Methods, an encoder and a decoder are configured for transition between frames with different internal sampling rates. Linear predictive (LP) filter parameters are converted from a sampling rate S1 to a sampling rate S2. A power spectrum of a LP synthesis filter is computed, at the sampling rate S1, using the LP filter parameters. The power spectrum of the LP synthesis filter is modified to convert it from the sampling rate S1 to the sampling rate S2. The modified power spectrum of the LP synthesis filter is inverse transformed to determine autocorrelations of the LP synthesis filter at the sampling rate S2. The autocorrelations are used to compute the LP filter parameters at the sampling rate S2.

The present disclosure relates to an apparatus for decoding an encoded Unified Audio and Speech stream. The apparatus comprises a core decoder for decoding the encoded Unified Audio and Speech stream. The core decoder includes an eSBR unit for extending a bandwidth of an input signal, the eSBR unit including a QMF based harmonic transposer. The QMF based harmonic transposer is configured to process the input signal in the QMF domain, in each of a plurality of synthesis subbands, to extend the bandwidth of the input signal. The QMF based harmonic transposer is configured to operate at least in part based on pre-computed information. The present disclosure further relates to corresponding methods and storage media.

The present disclosure relates to an apparatus for decoding an encoded Unified Audio and Speech stream. The apparatus comprises a core decoder for decoding the encoded Unified Audio and Speech stream. The core decoder includes an eSBR unit for extending a bandwidth of an input signal, the eSBR unit including a QMF based harmonic transposer. The QMF based harmonic transposer is configured to process the input signal in the QMF domain, in each of a plurality of synthesis subbands, to extend the bandwidth of the input signal. The QMF based harmonic transposer is configured to operate at least in part based on pre-computed information. The present disclosure further relates to corresponding methods and storage media.

A system and method for processing an audio input signal includes a microphone, a controller, and a communication link that may be coupled to a remote speaker. The microphone captures the audio input signal and communicates the audio input signal to the controller, and the controller is coupled to the communication link. The controller includes executable code to generate, via a linear noise reduction filtering algorithm, a first resultant based upon the audio input signal, and generate, via non-linear post filtering algorithm, a second resultant based upon the first resultant. An audio output signal is generated based upon the second resultant employing a feature restoration algorithm. The audio output signal is communicated, via the communication link, to a speaker that may be at a remote location.

A system and method for processing an audio input signal includes a microphone, a controller, and a communication link that may be coupled to a remote speaker. The microphone captures the audio input signal and communicates the audio input signal to the controller, and the controller is coupled to the communication link. The controller includes executable code to generate, via a linear noise reduction filtering algorithm, a first resultant based upon the audio input signal, and generate, via non-linear post filtering algorithm, a second resultant based upon the first resultant. An audio output signal is generated based upon the second resultant employing a feature restoration algorithm. The audio output signal is communicated, via the communication link, to a speaker that may be at a remote location.

A coding apparatus includes: a first coding circuit that codes an input signal selectively using coding in a time domain or a frequency domain according to the characteristic of the input signal in a core layer; and a second coding circuit that codes an error in coding by the first coding circuit using a coding method corresponding to the domain type of coding used in the core layer in an extension layer for the core layer.

A coding apparatus includes: a first coding circuit that codes an input signal selectively using coding in a time domain or a frequency domain according to the characteristic of the input signal in a core layer; and a second coding circuit that codes an error in coding by the first coding circuit using a coding method corresponding to the domain type of coding used in the core layer in an extension layer for the core layer.

A wireless audio system for encoding and decoding an audio signal using spectral bandwidth replication is provided. Bandwidth extension is performed in the time-domain, enabling low-latency audio coding.

A wireless audio system for encoding and decoding an audio signal using spectral bandwidth replication is provided. Bandwidth extension is performed in the time-domain, enabling low-latency audio coding.

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.