Search arXiv⌕ Search

arXiv · 2610.39168

Reading-Level Verification of Kanji Words in Japanese Financial Speech Synthesis

Abstract

Few-shot voice cloning makes it cheap to produce daily Japanese financial commentary with several synthetic speakers, but kanji words with more than one reading, such as the word for "market" (shijo vs. ichiba) or "overall trend" (taisei vs. oze), are still misread. A common quality gate transcribes each synthesized sentence with automatic speech recognition (ASR), converts both the input and the transcript to kana, and regenerates on mismatch. We show that this gate is unreliable in both directions: when the audio contains a misreading, the ASR language model frequently writes the expected kanji back, hiding the error, and when the audio is correct, ordinary recognition errors trigger needless regeneration. We propose a reading-level check that scores the expected reading and its confusable alternatives with a connectionist temporal classification (CTC) hiragana recognizer and flags a word when an alternative explains the audio better. On 540 utterances from three voice-cloned production voices covering 30 financial terms, with misreadings injected under controlled conditions, the transcript-based gate detected 78.9% of misreadings at a 44.4% false-alarm rate, whereas the proposed check detected 91.7% at 5.0% (AUC 0.990). On natural kanji input, it caught a front-end error that the transcript-based gate missed for one voice, and reduced regeneration requests from 38.9% to 6.1% of sentences.

Explore related subjects

Keep this discovery

Explore connections, maps & timelines

BibTeXRIS

Cheng Zhang. 2026-10-29. Reading-Level Verification of Kanji Words in Japanese Financial Speech Synthesis. https://arxiv.org/abs/2610.39168

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

KEEP EXPLORING

Related papers

IsoNet: Face-Guided Speech Extraction for Compact Microphone Arrays

Listening to one chosen talker in a crowded room is effortless for people but hard for small devices, whose microphone arrays are too compact for classical beamforming to separate nearby voices. We present IsoNet, a select-to-listen system: the user selects a face in the camera view, and a 4-microphone array with a 9.4 cm aperture returns that person's speech. IsoNet fuses multi-channel complex spectra, GCC-PHAT spatial cues, and a face embedding at the bottleneck of a U-Net mask estimator, trained with auxiliary direction-of-arrival supervision and an SNR curriculum on simulated reverberant VoxCeleb mixtures. On a hard test set spanning -1 to 10 dB SNR, IsoNet reaches 9.31 dB SI-SDR, a 4.85 dB improvement over the mixture, and raises PESQ from 1.40 to 2.13 and STOI from 0.72 to 0.84. The gains are largest where the task is hardest (5.71 dB SI-SDR improvement at -1 to 1 dB SNR), and IsoNet outperforms oracle-steered delay-and-sum and MVDR beamformers by 9.7 and 10.9 dB SI-SDR. Ablations show that visual and spatial cues are complementary, and a 4.9M-parameter audio-spatial variant stays within 0.16 dB of the full model while running 26 times faster than real time.

cs.SD↗

A learning-based method of automatic music transcription

Automatic Music Transcription (AMT) seeks to convert acoustic music recordings into symbolic musical notation, a process that remains remarkably difficult for polyphonic piano due to concurrent polyphony, harmonic partial interference, expressive timing, and severe pitch activation sparsity. In this paper, we formulate polyphonic piano transcription as a frame-level multi-label classification problem across the standard 88 piano pitches (MIDI notes 21 to 108). Raw audio waveforms are processed using the Constant-Q Transform (CQT) to construct a geometrically spaced log-frequency time-frequency representation that aligns with the chromatic musical scale across seven octaves (252 frequency bins per frame). We systematically investigate recurrent sequence models, contrasting unidirectional Long Short-Term Memory (LSTM) with Bidirectional LSTM (BiLSTM) backbones, and propose two architectural adaptations tailored to recurrent temporal representations: a modified additive attention module and a modified multihead attention mechanism equipped with hyperbolic tangent activation and head-wise summation. Furthermore, to mitigate the severe class imbalance between inactive and active notes (approximately 41:1), we evaluate Weighted Binary Cross-Entropy (WBCE) alongside standard Binary Cross-Entropy (BCE). Rigorous empirical evaluations on the MAPS and MAESTRO datasets demonstrate steady, compounding gains from bidirectionality and attention mechanisms. Our proposed Multihead Attention-BiLSTM architecture attains an F1-score of 88.20%, an accuracy of 78.90%, a precision of 0.89, and a recall of 0.86, substantially outperforming the baseline unidirectional LSTM (F1-score of 71.20%) while maintaining a compact parameter budget of 4.43M. Qualitative piano-roll visualizations confirm that our model generates faithful transcriptions even in complex harmonic passages.

cs.SD↗

Lightweight Underwater Acoustic Vessel Classification for Embedded Deployment

We propose a Raspberry Pi-based underwater vessel classification system for edge deployment under limited connectivity, bandwidth and power in offshore acoustic surveillance. Using ShipsEar, we compare six classifiers on a Raspberry Pi 5 through a complete audio-to-prediction pipeline. We assess classification performance, processing latency, CPU utilisation, memory consumption and monitored power. Supporting experiments examine acoustic feature selection, statistical pooling and Transformer patch configurations. The two-layer convolutional neural network (CNN) achieved a mean macro-F1 of \textbf{(99.14 \textpm{} 0.32)\%} across five random seeds. The radial basis function support vector machine (RBF-SVM) achieved \textbf{98.83\%}, with a median end-to-end latency of 28.109~ms compared with 120.655~ms for the CNN. This favourable trade-off motivated RBF-SVM deployment on the Raspberry Pi Zero 2 W. It maintained the same classification performance with a median end-to-end latency of \textbf{349.09~ms} and a sustained throughput of \textbf{2.962 classifications/s}, or \textbf{14.8} times the rate required for consecutive five-second audio segments.

cs.SD↗