Abstract: For producing a fingerprint of an audio signal, use is made of information defining a plurality of predetermined fingerprint modi, all of the fingerprint modi relating to the same type of fingerprint, the fingerprint modi, however, providing different fingerprints differing from each other with regard to their data volume, on the one hand, and to their characterizing strength for characterizing the audio signal, on the other hand, the fingerprint modi being predetermined such that a fingerprint in accordance with a fingerprint modus having a first characterizing strength is convertible to a fingerprint in accordance with a fingerprint modus having a second characterizing strength, without using the audio signal. A predetermined fingerprint modus of the plurality of predetermined fingerprint modi is set and subsequently used for computing a fingerprint using the audio signal.

Abstract: 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.

Abstract: In a method for signalling a noise substitution when coding an audio signal, the time-domain audio signal is first transformed into the frequency domain to obtain spectral values. The spectral values are subsequently grouped together to form groups of spectral values. On the basis of a detection establishing whether a group of spectral values is a noisy group or not, a codebook is allocated to a non-noisy or tonal group by means of a codebook number for redundancy coding of the same. If a group is noisy, an additional codebook number which does not refer to a codebook is allocated to it in order to signal that this group is noisy and therefore does not have to be redundancy coded. By signalling noise substitution by means of a Huffman codebook number for noisy groups of spectral values, which are e.g.

Abstract: An apparatus for analyzing an audio signal with regard to rhythm information of the audio signal by using an autocorrelation function comprises a filter bank for separating the audio signal into at least two sub-band signals. The sub-band signals are examined with regard to periodicities by an autocorrelation function, to obtain rhythm raw-information for the at least two sub-band signals. To reduce or eliminate the ambiguities of the autocorrelation function for periodical signals, the rhythm raw-information is postprocessed to obtain post-processed rhythm raw-information for the sub-band signal. The rhythm information of the audio signal is established based on the postprocessed rhythm raw-information.

Abstract: An apparatus for analyzing an audio signal with regard to rhythm information of the audio signal comprises a filterbank for dividing the audio signal into at least two sub-band signals. Every sub-band signal is examined with regard to a periodicity of the sub-band signal to obtain rhythm raw-information of every sub-band signal. The rhythm raw-information is subjected to a quality evaluation to obtain a significance measure for every sub-band signal. The rhythm information of the audio signal will finally be established by considering the significance measure of the sub-band signal and the rhythm raw-information. This enables a more robust analysis of the audio signal, since sub-band signals, where significant rhythm information are present, are preferred compared to sub-band signals where less significant rhythm information are present, when establishing the rhythm information.

Abstract: 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.

Abstract: A method for detecting a transient in a discrete-time audio signal is performed completely in the time domain and includes the step of segmenting the discrete-time audio signal so as to generate consecutive segments of the same length with unfiltered discrete-time audio signals xs(T-1). The discrete-time audio signal in a current segment is subsequently filtered. Then either the energy of the filtered discrete-time audio signal in the current segment can be compared with the energy of the filtered discrete-time audio signal in a preceding segment or a current relationship between the energy of the filtered discrete-time audio signal in the current segment and the energy of the unfiltered discrete-time audio signal in the current segment can be formed and this current relationship compared with a preceding corresponding relationship. On the basis of the one and/or the other of these comparisons it is detected whether a transient is present in the discrete-time audio signal.

Abstract: A method for detecting a transient in a discrete-time audio signal is performed completely in the time domain and includes the step of segmenting the discrete-time audio signal as to generate consecutive segments of the same length with unfiltered discrete-time audio signals. The discrete-time audio signal in a current segment is filtered. Either the energy of the filtered discrete-time audio signal in the current segment is compared with the energy of the filtered discrete-time audio signal in a preceding segment or a current relationship between the energy of the filtered discrete-time audio signal in the current segment and the energy of the unfiltered discrete-time audio signal in the current segment is formed and this current relationship compared with a preceding corresponding relationship. Whether a transient is present in the discrete-time audio signal is detected using one and/or the other of these comparisons.

Abstract: 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.

Abstract: In a method for concealing errors in an audio data stream the occurrence of an error is detected in the audio data stream, audio data prior to the occurrence of the fault being intact audio data. Thereafter a spectral energy of a subgroup of the intact audio data is calculated. After forming a pattern for substitute data on the basis of the spectral energy calculated for the subgroup of the intact audio data, substitute data for erroneous or missing audio data which correspond to the subgroup are created on the basis of the pattern.

Abstract: 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.

Abstract: Jointly processed stereophonic audio signal properties are identified using a stereophonic signal as reference signal and creating a signal for testing by processing the stereophonic signal, e.g. by coding and subsequently decoding it. Both signals are transformed into the frequency domain to create representative spectral data for the respective subbands. Correlation coefficients are determined for each subband both of the reference signal and also of the signal for testing on the basis of the spectral data of the channels of the reference signal or of the signal for testing. From the comparison of the correlation coefficients belonging to the same subband, jointly processed stereophonic audio signals are detected if at least one of the correlation coefficients of the signal for testing greatly exceeds the correlation coefficient of the reference signal for the same subband.

Abstract: The tonality of an audio signal is determined by a method which includes the steps of blockwise frequency transforming a digital input signal x(n) to create a real positive-value representation X(k) of the input signal, where k designates the index of a frequency line, and determining the tonality T of the signal component for the frequency line k according to the following equation: ##EQU1## where F.sub.1 is the filter function of a first digital filter with a first, differentiating characteristic, F.sub.2 is the filter function of a second digital filter with a second, flat or integrating characteristic or with a characteristic which is less strongly differentiating than the first characteristic, and d.sub.1 and d.sub.2 are integer constants which, depending on the filter parameters, are so chosen that the delays of the filters are compensated for in each case.

Abstract: In the case of coding a plurality of signals which are not independent of e another, a selection of the suitable type of coding is made as a function of a similarity measure.According to one aspect of the invention, the similarity measure is determined by firstly coding one of the signals according to the intensity-stereo method and then decoding it in order to create a signal affected by coding error, whereupon the latter signal and the associated non-coded signal are transformed into the frequency domain. In the frequency domain, a selection or evaluation of the actually audible spectral components, as well as of the signal affected by coding error and of the associated signal not affected by coding error, is undertaken using a listening threshold which is determined by a psycho-acoustic calculation. Intensity-stereo coding is undertaken in the case of a high similarity measure, whereas otherwise a separate coding of the channels is performed.

Abstract: A process for transmitting and/or storing digital signals of multiple chals. This process is suited, in particular, for transmitting the five channels of 3/2 stereophony as well as for transmitting two stereo channels and three additional commentary channels. In this manner, by way of illustration, television programs with multi-language audio signals can be transmitted. This process is distinguished in that by reduction of the to-be-transmitted data, only a bit rate of 384 kbit/s is required for transmission. The reduction of the data is achieved by the K input channels being imaged in segments onto the N.ltoreq.K virtual spectral data channels, by the spectral data channels being quantized, coded, and transmitted taking into consideration the principles of psychoacoustics, and by K output channels being reproduced from the transmitted bit stream with the aid of a transmitted list from the N.ltoreq.K spectral data channels.

Abstract: A process for reducing data in the transmission and/or storage of digital signals of several interdependent channels is described.As the signals from the channels are not independent of one another, an additional data quantity reduction is possible during transmission.In known processes for the coding of signals from two stereo channels, in the case of an unfavorable signal composition interference can be caused by the coding process and cannot be concealed by the signal actually present in the channel and consequently reduce listening enjoyment.In the process according to the invention and with the aid of the spectral values of corresponding blocks of several channels, there is a balancing of the necessary data rate for a separate coding of the two channels and for a joint coding. There is a joint coding of the channels for as long as the data rate for the joint coding does not exceed by a predeterminable value the data rate for the separate coding of the particular channel.

Abstract: In a method of coding a plurality of audio signals, the left and the right basic channel as well as the central channel are combined by joint stereo coding so as to obtain a jointly coded signal, which is decoded so as to provide simulated decoded signals. The simulated decoded signals and two surround channels are combined by matricization by means of a compatibility matrix so as to form compatible signals which are suitable for decoding by existing decoders.

Abstract: Disclosed is a process for the detection of errors, respectively perceptible disturbances, in the transmission of frequency-coded digital signals and, in particular, audio signals transmitted in blocks by a transmitter to a receiver.The present invention is distinguished by that from the frequency coefficients of earlier and, if need be, future blocks, a decision function is formed, on the basis of which the occurrence of an error is determined, and that the frequency coefficient containing an error is not longer utilized for the evaluation of subsequent blocks.