Abstract: In a method of coding discrete time signals (X1) sampled with a first sampling rate, second time signals (x2) are generated using the first time signals having a bandwidth corresponding to a second sampling rate, with the second sampling rate being lower than the first sampling rate. The second time signals are coded in accordance with a first coding algorithm. The coded second signals (X2c) are decoded again in order to obtain coded/decoded second time signals (X2cd) having a bandwidth corresponding to the second sampling frequency. The first time signals, by frequency domain transformation, become first spectral values (X1). Second spectral values (X2cd) are generated from the coded/decoded second time signals, the second spectral values being a representation of the coded/decoded time signals in the frequency domain.

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: 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.

Abstract: In coding of an audio signal, coded signals with low quality and bit rate on the one hand and coded signals with high quality and bit rate on the other hand are transmitted to a decoder. At first, the audio signal is coded with low bit rate and is transmitted to the decoder before an additional coded signal is transmitted to the decoder, which either alone or together with the first coded signal upon decoding thereof provides a decoded signal with high quality within the decoder. In this manner, a low-quality decoded signal is generated first in the decoder before decoding of the high-quality signal is possible.

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: A method of encoding time-discrete audio signals comprises the steps of weighting the time-discrete audio signal by means of window functions overlapping each other so as to form blocks, the window functions producing blocks of a first length for signals varying weakly with time and blocks of a second length for signals varying strongly with time. A start window sequence is selected for the transition from windowing with blocks of the first length to windowing with blocks of the second length, whereas a stop window sequence is selected for the opposite transition. The start window sequence is selected from at least two different start window sequences having different lengths, whereas the stop window sequence is selected from at least two different stop window sequences having different lengths. A method of decoding blocks of encoded audio signals selects a suitable inverse transformation as well as a suitable synthesis window as a reaction to side information associated with each block.

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: 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: A method is disclosed for determining estimates of the perceived noise masking level of audio signals as a function of frequency. By developing a randomness metric related to the euclidian distance between (i) actual frequency components amplitude and phase for each block of sampled values of the signal and (ii) predicted values for these components based on values in prior blocks, it is possible to form a tonality index which provides more detailed information useful in forming the noise masking function. Application of these techniques is illustrated in a coding and decoding context for audio recording or transmission. The noise spectrum is shaped based on a noise threshold and a tonality measure for each critical frequency-band (bark).

Abstract: Disclosed is an apparatus for checking audio signal processing systems. The apparatus has the following features:the apparatus is provided with a first input connection, to which the input signal of the audio processing system to be checked is transmitted, a second input connection, to which the output signal of said system is transmitted, and a signal processor.

Abstract: A method is disclosed for determining estimates of the perceived noise masking level of audio signals as a function of frequency. By developing a randomness metric related to the euclidian distance between (i) actual frequency components amplitude and phase for each block of sampled values of the signal and (ii) predicted values for these components based on values in prior blocks, it is possible to form a tonality index which provides more detailed information useful in forming the noise masking function. Application of these techniques is illustrated in a coding and decoding context for audio recording or transmission. The noise spectrum is shaped based on a noise threshold and a tonality measure for each critical frequency-band (bark).