REVIEW 2 major objections 5 minor 78 references
A Unifying Perspective on Audio Generative Modeling: Latent Representations and Modeling Strategies
T0 review · 2 major / 5 minor · reviewed 2026-08-14 · deepseek-v4-flash
Pith's one-line read Audio generative quality is determined by matching a latent's dependency horizon and conditional ambiguity to the model, not by choosing discrete or continuous tokens.
desk verdict Organizing audio generation around representation-model pairs is a genuine contribution, but the paper's prescriptive Principle 2 remains underdetermined by its own proxies. read the letter →
The pith
A machine-rendered reading of the paper's core claim, the machinery that carries it, and where it could break.
The reading
What carries the argument
The load-bearing machinery is Principle 2—'predict long-range structure, generate ambiguous detail'—made measurable by two diagnostics. Dependency horizon is captured by the context-ablation curve $h_\delta$, which asks how long useful history keeps improving validation loss; conditional ambiguity is captured by how many perceptually valid samples remain under fixed conditioning, or by the quality gap between a deterministic regressor and a distributional head. Around this pair the paper organizes the rest: a four-objective representation design space (burden, distortion, empirical modelability, streaming compatibility), the chain-rule factorizations of RVQ's time/depth grid and of sequential models, and critical-path network evaluations $E_G$ as a cost measure distinct from latency or token count. These pieces do the work of converting 'semantics is low-entropy, acoustics is high-entropy' from an intuition into placement questions about where global versus local capacity should go.
What would settle it
Run a controlled comparison of two latent configurations matched in reconstruction distortion, training compute, and inference cost—say a 12.5 Hz, eight-codebook stream versus a 75 Hz single-codebook stream—and measure downstream generation quality while sweeping usable context length. If the configurations show identical dependency-horizon and conditional-ambiguity profiles yet one is consistently easier for the same generator family to model, the claim that dependency horizon and ambiguity rather than token category or codebook depth govern modelability would be wrong.
Extended reading notes
Core claim
The central claim is that the distinction between discrete and continuous latents describes only the output interface, while dependency horizon, conditional ambiguity, and streaming requirements determine how that interface should be modeled. The paper proposes two principles: reconstruction quality does not determine modelability (Principle 1), and one should predict long-range structure while generating ambiguous detail (Principle 2). Concretely, after conditioning on the past, each latent variable is treated as either long-range structure $s_t$ or remaining realization detail $a_t$, with the chain rule $H(s_t,a_t|z_{<t})=H(s_t|z_{<t})+H(a_t|z_{<t},s_t)$ making plain that the two roles demand different modeling capacity. Dependency horizon is measured by a context-ablation curve $h_\delta=\min\{h:L(h)\le L(h_{\max})+\delta\}$, and conditional ambiguity by diversity among perceptually acceptable samples or by the gap between a deterministic predictor and a distributional head. Applied to representative systems, this yields the paper's sharpest conclusions: RVQ provides ordered capacity but not ordered semantics; AudioLM's semantic-then-acoustic cascade is one placement of the global/local boundary, not a universal template; and autoregressive, iterative, and hybrid schemes differ chiefly in how they trade retained conditional structure against critical-path generation cost.
Load-bearing premise
The framework's design advice rests on the assumption that a latent's variables can be split into long-range structure versus conditionally ambiguous detail, and that this split can be read off from context-ablation curves and ambiguity proxies; the paper concedes the split may remain entangled and that the controlled comparisons needed to verify it are still missing.
Editorial extensions
If this is right
- Codec and generator benchmarks should report generation quality, not just reconstruction, and should match perceptual distortion, training compute, inference cost, and lookahead when comparing latents.
- A latent whose context-ablation curve keeps improving with history should be handled by a long-context autoregressive or hybrid model; a latent whose remaining ambiguity is large under conditioning should be paired with a diffusion, flow, mixture, or masked-iterative head.
- Residual codebook order should not be read as a semantic hierarchy; semantic supervision or factor assignment is a deliberate placement decision, not a property RVQ already provides.
- Swapping a discrete codec for a continuous latent, or vice versa, does not remove the global-versus-local allocation problem; the same dependency and ambiguity analysis applies on both sides of the interface.
- For streaming systems, the framework predicts that hybrid designs—a causal outer process over blocks with an iterative or distributional inner head—are a natural fit, because they put long-range state on the serial path and ambiguous detail on the parallel inner path.
Reading between the lines
- Beyond the paper: if Principle 2 is causal rather than merely descriptive, measuring the context-ablation curve and conditional-ambiguity proxies on a candidate latent before training a full generator could serve as a cheap screening test; the paper does not itself propose such a protocol.
- Beyond the paper: the framework implies a direct experiment for the 'capacity is moved, not removed' claim—hold nominal bitrate fixed and compare low-frame-rate/deep-RVQ against high-frame-rate/shallow-RVQ at matched compute and distortion; if one configuration dominates on downstream generation despite identical dependency and ambiguity profiles, the exchange is not neutral.
- Beyond the paper: the semantic-rate curve suggested for continuous latents could be extended to discrete latents by sweeping codebook count and codebook size at fixed bitrate, turning the framework's diagnostic dimensions into a standard modelability report.
Editorial analysis
A structured set of objections, weighed in public.
Referee Report
Summary. This paper argues that audio generative models should be analyzed and designed as a coupled pair: a latent representation plus a distribution model matched to its dependency structure. It organizes representation design through four objectives (burden, distortion, empirical modelability, streaming compatibility) and distribution modeling through two diagnostic dimensions (dependency horizon and conditional ambiguity). The chain-rule factorizations in Equations (6), (9), (11), and (15) are exact identities, and the paper uses them to frame existing systems: RVQ provides ordered capacity but not ordered semantics; AudioLM's semantic–acoustic cascade is one possible placement of the global/local boundary; AR, iterative refinement, and hybrid schemes differ in how they trade dependency horizon against critical-path cost. The paper presents no new experiments; Table 1 is explicitly a worked literature analysis, and Section 7 openly lists the controlled comparisons that are still missing.
Significance. The paper's main value is as a conceptual framework and a common vocabulary for comparing representation–model pairs. Its strengths are its internal consistency: the core chain-rule equations are exact identities, no free parameters are fitted to make the framework work, and the paper is unusually transparent about what it does not validate (Section 7). The four-objective framing and the insistence that reconstruction quality does not determine modelability (Principle 1) are useful correctives to codec-only evaluation. The paper also poses its central design claim, Principle 2, as an explicit hypothesis rather than a proven result, and it identifies the controlled experiments needed to test it. If the operationalization gap discussed below can be closed, the framework would be a genuinely useful unifying perspective for the field.
major comments (2)
- [§2.3, Eq. (8)] The paper's actionable design guidance rests on Principle 2, but the proposed operational proxies do not determine a unique split between s_t and a_t. Equation (8) defines hδ as a history-length threshold on validation loss, and Section 2.3 defines conditional ambiguity via sample diversity and the deterministic-versus-distributional head gap. These measure different quantities, and neither is shown to characterize a partition of z_t. A variable such as speaker identity under text-only conditioning has a long dependency horizon (hδ is large, so it should receive global capacity) and high conditional ambiguity (multiple speakers are valid, so the ambiguity proxy says delegate it to a local or iterative generator). The two diagnostics therefore give opposite advice for the same variable, and Principle 2's recommendation is underdetermined. The paper honestly concedes in Section 2.3 that s_t and a_t 'need not be explicit coordinates' and in Section 7 that controlled comparisons are missing, but the framework's central design claim needs either a decision procedure that resolves such conflicts or an explicit downgrade of Principle 2 to a heuristic checklist.
- [§6.5] The design hypothesis for unified audio models restates Principle 2 under the names g and ℓ, and therefore inherits the same underdetermination. The claim that an error in g changes global continuation while multiple values of ℓ remain perceptually valid presupposes that the g/ℓ boundary is known before the framework is applied; but the paper provides no method to measure whether a given variable belongs to g or ℓ for an arbitrary latent. Section 7.4 lists possible boundaries (between tokenizers, RVQ layers, model stages, outer/inner loops) but does not say how to test which one is correct. This leaves the main design advice of the paper unable to be applied uniquely to a new representation–model pair.
minor comments (5)
- [§2.3] Principle 2 is labeled a principle but described in the same section as a 'design hypothesis'; recommend renaming it to 'Hypothesis 2' for consistency and to avoid overclaiming.
- [§2.3 and §4.2] The notation z_t = (s_t, a_t) is introduced for the chain-rule decomposition, but the aggregate symbols s and a used in Section 4.2 and Section 6.4 are never formally defined; please align the notation.
- [Table 1] The table's 'Global signal' column mixes variables explicitly exposed by the representation with external conditioning (for example, text conditioning in VALL-E and MusicGen); a separate column or a legend would clarify the distinction.
- [Throughout] There are several formatting and typographical issues, including 'V ALL-E' for 'VALL-E', 'V oicebox' for 'Voicebox', and inconsistent capitalization of 'VAE' and 'mel'; these should be cleaned up.
- [References] The manuscript cites a number of 2026 arXiv preprints (e.g., MOSS-Audio-Tokenizer, DashengTokenizer, WavCube, TADA) as established systems; for a journal version, please indicate their publication or review status, or mark them clearly as preprints.
Circularity Check
No significant circularity: the framework is organizational, its core equations are identities, and the author's prior systems are illustrative rather than load-bearing.
full rationale
The paper makes no fitted-input-called-prediction move. Equations (6), (7), (9), (11), and (15) are chain-rule identities used only to organize where dependencies can be placed, and the paper explicitly refuses to infer architecture from them: "The equality itself favors no architecture" (Section 4) and "They do not determine one architecture from the latent alone" (Section 6). Principle 2 is presented as a design hypothesis, not as a derived theorem: "This is the lens we use throughout the paper, but st and at need not be explicit coordinates of a learned latent. They denote functional roles that may remain entangled in practice" (Section 2.3). The operational proxies are offered as testable questions with an explicit caveat: "Neither proxy identifies a unique semantic-versus-acoustic split, but together they turn the two concepts into testable questions" (Section 2.3). Section 7 acknowledges the missing controlled comparisons: "The table does not establish that one allocation is optimal; that requires the controlled comparisons proposed in Section 7" (Section 6.4). Thus no empirical prediction is being passed off as already validated by the framework itself. The advice that long-horizon variables should receive global capacity is close to definitional if one identifies st as 'variables requiring long-context prediction', but the paper labels it a design hypothesis and leaves the st/at separation explicitly entangled, so this is not a hidden reduction of an output to an input. The only mild concern is the number of self-citations (UniAudio, LLM-Codec, HiFi-Codec, ALMTokenizer, WavCube, among others) used as worked examples, but the framework's internal logic also leans heavily on external systems (AudioLM, VALL-E, MusicGen, SoundStorm, CALM, DiTAR, DAC), and accepting or rejecting the self-cited papers would not change the identities in Equations (6)-(15) or the four-objective design space. No uniqueness theorem is imported from the authors' prior work, and no parameter is fitted and then renamed as a prediction.
Assumptions & free parameters
assumptions (4)
- standard math The exact chain-rule expansion of H(z) over time and codebook depth is the correct basis for comparing modeling schemes (Equations 6, 9, 11, 15).
- ad hoc to paper Long-range structure st and conditionally ambiguous detail at can be separated in audio latents (Principle 2, Section 2.3).
- domain assumption The context-ablation curve L(h) and its horizon hδ (Equation 8) give a meaningful measure of dependency horizon across model families and budgets.
- domain assumption Perceptual equivalence can serve as a conditioning-relative criterion for conditional ambiguity.
invented entities (2)
-
dependency horizon
-
conditional ambiguity
Cite this review
Pith. "Pith review of A Unifying Perspective on Audio Generative Modeling: Latent Representations and Modeling Strategies." pith.science (2026). https://pith.science/paper/JHBN3UEO
@misc{pith2026260808667,
author = {Pith},
title = {Pith review of: A Unifying Perspective on Audio Generative Modeling: Latent Representations and Modeling Strategies},
year = {2026},
howpublished = {\url{https://pith.science/paper/JHBN3UEO}},
note = {Machine review of arXiv:2608.08667}
}
read the original abstract
Every audio generative system makes two coupled decisions: what representation to generate, and how to model its distribution. This paper organizes audio generative modeling around this coupling. For representation design, we compare discrete, continuous, and hybrid latents through four objectives: representation burden, distortion, empirical modelability, and streaming compatibility. For distribution modeling, rather than treating a latent's difficulty as an intrinsic scalar, we use two diagnostic dimensions: dependency horizon, how far useful context extends, and conditional ambiguity, how much uncertainty remains after conditioning. These dimensions refine the common semantic-versus-acoustic intuition: variables with a long dependency horizon should receive global modeling capacity. Conditionally ambiguous detail may be delegated to a local or iterative generator. Applied to representative systems, this view shows that RVQ's residual order gives ordered capacity but not ordered semantics, that AudioLM's semantic-versus-acoustic cascade is one explicit placement of this boundary rather than a universal template, and that autoregression, iterative refinement, and hybrid designs differ chiefly in how they trade dependency horizon against critical-path generation cost. The distinction between discrete and continuous latents describes the output interface; dependency horizon, conditional ambiguity, and streaming determine how that interface should be modeled. Rather than cataloguing individual systems, we provide an evaluation and design framework for comparing representation-model pairs.
Figures
Figures from the paper (1 more)
Reference graph
Works this paper leans on
-
[1]
SoundStream: An End-to-End Neural Audio Codec
Zeghidour et al. SoundStream: An End-to-End Neural Audio Codec. https://arxiv.org/abs/2107. 03312, 2021. IEEE/ACM TASLP
work page 2021
-
[2]
High Fidelity Neural Audio Compression
Défossez et al. High Fidelity Neural Audio Compression. https://arxiv.org/abs/2210.13438, 2023. (EnCodec), TMLR
arXiv 2023
-
[3]
Neural Codec Language Models are Zero-Shot Text to Speech Synthesizers
Wang et al. Neural Codec Language Models are Zero-Shot Text to Speech Synthesizers. https://arxiv. org/abs/2301.02111, 2023. (V ALL-E)
arXiv 2023
-
[4]
Simple and Controllable Music Generation
Copet et al. Simple and Controllable Music Generation. https://arxiv.org/abs/2306.05284, 2023. (MusicGen), NeurIPS
arXiv 2023
-
[5]
UniAudio: An Audio Foundation Model Toward Universal Audio Generation
Yang et al. UniAudio: An Audio Foundation Model Toward Universal Audio Generation. https://arxiv. org/abs/2310.00704, 2024. ICML
arXiv 2024
-
[6]
Moshi: a speech-text foundation model for real-time dialogue
Défossez et al. Moshi: a speech-text foundation model for real-time dialogue. https://arxiv.org/abs/ 2410.00037, 2024. (incl. Mimi)
arXiv 2024
-
[7]
AudioLM: a Language Modeling Approach to Audio Generation
Borsos et al. AudioLM: a Language Modeling Approach to Audio Generation. https://arxiv.org/abs/ 2209.03143, 2022
arXiv 2022
-
[8]
DiTAR: Diffusion Transformer Autoregressive Modeling for Speech Generation
Jia et al. DiTAR: Diffusion Transformer Autoregressive Modeling for Speech Generation. https://arxiv. org/abs/2502.03930, 2025. ICML
arXiv 2025
Show all 78 references
-
[9]
Continuous Audio Language Models
Simon Rouard, Manu Orsini, Axel Roebel, Neil Zeghidour, and Alexandre Défossez. Continuous Audio Language Models. https://arxiv.org/abs/2509.06926, 2025. (CALM; consistency head over an audio V AE; from the Mimi/Moshi team)
2025
-
[10]
Cover and Joy A
Thomas M. Cover and Joy A. Thomas.Elements of Information Theory. Wiley-Interscience, 2nd edition, 2006
2006
-
[11]
UniAudio 1.5: LLM-driven Audio Codec is A Few-shot Audio Task Learner
Yang et al. UniAudio 1.5: LLM-driven Audio Codec is A Few-shot Audio Task Learner. https://arxiv. org/abs/2406.10056, 2024. (LLM-Codec), NeurIPS
2024 arXiv
-
[12]
The Information Bottleneck Method
Tishby, Pereira, and Bialek. The Information Bottleneck Method. https://arxiv.org/abs/physics/ 0004057, 2000. 18
2000
-
[13]
High-Fidelity Audio Compression with Improved RVQGAN
Kumar et al. High-Fidelity Audio Compression with Improved RVQGAN. https://arxiv.org/abs/2306. 06546, 2023. (DAC), NeurIPS
2023
-
[14]
Discrete Audio Tokens: More Than a Survey! https://arxiv.org/abs/2506.10274, 2025
Mousavi et al. Discrete Audio Tokens: More Than a Survey! https://arxiv.org/abs/2506.10274, 2025. TMLR
2025
-
[15]
HiFi-Codec: Group-residual Vector quantization for High Fidelity Audio Codec
Yang et al. HiFi-Codec: Group-residual Vector quantization for High Fidelity Audio Codec. https://arxiv. org/abs/2305.02765, 2023
2023 arXiv
-
[16]
WavTokenizer: an Efficient Acoustic Discrete Codec Tokenizer for Audio Language Modeling
Ji et al. WavTokenizer: an Efficient Acoustic Discrete Codec Tokenizer for Audio Language Modeling. https://arxiv.org/abs/2408.16532, 2025. ICLR
2025 arXiv
-
[17]
BigCodec: Pushing the Limits of Low-Bitrate Neural Speech Codec
Xin et al. BigCodec: Pushing the Limits of Low-Bitrate Neural Speech Codec. https://arxiv.org/abs/ 2409.05377, 2024
2024 arXiv
-
[18]
Finite Scalar Quantization: VQ-V AE Made Simple
Mentzer et al. Finite Scalar Quantization: VQ-V AE Made Simple. https://arxiv.org/abs/2309.15505,
-
[19]
SoundStorm: Efficient Parallel Audio Generation
Borsos et al. SoundStorm: Efficient Parallel Audio Generation. https://arxiv.org/abs/2305.09636, 2023
2023 arXiv
-
[20]
Masked Audio Generation using a Single Non-Autoregressive Transformer
Ziv et al. Masked Audio Generation using a Single Non-Autoregressive Transformer. https://arxiv.org/ abs/2401.04577, 2024. (MAGNeT), ICLR
2024 arXiv
-
[21]
w2v-BERT: Combining Contrastive Learning and Masked Language Modeling for Self-Supervised Speech Pre-Training.https://arxiv.org/abs/2108.06209, 2021
Chung et al. w2v-BERT: Combining Contrastive Learning and Masked Language Modeling for Self-Supervised Speech Pre-Training.https://arxiv.org/abs/2108.06209, 2021. ASRU
2021 arXiv
-
[22]
SpeechTokenizer: Unified Speech Tokenizer for Speech Large Language Models
Zhang et al. SpeechTokenizer: Unified Speech Tokenizer for Speech Large Language Models. https: //arxiv.org/abs/2308.16692, 2024. ICLR
2024 arXiv
-
[23]
SoCodec: A Semantic-Ordered Multi-Stream Speech Codec for Efficient Language Model Based TTS.https://arxiv.org/abs/2409.00933, 2024
Guo et al. SoCodec: A Semantic-Ordered Multi-Stream Speech Codec for Efficient Language Model Based TTS.https://arxiv.org/abs/2409.00933, 2024. SLT 2024
2024 arXiv
-
[24]
SemantiCodec: An Ultra Low Bitrate Semantic Audio Codec for General Sound
Liu et al. SemantiCodec: An Ultra Low Bitrate Semantic Audio Codec for General Sound. https://arxiv. org/abs/2405.00233, 2024
2024 arXiv
-
[25]
Codec Does Matter: Exploring the Semantic Shortcoming of Codec for Audio Language Model
Ye et al. Codec Does Matter: Exploring the Semantic Shortcoming of Codec for Audio Language Model. https://arxiv.org/abs/2408.17175, 2025. (X-Codec), AAAI
2025 arXiv
-
[26]
HuBERT: Self-Supervised Speech Representation Learning by Masked Prediction of Hidden Units
Hsu et al. HuBERT: Self-Supervised Speech Representation Learning by Masked Prediction of Hidden Units. https://arxiv.org/abs/2106.07447, 2021
2021 arXiv
-
[27]
WavLM: Large-Scale Self-Supervised Pre-Training for Full Stack Speech Processing
Chen et al. WavLM: Large-Scale Self-Supervised Pre-Training for Full Stack Speech Processing. https: //arxiv.org/abs/2110.13900, 2021
2021
-
[28]
MOSS-Audio-Tokenizer: Scaling Audio Tokenizers for Future Audio Foundation Models
Gong et al. MOSS-Audio-Tokenizer: Scaling Audio Tokenizers for Future Audio Foundation Models. https://arxiv.org/abs/2602.10934, 2026
2026
-
[29]
ALMTokenizer: A Low-bitrate and Semantic-rich Audio Codec Tokenizer for Audio Language Modeling.https://arxiv.org/abs/2504.10344, 2025
Yang et al. ALMTokenizer: A Low-bitrate and Semantic-rich Audio Codec Tokenizer for Audio Language Modeling.https://arxiv.org/abs/2504.10344, 2025. ICML
2025 arXiv
-
[30]
Fewer-token Neural Speech Codec with Time-invariant Codes
Ren et al. Fewer-token Neural Speech Codec with Time-invariant Codes. https://arxiv.org/abs/2310. 00014, 2024. (TiCodec), ICASSP
2024
-
[31]
Learning Source Disentanglement in Neural Audio Codec.https://arxiv.org/abs/2409.11228,
Bie et al. Learning Source Disentanglement in Neural Audio Codec.https://arxiv.org/abs/2409.11228,
-
[32]
NaturalSpeech 3: Zero-Shot Speech Synthesis with Factorized Codec and Diffusion Models
Ju et al. NaturalSpeech 3: Zero-Shot Speech Synthesis with Factorized Codec and Diffusion Models. https: //arxiv.org/abs/2403.03100, 2024. (FACodec), ICML. 19
2024 arXiv
-
[33]
CosyV oice: A Scalable Multilingual Zero-shot TTS Synthesizer based on Supervised Semantic Tokens.https://arxiv.org/abs/2407.05407, 2024
Du et al. CosyV oice: A Scalable Multilingual Zero-shot TTS Synthesizer based on Supervised Semantic Tokens.https://arxiv.org/abs/2407.05407, 2024
2024 arXiv
-
[34]
Seed-TTS: A Family of High-Quality Versatile Speech Generation Models
Anastassiou et al. Seed-TTS: A Family of High-Quality Versatile Speech Generation Models. https: //arxiv.org/abs/2406.02430, 2024
2024 arXiv
-
[35]
Stable Audio Open.https://arxiv.org/abs/2407.14358, 2024
Evans et al. Stable Audio Open.https://arxiv.org/abs/2407.14358, 2024
2024 arXiv
-
[36]
NaturalSpeech 2: Latent Diffusion Models are Natural and Zero-Shot Speech and Singing Synthesizers.arXiv preprint arXiv:2304.09116, 2023
Kai Shen, Zeqian Ju, Xu Tan, Yanqing Liu, Yichong Leng, Lei He, Tao Qin, Sheng Zhao, and Jiang Bian. NaturalSpeech 2: Latent Diffusion Models are Natural and Zero-Shot Speech and Singing Synthesizers.arXiv preprint arXiv:2304.09116, 2023
2023 arXiv
-
[37]
Make-An-Audio: Text-To-Audio Generation with Prompt-Enhanced Diffusion Models
Huang et al. Make-An-Audio: Text-To-Audio Generation with Prompt-Enhanced Diffusion Models. https: //arxiv.org/abs/2301.12661, 2023. ICML
2023 arXiv
-
[38]
DashengTokenizer: One layer is enough for unified audio understanding and generation
Dinkel et al. DashengTokenizer: One layer is enough for unified audio understanding and generation. https://arxiv.org/abs/2602.23765, 2026
2026
-
[39]
WavCube: Unifying Speech Representation for Understanding and Generation via Semantic- Acoustic Joint Modeling.https://arxiv.org/abs/2605.06407, 2026
Yang et al. WavCube: Unifying Speech Representation for Understanding and Generation via Semantic- Acoustic Joint Modeling.https://arxiv.org/abs/2605.06407, 2026
2026 arXiv
-
[40]
Ming-UniAudio: Speech LLM for Joint Understanding, Generation and Editing with Unified Representation
Yan and others (Inclusion AI). Ming-UniAudio: Speech LLM for Joint Understanding, Generation and Editing with Unified Representation. https://arxiv.org/abs/2511.05516, 2025. (MingTok-Audio continuous tokenizer)
2025
-
[41]
LoSATok: Low-dimensional Semantic-Acoustic Tokenizer for Cross-Domain Audio Understanding and Generation.https://arxiv.org/abs/2605.27840, 2026
Zhang et al. LoSATok: Low-dimensional Semantic-Acoustic Tokenizer for Cross-Domain Audio Understanding and Generation.https://arxiv.org/abs/2605.27840, 2026
2026 arXiv
-
[42]
Yaron Lipman, Ricky T. Q. Chen, Heli Ben-Hamu, Maximilian Nickel, and Matt Le. Flow Matching for Generative Modeling.https://arxiv.org/abs/2210.02747, 2023. ICLR
2023 arXiv
-
[43]
V oicebox: Text-Guided Multilingual Universal Speech Generation at Scale.https://arxiv.org/ abs/2306.15687, 2023
Le et al. V oicebox: Text-Guided Multilingual Universal Speech Generation at Scale.https://arxiv.org/ abs/2306.15687, 2023. NeurIPS
2023 arXiv
-
[44]
GIVT: Generative Infinite-V ocabulary Transformers
Tschannen et al. GIVT: Generative Infinite-V ocabulary Transformers. https://arxiv.org/abs/2312. 02116, 2024. ECCV
2024
-
[45]
Autoregressive Image Generation without Vector Quantization
Li et al. Autoregressive Image Generation without Vector Quantization. https://arxiv.org/abs/2406. 11838, 2024. (MAR), NeurIPS
2024
-
[46]
Hyperspherical Latents Improve Continuous-Token Autoregressive Generation
Guolin Ke and Hui Xue. Hyperspherical Latents Improve Continuous-Token Autoregressive Generation. https://arxiv.org/abs/2509.24335, 2025
2025
-
[47]
MaskGIT: Masked Generative Image Transformer
Chang et al. MaskGIT: Masked Generative Image Transformer. https://arxiv.org/abs/2202.04200,
-
[48]
Denoising Diffusion Probabilistic Models.Advances in Neural Information Processing Systems, 33:6840–6851, 2020
Jonathan Ho, Ajay Jain, and Pieter Abbeel. Denoising Diffusion Probabilistic Models.Advances in Neural Information Processing Systems, 33:6840–6851, 2020
2020
-
[49]
On Generative Spoken Language Modeling from Raw Audio
Lakhotia et al. On Generative Spoken Language Modeling from Raw Audio. https://arxiv.org/abs/ 2102.01192, 2021. TACL
2021 arXiv
-
[50]
Textually Pretrained Speech Language Models
Hassid et al. Textually Pretrained Speech Language Models. https://arxiv.org/abs/2305.13009, 2023. NeurIPS 2023
2023 arXiv
-
[51]
MusicLM: Generating Music From Text.https://arxiv.org/abs/2301.11325, 2023
Agostinelli et al. MusicLM: Generating Music From Text.https://arxiv.org/abs/2301.11325, 2023
2023 arXiv
-
[52]
Diffsound: Discrete Diffusion Model for Text-to-sound Generation
Yang et al. Diffsound: Discrete Diffusion Model for Text-to-sound Generation. https://arxiv.org/abs/ 2207.09983, 2023. IEEE/ACM TASLP. 20
2023 arXiv
-
[53]
InstructTTS: Modelling Expressive TTS in Discrete Latent Space with Natural Language Style Prompt.https://arxiv.org/abs/2301.13662, 2024
Yang et al. InstructTTS: Modelling Expressive TTS in Discrete Latent Space with Natural Language Style Prompt.https://arxiv.org/abs/2301.13662, 2024. IEEE/ACM TASLP
2024 arXiv
-
[54]
E2 TTS: Embarrassingly Easy Fully Non-Autoregressive Zero-Shot TTS
Eskimez et al. E2 TTS: Embarrassingly Easy Fully Non-Autoregressive Zero-Shot TTS. https://arxiv. org/abs/2406.18009, 2024
2024 arXiv
-
[55]
F5-TTS: A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching
Chen et al. F5-TTS: A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching. https: //arxiv.org/abs/2410.06885, 2025. ACL
2025 arXiv
-
[56]
Autoregressive Speech Synthesis without Vector Quantization
Meng et al. Autoregressive Speech Synthesis without Vector Quantization. https://arxiv.org/abs/2407. 08551, 2025. (MELLE), ACL
2025
-
[57]
VibeV oice Technical Report.https://arxiv.org/abs/2508.19205, 2025
Microsoft. VibeV oice Technical Report.https://arxiv.org/abs/2508.19205, 2025. Microsoft
2025 arXiv
-
[58]
DASB — Discrete Audio and Speech Benchmark
Mousavi et al. DASB — Discrete Audio and Speech Benchmark. https://arxiv.org/abs/2406.14294, 2024
2024 arXiv
-
[59]
TADA: A Generative Framework for Speech Modeling via Text-Acoustic Dual Alignment
Dang et al. TADA: A Generative Framework for Speech Modeling via Text-Acoustic Dual Alignment. https://arxiv.org/abs/2602.23068, 2026
2026
-
[60]
FlexiCodec: A Dynamic Neural Audio Codec for Low Frame Rates
Li et al. FlexiCodec: A Dynamic Neural Audio Codec for Low Frame Rates. https://arxiv.org/abs/ 2510.00981, 2026. ICLR
2026
-
[61]
Signal Estimation from Modified Short-Time Fourier Transform, 1984
Griffin and Lim. Signal Estimation from Modified Short-Time Fourier Transform, 1984. IEEE TASSP
1984
-
[62]
WaveNet: A Generative Model for Raw Audio
van den Oord et al. WaveNet: A Generative Model for Raw Audio. https://arxiv.org/abs/1609.03499, 2016
2016 arXiv
-
[63]
HiFi-GAN: GANs for Efficient and High Fidelity Speech Synthesis
Kong et al. HiFi-GAN: GANs for Efficient and High Fidelity Speech Synthesis. https://arxiv.org/abs/ 2010.05646, 2020. NeurIPS
2010 arXiv
-
[64]
Natural TTS Synthesis by Conditioning WaveNet on Mel Spectrogram Predictions
Shen et al. Natural TTS Synthesis by Conditioning WaveNet on Mel Spectrogram Predictions. https: //arxiv.org/abs/1712.05884, 2018. (Tacotron 2), ICASSP
2018 arXiv
-
[65]
Conditional Variational Autoencoder with Adversarial Learning for End-to-End Text-to-Speech
Kim et al. Conditional Variational Autoencoder with Adversarial Learning for End-to-End Text-to-Speech. https://arxiv.org/abs/2106.06103, 2021. (VITS), ICML
2021 arXiv
-
[66]
A Scale for the Measurement of the Psychological Magnitude Pitch
Stevens, V olkmann, and Newman. A Scale for the Measurement of the Psychological Magnitude Pitch. https://doi.org/10.1121/1.1915893, 1937. (the mel scale), J. Acoust. Soc. Am
1937 doi
-
[67]
Comparison of Parametric Representations for Monosyllabic Word Recognition in Continuously Spoken Sentences
Davis and Mermelstein. Comparison of Parametric Representations for Monosyllabic Word Recognition in Continuously Spoken Sentences. https://doi.org/10.1109/TASSP.1980.1163420, 1980. (MFCC), IEEE TASSP
1980
-
[68]
WaveGrad: Estimating Gradients for Waveform Generation
Chen et al. WaveGrad: Estimating Gradients for Waveform Generation. https://arxiv.org/abs/2009. 00713, 2020. ICLR 2021
2009
-
[69]
DiffWave: A Versatile Diffusion Model for Audio Synthesis
Kong et al. DiffWave: A Versatile Diffusion Model for Audio Synthesis. https://arxiv.org/abs/2009. 09761, 2020. ICLR 2021
2009
-
[70]
dMel: Speech Tokenization made Simple.https://arxiv.org/abs/2407.15835, 2024
Bai et al. dMel: Speech Tokenization made Simple.https://arxiv.org/abs/2407.15835, 2024
2024 arXiv
-
[71]
Grad-TTS: A Diffusion Probabilistic Model for Text-to-Speech
Popov et al. Grad-TTS: A Diffusion Probabilistic Model for Text-to-Speech. https://arxiv.org/abs/ 2105.06337, 2021. ICML 2021
2021 arXiv
-
[72]
DiffSinger: Singing V oice Synthesis via Shallow Diffusion Mechanism
Liu et al. DiffSinger: Singing V oice Synthesis via Shallow Diffusion Mechanism. https://arxiv.org/ abs/2105.02446, 2021. AAAI 2022
2021 arXiv
-
[73]
Sound Texture Perception via Statistics of the Auditory Periphery: Evidence from Sound Synthesis.https://doi.org/10.1016/j.neuron.2011.06.032, 2011
McDermott and Simoncelli. Sound Texture Perception via Statistics of the Auditory Periphery: Evidence from Sound Synthesis.https://doi.org/10.1016/j.neuron.2011.06.032, 2011. Neuron. 21
2011 doi
-
[74]
Neural Discrete Representation Learning
van den Oord et al. Neural Discrete Representation Learning. https://arxiv.org/abs/1711.00937, 2017. (VQ-V AE), NeurIPS
2017 arXiv
-
[75]
SNAC: Multi-Scale Neural Audio Codec.https://arxiv.org/abs/2410.14411, 2024
Siuzdak et al. SNAC: Multi-Scale Neural Audio Codec.https://arxiv.org/abs/2410.14411, 2024
2024 arXiv
-
[76]
An Image is Worth 32 Tokens for Reconstruction and Generation
Yu et al. An Image is Worth 32 Tokens for Reconstruction and Generation. https://arxiv.org/abs/2406. 07550, 2024. (TiTok), NeurIPS
2024
-
[77]
FlexTok: Resampling Images into 1D Token Sequences of Flexible Length
Bachmann et al. FlexTok: Resampling Images into 1D Token Sequences of Flexible Length. https: //arxiv.org/abs/2502.13967, 2025. ICML
2025 arXiv
-
[78]
Variable-rate discrete representation learning
Dieleman et al. Variable-rate discrete representation learning. https://arxiv.org/abs/2103.06089, 2021. 22 Appendix A Why Audio Needs a Latent A.1 Mel as the first widely used latent Generating audio in a latent space is older than learned codecs. The mel-spectrogram already a...
2021 arXiv
Reviewed August 14, 2026 · model on record in the stance chip above.
Discussion (0). Continue with ORCID to comment.