REVIEW 3 major objections 5 minor 59 references
ReverbMiipher: Generative Speech Restoration meets Reverberation Characteristics Controllability
T0 review · 3 major / 5 minor · reviewed 2026-08-15 · deepseek-v4-flash
Pith's one-line read The paper claims that a generative speech-restoration model can remove noise and codec artifacts while preserving the original reverberation, and that it does so by routing a dedicated reverb feature through the vocoder, outperforming the…
desk verdict ReverbMiipher is a solid incremental extension of Miipher-2 that beats a two-stage RIR baseline at preserving reverberation, but the claimed disentanglement of the reverb feature is asserted, not demonstrated. 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 central object is the 512-dimensional reverb feature c = R(x) produced by the ReverbEncoder, a stack of four Conformer layers with kernel size five followed by time-axis mean pooling, operating on the same 128-band log-mel input as the feature cleaner. This vector conditions the WaveFit vocoder as V(ĥ_s, c), and the training loss of Eq. (1) switches stochastically with probability q = 0.1 between the reverberant target s*r (when c is the true encoder output) and the anechoic target s (when c is a zero vector). The switching is what forces the encoder to specialize: the vocoder must learn to use c to reproduce the room, while the zero-vector case tells it that no reverb is also a valid condition, so any non-reverb information in c would have to be ignored or would corrupt the anechoic output. That mechanism is what separates the reverb representation from the speech content and makes downstream interpolation and sampling meaningful.
What would settle it
Take a fixed clean anechoic utterance s, extract reverb features c_i from a set of noisy utterances with different speakers and content, and synthesize V(ĥ_s, c_i). If the reverb feature is truly disentangled, the output should have the same ASR transcript and near-identical speaker embedding as s for every c_i, with only the room acoustics changing; a systematic drop in speaker similarity or rise in word error rate as c_i varies would falsify the disentanglement claim.
Extended reading notes
Core claim
ReverbMiipher's central claim is that reverberation can be treated as an explicit conditioning variable inside a generative speech-restoration model rather than something to remove or add afterwards. The ReverbEncoder's feature vector c is the carrier of the room acoustics: when c is present, the vocoder is trained to output s*r, the clean speech convolved with the original room impulse response, and when c is the zero vector, it must output the anechoic source s. Because the model receives only the noisy input, the encoder must pull the reverb information out of the degraded mixture, and the random switching with probability q = 0.1 is the training signal that prevents c from smuggling in speaker identity, content, or noise. The paper then shows that c varies smoothly with RT60 and DRR in a PCA projection, that linear interpolation between two c vectors produces intermediate reverberation, and that sampling from the PCA plane gives new but structurally intact reverberation effects. The claim is that this one-stage, feature-conditioned design outperforms the two-stage baseline of Miipher-2 restoration followed by convolution with a simulated RIR chosen with known RT60 and DRR.
Load-bearing premise
The whole control scheme rests on the assumption that training with a 10% zero-vector switching probability makes the reverb feature encode only reverberation; if that vector also leaks speaker identity or content, the demonstrated interpolation and sampling are changing the speech itself rather than just the room.
Editorial extensions
If this is right
- Reverb preservation becomes a built-in behavior of speech restoration rather than a post-hoc step, so the acoustic environment of the original recording is retained in applications such as augmented reality, virtual reality, and video generation.
- The reverb feature can be transferred between utterances, interpolated, or sampled from a PCA plane, enabling the creation of datasets with controllable spatial characteristics.
- Because the model is trained end-to-end rather than relying on explicit RIR estimation, it avoids the error propagation and fidelity losses of the two-stage clean-then-convolve pipeline.
- The approach beats the simulated-RIR baseline even when that baseline is given perfect knowledge of RT60 and DRR, indicating that the advantage comes from the architecture itself rather than from having better room information.
Reading between the lines
- If the disentanglement holds, the same conditioning trick could be extended to other acoustic attributes, such as environmental sounds, effectively turning speech restoration into a controllable acoustic-scene remixer.
- A testable extension is cross-speaker or cross-language reverb transfer: swapping the reverb feature between utterances with different speakers or content should leave the transcript and speaker identity untouched, which can be verified with ASR and speaker-embedding similarity.
- The smooth PCA-plane structure suggests the reverb feature manifold may be low-dimensional; an explicit dimensionality analysis or a learned controllable latent could give even finer control than linear interpolation.
- The subjective evaluation uses pairwise ranking focused on reverberation similarity, so a future test that also measures overall perceived quality would clarify whether reverb preservation comes at any cost to other perceptual dimensions.
Signed reviews
Editorial analysis
A structured set of objections, weighed in public.
Referee Report
Summary. The paper proposes ReverbMiipher, an extension of the Miipher-2 speech restoration model that adds a ReverbEncoder producing a 512-dimensional utterance-level feature c from the noisy log-mel input. The vocoder is conditioned on c, and training uses stochastic zero-vector replacement so that setting c to the zero vector yields clean speech while using the extracted c yields speech with reverberation. Experiments compare ReverbMiipher against Miipher-2 and a two-stage baseline (Miipher-2 followed by convolution with an oracle-parameter simulated RIR) using pairwise subjective ranking and objective MCD/GPE/SPK-sim against the reverberant reference. The paper also presents PCA visualization, linear interpolation of c, and sampling from the PCA plane as reverberation control applications.
Significance. If the claimed disentanglement holds, ReverbMiipher is a meaningful step toward speech restoration that preserves and controls the acoustic environment rather than removing it. The strongest evidence is the subjective ranking (mean rank 1.49 versus 2.01 and 2.50, p < 1e-8) and the improvement in MCD and SPK-sim over an upper-bound two-stage baseline that is given oracle RT60 and DRR values. The proposed use of a learned reverb feature with interpolation and PCA sampling is a practical and potentially generalizable interface. However, the paper's central controllability claim rests on an untested disentanglement assumption, and the artifact-removal claim lacks direct objective metrics.
major comments (3)
- [§3.2, Eq. (1)] The stochastic zero-vector replacement does not enforce the claimed disentanglement. In the branch U(0,1) <= q, the loss L_V(s, V(h_s, 0)) updates only the vocoder, not the ReverbEncoder, since c is fixed to 0. In the other branch, the ReverbEncoder is trained solely to minimize L_V(s*r, V(h_s, c)); any information in c that helps reconstruct the reverberant target, including speaker identity, content, or residual noise, is useful for this loss. The statement that artifacts in c 'will be treated as noise by the network' (§3.2) is an assertion without gradient-level justification. Since h_s is an imperfect estimate, c has a strong incentive to carry content and speaker information as a shortcut. Consequently, the interpolation and sampling demonstrations in §5 may change speaker or content rather than reverberation only. The authors should provide a quantitative disentanglement test, e.g., swap or interpolate c between utterances with different speakers or content and measure speaker similarity and word error rate of the output, or add an explicit information-bottleneck or adversarial objective.
- [§4.4, Table 2] The abstract and §1 claim that ReverbMiipher 'removes other artifacts' such as noise and codec distortions, but Table 2 reports only MCD, GPE, and SPK-sim computed against the reverberant reference s*r. These metrics do not directly measure artifact removal (e.g., noise audibility, codec distortion) and are also consistent with c leaking content or speaker information, since such leakage would help match s*r. I recommend adding standard restoration metrics such as PESQ, STOI, or DNSMOS against the clean reference (or a no-reference metric), and possibly a subjective quality MOS, so that the artifact-removal claim is supported independently of the reverberation-preservation claim.
- [§5.2–§5.3, Figs. 4–5] The controllability experiments are qualitative only. Fig. 4b and Fig. 5 show spectrograms and state that harmonic structure or original speech structure is preserved, but no objective measurement of content or speaker consistency is reported. Given the disentanglement concern in §3.2, these demonstrations do not establish that varying alpha or sampling c controls only reverberation. I suggest measuring, across the interpolation path alpha in [0,1] and across sampled points, word error rate (or phoneme error rate) and speaker embedding cosine similarity against the source utterance, as well as an objective reverb-parameter estimate (e.g., RT60 and DRR) of the output.
minor comments (5)
- [§4.2, §6] There are minor typos: 'V orbis' should be 'Vorbis' in §4.2, and 'better than to the baseline' should be 'better than the baseline' in §6.
- [Table 2] The objective results are reported without error bars or confidence intervals; with 1070 evaluation utterances, pairwise significance tests would strengthen the objective comparison.
- [Fig. 3] Clarify the treatment of anechoic samples, since RT60 is nominally 0 and DRR is undefined for an anechoic condition; the text says anechoic samples cluster at upper right, which is confusing given the axis ranges.
- [§5.2] Typo: 'uttrances' should be 'utterances', and 'Let c1 and c2 be a reverb features' should be 'Let c1 and c2 be reverb features'.
- [§3.2] The switching probability q is not ablated; a brief sensitivity study or a justification for q=0.1 would help, since q controls the tradeoff between preservation and controllability.
Circularity Check
No significant circularity: ReverbMiipher's central claims rest on external supervised targets and held-out benchmarks.
full rationale
The paper's core derivation chain is supervised learning against external ground-truth signals, not a self-referential prediction. Equation (1) trains ReverbMiipher with targets s*r and s, both defined independently of the model, and the objective evaluations in Table 2 compare model outputs to the held-out reverberant reference s*r; the subjective evaluation in Table 1 compares listener rankings against the same external reference. The reverb feature c is a learned representation, not a fitted parameter renamed as a prediction; the interpolation and PCA-sampling demonstrations in Section 5 are empirical case studies rather than derived predictions. The paper cites prior work by overlapping authors ([26] Miipher-2, [45] WaveFit, [48] FLEURS-R) as building blocks, but these are pre-existing components with their own external training objectives and are not invoked as unverified uniqueness theorems or ansatz-smuggling citations. The disentanglement of c is asserted via the stochastic zero-vector strategy rather than rigorously demonstrated, which is a validity or robustness concern, but it is not a circular reduction: no equation in the paper is equivalent to its own input by construction, and no fitted parameter is passed off as a prediction. Therefore the appropriate circularity score is 0.
Assumptions & free parameters
free parameters (3)
- switching probability q =
0.1
- reverb feature dimension D =
512
- number of ReverbEncoder Conformer layers =
4
assumptions (4)
- domain assumption The clean SSL feature hs predicted by the feature cleaner is anechoic and contains all information needed to reconstruct the clean speech s.
- domain assumption A single 512-dimensional time-pooled vector can encode the relevant reverberation characteristics of an RIR from a noisy log-mel spectrogram.
- ad hoc to paper Gradient updates with stochastic zero-vector replacement converge to a solution where c carries only reverberation and the vocoder ignores any leakage.
- domain assumption Image-method-simulated RIRs used in training are representative enough of real RIRs from the MIT survey at test time.
Cite this review
Pith. "Pith review of ReverbMiipher: Generative Speech Restoration meets Reverberation Characteristics Controllability." pith.science (2026). https://pith.science/paper/BQO7R24T
@misc{pith2026250505077,
author = {Pith},
title = {Pith review of: ReverbMiipher: Generative Speech Restoration meets Reverberation Characteristics Controllability},
year = {2026},
howpublished = {\url{https://pith.science/paper/BQO7R24T}},
note = {Machine review of arXiv:2505.05077}
}
read the original abstract
Reverberation encodes spatial information regarding the acoustic source environment, yet traditional Speech Restoration (SR) usually completely removes reverberation. We propose ReverbMiipher, an SR model extending parametric resynthesis framework, designed to denoise speech while preserving and enabling control over reverberation. ReverbMiipher incorporates a dedicated ReverbEncoder to extract a reverb feature vector from noisy input. This feature conditions a vocoder to reconstruct the speech signal, removing noise while retaining the original reverberation characteristics. A stochastic zero-vector replacement strategy during training ensures the feature specifically encodes reverberation, disentangling it from other speech attributes. This learned representation facilitates reverberation control via techniques such as interpolation between features, replacement with features from other utterances, or sampling from a latent space. Objective and subjective evaluations confirm ReverbMiipher effectively preserves reverberation, removes other artifacts, and outperforms the conventional two-stage SR and convolving simulated room impulse response approach. We further demonstrate its ability to generate novel reverberation effects through feature manipulation.
Figures
Reference graph
Works this paper leans on
- [1]
-
[2]
W. Bailey and B. M. Fazenda, “The effect of reverberation and audio spatialization on egocentric distance estimation of objects in stereoscopic virtual reality,” J. Acoust. Soc. Am. , 2017
work page 2017
-
[3]
D. R. Begault, E. M. Wenzel, and M. R. Anderson, “Direct comparison of the impact of head tracking, reverberation, and individualized head- related transfer functions on the spatial perception of a virtual speech source,” J. AES, 2001
work page 2001
-
[4]
Auditory-induced presence in mixed reality environments and related technology,
P. Larsson, A. V ¨aljam¨ae, et al. , “Auditory-induced presence in mixed reality environments and related technology,” The engineering of mixed reality systems, 2010
work page 2010
-
[5]
High-resolution image synthesis with latent diffusion models,
R. Rombach, A. Blattmann, et al., “High-resolution image synthesis with latent diffusion models,” in CVPR, 2022
work page 2022
-
[6]
AudioLM: a language modeling approach to audio generation,
Z. Borsos, R. Marinier, et al., “AudioLM: a language modeling approach to audio generation,” IEEE/ACM Trans. Audio, Speech and Lang. Proc. , 2023
work page 2023
-
[7]
P. A. Naylor and N. D. Gaubitch, Speech Dereverberation. Springer, 2010
work page 2010
-
[8]
Speech dereverberation based on variance-normalized delayed linear prediction,
T. Nakatani, T. Yoshioka, et al. , “Speech dereverberation based on variance-normalized delayed linear prediction,” IEEE/ACM Trans. Audio, Speech and Lang. Proc. , 2010
work page 2010
Show all 59 references
-
[9]
The REVERB challenge: A common evaluation framework for dereverberation and recognition of reverberant speech,
K. Kinoshita, M. Delcroix, et al. , “The REVERB challenge: A common evaluation framework for dereverberation and recognition of reverberant speech,” in WASPAA, 2013
2013
-
[10]
Learning spectral mapping for speech dereverberation,
K. Han, Y . Wang, and D. Wang, “Learning spectral mapping for speech dereverberation,” in ICASSP, 2014
2014
-
[11]
Neural network-based spectrum estimation for online wpe dereverberation
K. Kinoshita, M. Delcroix, et al. , “Neural network-based spectrum estimation for online wpe dereverberation.” in Interspeech, 2017
2017
-
[12]
Fundamentals, present and future perspectives of speech enhancement,
N. Das, S. Chakraborty, et al. , “Fundamentals, present and future perspectives of speech enhancement,” Int. J. of Speech Tech. , 2021
2021
-
[13]
Parametric resynthesis with neural vocoders,
S. Maiti and M. I. Mandel, “Parametric resynthesis with neural vocoders,” in WASPAA, 2019
2019
-
[14]
SelfRemaster: Self-supervised speech restoration with analysis-by-synthesis approach using channel modeling,
T. Saeki, S. Takamichi, et al. , “SelfRemaster: Self-supervised speech restoration with analysis-by-synthesis approach using channel modeling,” in Interspeech, 2022
2022
-
[15]
HiFi-GAN-2: Studio-quality speech enhancement via generative adversarial networks conditioned on acoustic features,
J. Su, Z. Jin, and A. Finkelstein, “HiFi-GAN-2: Studio-quality speech enhancement via generative adversarial networks conditioned on acoustic features,” in WASPAA, 2021
2021
-
[16]
V oiceFixer: A unified framework for high-fidelity speech restoration,
H. Liu, X. Liu, et al. , “V oiceFixer: A unified framework for high-fidelity speech restoration,” in Interspeech, 2022
2022
-
[17]
Universal speech enhancement with score- based diffusion,
J. Serr `a, S. Pascual, et al. , “Universal speech enhancement with score- based diffusion,” arXiv preprint arXiv:2206.03065 , 2022
2022 arXiv
-
[18]
Miipher: A robust speech restoration model integrating self-supervised speech and text representations,
Y . Koizumi, H. Zen, et al. , “Miipher: A robust speech restoration model integrating self-supervised speech and text representations,” in WASPAA, 2023
2023
-
[19]
Speech enhancement and dereverberation with diffusion-based generative models,
J. Richter, S. Welker, et al. , “Speech enhancement and dereverberation with diffusion-based generative models,” IEEE/ACM Trans. Audio, Speech and Lang. Proc. , 2023
2023
-
[20]
Diffusion models for audio restoration: A review,
J.-M. Lemercier, J. Richter, et al., “Diffusion models for audio restoration: A review,” IEEE Signal Process Mag. , 2024
2024
-
[21]
Universal score-based speech enhancement with high content preservation,
R. Scheibler, Y . Fujita,et al., “Universal score-based speech enhancement with high content preservation,” in Interspeech, 2024
2024
-
[22]
LLaSE-G1: Incentivizing generaliza- tion capability for llama-based speech enhancement,
B. Kang, X. Zhu, et al. , “LLaSE-G1: Incentivizing generaliza- tion capability for llama-based speech enhancement,” arXiv preprint arXiv:2503.00493, 2025
2025 arXiv
-
[23]
Genhancer: High-fidelity speech enhancement via generative modeling on discrete codec tokens,
H. Yang, J. Su, et al. , “Genhancer: High-fidelity speech enhancement via generative modeling on discrete codec tokens,” in Interspeech, 2024
2024
-
[24]
Joint semantic knowledge distillation and masked acoustic modeling for full-band speech restoration with improved intelligibility,
X. Liu, X. Li, et al. , “Joint semantic knowledge distillation and masked acoustic modeling for full-band speech restoration with improved intelligibility,” in ICASSP, 2025
2025
-
[25]
DiTSE: High-fidelity generative speech enhancement via latent diffusion transformers,
H. R. Guimar ˜aes, J. Su, et al. , “DiTSE: High-fidelity generative speech enhancement via latent diffusion transformers,” arXiv preprint arXiv:2504.09381, 2025
2025
-
[26]
Miipher-2: A universal speech restora- tion model for million-hour scale data restoration,
S. Karita, Y . Koizumi, et al. , “Miipher-2: A universal speech restora- tion model for million-hour scale data restoration,” arXiv preprint arXiv:2505.04457, 2025
2025 arXiv
-
[27]
Photorealistic video generation with diffusion models,
A. Gupta, L. Yu, et al. , “Photorealistic video generation with diffusion models,” in ECCV, 2023
2023
-
[28]
Stable video diffusion: Scal- ing latent video diffusion models to large datasets,
A. Blattmann, T. Dockhorn, et al. , “Stable video diffusion: Scal- ing latent video diffusion models to large datasets,” arXiv preprint arXiv:2311.15127, 2023
2023 arXiv
-
[29]
Image method for efficiently simulating small-room acoustics,
J. B. Allen and D. A. Berkley, “Image method for efficiently simulating small-room acoustics,” J. Acoust. Soc. Am. , 1979
1979
-
[30]
Simulating room transfer functions between transducers mounted on audio devices using a modified image source method,
Z. Xu, A. Herzog, et al. , “Simulating room transfer functions between transducers mounted on audio devices using a modified image source method,” The J. of the Acoust. Soc. Am. , 2024
2024
-
[31]
Fdtd methods for 3-d room acoustics simulation with high-order accuracy in space and time,
B. Hamilton and S. Bilbao, “Fdtd methods for 3-d room acoustics simulation with high-order accuracy in space and time,” IEEE/ACM Trans. Audio, Speech and Lang. Proc. , 2017
2017
-
[32]
Learning neural acoustic fields,
A. Luo, Y . Du, et al. , “Learning neural acoustic fields,” in NeurIPS, 2022
2022
-
[33]
IR-GAN: Room Impulse Response Generator for Far-Field Speech Recognition,
A. Ratnarajah, Z. Tang, and D. Manocha, “IR-GAN: Room Impulse Response Generator for Far-Field Speech Recognition,” in Interspeech, 2021
2021
-
[34]
Filtered noise shaping for time domain room impulse response estimation from reverberant speech,
C. J. Steinmetz, V . K. Ithapu, and P. Calamia, “Filtered noise shaping for time domain room impulse response estimation from reverberant speech,” in WASPAA, 2021
2021
-
[35]
Reverberation-controllable voice conver- sion using reverberation time estimator,
Y . Choi, C. Xie, and T. Toda, “Reverberation-controllable voice conver- sion using reverberation time estimator,” in Interspeech, 2023
2023
-
[36]
Multi-source spatial knowledge understanding for immersive visual text-to-speech,
S. He and R. Liu, “Multi-source spatial knowledge understanding for immersive visual text-to-speech,” in ICASSP, 2025
2025
-
[37]
Supervised speech separation based on deep learning: An overview,
D. Wang and J. Chen, “Supervised speech separation based on deep learning: An overview,” IEEE/ACM Trans. Audio, Speech and Lang. Proc., 2018
2018
-
[38]
A survey and an extensive evaluation of popular audio declipping methods,
P. Z´aviˇska, P. Rajmic, et al. , “A survey and an extensive evaluation of popular audio declipping methods,” IEEE Journal of Selected Topics in Signal Processing, 2020
2020
-
[39]
A deep neural network approach to speech bandwidth expansion,
K. Li and C.-H. Lee, “A deep neural network approach to speech bandwidth expansion,” in ICASSP, 2015
2015
-
[40]
Audio super-resolution using neural nets,
V . Kuleshov, S. Z. Enam, and S. Ermon, “Audio super-resolution using neural nets,” in ICLR (Workshop Track), 2017
2017
-
[41]
Neurips 2024 competition proposal: Urgent challenge,
W. Zhang, R. Scheibler, et al. , “Neurips 2024 competition proposal: Urgent challenge,” in NeurIPS 2024 Competition Track
2024
-
[42]
Google USM: Scaling automatic speech recognition beyond 100 languages,
Y . Zhang, W. Han, et al. , “Google USM: Scaling automatic speech recognition beyond 100 languages,” 2023
2023
-
[43]
Towards a unified view of parameter-efficient transfer learning,
J. He, C. Zhou, et al. , “Towards a unified view of parameter-efficient transfer learning,” in ICLR, 2022
2022
-
[44]
Conformer: Convolution-augmented transformer for speech recognition,
A. Gulati, C.-C. Chiu, et al. , “Conformer: Convolution-augmented transformer for speech recognition,” in Interspeech, 2020
2020
-
[45]
WaveFit: An iterative and non- autoregressive neural vocoder based on fixed-point iteration,
Y . Koizumi, K. Yatabe, et al. , “WaveFit: An iterative and non- autoregressive neural vocoder based on fixed-point iteration,” in SLT, 2023
2023
-
[46]
Pyroomacoustics: A python package for audio room simulation and array processing algorithms,
R. Scheibler, E. Bezzam, and I. Dokmani ´c, “Pyroomacoustics: A python package for audio room simulation and array processing algorithms,” in ICASSP, 2018
2018
-
[47]
Single-channel blind direct-to- reverberation ratio estimation using masking
W. Mack, S. Deng, and E. A. Habets, “Single-channel blind direct-to- reverberation ratio estimation using masking.” in Interspeech, 2020
2020
-
[48]
FLEURS-R: A restored multilingual speech corpus for generation tasks,
M. Ma, Y . Koizumi, et al. , “FLEURS-R: A restored multilingual speech corpus for generation tasks,” in Interspeech, 2024
2024
-
[49]
EARS: An anechoic fullband speech dataset benchmarked for speech enhancement and dereverberation,
J. Richter, Y .-C. Wu, et al. , “EARS: An anechoic fullband speech dataset benchmarked for speech enhancement and dereverberation,” in Interspeech, 2024
2024
-
[50]
WHAM!: Extending speech separation to noisy environments,
G. Wichern, J. Antognini, et al. , “WHAM!: Extending speech separation to noisy environments,” in Interspeech, 2019
2019
-
[51]
Statistics of natural reverberation enable perceptual separation of sound and space,
J. Traer and J. H. McDermott, “Statistics of natural reverberation enable perceptual separation of sound and space,” Proceedings of the National Academy of Sciences , 2016
2016
-
[52]
Adam: A method for stochastic optimization
D. P. Kingma and J. Ba, “Adam: A method for stochastic optimization.” in ICLR, 2015
2015
-
[53]
W. J. Conover, Practical nonparametric statistics . john wiley & sons, 1999
1999
-
[54]
Multiple comparisons among means,
O. J. Dunn, “Multiple comparisons among means,” Journal of the American statistical association , 1961
1961
-
[55]
Mel-cepstral distance measure for objective speech quality assessment,
R. Kubichek, “Mel-cepstral distance measure for objective speech quality assessment,” in PACRIM, 1993
1993
-
[56]
Reducing f0 frame error of f0 tracking algorithms under noisy conditions with an unvoiced/voiced classification frontend,
W. Chu and A. Alwan, “Reducing f0 frame error of f0 tracking algorithms under noisy conditions with an unvoiced/voiced classification frontend,” in ICASSP, 2009
2009
-
[57]
Transfer learning from speaker verification to multispeaker text-to-speech synthesis,
Y . Jia, Y . Zhang,et al. , “Transfer learning from speaker verification to multispeaker text-to-speech synthesis,” in NeurIPS, 2018
2018
-
[58]
Sample efficient adaptive text-to-speech,
Y . Chen, Y . Assael,et al. , “Sample efficient adaptive text-to-speech,” in ICLR, 2019
2019
-
[59]
New method of measuring reverberation time,
M. R. Schroeder, “New method of measuring reverberation time,” The Journal of the Acoustical Society of America , 1965
1965
Reviewed August 15, 2026 · model on record in the stance chip above.
Discussion (0). Continue with ORCID to comment.