Nikolaus Rettelbach

Bio: Nikolaus Rettelbach is an academic researcher. The author has contributed to research in topics: Speech coding & Audio signal. The author has an hindex of 11, co-authored 13 publications receiving 695 citations.

Multi-resolution switched audio encoding/decoding scheme

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

Unified speech and audio coding scheme for high quality at low bitrates

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.

MPEG Unified Speech and Audio Coding - The ISO/MPEG Standard for High-Efficiency Audio Coding of All Content Types

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

Low bitrate audio encoding/decoding scheme having cascaded switches

Abstract: An audio encoder comprises a first information sink oriented encoding branch such as a spectral domain encoding branch, a second information source or SNR oriented encoding branch such as an LPC-domain encoding branch, and a switch for switching between the first encoding branch and the second encoding branch, wherein the second encoding branch comprises a converter into a specific domain different from the spectral domain such as an LPC analysis stage generating an excitation signal, and wherein the second encoding branch furthermore comprises a specific domain coding branch such as LPC domain processing branch, and a specific spectral domain coding branch such as LPC spectral domain processing branch, and an additional switch for switching between the specific domain coding branch and the specific spectral domain coding branch An audio decoder comprises a first domain decoder such as a spectral domain decoding branch, a second domain decoder such as an LPC domain decoding branch for decoding a signal such as an excitation signal in the second domain, and a third domain decoder such as an LPC-spectral decoder branch and two cascaded switches for switching between the decoders

A Novel Scheme for Low Bitrate Unified Speech and Audio Coding – MPEG RM0

Abstract: Coding of speech signals at low bitrates, such as 16 kbps, has to rely on an efficient speech reproduction model to achieve reasonable speech quality. However, for audio signals not fitting to the model this approach generally fails. On the other hand, generic audio codecs, designed to handle any kind of audio signal, tend to show unsatisfactory results for speech signals, especially at low bitrates. To overcome this, a process was initiated by ISO/MPEG, aiming to standardize a new codec with consistent high quality for speech, music and mixed content over a broad range of bitrates. After a formal listening test evaluating several proposals MPEG has selected the best performing codec as the reference model for the standardization process. This paper describes this codec in detail and shows that the new reference model reaches the goal of consistent high quality for all signal types.

MPEG-H 3D Audio—The New Standard for Coding of Immersive Spatial Audio

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.

Method and an apparatus for processing an audio signal

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.

Multi-resolution switched audio encoding/decoding scheme

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

A harmonic bandwidth extension method for audio codecs

Abstract: Today's efficient audio codecs for low bitrate application scenarios often rely on parametric coding of the upper frequency band portion of a signal while the lower frequency band portion of the same is conveyed by a waveform preserving coding method. At the decoder, the upper frequency signal is approximated from the lower frequency data using the upper frequency band parameters. However, commonly used methods of bandwidth extension almost inevitably suffer from a sensation of unpleasant roughness, which is especially present for tonal music items. In this paper we expose the origin of the roughness and propose a bandwidth extension method, which does not introduce roughness into the reconstructed audio signal. A listening test demonstrates the advantage of the proposed method compared to a standard bandwidth extension.

MPEG-H Audio—The New Standard for Universal Spatial / 3D Audio Coding

Abstract: Recently, a new generation of spatial audio formats were introduced that include elevated loudspeakers and surpass traditional surround sound formats, such as 5.1, in terms of spatial realism. To facilitate high-quality bitrate-efficient distribution and flexible reproduction of 3D sound, the MPEG standardization group recently started the MPEG-H Audio Coding development for the universal carriage of encoded 3D sound from channel-based, object-based and HOA-based input. High quality reproduction is supported 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 incompatibility between various 3D formats. The paper describes the current status of the standardization project and provides an overview of the system architecture, its capabilities and performance.