Jul 2026· International Research Journal of Multidisciplinary Technovation· pp. 266-287· 0 citations· 29 references
Abstract
Voice conversion (VC) is an emerging technology in speech processing that aims to modify an utterance so that it sounds as though it were spoken by another speaker while preserving its linguistic content. High-quality voice conversion has broad applications, including speech synthesis, assistive communication, and entertainment applications such as multilingual dubbing. However, current embedding-guided voice conversion (EGVC) frameworks often struggle with generalization and naturalness under regional data-scarcity conditions. This study explores these limitations by evaluating an EGNVC framework adapted for low-resource regional-language pairs. The proposed framework incorporates the Harvest pitch-extraction algorithm alongside pretrained speaker representations to guide cross-gender pitch transitions while attempting to preserve speaker-identity profiles. Experimental results show that, although the framework successfully shifts macro-level pitch contours across genders, spectral alterations lead to substantial acoustic distortion and reduced intelligibility. Specifically, Kannada speech conversion achieved a localized objective intelligibility score of STOI = 0.12, whereas Malayalam transformations exhibited substantial spectral variation, with an MCD of 169.75, highlighting significant language-specific barriers to regional voice conversion. Kannada achieved higher intelligibility (STOI = 0.12) than Malayalam, whereas Malayalam required greater spectral modification (MCD = 169.75), indicating language-specific challenges in voice conversion. These baseline metrics delineate the empirical limitations of current embedding-guided architectures for Dravidian languages and indicate that substantial advances in spectral mapping are required before such systems can be integrated into real-time assistive or localized voice-synthesis applications.
A Conditional Speaker Normalization module is introduced that conditions the normalization process on speaker embeddings derived from either speaker verification systems or a speaker-height estimation model to mitigate the speaker-dependent nature of Log-Area Ratios.
The resulting 82M-parameter model, Wayu-Paxa-TTS-Edge, enables on-device Thai TTS without reference audio and achieves the lowest pause-placement error and intra-word pause rates among the three systems.
Kunat Pipatanakul, Potsawee Manakul, Warit Sirichotedumrong et al.· 1 citation
To the authors' knowledge, Persona-ASR is the first TS-ASR system for the Kazakh language, establishing a foundation for multilingual personalized ASR in low-resource settings.
Rakhat Meiramov, Tomiris Rakhimzhanova, Adil Taibassarov et al.· Machine Learning and Knowled...· 0 citations
This paper presents a novel approach to controllable singing voice conversion through the application of the information-bottleneck principle and provides a comprehensive theoretical analysis of the information-bottleneck framework adapted for sequential audio data and outline a robust architecture designed to exploit...
Anna Pellegrini· Global Media and Social Scie...· 0 citations
CASA, a simpler architecture combining Whisper-medium and Qwen3.5-2B that achieves state-of-the-art performance while providing a more interpretable separation between speech delivery and content, is proposed.
Nhan Phan, Ilona Lähteenmäki, Anna von Zansen et al.· 0 citations
Speech assessment tools for speakers with speech pathology must be both accurate and interpretable if they are to be adopted in clinical practice. Existing reference-audio measures such as the Neural Acoustic Distance (NAD) reach high speaker-level correlations with listener intelligibility scores but operate on self-s...
B. Halpern, Thomas B. Tienkamp, D. Abur et al.· 0 citations
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