Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V.

Other

Heinz Gerhauser

Karlheinz Brandenburg

Martin Dietz

Ernst Eberlein

J&uuml;rgen Herre

Harald Popp

Dieter Seitzer

Bodo Teichmann

Uwe Gbur

Jurgen Herre

Roland Plankenbuhler

Bernhard Grill

Rainer Perthold

Oliver Kunz

Karl-Heinz Brandenburg

Hartmut Schott

James David Johnston

Thomas Sporer

Schuyler Reynier Quackenbush

Albert Heuberger

Alexander Zink

Stefan Lipp

Markus Prosch

Maximilian Hofmann

Olaf Korte

Stephan Buchholz

Gerald Ulbricht

Thomas Von Der Gruen

Sabah Badri

Rainer Buchta

Ingmar Bretz

Martin Sieler

Rolf Kapust

Christian Schmidmer

Michael Keyhl

Ingo Rogalsky

Jurgen Zeller

Karl Heinz Brandenburg

Detlev Von Braumuller

Nuria Llombart-Juan

Roland Plankenbühler

Wolfgang Schildbach

Andre Heuer

Gunter Rohmer

Stefan Kohler

Roland Brand

Florian Oelmaier

A system for presenting information regarding an object provided within a plurality of information sources to a user or a presentation means depending on a location of the user or the presentation means includes a means for determining the location, a means for selecting the information to be presented depending on the determined location and on one or several pre-settable selection criteria defining an information source from the plurality of information sources, and a means for outputting the information to be presented.

Method and device fore reproducing information

In a method for coding an audio signal digitized at a low sampling rate to obtain time domain audio samples. A frequency domain representation of the time domain audio samples is produced. The frequency domain representation includes successive frequency lines. These frequency lines are grouped into a plurality of scale factor bands. The successive frequency lines in a scale factor band are coded with the same scale factor. A plurality of regions is formed by grouping the scale factor bands, wherein successive scale factor bands form a region within which all the scale factors are coded with the same number of bits, which is determined according to the largest scale factor of the region. The scale factors assigned to scale factor bands within the highest region that includes the higher frequency successive frequency lines are set to zero. The frequency lines in the highest region are coded using the zero-valued scale factors that correspond to a multiplication factor of 1. The scale factors for the highest region, however, are not coded. Thus, the bits that would be required for coding these zero-valued scale factors are saved and can be used for a finer quantization of the rest of the spectrum. Additionally, this coding method when applied to ISO/IEC 13818-3 as a low sampling rate modification thereof only requires minimal changes with respect to this Standard.

Process of low sampling rate digital encoding of audio signals

A method of coding stereo audio spectral values first carries out grouping of those values in scale factor bands, with which scale factors are associated. Sections are formed next, each comprising at least one scale factor band. The spectral values are coded within at least one section with a code book assigned to the section, out of a plurality of code books each with a code book number assigned to it, the number of the code book used being transmitted as side information to the coded stereo audio spectral values. At least one additional code book number is provided, which does not refer to a code book but shows information relevant to the section to which it is assigned. A method of decoding stereo audio spectral values which are partly coded by the intensity stereo process and which have side information uses the relevant information, showing the additional code book numbers, to cancel the existing coding of the stereo audio spectral values.

Process for coding and decoding stereophonic spectral values

In a method for estimating a non-linear characteristic of an amplifier, an input signal is provided, a reference signal is determined based on the power of the input signal, the input signal is amplified using said amplifier in order to produce an amplified signal, a measurement signal is determined based on the power of the amplified signal and respective samples of the measurement signal are associated with respective samples of the reference signal. A gain value of said amplifier for each sample is determined from the respective samples of the reference signal and the measurement signal. An operating range of the amplifier is divided into a plurality of power sections and an averaged gain value is produced for each power section. Thereafter, a measure of the deviation of the gain values associated with a power section from said averaged gain value is determined. Then, averaged gain values are rejected which have a measure of the deviation which exceeds a predetermined threshold. Finally, an interpolation is performed using averaged gain values which have not been rejected in order to produce an estimation of said non-linear characteristic of said amplifier.

System for estimating a non-linear characteristic of an amplifier

The present invention permits a combination of a scalable audio coder with the TNS technique. In a method for coding time signals sampled in a first sampling rate, second time signals are first generated whose sampling rate is smaller than the first sampling rate. The second time signals are then coded according to a first coding algorithm and written into a bit stream. The coded second time signals are, however, decoded again, and, like the first time signals, transformed into the frequency domain. From a spectral representation of the first time signals, TNS prediction coefficients are calculated. The transformed output signal of the coder/decoder with the first coding algorithm, like the spectral representation of the first time signal, undergoes a prediction over the frequency to obtain residual spectral values for both signals, though only the prediction coefficients calculated on the basis of the first time signals are used. These two signals are evaluated against each other. The evaluated residual spectral values are then coded by means of a second coding algorithm to obtain coded evaluated residual spectral values, which, together with the side information containing the calculated prediction coefficients, are written into the bit stream.

Method and a device for coding audio signals and a method and a device for decoding a bit stream

A method for coding or decoding an audio signal combines the advantages of TNS processing and noise substitution. A time-discrete audio signal is initially transformed to the frequency domain in order to obtain spectral values of the temporal audio signal. Subsequently, a prediction of the spectral values in relation to frequency is carried out in order to obtain spectral residual values. Within the spectral residual values, areas are detected encompassing spectral residual values with noise properties. The spectral residual values in the noise areas are noise-substituted, whereupon information concerning the noise areas and noise substitution is incorporated into side information pertaining to a coded audio signal. Thus, considerable bit savings in case of transient signals can be achieved.

Method for coding an audio signal

In the coding and decoding of stereo audio spectral values both the intensity stereo process and prediction are used in order to achieve high data compression. If intensity stereo coding is active in one section of scale factor bands, the prediction for the right channel in that range is deactivated, whereby the results of the prediction are not used to form the coded stereo audio spectral values. To allow further adaptation of the prediction for the right channel, the predictor of the right channel is fed with stereo audio spectral values for the channel, which again are intensity stereo decoded.

Coding and decoding of audio signals by using intensity stereo and prediction processes

Consumption measurement system for remote reading

Method for the cascaded coding and decoding of audio data

Method and device for determining the tonality of an audio signal

A digital adaptive transformation coding method for the transmission and/or torage of audio signals, specifically music signals, wherein N scanned values of the audio signal are transformed into M spectral coefficients, and the coefficients are split up into frequency groups, quantized and then coded. The quantized maximum value of each frequency group is used to define the coarse variation of the spectrum. The same number of bits is assigned to all values in a frequency group. The bits are assigned to the individual frequency groups as a function of the quantized maximum value present in the particular frequency group. A multi-signal processor system is disclosed which is specifically designed for implementation of this method.

Digital adaptive transformation coding method

Process for simultaneous transmission of signals from N signal sources

Apparatus for checking audio signal processing systems

Growing community of inventors

Waischenfeld, Germany

Heinz Gerhauser