Search arXiv⌕ Search

arXiv · 2610.04757

Prompt-Consistency Inference for Zero-Shot Flow-Matching Text-to-Speech Models

Abstract

In recent years, flow-matching models have produced significant improvements in zero-shot text-to-speech synthesis. Conditioned on an audio prompt and text, these models learn a velocity field and generate speech by iteratively solving an ODE. During inference, the solver evolves a single state spanning both the prompt and the region to be generated, although only the generated region is ultimately retained. The discarded prompt state, however, still matters; its intermediate values influence generation through the velocity field that couples the two regions. As sampling continues, this state can drift away from the prescribed conditional path, introducing a discrepancy into subsequent generation updates. Unlike the unknown generated trajectory, the prompt path is available in closed form from the reference audio and the initial noise. We exploit this observation with Prompt-Consistency Inference (PCI), a training-free rule that restores the prompt block to its analytic value before each velocity evaluation, while leaving the generated block unchanged. PCI improves speaker similarity and intelligibility across the evaluated flow-matching TTS backbones without additional network evaluations. Our ablation studies further show that PCI keeps post-step prompt discrepancies smaller and that corrections covering the later sampling stages recover much of the observed similarity gain.

Explore related subjects

Keep this discovery

Explore connections, maps & timelines

BibTeXRIS

Vasily Zadorozhnyy, Can Goksen, Kazuhito Koishida, Dung Tran. 2026-10-03. Prompt-Consistency Inference for Zero-Shot Flow-Matching Text-to-Speech Models. https://arxiv.org/abs/2610.04757

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

KEEP EXPLORING

Related papers

Combining Masked Language Modeling and Cross-Modal Contrastive Learning for Prosody-Aware TTS

Text encoders in neural text-to-speech (TTS) see only linguistic content, so prosody must be inferred from paired data and synthesised intonation is often flat. Contrastive text-audio pretraining (CLAPSpeech) injects prosodic knowledge by contrasting acoustic realisations of the same phoneme, but it never trains the encoder to tell different phonemes apart, and its interaction with masked language modelling (MLM) is unexplored. We study a curriculum for a speaker-conditioned phoneme-and-subword encoder: MLM, contrastive alignment over batches of different phonemes, and CLAPSpeech-style same-phoneme refinement. On Russian conversational speech, with each variant conditioning an identical Grad-TTS model, MLM followed by mixed-phoneme alignment gives the lowest word error rate (17.6% versus 32.8% for same-phoneme training alone) and the best quality and intonation ratings. Adding same-phoneme refinement improves same-phoneme retrieval yet yields the worst synthesis: embedding-space retrieval does not predict usefulness for generation.

cs.SD↗

Beyond Naturalness: Probing Automated Text-To-Speech Evaluators on Linguistically Grounded Dimensions

Automated Text-to-Speech (TTS) evaluation methods (Mean Opinion Score (MOS) predictors and Audio Large Language Model (Audio-LLM) judges) are expected to reflect human perception, yet it is unclear how well they capture the distinct aspects of speech that listeners actually perceive. We deconstruct "naturalness" into a linguistically grounded annotation schema spanning 10 distinct perceptual dimensions, and use it to construct the first dimension-level meta-evaluation benchmark for TTS, comprising 860 utterances annotated by trained linguist raters. Results from benchmarking four MOS predictors and four Audio-LLM judges reveal that MOS predictors are more sensitive to acoustic signal-level properties at the prosodic and paralinguistic levels, while Audio-LLM judges show selective, prompt-dependent detection that favors word-level errors and does not generalize across all dimensions. Neither class reliably captures a breadth of linguistically structured, human-perceptible speech errors. Our dataset, annotation schema, and evaluation code are publicly released to support more targeted and interpretable TTS evaluation.

cs.SD↗

Qwen-Audio-Jev: A Unified Interface for Audio Decisions

Large audio language models (LALMs) have made substantial progress in audio understanding. However, autoregressive text generation introduces computational overhead and latency in high-throughput applications that require frequent selection, judgment, and scoring. To address this limitation, we introduce Qwen-Audio-Jev, which augments Qwen-Audio-3.1-ASR-Flash-Next with a dedicated Jev decision head. The model unifies selection, binary judgment, and scoring through three decision primitives: CHOICE, NOUL, and SCORE. Given an audio input, a natural-language question, and an explicit answer specification, Qwen-Audio-Jev directly outputs a task-conditioned probability distribution or a numerical score through typed decision readouts. Evaluation across 18 benchmarks demonstrates the model's effectiveness: CHOICE achieves 78.40 percent accuracy on MMAU and 83.14 percent on MMSU; NOUL achieves 93.76 percent accuracy on the VoxCeleb1-O speaker verification benchmark; and SCORE achieves a normalized mean absolute error of 0.1235 on NISQA overall quality assessment. These results highlight the potential of Qwen-Audio-Jev for efficient audio decision-making in applications such as large-scale audio screening, speaker verification, and automated quality assessment.

cs.SD↗