattiin, että tällä formanttien korostamisella voidaan peittää kvantisointikohinaa, mutta samalla sointiväri muuttuu epäluonnollisemmaksi. Tekstissä esitetään malli optimaalisten muokkausvakioiden valitsemiseksi perkeptuaalisen signaali-kohinasuhteen funktiona.This thesis discusses the perceptual annoyance of several audio coding artifacts that have become of interest during the development of USAC, a new low-bitrate speech and audio coder. A total of four different coding-related phenomena, all of which are explained below, were investigated in this study. All artifacts were artificially generated using MATLAB(R) and evaluated in listening tests with approximately ten participants in each. This work was commissioned by Fraunhofer IIS, Germany - a leader in audio coding technology and the home of MP3. In audio coding, signals are usually processed in frames with a length of 20 to 50 milliseconds, which may cause rapid variations in artifacts. In our tests, the level of critical-bandwidth noise or single harmonics was altered with various speeds. The results suggest that moderate-speed variations are considered the most annoying. Harmonic bandwidth extension is a method that generates artificial harmonics by stretching spectra in frequency. It is useful in audio compression because upper harmonics need not be encoded explicitly, but can be approximately reconstructed in the decoding phase. However, the generated harmonic patch will inevitably be incomplete, which may cause a false additional pitch sensation. The perceived strength of this ghost pitch was examined with synthetic tones as a function of fundamental and crossover frequencies. The masking curve of a signal frame can be efficiently modelled with a spectral envelope. It can then be used for transferring the frame to the perceptual domain for quantization. The resulting quantization noise will be less audible if the smoothness of the envelope is properly adjusted in the first place by modifying the transfer function with a constant. A proposal for the optimal constant value is provided in this study. Strong parts of a signal spectrum can be boosted and weak parts diminished by multiplying the spectrum with its modified envelope. This technique, known as formant enhancement, enables a better masking of quantization noise, but tends to render the overall tone unnatural. A model for selecting the optimal spectrum modification parameter values as a function of perceptual signal-to-noise ratio is proposed