An audio signal encoding method includes obtaining a current frame of an audio signal, where the current frame includes a high frequency band signal and a low frequency band signal; obtaining a parameter of bandwidth extension of the current frame based on the high frequency band signal, the low frequency band signal, and configuration information of the bandwidth extension; obtaining tile information, where the tile information indicates a first frequency range in which tonal component detection needs to be performed on the high frequency band signal; performing tonal component detection in the first frequency range to obtain information about a tonal component of the high frequency band signal; and performing bitstream multiplexing on the parameter of the bandwidth extension and the information of the tonal component to obtain a payload bitstream.

An audio signal encoding method includes obtaining a current frame of an audio signal, where the current frame includes a high frequency band signal and a low frequency band signal; obtaining a parameter of bandwidth extension of the current frame based on the high frequency band signal, the low frequency band signal, and configuration information of the bandwidth extension; obtaining tile information, where the tile information indicates a first frequency range in which tonal component detection needs to be performed on the high frequency band signal; performing tonal component detection in the first frequency range to obtain information about a tonal component of the high frequency band signal; and performing bitstream multiplexing on the parameter of the bandwidth extension and the information of the tonal component to obtain a payload bitstream.

A method and device for encoding a stereo sound signal comprise stereo encoders using stereo modes operating in time domain (TD), in frequency domain (FD) or in modified discrete Fourier transform (MDCT) domain. A controller controls switching between the TD, FD and MDCT stereo modes. Upon switching from one stereo mode to the other, the switching controller may (a) recalculate at least one length of down-processed/mixed signal in a current frame of the stereo sound signal, (b) reconstruct a down-processed/mixed signal and also other signals related to the other stereo mode in the current frame, (c) adapt data structures and/or memories for coding the stereo sound signal in the current frame using the other stereo mode, and/or (d) alter a TD stereo channel down-mixing to maintain a correct phase of left and right channels of the stereo sound signal. Corresponding stereo sound signal decoding method and device are described.

The coding efficiency of an audio codec using a controllable—switchable or even adjustable—harmonic filter tool is improved by performing the harmonicity-dependent controlling of this tool using a temporal structure measure in addition to a measure of harmonicity in order to control the harmonic filter tool. In particular, the temporal structure of the audio signal is evaluated in a manner which depends on the pitch. This enables to achieve a situation-adapted control of the harmonic filter tool so that in situations where a control made solely based on the measure of harmonicity would decide against or reduce the usage of this tool, although using the harmonic filter tool would, in that situation, increase the coding efficiency, the harmonic filter tool is applied, while in other situations where the harmonic filter tool may be inefficient or even destructive, the control reduces the appliance of the harmonic filter tool appropriately.

The coding efficiency of an audio codec using a controllable—switchable or even adjustable—harmonic filter tool is improved by performing the harmonicity-dependent controlling of this tool using a temporal structure measure in addition to a measure of harmonicity in order to control the harmonic filter tool. In particular, the temporal structure of the audio signal is evaluated in a manner which depends on the pitch. This enables to achieve a situation-adapted control of the harmonic filter tool so that in situations where a control made solely based on the measure of harmonicity would decide against or reduce the usage of this tool, although using the harmonic filter tool would, in that situation, increase the coding efficiency, the harmonic filter tool is applied, while in other situations where the harmonic filter tool may be inefficient or even destructive, the control reduces the appliance of the harmonic filter tool appropriately.

An audio signal coding method is provided that includes: obtaining a current frame of an audio signal; obtaining a coding parameter based on a power spectrum ratio of a current frequency in a current frequency area of at least a part of signals of the current frame, where the coding parameter indicates tonal component information of the at least a part of signals, the tonal component information includes at least one of location information of a tonal component, quantity information of tonal components, amplitude information of the tonal component, or energy information of the tonal component, and the power spectrum ratio of the current frequency is a ratio of a value of a power spectrum of the current frequency to a mean value of power spectrums of the current frequency area; and performing bitstream multiplexing on the coding parameter to obtain a coded bitstream.

An audio signal coding method is provided that includes: obtaining a current frame of an audio signal; obtaining a coding parameter based on a power spectrum ratio of a current frequency in a current frequency area of at least a part of signals of the current frame, where the coding parameter indicates tonal component information of the at least a part of signals, the tonal component information includes at least one of location information of a tonal component, quantity information of tonal components, amplitude information of the tonal component, or energy information of the tonal component, and the power spectrum ratio of the current frequency is a ratio of a value of a power spectrum of the current frequency to a mean value of power spectrums of the current frequency area; and performing bitstream multiplexing on the coding parameter to obtain a coded bitstream.

A speech coding method of reducing error propagation due to voice packet loss, is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame, the excitation of a next frame is obtained according to the reduced or limited pitch gain value of the first subframe, and the next frame is encoded according to the obtained excitation. The method is used for a voiced speech class.

A speech coding method of reducing error propagation due to voice packet loss, is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame, the excitation of a next frame is obtained according to the reduced or limited pitch gain value of the first subframe, and the next frame is encoded according to the obtained excitation. The method is used for a voiced speech class.

An audio processing unit (APU) is disclosed. The APU includes a buffer memory configured to store at least one frame of an encoded audio bitstream, where the encoded audio bitstream includes audio data and a metadata container. The metadata container includes a header and one or more metadata payloads after the header. The one or more metadata payloads include dynamic range compression (DRC) metadata, and the DRC metadata is or includes profile metadata indicative of whether the DRC metadata includes dynamic range compression (DRC) control values for use in performing dynamic range compression in accordance with at least one compression profile on audio content indicated by at least one block of the audio data.