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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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The abstract, sentence by sentence
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.
background contribution method result interpretation main claim
| # | sentence | role | the authors' own work | states the main conclusion | strength |
|---|---|---|---|---|---|
| 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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References
How the literature uses it
3 arXiv papers cite this in 5 sentences
Papers citing it · 1 sentence without a confident label
Compares with it 1 paper
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This is unlike [ref] and [ref] which enforce a specific bitrate during training and require training a different model for each target bitrate.
SoundStream: An End-to-End Neural Audio Codec · Related work · 0.98
Cites it as background 3 papers
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The recent improvements of deep neural network (DNN) models and learning methods have open new horizons for highly structured sequential data-related tasks, including automatic speech recognition (ASR) ( [ref] ) as well as audio compression ( [ref] , [ref] , [ref] ).
Deep Neural Networks and End-to-End Learning for Audio Compression · Introduction · 0.96
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An end-to-end audio codec targeting general audio at high bitrates (i.e., above $64$kbps) was proposed in [ref] .
SoundStream: An End-to-End Neural Audio Codec · Related work · 0.81
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In the speech signal processing, analysis/synthesis methods have been developed based on simple autoencoder (AE) [ref] or cascading AE structures [ref] .
Conditional variational autoencoder to improve neural audio synthesis for polyphonic music sound · Introduction · 1.00
What it cites · 5 sentences without a confident label
Uses the method or tool of 1 paper
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We use Adam optimizer [ref] with the batch size of 128 frames.
Adam: A Method for Stochastic Optimization · Experimental setup · 1.00
Compares with 1 paper
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We first show that for the raw PCM input CMRL outperforms AMR-WB and Kankanahalli-Net in terms of objective metrics in the experimental setup proposed in [ref] , where the use of LPC was not tested.
End-to-End Optimized Speech Coding with Deep Neural Networks · Experiments · 0.94
Cites as background 2 papers
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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
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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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