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17th Annual Conference of the International-speech-Communication-Association (INTERspeech 2016)

作者： Korse, Srikanth Jaehnel, Tobias Baeckstroem, Tom Fraunhofer IIS Erlangen Germany Friedrich Alexander Univ Int Audio Labs Erlangen Germany

ISBN: (纸本)9781510833135

speech and audio codecs model the overall shape of the signal spectrum using envelope models. In speech coding the predominant approach is linear predictive coding, which offers high coding efficiency at the cost of computational complexity and a rigid systems design. audio codecs are usually based on scale factor bands, whose calculation and coding is simple, but whose coding efficiency is lower than that of linear prediction. In the current work we propose an entropy coding approach for scale factor bands, with the objective of reaching the same coding efficiency as linear prediction, but simultaneously retaining a low computational complexity. The proposed method is based on quantizing the distribution of spectral mass using beta distributions. Our experiments show that the perceptual quality achieved with the proposed method is similar to that of linear predictive models with the same bit rate, while the design simultaneously allows variable bit-rate coding and can easily be scaled to different sampling rates. The algorithmic complexity of the proposed method is less than one third of traditional multi-stage vector quantization of linear predictive envelopes.

关键词： spectral envelope arithmetic coding speech and audio coding

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End-to-end Optimization of Source Models for speech and audio coding Using a Machine Learning Framework 20

End-to-end Optimization of Source Models for Speech and Audi...

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Interspeech Conference

作者： Backstrom, Tom Aalto Univ Dept Signal Proc & Acoust Espoo Finland

speech coding is the most commonly used application of speech processing. Accumulated layers of improvements have however made codecs so complex that optimization of individual modules becomes increasingly difficult. This work introduces machine learning methodology to speech and audio coding, such that we can optimize quality in terms of overall entropy. We can then use conventional quantization, coding and perceptual models without modification such that the codec adheres to conventional requirements on algorithmic complexity, latency and robustness to packet loss. Experiments demonstrate that end-to-end optimization of quantization accuracy of the spectral envelope can be used for a lossless reduction in bitrate of 0.4 kbits/s.

关键词： speech and audio coding end-to-end optimization speech source modeling

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Postfiltering with Complex Spectral Correlations for speech and audio coding 19

Postfiltering with Complex Spectral Correlations for Speech ...

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19th Annual Conference of the International-speech-Communication-Association (INTERspeech 2018)

作者： Das, Sneha Backstrom, Tom Aalto Univ Dept Signal Proc & Acoust Espoo Finland

ISBN: (纸本)9781510872219

State-of-the-art speech codecs achieve a good compromise between quality, bitrate and complexity. However, retaining performance outside the target bitrate range remains challenging. To improve performance, many codecs use pre- and post-filtering techniques to reduce the perceptual effect of quantization-noise. In this paper, we propose a postfiltering method to attenuate quantization noise which uses the complex spectral correlations of speech signals. Since conventional speech codecs cannot transmit information with temporal dependencies as transmission errors could result in severe error propagation, we model the correlation offline and employ them at the decoder, hence removing the need to transmit any side information. Objective evaluation indicates an average 4 dB improvement in the perceptual SNR of signals using the context based post-filter. with respect to the noisy signal, and an average 2 dB improvement relative to the conventional Wiener filter. These results are confirmed by an improvement of up to 30 MUSHRA points in a subjective listening test.

关键词： speech and audio coding noise reduction temporal correlation post-filtering

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End-to-end Optimized Multi-stage Vector Quantization of Spectral Envelopes for speech and audio coding 22

End-to-end Optimized Multi-stage Vector Quantization of Spec...

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Interspeech Conference

作者： Vali, Mohammad Hassan Backstrom, Tom Aalto Univ Dept Signal Proc & Acoust Espoo Finland

ISBN: (纸本)9781713836902

Spectral envelope modeling is an instrumental part of speech and audio codecs, which can be used to enable efficient entropy coding of spectral components. Overall optimization of codecs, including envelope models, has however been difficult due to the complicated interactions between different modules of the codec. In this paper, we study an end-to-end optimization methodology to optimize all modules in a codec integrally with respect to each other while capturing all these complex interactions with a global loss function. For the quantization of the spectral envelope parameters with a fixed bitrate, we use multistage vector quantization which gives high quality, but yet has a computational complexity which can be realistically applied in embedded devices. The obtained results demonstrate benefits in terms of PESQ and PSNR in comparison to the 3GPP EVS, as well as our recently proposed PyAWNeS codecs.

关键词： speech and audio coding spectral envelope modeling multi-stage vector quantization end-to-end optimization

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CANDIDATE PROPOSAL FOR ITU-T SUPER-WIDEBAND speech and audio coding

CANDIDATE PROPOSAL FOR ITU-T SUPER-WIDEBAND SPEECH AND AUDIO...

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IEEE International Conference on Acoustics, speech and Signal Processing

作者： Geiser, Bernd Krueger, Hauke Loellmann, Heinrich W. Vary, Peter Zhang, Deming Wan, Hualin Li, Hai Ting Zhang, Li Bin Univ Aachen Rhein Westfal TH Aachen Inst Commun Syst & Data Proc I D Aachen Germany Huawei Technol Co Lt Beijing Peoples R China

ISBN: (纸本)9781424423538

This paper describes the speech and audio codec that has been submitted to ITU-T by Huawei and ETRI as a candidate for the upcoming super-wideband and stereo extensions of Rec. G.729.1 and G.718. The core codec in the current implementation is G.729.1 and the encoded frequency range is increased from 7 kHz to 14 kHz. Therefore, the maximum bit rate is raised from 32 kbit/s to 64 kbit/s by adding five bitstream layers. A comprehensive overview of the codec is presented with a focus on the mono coding components. The results of the listening tests that have been conducted during the ITU-T qualification phase are summarized. The proposed codec passes all quality requirements for mono input signals.

关键词： speech and audio coding ITU standardization

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Dithered Quantization for Frequency-Domain speech and audio coding 19

Dithered Quantization for Frequency-Domain Speech and Audio ...

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19th Annual Conference of the International-speech-Communication-Association (INTERspeech 2018)

作者： Backstrom, Tom Fischer, Johannes Das, Sneha Aalto Univ Dept Signal Proc & Acoust Espoo Finland Friedrich Alexander Univ FAU Int Audio Labs Erlangen Erlangen Germany

ISBN: (纸本)9781510872219

A common issue in coding speech and audio in the frequency domain, which appears with decreasing bitrate, is that quantization levels become increasingly sparse. With low accuracy, high-frequency components are typically quantized to zero, which leads to a muffled output signal and musical noise. Bandwidth extension and noise-filling methods attempt to treat the problem by inserting noise of similar energy as the original signal, at the cost of low signal to noise ratio. Dithering methods however provide an alternative approach, where both accuracy and energy are retained. We propose a hybrid coding approach where low-energy samples are quantized using dithering, instead of the conventional uniform quantizer. For dithering, we apply 1 bit quantization in a randomized sub-space. We further show that the output energy can be adjusted to the desired level using a scaling parameter. Objective measurements and listening tests demonstrate the advantages of the proposed methods.

关键词： speech and audio coding dithering noise filling perceptual coding 1 bit quantization

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Lattice Vector Quantization Applied to speech and audio coding

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ZTE Communications 2012年第2期10卷 25-33页

作者： Minjie Xie ZTE USA Inc. Richardson TX 75080 USA

Lattice vector quantization （LVQ） has been used for real-time speech and audio coding systems. Compared with conventional vector quantization, LVQ has two main advantages： It has a simple and fast encoding process, and it significantly reduces the amount of memory required. Therefore, LVQ is suitable for use in low-complexity speech and audio coding. In this paper, we describe the basic concepts of LVQ and its advantages over conventional vector quantization. We also describe some LVQ techniques that have been used in speech and audio coding standards of international standards developing organizations （SDOs）.

关键词： Vector quantization lattice vector quantization speech and audio coding transform coding

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Advances in speech and audio Processing and coding 6

Advances in Speech and Audio Processing and Coding

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6th International Conference on Information, Intelligence, Systems and Applications (IISA)

作者： Spanias, Andreas Arizona State Univ SenSIP Ctr Sch ECEE Tempe AZ 85287 USA

ISBN: (纸本)9781467393119

This plenary session will cover speech processing research advances with the emphasis on speech and audio coding methods. In the session, we will discuss the fundamental principles, techniques, and algorithms used in current coding applications including a summary of codecs for telecommunication standards. The session will start with a discussion on: the basic speech representation methods, the performance measures used to evaluate coded speech, and the role of the standards. Brief algorithm descriptions include: ADPCM, sub-band coding, adaptive transform coding, sinusoidal transform coding (STC), linear predictive coding (LPC), and analysis-by-synthesis LPC (sparse excitation, code excited LPC, and ACELP). The presentation will feature audio, and computer demonstrations of recent speech coding standards including voice-over IP algorithms. The plenary session will also cover wideband audio standards such as MPEG audio and other layers (e.g., MP3, AAC). Recent algorithms will also be described including the following: Variable-Rate Multimode Wideband (VMR-WB), Speex, G722.1, OGG Vorbis 2012, iLBC, SELT, SILK, Opus 2013, Qualcomm wideband 5G codecs. At the end of the session, we will cover briefly recent applications that use voice features for detecting speech pathologies, and also discuss how long-term speech parameters can be used as predictors of other diseases such as tremors, Alzheimer's etc.

关键词： speech and audio coding Standardized Codecs

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Advances in speech and audio Processing and coding

Advances in Speech and Audio Processing and Coding

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International Conference on Information, Intelligence, Systems and Applications

作者： Andreas Spanias SenSIP Center School of ECEE Arizona State University Tempe AZ 85287. USA

ISBN: (纸本)9781467393126

关键词： speech and audio coding Standardized Codecs

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Anthropomorphic coding of speech and audio: A model inversion approach

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EURASIP JOURNAL ON APPLIED SIGNAL PROCESSING 2005年第9期2005卷 1334-1349页

作者： Feldbauer, C Kubin, G Kleijn, WB Graz Univ Technol Signal Proc & Speech Commun Lab A-8010 Graz Austria Royal Inst Technol KTH Dept Signal Sensors & Syst S-10044 Stockholm Sweden

Auditory modeling is a well-established methodology that provides insight into human perception and that facilitates the extraction of signal features that are most relevant to the listener. The aim of this paper is to provide a tutorial on perceptual speech and audio coding using an invertible auditory model. In this approach, the audio signal is converted into an auditory representation using an invertible auditory model. The auditory representation is quantized and coded. Upon decoding, it is then transformed back into the acoustic domain. This transformation converts a complex distortion criterion into a simple one, thus facilitating quantization with low complexity. We briefly review past work on auditory models and describe in more detail the components of our invertible model and its inversion procedure, that is, the method to reconstruct the signal from the output of the auditory model. We summarize attempts to use the auditory representation for low-bit-rate coding. Our approach also allows the exploitation of the inherent redundancy of the human auditory system for the purpose of multiple description (joint source-channel) coding.

关键词： speech and audio coding auditory representation auditory model inversion auditory synthesis perceptual domain coding multiple description coding

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