摘要:
When several signals (k1, k2) that are not independent from each other are coded, the appropriate type of coding is selected depending on a similarity measurement. In one aspect of the invention, the similarity measurement is determined by one of the signals (k1, k2) being at first coded according to the intensity stereo process and then decoded, in order to create a signal (ki1, ki2) affected by a coding error. Said signal and the corresponding non-coded signal are then transformed into the frequency range. The actually audible spectral components of both the signals (ki1, ki2) affected by a coding error and the corresponding signal (k1, k2) unaffected by a coding error are selected or evaluated in the frequency range by using a listening threshold determined by a psychoacoustic calculation. When the degree of similarity is high, an intensity stereo coding process is carried out, otherwise the channels are separately coded.
摘要:
Dispositif pour le contrôle de systèmes de traitement de signaux sonores présentant les caractéristiques suivantes: Le dispositif comporte une première borne d'entrée, au niveau de laquelle est appliqué le signal d'entrée du système de traitement de signaux sonores à contrôler, une deuxième borne d'entrée, au niveau de laquelle est appliqué le signal de sortie du système, et un processeur de signaux. Le processeur de signaux calcule, par corrélation entre les deux signaux présents aux deux bornes d'entrée, le retard du signal du système à contrôler. Le processeur de signaux produit en permanence un signal différentiel à partir du signal présent pendant une période déterminée au niveau de la première borne d'entrée et du signal présent ensuite au niveau de la deuxième borne d'entrée après une période égale au retard du signal. Le processeur de signaux calcule la composition spectrale du signal présent pendant la période déterminée au niveau de la première borne d'entrée et celle du signal différentiel correspondant. A partir de la composition spectrale du signal présent au niveau de la première borne d'entrée, le processeur de signaux calcule le seuil d'audition de l'oreille humaine et le compare au signal différentiel correspondant.
标题翻译:VERFAHREN UND VORRICHTUNG ZUM EXTRAHIEREN EINER SIGNALKENNUNG,VERFAHREN UND VORRICHUNG ZUM ERZEUGEN EINERDAZUGEHÖRIGENDATABANK und Verfahren und Vorrichtung zum Referenzieren eines Such-Zeitsignals
摘要:
The invention relates to a method for extracting a signal identifier from a time signal, according to which the temporal occurrence of signal edges is detected in the time signal (12), whereby a signal edge has a specified temporal length. In addition, the temporal interval between two selected detected signal edges is determined (14). A frequency value is calculated (16) from said determined interval and is assigned to a time of occurrence of the frequency value in the time signal in order to obtain a co-ordinate tuple from the frequency value and the time of occurrence for said frequency value. A signal identifier is created from a plurality of co-ordinate tuples (18), each co-ordinate tuple containing a frequency value and a time of occurrence, in such a way that the signal identifier comprises a sequence of signal identifier values, which reproduce the temporal course of the time signal. The extracted signal identifier is based on signal edges of the time signal and reproduces the temporal course of the time signal. The signal identifier thus characterises the time signal and is stable in relation to changes to said time signal.
摘要:
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. By the fact that the information to be introduced are introduced into the data stream without changing to the time domain, the block rastering underlying the short-term spectrum will not be touched, so that introducing a watermark will not lead to tandem encoding effects.
摘要:
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.
摘要:
The invention makes it possible to combine a scaleable audio coder with TNS technology. According to the inventive method for encoding time signals (x1) sampled in a first sampling rate, second time signals (x2) with a sampling rate smaller than the first sampling rate are generated (12). The second time signals (x2) are then encoded (14) according to a first coding algorithm, and written into a bit stream (xAUS) (16). The encoded second time signals (x2c) are then decoded (14) again and are transformed (23, 24) into the frequency range, as are the first time signals. TNS prediction coefficients are then calculated (25) from a spectral representation of the first time signals (X1). The transformed output signal (X2cd) of the coder/decoder (14) with the first coding algorithm and the spectral representation (X1) of the first time signal are subjected to a prediction of the frequency (27) in order to obtain spectral residual values for both signals using the prediction coefficients calculated on the basis of the first time signals alone. These two signals are evaluated against each other (26, 28). The evaluated spectral residual values (Xb) are then encoded by means of a second coding algorithm in order to obtain coded evaluated spectral residual values (Xcb). These evaluated spectral residual values are written into the bit stream (xAUS) in addition to side information with the prediction coefficients.
摘要:
The invention relates to a method for signalling a noise substitution during audio signal coding. According to said method, the audio signal is first transformed in the frequency range to obtain spectral values. The spectral values are subsequently grouped to form spectral value groups. On the basis of a detection whether a group of spectral values is a noise group or not, a coding table is allocated to a non-noise group or a tonal group by means of a coding table number for redundancy coding of the same. If a group is a noise group it is allocated an additional coding table number which does not refer to a coding table in order to signal that this group is a noise group and that it must not be redundancy coded. By signalling noise substitution by means of a Huffman-code table number for noise groups of spectral values which are for instance scale factor band sections and which must not be redundancy coded, an opportunity is provided for implementing availability of a noise substitution in a scale factor band in the bit flow syntax of the MPEG-2 Advanced Audio coding Standard, without intervening in the basic coding structure and without having to touch the structure of the existing bit flow syntax.