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Cascaded Cross-Module Residual Learning towards Lightweight End-to-End Speech Coding

Audio and Speech Processing · Submitted 18 Jun 2019

abstract compiled · 2026-09-23 · no language-model review

The authors' main claim The proposed method shows better objective performance than AMR-WB and the state-of-the-art DNN-based speech codec with a similar network architecture.

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arXiv ID1906.07769 (version 4)
Pith ID4RTCQAWN, Pith's permanent record of this paper and its review
CitationJongmo Sung, Kai Zhen, Minje Kim, Mi Suk Lee, Seungkwon Beack. "Cascaded Cross-Module Residual Learning towards Lightweight End-to-End Speech Coding." arXiv:1906.07769 (2019). https://pith.science/paper/4RTCQAWN
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arXiv listing abstract v4. Results and the main claim are underlined; .

Speech codecs learn compact representations of speech signals to facilitate data transmission. Many recent deep neural network (DNN) based end-to-end speech codecs achieve low bitrates and high perceptual quality at the cost of model complexity. We propose a cross-module residual learning (CMRL) pipeline as a module carrier with each module reconstructing the residual from its preceding modules. CMRL differs from other DNN-based speech codecs, in that rather than modeling speech compression problem in a single large neural network, it optimizes a series of less-complicated modules in a two-phase training scheme. The proposed method shows better objective performance than AMR-WB and the state-of-the-art DNN-based speech codec with a similar network architecture. As an end-to-end model, it takes raw PCM signals as an input, but is also compatible with linear predictive coding (LPC), showing better subjective quality at high bitrates than AMR-WB and OPUS. The gain is achieved by using only 0.9 million trainable parameters, a significantly less complex architecture than the other DNN-based codecs in the literature.

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What the classifier reports for each sentence. The number is the probability it gave the label.
#sentencerolethe authors' own workstates the main conclusionstrength
s0 Speech codecs learn compact representations of speech signals to facilitate data transmission. background 1.00 0.04 0.02 asserted 1.00
s1 Many recent deep neural network (DNN) based end-to-end speech codecs achieve low bitrates and high perceptual quality at the cost of model complexity. background 0.89 0.06 0.04 asserted 1.00
s2 We propose a cross-module residual learning (CMRL) pipeline as a module carrier with each module reconstructing the residual from its preceding modules. contribution 0.67 0.90 0.10 asserted 0.99
s3 CMRL differs from other DNN-based speech codecs, in that rather than modeling speech compression problem in a single large neural network, it optimizes a series of less-complicated modules in a two-phase training scheme. method 0.85 0.78 0.08 asserted 1.00
s4 The proposed method shows better objective performance than AMR-WB and the state-of-the-art DNN-based speech codec with a similar network architecture. result 1.00 0.94 0.34 asserted 1.00
s5 As an end-to-end model, it takes raw PCM signals as an input, but is also compatible with linear predictive coding (LPC), showing better subjective quality at high bitrates than AMR-WB and OPUS. result 0.78 0.91 0.14 asserted 1.00
s6 The gain is achieved by using only 0.9 million trainable parameters, a significantly less complex architecture than the other DNN-based codecs in the literature. result 0.86 0.92 0.28 asserted 1.00

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The paper
Original abstract

Speech codecs learn compact representations of speech signals to facilitate data transmission. Many recent deep neural network (DNN) based end-to-end speech codecs achieve low bitrates and high perceptual quality at the cost of model complexity. We propose a cross-module residual learning (CMRL) pipeline as a module carrier with each module reconstructing the residual from its preceding modules. CMRL differs from other DNN-based speech codecs, in that rather than modeling speech compression problem in a single large neural network, it optimizes a series of less-complicated modules in a two-phase training scheme. The proposed method shows better objective performance than AMR-WB and the state-of-the-art DNN-based speech codec with a similar network architecture. As an end-to-end model, it takes raw PCM signals as an input, but is also compatible with linear predictive coding (LPC), showing better subjective quality at high bitrates than AMR-WB and OPUS. The gain is achieved by using only 0.9 million trainable parameters, a significantly less complex architecture than the other DNN-based codecs in the literature.

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3 arXiv papers cite this in 5 sentences

Each citing sentence is taken from the citing paper's LaTeX source and typed by Pith's classifier. A label appears at probability 0.8 or more (0.95 for support), a contest at 0.7 or more; the quoted words are the citing authors'. How each label was checked · JSON

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  • There have been various conventional speech coding methodologies, including linear predictive coding (LPC) [ref] , adaptive encoding [ref] , and perceptual weighting [ref] among other domain specific knowledge about the speech signals, that are used to construct classic codecs, such as AMR-WB [ref] and OPUS [ref] with high perceptual quality.

    High-Quality, Low-Delay Music Coding in the Opus Codec · Introduction · 0.98

  • Many DNN methods [ref] [ref] take inputs in time-frequency (T-F) domain from short time Fourier transform (STFT) or modified discrete cosine transform (MDCT), etc.

    Composition of Deep and Spiking Neural Networks for Very Low Bit Rate Speech Coding · Introduction · 0.81

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