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.
摘要:
An assignment of one of phase set of different loss concealment tools of an audio decoder to a portion of the audio signal to be decoded from a data stream, which portion is affected by loss, that is the selection out of the set of different loss concealment tools, may be made in a manner leading to a more pleasant loss concealment if the assignment/selection is done based on two measures: A first measure which is determined measures a spectral position of a spectral centroid of a spectrum of the audio signal and a second measure which is determined measures a temporal predictability of the audio signal. The assigned or selected loss concealment tool may then be used to recover the portion of the audio signal.
摘要:
An apparatus (100) for generating spectral replacement values for an audio signal is provided. The apparatus (100) comprises a buffer unit (110) for storing previous spectral values relating to a previously received error-free audio frame. Moreover, the apparatus (100) comprises a concealment frame generator (120) for generating the spectral replacement values, when a current audio frame has not been received or is erroneous. The previously received error-free audio frame comprises filter information, the filter information having associated a filter stability value indicating a stability of a prediction filter. The concealment frame generator (120) is adapted to generate the spectral replacement values based on the previous spectral values and based on the filter stability value.
摘要:
A device for masking a fault in a faulty or potentially faulty information unit generates output values of a forward decoder (14) and of a backward decoder (16), which differ from one another thereby indicating (18) a fault masking area. The different values for one and the same information unit are checked with regard to a predetermined criterion in order to select (22) the value that fulfills, i.e. is plausible for, the predetermined criterion. This enables, without impairing a compression rate, an elimination or reduction of the continuation faults introduced during a decoding of blocks comprised of reversible code words of a variable length.
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 includes a receiving interface for receiving one or more frames comprising information on a plurality of audio signal samples of an audio signal spectrum of the encoded audio signal, and a processor for generating the reconstructed audio signal. The processor is configured to generate the reconstructed audio signal by fading a modified spectrum to a target spectrum, if a current frame is not received by the receiving interface or if the current frame is received by the receiving interface but is corrupted, wherein the modified spectrum includes a plurality of modified signal samples, wherein, for each of the modified signal samples of the modified spectrum, an absolute value of the modified signal sample is equal to an absolute value of one of the audio signal samples of the audio signal spectrum.
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.