摘要:
A multi-mode audio signal decoder has a spectral value determinator to obtain sets of decoded spectral coefficients for a plurality of portions of an audio content and a spectrum processor configured to apply a spectral shaping to a set of spectral coefficients in dependence on a set of linear-prediction-domain parameters for a portion of the audio content encoded in a linear-prediction mode, and in dependence on a set of scale factor parameters for a portion of the audio content encoded in a frequency-domain mode. The audio signal decoder has a frequency-domain-to-time-domain converter configured to obtain a time-domain audio representation on the basis of a spectrally-shaped set of decoded spectral coefficients for a portion of the audio content encoded in the linear-prediction mode and for a portion of the audio content encoded in the frequency domain mode. An audio signal encoder is also described.
摘要:
For adding additional data, such as multi-channel extension data, to base data, such as conventional stereo data, a test fingerprint of test data relating to a test time instant of the test data is provided. The test data equals the additional data or the base data or depends on the additional data or the base data in parametric manner. Using the test fingerprint, reference time instant information is determined, which depends on a reference time instant in reference data, the reference data being the conventional stereo data. Finally, the additional data or the base data is manipulated, namely using the reference time instant information and the test time instant information, to obtain manipulated data, by which synchronous reproduction of the data information can be performed. Thus, a robust and flexible possibility for synchronous, especially late extension of base data by additional data is obtained.
摘要:
For a multi-channel audio signal, parametric coding is applied to different subsets of audio input channels for different frequency regions. For example, for a 5.1 surround sound signal having five regular channels and one low-frequency (LFE) channel, binaural cue coding (BCC) can be applied to all six audio channels for sub-bands at or below a specified cut-off frequency, but to only five audio channels (excluding the LFE channel) for sub-bands above the cut-off frequency. Such frequency-based coding of channels can reduce the encoding and decoding processing loads and/or size of the encoded audio bitstream relative to parametric coding techniques that are applied to all input channels over the entire frequency range.
摘要:
An input multi-channel representation is converted into a different output multi-channel representation of a spatial audio signal, in that an intermediate representation of the spatial audio signal is derived, the intermediate representation having direction parameters indicating a direction of origin of a portion of the spatial audio signal; and in that the output multi-channel representation of the spatial audio signal is generated using the intermediate representation of the spatial audio signal.
摘要:
An apparatus for encoding a sequence of samples of an audio signal, with each sample within the sequence having an original position, includes a sorter for sorting the samples depending on their sizes, in order to obtain a sorted sequence of samples, with each sample having a sorting position within the sorted sequence. Furthermore, the apparatus has an encoder for encoding the sorted samples and information on a relation between the original and sorting positions of the samples.
摘要:
In processing a multi-channel audio signal having at least three original channels, a first downmix channel and a second downmix channel are provided, which are derived from the original channels. For a selected original channel of the original channels, channel side information are calculated such that a downmix channel or a combined downmix channel including the first and the second downmix channels, when weighted using the channel side information, results in an approximation of the selected original channel. The channel side information and the first and second downmix channels form output data to be transmitted to a decoder, which, in case of a low level decoder only decodes the first and second downmix channels or, in case of a high level decoder provides a full multi-channel audio signal based on the downmix channels and the channel side information. Since the channel side information only occupy a low number of bits, and since the decoder does not use dematrixing, an efficient and high quality multi-channel extension for stereo players and enhanced multi-channel players is obtained.
摘要:
An apparatus for processing a multi-channel signal includes a means for determining a similarity between a first one of two channels and a second one of the two channels. Furthermore, a means for performing a prediction filtering of the spectral coefficients is provided, which is formed to perform a prediction filtering with only a single prediction filter for both channels in case of high similarity between the first and the second channel, and to perform a prediction filtering with two separate prediction filters in case of a dissimilarity between the first and the second channel. With this, an introduction of stereo artifacts and a deterioration of the coding gain in stereo coding techniques are avoided.
摘要:
For analyzing an information signal having a sequence of blocks of information units, wherein a plurality of consecutive blocks of the sequence of blocks represents an information entity, using a sequence of fingerprints for the sequence of blocks, identification results are provided for consecutive fingerprints, wherein an identification result represents an association of a block of information units with a predetermined information entity. Then at least two hypotheses are formed from the identification results for the consecutive fingerprints, wherein a first hypothesis is an assumption for the association of the sequence of blocks with a first information entity, and wherein the second hypothesis is an assumption for the association of the sequence of blocks with the second information entity. Then various hypotheses are examined to obtain an examination result on the basis of which there is then made a statement on the information signal. This achieves a meaningful and reliable time-continuous analysis of an information signal.
摘要:
In a method for concealing an error in an encoded audio signal a set of spectral coefficients is subdivided into at least two sub-bands (14), whereupon the sub-bands are subjected to a re-verse transform (16). A specific prediction is performed (18) for each quasi time signal of a sub-band to obtain an estimated temporal representation for a sub-band of a set of spectral coefficients following the current set. A forward transform (20) of the time signal of each sub-band provides estimated spectral coefficients which can be used (28) instead of erroneous spectral coefficients of a following set of spectral coefficients, e.g. in order to conceal transmission errors. Transforming at the sub-band level provides independence from transform characteristics such as block length, window type and MDCT algorithm while at the same time preserving spectral processing for error concealment. Thus the spectral characteristics of audio signals can also be taken into account during error concealment.
摘要:
For embedding watermark information into an information signal including audio and/or video information, first of all a synchronization sequence with a plurality of synchronization sequence units and a data sequence with a plurality of data sequence units are provided, wherein between the data sequence and the synchronization sequence a time shift is present and wherein a degree of shifting depends on the watermark information. A combination means generates a combination sequence having a plurality of combination sequence units from the synchronization sequence and the data sequence shifted with regard to the synchronization sequence, wherein the combination sequence units are derived from synchronization sequence units and shifted data sequence units. The combination sequence is combined with the information signal in order to embed the watermark information into the information signal. A watermark extractor receives a synchronization sequence correlation peak for every data sequence correlation peak associated with the same and therefore determines the watermark information on the basis of the time interval between the synchronization sequence correlation peak and the data sequence correlation peak in a secure and robust way. The concept is robust, provides a high data rate and is simultaneously flexible with regard to the weighting of synchronization energy and data energy and with regard to the robustness on the one hand and data rate on the other hand, respectively.