摘要:
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. Then, the greatest common divisor of the analysis spectral coefficients or the spectral coefficients derived from the analysis spectral coefficients will be calculated, corresponding to the quantization step width used when quantizing the encoding algorithm or an integer multiple of it. Then, 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. Thus, all parameters used for the quantization of the original time signal are known, so that for quantizing the analysis time signal no longer full iteration loops have to be performed, which are, on the one hand, very computing time intensive and, on the other hand, introduce tandem encoding distortions.
摘要:
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.
摘要:
In determining a coding block raster on which a decoded signal is based, a segment of the decoded signal is picked out first, said segment beginning at a certain output sampling value of the decoded signal. Said segment is then converted into a spectral representation, whereupon said spectral representation is then evaluated in relation to a predetermined criterion in order to obtain an evaluation result for the segment. This procedure is repeated for a plurality of different segments beginning at different output sampling values each, in order to obtain a plurality of evaluation results. Finally, the plurality of the evaluation results is searched in order to establish the evaluation result that has an extreme value as compared to the other evaluation results, in such a way that it can be assumed that the segment to which this evaluation result is allocated matches the coding block raster on which the decoded signal is based. This method can be used to determine the coding block raster for any decoded signal that has no explicit information about its coding block raster.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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. The division into a central wave-field synthesis module and the plurality of distributed loudspeaker modules allows that audio reproduction systems that are scalable with regard to the price can be generated in order to offer systems of different size scalable in price particularly for cinema reproduction rooms varying strongly in size.