摘要:
An audio signal decoder includes a context-based spectral value decoder configured to decode a codeword describing one or more spectral values or at least a portion of a number representation thereof in dependence on a context state. The audio signal decoder also includes a context state determinator configured to determine a current context state in dependence on one or more previously decoded spectral values and a time warping frequency-domain-to-time-domain converter configured to provide a time-warped time-domain representation of a given audio frame on the basis of a set of decoded spectral values provided by the context-based spectral value decoder and in dependence on the time warp information. The context-state determinator is configured to adapt the determination of the context state to a change of a fundamental frequency between subsequent audio frames. An audio signal encoder applies a comparable concept.
摘要:
An apparatus for generating a bandwidth extended audio signal from an input signal, includes a patch generator for generating one or more patch signals from the input signal, wherein the patch generator is configured for performing a time stretching of subband signals from an analysis filterbank, and wherein the patch generator further includes a phase adjuster for adjusting phases of the subband signals using a filterbank-channel dependent phase correction.
摘要:
The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.
摘要:
A parameter transformer generates level parameters, indicating an energy relation between a first and a second audio channel of a multi-channel audio signal associated to a multi-channel loudspeaker configuration. The level parameter are generated based on object parameters for a plurality of audio objects associated to a down-mix channel, which is generated using object audio signals associated to the audio objects. The object parameters have an energy parameter indicating an energy of the object audio signal. To derive the coherence and the level parameters, a parameter generator is used, which combines the energy parameter and object rendering parameters, which depend on a desired rendering configuration.
摘要:
An apparatus for synthesizing a rendered output signal having a first audio channel and a second audio channel includes a decorrelator stage for generating a decorrelator signal based on a downmix signal, and a combiner for performing a weighted combination of the downmix signal and a decorrelated signal based on parametric audio object information, downmix information and target rendering information. The combiner solves the problem of optimally combining matrixing with decorrelation for a high quality stereo scene reproduction of a number of individual audio objects using a multichannel downmix.
摘要:
The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.
摘要:
An audio signal decoder configured to provide a decoded audio signal representation on the basis of an encoded audio signal representation including a sampling frequency information, an encoded time warp information and an encoded spectrum representation includes a time warp calculator and a warp decoder. The time warp calculator is configured to adapt a mapping rule for mapping codewords of the encoded time warp information onto decoded time warp values describing the decoded time warp information in dependence on the sampling frequency information. The warp decoder is configured to provide the decoded audio signal representation on the basis of the encoded spectrum representation and in dependence on the decoded time warp information.
摘要:
The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.
摘要:
The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.
摘要:
The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.