A broadcast system may have less restrictive timing constraints by providing synchronous processing chains for the HD audio portion and the FM audio portion so that no samples are added or removed from when the input audio is first sampled at an input rate and when the signals are combined and output by a digital analog converter operating at an output rate. The signals can be buffered within the synchronous processing paths and the state of the buffer can be used to control the input rate of the sampler. Graceful change over across multiple input sources can be achieved provided all input source input rates are controlled to by the overall output rate and all input sources are phase aligned to produce output symbols at the same time.

A broadcast system may have less restrictive timing constraints by providing synchronous processing chains for the HD audio portion and the FM audio portion so that no samples are added or removed from when the input audio is first sampled at an input rate and when the signals are combined and output by a digital analog converter operating at an output rate. The signals can be buffered within the synchronous processing paths and the state of the buffer can be used to control the input rate of the sampler. Graceful change over across multiple input sources can be achieved provided all input source input rates are controlled to by the overall output rate and all input sources are phase aligned to produce output symbols at the same time.

An audio encoding bit rate prediction model training method is performed by a computer device. The method includes: obtaining a sample audio feature parameter corresponding to each of sample audio frames in a first sample audio; performing encoding bit rate prediction on the sample audio feature parameter through an encoding bit rate prediction model, to obtain a sample encoding bit rate for each of the sample audio frames; performing audio encoding on the sample audio frames based on the corresponding sample encoding bit rates to generate sample audio data corresponding to the sample audio frames; performing audio decoding on the sample audio data, to obtain a second sample audio corresponding to the sample audio data; and training the encoding bit rate prediction model based on the first sample audio and the second sample audio until a sample encoding quality score reaches a target encoding quality score.

An audio encoding bit rate prediction model training method is performed by a computer device. The method includes: obtaining a sample audio feature parameter corresponding to each of sample audio frames in a first sample audio; performing encoding bit rate prediction on the sample audio feature parameter through an encoding bit rate prediction model, to obtain a sample encoding bit rate for each of the sample audio frames; performing audio encoding on the sample audio frames based on the corresponding sample encoding bit rates to generate sample audio data corresponding to the sample audio frames; performing audio decoding on the sample audio data, to obtain a second sample audio corresponding to the sample audio data; and training the encoding bit rate prediction model based on the first sample audio and the second sample audio until a sample encoding quality score reaches a target encoding quality score.

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 present disclosure provides a communication method, apparatus, and system for a digital enhanced cordless telecommunications (DECT) base station. The method includes: determining, based on a communication connection request sent by a handset, whether a base station satisfies a wideband (WB) voice communication requirement of the handset, and returning communication acknowledgment information; and enabling the base station to perform WB voice communication with the handset if the communication acknowledgment information is a positive acknowledgment, or enabling the base station to perform narrowband (NB) voice communication with the handset if the communication acknowledgment information is a negative acknowledgment. The present disclosure can implement WB voice communication between a DECT base station and more than six handsets.

The present disclosure provides a communication method, apparatus, and system for a digital enhanced cordless telecommunications (DECT) base station. The method includes: determining, based on a communication connection request sent by a handset, whether a base station satisfies a wideband (WB) voice communication requirement of the handset, and returning communication acknowledgment information; and enabling the base station to perform WB voice communication with the handset if the communication acknowledgment information is a positive acknowledgment, or enabling the base station to perform narrowband (NB) voice communication with the handset if the communication acknowledgment information is a negative acknowledgment. The present disclosure can implement WB voice communication between a DECT base station and more than six handsets.

Various embodiments provide a method and an apparatus for determining a weighting factor during stereo signal encoding. In those embodiments, a parameter value corresponding to the encoding mode of the to-be-encoded signal is determining based on an encoding mode of a to-be-encoded signal in a stereo signal and a correspondence between an encoding mode and a parameter value. Based on the determined parameter value and an energy spectrum of a linear prediction filter corresponding to an original line spectral frequency parameter of the to-be-encoded signal is a weighting factor for calculating a distance between the original line spectral frequency parameter and a target original line spectral frequency parameter is calculated.

An audio encoding/decoding apparatus and method using vector quantized residual error features are disclosed. An audio signal encoding method includes outputting a bitstream of a main codec by encoding an original signal, decoding the bitstream of the main codec, determining a residual error feature vector from a feature vector of a decoded signal and a feature vector of the original signal, and outputting a bitstream of additional information by encoding the residual error feature vector.

Disclosed are a method of encoding and decoding an audio signal and an encoder and a decoder performing the method. The method of encoding an audio signal includes identifying an input signal, and generating a bitstring of each encoding layer by applying, to the input signal, an encoding model including a plurality of successive encoding layers that encodes the input signal, in which a current encoding layer among the encoding layers is trained to generate a bitstring of the current encoding layer by encoding an encoded signal which is a signal encoded in a previous encoding layer and quantizing an encoded signal which is a signal encoded in the current encoding layer.