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Author

Stefan Bayer

Bio: Stefan Bayer is an academic researcher from Philips. The author has contributed to research in topics: Audio signal & Encoder. The author has an hindex of 18, co-authored 52 publications receiving 1191 citations.


Papers
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Patent
06 Dec 2012
TL;DR: In this paper, a signal analyzer for analyzing the audio signal is provided, which determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded message from a second encoding branch.
Abstract: An audio encoder for encoding an audio signal has a first coding branch, the first coding branch comprising a first converter for converting a signal from a time domain into a frequency domain Furthermore, the audio encoder has a second coding branch comprising a second time/frequency converter Additionally, a signal analyzer for analyzing the audio signal is provided The signal analyzer, on the hand, determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded signal from a second encoding branch On the other hand, the signal analyzer determines a time/frequency resolution to be applied by the converters when generating the encoded signals An output interface includes, in addition to the first encoded signal and the second encoded signal, a resolution information identifying the resolution used by the first time/frequency converter and used by the second time/frequency converter

128 citations

Proceedings ArticleDOI
19 Apr 2009
TL;DR: This new codec forms the basis of the reference model in the ongoing MPEG standardization activity for Unified Speech and Audio Coding, which results in a codec that exhibits consistently high quality for speech, music and mixed audio content.
Abstract: Traditionally, speech coding and audio coding were separate worlds. Based on different technical approaches and different assumptions about the source signal, neither of the two coding schemes could efficiently represent both speech and music at low bitrates. This paper presents a unified speech and audio codec, which efficiently combines techniques from both worlds. This results in a codec that exhibits consistently high quality for speech, music and mixed audio content. The paper gives an overview of the codec architecture and presents results of formal listening tests comparing this new codec with HE-AAC(v2) and AMR-WB+. This new codec forms the basis of the reference model in the ongoing MPEG standardization activity for Unified Speech and Audio Coding.

108 citations

Patent
30 Jun 2006
TL;DR: In this article, a controller is connected for providing the time-varying control signal, which depends on the audio signal, and the controller is introduced to an encoding processor having different encoding algorithms adapted to a specific signal pattern.
Abstract: An audio encoder, an audio decoder or an audio processor includes a filter for generating a filtered audio signal, the filter having a variable warping characteristic, the characteristic being controllable in response to a time-varying control signal, the control signal indicating a small or no warping characteristic or a comparatively high warping characteristic. Furthermore, a controller is connected for providing the time-varying control signal, which depends on the audio signal. The filtered audio signal can be introduced to an encoding processor having different encoding algorithms, one of which is a coding algorithm adapted to a specific signal pattern. Alternatively, the filter is a post-filter receiving a decoded audio signal.

95 citations

Journal Article
TL;DR: All aspects of this standardization eort are outlined, starting with the history and motivation of the MPEG work item, describing all technical features of the nal system, and further discussing listening test results and performance numbers which show the advantages of the new system over current state-of-the-art codecs.
Abstract: In early 2012 the ISO/IEC JTC1/SC29/WG11 (MPEG) nalized the new MPEG-D Unied Speech and Audio Coding standard The new codec brings together the previously separated worlds of general audio coding and speech coding It does so by integrating elements from audio coding and speech coding into a unied system The present publication outlines all aspects of this standardization eort, starting with the history and motivation of the MPEG work item, describing all technical features of the nal system, and further discussing listening test results and performance numbers which show the advantages of the new system over current state-of-the-art codecs

88 citations

Patent
06 Jul 2009
TL;DR: An apparatus for encoding comprises a first domain converter (510), a switchable bypass (50), a second domain Converter (410), a first processor (420) and a second processor (520) to obtain an encoded audio signal having different signal portions represented by coded data in different domains, which have been coded by different coding algorithms as discussed by the authors.
Abstract: An apparatus for encoding comprises a first domain Converter (510), a switchable bypass (50), a second domain Converter (410), a first processor (420) and a second processor (520) to obtain an encoded audio signal having different signal portions represented by coded data in different domains, which have been coded by different coding algorithms. Corresponding decoding stages in the decoder together with a bypass for bypassing a domain Converter allow the generation of a decoded audio signal with high quality and low bit rate.

85 citations


Cited by
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Book ChapterDOI
02 Jul 2018
TL;DR: SiSEC 2018 as mentioned in this paper was focused on audio and pursued the effort towards scaling up and making it easier to prototype audio separation software in an era of machine-learning-based systems.
Abstract: This paper reports the organization and results for the 2018 community-based Signal Separation Evaluation Campaign (SiSEC 2018). This year’s edition was focused on audio and pursued the effort towards scaling up and making it easier to prototype audio separation software in an era of machine-learning based systems. For this purpose, we prepared a new music separation database: MUSDB18, featuring close to 10 h of audio. Additionally, open-source software was released to automatically load, process and report performance on MUSDB18. Furthermore, a new official Python version for the BSS Eval toolbox was released, along with reference implementations for three oracle separation methods: ideal binary mask, ideal ratio mask, and multichannel Wiener filter. We finally report the results obtained by the participants.

250 citations

Patent
30 Dec 2008
TL;DR: In this article, a linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal.
Abstract: The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

170 citations

Journal ArticleDOI
TL;DR: This paper provides an overview of the MPEG-H 3D Audio project and technology and an assessment of the system capabilities and performance.
Abstract: The science and art of Spatial Audio is concerned with the capture, production, transmission, and reproduction of an immersive sound experience. Recently, a new generation of spatial audio technology has been introduced that employs elevated and lowered loudspeakers and thus surpasses previous ‘surround sound’ technology without such speakers in terms of listener immersion and potential for spatial realism. In this context, the ISO/MPEG standardization group has started the MPEG-H 3D Audio development effort to facilitate high-quality bitrate-efficient production, transmission and reproduction of such immersive audio material. The underlying format is designed to provide universal means for carriage of channel-based, object-based and Higher Order Ambisonics based input. High quality reproduction is provided for many output formats from 22.2 and beyond down to 5.1, stereo and binaural reproduction—independently of the original encoding format, thus overcoming the incompatibility between various 3D formats. This paper provides an overview of the MPEG-H 3D Audio project and technology and an assessment of the system capabilities and performance.

147 citations

Patent
Hyen O Oh1, Yang Won Jung1
29 Jul 2009
TL;DR: In this article, an apparatus for processing an audio signal and method thereof is described, which includes receiving, by an audio processing apparatus, a signal including a first data of a first block encoded with rectangular coding scheme and a second data of the second block encoded in non-rectangular coding scheme.
Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part.

136 citations