REVIEW 5 major objections 4 minor 48 references
Comparative Analysis of Fast and High-Fidelity Neural Vocoders for Low-Latency Streaming Synthesis in Resource-Constrained Environments
T0 review · 5 major / 4 minor · reviewed 2026-08-07 · deepseek-v4-flash
Pith's one-line read This paper claims that a multi-stream extension of the Wavehax vocoder delivers the best throughput among compared neural vocoders at sub-80 ms streaming latency on a single CPU, with near-non-causal quality at one frame of lookahead…
desk verdict MS-Wavehax is a genuinely compact, high-quality streaming vocoder, but the central throughput claim is partly confounded by an uneven ONNX STFT replacement across models. 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 mechanism is multi-stream synthesis with trainable filters combined with streaming adaptations. A signal analysis filter (length 63) decomposes the input waveform into four subscale signals; each subscale goes through STFT with 20 ms frames and a 10 ms shift, producing subscale spectrograms that are concatenated with projected mel-spectrogram features into a 12-channel representation. A series of 2D convolutional layers and residual blocks processes this representation, and a final pointwise convolution outputs eight spectrograms (real and imaginary parts of four subscale signals), which are converted back to the time domain via iSTFT and merged by a synthesis filter. Streaming feasibility comes from ring-buffer caching applied to all convolutional, STFT, and iSTFT layers, and from replacing layer normalization with batch normalization, since batch norm does not require computing statistics over the whole sequence at inference. The throughput analysis works through im2col matrix sizes: 2D convolutions produce input matrices that grow with chunk size T and the frequency dimension, increasing main-memory traffic at larger chunks, while 1D convolutions have large but constant filter matrices that are reused from cache.
What would settle it
Run the same streaming RTF comparison on a low-end mobile CPU (e.g., an ARM Cortex-A series core with small L1/L2 caches and no large shared L3) using the same ONNX Runtime pipeline and chunk sizes of 1, 4, 8, and 16 frames. If Vocos already matches or beats MS-Wavehax at chunk sizes below 8 frames, or if MS-Wavehax fails to keep the lowest RTF below 80 ms latency, then the paper's central 'highest throughput under low-latency' claim does not generalize beyond the tested EPYC configuration. A second check: switch the convolution backend from im2col-based GeMM to direct convolution on the same EPYC CPU and see whether the RTF crossover between MS-Wavehax and Vocos shifts or disappears, which would confirm or falsify the proposed memory-bandwidth mechanism.
Extended reading notes
Core claim
On the paper's own terms, the central discovery is that multi-stream decomposition of the harmonic-prior-based Wavehax architecture yields a vocoder that is both subjectively high-quality and uniquely suited to low-latency CPU streaming. MS-Wavehax splits the input waveform with an analysis filter into four subscale signals, converts each to a complex spectrogram, concatenates them with mel-spectrogram features, processes them with 2D convolutions, and recombines the streams via a synthesis filter. In listening tests for both analysis-synthesis and text-to-speech, non-causal MS-Wavehax and causal MS-Wavehax with one-frame lookahead score highest among the vocoders. With 0.332 million parameters and about 1.576 GMACs per second of audio in PyTorch, it achieves the best real-time factor at small chunk sizes and requires only 2.4% of HiFi-GAN V1's model size. The paper further shows that as chunk size grows, memory transfer in 2D convolutions becomes the bottleneck, which is why the small 2D-convolution-based MS-Wavehax wins at low latency while the 1D-convolution-based Vocos wins at large chunk sizes.
Load-bearing premise
The central throughput claim assumes that the im2col-based memory-bandwidth behavior measured on a single AMD EPYC 7302 CPU with ONNX Runtime generalizes to other resource-constrained devices; if a target device's cache hierarchy or convolution backend changes the balance between data transfer and compute, the reported throughput ranking and recommended chunk sizes could change.
Editorial extensions
If this is right
- For chunk sizes corresponding to less than 80 ms of additional latency, MS-Wavehax gives the lowest real-time factor among the tested vocoders in a single-thread CPU environment, making it suitable for interactive real-time applications.
- With one frame (10 ms) of lookahead, causal MS-Wavehax matches its non-causal quality in listening tests, indicating that near-zero-future-context streaming is feasible without audible degradation.
- The model size of MS-Wavehax is about 2.4% of HiFi-GAN V1's parameter count (0.332M versus 13.82M), so it can fit into on-device memory budgets where larger vocoders are impractical.
- The im2col memory-bandwidth analysis predicts that 2D-convolution vocoders lose throughput relative to 1D-convolution vocoders as chunk size grows, providing a concrete design rule for choosing streaming chunk sizes.
- Replacing layer normalization with batch normalization and adding caching does not harm quality and, in the paper's internal comparisons, even improves objective metrics.
- The relationship between latency and throughput depends on hardware cache hierarchy and memory bandwidth, so the optimal chunk size and vocoder architecture should be chosen according to the target device.
- The paper's analysis is performed on a single AMD EPYC 7302 CPU using ONNX Runtime, and the practical recommendations assume this behavior generalizes to other resource-constrained CPUs and inference backends.
- The multi-stream decomposition's benefit may extend beyond efficiency: partitioning the spectrogram into smaller sub-spectrograms makes spectral structures more locally consistent, which the paper argues improves how 2D convolutional kernels capture dependencies.
Editorial analysis
A structured set of objections, weighed in public.
Referee Report
Summary. The paper proposes multi-stream Wavehax (MS-Wavehax), a compact neural vocoder that extends Wavehax with four-stream subscale synthesis using trainable analysis/synthesis filters, and also introduces a multi-stream version of Vocos. The authors evaluate streaming synthesis on a single CPU with ONNX Runtime, reporting real-time factors (RTF) as a function of chunk size, and present a theoretical im2col-based analysis of why 1D ConvNeXt-based Vocos overtakes 2D Wavehax variants at larger chunk sizes. Subjective MOS evaluations in analysis-synthesis and text-to-speech tasks are reported for the baseline and multi-stream models, including causal MS-Wavehax with and without one-frame lookahead. The central claims are that MS-Wavehax offers high speech quality under causal and non-causal conditions, achieves the highest throughput at low latency (<80 ms), and requires only 2.4% of HiFi-GAN V1's parameter count.
Significance. If the central claims hold, this is a practically valuable result: a vocoder of 0.332M parameters that nearly matches non-causal quality with one frame of lookahead would be directly useful for on-device streaming TTS and voice conversion. The paper also provides a useful methodological discussion of how streaming overhead, parameter loading, and cache behavior interact with 1D versus 2D convolution design, and the im2col matrix-size analysis in Section 3.2 and Table 2 is a clear, concrete contribution. The experimental setup uses public corpora (JVS, JSUT), a standard MOS protocol with 30 listeners, and detailed hyperparameters, which are strengths. However, as detailed below, the throughput comparison is confounded by the ONNX STFT/iSTFT replacement, and the subjective comparison lacks causal baselines and statistical testing, so the main claims are not yet fully established.
major comments (5)
- [Section 3.2, Table 1, Figure 2] The central throughput claim is confounded by the ONNX STFT/iSTFT replacement. Table 1 shows that replacing torch.stft/torch.istft with convolution-based implementations changes per-second MACs from 1.348B to 46.96B for Vocos (a 35x increase), from 1.787B to 13.03B for Wavehax (a 7.3x increase), and from 1.576B to 2.291B for MS-Wavehax (a 1.45x increase). Since Figure 2 reports RTF measured with ONNX Runtime, the ranking at small chunk sizes may reflect the relative overhead of the conv-based STFT implementation rather than the multi-stream architecture or the 2D-versus-1D convolution behavior analyzed in the im2col discussion. The manuscript does not provide a per-layer RTF breakdown or an ablation that holds the STFT implementation fixed across all models. To support the claim that MS-Wavehax has the highest streaming throughput, the authors should either report RTF with a common STFT/iSTFT implementation (e.g., native PyTorch or a backend optimized for all models), or explicitly isolate and compare the STFT overhead per model.
- [Section 4.2, Figure 4] The MOS evaluation does not include causal baselines. The non-causal set includes HiFi-GAN, iSTFTNet, Vocos, and Wavehax, but the only causal models evaluated are MS-Wavehax with lookahead 0 and 1. Thus the claim that MS-Wavehax 'delivers high speech quality under causal conditions' cannot be compared against causal variants of any other vocoder. For example, a causal HiFi-GAN or causal Vocos would be a natural baseline to determine whether the quality gap under causality is specific to MS-Wavehax or a general property of causal streaming. The authors should add at least one strong causal baseline or soften the claim accordingly.
- [Section 4.2, Figure 4] No statistical significance testing is reported for the MOS results. The figure shows 95% confidence intervals, but with 30 listeners and 10 samples per model, overlapping intervals do not establish equivalent or superior quality. The manuscript states that MS-Wavehax 'achieved the highest MOS in both tasks' and that 'the performance gap between MS-Wavehax with LA=1 and LA=0 disappeared in TTS,' but these conclusions are not supported by any pairwise significance test (e.g., matched bootstrap, Wilcoxon, or ANOVA with a post-hoc test). The authors should add an appropriate significance test or explicitly present the comparison as descriptive rather than inferential.
- [Section 2.2 and Section 2.3] Two load-bearing design choices are justified only by 'internal experiments' that are never shown: the selection of trainable analysis/synthesis filters over PQMF and DWT, and the replacement of layer normalization with batch normalization. No objective metric values, architectures, or ablations are provided for these choices, which makes the proposed method difficult to reproduce and means the reader cannot assess whether these choices are robust or tuned to the specific corpora used. The authors should report the numerical results of these internal experiments or omit the claims and treat the choices as fixed design decisions.
- [Section 3.2] The generalizability of the throughput analysis to 'resource-constrained environments' is supported by measurements on a single AMD EPYC 7302 CPU with a 16 MiB L3 cache, which is a server-class processor rather than a typical edge or mobile CPU. The paper acknowledges cache and SIMD dependencies in the discussion, but no data is provided for a range of hardware (e.g., ARM mobile cores, lower-power x86, or devices with smaller caches). The practical recommendations about chunk sizes and 1D vs 2D convolution design should either be explicitly scoped to this one CPU or validated on at least one additional representative device.
minor comments (4)
- [Section 2.2] The sentence 'These spectrograms are then concatenated with the projected mel-spectrogram features, resulting in a 12-channel representation' appears twice verbatim; one copy should be removed.
- [Section 3.2 and Table 1] The table caption lists 'MACs' for PyTorch and ONNX, but the figure and text never clarify whether the PyTorch numbers use the native torch.stft/torch.istft or the same convolution-based replacement; this should be stated explicitly for both columns.
- [Section 4.2] The explanation for MS-Vocos underperforming Vocos is speculative ('We speculate that estimating inter-stream dependencies is particularly challenging...') and is not tested. If this claim is retained, it would benefit from a simple diagnostic, such as varying the latent channel count in MS-Vocos.
- [General] There are several typographical artifacts, such as 'V ocoder' in the title and repeated 'V ocos' spacing in the references; a final proofreading pass would be helpful.
Circularity Check
No significant circularity: the central throughput and quality claims are empirical comparisons against external baselines and corpora; self-citations are architectural precedents, not load-bearing evidence.
full rationale
The paper's derivation chain is empirical rather than definitional. MS-Wavehax's low-latency throughput is measured via ONNX Runtime RTF experiments (Section 3.2, Figure 2), its compactness is reported as measured parameter counts and MACs (Table 1), and its quality claims rest on listener MOS evaluations (Section 4.2, Figure 4). No fitted constant is renamed as a prediction, and no equation reduces the claimed result to its own inputs by construction. The im2col analysis (Table 2, Figure 3) is an explanatory account of the observed RTF reversal, not a derivation that forces the outcome. The paper does cite prior work by overlapping authors: Wavehax [12] and multi-stream variants [13, 27]. These self-citations establish the architectural lineage from which MS-Wavehax is built, but the paper's new claims are not justified solely by those citations; they are supported by new measurements against external baselines (HiFi-GAN, iSTFTNet, Vocos) and external corpora (JVS, JSUT). The Table 1 note that torch.stft/torch.istft were replaced by convolution-based implementations for ONNX conversion is a potential confound for the cross-model throughput comparison, since MACs change disproportionately across models (e.g., Vocos 1.348B to 46.96B vs. MS-Wavehax 1.576B to 2.291B). This is a validity or generalization concern, not circularity: the reported RTF ranking may depend on the STFT implementation choice, but nothing in the argument assumes its own conclusion by construction. Section 3.2's footnote also acknowledges that the im2col analysis assumes a particular backend and that other implementations may behave differently. These caveats lower confidence in the generality of the central claim but do not make the claim circular. Overall, the paper is self-contained against external benchmarks, so the appropriate finding is no significant circularity.
Assumptions & free parameters
free parameters (4)
- Number of subscale streams =
4
- MS-Wavehax STFT frame length/frame shift =
20 ms / 10 ms
- Analysis/synthesis filter length =
63
- Hidden channels in MS-Wavehax =
64
assumptions (4)
- domain assumption Trainable subscale filters outperform PQMF and DWT for the tested vocoders
- domain assumption Batch normalization is a suitable replacement for layer normalization in streaming and improves objective metrics
- domain assumption ONNX Runtime with im2col-based convolutions is representative of streaming inference on resource-constrained CPUs
- domain assumption MOS from 30 listeners on 10 samples per model can distinguish model quality
Cite this review
Pith. "Pith review of Comparative Analysis of Fast and High-Fidelity Neural Vocoders for Low-Latency Streaming Synthesis in Resource-Constrained Environments." pith.science (2026). https://pith.science/paper/Y5BH6M2N
@misc{pith2026250603554,
author = {Pith},
title = {Pith review of: Comparative Analysis of Fast and High-Fidelity Neural Vocoders for Low-Latency Streaming Synthesis in Resource-Constrained Environments},
year = {2026},
howpublished = {\url{https://pith.science/paper/Y5BH6M2N}},
note = {Machine review of arXiv:2506.03554}
}
read the original abstract
In real-time speech synthesis, neural vocoders often require low-latency synthesis through causal processing and streaming. However, streaming introduces inefficiencies absent in batch synthesis, such as limited parallelism, inter-frame dependency management, and parameter loading overhead. This paper proposes multi-stream Wavehax (MS-Wavehax), an efficient neural vocoder for low-latency streaming, by extending the aliasing-free neural vocoder Wavehax with multi-stream decomposition. We analyze the latency-throughput trade-off in a CPU-only environment and identify key bottlenecks in streaming neural vocoders. Our findings provide practical insights for optimizing chunk sizes and designing vocoders tailored to specific application demands and hardware constraints. Furthermore, our subjective evaluations show that MS-Wavehax delivers high speech quality under causal and non-causal conditions while being remarkably compact and easily deployable in resource-constrained environments.
Figures
Reference graph
Works this paper leans on
-
[1]
Introduction Advances in deep learning have led to remarkable improve- ments in speech synthesis, enabling highly natural speech generation. One key component of this progress is neural vocoders, which synthesize audio waveforms from acoustic fea- tures. Specifically, neural vocoders based on generative adver- sarial networks [1], such as MelGAN [2] and H...
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[2]
Proposed method This section introduces V ocos [7] and Wavehax [12], along with our proposed MS-V ocos and MS-Wavehax, which inte- grate multi-stream synthesis [13]. Additionally, we elaborate on techniques to facilitate streaming feasibility. 2.1. Overview of V ocos and Wavehax V ocos repurposes the ConvNeXt [20] model, originally devel- oped for image p...
work page Pith review arXiv 2025
-
[3]
Analysis of latency vs. throughput We analyze the relationship between latency and throughput via block streaming synthesis using several neural vocoders. 3.1. Model details We compared four neural vocoders, HiFi-GAN (V1) [3], iSTFT- Net (V1) [4], V ocos [7], and Wavehax [12], along with their multi-stream variants, each configured with four streams. All ...
-
[4]
All models assumed a 100-dimensional mel-spectrogram and a 1-dimensionalF 0 input. Wavehax and MS-Wavehax utilized F0 for generating prior signals, whereas other models concate- nate it with the mel-spectrogram, resulting in a 101-dimensional input feature. The sampling rate and frame shift are fixed at 24 kHz and 10 ms, respectively. Each model was adapt...
-
[5]
Speech quality evaluation In the experiments in Section 3, MS-Wavehax achieved the highest throughput under low-latency conditions. Next, we evaluate its speech quality under causal and non-causal condi- tions, compared to the vocoders described in Section 3.1. 1Although our analysis focuses on a particular scenario where backend-level optimizations depen...
-
[6]
Acknowledgements This work was supported in part by JST AIP Acceleration Re- search JPMJCR25U5, Japan, and in part by the Japan Society for the Promotion of Science (JSPS) Grants-in-Aid for Scien- tific Research (KAKENHI) under Grant 24KJ1236
-
[7]
Conclusion This study examined neural vocoders for streaming applica- tions. Our analysis revealed that streaming throughput de- pends on overhead from data and parameter loading as well as computational complexity. This overhead varies with the streaming unit sizes and the choice of 1D and 2D convolutions. These findings provide insights into designing s...
-
[9]
Generative Ad- versarial Nets,
I. Goodfellow, J. Pouget-Abadie, M. Mirzaet al., “Generative Ad- versarial Nets,” inProc. NeurIPS, vol. 27, 2014, pp. 2672–2680
work page 2014
Show all 48 references
-
[10]
MelGAN: Generative Adversarial Networks for Conditional Waveform Synthesis,
K. Kumar, R. Kumar, T. de Boissiereet al., “MelGAN: Generative Adversarial Networks for Conditional Waveform Synthesis,” in Proc. NeurIPS, 2019, pp. 14 910–14 921
2019
-
[11]
HiFi-GAN: Generative Adversarial Networks for Efficient and High Fidelity Speech Synthesis,
J. Kong, J. Kim, and J. Bae, “HiFi-GAN: Generative Adversarial Networks for Efficient and High Fidelity Speech Synthesis,” in Proc. NeurIPS, vol. 33, 2020, pp. 17 022–17 033
2020
-
[12]
iSTFTNet: Fast and Lightweight Mel-Spectrogram V ocoder Incorporating Inverse Short-Time Fourier Transform,
T. Kaneko, K. Tanaka, H. Kameokaet al., “iSTFTNet: Fast and Lightweight Mel-Spectrogram V ocoder Incorporating Inverse Short-Time Fourier Transform,” inProc. ICASSP, 2022, pp. 6207–6211
2022
-
[13]
iSTFTNet2: Faster and More Lightweight iSTFT-Based Neural V ocoder Using 1D- 2D CNN,
T. Kaneko, H. Kameoka, K. Tanakaet al., “iSTFTNet2: Faster and More Lightweight iSTFT-Based Neural V ocoder Using 1D- 2D CNN,” inProc. Interspeech, 2023, pp. 4369–4373
2023
-
[14]
APNet: An All-Frame-Level Neural V ocoder Incorporating Direct Prediction of Amplitude and Phase Spectra,
Y . Ai and Z.-H. Ling, “APNet: An All-Frame-Level Neural V ocoder Incorporating Direct Prediction of Amplitude and Phase Spectra,”IEEE/ACM TASLP, 2023
2023
-
[15]
V ocos: Closing the gap between time-domain and Fourier-based neural vocoders for high-quality audio synthesis,
H. Siuzdak, “V ocos: Closing the gap between time-domain and Fourier-based neural vocoders for high-quality audio synthesis,” inProc. ICLR, 2024
2024
-
[16]
AC-VC: Non-Parallel Low La- tency Phonetic Posteriorgrams Based V oice Conversion,
D. Ronssin and M. Cernak, “AC-VC: Non-Parallel Low La- tency Phonetic Posteriorgrams Based V oice Conversion,” inProc. ASRU, 2021, pp. 710–716
2021
-
[17]
Low-latency real-time non-parallel voice conversion based on cyclic variational autoencoder and multiband WaveRNN with data-driven linear prediction,
P. L. Tobing and T. Toda, “Low-latency real-time non-parallel voice conversion based on cyclic variational autoencoder and multiband WaveRNN with data-driven linear prediction,” inProc. SSW, 2021, pp. 142–147
2021
-
[18]
An Investigation of Streaming Non-Autoregressive sequence-to-sequence V oice Con- version,
T. Hayashi, K. Kobayashi, and T. Toda, “An Investigation of Streaming Non-Autoregressive sequence-to-sequence V oice Con- version,” inProc. ICASSP, 2022, pp. 6802–6806
2022
-
[19]
Streaming non-autoregressive model for any-to-many voice conversion,
Z. Chen, H. Miao, and P. Zhang, “Streaming non-autoregressive model for any-to-many voice conversion,”arXiv:2206.07288, 2022
2022 arXiv
-
[20]
Wavehax: Aliasing-Free Neural Waveform Synthesis Based on 2D Convo- lution and Harmonic Prior for Reliable Complex Spectrogram Es- timation,
R. Yoneyama, A. Miyashita, R. Yamamotoet al., “Wavehax: Aliasing-Free Neural Waveform Synthesis Based on 2D Convo- lution and Harmonic Prior for Reliable Complex Spectrogram Es- timation,”arXiv:2411.06807, 2024
2024
-
[21]
Multi-Stream HiFi-GAN with Data-Driven Waveform Decomposition,
T. Okamoto, T. Toda, and H. Kawai, “Multi-Stream HiFi-GAN with Data-Driven Waveform Decomposition,” inProc. ASRU, 2021, pp. 610–617
2021
-
[22]
Implementation of DNN-based real-time voice conversion and its improvements by audio data augmentation and mask-shaped device,
R. Arakawa, S. Takamichi, and H. Saruwatari, “Implementation of DNN-based real-time voice conversion and its improvements by audio data augmentation and mask-shaped device,” inProc. SSW, 2019, pp. 93–98
2019
-
[23]
Real-Time, Full-Band, Online DNN-Based V oice Conversion System Using a Single CPU,
T. Saeki, Y . Saito, S. Takamichiet al., “Real-Time, Full-Band, Online DNN-Based V oice Conversion System Using a Single CPU,” inProc. Interspeech, 2020, pp. 1021–1022
2020
-
[24]
Fregrad: Lightweight and Fast Frequency-Aware Diffusion V ocoder,
T. D. Nguyen, J.-H. Kim, Y . Janget al., “Fregrad: Lightweight and Fast Frequency-Aware Diffusion V ocoder,” inProc. ICASSP, 2024, pp. 10 736–10 740
2024
-
[25]
Incremental Text-to-Speech Syn- thesis with Prefix-to-Prefix Framework,
M. Ma, B. Zheng, K. Liuet al., “Incremental Text-to-Speech Syn- thesis with Prefix-to-Prefix Framework,” inProc. EMNLP, 2020, pp. 3886–3896
2020
-
[26]
Neural iTTS: Toward Synthesizing Speech in Real-time with End-to-end Neural Text- to-Speech Framework,
T. Yanagita, S. Sakti, and S. Nakamura, “Neural iTTS: Toward Synthesizing Speech in Real-time with End-to-end Neural Text- to-Speech Framework,” inProc. SSW, 2019, pp. 183–188
2019
-
[27]
High Qual- ity Streaming Speech Synthesis with Low, Sentence-Length- Independent Latency,
N. Ellinas, G. Vamvoukakis, K. Markopouloset al., “High Qual- ity Streaming Speech Synthesis with Low, Sentence-Length- Independent Latency,” inProc. Interspeech, 2020, pp. 2022–2026
2020
-
[28]
A ConvNet for the 2020s,
Z. Liu, H. Mao, C. Wuet al., “A ConvNet for the 2020s,” inProc. CVPR, 2022, pp. 11 966–11 976
2022
-
[29]
Design and evaluation of parallel quadrature mirror filters (PQMF),
C. Galand and D. Esteban, “Design and evaluation of parallel quadrature mirror filters (PQMF),” inProc. ICASSP, vol. 8, 1983, pp. 224–227
1983
-
[30]
Multi-band MelGAN: Faster Waveform Generation for High-Quality Text-to-Speech,
G. Yang, S. Yang, K. Liuet al., “Multi-band MelGAN: Faster Waveform Generation for High-Quality Text-to-Speech,” inProc. SLT, 2021
2021
-
[31]
Fre-GAN: Adversar- ial Frequency-Consistent Audio Synthesis,
J.-H. Kim, S.-H. Lee, J.-H. Leeet al., “Fre-GAN: Adversar- ial Frequency-Consistent Audio Synthesis,” inProc. Interspeech, 2021, pp. 2197–2201
2021
-
[32]
Harvest: A High-Performance Fundamental Fre- quency Estimator from Speech Signals,
M. Morise, “Harvest: A High-Performance Fundamental Fre- quency Estimator from Speech Signals,” inProc. Interspeech, 2017, pp. 2321–2325
2017
-
[33]
Perceptual evaluation of speech quality (PESQ)-a new method for speech quality assess- ment of telephone networks and codecs,
A. Rix, J. Beerends, M. Hollieret al., “Perceptual evaluation of speech quality (PESQ)-a new method for speech quality assess- ment of telephone networks and codecs,” inProc. ICASSP, vol. 2, 2001, pp. 749–752 vol.2
2001
-
[34]
UTMOS: UTokyo-SaruLab System for V oiceMOS Challenge 2022,
T. Saeki, D. Xin, W. Nakataet al., “UTMOS: UTokyo-SaruLab System for V oiceMOS Challenge 2022,” inProc. Interspeech, 2022, pp. 4521–4525
2022
-
[35]
Lightweight and High-Fidelity End-to-End Text-to-Speech with Multi-Band Generation and Inverse Short-Time Fourier Transform,
M. Kawamura, Y . Shirahata, R. Yamamotoet al., “Lightweight and High-Fidelity End-to-End Text-to-Speech with Multi-Band Generation and Inverse Short-Time Fourier Transform,” inProc. ICASSP, 2023, pp. 1–5
2023
-
[36]
Aggregate
(Mel-model), which combines a Transformer encoder [37] with a decoder based on optimal-transport conditional flow matching [38]. We used the official implementation [39] and set the mel-spectrogram hyperparameters to match those of the A/S task. We developed anF 0 prediction m...
-
[37]
Developing Real-Time Stream- ing Transformer Transducer for Speech Recognition on Large- Scale Dataset,
X. Chen, Y . Wu, Z. Wanget al., “Developing Real-Time Stream- ing Transformer Transducer for Speech Recognition on Large- Scale Dataset,” inProc. ICASSP, 2021, pp. 5904–5908
2021
-
[38]
Layer Normalization,
J. L. Ba, J. R. Kiros, and G. E. Hinton, “Layer Normalization,” arXiv:1607.06450, 2016
2016 arXiv
-
[39]
Batch Normalization: Accelerating Deep Network Training by Reducing Internal Covariate Shift,
S. Ioffe and C. Szegedy, “Batch Normalization: Accelerating Deep Network Training by Reducing Internal Covariate Shift,” in Proc. ICML, 2015, pp. 448–456
2015
-
[40]
JVS corpus: free Japanese multi-speaker voice corpus,
S. Takamichi, K. Mitsui, Y . Saitoet al., “JVS corpus: free Japanese multi-speaker voice corpus,”arXiv:1908.06248, 2019
1908 arXiv
-
[41]
BigVGAN: A Universal Neural V ocoder with Large-Scale Training,
S. gil Lee, W. Ping, B. Ginsburget al., “BigVGAN: A Universal Neural V ocoder with Large-Scale Training,” inProc. ICLR, 2023
2023
-
[42]
UnivNet: A Neural V ocoder with Multi-Resolution Spectrogram Discriminators for High-Fidelity Waveform Generation,
W. Jang, D. Lim, J. Yoonet al., “UnivNet: A Neural V ocoder with Multi-Resolution Spectrogram Discriminators for High-Fidelity Waveform Generation,” inProc. Interspeech, 2021, pp. 2207– 2211
2021
-
[43]
JSUT corpus: free large-scale Japanese speech corpus for end-to-end speech synthe- sis,
R. Sonobe, S. Takamichi, and H. Saruwatari, “JSUT corpus: free large-scale Japanese speech corpus for end-to-end speech synthe- sis,”arXiv:1711.00354, 2017
2017 arXiv
-
[44]
Matcha-TTS: A Fast TTS Architecture with Conditional Flow Matching,
S. Mehta, R. Tu, J. Beskowet al., “Matcha-TTS: A Fast TTS Architecture with Conditional Flow Matching,” inProc. ICASSP, 2024, pp. 11 341–11 345
2024
-
[45]
Attention is All you Need,
A. Vaswani, N. Shazeer, N. Parmaret al., “Attention is All you Need,” inProc. NeurIPS, vol. 30, 2017
2017
-
[46]
Flow Matching for Generative Modeling,
Y . Lipman, R. T. Q. Chen, H. Ben-Hamuet al., “Flow Matching for Generative Modeling,” inProc. ICLR, 2023
2023
-
[47]
Matcha-tts
shivammehta25. Matcha-tts. [Online]. Available: https://github. com/shivammehta25/Matcha-TTS
-
[48]
jsut-label
sarulab-speech. jsut-label. [Online]. Available: https://github. com/sarulab-speech/jsut-label
-
[49]
What the Future Brings: Investigating the Impact of Lookahead for Incremental Neural TTS,
B. Stephenson, L. Besacier, L. Girinet al., “What the Future Brings: Investigating the Impact of Lookahead for Incremental Neural TTS,” inProc. Interspeech, 2020, pp. 215–219
2020
Reviewed August 7, 2026 · model on record in the stance chip above.
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