Abstract:
An audio coding method and apparatus, where the method includes, for each audio frame in audio, when a signal characteristic of the audio frame and a signal characteristic of a previous audio frame meet a preset modification condition, determining a first modification weight according to linear spectral frequency (LSF) differences of the audio frame and the LSF differences of the previous audio frame, modifying a linear predictive parameter of the audio frame according to the determined first modification weight, and coding the audio frame according to a modified linear predictive parameter of the audio frame. According to the present disclosure, audio having a wider bandwidth can be coded while a bit rate remains unchanged or a bit rate slightly changes and a spectrum between audio frames is steadier.
Abstract:
Present disclosure provides a signal processing method and device. Spectral coefficients of a current frame of a frequency-domain audio signal are divided into N sub-bands. N is a positive integer greater than 1. According to an energy attribute and a spectral attribute of a first subset of the N sub-bands, whether to modify original envelope values of sub-bands in the first subset is determined. A frequency range of each of the M sub-bands in the first subset is lower than a frequency range of each of the K sub-bands. Based on a determination that the original envelope values of the M sub-bands need to be modified, the original envelope values of the M sub-bands are modified individually to obtain modified envelope values of the M sub-bands. Encoding bits are allocated to each of the N sub-bands according to the modified envelope values of the M sub-bands and original envelope values of the K sub-bands.
Abstract:
An audio coding method and apparatus, where the method includes, for each audio frame in audio, when a signal characteristic of the audio frame and a signal characteristic of a previous audio frame meet a preset modification condition, determining a first modification weight according to linear spectral frequency (LSF) differences of the audio frame and LSF differences of the previous audio frame, modifying a linear predictive parameter of the audio frame according to the determined first modification weight, and coding the audio frame according to a modified linear predictive parameter of the audio frame. According to the present disclosure, audio having a wider bandwidth can be coded while a bit rate remains unchanged or a bit rate slightly changes and a spectrum between audio frames is steadier.
Abstract:
A frame loss compensation processing method and apparatus is presented, where the method includes, when a ith frame is a lost frame, estimating a spectrum frequency parameter, a pitch period, and a gain of the ith frame according to at least one of an inter-frame relationship between first N frames of the ith frame or an intra-frame relationship between first N frames of the ith frame. A parameter of the ith frame is determined using the signal correlation between the first N frames, the signal energy stability between the first N frames, intra-frame signal correlation of each frame, and intra-frame signal energy stability of each frame.
Abstract:
An encoding method, a decoding method, an encoding apparatus, a decoding apparatus, a transmitter, a receiver, and a communications system. The encoding method includes: dividing a to-be-encoded time-domain signal into a low band signal and a high band signal; performing encoding on the low band signal to obtain a low frequency encoding parameter; performing encoding on the high band signal to obtain a high frequency encoding parameter, and obtaining a synthesized high band signal; performing short-time post-filtering processing on the synthesized high band signal to obtain a short-time filtering signal; and calculating a high frequency gain based on the high band signal and the short-time filtering signal. A technical solution according to the embodiments of the present invention can improve an encoding and/or decoding effect.
Abstract:
Embodiments of the present invention relate to the communications field, and provide a data processing method and apparatus, which can resolve a problem of excessively small capacity of a base station for data transmission, and improve capacity of the base station for data transmission. In one embodiment, a base station obtains a wireless data packet, performs protocol conversion on the wireless data packet to generate a transmission data packet, adapts the transmission data packet to generate a standard data packet, compresses and optimizes the standard data packet to generate compressed data, and adds the compressed data to a preset data packet. The present invention is used for data processing.
Abstract:
Present disclosure provide an encoding method and apparatus, which relate to the communications field and can perform proper quantization bit allocation for spectral coefficients of an audio signal, thereby improving quality of a signal obtained by a decoder by means of decoding. The method includes: after splitting spectral coefficients of a current data frame into subbands, acquiring quantized frequency envelope values of the subbands; modifying quantized frequency envelope values of subbands of a first quantity in the subbands; allocating quantization bits to the subbands according to modified quantized frequency envelope values of the subbands of the first quantity; quantizing a spectral coefficient of a subband to which a quantization bit is allocated in the subbands; and writing the quantized spectral coefficient of the subband to which a quantization bit is allocated into a bitstream.
Abstract:
The present invention provide a bandwidth extension method and apparatus. The method includes: acquiring a bandwidth extension parameter, where the bandwidth extension parameter includes one or more of the following parameters: a linear predictive coefficient (LPC), a line spectral frequency (LSF) parameter, a pitch period, a decoding rate, an adaptive codebook contribution, and an algebraic codebook contribution; and performing, according to the bandwidth extension parameter, bandwidth extension on a decoded low-frequency signal, to obtain a high frequency band signal. The high frequency band signal recovered by using the bandwidth extension method and apparatus in the embodiments of the present invention is close to an original high frequency band signal, and the quality is satisfactory.
Abstract:
The present invention provides a method for switching a working mode on a relay network, a base station, a relay node, and a communications system. A donor eNB (DeNB) sends an RN reconfiguration message to a relay node RN in frequency division duplex FDD mode, so that the RN switches from the FDD mode to a half-duplex frequency division duplex H-FDD mode. The DeNB receives an acknowledgment message sent by the RN in FDD mode. When the DeNB verifies, according to the RN reconfiguration message and the acknowledgement message, that the RN has already started to switch from the FDD mode to the H-FDD mode, the DeNB switches from the FDD mode to the H-FDD mode.
Abstract:
Embodiments of this application provide a timing advance determining method and a communication apparatus, to improve precision of calculating a timing advance (TA) by a terminal, and reduce inter-symbol interference (ISI). The method includes: A first network device determines a first parameter based on a first delay compensation value, where the first delay compensation value is a delay compensation made by the first network device for receiving a signal sent by a terminal, the first parameter indicates a difference between a round-trip delay of a feeder link in a non-terrestrial network NTN and the first delay compensation value, and the difference is used to determine a TA used by the terminal for signal sending; and the first network device sends the first parameter.