Abstract: A system and a corresponding method for assisting selective hearing are provided. The system includes a detector for detecting an audio source signal portion of one or more audio sources by using at least two received microphone signals of a hearing environment. In addition, the system includes a position determiner for allocating position information to each of the one or more audio sources. In addition, the system includes an audio type classifier for assigning an audio source signal type to the audio source signal portion of each of the one or more audio sources. In addition, the system includes a signal portion modifier for varying the audio source signal portion of at least one audio source of the one or more audio sources depending on the audio signal type of the audio source signal portion of the at least one audio source so as to obtain a modified audio signal portion of the at least one audio source. In addition, the system includes a signal generator.

Abstract: A device for determining room-optimized transfer functions for a listening room serving for room-optimized post-processing of audio signals in spatial production, is configured to analyze room acoustics of the listening room and to determine, based on the analysis of the room acoustics, the room-optimized transfer functions for the listening room where the spatial reproduction by means of a binaural close-range sound transducer is to take place. The spatial reproduction of the audio signals by means of the binaural close-range sound transducer may then be emulated using known head-related transfer functions und using the room-optimized transfer functions, wherein a room to be synthesized may be emulated based on the head-related transfer functions, and wherein the listening room may be emulated based on the room-optimized transfer functions.

Abstract: A device for determining room-optimized transfer functions for a listening room serving for room-optimized post-processing of audio signals in spatial production, is configured to analyze room acoustics of the listening room and to determine, based on the analysis of the room acoustics, the room-optimized transfer functions for the listening room where the spatial reproduction by means of a binaural close-range sound transducer is to take place. The spatial reproduction of the audio signals by means of the binaural close-range sound transducer may then be emulated using known head-related transfer functions und using the room-optimized transfer functions, wherein a room to be synthesized may be emulated based on the head-related transfer functions, and wherein the listening room may be emulated based on the room-optimized transfer functions.

Abstract: Techniques for introducing information into a data stream first obtains the spectral values of the short-term spectrum of the audio signal. Separately, information to be introduced are combined with a spread sequence obtaining a spread information signal, whereupon a spectral representation of the spread information is generated, then weighted with an established psychoacoustic maskable noise energy to generate a weighted information signal, wherein energy of the introduced information is substantially equal to or below the psychoacoustic masking threshold. The weighted information signal and the spectral values of the short-term spectrum of the audio signal are then summed and afterwards processed again to obtain a processed data stream including audio information and information to be introduced.

Abstract: Techniques for introducing information into a data stream first obtains the spectral values of the short-term spectrum of the audio signal. Separately, information to be introduced are combined with a spread sequence obtaining a spread information signal, whereupon a spectral representation of the spread information is generated, then weighted with an established psychoacoustic maskable noise energy to generate a weighted information signal, wherein energy of the introduced information is substantially equal to or below the psychoacoustic masking threshold. The weighted information signal and the spectral values of the short-term spectrum of the audio signal are then summed and afterwards processed again to obtain a processed data stream including audio information and information to be introduced.

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: An inventive method for introducing information into a data stream including data about spectral values representing a short-term spectrum of an audio signal first performs a processing of the data stream to obtain the spectral values of the short-term spectrum of the audio signal. Apart from that, the information to be introduced are combined with a spread sequence to obtain a spread information signal, whereupon a spectral representation of the spread information is generated which will then be weighted with an established psychoacoustic maskable noise energy to generate a weighted information signal, wherein the energy of the introduced information is substantially equal to or below the psychoacoustic masking threshold. The weighted information signal and the spectral values of the short-term spectrum of the audio signal will then be summed and afterwards processed again to obtain a processed data stream including both audio information and information to be introduced.

Abstract: An integer transform, which provides integer output values, carries out the TDAC function of a MDCT in the time domain before the forward transform. In overlapping windows, this results in a Givens rotation which may be represented by lifting matrices, wherein time-discrete sampled values of an audio signal may at first be summed up on a pair-wise basis to build a vector so as to be sequentially provided with a lifting matrix. After each multiplication of a vector by a lifting matrix, a rounding step is carried out such that, on the output-side, only integers will result. By transforming the windowed integer sampled value with an integer transform, a spectral representation with integer spectral values may be obtained. The inverse mapping with an inverse rotation matrix and corresponding inverse lifting matrices results in an exact reconstruction.

Abstract: An inventive method for introducing information into a data stream including data about spectral values representing a short-term spectrum of an audio signal first performs a processing of the data stream to obtain the spectral values of the short-term spectrum of the audio signal. Apart from that, the information to be introduced are combined with a spread sequence to obtain a spread information signal, whereupon a spectral representation of the spread information is generated which will then be weighted with an established psychoacoustic maskable noise energy to generate a weighted information signal, wherein the energy of the introduced information is substantially equal to or below the psychoacoustic masking threshold. The weighted information signal and the spectral values of the short-term spectrum of the audio signal will then be summed and afterwards processed again to obtain a processed data stream including both audio information and information to be introduced.

Abstract: An inventive method for introducing information into a data stream including data about spectral values representing a short-term spectrum of an audio signal first performs a processing of the data stream to obtain the spectral values of the short-term spectrum of the audio signal. Apart from that, the information to be introduced are combined with a spread sequence to obtain a spread information signal, whereupon a spectral representation of the spread information is generated which will then be weighted with an established psychoacoustic maskable noise energy to generate a weighted information signal, wherein the energy of the introduced information is substantially equal to or below the psychoacoustic masking threshold. The weighted information signal and the spectral values of the short-term spectrum of the audio signal will then be summed and afterwards processed again to obtain a processed data stream including both audio information and information to be introduced.

Abstract: An apparatus for compiling a test comprises a database having a plurality of test tasks stored therein, each test task being associated with a task type, means for selecting test tasks from the database to obtain a multitude of selected test tasks, and means for outputting the selected test tasks of the test to a user. The means for selecting test tasks comprises means for selecting, for a task type, at least one test task from the database and for taking the selected test task over to the multitude of selected test tasks if a test task for the task type is available in the database, and an exception-handling logic configured to search the database, for a task type for which no test task is available in the database, for a replacement test task according to a given replacement rule and take same over to the multitude of selected tasks.

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 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: In a method for generating an identifier for an audio signal including a tone generated by an instrument, a discrete amplitude-time representation of the audio signal is generated at first, wherein the amplitude-time representation, for a plurality of subsequent points in time, comprises a plurality of subsequent amplitude values, wherein a point in time is associated to each amplitude value. Subsequently, an identifier for the audio signal is extracted from the amplitude-time representation. An instrument database is formed from several identifiers for several audio signals including tones of several instruments. By means of a test identifier for an audio signal having been produced by an unknown instrument, the type of the test instrument is determined using the instrument database. A precise instrument identification can be obtained by using the amplitude-time representation of a tone produced by an instrument for identifying a musical instrument.

Abstract: In a method for generating a second data stream from a first data stream, which comprises a first header and a first payload data block with payload data, the first header is initially extracted from the first data stream. Subsequently the second header for the second data stream is generated. Then, at least a part of the first header is entered into the second header, the part of the first header including information, which allows conclusions as to the origin of the payload data. Finally, the second payload data block is generated, which comprises the same payload data, so as to obtain the complete second data stream. The method according to the invention enables device-specific encrypting of payload data, a flexible, device-specific “copy” for other devices of a user and, in particular, complete documentation of the origin of the present copy, such that effective copyright protection may be realized.

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: In a method for transferring a music signal into a note-based description, a frequency-time representation of the music signal is first generated, the frequency-time representation comprising coordinate tuples, a coordinate tuple including a frequency value and a time value, the time value indicating the time of occurrence of the assigned frequency in the music signal. Thereupon, a fit function will be calculated as a function of the time, the course of which is determined by the coordinate tuples of the frequency-time representation. For time-segmenting the frequency-time representation, at least two adjacent extreme values of the fit function will be determined. On the basis of the determined extreme values, a segmenting will be carried out, a segment being limited by two adjacent extreme values of the fit function, the time length of the segments indicating a time length of a note for the segment. For pitch determination, a pitch for the segment using coordinate tuples in the segment will be determined.

Abstract: For characterizing an information signal having an amplitude-time waveform with local extreme values, at first the local extreme values of the information signal are determined, wherein a local extreme value is defined by a time instant and an amplitude. Furthermore, area information of valleys or mountains of the information signal in case of a one-dimensional amplitude of the information signal or volume information in case of a two-dimensional amplitude of the information signal of valleys or mountains is ascertained. A valley or mountain is defined by a temporal section of the information signal, wherein the section of the information signal extends from the time instant of a local extreme value to a temporarily adjacent value of the information signal having the same amplitude as the local extreme value.

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: 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.