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:
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:
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 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:
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:
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:
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:
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:
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.
摘要:
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 pre-determined 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. The convertibility feature of the fingerprints having been produced by the different fingerprint modi enables setting a flexible compromise between the data volume and the characterizing strength for certain applications without having to re-generate a fingerprint database with each change of the fingerprint modus. Fingerprint representations scaled with regard to time or frequency may readily be converted to a different fingerprint modus.