123,480 research outputs found
Letter-based speech synthesis
Initial attempts at performing text-to-speech conversion based on standard orthographic units are presented, forming part of a larger scheme of training TTS systems on features that can be trivially extracted from text. We evaluate the possibility of using the technique of decision-tree-based context clustering conventionally used in HMM-based systems for parametertying to handle letter-to-sound conversion. We present the application of a method of compound-feature discovery to corpusbased speech synthesis. Finally, an evaluation of intelligibility of letter-based systems and more conventional phoneme-based systems is presented
SNAC: Speaker-normalized affine coupling layer in flow-based architecture for zero-shot multi-speaker text-to-speech
Zero-shot multi-speaker text-to-speech (ZSM-TTS) models aim to generate a
speech sample with the voice characteristic of an unseen speaker. The main
challenge of ZSM-TTS is to increase the overall speaker similarity for unseen
speakers. One of the most successful speaker conditioning methods for
flow-based multi-speaker text-to-speech (TTS) models is to utilize the
functions which predict the scale and bias parameters of the affine coupling
layers according to the given speaker embedding vector. In this letter, we
improve on the previous speaker conditioning method by introducing a
speaker-normalized affine coupling (SNAC) layer which allows for unseen speaker
speech synthesis in a zero-shot manner leveraging a normalization-based
conditioning technique. The newly designed coupling layer explicitly normalizes
the input by the parameters predicted from a speaker embedding vector while
training, enabling an inverse process of denormalizing for a new speaker
embedding at inference. The proposed conditioning scheme yields the
state-of-the-art performance in terms of the speech quality and speaker
similarity in a ZSM-TTS setting.Comment: Accepted to IEEE Signal Processing Letter
SMaTTS: standard malay text to speech system
This paper presents a rule-based text- to- speech
(TTS) Synthesis System for Standard Malay, namely SMaTTS. The
proposed system using sinusoidal method and some pre- recorded
wave files in generating speech for the system. The use of phone
database significantly decreases the amount of computer memory
space used, thus making the system very light and embeddable. The
overall system was comprised of two phases the Natural Language
Processing (NLP) that consisted of the high-level processing of text
analysis, phonetic analysis, text normalization and morphophonemic
module. The module was designed specially for SM to overcome
few problems in defining the rules for SM orthography system before
it can be passed to the DSP module. The second phase is the Digital
Signal Processing (DSP) which operated on the low-level process of
the speech waveform generation. A developed an intelligible and
adequately natural sounding formant-based speech synthesis system
with a light and user-friendly Graphical User Interface (GUI) is
introduced. A Standard Malay Language (SM) phoneme set and an
inclusive set of phone database have been constructed carefully for
this phone-based speech synthesizer. By applying the generative
phonology, a comprehensive letter-to-sound (LTS) rules and a
pronunciation lexicon have been invented for SMaTTS. As for the
evaluation tests, a set of Diagnostic Rhyme Test (DRT) word list was
compiled and several experiments have been performed to evaluate
the quality of the synthesized speech by analyzing the Mean Opinion
Score (MOS) obtained. The overall performance of the system as
well as the room for improvements was thoroughly discussed
Nearest Neighbor-Based Indonesian G2P Conversion
Grapheme-to-phoneme conversion (G2P), also known as letter-to-sound conversion, is an important module in both speech synthesis and speech recognition. The methods of G2P give varying accuracies for different languages although they are designed to be language independent. This paper discusses a new model based on pseudo nearest neighbor rule (PNNR) for Indonesian G2P. In this model, partial orthogonal binary code for graphemes, contextual weighting, and neighborhood weighting are introduced. Testing to 9,604 unseen words shows that the model parameters are easy to be tuned to reach high accuracy. Testing to 123 sentences containing homographs shows that the model could disambiguate homographs if it uses long graphemic context. Compare to information gain tree, PNNR gives slightly higher phoneme error rate, but it could disambiguate homographs
A High Quality Text-To-Speech System Composed of Multiple Neural Networks
While neural networks have been employed to handle several different
text-to-speech tasks, ours is the first system to use neural networks
throughout, for both linguistic and acoustic processing. We divide the
text-to-speech task into three subtasks, a linguistic module mapping from text
to a linguistic representation, an acoustic module mapping from the linguistic
representation to speech, and a video module mapping from the linguistic
representation to animated images. The linguistic module employs a
letter-to-sound neural network and a postlexical neural network. The acoustic
module employs a duration neural network and a phonetic neural network. The
visual neural network is employed in parallel to the acoustic module to drive a
talking head. The use of neural networks that can be retrained on the
characteristics of different voices and languages affords our system a degree
of adaptability and naturalness heretofore unavailable.Comment: Source link (9812006.tar.gz) contains: 1 PostScript file (4 pages)
and 3 WAV audio files. If your system does not support Windows WAV files, try
a tool like "sox" to translate the audio into a format of your choic
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