Search arXiv⌕ Search

arXiv · 2610.38316

DRC-TTS: A Dual-Resolution Codec-Based Architecture for Lexicon-Free Malay Speech Synthesis

Abstract

Modern discrete neural audio codecs facilitate speech synthesis by replacing continuous acoustic representations with low-frame-rate discrete token sequences. For Malay speech synthesis, character-level input reduces reliance on pronunciation lexicons. However, combining character-level inputs with low-frame-rate discrete codecs introduces a temporal alignment challenge. When predicted character durations are quantized into frame counts at 12.5 Hz, durations shorter than one codec frame may be assigned zero frames. Consequently, the corresponding character representations may be lost during length regulation, potentially leading to pronunciation omissions. This study proposes DRC-TTS, a dual-resolution codec-based architecture for lexicon-free Malay speech synthesis. The model expands character representations at a temporal resolution of 50 Hz and subsequently downsamples them through strided convolution to 12.5 Hz for discrete codec-token prediction. Objective evaluations show that the proposed model reduces the character error rate (CER) from 27.11% for the baseline to 21.86%, an absolute reduction of 5.25 percentage points. The proposed model also achieves a mean opinion score (MOS) of 4.30, compared with 4.12 for the baseline. These results suggest that the dual-resolution design may improve character-level alignment and perceived speech naturalness under the evaluated conditions.

Explore related subjects

Keep this discovery

Explore connections, maps & timelines

BibTeXRIS

Tianhui Su, Tien-Ping Tan. 2026-10-28. DRC-TTS: A Dual-Resolution Codec-Based Architecture for Lexicon-Free Malay Speech Synthesis. https://arxiv.org/abs/2610.38316

Cite the original work for its findings. Save a collection to share your selection of sources.

KEEP EXPLORING

Related papers

Array-Agnostic High-Order Ambisonics Encoding via Diffusion Posterior Sampling

Ambisonics provides a device-independent representation of three-dimensional sound fields; however, extracting these coefficients from practical microphone array recordings remains strongly dependent on array geometry, acoustic transfer functions, and spatial sampling resolution. This pipeline is typically divided into two distinct tasks: encoding, which entails reconstructing the physically resolvable spatial components, and upscaling, which requires generating higher-order coefficients. Model-based approaches often fail on irregular arrays due to ill-conditioned transfer matrices, whereas conventional neural encoders remain strictly tied to the specific acquisition setup seen during training. Furthermore, existing upscaling methods usually assume ideal low-order inputs, ignoring device-specific array artifacts. We present ADEUPS, a unified framework that couples an array-independent Ambisonic diffusion prior with an explicit physical acquisition model during inference, enabling zero-shot encoding and upscaling across diverse array configurations. We train our prior using an order-masking strategy and perform inference via score blending, which we demonstrate to be crucial for solving the underlying inverse problem. Through extensive numerical evaluations across various array geometries, we show that ADEUPS consistently outperforms task- and array-specific baselines. Finally, we conduct subjective listening tests to confirm the perceptual effectiveness of our approach.

eess.AS↗

Conformal Prediction-Aware DOA Tracking

We introduce an uncertainty-aware DOA-tracking framework combining deep posterior estimation, conformal-prediction calibration, and temporal smoothing. The approach enhances stability against reverberation and burst-like directional noise, yielding more reliable trajectories and greater robustness than conventional DOA-tracking methods.

eess.AS↗

The Importance of Speech Genre for End-of-Turn Detector Training

End-of-turn (EoT) detection has been studied mostly through features, models, and modalities, while the speech genre of the training data has drawn far less attention. We isolate this factor. Using one reproducible annotation pipeline, we build a matched pair of corpora, multi-turn human-human conversational and single-turn human-machine, and train the same streaming ConvLSTM EoT detector on each. Across four test sets the conversational model reaches up to +0.45 higher F1, driven by fewer false alarms on the mid-turn pauses and hesitations that are rare in single-turn corpora. This conversational-training effect is not specific to frame-level decoding. Under a matched complete-or-not test it also appears at the utterance level, across two encoders, but only on conversational test data. Testing only on single-turn speech, as most prior work does, therefore hides the benefit of conversational training data. Against open-source baselines, our 1.8M-parameter model attains the highest F1 on every test set.

eess.AS↗