G10L19/265

STEREO AUDIO ENCODER AND DECODER

The present disclosure provides methods, devices and computer program products for encoding and decoding a stereo audio signal based on an input signal. According to the disclosure, a hybrid approach of using both parametric stereo coding and a discrete representation of the stereo audio signal is used which may improve the quality of the encoded and decoded audio for certain bitrates.

EFFICIENT COMBINED HARMONIC TRANSPOSITION

The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular, a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described. The system may comprise an analysis filter bank (501) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank (501) has a frequency resolution of Δf. The system further comprises a nonlinear processing unit (502) configured to determine a set of synthesis subband signals from the set of analysis subband signals using a transposition order P; wherein the set of synthesis subband signals comprises a portion of the set of analysis subband signals phase shifted by an amount derived from the transposition order P; and a synthesis filter bank (504) configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein the synthesis filter bank (504) has a frequency resolution of FΔf; with F being a resolution factor, with F≥1; wherein the transposition order P is different from the resolution factor F.

DECODER, ENCODER, AND METHOD FOR INFORMED LOUDNESS ESTIMATION IN OBJECT-BASED AUDIO CODING SYSTEMS

A decoder for generating an audio output signal having one or more audio output channels includes a receiving interface for receiving an audio input signal including a plurality of audio object signals, for receiving loudness information on the audio object signals, and for receiving rendering information indicating whether one or more of the audio object signals shall be amplified or attenuated. Moreover, the decoder includes a signal processor for generating the one or more audio output channels of the audio output signal. The signal processor is configured to determine a loudness compensation value depending on the loudness information and depending on the rendering information. Furthermore, the signal processor is configured to generate the one or more audio output channels of the audio output signal from the audio input signal depending on the rendering information and depending on the loudness compensation value. Moreover, an encoder is provided.

Model Based Prediction in a Critically Sampled Filterbank
20230306974 · 2023-09-28 · ·

The present document relates to audio source coding systems. In particular, the present document relates to audio source coding systems which make use of linear prediction in combination with a filterbank. A method for estimating a first sample (615) of a first subband signal in a first subband of an audio signal is described. The first subband signal of the audio signal is determined using an analysis filterbank (612) comprising a plurality of analysis filters which provide a plurality of subband signals in a plurality of subbands from the audio signal, respectively. The method comprises determining a model parameter (613) of a signal model; determining a prediction coefficient to be applied to a previous sample (614) of a first decoded subband signals derived from the first subband signal, based on the signal model, based on the model parameter (613) and based on the analysis filterbank (612); wherein a time slot of the previous sample (614) is prior to a time slot of the first sample (615); and determining an estimate of the first sample (615) by applying the prediction coefficient to the previous sample (614).

Parametric reconstruction of audio signals

An encoding system encodes an N-channel audio signal (X), wherein N≥3, as a single-channel downmix signal (Y) together with dry and wet upmix parameters ({tilde over (C)}, {tilde over (P)}). In a decoding system, a decorrelating section outputs, based on the downmix signal, an (N−1)-channel decorrelated signal (Z); a dry upmix section maps the downmix signal linearly in accordance with dry upmix coefficients (C) determined based on the dry upmix parameters; a wet upmix section populates an intermediate matrix based on the wet upmix parameters and knowing that the intermediate matrix belongs to a predefined matrix class, obtains wet upmix coefficients (P) by multiplying the intermediate matrix by a predefined matrix, and maps the decorrelated signal linearly in accordance with the wet upmix coefficients; and a combining section combines outputs from the upmix sections to obtain a reconstructed signal ({circumflex over (X)}) corresponding to the signal to be reconstructed.

Cross Product Enhanced Harmonic Transposition
20210366500 · 2021-11-25 · ·

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.

Post filter for audio signals

In some embodiments, a pitch filter for filtering a preliminary audio signal generated from an audio bitstream is disclosed. The pitch filter has an operating mode selected from one of either: (i) an active mode where the preliminary audio signal is filtered using filtering information to obtain a filtered audio signal, and (ii) an inactive mode where the pitch filter is disabled. The preliminary audio signal is generated in an audio encoder or audio decoder having a coding mode selected from at least two distinct coding modes, and the pitch filter is capable of being selectively operated in either the active mode or the inactive mode while operating in the coding mode based on control information.

Method, apparatus, and system for processing audio data
11183197 · 2021-11-23 · ·

A method for processing audio data includes obtaining a first noise frame of an audio signal, wherein the first noise frame includes a first low-band signal and a first high-band signal, obtaining a first low-band parameter corresponding to the first low-band signal and a first high-band parameter corresponding to the first high-band signal, encoding a first silence insertion descriptor (SID) corresponding to the first noise frame to comprise the first low-band parameter and the first high-band parameter, obtaining a second noise frame of the audio signal, wherein the second noise frame includes a second low-band signal and a second high-band signal, where the first noise frame is prior to the second noise frame in the audio signal, and determining whether a second SID corresponding to the second noise frame should comprise a second high-band parameter of the second high-band signal.

CROSS PRODUCT ENHANCED HARMONIC TRANSPOSITION
20230298606 · 2023-09-21 · ·

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.

SERVER AND METHOD FOR CONTROLLING SERVER

A display apparatus and a server which implements an interactive system are disclosed. The server includes a communicator which receives text information corresponding to a user voice collected at the display apparatus from the display apparatus, and a controller which extracts an utterance component from the text information and controls so that a query to search contents is generated using the extracted utterance component and transmitted to an external server which categorizes metadata of the content under each item and stores the same, in which the controller generates the query by adding a preset item to a criteria to search a content, when a number of criteria to categorize the content under an item corresponding to the extracted utterance component is less than a preset number.