Adaptive Variable Degree-k Zero-Trees for Re-Encoding of Perceptually Quantized Wavelet-Packet Transformed Audio and High Quality Speech

A fast, efficient and scalable algorithm is proposed, in this paper, for re-encoding of perceptually quantized wavelet-packet transform (WPT) coefficients of audio and high quality speech and is called "adaptive variable degree-k zero-trees" (AVDZ). The quantization process is carried out by taking into account some basic perceptual considerations, and achieves good subjective quality with low complexity. The performance of the proposed AVDZ algorithm is compared with two other zero-tree-based schemes comprising: 1- Embedded Zero-tree Wavelet (EZW) and 2- The set partitioning in hierarchical trees (SPIHT). Since EZW and SPIHT are designed for image compression, some modifications are incorporated in these schemes for their better matching to audio signals. It is shown that the proposed modifications can improve their performance by about 15-25%. Furthermore, it is concluded that the proposed AVDZ algorithm outperforms these modified versions in terms of both output average bit-rates and computation times.Comment: 30 pages (Double space), 15 figures, 5 tables, ISRN Signal Processing (in Press

Scalable Speech Coding for IP Networks

The emergence of Voice over Internet Protocol (VoIP) has posed new challenges to the development of speech codecs. The key issue of transporting real-time voice packet over IP networks is the lack of guarantee for reasonable speech quality due to packet delay or loss. Most of the widely used narrowband codecs depend on the Code Excited Linear Prediction (CELP) coding technique. The CELP technique utilizes the long-term prediction across the frame boundaries and therefore causes error propagation in the case of packet loss and need to transmit redundant information in order to mitigate the problem. The internet Low Bit-rate Codec (iLBC) employs the frame-independent coding and therefore inherently possesses high robustness to packet loss. However, the original iLBC lacks in some of the key features of speech codecs for IP networks: Rate flexibility, Scalability, and Wideband support. This dissertation presents novel scalable narrowband and wideband speech codecs for IP networks using the frame independent coding scheme based on the iLBC. The rate flexibility is added to the iLBC by employing the discrete cosine transform (DCT) and iii the scalable algebraic vector quantization (AVQ) and by allocating different number of bits to the AVQ. The bit-rate scalability is obtained by adding the enhancement layer to the core layer of the multi-rate iLBC. The enhancement layer encodes the weighted iLBC coding error in the modified DCT (MDCT) domain. The proposed wideband codec employs the bandwidth extension technique to extend the capabilities of existing narrowband codecs to provide wideband coding functionality. The wavelet transform is also used to further enhance the performance of the proposed codec. The performance evaluation results show that the proposed codec provides high robustness to packet loss and achieves equivalent or higher speech quality than state-of-the-art codecs under the clean channel condition

Artificial Bandwidth Extension of Speech Signals using Neural Networks

Although mobile wideband telephony has been standardized for over 15 years, many countries still do not have a nationwide network with good coverage. As a result, many cellphone calls are still downgraded to narrowband telephony. The resulting loss of quality can be reduced by artificial bandwidth extension. There has been great progress in bandwidth extension in recent years due to the use of neural networks. The topic of this thesis is the enhancement of artificial bandwidth extension using neural networks. A special focus is given to hands-free calls in a car, where the risk is high that the wideband connection is lost due to the fast movement. The bandwidth of narrowband transmission is not only reduced towards higher frequencies above 3.5 kHz but also towards lower frequencies below 300 Hz. There are already methods that estimate the low-frequency components quite well, which will therefore not be covered in this thesis. In most bandwidth extension algorithms, the narrowband signal is initially separated into a spectral envelope and an excitation signal. Both parts are then extended separately in order to finally combine both parts again. While the extension of the excitation can be implemented using simple methods without reducing the speech quality compared to wideband speech, the estimation of the spectral envelope for frequencies above 3.5 kHz is not yet solved satisfyingly. Current bandwidth extension algorithms are just able to reduce the quality loss due to narrowband transmission by a maximum of 50% in most evaluations. In this work, a modification for an existing method for excitation extension is proposed which achieves slight improvements while not generating additional computational complexity. In order to enhance the wideband envelope estimation with neural networks, two modifications of the training process are proposed. On the one hand, the loss function is extended with a discriminative part to address the different characteristics of phoneme classes. On the other hand, by using a GAN (generative adversarial network) for the training phase, a second network is added temporarily to evaluate the quality of the estimation. The neural networks that were trained are compared in subjective and objective evaluations. A final listening test addressed the scenario of a hands-free call in a car, which was simulated acoustically. The quality loss caused by the missing high frequency components could be reduced by 60% with the proposed approach.Obwohl die mobile Breitbandtelefonie bereits seit über 15 Jahren standardisiert ist, gibt es oftmals noch kein flächendeckendes Netz mit einer guten Abdeckung. Das führt dazu, dass weiterhin viele Mobilfunkgespräche auf Schmalbandtelefonie heruntergestuft werden. Der damit einhergehende Qualitätsverlust kann mit künstlicher Bandbreitenerweiterung reduziert werden. Das Thema dieser Arbeit sind Methoden zur weiteren Verbesserungen der Qualität des erweiterten Sprachsignals mithilfe neuronaler Netze. Ein besonderer Fokus liegt auf der Freisprech-Telefonie im Auto, da dabei das Risiko besonders hoch ist, dass durch die schnelle Fortbewegung die Breitbandverbindung verloren geht. Bei der Schmalbandübertragung fehlen neben den hochfrequenten Anteilen (etwa 3.5–7 kHz) auch tiefe Frequenzen unterhalb von etwa 300 Hz. Diese tieffrequenten Anteile können mit bereits vorhandenen Methoden gut geschätzt werden und sind somit nicht Teil dieser Arbeit. In vielen Algorithmen zur Bandbreitenerweiterung wird das Schmalbandsignal zu Beginn in eine spektrale Einhüllende und ein Anregungssignal aufgeteilt. Beide Anteile werden dann separat erweitert und schließlich wieder zusammengeführt. Während die Erweiterung der Anregung nahezu ohne Qualitätsverlust durch einfache Methoden umgesetzt werden kann ist die Schätzung der spektralen Einhüllenden für Frequenzen über 3.5 kHz noch nicht zufriedenstellend gelöst. Mit aktuellen Methoden können im besten Fall nur etwa 50% der durch Schmalbandübertragung reduzierten Qualität zurückgewonnen werden. Für die Anregungserweiterung wird in dieser Arbeit eine Variation vorgestellt, die leichte Verbesserungen erzielt ohne dabei einen Mehraufwand in der Berechnung zu erzeugen. Für die Schätzung der Einhüllenden des Breitbandsignals mithilfe neuronaler Netze werden zwei Änderungen am Trainingsprozess vorgeschlagen. Einerseits wird die Kostenfunktion um einen diskriminativen Anteil erweitert, der das Netz besser zwischen verschiedenen Phonemen unterscheiden lässt. Andererseits wird als Architektur ein GAN (Generative adversarial network) verwendet, wofür in der Trainingsphase ein zweites Netz verwendet wird, das die Qualität der Schätzung bewertet. Die trainierten neuronale Netze wurden in subjektiven und objektiven Tests verglichen. Ein abschließender Hörtest diente zur Evaluierung des Freisprechens im Auto, welches akustisch simuliert wurde. Der Qualitätsverlust durch Wegfallen der hohen Frequenzanteile konnte dabei mit dem vorgeschlagenen Ansatz um etwa 60% reduziert werden

A-Interface Over Internet Protocol For User-Plane Connection Optimization In GSM/EDGE Radio Access Network

This thesis will cover a detailed study about the main motivations and benefits from using IP as a transport protocol for specifically A-interface in GERAN for Circuit Switched User-Plane (CS-UP) connection, in addition to the required protocols. The main study in this document will be around Real Time Protocol (RTP), Real Time Control Protocol (RTCP) negotiation for RTP packets multiplexing, for both cases, with and without RTP header compression. The focus will be about the communication between the Base Station Controller (BSC) and the Media GateWay (MGW), the bandwidth gain in accordance to the multiplexing delay for processing and buffering, the voice Quality of Service (QoS) and some other parameters

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

Media gateway utilizando um GPU

Mestrado em Engenharia de Computadores e Telemátic

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

Models and analysis of vocal emissions for biomedical applications

This book of Proceedings collects the papers presented at the 3rd International Workshop on Models and Analysis of Vocal Emissions for Biomedical Applications, MAVEBA 2003, held 10-12 December 2003, Firenze, Italy. The workshop is organised every two years, and aims to stimulate contacts between specialists active in research and industrial developments, in the area of voice analysis for biomedical applications. The scope of the Workshop includes all aspects of voice modelling and analysis, ranging from fundamental research to all kinds of biomedical applications and related established and advanced technologies