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检索条件"主题词=Audio coding"

共 1193 条记录，以下是21-30 订阅

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OpenACE: An Open Benchmark for Evaluating audio coding Performance

OpenACE: An Open Benchmark for Evaluating Audio Coding Perfo...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Jozef Coldenhoff Niclas Granqvist Milos Cernak Logitech Europe S.A. Lausanne Switzerland

ISBN: (数字)9798350368741

ISBN: (纸本)9798350368758

audio and speech coding lack unified evaluation and open-source testing. Many candidate systems were evaluated on proprietary, non-reproducible, or small data, and machine learning-based codecs are often tested on datasets with similar distributions as trained on, which is unfairly compared to digital signal processing-based codecs that usually work well with unseen data. This paper presents a full-band audio and speech coding quality benchmark with more variable content types, including traditional open test vectors. An example use case of audio coding quality assessment is presented with open-source Opus, 3GPP’s EVS, and recent ETSI’s LC3 with LC3+ used in Bluetooth LE audio profiles. Besides, quality variations of emotional speech encoding at 16 kbps are shown. The proposed open-source benchmark contributes to audio and speech coding democratization and is available at https://***/JozefColdenhoff/OpenACE.

关键词： Codecs Speech coding audio coding Working environment noise Benchmark testing Data augmentation Data models Vectors Reverberation Speech processing

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Perceptual audio coding: A 40-Year Historical Perspective

Perceptual Audio Coding: A 40-Year Historical Perspective

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Jürgen Herre Schuyler Quackenbush Minje Kim Jan Skoglund International Audio Laboratories Erlangen Erlangen Germany Audio Research Labs Westfield NJ University of Illinois at Urbana-Champaign Urbana IL Google LLC San Francisco CA

ISBN: (数字)9798350368741

ISBN: (纸本)9798350368758

In the history of audio and acoustic signal processing, perceptual audio coding has certainly excelled as a bright success story by its ubiquitous deployment in virtually all digital media devices, such as computers, tablets, mobile phones, set-top-boxes, and digital radios. From a technology perspective, perceptual audio coding has undergone tremendous development from the first very basic perceptually driven coders (including the popular mp3 format) to today’s full-blown integrated coding/rendering systems. This paper provides a historical overview of this research journey by pinpointing the pivotal development steps in the evolution of perceptual audio coding. Finally, it provides thoughts about future directions in this area.

关键词： Computers audio coding Speech coding Media Digital communication Mobile handsets Acoustics History Speech processing Digital audio players

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Variable Bitrate Residual Vector Quantization for audio coding

Variable Bitrate Residual Vector Quantization for Audio Codi...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Yunkee Chae Woosung Choi Yuhta Takida Junghyun Koo Yukara Ikemiya Zhi Zhong Kin Wai Cheuk Marco A. Martínez-Ramírez Kyogu Lee Wei-Hsiang Liao Yuki Mitsufuji Sony AI IPAI Sony Group Corporation Tokyo Japan AIIS Department of Intelligence and Information Seoul National University

ISBN: (数字)9798350368741

ISBN: (纸本)9798350368758

Recent state-of-the-art neural audio compression models have progressively adopted residual vector quantization (RVQ). Despite this success, these models employ a fixed number of codebooks per frame, which can be suboptimal in terms of rate-distortion tradeoff, particularly in scenarios with simple input audio, such as silence. To address this limitation, we propose variable bitrate RVQ (VRVQ) for audio codecs, which allows for more efficient coding by adapting the number of codebooks used per frame. Furthermore, we propose a gradient estimation method for the non-differentiable masking operation that transforms from the importance map to the binary importance mask, improving model training via a straight-through estimator. We demonstrate that the proposed training framework achieves superior results compared to the baseline method and shows further improvement when applied to the current state-of-the-art codec. audio samples are available at: https://***/***/

关键词： Training Adaptation models Codecs audio coding Vector quantization Bit rate Rate-distortion Estimation Transforms Vectors

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Post-Training Latent Dimension Reduction in Neural audio coding

Post-Training Latent Dimension Reduction in Neural Audio Cod...

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European Signal Processing Conference (EUSIPCO)

作者： Thomas Muller Stéphane Ragot Pierrick Philippe Pascal Scalart Orange Innovation IRISA University of Rennes Lannion France Orange Innovation Lannion France Orange Innovation Rennes France IRISA University of Rennes Lannion France

ISBN: (数字)9789464593617

ISBN: (纸本)9798331519773

This work addresses the problem of latent space quantization in neural audio coding. A covariance analysis of latent space is performed on several pre-trained audio coding models (Lyra V2, EnCodec, audioDec). It is proposed to truncate latent space dimension using a fixed linear transform. The Karhunen-Loève transform (KLT) is applied on learned residual vector quantization (RVQ) codebooks. The proposed method is applied in a backward-compatible way to EnCodec, and we show that quantization complexity and codebook storage are reduced (by 43.4%), with no noticeable difference in subjective AB tests.

关键词： Dimensionality reduction Degradation Codecs audio coding Vector quantization Europe Transforms Vectors Complexity theory Decorrelation

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HARP-NET: HYPER-AUTOENCODED RECONSTRUCTION PROPAGATION FOR SCALABLE NEURAL audio coding

HARP-NET: HYPER-AUTOENCODED RECONSTRUCTION PROPAGATION FOR S...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics (WASPAA)

作者： Petermann, Darius Beack, Seungkwon Kim, Minje Indiana Univ Dept Intelligent Syst Engn Bloomington IN 47408 USA Elect & Telecommun Res Inst Daejeon 34129 South Korea

ISBN: (纸本)9781665448703

We propose a novel autoencoder architecture that improves the architectural scalability of general-purpose neural audio coding models. An autoencoder-based codec employs quantization to turn its bottleneck layer activation into bitstrings, a process that hinders information flow between the encoder and decoder parts. To circumvent this issue, we employ additional skip connections between the corresponding pair of encoder-decoder layers. The assumption is that, in a mirrored autoencoder topology, a decoder layer reconstructs the intermediate feature representation of its corresponding encoder layer. Hence, any additional information directly propagated from the corresponding encoder layer helps the reconstruction. We implement this kind of skip connections in the form of additional autoencoders, each of which is a small codec that compresses the massive data transfer between the paired encoder-decoder layers. We empirically verify that the proposed hyperautoencoded architecture improves perceptual audio quality compared to an ordinary autoencoder baseline.

关键词： audio coding deep learning U-Net autoencoders

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Native Multi-Band audio coding Within Hyper-Autoencoded Reconstruction Propagation Networks

Native Multi-Band Audio Coding Within Hyper-Autoencoded Reco...

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International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Darius Petermann Inseon Jang Minje Kim Department of Intelligent Systems Engineering Indiana University Bloomington IN USA Electronics and Telecommunications Research Institute Daejeon South Korea

Spectral sub-bands do not portray the same perceptual relevance. In audio coding, it is therefore desirable to have independent control over each of the constituent bands so that bitrate assignment and signal reconstruction can be achieved efficiently. In this work, we present a novel neural audio coding network that natively supports a multi-band coding paradigm. Our model extends the idea of compressed skip connections in the U-Net-based codec, allowing for independent control over both core and high band-specific reconstructions and bit allocation. Our system reconstructs the full-band signal mainly from the condensed core-band code, therefore exploiting and showcasing its bandwidth extension capabilities to its fullest. Meanwhile, the low-bitrate high-band code helps the high-band reconstruction similarly to MPEG audio codecs' spectral bandwidth replication. MUSHRA tests show that the proposed model not only improves the quality of the core band by explicitly assigning more bits to it but retains a good quality in the high-band as well.

关键词： Codes Codecs audio coding Bit rate Neural networks Transform coding Signal processing algorithms

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High Quality audio coding with Mdctnet

High Quality Audio Coding with Mdctnet

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International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Grant Davidson Mark Vinton Per Ekstrand Cong Zhou Lars Villemoes Lie Lu Dolby Laboratories Inc. San Francisco CA USA Dolby Sweden AB Stockholm Sweden

We propose a neural audio generative model, MDCTNet, operating in the perceptually weighted domain of an adaptive modified discrete cosine transform (MDCT). The architecture of the model captures correlations in both time and frequency directions with recurrent layers (RNNs). An audio coding system is obtained by training MDCTNet on a diverse set of fullband monophonic audio signals at 48 kHz sampling, conditioned by a perceptual audio encoder. In a subjective listening test with ten excerpts chosen to be balanced across content types, yet critical for both codecs, the mean performance of the proposed system for 24 kb/s variable bitrate (VBR) is similar to that of Opus at twice the bitrate.

关键词： Training Adaptation models Time-frequency analysis Codecs audio coding Bit rate Signal processing

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Low Delay Robust audio coding by Noise Shaping, Fractional Sampling, and Source Prediction

Low Delay Robust Audio Coding by Noise Shaping, Fractional S...

引用

Data Compression Conference (DCC)

作者： Ostergaard, Jan Aalborg Univ Dept Elect Syst Fredrik Bajers Vej 7 DK-9000 Aalborg Denmark

ISBN: (纸本)9781665403337

It was recently shown that the combination of source prediction, two-times oversampling, and noise shaping, can be used to obtain a robust (multiple-description) audio coding framework for networks with packet loss probabilities less than 10%. Specifically, it was shown that audio signals could be encoded into two descriptions (packets), which were separately sent over a communication channel. Each description yields a desired performance by itself, and when they are combined, the performance is improved. This paper extends the previous work to an arbitrary number of descriptions (packets) by using fractional oversampling and a new decoding principle. We demonstrate that, due to source aliasing, existing MSE optimized reconstruction rules from noisy sampled data, performs poorly from a perceptual point of view. A simple reconstruction rule is proposed, that improves the PEAQ objective difference grades (ODG) by more than 2 points. The proposed audio coder enables low-delay high-quality audio streaming on networks with late packet arrivals or packet losses. With a coding delay of 2.5 ms, and a total bitrate of 300 kbps, it is demonstrated that mean PEAQ ODGs around -0.65 can be obtained for 48 kHz (mono) music (pop & rock), and packet loss probabilities of 20%.

关键词： audio coding fractional sampling low delay Multiple descriptions noise shaping source predictions

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Multiple Description audio coding for Wireless Low-Frequency Sound Zones

Multiple Description Audio Coding for Wireless Low-Frequency...

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Data Compression Conference (DCC)

作者： Jan Østergaard Christian Pedersen Mo Zhou Niels de Koeijer Martin Møller Aalborg University Fredrik Bajers Vej 7B Aalborg Denmark TBang & Olufsen Bang og Olufsen Alle Struer Denmark

We present a joint design of sound zone control filters and robust audio coding for wireless low frequency sound zones. The audio signal is filtered using sound zone control filters and encoded using a multiple-description coder. The control filters and the multiple-description coder are combined in a nested loop. The inner loop performs filtering for sound zone control and generates multiple descriptions using oversampling and closed-loop prediction. The outer loop performs noise shaping and guarantees a trade-off between robustness and quality of the descriptions. A closed-form expression for the optimal sound-zone control filters are provided, and a simulation study demonstrates that even at moderate packet loss rates, a significant gain is possible compared to not using multiple descriptions.

关键词： Wireless communication Closed-form solutions Finite impulse response filters audio coding Filtering Packet loss Robustness

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End-to-End Neural audio coding in the MDCT Domain

End-to-End Neural Audio Coding in the MDCT Domain

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International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Hyungseob Lim Jihyun Lee Byeong Hyeon Kim Inseon Jang Hong-Goo Kang Department of Electrical and Electronic Engineering Yonsei University Seoul South Korea Electronics and Telecommunications Research Institute Daejeon South Korea

Modern deep neural network (DNN)-based audio coding approaches utilize complicated non-linear functions (e.g., convolutional neural networks and non-linear activations), which leads to high complexity and memory usage. However, their decoded audio quality is still not much higher than that of signal processing-based legacy codecs. In this paper, we propose an effective frequency-domain neural audio coding paradigm that adopts the modified discrete cosine transform (MDCT) for analysis and synthesis and DNNs for the quantization of variables. It includes an efficient method to encode MDCT bins as well as a mechanism to adapt the quantization level of each bin. Our neural audio codec is trained in an end-to-end manner with the help of psychoacoustics-based perceptual loss, removing the burden of module-by-module fine-tuning. Experimental results show that our proposed model’s performance is comparable with the MP3 codec at around 64 and 48 kbps bit-rates for mono signals.

关键词： Codecs Quantization (signal) audio coding Frequency-domain analysis Neural networks Psychoacoustic models MONOS devices

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