Elastic Time: Dynamic Frame Rate Bottlenecks for Neural Audio Coding
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Introduction to the show: ident: AI Radio. Generated commentary on the latest Artificial Intelligence papers.
Tom: Today's paper: "Elastic Time: Dynamic Frame Rate Bottlenecks for Neural Audio Coding".
Jane: The gist The Elastic Time method introduces a dynamic frame-rate bottleneck that converts fixed-frame-rate autoencoders into dynamic ones by learning a lightweight latent predictor to decide which frames can be…
Tom: First, who's behind it and why it matters.
Paper summary: Tom: So, to summarize Elastic Time: Dynamic Frame Rate Bottlenecks for Neural Audio Coding, they claim that existing neural audio autoencoders often use a fixed latent frame rate which means they allocate equal temporal budget across parts of the audio that have very different information density.
Jane: And this usually results in unnecessarily long sequences when you need high fidelity but only need a certain level of detail in specific sections.
Lu: The authors introduce Elastic Time as a dynamic frame-rate bottleneck that converts those fixed-frame-rate autoencoders into dynamic ones by learning a lightweight latent predictor to decide which frames can be skipped and later reconstructed.
Meng: So they are adding this predictor to the existing structure without having to modify the number of latent dimensions, which is smart because it keeps the underlying model structure intact.
Lalam: The goal is enabling efficient greedy boundary selection at inference time, allowing for a per-sample target kept length N, which corresponds to a kept fraction rho equal to N divided by T.
Tom: It really matters because this method enables deployment-time rate control while improving the efficiency-quality tradeoffs relative to baselines.
Conclusion: Jane: Looking at the title, Elastic Time suggests a system that can adjust its temporal resolution based on content, which is really about making models adaptive to the input itself.
Tom: And the authors are Dimitrios Bralios and Paris Smaragdis and Minje Kim from Illinois Urbana-Champaign and MIT; they've shown how this dynamic frame-rate approach works by using a learned latent predictor.
Lu: What this means for us is that we can potentially deploy these audio models much more efficiently because the computational load scales with the actual information present in the audio, not just a fixed number of frames.
Meng: Practically, it means less time on inference for users who don't need every single frame at maximum resolution and better fidelity where it matters most.
Lalam: This research gives us a flexible mechanism for adjusting temporal resolution in audio processing without needing external semantic supervision to dictate the changes.
University of Illinois Urbana-Champaign · Massachusetts Institute of Technology
cs.SD, cs.LG, eess.AS
Submitted: 2026-06-25
Updated: 2026-06-25
Comments: Interspeech 2026
Journal ref: Interspeech 2026
DOI: 10.21437/Interspeech.2026-3031
Code: https://github.com/dbralios/elastic-time
License: http://creativecommons.org/licenses/by/4.0/
Importance score: 79/100
The gist: The gist The Elastic Time method introduces a dynamic frame-rate bottleneck that converts fixed-frame-rate autoencoders into dynamic ones by learning a lightweight latent predictor to decide which
Key concepts
- Lightweight Predictor
- This is a small model trained to predict future latent frames using past context. It learns the short-term dynamics of the latent space, enabling the system to anticipate which frames are redundant and can be safely dropped during inference for faster processing.
- Chunk Procedure
- The process of partitioning a long sequence of latents into variable-length segments based on a predictability criterion. The goal is to identify segments where most frames can be approximated by an 'anchor' frame, which are then candidates for removal or skipping.
- Boundary Selection
- Deciding where to split the latent sequence into these variable-length chunks. Two methods are used: a fast greedy procedure and an exact dynamic program (DP) that calculates a cost function based on predicted reconstruction error to find the optimal split points.
Terminology
Summary
The gist The Elastic Time method introduces a dynamic frame-rate bottleneck that converts fixed-frame-rate autoencoders into dynamic ones by learning a lightweight latent predictor to decide which frames can be skipped during inference, enabling efficient rate control.
Proposed Method
-
The core technical contribution is a
lightweight predictor trained to predict future latent frames from past context
. -
This predictor models short-term latent dynamics along the temporal axis by predicting the next frame given the previous predicted frame and recurrent state.
-
The process involves a
chunk procedure that partitions the latent sequence into N variable-length segments according to a predictability criterion, retaining one “anchor” frame from each segment
. -
The
dechunking procedure produces the full-length reconstructed latent sequence yˆ, using P
. -
Within each segment m (with length lm > 1),
we perform one autoregressive predictor rollout for lm − 1 steps, starting from the anchor frame yam
1), we perform one autoregressive predictor rollout for lm − 1 steps, starting from the anchor frame yam>.
Boundary Selection
The chunking is guided by the assumption that latent predictability implies redundancy: if a segment of latents can be approximated from its anchor via P, then these frames can be dropped
.
Two solvers are proposed for boundary selection: (i) an approximate but efficient greedy procedure
and (ii) an exact dynamic program (DP)
The greedy algorithm runs in O(T log T) time after initializing local costs
.
The DP formulation defines the cost as c(s, l) = Xl−1 i=1 d yˆ (s) s+i, ys+i
.
Training Objectives
The model is trained with two coupled objectives: (i) accurate multi-step latent prediction and (ii) preservation of reconstruction quality after chunking/dechunking
.
The predictor P is trained using a rollout prediction loss Lroll
which penalizes multi-step rollout error
.
The model also uses an auxiliary prediction loss Lvalid computed on the frames replaced during this transformation
.
Evaluation and Results
The evaluation compares four main variants: two greedybased models, one trained at a fixed kept fraction ρ = 0.5 (denoted ET-greedy@0.5) and one trained over a range ρ ∼ U(0.5, 0.995) (ET-greedy), and two DP-based models
.
The results show that greedy and DP chunking perform similarly
.
ET generally achieves lower mel-d and FAD on SongDescriber and DAPS, with the clearest gains in FAD
.
The conclusion is that dynamic frame-rate scalability could improve the efficiency of downstream generation and long-context audio modeling
.
Conclusion
Elastic Time enables framerate scalable models via content-adaptive latent decimation
.
The method is implemented as a Re-Bottleneck plug-in on top of a frozen pretrained autoencoder
.
It achieves competitive efficiency-quality tradeoffs relative to fixed-rate and prior dynamic chunking baselines
.
The overall impact is that the method allows for deployment-time rate control without external semantic supervision
.
Acknowledgments
This work was supported by Electronics and Telecommunications Research Institute (ETRI) grant funded by the Korean government
.
References
The references listed are [1] A. Defossez, J. Copet, G. Synnaeve, and Y. Adi, “High fidelity neural audio compression,” Transactions on Machine Learning Research, 2023. [5] Z. Evans, J. D. Parker, C. Carr, Z. Zukowski, J. Taylor, and J. Pons, “Stable audio open,” in Proc ICASSP IEEE 2025 1–5 <ref:2606.27320#pg6>. [17] Y. Chae, W. Choi, Y. Takida, J. Koo, Y. Ikemiya, Z. Zhong, K. W Cheuk et al., “Variable bitrate residual vector quantization for audio coding,” in Proc ICASSP IEEE 2025 1–5. [19] J. Li, Y. Qian, Y Hu, leying zhang et al., “Flexicodec: A dynamic neural audio codec for low frame rates,” in Proc ICLR 2026. [23] D. Bralios, J. Casebeer, and P. Smaragdis, “Re-bottleneck: Latent restructuring for neural audio autoencoders,” in 2025 IEEE 35th International Workshop on Machine Learning for Signal Processing (MLSP) IEEE 2025 1–6 <ref:2606.27320#pg2>. [43] C. J. Steinmetz and J. D. Reiss, “auraloss: Audio focused loss functions in pytorch,” in Digital music research network one-day workshop (DMRN+ 15) 2020 <ref:2606.
Improvements for AI systems
-
The Elastic Time (ET) module enables
deployment-time rate control
by learning alightweight latent predictor used to decide which frames can be skipped and later reconstructed.
This allows downstream models to adapt their temporal resolution dynamically based on the required compression factor, facilitating more efficient deployment. -
The system supports frame-rate scalability by allowing the model to adjust its effective latent sequence length at inference time. By using a
predictability criterion
guided by the learned predictor P, it can performefficient greedy boundary selection
or an exact dynamic programming procedure to select segments, effectively convertingfixed-frame-rate autoencoders into dynamic ones.
-
The training objective optimizes for both prediction accuracy and reconstruction quality simultaneously. The loss function is defined as
Lpred = λrollLroll + λvalidLvalid,
ensuring that the predictor P is trained to producestable rollouts
while maintaining compatibility with the base model's reconstruction ability, which is crucial for preserving perceptual quality during dynamic decimation. -
The Re-Bottleneck framework allows for the reuse of
pretrained, publicly available fixed-rate models
by focusing learning on a latent-space module that specializes indynamic temporal allocation while preserving the base model’s strong reconstruction ability.
This significantly reduces the computational cost associated with training audio autoencoders from scratch. -
The method provides a flexible mechanism for adjusting temporal resolution, which is expected to
facilitate more efficient downstream modeling for generation and long-context tasks,
such as autoregressive next-token prediction or diffusion-based generative models that are bottlenecked by latent sequence length.
Abstract
Neural audio autoencoders have become a core component of compression, feature extraction, and generation. However, while existing systems support variable bitrate, the vast majority of models still operate at a fixed latent frame-rate, allocating equal temporal budget to regions with very different information density, which can result in unnecessarily long sequences. We introduce Elastic Time, a dynamic frame-rate bottleneck that converts fixed-frame-rate autoencoders to dynamic ones. Our method learns a lightweight latent predictor used to decide which frames can be skipped and later reconstructed, enabling efficient greedy boundary selection at inference. Experiments show our method enables deployment-time rate control while improving efficiency-quality tradeoffs relative to baselines. Overall, we provide a flexible mechanism for adjusting temporal resolution in audio autoencoders, potentially facilitating more efficient downstream modeling for generation and long-context tasks.
Sources
- SpectroStream: A Versatile Neural Codec for General Audio
- Moshi: a speech-text foundation model for real-time dialogue
- SoundStorm: Efficient Parallel Audio Generation
- Back to Ear: Perceptually Driven High Fidelity Music Reconstruction
- SALAD-VAE: Semantic Audio Compression with Language-Audio Distillation
- Beyond Fixed Frames: Dynamic Character-Aligned Speech Tokenization
- Dynamic Chunking for End-to-End Hierarchical Sequence Modeling
- GLU Variants Improve Transformer
- Moisesdb: A dataset for source separation beyond 4-stems
- Muon is Scalable for LLM Training
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