Large Audio-Language Models (LALMs) have set new benchmarks in speech processing, but their deployment is hindered by the KV cache memory footprint during long-context inference. The KV cache grows linearly with audio duration, severely limiting on-device and streaming deployment.
- Core Problem
- Technical Solution
- Audio-Critical Head Identification
- Spectral Score Smoothing (SSS)
- AudioKV Framework
- Experimental Results
- Comparison with Existing Methods
- Open Questions
AudioKV: KV Cache Eviction in Efficient Large Audio-Language Models
Core Problem
Large Audio-Language Models (LALMs) have set new benchmarks in speech processing, but their deployment is hindered by the KV cache memory footprint during long-context inference. The KV cache grows linearly with audio duration, severely limiting on-device and streaming deployment.
General KV cache compression techniques (SnapKV, AdaKV, etc.) excel in text-only LLMs but fail in the audio domain because they overlook the intrinsic temporal continuity of acoustic signals.
Two specific failures: 1. Head-agnostic compression: Uniformly compressing all attention heads is suboptimal because only a subset of heads captures acoustic information 2. Temporal clustering: Raw importance scores cause Top-K selection to concentrate on small token clusters, violating speech's temporal continuity
Technical Solution
1. Audio-Critical Head Identification
Insight: Attention heads in LALMs are modality-specialized. Only a small fraction consistently attends to audio tokens.
Method:
1. Use WhisperX to get word-level timestamps and confidence scores from ASR
2. Retain high-confidence words (τ = 0.95) as reliable anchors
3. Convert timestamps to audio-token index spans
4. For each attention head, measure overlap between high-attention audio tokens and ground-truth audio spans
5. Compute per-head audio-grounding scores S_{l,h}
Head-aware KV cache allocation:
b_{l,h} = w + r + B_score · S_{l,h} / Σ_{l',h'} S_{l',h'}
Where:
- w = fixed local window size
- r = uniform baseline allocation
- B_score = remaining budget distributed proportionally by modality relevance
2. Spectral Score Smoothing (SSS)
Insight: Speech signals encode information in temporally continuous acoustic representations. Importance scores should be smoothly distributed across the temporal sequence.
Problem with existing methods: Top-K selection on raw scores creates dense clusters, violating temporal continuity. Pooling-based smoothing is inherently local.
SSS Method:
1. Treat importance scores as a 1D temporal signal
2. Apply FFT-based global low-pass filter to suppress high-frequency noise
3. Recover smooth global trends
4. Interpolate filtered signal with original scores using mixing coefficient α
smoothed_scores = α · FFT_lowpass(original_scores) + (1 - α) · original_scores
Advantages over pooling: - Global smoothing over entire importance signal (not local neighborhood) - Frequency-aware: separates semantic relevance (low-freq) from spurious alignments (high-freq) - Model-agnostic and computationally efficient (FFT is O(n log n))
AudioKV Framework
Combines both components: 1. Offline: Identify audio-critical heads via ASR-aligned probing 2. Inference-time: Apply SSS to importance scores, then allocate head-aware KV budgets
Experimental Results
Evaluated on Qwen and Gemma series LALMs across ASR and speech translation benchmarks:
Key result: At 40% compression ratio on Qwen3-Omni-30B: - AudioKV: 0.45% accuracy drop (near-full accuracy) - Traditional methods: Catastrophic degradation with repetition failures
AudioKV substantially mitigates ASR degradation compared to head-agnostic baselines while maintaining the same cache budget.
Comparison with Existing Methods
| Method | Head-Aware | Temporal Continuity | Audio-Specific | 40% Compression |
|---|---|---|---|---|
| SnapKV | No | Pooling (local) | No | Catastrophic |
| AdaKV | No | Heuristic | No | Catastrophic |
| PyramidKV | Layer-wise | No | No | Degraded |
| AudioKV | Yes | FFT (global) | Yes | 0.45% drop |
Open Questions
-
Head stability: Are audio-critical heads consistent across different tasks, speakers, and languages?
-
SSS parameter sensitivity: How to choose the mixing coefficient
αand cutoff frequency optimally? -
Extension to video: Can similar head-aware + temporal continuity approaches work for video KV cache?
-
Streaming scenario: How to adapt AudioKV for true streaming where future context is unavailable for FFT?
See Also
- speech-llm — Speech-language model architectures
- qwen3-omni — Qwen3-Omni model family used in AudioKV experiments
- full-duplex-speech-model — Full-duplex speech models