摘要:
An audio decoder configured to produce an audio signal from a bitstream containing audio frames includes: a core band decoding module configured to derive a directly decoded core band audio signal from the bitstream; a bandwidth extension module configured to derive a parametrically de-coded bandwidth extension audio signal from the core band audio signal and from the bitstream, wherein the bandwidth extension audio signal is based on a frequency domain signal having at least one frequency band; and a combiner configured to combine the core band audio signal and the bandwidth extension audio signal so as to produce the audio signal; wherein the bandwidth extension module includes an energy adjusting module being configured in such way that in a current audio frame in which an audio frame loss occurs, an adjusted signal energy for the cur-rent audio frame for the at least one frequency band is set.
摘要:
An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal is provided, having: a receiving interface for receiving one or more frames, a coefficient generator, and a signal reconstructor. The coefficient generator is configured to determine one or more first audio signal coefficients, and one or more noise coefficients. Moreover, the coefficient generator is configured to generate one or more second audio signal coefficients, depending on the one or more first audio signal coefficients and depending on the one or more noise coefficients. The audio signal reconstructor is configured to reconstruct a first portion of the reconstructed audio signal depending on the one or more first audio signal coefficients and the audio signal reconstructor is configured to reconstruct a second portion of the reconstructed audio signal depending on the one or more second audio signal coefficients, if the current frame is not received by the receiving interface or if the current frame being received by the receiving interface is corrupted.
摘要:
The present invention is based on the finding that parameters including a first set of parameters of a representation of a first portion of an original signal and including a second set of parameters of a representation of a second portion of the original signal can be efficiently encoded, when the parameters are arranged in a first sequence of tuples and in a second sequence of tuples, wherein the first sequence of tuples comprises tuples of parameters having two parameters from a single portion of the original signal and wherein the second sequence of tuples comprises tuples of parameters having one parameter from the first portion and one parameter from the second portion of the original signal. An efficient encoding can be achieved using a bit estimator to estimate the number of necessary bits to encode the first and the second sequence of tuples, wherein only the sequence of tuples is encoded, that results in the lower number of bits.
摘要:
The invention relates to an entropy encoder (10) for generating a data flow, which has two reference points consisting of variable-length code words. Said device comprises a first unit (16) for writing at least one portion of a code word into the data flow in a first direction of writing, starting from a first reference point, and comprises a second device (18) for writing at least one portion of a code word into the data flow in a second direction of writing which is opposite to the first directions of writing, starting from the other reference point. In particular, if a raster with multiple segments is used to write the variable-length code words into the data flow, then in the best-case scenario the number of code words that can be written, beginning at the raster points, is doubled in such a way that the data flow of variable-length code words is resistant with regard to the propagation of sequence errors.
摘要:
The invention relates to a method for generating a data flow from variable-length code words which are subdivided into a number of code word sets, whereby a raster comprising segments is determined for the data flow in which two adjacent raster points (41, 42) define a segment (40). The code words (1-6) from the first set are written into the data flow, beginning at raster points. Code words from the second set are then written into the data flow according to a predetermined allocation specification, whereby each code word from the second set is allocated to a separate segment. Whole code words or portions of code words which cannot be written according to their allocation are saved and entered into the data flow in additional writing attempts, whereby the allocation according to a predetermined specification is changed from one attempt to the next. This procedure is repeated in a similar manner for any additional data set that may exist. In this way, the ends of code words from the second set are decoupled from the beginnings of following code words from the second set, as the corresponding code words in a set are written in a segment by segment manner. This reduces the propagation of errors.
摘要:
Audio receiver processor for processing an error protected frame, comprising: a receiver interface (300) for receiving the error protected frame to obtain a received error protected frame; an error protection processor (302) for processing the received error protected frame to obtain an encoded audio frame, wherein the error protection processor is configured to check whether a codeword of a first predefined subset of codewords of the encoded audio frame comprises an error, and an error concealer or an error concealment indicator (304) configured to perform a frame loss concealment operation or to generate a frame loss concealment indication in case of a detected error in the first predefined subset of the codewords.
摘要:
The purpose of the invention is to bridge the gap between parametric multi-channel audio coding and matrixed-surround multi-channel coding by gradually improving the sound of an up-mix signal while raising the bit-rate consumed by the side-information starting from 0 up to the bit-rates of the parametric methods. More specifically, it provides a method of flexibly choosing an "operating point" somewhere between matrixed-surround (no side-information, limited audio quality) and fully parametric reconstruction (full side-information rate required, good quality). This operating point can be chosen dynamically (i.e. varying over time) and in response to the permissible side-information rate, as it is dictated by the individual application.
摘要:
The purpose of the invention is to bridge the gap between parametric multi-channel audio coding and matrixed-surround multi-channel coding by gradually improving the sound of an up-mix signal while raising the bit-rate consumed by the side-information starting from 0 up to the bit-rates of the parametric methods. More specifically, it provides a method of flexibly choosing an "operating point" somewhere between matrixed-surround (no side-information, limited audio quality) and fully parametric reconstruction (full side-information rate required, good quality). This operating point can be chosen dynamically (i.e. varying over time) and in response to the permissible side-information rate, as it is dictated by the individual application.