Abstract: An encoding method, a decoding method, an encoding apparatus, a decoding apparatus, a transmitter, a receiver, and a communications system, where 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, 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.

Abstract: The present invention disclose a speech/audio bitstream decoding method including: acquiring a speech/audio decoding parameter of a current speech/audio frame, where the foregoing current speech/audio frame is a redundant decoded frame or a speech/audio frame previous to the foregoing current speech/audio frame is a redundant decoded frame; performing post processing on the acquired speech/audio decoding parameter according to speech/audio parameters of X speech/audio frames, where the foregoing X speech/audio frames include M speech/audio frames previous to the foregoing current speech/audio frame and/or N speech/audio frames next to the foregoing current speech/audio frame; and recovering a speech/audio signal by using the post-processed speech/audio decoding parameter of the foregoing current speech/audio frame. The technical solutions of the present invention help improve quality of an output speech/audio signal.

Abstract: An audio encoding method and a related apparatus are disclosed. The audio coding method includes: obtaining an audio signal; obtaining spectral coefficients of a current frame of the audio signal; obtaining a set of reference coding parameters of the current frame; determining whether the set of reference coding parameters satisfy a preset parameter condition; when it is determined that the reference coding parameters satisfy the preset parameter condition, encoding the spectral coefficients of the current frame using a transform coded excitation (TCX) algorithm. The audio encoding method and the related apparatus help improve encoding quality or encoding efficiency in audio signal encoding.

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 inter-channel phase difference (IPD) parameter extraction method and apparatus, where the extraction method includes obtaining a parameter obtaining an information extraction manner for a current frame of a multi-channel signal, obtaining an IPD parameter extraction manner for the current frame based on the parameter obtaining the information extraction manner, where the obtained IPD parameter extraction manner is one of at least two preset IPD parameter extraction manners, and obtaining an IPD parameter of the current frame based on the obtained IPD parameter extraction manner for the current frame.

Abstract: A method and an apparatus for processing a temporal envelope of an audio signal, and an encoder, where when multiple temporal envelopes are solved, continuity of signal energy can be well maintained, and in addition, complexity of calculating a temporal envelope is reduced. The method includes obtaining a high-band signal of the current frame audio signal according to the received current frame audio signal, dividing the high-band signal of the current frame signal into M subframes according to a predetermined temporal envelope quantity M, where M is an integer greater than or equal to two, calculating a temporal envelope of each of the subframes, performing windowing on the first subframe of the M subframes and the last subframe of the M subframes using an asymmetric window function, and performing windowing on a subframe except the first subframe and the last subframe of the M subframes.

Abstract: A method and device for decoding a signal. The method for decoding a signal includes: obtaining spectral coefficients of sub-bands from a received bitstream by means of decoding; classifying sub-bands in which the spectral coefficients are located into a sub-band with saturated bit allocation and a sub-band with unsaturated bit allocation; performing noise filling on a spectral coefficient that has not been obtained by means of decoding and is in the sub-band with unsaturated bit allocation, so as to restore the spectral coefficient that has not been obtained by means of decoding; and obtaining a frequency domain signal according to the spectral coefficients obtained by means of decoding and the restored spectral coefficient. Therefore, a sub-band with unsaturated bit allocation in a frequency domain signal may be obtained by classification, thereby improving signal decoding quality.

Abstract: The present disclosure provide a signal processing method and apparatus. The method includes: determining a total quantity of to-be-allocated bits corresponding to a current frame; implementing primary bit allocation on to-be-processed sub-bands; performing a primary information unit quantity determining operation for each sub-band that has undergone the primary bit allocation; selecting sub-bands for secondary bit allocation from the to-be-processed sub-bands according to at least one of a sub-band characteristic of each sub-band of the to-be-processed sub-bands or the total quantity of surplus bits; implementing secondary bit allocation on the sub-bands for secondary bit allocation; and performing a secondary information unit quantity determining operation for each sub-band of the sub-bands for secondary bit allocation.

Abstract: Present disclosure disclose a voice signal processing method, includes: receiving a first voice coded signal from a first terminal; performing voice decoding processing on the first voice coded signal to obtain a voice decoding parameter and a first voice decoded signal; performing, by using the voice decoding parameter, virtual bandwidth extension processing to obtain a bandwidth extension voice decoded signal corresponding to the first voice decoded signal; after combining the first voice decoded signal and the bandwidth extension voice decoded signal, performing voice coding processing to obtain a second voice coded signal; and sending the second voice coded signal to a second terminal that establishes a call connection to the first terminal, where a maximum frequency bandwidth supported by the first terminal is less than a maximum frequency bandwidth supported by the second terminal. Thus, Service quality of terminals that have asymmetric maximum frequency bandwidth support capabilities can be improved.

Abstract: Embodiments of the present disclosure provide signal encoding and decoding methods and devices. The method includes: determining, a quantity k of subbands to be encoded, where i is a positive number, and k is a positive integer; selecting, according to quantized envelopes of all subbands, k subbands from all the subbands, or selecting k subbands from all subbands according to a psychoacoustic model; and performing a first-time encoding operation on spectral coefficients of the k subbands. In some embodiments of the present disclosure, the quantity k of subbands to be encoded is determined according to the quantity of available bits and the first saturation threshold, and encoding is performed on the k subbands that are selected from all the subbands, instead of on an entire frequency band.

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 application can improve an encoding and/or decoding effect.

Abstract: Embodiments of the present disclosure provide a decoding method and a decoding apparatus. The decoding method includes: in a case in which it is determined that a current frame is a lost frame, synthesizing a high frequency band signal; determining subframe gains of multiple subframes of the current frame; determining a global gain of the current frame; and adjusting, according to the global gain and the subframe gains of the multiple subframes, the synthesized high frequency band signal to obtain a high frequency band signal of the current frame. A subframe gain of the current frame is obtained according to a gradient between subframe gains of subframes previous to the current frame, so that transition before and after frame loss is more continuous, thereby reducing noise during signal reconstruction, and improving speech quality.

Abstract: Present disclosure pertains to an audio signal processing method and device. A current frame of a frequency-domain audio signal has N frequency sub-bands. According to an energy attribute and a spectral attribute of a first subset of M sub-bands, whether to adjust envelope values of the M sub-bands is determined. Based on a determination that the energy envelope values of the M sub-bands need to be adjusted, the energy 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 adjusted envelope values of the M sub-bands and energy envelope information of K sub-bands of a second subset. The first subset and the second subset have no overlap in frequency, and N=M+K.

Abstract: The present invention provides 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 band 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: A method and an apparatus for processing a temporal envelope of an audio signal, and an encoder are disclosed. When multiple temporal envelopes are solved, continuity of signal energy can be well maintained, and in addition, complexity of calculating a temporal envelope is reduced. The method includes: obtaining a high-band signal of the current frame audio signal according to the received current frame audio signal; dividing the high-band signal of the current frame signal into M subframes according to a predetermined temporal envelope quantity M, where M is an integer that is greater than or equal to 2; calculating a temporal envelope of each of the subframes; performing windowing on the first subframe of the M subframes and the last subframe of the M subframes by using an asymmetric window function; and performing windowing on a subframe except the first subframe and the last subframe of the M subframes.

Abstract: The present invention discloses a speech/audio signal processing method and apparatus.

Abstract: An audio signal decoding method includes obtaining mode information of a high frequency band signal of an audio signal and indices of a low frequency band signal of the audio signal by parsing a received bitstream, obtaining the low frequency band signal based on the indices, predicting an excitation signal of a high frequency band signal based on the low frequency band signal, and reconstructing the high frequency band signal based on the frequency envelope and the excitation signal. A manner for obtaining the frequency envelope of the high frequency band signal when mode information indicates the high frequency band signal is a harmonic type signal is different from a manner for obtaining the frequency envelope of the high frequency band signal when the mode information indicates the high frequency band signal is a non-harmonic type signal.

Abstract: Embodiments of the present disclosure provide signal encoding and decoding methods and devices. The method includes: determining, a quantity k of subbands to be encoded, where i is a positive number, and k is a positive integer; selecting, according to quantized envelopes of all subbands, k subbands from all the subbands, or selecting k subbands from all subbands according to a psychoacoustic model; and performing a first-time encoding operation on spectral coefficients of the k subbands. In the embodiments of the present disclosure, the quantity k of subbands to be encoded is determined according to the quantity of available bits and the first saturation threshold, and encoding is performed on the k subbands that are selected from all the subbands, instead of on an entire frequency band, which can reduce spectrum holes of a signal obtained through decoding, and therefore, can improve auditory quality of an output signal.

Abstract: The present disclosure provides a speech/audio signal processing method based on wideband switching and a coding apparatus. The method includes: if a first wideband speech/audio signal is a harmonic signal, adjusting a determining condition for determining that a second wideband speech/audio signal is a harmonic signal, to obtain a first determining condition, where the first wideband speech/audio signal is a signal before wideband switching, and the second wideband speech/audio signal is a signal after the wideband switching; and determining, according to the first determining condition, whether the second wideband speech/audio signal is a harmonic signal. In the case of wideband switching, signal types of speech/audio signals remain as consistent as possible before and after the switching, so that continuity of the speech/audio signal decoded by a decoder device is ensured as much as possible, further improving speech communication service quality.

Abstract: The present disclosure provide a signal processing method and apparatus. The method includes: determining a total quantity of to-be-allocated bits corresponding to a current frame; implementing primary bit allocation on to-be-processed sub-bands; performing a primary information unit quantity determining operation for each sub-band that has undergone the primary bit allocation; selecting sub-bands for secondary bit allocation from the to-be-processed sub-bands according to at least one of a sub-band characteristic of each sub-band of the to-be-processed sub-bands or the total quantity of surplus bits; implementing secondary bit allocation on the sub-bands for secondary bit allocation; and performing a secondary information unit quantity determining operation for each sub-band of the sub-bands for secondary bit allocation.