G10L19/24

Audio encoder and bandwidth extension decoder

An audio encoder for providing an output signal using an input audio signal includes a patch generator, a comparator and an output interface. The patch generator generates at least one bandwidth extension high-frequency signal, wherein a bandwidth extension high-frequency signal includes a high-frequency band. The high-frequency band of the bandwidth extension high-frequency signal is based on a low frequency band of the input audio signal. A comparator calculates a plurality of comparison parameters. A comparison parameter is calculated based on a comparison of the input audio signal and a generated bandwidth extension high-frequency signal. Each comparison parameter of the plurality of comparison parameters is calculated based on a different offset frequency between the input audio signal and a generated bandwidth extension high-frequency signal. Further, the comparator determines a comparison parameter from the plurality of comparison parameters, wherein the determined comparison parameter fulfills a predefined criterion.

METHOD FOR ENCODING MULTI-CHANNEL AUDIO SIGNAL AND ENCODING DEVICE FOR PERFORMING ENCODING METHOD, AND METHOD FOR DECODING MULTI-CHANNEL AUDIO SIGNAL AND DECODING DEVICE FOR PERFORMING DECODING METHOD

An encoding method for a multi-channel audio signal, an encoding apparatus for performing the encoding method, and a decoding method for a multi-channel audio signal and a decoding apparatus for performing the decoding method are disclosed. A method and apparatus of bypassing an MPEG Surround (MPS) standard operation and using an arbitrary tree when a number of audio signals of N channels exceeds a channel number defined in an MPS standard, is disclosed.

TRANSMISSION-AGNOSTIC PRESENTATION-BASED PROGRAM LOUDNESS

This disclosure falls into the field of audio coding, in particular it is related to the field of providing a framework for providing loudness consistency among differing audio output signals. In particular, the disclosure relates to methods, computer program products and apparatus for encoding and decoding of audio data bitstreams in order to attain a desired loudness level of an output audio signal.

TRANSMISSION-AGNOSTIC PRESENTATION-BASED PROGRAM LOUDNESS

This disclosure falls into the field of audio coding, in particular it is related to the field of providing a framework for providing loudness consistency among differing audio output signals. In particular, the disclosure relates to methods, computer program products and apparatus for encoding and decoding of audio data bitstreams in order to attain a desired loudness level of an output audio signal.

System for maintaining reversible dynamic range control information associated with parametric audio coders

On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (α) from the bitstream, where 1≤m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

System for maintaining reversible dynamic range control information associated with parametric audio coders

On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (α) from the bitstream, where 1≤m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

HARMONIC TRANSPOSITION IN AN AUDIO CODING METHOD AND SYSTEM
20230027660 · 2023-01-26 · ·

The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length L.sub.a, extracting a frame of the input signal, and an analysis transformation unit of order M transforming the samples into M complex coefficients. M is a function of the transposition factor T. The system further comprises a nonlinear processing unit altering the phase of the complex coefficients by using the transposition factor T, a synthesis transformation unit of order M transforming the altered coefficients into M altered samples, and a synthesis window of length L.sub.s, generating a frame of the output signal.

HARMONIC TRANSPOSITION IN AN AUDIO CODING METHOD AND SYSTEM
20230027660 · 2023-01-26 · ·

The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length L.sub.a, extracting a frame of the input signal, and an analysis transformation unit of order M transforming the samples into M complex coefficients. M is a function of the transposition factor T. The system further comprises a nonlinear processing unit altering the phase of the complex coefficients by using the transposition factor T, a synthesis transformation unit of order M transforming the altered coefficients into M altered samples, and a synthesis window of length L.sub.s, generating a frame of the output signal.

Audio decoding device, audio coding device, audio decoding method, audio coding method, audio decoding program, and audio coding program
11562760 · 2023-01-24 · ·

An objective of the present invention is to correct a temporal envelope shape of a decoded signal with a small information volume and to reduce perceptible distortions. An audio decoding device which decodes a coded audio signal and outputs an audio signal comprises: a coded series analysis unit that analyzes a coded series which contains the coded audio signal; an audio decoding unit that receives from the coded series analysis unit the coded series which contains the coded audio signal and decodes same, obtaining an audio signal; a temporal envelope shape establishment unit that receives information from the coded series analysis unit and/or the audio decoding unit, and, on the basis of the information, establishes a temporal envelope shape of the decoded audio signal; and a temporal envelope correction unit that, on the basis of the temporal envelope shape which is established with the temporal envelope shape establishment unit, corrects the temporal envelope shape of the decoded audio signal and outputs same.

Audio decoding device, audio coding device, audio decoding method, audio coding method, audio decoding program, and audio coding program
11562760 · 2023-01-24 · ·

An objective of the present invention is to correct a temporal envelope shape of a decoded signal with a small information volume and to reduce perceptible distortions. An audio decoding device which decodes a coded audio signal and outputs an audio signal comprises: a coded series analysis unit that analyzes a coded series which contains the coded audio signal; an audio decoding unit that receives from the coded series analysis unit the coded series which contains the coded audio signal and decodes same, obtaining an audio signal; a temporal envelope shape establishment unit that receives information from the coded series analysis unit and/or the audio decoding unit, and, on the basis of the information, establishes a temporal envelope shape of the decoded audio signal; and a temporal envelope correction unit that, on the basis of the temporal envelope shape which is established with the temporal envelope shape establishment unit, corrects the temporal envelope shape of the decoded audio signal and outputs same.