DESIGN AND EVALUATION OF HARMONIC SPEECH ENHANCEMENT AND BANDWIDTH EXTENSION

Improving the quality and intelligibility of speech signals continues to be an important topic in mobile communications and hearing aid applications. This thesis explored the possibilities of improving the quality of corrupted speech by cascading a log Minimum Mean Square Error (logMMSE) noise reduction system with a Harmonic Speech Enhancement (HSE) system. In HSE, an adaptive comb filter is deployed to harmonically filter the useful speech signal and suppress the noisy components to noise floor. A Bandwidth Extension (BWE) algorithm was applied to the enhanced speech for further improvements in speech quality. Performance of this algorithm combination was evaluated using objective speech quality metrics across a variety of noisy and reverberant environments. Results showed that the logMMSE and HSE combination enhanced the speech quality in any reverberant environment and in the presence of multi-talker babble. The objective improvements associated with the BWE were found to be minima

Concatenative speech synthesis: a Framework for Reducing Perceived Distortion when using the TD-PSOLA Algorithm

This thesis presents the design and evaluation of an approach to concatenative speech synthesis using the Titne-Domain Pitch-Synchronous OverLap-Add (I'D-PSOLA) signal processing algorithm. Concatenative synthesis systems make use of pre-recorded speech segments stored in a speech corpus. At synthesis time, the `best' segments available to synthesise the new utterances are chosen from the corpus using a process known as unit selection. During the synthesis process, the pitch and duration of these segments may be modified to generate the desired prosody. The TD-PSOLA algorithm provides an efficient and essentially successful solution to perform these modifications, although some perceptible distortion, in the form of `buzzyness', may be introduced into the speech signal. Despite the popularity of the TD-PSOLA algorithm, little formal research has been undertaken to address this recognised problem of distortion. The approach in the thesis has been developed towards reducing the perceived distortion that is introduced when TD-PSOLA is applied to speech. To investigate the occurrence of this distortion, a psychoacoustic evaluation of the effect of pitch modification using the TD-PSOLA algorithm is presented. Subjective experiments in the form of a set of listening tests were undertaken using word-level stimuli that had been manipulated using TD-PSOLA. The data collected from these experiments were analysed for patterns of co- occurrence or correlations to investigate where this distortion may occur. From this, parameters were identified which may have contributed to increased distortion. These parameters were concerned with the relationship between the spectral content of individual phonemes, the extent of pitch manipulation, and aspects of the original recordings. Based on these results, a framework was designed for use in conjunction with TD-PSOLA to minimise the possible causes of distortion. The framework consisted of a novel speech corpus design, a signal processing distortion measure, and a selection process for especially problematic phonemes. Rather than phonetically balanced, the corpus is balanced to the needs of the signal processing algorithm, containing more of the adversely affected phonemes. The aim is to reduce the potential extent of pitch modification of such segments, and hence produce synthetic speech with less perceptible distortion. The signal processingdistortion measure was developed to allow the prediction of perceptible distortion in pitch-modified speech. Different weightings were estimated for individual phonemes,trained using the experimental data collected during the listening tests.The potential benefit of such a measure for existing unit selection processes in a corpus-based system using TD-PSOLA is illustrated. Finally, the special-case selection process was developed for highly problematic voiced fricative phonemes to minimise the occurrence of perceived distortion in these segments. The success of the framework, in terms of generating synthetic speech with reduced distortion, was evaluated. A listening test showed that the TD-PSOLA balanced speech corpus may be capable of generating pitch-modified synthetic sentences with significantly less distortion than those generated using a typical phonetically balanced corpus. The voiced fricative selection process was also shown to produce pitch-modified versions of these phonemes with less perceived distortion than a standard selection process. The listening test then indicated that the signal processing distortion measure was able to predict the resulting amount of distortion at the sentence-level after the application of TD-PSOLA, suggesting that it may be beneficial to include such a measure in existing unit selection processes. The framework was found to be capable of producing speech with reduced perceptible distortion in certain situations, although the effects seen at the sentence-level were less than those seen in the previous investigative experiments that made use of word-level stimuli. This suggeststhat the effect of the TD-PSOLA algorithm cannot always be easily anticipated due to the highly dynamic nature of speech, and that the reduction of perceptible distortion in TD-PSOLA-modified speech remains a challenge to the speech community

Precise Estimation of Vocal Tract and Voice Source Characteristics

This thesis addresses the problem of quality degradation in speech produced by parameter-based speech synthesis, within the framework of an articulatory-acoustic forward mapping. I first investigate current problems in speech parameterisation, and point out the fact that conventional parameterisation inaccurately extracts the vocal tract response due to interference from the harmonic structure of voiced speech. To overcome this problem, I introduce a method for estimating filter responses more precisely from periodic signals. The method achieves such estimation in the frequency domain by approximating all the harmonics observed in several frames based on a least squares criterion. It is shown that the proposed method is capable of estimating the response more accurately than widely-used frame-by-frame parameterisation, for simulations using synthetic speech and for an articulatory-acoustic mapping using actual speech. I also deal with the source-filter separation problem and independent control of the voice source characteristic during speech synthesis. I propose a statistical approach to separating out the vocal-tract filter response from the voice source characteristic using a large articulatory database. The approach realises such separation for voiced speech using an iterative approximation procedure under the assumption that the speech production process is a linear system composed of a voice source and a vocal-tract filter, and that each of the components is controlled independently by different sets of factors. Experimental results show that controlling the source characteristic greatly improves the accuracy of the articulatory-acoustic mapping, and that the spectral variation of the source characteristic is evidently influenced by the fundamental frequency or the power of speech. The thesis provides more accurate acoustical approximation of the vocal tract response, which will be beneficial in a wide range of speech technologies, and lays the groundwork in speech science for a new type of corpus-based statistical solution to the source-filter separation problem

A Prosodic Turkish text-to-speech synthesizer

Naturalness in Text-to-Speech systems is very important in achieving high quality waveform. The naturalness of the waveform is highly correlated with phonetic coverage and prosodic features such as, duration and F0 contour. Duration determines the timing for the synthesized phoneme, whereas F0 contour determines fundamental frequency component of the waveform. This thesis presents the development of a prosodic Text-to-Speech System for Turkish Language using the Festival Tool [31]. We describe a complete realization of a new male voice, covering allophones of Turkish using duration and F0 parameters. The duration of the allophones and the word stress have been studied extensively. Sentence stress and phrasal stress are also discussed by in less detail. Carrier words are designed approximately for all allophone-allophone combinations. 1680 carrier words are recorded in a sound-proof recording studio. LPC (linear predictive coding) and RES (residual) parameters are computed. The text normalisation module is implemented for abbreviations and numbers. Durations for the allophones are entered. Sentence level and word level F0 generation modules are implemented. By increasing the number of phonemes and giving prosody we obtained a more natural sounding Text-to-Speech System for Turkish Language

A configurable vector processor for accelerating speech coding algorithms

The growing demand for voice-over-packer (VoIP) services and multimedia-rich applications has made increasingly important the efficient, real-time implementation of low-bit rates speech coders on embedded VLSI platforms. Such speech coders are designed to substantially reduce the bandwidth requirements thus enabling dense multichannel gateways in small form factor. This however comes at a high computational cost which mandates the use of very high performance embedded processors. This thesis investigates the potential acceleration of two major ITU-T speech coding algorithms, namely G.729A and G.723.1, through their efficient implementation on a configurable extensible vector embedded CPU architecture. New scalar and vector ISAs were introduced which resulted in up to 80% reduction in the dynamic instruction count of both workloads. These instructions were subsequently encapsulated into a parametric, hybrid SISD (scalar processor)–SIMD (vector) processor. This work presents the research and implementation of the vector datapath of this vector coprocessor which is tightly-coupled to a Sparc-V8 compliant CPU, the optimization and simulation methodologies employed and the use of Electronic System Level (ESL) techniques to rapidly design SIMD datapaths

The Development of Synthetic Speech Aids for Patients With Acquired Disability

Patients suffering from a variety of speech disorders can benefit from synthetic speech. This study concentrates on the dysarthric patients with acquired speech loss as these patients have intact intellect and are more likely to benefit from synthetic speech. The physical skills of these patients vary enormously and their needs and situations are different. The main part of this work is concerned with the design, development and evaluation of a range of speech aids to meet these varying needs and skills. Three methods of speech synthesis are used and their performance has been investigated by using a Diagnostic Rhyme Test to measure the intelligibility of individual words. The results of this trial showed Adaptive Differential Pulse Code Modulation (ADPCM) to be more intelligible than Linear Predictive Coding (LPC), both these methods being more intelligible than constructive synthesis. A further trial was conducted to measure the speech quality of phrases produced by the synthesisers. This showed listeners preferred listening to phrases constructed of LPC words than to phrases generated using Phoneme based synthesisers. Phrases with mixed LPC and constructed words were preferred to phrases of constructed words. The devices that were developed use different methods of synthesis and the choice of method was guided by these trials. The Pocket Speech Aid is a rapid access limited vocabulary communication aid which uses ADPCM synthesis. Direct selection is the method used to give users access to eight phrases. The Pocket Speech Aid has been very successful in practice. When used as a telephone aid eight out of ten patients increased their communication ability and when used as a conversation prompter ten out of fourteen patients were able to steer the direction of real time conversations. This device has generated a great deal of interest from other centres and the demand for the device which is currently being manufactured confirms that it has a role to play in assisting those with communication difficulties. The Macleod Unit was named after a remarkable patient suffering from Motor Neurone Disease who realised his speech would soon be lost and had the foresight to select a vocabulary and record the words on a cassette recorder. His 625 word vocabulary was transferred to the speech aid which uses an encoding method of word selection. Clinical feedback showed the device to be of benefit for this highly motivated individual but was less successful for other patients in this group who found the cognitive effort to select codes too great. An unlimited vocabulary device based on the commercially available VOTRAX which uses constructive synthesis was developed but this device was rejected because of the robotic sounding voice. A further unlimited vocabulary device prototype, the Uvocom, was designed to improve the speech quality and to investigate if there is a need for an unlimited vocabulary. The Uvocom uses a core vocabulary of 1000 LPC words and uses Phoneme back-up for words not stored in the core vocabulary. Trials with the Uvocom have indicated that quality speech in an unlimited vocabulary device is likely to benefit a small number of patients who have the physical skills to operate such a device. Finally, some indication is given of the directions in which future work could progress based on the proven success of the Pocket Speech Aid

Unit selection and waveform concatenation strategies in Cantonese text-to-speech.

Oey Sai Lok.Thesis (M.Phil.)--Chinese University of Hong Kong, 2005.Includes bibliographical references.Abstracts in English and Chinese.Chapter 1. --- Introduction --- p.1Chapter 1.1 --- An overview of Text-to-Speech technology --- p.2Chapter 1.1.1 --- Text processing --- p.2Chapter 1.1.2 --- Acoustic synthesis --- p.3Chapter 1.1.3 --- Prosody modification --- p.4Chapter 1.2 --- Trends in Text-to-Speech technologies --- p.5Chapter 1.3 --- Objectives of this thesis --- p.7Chapter 1.4 --- Outline of the thesis --- p.9References --- p.11Chapter 2. --- Cantonese Speech --- p.13Chapter 2.1 --- The Cantonese dialect --- p.13Chapter 2.2 --- Phonology of Cantonese --- p.14Chapter 2.2.1 --- Initials --- p.15Chapter 2.2.2 --- Finals --- p.16Chapter 2.2.3 --- Tones --- p.18Chapter 2.3 --- Acoustic-phonetic properties of Cantonese syllables --- p.19References --- p.24Chapter 3. --- Cantonese Text-to-Speech --- p.25Chapter 3.1 --- General overview --- p.25Chapter 3.1.1 --- Text processing --- p.25Chapter 3.1.2 --- Corpus based acoustic synthesis --- p.26Chapter 3.1.3 --- Prosodic control --- p.27Chapter 3.2 --- Syllable based Cantonese Text-to-Speech system --- p.28Chapter 3.3 --- Sub-syllable based Cantonese Text-to-Speech system --- p.29Chapter 3.3.1 --- Definition of sub-syllable units --- p.29Chapter 3.3.2 --- Acoustic inventory --- p.31Chapter 3.3.3 --- Determination of the concatenation points --- p.33Chapter 3.4 --- Problems --- p.34References --- p.36Chapter 4. --- Waveform Concatenation for Sub-syllable Units --- p.37Chapter 4.1 --- Previous work in concatenation methods --- p.37Chapter 4.1.1 --- Determination of concatenation point --- p.38Chapter 4.1.2 --- Waveform concatenation --- p.38Chapter 4.2 --- Problems and difficulties in concatenating sub-syllable units --- p.39Chapter 4.2.1 --- Mismatch of acoustic properties --- p.40Chapter 4.2.2 --- "Allophone problem of Initials /z/, Id and /s/" --- p.42Chapter 4.3 --- General procedures in concatenation strategies --- p.44Chapter 4.3.1 --- Concatenation of unvoiced segments --- p.45Chapter 4.3.2 --- Concatenation of voiced segments --- p.45Chapter 4.3.3 --- Measurement of spectral distance --- p.48Chapter 4.4 --- Detailed procedures in concatenation points determination --- p.50Chapter 4.4.1 --- Unvoiced segments --- p.50Chapter 4.4.2 --- Voiced segments --- p.53Chapter 4.5 --- Selected examples in concatenation strategies --- p.58Chapter 4.5.1 --- Concatenation at Initial segments --- p.58Chapter 4.5.1.1 --- Plosives --- p.58Chapter 4.5.1.2 --- Fricatives --- p.59Chapter 4.5.2 --- Concatenation at Final segments --- p.60Chapter 4.5.2.1 --- V group (long vowel) --- p.60Chapter 4.5.2.2 --- D group (diphthong) --- p.61References --- p.63Chapter 5. --- Unit Selection for Sub-syllable Units --- p.65Chapter 5.1 --- Basic requirements in unit selection process --- p.65Chapter 5.1.1 --- Availability of multiple copies of sub-syllable units --- p.65Chapter 5.1.1.1 --- "Levels of ""identical""" --- p.66Chapter 5.1.1.2 --- Statistics on the availability --- p.67Chapter 5.1.2 --- Variations in acoustic parameters --- p.70Chapter 5.1.2.1 --- Pitch level --- p.71Chapter 5.1.2.2 --- Duration --- p.74Chapter 5.1.2.3 --- Intensity level --- p.75Chapter 5.2 --- Selection process: availability check on sub-syllable units --- p.77Chapter 5.2.1 --- Multiple copies found --- p.79Chapter 5.2.2 --- Unique copy found --- p.79Chapter 5.2.3 --- No matched copy found --- p.80Chapter 5.2.4 --- Illustrative examples --- p.80Chapter 5.3 --- Selection process: acoustic analysis on candidate units --- p.81References --- p.88Chapter 6. --- Performance Evaluation --- p.89Chapter 6.1 --- General information --- p.90Chapter 6.1.1 --- Objective test --- p.90Chapter 6.1.2 --- Subjective test --- p.90Chapter 6.1.3 --- Test materials --- p.91Chapter 6.2 --- Details of the objective test --- p.92Chapter 6.2.1 --- Testing method --- p.92Chapter 6.2.2 --- Results --- p.93Chapter 6.2.3 --- Analysis --- p.96Chapter 6.3 --- Details of the subjective test --- p.98Chapter 6.3.1 --- Testing method --- p.98Chapter 6.3.2 --- Results --- p.99Chapter 6.3.3 --- Analysis --- p.101Chapter 6.4 --- Summary --- p.107References --- p.108Chapter 7. --- Conclusions and Future Works --- p.109Chapter 7.1 --- Conclusions --- p.109Chapter 7.2 --- Suggested future works --- p.111References --- p.113Appendix 1 Mean pitch level of Initials and Finals stored in the inventory --- p.114Appendix 2 Mean durations of Initials and Finals stored in the inventory --- p.121Appendix 3 Mean intensity level of Initials and Finals stored in the inventory --- p.124Appendix 4 Test word used in performance evaluation --- p.127Appendix 5 Test paragraph used in performance evaluation --- p.128Appendix 6 Pitch profile used in the Text-to-Speech system --- p.131Appendix 7 Duration model used in Text-to-Speech system --- p.13

Concatenative speech synthesis : a framework for reducing perceived distortion when using the TD-PSOLA algorithm

This thesis presents the design and evaluation of an approach to concatenative speech synthesis using the Titne-Domain Pitch-Synchronous OverLap-Add (I'D-PSOLA) signal processing algorithm. Concatenative synthesis systems make use of pre-recorded speech segments stored in a speech corpus. At synthesis time, the `best' segments available to synthesise the new utterances are chosen from the corpus using a process known as unit selection. During the synthesis process, the pitch and duration of these segments may be modified to generate the desired prosody. The TD-PSOLA algorithm provides an efficient and essentially successful solution to perform these modifications, although some perceptible distortion, in the form of `buzzyness', may be introduced into the speech signal. Despite the popularity of the TD-PSOLA algorithm, little formal research has been undertaken to address this recognised problem of distortion. The approach in the thesis has been developed towards reducing the perceived distortion that is introduced when TD-PSOLA is applied to speech. To investigate the occurrence of this distortion, a psychoacoustic evaluation of the effect of pitch modification using the TD-PSOLA algorithm is presented. Subjective experiments in the form of a set of listening tests were undertaken using word-level stimuli that had been manipulated using TD-PSOLA. The data collected from these experiments were analysed for patterns of co- occurrence or correlations to investigate where this distortion may occur. From this, parameters were identified which may have contributed to increased distortion. These parameters were concerned with the relationship between the spectral content of individual phonemes, the extent of pitch manipulation, and aspects of the original recordings. Based on these results, a framework was designed for use in conjunction with TD-PSOLA to minimise the possible causes of distortion. The framework consisted of a novel speech corpus design, a signal processing distortion measure, and a selection process for especially problematic phonemes. Rather than phonetically balanced, the corpus is balanced to the needs of the signal processing algorithm, containing more of the adversely affected phonemes. The aim is to reduce the potential extent of pitch modification of such segments, and hence produce synthetic speech with less perceptible distortion. The signal processingdistortion measure was developed to allow the prediction of perceptible distortion in pitch-modified speech. Different weightings were estimated for individual phonemes,trained using the experimental data collected during the listening tests.The potential benefit of such a measure for existing unit selection processes in a corpus-based system using TD-PSOLA is illustrated. Finally, the special-case selection process was developed for highly problematic voiced fricative phonemes to minimise the occurrence of perceived distortion in these segments. The success of the framework, in terms of generating synthetic speech with reduced distortion, was evaluated. A listening test showed that the TD-PSOLA balanced speech corpus may be capable of generating pitch-modified synthetic sentences with significantly less distortion than those generated using a typical phonetically balanced corpus. The voiced fricative selection process was also shown to produce pitch-modified versions of these phonemes with less perceived distortion than a standard selection process. The listening test then indicated that the signal processing distortion measure was able to predict the resulting amount of distortion at the sentence-level after the application of TD-PSOLA, suggesting that it may be beneficial to include such a measure in existing unit selection processes. The framework was found to be capable of producing speech with reduced perceptible distortion in certain situations, although the effects seen at the sentence-level were less than those seen in the previous investigative experiments that made use of word-level stimuli. This suggeststhat the effect of the TD-PSOLA algorithm cannot always be easily anticipated due to the highly dynamic nature of speech, and that the reduction of perceptible distortion in TD-PSOLA-modified speech remains a challenge to the speech community.EThOS - Electronic Theses Online ServiceGBUnited Kingdo

Adaptive pattern recognition of hand-written Chinese characters using Fourier transforms of projection profiles

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