--- id: cao26b_interspeech title: "Audio-NSP: Data-Centric Semi-Autoregressive Generation for Large Audio-Language Models" authors: - Liang Cao - Xize Cheng - Dongjie Fu - Weihao Wu - Fuming You - Zhiyong Wu - Haifeng Hu year: 2026 doi: 10.21437/Interspeech.2026-1737 isca_url: https://www.isca-archive.org/interspeech_2026/cao26b_interspeech.html pdf_url: https://www.isca-archive.org/interspeech_2026/cao26b_interspeech.pdf session: Efficient Inference for ASR and Speech LMs topics: - speech-llm - asr - tts category: speech-llm-dialogue labels: - efficient-on-device - generative-model institutions: - Tsinghua University - Zhejiang University - Tencent funding: - National Natural Science Foundation of China code: url: "" license: "" open_to_collaboration: false wiki_frontmatter: id: cao26b_interspeech category: speech-llm-dialogue labels: - efficient-on-device - generative-model institutions: - Tsinghua University - Zhejiang University - Tencent updated: 2026-09-29 confidence: full-paper digest: v2 source: https://doi.org/10.21437/Interspeech.2026-1737 pdf: https://www.isca-archive.org/interspeech_2026/cao26b_interspeech.pdf wiki_url: https://interspeech-2026-wiki.vercel.app/papers/cao26b_interspeech/ markdown_url: https://interspeech-2026-wiki.vercel.app/papers/cao26b_interspeech/markdown.md --- # Audio-NSP: Data-Centric Semi-Autoregressive Generation for Large Audio-Language Models *Liang Cao, Xize Cheng, Dongjie Fu, Weihao Wu, Fuming You, Zhiyong Wu, Haifeng Hu* [PDF](https://www.isca-archive.org/interspeech_2026/cao26b_interspeech.pdf) · [ISCA page](https://www.isca-archive.org/interspeech_2026/cao26b_interspeech.html) · [DOI](https://doi.org/10.21437/Interspeech.2026-1737) **Category:** `speech-llm-dialogue` · **Labels:** `efficient-on-device`, `generative-model` **TL;DR** — Audio-NSP enables zero-overhead, semi-autoregressive parallel generation in large audio-language models (LALMs) via blockwise SFT and a modality-aware dynamic truncation strategy, achieving up to 3.42x speedups while preserving generation quality. ## Key contributions - Introduces Audio-NSP, a data-centric semi-autoregressive framework that activates parallel token generation via SFT with zero structural modifications or auxiliary heads. - Identifies the decoding dilemma caused by the modal entropy gap between deterministic text tokens and high-entropy, one-to-many acoustic speech tokens. - Proposes a modality-aware dynamic truncation strategy that applies strict thresholds for text and lenient thresholds for audio to prevent degeneration or fidelity loss. - Achieves up to 3.42x speedups across ASR, Spoken Question Answering (SQA), and TTS while outperforming fixed-length Multi-Token Prediction (MTP) baselines. ## Problem Large Audio-Language Models (LALMs) unify speech and text in an autoregressive fashion but suffer from severe inference latency because discrete audio sequences are orders of magnitude longer than text. Prior parallel generation approaches either require training native non-autoregressive models from scratch—destroying pretrained cross-modal reasoning—or rely on auxiliary multi-token prediction (MTP) heads and speculative decoding. These auxiliary head methods introduce memory overhead and struggle to model continuous, high-entropy acoustic mappings, leading to severe audio quality degradation. This paper matters because it resolves the latency bottleneck of LALMs without sacrificing foundation model architecture or audio fidelity. ## Method Audio-NSP appends multiple prediction blocks to the original sequence during training to bypass standard in-place masking inefficiencies. For a sequence X of length L, K starting indices are sampled from the response, and blocks B^(k) of length W are constructed using an anchor token followed by W-1 learnable mask tokens M. Positional embeddings are explicitly matched to original sequence logical positions (P(xtk) = tk, P(Mj) = tk + j), and a customized attention mask ensures causal attention for prefixes, historical context for blocks, and bidirectional intra-block visibility while isolating different blocks. Training uses standard Cross-Entropy loss computed solely on the masked tokens. During inference, the model operates in a predict-verify-accept cycle. At step t, W-1 mask tokens are appended, probability distributions are predicted in parallel, and an optimal acceptance length L is determined using modality-aware dynamic truncation. Because text tokens exhibit low entropy and sharp distributions while acoustic tokens have flatter distributions and higher entropy due to one-to-many speech mappings, a uniform threshold causes single-token degeneration. The acceptance length is defined as L = max {k | forall i <= k, C(xt+i) > tau(mi)}, assigning a strict threshold tau_text = 0.8 for text and a lenient threshold tau_audio = 0.2 for audio. The framework uses VITA-Audio-Plus-Vanilla as a backbone, block length W = 4, dynamic sequence packing up to maximum context length L_max, learning rate 5e-6, batch size 256, and is trained for 140k steps. ## Experimental setup Evaluated on WenetSpeech-meeting, WenetSpeech-net, AIShell-test, LibriSpeech (clean and other) for ASR; LlamaQuestion, TriviaQA, WebQuestion for SQA (Spoken Question Answering); and Seed-test-zh, Seed-test-en, Seed-test-hard, LibriTTS-test-clean for TTS. Metrics include WER/CER for ASR/TTS, Accuracy for SQA, and Tokens Per Step (TPS) for inference speedup. Baselines include VITA-Base and VITA-MTP (VITA-Audio-Plus-Boost). ## Results Audio-NSP achieves an average ASR WER of 5.71 (a minor +1.20% degradation vs 6.65 for MTP) with an average speedup of 3.11x, peaking at 3.42x on LibriSpeech-clean with a WER of 2.88 (vs 3.13 for MTP). In SQA tasks, it matches or outperforms baselines while reaching up to 254 TPS. For TTS, it strictly outperforms MTP across all benchmarks, achieving an average WER/CER of 4.06 (+0.29 degradation vs 4.45 for MTP) with ~1.93x speedup. The more moderate speedup in TTS is a deliberate design choice: continuous acoustic representations have high entropy and low error tolerance, causing the system to safely fall back to finer-grained steps during high-uncertainty transitions. | System / Condition | ASR LibriSpeech-clean (WER_↓_) | ASR AIShell-test (WER_↓_) | TTS Seed-test-en (WER_↓_) | Speedup (TPS) | |---|---|---|---|---| | VITA-Base | 2.00 | 1.94 | 1.85 | 1.00x | | VITA-MTP | 3.13 | 4.72 | 2.21 | ~3.0x | | Audio-NSP (Ours) | 2.88 | 2.29 | 2.13 | 3.42x | ## Limitations The approach relies on a pre-trained LALM backbone and inherits its tokenization and sequence length constraints (L_max). The audio speedup is intentionally capped compared to text due to modality-aware fallback mechanisms required to protect acoustic fidelity. Evaluation is constrained to specific open-source ASR, SQA, and TTS corpora, leaving multilingual cross-lingual transfer at massive scale untested. ## Why read this Speech and ML engineers looking to deploy or accelerate LALMs without retraining foundation models from scratch should read this to understand how to design blockwise parallel generation and handle the modal entropy gap between text and speech. ## Code None released (as of this page's `updated` date). If you are an author with a repo, please claim this entry — see CONTRIBUTING.md. ## Applications Real-time conversational AI assistants, low-latency spoken dialogue systems, and high-throughput speech synthesis and recognition services. ## Institutions / 機構 Tsinghua University, Zhejiang University, Tencent **Funding / 經費:** National Natural Science Foundation of China ## Related - [An Efficient vLLM-Based Inference Pipeline for Unified Audio Understanding and Generation](wang26w_interspeech.md) — same problem · relatedness 2.3/3 - [HybridCodec: Modeling Discrete and Continuous Representations For Efficient Speech Language Models](ploujnikov26_interspeech.md) — same problem · relatedness 2.2/3 - [U-Codec: Neural Speech Codec under Extreme Temporal Compression for Fast High-Fidelity Speech Generation](yang26n_interspeech.md) — same problem · relatedness 2.0/3 - [Accelerating End-to-End ASR via Semi-Autoregressive Speculative Decoding](wu26g_interspeech.md) — shared technique · relatedness 2.0/3 - [AdaTS: Adaptive Token Sampling for Efficient Speech Language Models](sannigrahi26_interspeech.md) — same problem · relatedness 2.0/3 All 950k paper pairs scored by TypeSafe Jev (`scripts/related/`); relatedness 0 = unrelated … 3 = directly comparable.