Comparison of CELP speech coder with a wavelet method

This thesis compares the speech quality of Code Excited Linear Predictor (CELP, Federal Standard 1016) speech coder with a new wavelet method to compress speech. The performances of both are compared by performing subjective listening tests. The test signals used are clean signals (i.e. with no background noise), speech signals with room noise and speech signals with artificial noise added. Results indicate that for clean signals and signals with predominantly voiced components the CELP standard performs better than the wavelet method but for signals with room noise the wavelet method performs much better than the CELP. For signals with artificial noise added, the results are mixed depending on the level of artificial noise added with CELP performing better for low level noise added signals and the wavelet method performing better for higher noise levels

Vector Sum Excited Linear Prediction (VSELP) speech coding at 4.8 kbps

Code Excited Linear Prediction (CELP) speech coders exhibit good performance at data rates as low as 4800 bps. The major drawback to CELP type coders is their larger computational requirements. The Vector Sum Excited Linear Prediction (VSELP) speech coder utilizes a codebook with a structure which allows for a very efficient search procedure. Other advantages of the VSELP codebook structure is discussed and a detailed description of a 4.8 kbps VSELP coder is given. This coder is an improved version of the VSELP algorithm, which finished first in the NSA's evaluation of the 4.8 kbps speech coders. The coder uses a subsample resolution single tap long term predictor, a single VSELP excitation codebook, a novel gain quantizer which is robust to channel errors, and a new adaptive pre/postfilter arrangement

Comparison of Wideband Earpiece Integrations in Mobile Phone

Perinteisesti puhelinverkoissa välitettävä puhe on ollut kapeakaistaista, kaistan ollessa 300 - 3400 Hz. Voidaan kuitenkin olettaa, että laajakaistaiset puhepalvelut tulevat saamaan markkinoilla enemmän jalansijaa tulevina vuosina. Tässä lopputyössä esitellään puheenkoodauksen perusteet laajakaistaisen adaptiivisen moninopeuspuhekoodekin (AMR-WB) kanssa. Laajakaistainen puhekoodekki laajentaa puhekaistan 50-7000 Hz käyttäen 16 kHz näytetaajuutta. Käytännössä laajempi kaista tarkoittaa parannuksia puheen ymmärrettävyyteen ja tekee siitä luonnollisemman ja mukavamman kuuloista. Tämän lopputyön päätavoite on vertailla kahden eri laajakaistaisen matkapuhelinkuulokkeen integrointia. Kysymys kuuluu, kuinka paljon käyttäjä hyötyy isommasta kuulokkeesta matkapuhelimessa? Kuulokkeiden suorituskyvyn selvittämiseksi niille tehtiin objektiivisia mittauksia vapaakentässä. Mittauksia tehtiin myös puhelimelle pää- ja torsosimulaattorissa (HATS) johdottamalla kuuloke suoraan vahvistimelle, sekä lisäksi puhelun ollessa aktiivisena GSM ja WCDMA verkoissa. Objektiiviset mittaukset osoittivat kahden eri integroinnin väliset erot kuulokkeiden taajuusvasteessa ja särössä erityisesti matalilla taajuuksilla. Lopuksi tehtiin kuuntelukoe tarkoituksena selvittää erottaako loppukäyttäjä pienemmän ja isomman kuulokkeen välistä eroa käyttäen kapeakaistaisia ja laajakaistaisia puhelinääninäytteitä. Kuuntelukokeen tuloksien pohjalta voidaan sanoa, että käyttäjä erottaa kahden eri integroinnin erot ja miespuhuja hyötyy naispuhujaa enemmän isommasta kuulokkeesta laajakaistaisella puhekoodekilla.The speech in telecommunication networks has been traditionally narrowband ranging from 300 Hz to 3400 Hz. It can be expected that wideband speech call services will increase their foothold in the markets during the coming years. In this thesis speech coding basics with adaptive multirate wideband (AMR-WB) are introduced. The wideband codec widens the speech band to new range from 50 Hz to 7000 Hz using 16 kHz sampling frequency. In practice the wider band means improvements to speech intelligibility and makes it more natural and comfortable to listen to. The main focus of this thesis work is to compare two different wideband earpiece integrations. The question is how much the end-user will benefit from using a larger earpiece in a mobile phone? To find out speaker performance, objective measurements in free field were done for the earpiece modules. Measurements were performed also for the phone on head and torso simulator (HATS) by wiring the earpieces directly to a power amplifier and with over the air on GSM and WCDMA networks. The results of objective measurements showed differences between the earpiece integrations especially on low frequencies in frequency response and distortion. Finally the subjective listening test is done for comparison to see if the end-user notices the difference between smaller and larger earpiece integrations using narrowband and wideband speech samples. Based on these subjective test results it can be said that the user can differentiate between two different integrations and that a male speaker benefits more from a larger earpiece than a female speaker

The development of speech coding and the first standard coder for public mobile telephony

This thesis describes in its core chapter (Chapter 4) the original algorithmic and design features of the ??rst coder for public mobile telephony, the GSM full-rate speech coder, as standardized in 1988. It has never been described in so much detail as presented here. The coder is put in a historical perspective by two preceding chapters on the history of speech production models and the development of speech coding techniques until the mid 1980s, respectively. In the epilogue a brief review is given of later developments in speech coding. The introductory Chapter 1 starts with some preliminaries. It is de- ??ned what speech coding is and the reader is introduced to speech coding standards and the standardization institutes which set them. Then, the attributes of a speech coder playing a role in standardization are explained. Subsequently, several applications of speech coders - including mobile telephony - will be discussed and the state of the art in speech coding will be illustrated on the basis of some worldwide recognized standards. Chapter 2 starts with a summary of the features of speech signals and their source, the human speech organ. Then, historical models of speech production which form the basis of di??erent kinds of modern speech coders are discussed. Starting with a review of ancient mechanical models, we will arrive at the electrical source-??lter model of the 1930s. Subsequently, the acoustic-tube models as they arose in the 1950s and 1960s are discussed. Finally the 1970s are reviewed which brought the discrete-time ??lter model on the basis of linear prediction. In a unique way the logical sequencing of these models is exposed, and the links are discussed. Whereas the historical models are discussed in a narrative style, the acoustic tube models and the linear prediction tech nique as applied to speech, are subject to more mathematical analysis in order to create a sound basis for the treatise of Chapter 4. This trend continues in Chapter 3, whenever instrumental in completing that basis. In Chapter 3 the reader is taken by the hand on a guided tour through time during which successive speech coding methods pass in review. In an original way special attention is paid to the evolutionary aspect. Speci??cally, for each newly proposed method it is discussed what it added to the known techniques of the time. After presenting the relevant predecessors starting with Pulse Code Modulation (PCM) and the early vocoders of the 1930s, we will arrive at Residual-Excited Linear Predictive (RELP) coders, Analysis-by-Synthesis systems and Regular- Pulse Excitation in 1984. The latter forms the basis of the GSM full-rate coder. In Chapter 4, which constitutes the core of this thesis, explicit forms of Multi-Pulse Excited (MPE) and Regular-Pulse Excited (RPE) analysis-by-synthesis coding systems are developed. Starting from current pulse-amplitude computation methods in 1984, which included solving sets of equations (typically of order 10-16) two hundred times a second, several explicit-form designs are considered by which solving sets of equations in real time is avoided. Then, the design of a speci??c explicitform RPE coder and an associated eÆcient architecture are described. The explicit forms and the resulting architectural features have never been published in so much detail as presented here. Implementation of such a codec enabled real-time operation on a state-of-the-art singlechip digital signal processor of the time. This coder, at a bit rate of 13 kbit/s, has been selected as the Full-Rate GSM standard in 1988. Its performance is recapitulated. Chapter 5 is an epilogue brie y reviewing the major developments in speech coding technology after 1988. Many speech coding standards have been set, for mobile telephony as well as for other applications, since then. The chapter is concluded by an outlook

An investigation into glottal waveform based speech coding

Coding of voiced speech by extraction of the glottal waveform has shown promise in improving the efficiency of speech coding systems. This thesis describes an investigation into the performance of such a system. The effect of reverberation on the radiation impedance at the lips is shown to be negligible under normal conditions. Also, the accuracy of the Image Method for adding artificial reverberation to anechoic speech recordings is established. A new algorithm, Pre-emphasised Maximum Likelihood Epoch Detection (PMLED), for Glottal Closure Instant detection is proposed. The algorithm is tested on natural speech and is shown to be both accurate and robust. Two techniques for giottai waveform estimation, Closed Phase Inverse Filtering (CPIF) and Iterative Adaptive Inverse Filtering (IAIF), are compared. In tandem with an LF model fitting procedure, both techniques display a high degree of accuracy However, IAIF is found to be slightly more robust. Based on these results, a Glottal Excited Linear Predictive (GELP) coding system for voiced speech is proposed and tested. Using a differential LF parameter quantisation scheme, the system achieves speech quality similar to that of U S Federal Standard 1016 CELP at a lower mean bit rate while incurring no extra delay

Novel Pitch Detection Algorithm With Application to Speech Coding

This thesis introduces a novel method for accurate pitch detection and speech segmentation, named Multi-feature, Autocorrelation (ACR) and Wavelet Technique (MAWT). MAWT uses feature extraction, and ACR applied on Linear Predictive Coding (LPC) residuals, with a wavelet-based refinement step. MAWT opens the way for a unique approach to modeling: although speech is divided into segments, the success of voicing decisions is not crucial. Experiments demonstrate the superiority of MAWT in pitch period detection accuracy over existing methods, and illustrate its advantages for speech segmentation. These advantages are more pronounced for gain-varying and transitional speech, and under noisy conditions

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

Vector adaptive predictive coder for speech and audio

A real-time vector adaptive predictive coder which approximates each vector of K speech samples by using each of M fixed vectors in a first codebook to excite a time-varying synthesis filter and picking the vector that minimizes distortion. Predictive analysis for each frame determines parameters used for computing from vectors in the first codebook zero-state response vectors that are stored at the same address (index) in a second codebook. Encoding of input speech vectors s.sub.n is then carried out using the second codebook. When the vector that minimizes distortion is found, its index is transmitted to a decoder which has a codebook identical to the first codebook of the decoder. There the index is used to read out a vector that is used to synthesize an output speech vector s.sub.n. The parameters used in the encoder are quantized, for example by using a table, and the indices are transmitted to the decoder where they are decoded to specify transfer characteristics of filters used in producing the vector s.sub.n from the receiver codebook vector selected by the vector index transmitted