arXiv subjects

Sri Rama Murty Kodukula

Publications and source records attributed to Sri Rama Murty Kodukula.

4 recordsLinked to original sources

arXiv2026

Self-Supervised Source-Filter Model of Speech Production

Deep learning models of speech seldom represent the source-filter process by which speech is produced, relying instead on latent features with no physical correspondence to articulation or phonation. We propose SoFilNet, a self-supervised, differentiable Source-Filter Network that estimates the parameters of speech production directly from the raw waveform: a source, specified by fundamental frequency (F0), voicing and the aperiodicity of five frequency bands, and a filter, specified by ordered formant poles, additional poles that shape the rest of the spectral envelope, and antiresonance zeros. The model is trained only to reconstruct the input speech; no formant or pitch annotation is used. Since per-bin spectral losses favor blurred spectra wherever the model is uncertain, we train F0 and voicing against autocorrelation pseudo-labels refined with the model's own filter, aperiodicity against the periodic fraction of each band, and formant sharpness against spectral contrast. On VTR-FORMANTS, the estimated formants attain the lowest mean error among the reported methods, including supervised trackers, and on the unseen PTDB-TUG corpus the F0 frame error matches the best classical tracker and is lower than that of all learned trackers. The parameters are also consistent with speech production theory: aperiodicity rises with frequency and orders phone classes by the turbulence of their excitation, and the zeros deepen on nasal consonants, the lower one rising with place of articulation.

eess.AS↗

arXiv2026

CORD-KWS: Calibrated, Order-Aware Detection for Open-Vocabulary Keyword Spotting

Open-vocabulary keyword spotting (KWS) must detect arbitrary keywords without retraining. Cross-attention-based models achieve state-of-the-art performance but require pairwise interaction between the audio and each keyword, recomputing the fused representation for every audio--keyword pair. Embedding-based models avoid this computational cost through independent encoding and similarity scoring, yet remain inferior on challenging benchmarks such as LibriPhrase-hard. We show that this gap is primarily a training issue rather than a limitation of model capacity. While contrastive learning encourages correct keywords to rank above negatives, it does not explicitly calibrate absolute similarity scores, which are critical for applying a fixed threshold across keyword detection. We propose Calibrated, Order-aware Detection KWS (CORD-KWS), a training framework that introduces two complementary objectives while retaining the same encoders, cosine scoring, and O(1) keyword enrollment: a calibrated detection head that supervises absolute scores, and a frame-level CTC objective that preserves phonetic order information. CORD-KWS achieves EERs of 0.43% and 9.64% on LibriPhrase-easy and LibriPhrase-hard, respectively, outperforming existing embedding- and cross-attention-based models.

cs.SD↗

arXiv2026

Enhancing Audio Captioning with Auxiliary AudioSet Semantics

Automatic Audio Captioning (AAC) seeks to generate natural language descriptions of complex acoustic scenes, bridging auditory perception and language understanding. However, word-selection indeterminacy and increasing reliance on large-scale sequence-to-sequence or LLM-based models limit practical deployment. We propose a resource-efficient AAC framework that explicitly grounds caption generation in auxiliary AudioSet semantics. Frame-level acoustic representations extracted using a ConvNeXt encoder are augmented with top-$K$ predicted AudioSet keywords, providing structured contextual cues for decoding. A compact six-layer BART-style decoder conditions on this joint acoustic-semantic representation, enabling caption generation without LLM-scale decoding. The proposed design balances semantic grounding and computational efficiency within a compact architecture. Evaluations on Clotho V2 and AudioCaps confirm competitive caption quality under practical deployment constraints.

eess.AS↗

arXiv2026

Shared Representation Learning for Reference-Guided Targeted Sound Detection

Human listeners exhibit the remarkable ability to segregate a desired sound from complex acoustic scenes through selective auditory attention, motivating the study of Targeted Sound Detection (TSD). The task requires detecting and localizing a target sound in a mixture when a reference audio of that sound is provided. Prior approaches, rely on generating a sound-discriminative conditional embedding vector for the reference and pairing it with a mixture encoder, jointly optimized with a multi-task learning approach. In this work, we propose a unified encoder architecture that processes both the reference and mixture audio within a shared representation space, promoting stronger alignment while reducing architectural complexity. This design choice not only simplifies the overall framework but also enhances generalization to unseen classes. Following the multi-task training paradigm, our method achieves substantial improvements over prior approaches, surpassing existing methods and establishing a new state-of-the-art benchmark for targeted sound detection, with a segment-level F1 score of 83.15% and an overall accuracy of 95.17% on the URBAN-SED dataset.

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