摘要:
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. The evaluated residual spectral values are then coded by means of a second coding algorithm to obtain coded evaluated residual spectral values, which, together with the side information containing the calculated prediction coefficients, are written into the bit stream.
摘要:
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.
摘要:
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.
摘要:
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. sections made up of scale factor bands which do not have to be redundancy coded, an opportunity is provided to indicate the presence of a noise substitution in a scale factor band in the bit stream syntax of the MPEG-2 Advanced Audio Coding (AAC) Standard without having to interfere with the basic coding structure and without having to meddle with the structure of the existing bit stream syntax.
摘要:
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.
摘要:
A method of coding a time-discrete stereo signal, the stereo signal having a first and a second channel, permits scalable stereo coding. At first, a mono signal is formed from the stereo signal, which is then coded, whereupon the coded mono signal is transmitted to a bit stream. Thereafter, the coded mono singal is decoded again, whereupon stereo information is formed on the basis of the coded/decoded mono signal and the first and second channels, with such stereo information being coded and being also written into the bit stream in order to obtain a bit stream comprising a complete coded monolayer as well as a layer with coded stereo information.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.said signal processor ascertains the signal delay time of said system to be checked by means of correlating said signals received at said two input connections,said signal processor always composes the difference signal from said signal received at said first input connection during a specific time span and said signal received at said second input connection, lagging by the signal delay time,said signal processor ascertains the spectral composition of said signal received at said first input connection during said specific time span and of said respective difference signal,said signal processor ascertains the hearing threshold of the human ear from said spectral composition and compares the ascertained hearing threshold with the respective difference signal.