During the last decade, CD-quality digital audio has essentially replaced analog audio. Emerging digital audio applications for network, wireless, and multimedia computing systems face a series of constraints such as reduced channel bandwidth, limited storage capacity, and low cost. These new applications have created a demand for high-quality digital audio delivery at low bit rates. In response to this need, considerable research has been devoted to the development of algorithms for perceptually transparent coding of high-fidelity (CD-quality) digital audio. As a result, many algorithms have been proposed, and several have now become international and/or commercial product standards. This paper reviews algorithms for perceptually transparent coding of CD-quality digital audio, including both research and standardization activities. This paper is organized as follows. First, psychoacoustic principles are described, with the MPEG psychoacoustic signal analysis model 1 discussed in some detail. Next, filter bank design issues and algorithms are addressed, with a particular emphasis placed on the modified discrete cosine transform, a perfect reconstruction cosine-modulated filter bank that has become of central importance in perceptual audio coding. Then, we review methodologies that achieve perceptually transparent coding of FM- and CD-quality audio signals, including algorithms that manipulate transform components, subband signal decompositions, sinusoidal signal components, and linear prediction parameters, as well as hybrid algorithms that make use of more than one signal model. These discussions concentrate on architectures and applications of those techniques that utilize psychoacoustic models to exploit efficiently masking characteristics of the human receiver. Several algorithms that have become international and/or commercial standards receive in-depth treatment, including the ISO/IEC MPEG family (-1, -2, -4), the Lucent Technologies PAC/EPAC/MPAC, the Dolby AC-2/AC-3, and the Sony ATRAC/SDDS algorithms. Then, we describe subjective evaluation methodologies in some detail, including the ITU-R BS.1116 recommendation on subjective measurements of small impairments. This paper concludes with a discussion of future research directions.

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Perceptual coding of digital audio

Disclosed herein are methods and systems for encoding digital watermarks into content signals. Also disclosed are systems and methods for detecting and/or verifying digital watermarks in content signals. According to one embodiment, a system for encoding of digital watermark information includes: a window identifier for identifying a sample window in the signal; an interval calculator for determining a quantization interval of the sample window; and a sampler for normalizing the sample window to provide normalized samples. According to another embodiment, a system for pre-analyzing a digital signal for encoding at least one digital watermark using a digital filter is disclosed. According to another embodiment, a method for pre-analyzing a digital signal for encoding digital watermarks comprises: (1) providing a digital signal; (2) providing a digital filter to be applied to the digital signal; and (3) identifying an area of the digital signal that will be affected by the digital filter based on at least one measurable difference between the digital signal and a counterpart of the digital signal selected from the group consisting of the digital signal as transmitted, the digital signal as stored in a medium, and the digital signal as played backed. According to another embodiment, a method for encoding a watermark in a content signal includes the steps of (1) splitting a watermark bit stream; and (2) encoding at least half of the watermark bit stream in the content signal using inverted instances of the watermark bit stream. Other methods and systems for encoding/decoding digital watermarks are also disclosed.

Optimization methods for the insertion, protection, and detection of digital watermarks in digitized data

A method for applying a digital watermark to a content signal is disclosed. In accordance with such a method, a watermarking key is identified. The watermarking key includes a binary sequence and information describing application of that binary sequence to the content signal. The digital watermark is then encoded within the content signal at one or more locations determined by the watermarking key.

Method and system for digital watermarking

A modular intelligent transportation system, comprising an environmentally protected enclosure, a system communications bus, a processor module, communicating with said bus, having a image data input and an audio input, the processor module analyzing the image data and/or audio input for data patterns represented therein, having at least one available option slot, a power supply, and a communication link for external communications, in which at least one available option slot can be occupied by a wireless local area network access point, having a communications path between said communications link and said wireless access point, or other modular components.

Modular intelligent transportation system

We report on experiments done in an attempt to improve the performance of a music similarity measure which we introduced in [2] The technique aims at comparing music titles on the basis of their global “timbre”, which has many applications in the field of Music Information Retrieval Such measures of timbre similarity have seen a growing interest lately, and every contribution (including ours) is yet another instantiation of the same basic pattern recognition architecture, only with different algorithm variants and parameters Most give encouraging results with a little effort, and imply that nearperfect results would just extrapolate by fine-tuning the algorithms’ parameters However, such systematic testing over large, inter-dependent parameter spaces is both difficult and costly, as it requires to work on a whole general meta-database architecture This paper contributes in two ways to the current state of the art We report on extensive tests over very many parameters and algorithmic variants, either already envisioned in the literature or not This leads to an improvement over existing algorithms of about 15% R-precision But most importantly, we describe many variants that surprisingly do not lead to any substancial improvement Moreover, our simulations suggest the existence of a “glass ceiling” at R-precision about 65% which cannot probably be overcome by pursuing such variations on the same theme

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Improving Timbre Similarity : How high’s the sky ?

Reference LCAV-CONF-2005-031 URL: www.aes.org Record created on 2005-10-07, modified on 2016-08-08

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The Reference Model Architecture for MPEG Spatial Audio Coding

Reference LCAV-CONF-2005-029 URL: www.aes.org Record created on 2005-10-07, modified on 2017-05-12

MPEG Spatial Audio Coding / MPEG Surround: Overview and Current Status

1 Philips Research Laboratories, 5656 AA, Eindhoven, The Netherlands jeroen.breebaart@philips.com 2 Fraunhofer Institute for Integrated Circuits IIS, 91058 Erlangen, Germany {hrr;pts}@iis.fraunhofer.de 3 Coding Technologies, 11352 Stockholm, Sweden {lv;kk}@codingtechnologies.com 4 Vast Audio, NSW 1430 Sydney, Australia craig@ee.usyd.edu.au 5 Philips Applied Technologies, 5616 LW Eindhoven, The Netherlands jeroen.koppens@philips.com

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Multi-Channel Goes Mobile: MPEG Surround Binaural Rendering

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.

Audio encoder, audio decoder and audio processor having a dynamically variable warping characteristic

In order to analyze an information signal, which has a series of blocks of information units, whereby a number of successive blocks of the series of blocks depicts an information entity, identification results for successive fingerprints are prepared (12) while using a series of fingerprints for the series of blocks, whereby an identification result depicts an association of a block of information units with a predetermined information entity. After this, at least two hypotheses are formed (14) from the identification results for the successive fingerprints. A first hypothesis is an assumption for the association of the series of blocks with a first information entity, and the second hypothesis is an assumption for the association of the series of blocks with the second information entity. Afterwards, different hypotheses are tested (16) in order to obtain a test result on the basis of which an assertion concerning the information signal is made (20). This results in obtaining a meaningful and reliable continuous-time analysis of an information signal.

Juergen Herre

Papers

The Reference Model Architecture for MPEG Spatial Audio Coding

MPEG Spatial Audio Coding / MPEG Surround: Overview and Current Status

Multi-Channel Goes Mobile: MPEG Surround Binaural Rendering

Audio encoder, audio decoder and audio processor having a dynamically variable warping characteristic

Device and method for analyzing an information signal