Abstract: A loudspeaker diaphragm includes a tapering edge region. Thus, it is possible to prevent or considerably reduce wave reflection and uncontrolled wave propagation caused thereby in a cheap and effective manner.

Abstract: An apparatus for scalable encoding a spectrum of a signal including audio and/or video information, with the spectrum comprising binary spectral values, includes a means for generating a first sub-scaling layer and a second sub-scaling layer in addition to a means for forming the encoded signal, with the means for forming being implemented so as to include the first sub-scaling layer and the second sub-scaling layer into the encoded signal that the first and the second sub-scaling layer are separately decodable from each other. In contrast to a full-scaling layer, a sub-scaling layer includes only the bits of a certain order of a part of the binary spectral values in the band, so that, by additionally decoding a sub-scaling layer, a more finely controllable and a more finely scalable precision gain may be achieved.

Abstract: A device for synchronizing an audio signal with a film comprising frames having imprinted time codes. A detector detects the imprinted time code for the sequence of frames to obtain a detected sequence of time codes. A time code generator generates a sequence of synthesis time codes based on an initial value. A decoder decodes a time code of the detected sequence of time codes obtain the initial value. A comparator compares a detected time code and a corresponding synthesis time code and, when a phase deviation is found to be above a deviation threshold value, a time code manipulator manipulates the synthesis time code for this frame in so far that it is changed with regard to its temporal length. An audio processor provides the samples of the audio signal associated with this frame in a time-controlled way in response to a detection of the synthesis time code for a frame.

Abstract: A time-discrete audio signal is processed to provide a quantization block with quantized spectral values. Furthermore, an integer spectral representation is generated from the time-discrete audio signal using an integer transform algorithm. The quantization block having been generated using a psychoacoustic model is inversely quantized and rounded to then form a difference between the integer spectral values and the inversely quantized rounded spectral values. The quantization block alone provides a lossy psychoacoustically coded/decoded audio signal after the decoding, whereas the quantization block, together with the combination block, provides a lossless or almost lossless coded and again decoded audio signal in the decoding. By generating the differential signal in the frequency domain, a simpler coder/decoder structure results.

Abstract: For the reduction of the rounding error, a first and a second non-integer input value are provided and combined, for example by addition, in non-integer state to obtain a non-integer result value which is rounded and added to a third input value. Thus, the rounding error may be reduced at an interface between two rotations divided into lifting steps or between a first rotation divided into lifting steps and a first lifting step of a subsequent multi-dimensional lifting sequence.

Abstract: Analyzing an analysis time signal that has been generated from encoding and decoding and original time signal according to an encoding algorithm. The encoding block raster underlying the analysis time signal used by the encoding algorithm is determined. The analysis time signal is converted from its timely representation of analysis spectral coefficients to a spectral representation by using the established encoding block raster. At least two analysis spectral coefficients are grouped. The greatest common divisor of the analysis spectral coefficients are calculated, corresponding to the quantization step width used when quantizing the encoding algorithm or an integer multiple of it. In the case of an audio signal, the scale factor can easily be established for this group of spectral coefficients, i.e., for a scale factor band, from the quantization step width. All parameters used for the quantization of the original time signal are known; full iteration loops need not be performed.

Abstract: An apparatus for generating a control signal for a film event system is described for synchronizing film events with an image reproduction, wherein a film comprises film information applied in a time sequence, and the apparatus comprises a means for storing the film information, wherein a time scale is associated to the stored film information, a means for receiving a section read from the film, a means for comparing the read section to the stored film information and a means for determining the control signal based on the comparison and the time scale.

Abstract: When processing a signal having a sequence of discrete values, wherein there is a first frequency range, in which the signal has a high energy, and wherein there is a second frequency range, in which the signal has a low energy, the sequence of discrete values is first manipulated to obtain a sequence of manipulated values, so that at least one of the manipulated values is non-integer. Then the sequence of manipulated values is rounded to obtain a sequence of manipulated values. The rounding is formed to effect a spectral shaping of a generated rounding error so that a spectrally shaped rounding error has a higher energy in the first frequency range than in the second frequency range. By spectrally shaping the rounding error so that the rounding error does not have any energy either in the storage areas where there is no signal energy, an especially efficient coding is obtained particularly in connection with a lossless coding context.

Abstract: A loudspeaker system including a plurality of sonic converters fixed in or to the rear of sound passage openings of a carrier plate is disclosed. The carrier plate is configured in the form of a wall plate for interior spaces or outside facades of buildings and the sonic converters are integrated into the carrier plate. The loudspeaker system or loudspeaker wall plate, respectively, permit the realization of an optimum acoustic irradiation of a space without aesthetically annoying loudspeaker boxes.

Abstract: For converting first and second blocks of discrete values into a transformed representation, the first block is transformed according to a first transformation rule and then rounded. Then, the rounded transformed values are summed with the second block of original discrete values, to then process the summation result according to a second transformation rule. The output values of the transformation via the second transformation rule are again rounded and then subtracted from the original discrete values of the first block of discrete values to obtain a block of integer output values of the transformed representation. By this multi-dimensional lifting scheme, a lossless integer transformation is obtained, which can be reversed by applying the same transformation rule, but with different signs in summation and subtraction, respectively, so that an inverse integer transformation can also be obtained.

Abstract: For a level correction in a wave field synthesis system having a wave field synthesis module and an array of loudspeakers for providing sound to a presentation region, a correction value which is based on a set amplitude state in a presentation region is determined, the set amplitude state depending on a position of the virtual source or a type of the virtual source, and the actual amplitude state in the presentation region depending on the component signals for the loudspeakers due to the virtual source. The correction value determined is fed to a manipulator manipulating the audio signal associated to the virtual source before feeding to the wave field synthesis module, or the component signals for the individual loudspeakers due to the virtual source are manipulated to reduce a deviation between a set amplitude state and an actual amplitude state at one point or several points in the presentation region.

Abstract: In a wave field synthesis apparatus for driving an array of loudspeakers with drive signals, the loudspeakers being arranged at different defined positions, a drive signal for a loudspeaker being based on an audio signal associated with a virtual source having a virtual position with reference to the loudspeaker array and on the defined position of the loudspeaker, at first relevant loudspeakers of the loudspeaker array are determined on the basis of the position of the virtual source, a predefined listener position, and the defined positions of the loudspeakers, so that artifacts due to loudspeaker signals moving opposite to a direction from the virtual source to the predefined listener position are reduced.

Abstract: For reducing Doppler artifacts in the wave-field synthesis due to delay changes from one time to a second time, first, the delay for the first time and the delay for the second time are determined. Then, a value of an audio signal delayed by the first delay for the current time and the value for the audio signal delayed by the second delay for the current time are determined. Then, the first value is weighted by a first weighting factor and a second value is averaged with a second weighting factor, whereupon the two weighted values are added up to obtain a discrete value for the current time of the component in a loudspeaker signal for a loudspeaker based on a virtual source. Thus, by knowing a delay present at a later time, panning is obtained from a delay to a subsequent delay, which reduces undesired Doppler artifacts.

Abstract: A device for synchronizing an audio signal with a film comprising frames, wherein each frame comprises an imprinted time code, includes means for detecting the imprinted time code for the sequence of frames in order to obtain a detected sequence of time codes. Further, a time code generator is provided which is implemented to generate a sequence of synthesis time codes based on an initial value. A decoder is further provided to decode a time code of the detected sequence of time codes in order to provide the initial value for the time code generator. A detected time code and a corresponding synthesis time code are compared in order to then, when a phase deviation is found to be above a deviation threshold value, manipulate the synthesis time code for this frame in so far that it is changed with regard to its temporal length.

Abstract: A system for evaluating the quality of an audio test signal derived from an audio reference signal by coding and decoding, said audio test signal and said audio reference signal each comprising a plurality of channels, comprises a unit for converting the audio reference signal into a first audio reference sum signal at a first reference point and into a second audio reference sum signal at a second reference point and for converting the audio test signal into a first audio test sum signal at the first reference point and into a second audio test sum signal at the second reference point, the audio reference sum signals and the audio test sum signals at the first and second reference points being a superposition of the respective channels, which can be emitted by a plurality of loudspeakers, weighted with a respective transfer function between the respective loudspeaker and the reference point in question, and a unit for evaluating the quality of the audio test sum signals while taking into consideration the au

Abstract: For analyzing an information signal consisting of a superposition of partial signals, wherein a partial signal originates from an individual source, the information signal is first decomposed into several component signals. Then one or more features are calculated for each component signal, wherein the feature(s) is/are defined so that it/they provide(s) a statement on an information content of the component signal. Finally, an association of the component signals with at least two subspaces is performed on the basis of the features calculated for the component signals. By feature extraction on the component signal level, an efficient subspace decomposition which is meaningful with respect to a present audio scene is achieved.

Abstract: The apparatus for determining an impulse response in an environment in which a speaker and a microphone are placed works using an audio signal. Means for spectrally coloring a test signal, which preferably is a pseudonoise signal, works using a psychoacoustic masking threshold of the audio signal to obtain a colored test signal, which is embedded in the audio signal to obtain a measuring signal, which can be fed to the speaker. Means for determining the impulse response preferably performs a cross-correlation of a reaction signal received via the microphone from the environment and the test signal or the colored test signal. With this, an impulse response of an environment may also be determined during the presentation of an audio piece to provide an optimal description of environment for a wave-field synthesis.

Abstract: An apparatus for suppressing feedback in an environment where a microphone and a loudspeaker are located, comprises a means for embedding a test signal into a loudspeaker signal, a microphone signal or a modified microphone signal, preferably by using a psychoacoustic masking threshold by using a pseudo-noise test signal, a means for determining a characteristic of a transmission channel in the environment between the loudspeaker and the microphone by using the embedded test signal and the microphone signal, a filter for filtering the loudspeaker signal to obtain a filtered loudspeaker signal, wherein the filter is adaptable to be adapted with regard to its filter characteristic to the characteristic of the transmission channel by the means for determining, as well as a means for subtracting the filtered loudspeaker signal from the microphone signal to obtain the modified microphone signal, in which the feedback is reduced due to the loudspeaker signal.

Abstract: An audio reproduction system is divided into a central wave-field synthesis module and a plurality of loudspeaker modules disposed in a distributed way, wherein synthesis signals for the individual loudspeakers as well as corresponding channel information associated to the synthesis signals are calculated in the central wave-field synthesis module. The synthesis signals for a loudspeaker as well as associated channel information will then be transmitted to respective loudspeaker modules via a transmission path, wherein every loudspeaker module obtains the synthesis signals and associated channel information intended for the loudspeaker associated to the loudspeaker module. A distributed audio rendering and digital/analog converting takes place in the loudspeaker module to generate the actually analog loudspeaker signals in a distributed way in spatial proximity to every loudspeaker.

Abstract: An apparatus for analyzing an analysis time signal that has been generated from encoding and decoding an original time signal according to an encoding algorithm first, wherein first the encoding block raster underlying the analysis time signal used by the encoding algorithm is determined. Thereupon, the analysis time signal will be converted from its timely representation comprising a plurality of analysis spectral coefficients, to a spectral representation by using the established encoding block raster. Then, at least two analysis spectral coefficients or at least two spectral coefficients derived from the analysis spectral coefficients by multiplication of an encoding amplification factor or by multiplication with a compression function are grouped.