Pith. sign in

REVIEW 4 major objections 5 minor 30 references

Scalable Speech Enhancement with Dynamic Channel Pruning

T0 review · 4 major / 5 minor · reviewed 2026-08-11 · deepseek-v4-flash

Pith's one-line read A speech-enhancement network that prunes channels at runtime saves 29.6% of MACs while losing 0.75% PESQ.

desk verdict A solid, incremental transfer of channel gating to speech enhancement; the headline efficiency claim is slightly oversold but the core idea holds. read the letter →

arxiv 2412.17121 v1 pith:5TNNUR4D submitted 2024-12-22 eess.AS cs.LGcs.SD

classification eess.AScs.LGcs.SD
keywords speechenhancementdynamicchannelpruningneuralnetworksconvolutionalgatingmoduleedgeAIPESQefficientinference
verification ladder T0 review T1 audit T2 compute T3 formal

The pith

A machine-rendered reading of the paper's core claim, the machinery that carries it, and where it could break.

The reading

Speech enhancement models for headsets and speakerphones are usually static in compute, even though noise conditions vary. This paper claims that a network can instead decide at runtime, per input, which convolutional channels are worth computing, and that this dynamic channel pruning carries over from image tasks to audio-to-audio speech enhancement. On a custom convolutional architecture, training with a 25 percent channel-use target saves 29.6 percent of multiply-accumulate operations while PESQ, a standard speech-quality score, drops by a reported 0.75 percent. The authors present Pareto comparisons indicating that most dynamic models are more efficient than static baselines at matched quality, especially for shallow networks. If the claim holds, resource-constrained devices could run larger enhancement models that spend more computation on difficult acoustic conditions and less on easy ones.

What carries the argument

The load-bearing mechanism is the per-block gating module $G_i$ and its binary mask. Given the block input, the module pools over a window of $L_{\mathrm{pool}}$ frames, passes the pooled vector through two pointwise convolutions with a ReLU in between, and applies a Heaviside step (relaxed during training by surrogate gradients or a Concrete distribution) to produce a mask selecting which of the block's $C_{\mathrm{res}}$ output channels stay active. The mask is used two ways: in training it is multiplied onto the block output before the residual addition, and in inference it determines which filters in the final pointwise convolution are retrieved and executed. An auxiliary squared-error loss pulls the mean fraction of active channels toward a target pruning ratio $\Phi_{\mathrm{trgt}}$. Together these pieces let the network spend computation according to input difficulty, which is what the paper argues enables the Pareto improvement.

What would settle it

Run the same test utterances through two versions of the model: one that multiplies the trained binary mask into the static graph, as done in training, and one that genuinely skips the masked channels' convolutions at inference, then compare PESQ and output waveforms; any material difference between the two regimes would falsify the transfer assumption behind the reported savings.

Watch

Extended reading notes

Core claim

The central discovery is that dynamic channel pruning can be applied to a fully convolutional, frequency-domain speech-enhancement network without destroying quality. The proposed network, Conv-FSENet, is built from stacks of dilated depthwise-separable convolutional blocks with residual connections; each block is paired with a lightweight gating module that pools the block input over time, computes raw scores with pointwise convolutions, and binarizes them into a channel mask. During training the mask multiplies the block output in a static graph; during inference the mask decides which channels' filters are retrieved and which activations are actually computed, so the savings come from skipping work rather than from a smaller model. The gating and backbone are trained jointly with an auxiliary loss that enforces a target pruning ratio, and three ways of backpropagating through the binary step are compared, with a surrogate-gradient approach giving the best quality/saving balance. The authors report that most dynamic variants are more Pareto-efficient than static baselines, especially for shallow networks, and that the mask patterns correlate with noise and speech content in different blocks.

Load-bearing premise

The reported savings rest on the assumption that a channel whose computation is skipped at inference behaves exactly as if it had been computed and then multiplied by zero in the trained graph.

Editorial extensions

If this is right

  • On embedded devices, this mechanism would let a speech-enhancement model spend more computation on difficult, noisy passages and less on easy ones, rather than applying a fixed compute budget to every frame.
  • The target pruning ratio acts as a tunable knob: models trained with 10%, 25%, 50%, and 75% targets span a range of PESQ-versus-MAC operating points, and most sit on or near the Pareto front of the static baselines.
  • The learned masks expose channels that are almost always active and channels that are almost never used, so deployment can first statically prune the rarely used ones; the paper reports this heuristic raises average MAC savings from 29.6% to 31.3%.
  • Causal variants keep roughly 30% MAC reduction with a PESQ penalty of about 5% relative to non-causal models, indicating the approach is compatible with streaming constraints.
  • The authors state the result as a step toward applying dynamic channel pruning to more complex speech-enhancement networks and other audio-to-audio tasks.

Reading between the lines

Editorial extensions of the paper, not claims the author makes directly.

  • A testable extension the paper leaves implicit is whether the reported MAC reduction turns into wall-clock latency and energy savings on real hardware, since the gating module and its moving-average pooling also cost time and power.
  • The 29.6 percent MAC reduction at a 25 percent channel target is well below what naively skipping 75 percent of channels would suggest; training with a loss that directly minimizes MACs or gating overhead might close that gap.
  • The observed correlations between mask activity and noise or speech presence in different blocks suggest the gating signal could double as a lightweight acoustic-condition detector, though the paper does not pursue this.
  • Since the pooling window is fixed to the receptive field and grows with depth, the paper's own explanation for weaker adaptiveness in deep networks could be tested by decoupling $L_{\mathrm{pool}}$ from $L_{\mathrm{RF}}$ and sweeping shorter windows.
Share X Bluesky LinkedIn Reddit HN

Signed reviews

No signed human review yet.

Editorial analysis

A structured set of objections, weighed in public.

Desk editor's note, referee report, and a circularity audit.

Referee Report

4 major / 5 minor

Summary. The paper introduces Dynamic Channel Pruning (DynCP) for speech enhancement, applied to a custom fully convolutional architecture named Conv-FSENet. A lightweight gating module per block predicts a binary mask that selects which channels of the block's final pointwise convolution are computed at inference. The model is trained end-to-end with surrogate-gradient or Binary Concrete relaxation of the binarization, plus an auxiliary loss that enforces an average target pruning ratio. Experiments on VoiceBank+DEMAND report PESQ, SI-SDR, and MACs per STFT frame, comparing static baselines and dynamic variants across causal/non-causal settings and different binarization strategies. The main claimed result is that the dynamic model saves 29.6% of MACs while incurring only a 0.75% drop in PESQ when trained to use 25% of channels on average.

Significance. If the reported trade-off is reliable, this is a useful step toward input-adaptive computation for on-device speech enhancement, and the paper is, to my knowledge, the first to transfer dynamic channel pruning to audio-to-audio enhancement. The architecture and gating formulation are clearly presented, and the comparison across surrogate gradients is informative. The strength of the claim, however, depends on precise and reproducible accounting of both the PESQ drop and the MAC savings, and the current manuscript does not yet provide the necessary statistical or measurement detail.

major comments (4)
  1. [Abstract and Sec. 7 vs. Table 2] The headline claim of a 0.75% PESQ drop is not supported by the numbers in Table 2. For the non-causal SuperSpike row, PESQ is 2.90 versus 2.92 for the baseline, which is a 0.68% relative drop; for the causal SuperSpike row the drop is 1.44% (2.73 vs. 2.77); for the non-causal Sigmoid row it is 1.03% (2.89 vs. 2.92). The abstract and conclusion should state the exact configuration and whether the percentage is relative or absolute. Given that the differences are as small as 0.02 PESQ, single-run results without error bars are insufficient to support the claimed trade-off.
  2. [Sec. 4 and Fig. 3] The equivalence between training-time mask multiplication and inference-time channel skipping should be made explicit and verified. In the training graph all channels of the final PW⊛ are computed and then masked, whereas at inference only active output channels are computed. For a pointwise convolution this is mathematically equivalent provided no normalization or nonlinearity is applied after the mask; the text should confirm that BatchNorm and PReLU occur before the final PW⊛ and that no operation is applied after the mask beyond the skip addition. More importantly, the paper must state how the reported MAC counts were computed: fvcore's flop_count on a static graph would count all channels, so the 29.6% reduction can only be reproduced if the actual inference-time active-channel masks were used. The cost of the gating module (two PW⊛ in Gi) must also be explicitly included or excluded from the reported MACs.
  3. [Sec. 5 (Training) and Table 2] No random seed or number of runs is reported, and the early-stopping protocol is described ambiguously ('interrupt the training after 20 epochs without improvement' and 'decay our learning rate by a factor of 0.5 after 3 validation rounds'). Since the central result is a very small PESQ difference, the paper should provide mean and standard deviation over at least three seeds, or clearly state that only one run was performed and temper the claims accordingly.
  4. [Sec. 6 (Discussion of deeper networks)] The attribution of degraded dynamism in deeper networks to Lpool = LRF is plausible but confounded: increasing Ns changes both the receptive field and the number of blocks/gating modules. Without an ablation that varies Lpool independently of network depth, the statement that 'smaller networks react faster... exhibiting higher adaptiveness' is not directly supported by the presented experiments.
minor comments (5)
  1. [Throughout] The architecture name is written inconsistently as 'Conv-FSENet' and 'ConvFSE-Net'; please unify.
  2. [Sec. 3.1, Eq. (2)] The complex mask cM is used before its definition; define the mask and its dimensions at first use.
  3. [Sec. 4, Eq. (5)] The notation in LDCP is slightly confusing: the mask tensor is indexed over batch N, time L, and block I, but the equation averages over N, L, I per channel c; spell out that the MSE is computed after averaging across those dimensions.
  4. [Sec. 5 (Model)] The text says 'Φtrgt = 0.25 unless noted otherwise', but Fig. 4 uses targets 10%, 25%, 50%, 75%; clarify how '25% of channels' relates to the measured 29.6% MAC reduction.
  5. [Sec. 6, Fig. 4] The Pareto-front plot would benefit from error bars or at least point labels for the specific configurations in Table 2; currently it is hard to map the plotted points to the tabulated rows.

Circularity Check

0 steps flagged · score 1.0 of 10

No significant circularity: the reported MAC/PESQ trade-offs are measured outcomes under a user-set pruning target, with only background self-citations.

full rationale

The paper's central claims—a 29.6% MAC saving at a 0.75% PESQ drop and the Pareto-front improvements in Fig. 4—are empirical results. PESQ and SI-SDR are computed on the held-out VoiceBank+DEMAND test set, and kMACs are computed with fvcore from the actual inference-time mask activity. The LDCP loss (Eq. 5) does nudge the time/block-averaged mask toward a user-chosen target Phi_trgt, but that target is an operating point, not a fitted parameter, and it does not by itself determine the measured PESQ, SI-SDR, or exact kMAC count; the reported trade-offs therefore have independent empirical content. The only self-citations ([5] in related work and [19] for the SuperSpike surrogate) are background and non-load-bearing; no uniqueness theorem or prior result by the same authors is used to forbid alternatives or to define the outcome. The train/inference gap between masked training and inference-time channel skipping is a genuine validity risk about whether measured PESQ transfers to actual skipping, but it is a correctness/engineering assumption, not a circular reduction of the output to the input. Hence no significant circularity.

Assumptions & free parameters 5 free parameters · 4 assumptions · 0 invented entities

The central claim rests on benchmark validity, quality metrics, a hardware-cost proxy, and the stability of surrogate-gradient training; none of these are proven in the paper, and hardware savings are not directly measured.

free parameters (5)
  • Phi_trgt (target pruning ratio) = 0.25 for main claim; 0.10, 0.50, 0.75 in Pareto sweep
    Chosen by hand to define the operating point; the headline result selects the 25% setting.
  • alpha in LSE = 0.3
    Chosen weight balancing complex-domain and magnitude-domain loss terms in Eq. (3).
  • c in LSE = 0.3
    Chosen compression exponent for dynamic range compression in the loss.
  • Cgate = 16
    Hidden feature dimension of the gating module, chosen by hand.
  • Lpool = LRF (receptive field length)
    Pooling window length for the gating module, set equal to the model receptive field.
assumptions (4)
  • domain assumption The VoiceBank+DEMAND dataset is a representative benchmark for speech enhancement, and the provided noisy-clean pairs are correct.
    All experiments and headline numbers are measured on this dataset; no independent test on other corpora is performed.
  • domain assumption PESQ and SI-SDR quantify the relevant quality of enhanced speech, and MACs are a valid proxy for computational cost on the target embedded devices.
    The paper does not measure latency or energy on actual hardware; savings are estimated from fvcore MAC counts.
  • domain assumption The surrogate-gradient training procedure (Sigmoid, SuperSpike, or Binary Concrete) yields a well-behaved minimizer of the non-differentiable gating objective.
    Empirically validated in Table 2, but no theoretical guarantee; the paper itself notes the Binary Concrete variant underperforms.
  • domain assumption Standard backpropagation and Adam training of the joint backbone-gating system converges to a solution where the mask reflects input-dependent channel importance.
    The paper relies on the training dynamics to produce useful masks; the failure mode of inactive channels never receiving gradient is acknowledged in Sec. 6 and motivates the stochastic variant.

how reviews work

0 comments
Cite this review

Pith. "Pith review of Scalable Speech Enhancement with Dynamic Channel Pruning." pith.science (2026). https://pith.science/paper/5TNNUR4D

@misc{pith2026241217121,
  author       = {Pith},
  title        = {Pith review of: Scalable Speech Enhancement with Dynamic Channel Pruning},
  year         = {2026},
  howpublished = {\url{https://pith.science/paper/5TNNUR4D}},
  note         = {Machine review of arXiv:2412.17121}
}
read the original abstract

Speech Enhancement (SE) is essential for improving productivity in remote collaborative environments. Although deep learning models are highly effective at SE, their computational demands make them impractical for embedded systems. Furthermore, acoustic conditions can change significantly in terms of difficulty, whereas neural networks are usually static with regard to the amount of computation performed. To this end, we introduce Dynamic Channel Pruning to the audio domain for the first time and apply it to a custom convolutional architecture for SE. Our approach works by identifying unnecessary convolutional channels at runtime and saving computational resources by not computing the activations for these channels and retrieving their filters. When trained to only use 25% of channels, we save 29.6% of MACs while only causing a 0.75% drop in PESQ. Thus, DynCP offers a promising path toward deploying larger and more powerful SE solutions on resource-constrained devices.

Discussion (0). Continue with ORCID to comment.

Reference graph

Works this paper leans on

30 extracted references · 21 canonical work pages

  1. [1]

    INTRODUCTION Real-time speech enhancement (SE) and noise suppression can facili- tate communication in noisy environments and have become ubiqui- tous in devices such as speakerphones, earbuds, and headsets. Thanks to the recent advances in deep learning, SE solutions based on neural networks made considerable strides, surpassing digital signal process- i...

  2. [2]

    Scalable Speech Enhancement with Dynamic Channel Pruning

    RELA TED WORK Dynamic Neural Networks are a class of neural networks where some aspects of the computational graph change at inference-time, usually based on the input data. We refer the interested reader to [3] for a comprehensive survey of DynNN techniques. In this section, we focus exclusively on sample-wise techniques applied to audio applications whe...

  3. [3]

    THE CONV-FSENET ARCHITECTURE 3.1. Problem formulation We consider the problem of single-channel speech enhancement in the time-frequency domain, where our input signal x(t) is a mixture of target speech s(t) and background noise n(t). In the Short-Time Fourier Transform (STFT) domain, we have: X(l, f) = S(l, f) + N (l, f) (1) where l is the STFT frame ind...

  4. [4]

    binary special case

    DYNAMIC CHANNEL PRUNING The number of multiply-accumulate operations (MACs) performed by each type of convolution can be approximated as: MACPW⊛ ≈ L · Cconv · Cres MACDDW⊛ ≈ L · Cconv · k (4) Since Cres ≫ k, we will ignore the impact of the middle DDW⊛and only concentrate on the last PW⊛, leaving the other layers unaffected. To determine which convolution...

  5. [5]

    Similarly, the test set includes 824 samples from two other speakers mixed with unseen noise at SNR between 17.5 dB and 2.5 dB

    EXPERIMENTAL SETUP Datasets We trained and evaluated our models on the V oice- Bank+DEMAND Dataset [21], consisting of 11 752 pairs of noisy- clean samples from 28 speakers with SNR between 15 dB and 0 dB. Similarly, the test set includes 824 samples from two other speakers mixed with unseen noise at SNR between 17.5 dB and 2.5 dB. All the data is downsam...

  6. [6]

    4, we relate the denoising performances and computational efficiency of the Conv-FSENet static baselines with their DynCP counterparts

    RESULTS In Fig. 4, we relate the denoising performances and computational efficiency of the Conv-FSENet static baselines with their DynCP counterparts. Although our dynamic variants experience a drop in PESQ, we benefit from a significant reduction in MACs (between 7 % and 39 %, depending on Φtrgt), which makes most of our DynCP models more Pareto-efficie...

  7. [7]

    Com- pared to the static baseline in Table 2, our dynamic models can save up to 29.6 % of MACs while only incurring a 0.75 % drop in PESQ

    CONCLUSION We presented ConvFSE-Net, a neural network architecture for SE, and extended it with a dynamic pruning system that learns to skip unnecessary convolutional channels based on the input data. Com- pared to the static baseline in Table 2, our dynamic models can save up to 29.6 % of MACs while only incurring a 0.75 % drop in PESQ. Thus, our results...

  8. [8]

    Real Time Speech Enhancement in the Waveform Domain,

    Alexandre Defossez, Gabriel Synnaeve, and Yossi Adi, “Real Time Speech Enhancement in the Waveform Domain,” Sept. 2020, arXiv:2006.12847 [cs, eess, stat]

Show all 30 references
  1. [9]

    TFCN: Temporal-Frequential Convolutional Network for Single-Channel Speech Enhance- ment,

    Xupeng Jia and Dongmei Li, “TFCN: Temporal-Frequential Convolutional Network for Single-Channel Speech Enhance- ment,” Jan. 2022, arXiv:2201.00480 [eess]

  2. [10]

    Dynamic Neural Networks: A Survey,

    Yizeng Han, Gao Huang, Shiji Song, Le Yang, Honghui Wang, and Yulin Wang, “Dynamic Neural Networks: A Survey,” Dec. 2021, arXiv:2102.04906 [cs]

  3. [11]

    Don’t shoot butterfly with rifles: Multi- channel Continuous Speech Separation with Early Exit Trans- former,

    Sanyuan Chen, Yu Wu, Zhuo Chen, Takuya Yoshioka, Shujie Liu, and Jinyu Li, “Don’t shoot butterfly with rifles: Multi- channel Continuous Speech Separation with Early Exit Trans- former,” Oct. 2020, arXiv:2010.12180 [cs, eess]

  4. [12]

    Dynamic nsNET2: Efficient Deep Noise Suppression with Early Exiting,

    Riccardo Miccini, Alaa Zniber, Cl ´ement Laroche, Tobias Piechowiak, Martin Schoeberl, Luca Pezzarossa, Ouassim Kar- rakchou, Jens Sparsø, and Mounir Ghogho, “Dynamic nsNET2: Efficient Deep Noise Suppression with Early Exiting,” in 2023 IEEE 33rd International Workshop on Mach...

  5. [13]

    Latent Iterative Refinement for Modular Source Separation,

    Dimitrios Bralios, Efthymios Tzinis, Gordon Wichern, Paris Smaragdis, and Jonathan Le Roux, “Latent Iterative Refinement for Modular Source Separation,” Nov. 2022, arXiv:2211.11917 [cs, eess]

  6. [14]

    Slim-Tasnet: A Slimmable Neural Network for Speech Separation,

    Mohamed Elminshawi, Srikanth Raj Chetupalli, and Emanu¨el A. P. Habets, “Slim-Tasnet: A Slimmable Neural Network for Speech Separation,” in 2023 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), New Paltz, NY , USA, Oct. 2023, pp. 1–5, IEEE

  7. [15]

    Runtime Neural Pruning,

    Ji Lin, Yongming Rao, Jiwen Lu, and Jie Zhou, “Runtime Neural Pruning,” in Advances in Neural Information Processing Systems. 2017, vol. 30, Curran Associates, Inc

  8. [16]

    Channel Gating Neural Networks,

    Weizhe Hua, Yuan Zhou, Christopher M De Sa, Zhiru Zhang, and G. Edward Suh, “Channel Gating Neural Networks,” in Ad- vances in Neural Information Processing Systems. 2019, vol. 32, Curran Associates, Inc

  9. [17]

    Dynamic Channel Pruning: Feature Boosting and Suppression,

    Xitong Gao, Yiren Zhao, Lukasz Dudziak, Robert Mullins, and Cheng-zhong Xu, “Dynamic Channel Pruning: Feature Boosting and Suppression,” Jan. 2019, arXiv:1810.05331 [cs]

  10. [18]

    Runtime Network Routing for Efficient Image Classification,

    Yongming Rao, Jiwen Lu, Ji Lin, and Jie Zhou, “Runtime Network Routing for Efficient Image Classification,” IEEE Transactions on Pattern Analysis and Machine Intelligence, vol. 41, no. 10, pp. 2291–2304, Oct. 2019, Conference Name: IEEE Transactions on Pattern Analysis and Mac...

  11. [19]

    Dynamic Neural Net- work Channel Execution for Efficient Training,

    Simeon E. Spasov and Pietro Lio, “Dynamic Neural Net- work Channel Execution for Efficient Training,” May 2019, arXiv:1905.06435 [cs, stat]

  12. [20]

    Learning to Inference with Early Exit in the Progressive Speech Enhancement,

    Andong Li, Chengshi Zheng, Lu Zhang, and Xiaodong Li, “Learning to Inference with Early Exit in the Progressive Speech Enhancement,” June 2021, arXiv:2106.11730 [cs, eess]

  13. [21]

    Conv-TasNet: Surpassing Ideal Time-Frequency Magnitude Masking for Speech Sep- aration,

    Yi Luo and Nima Mesgarani, “Conv-TasNet: Surpassing Ideal Time-Frequency Magnitude Masking for Speech Sep- aration,” IEEE/ACM Transactions on Audio, Speech, and Lan- guage Processing, vol. 27, no. 8, pp. 1256–1266, Aug. 2019, arXiv:1809.07454 [cs, eess]

  14. [22]

    Dynamic Slimmable Network for Speech Sepa- ration,

    Mohamed Elminshawi, Srikanth Raj Chetupalli, and Emanu¨el A. P. Habets, “Dynamic Slimmable Network for Speech Sepa- ration,” IEEE Signal Processing Letters, pp. 1–5, 2024, Confer- ence Name: IEEE Signal Processing Letters

  15. [23]

    TCNN: Temporal Con- volutional Neural Network for Real-time Speech Enhancement in the Time Domain,

    Ashutosh Pandey and DeLiang Wang, “TCNN: Temporal Con- volutional Neural Network for Real-time Speech Enhancement in the Time Domain,” in 2019 IEEE International Conference on Acoustics, Speech and Signal Processing, Brighton, United Kingdom, May 2019, pp. 6875–6879, IEEE

  16. [24]

    An Empirical Evaluation of Generic Convolutional and Recurrent Networks for Sequence Modeling,

    Shaojie Bai, J. Zico Kolter, and Vladlen Koltun, “An Empirical Evaluation of Generic Convolutional and Recurrent Networks for Sequence Modeling,” arXiv:1803.01271 [cs], Mar. 2018, arXiv: 1803.01271

  17. [25]

    79–86, Springer International Publishing, 2020

    Sebastian Braun and Ivan Tashev, Data Augmentation and Loss Normalization for Deep Noise Suppression, p. 79–86, Springer International Publishing, 2020

  18. [26]

    Resource-Efficient Speech Quality Prediction through Quantization Aware Train- ing and Binary Activation Maps,

    Mattias Nilsson, Riccardo Miccini, Cl ´ement Laroche, To- bias Piechowiak, and Friedemann Zenke, “Resource-Efficient Speech Quality Prediction through Quantization Aware Train- ing and Binary Activation Maps,” July 2024, arXiv:2407.04578 [cs, eess]

  19. [27]

    The Concrete Distribution: A Continuous Relaxation of Discrete Random Variables,

    Chris J. Maddison, Andriy Mnih, and Yee Whye Teh, “The Concrete Distribution: A Continuous Relaxation of Discrete Random Variables,” Mar. 2017, arXiv:1611.00712 [cs, stat]

  20. [28]

    Investigating RNN-based speech enhance- ment methods for noise-robust Text-to-Speech,

    Cassia Valentini-Botinhao, Xin Wang, Shinji Takaki, and Ju- nichi Yamagishi, “Investigating RNN-based speech enhance- ment methods for noise-robust Text-to-Speech,” in 9th ISCA Workshop on Speech Synthesis Workshop. Sept. 2016, pp. 146– 152, ISCA

  21. [29]

    Per- ceptual evaluation of speech quality (PESQ)-a new method for speech quality assessment of telephone networks and codecs,

    A.W. Rix, J.G. Beerends, M.P. Hollier, and A.P. Hekstra, “Per- ceptual evaluation of speech quality (PESQ)-a new method for speech quality assessment of telephone networks and codecs,” in 2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedin...

  22. [30]

    SDR - half-baked or well done?,

    Jonathan Le Roux, Scott Wisdom, Hakan Erdogan, and John R. Hershey, “SDR - half-baked or well done?,” Nov. 2018, arXiv:1811.02508 [cs, eess]

Pith tools

Reviewed August 11, 2026 · model on record in the stance chip above.