Time-domain concatenative text-to-speech synthesis.

A concatenation framework for time-domain concatenative speech synthesis (TDCSS) is presented and evaluated. In this framework, speech segments are extracted from CV, VC, CVC and CC waveforms, and abutted. Speech rhythm is controlled via a single duration parameter, which specifies the initial portion of each stored waveform to be output. An appropriate choice of segmental durations reduces spectral discontinuity problems at points of concatenation, thus reducing reliance upon smoothing procedures. For text-to-speech considerations, a segmental timing system is described, which predicts segmental durations at the word level, using a timing database and a pattern matching look-up algorithm. The timing database contains segmented words with associated duration values, and is specific to an actual inventory of concatenative units. Segmental duration prediction accuracy improves as the timing database size increases. The problem of incomplete timing data has been addressed by using `default duration' entries in the database, which are created by re-categorising existing timing data according to articulation manner. If segmental duration data are incomplete, a default duration procedure automatically categorises the missing speech segments according to segment class. The look-up algorithm then searches the timing database for duration data corresponding to these re-categorised segments. The timing database is constructed using an iterative synthesis/adjustment technique, in which a `judge' listens to synthetic speech and adjusts segmental durations to improve naturalness. This manual technique for constructing the timing database has been evaluated. Since the timing data is linked to an expert judge's perception, an investigation examined whether the expert judge's perception of speech naturalness is representative of people in general. Listening experiments revealed marked similarities between an expert judge's perception of naturalness and that of the experimental subjects. It was also found that the expert judge's perception remains stable over time. A synthesis/adjustment experiment found a positive linear correlation between segmental durations chosen by an experienced expert judge and duration values chosen by subjects acting as expert judges. A listening test confirmed that between 70% and 100% intelligibility can be achieved with words synthesised using TDCSS. In a further test, a TDCSS synthesiser was compared with five well-known text-to-speech synthesisers, and was ranked fifth most natural out of six. An alternative concatenation framework (TDCSS2) was also evaluated, in which duration parameters specify both the start point and the end point of the speech to be extracted from a stored waveform and concatenated. In a similar listening experiment, TDCSS2 stimuli were compared with five well-known text-tospeech synthesisers, and were ranked fifth most natural out of six

A Context-based Numeral Reading Technique for Text to Speech Systems

This paper presents a novel technique for context based numeral reading in Indian language text to speech systems. The model uses a set of rules to determine the context of the numeral pronunciation and is being integrated with the waveform concatenation technique to produce speech out of the input text in Indian languages. For this purpose, the three Indian languages Odia, Hindi and Bengali are considered. To analyze the performance of the proposed technique, a set of experiments are performed considering different context of numeral pronunciations and the results are compared with existing syllable-based technique. The results obtained from different experiments shows the effectiveness of the proposed technique in producing intelligible speech out of the entered text utterances compared to the existing technique even with very less storage and execution time

FPGA-based Implementation of Concatenative Speech Synthesis Algorithm

The main aim of a text-to-speech synthesis system is to convert ordinary text into an acoustic signal that is indistinguishable from human speech. This thesis presents an architecture to implement a concatenative speech synthesis algorithm targeted to FPGAs. Many current text-to-speech systems are based on the concatenation of acoustic units of recorded speech. Current concatenative speech synthesizers are capable of producing highly intelligible speech. However, the quality of speech often suffers from discontinuities between the acoustic units, due to contextual differences. This is the easiest method to produce synthetic speech. It concatenates prerecorded acoustic elements and forms a continuous speech element. The software implementation of the algorithm is performed in C whereas the hardware implementation is done in structural VHDL. A database of acoustic elements is formed first with recording sounds for different phones. The architecture is designed to concatenate acoustic elements corresponding to the phones that form the target word. Target word corresponds to the word that has to be synthesized. This architecture doesn\u27t address the form discontinuities between the acoustic elements as its ultimate goal is the synthesis of speech. The Hardware implementation is verified on a Virtex (v800hq240-4) FPGA device

Study on phonetic context of Malay syllables towards the development of Malay speech synthesizer [TK7882.S65 H233 2007 f rb].

Pensintesis sebutan Bahasa Melayu telah berkembang daripada teknik pensintesis berparameter (pemodelan penyebutan manusia dan pensintesis berdasarkan formant) kepada teknik pensintesis tidak berparameter (pensintesis sebutan berdasarkan pencantuman). Speech synthesizer has evolved from parametric speech synthesizer (articulatory and formant synthesizer) to non-parametric synthesizer (concatenative synthesizer). Recently, the concatenative speech synthesizer approach is moving towards corpusbased or unit selection technique

The Cerevoice Blizzard Entry 2007: Are Small Database Errors Worse than Compression Artifacts?

In commercial systems the memory footprint of unit selection systems is often a key issue. This is especially true for PDAs and other embedded devices. In this years Blizzard entry CereProc R○gave itself the criteria that the full database system entered would have a smaller memory footprint than either of the two smaller database entries. This was accomplished by applying speex speech compression to the full database entry. In turn a set of small database techniques used to improve the quality of small database systems in last years entry were extended. Finally, for all systems, two quality control methods were applied to the underlying database to improve the lexicon and transcription match to the underlying data. Results suggest that mild audio quality artifacts introduced by lossy compression have almost as much impact on MOS perceived quality as concatenation errors introduced by sparse data in the smaller systems with bulked diphones. Index Terms: speech synthesis, unit selection. 1

Vowel classification based approach for Telugu Text-to-Speech System using symbol concatenation

Telugu is one of the oldest languages in India. This paper describes the development of Telugu Text-to-Speech System (TTS) using vowel classification. Vowels are most important class of sound in most Indian languages. The duration of vowel is longer than consonants and is most significant. Here vowels are categorized as starting middle and end according to the position of occurrence in a word. The algorithm developed by us involves analysis of a sentence in terms of words and then symbols involving combination of pure consonants and vowels. Wave files are being merged as per the requirement to generate the modified consonants influenced by deergalu (vowel sign) and yuktaksharas generate the speech from a text. Speech unit database consisting of vowels (starting, middle and end) and consonants is developed. We evaluated our TTS using Mean Opinion Score (MOS) for intelligibility and voice quality with and without using vowel classification from sixty five listeners, and got better results with vowel classification