Abstract: An audio encoding method and a related apparatus are disclosed. The audio coding method includes: performing a time-frequency transformation on a current frame of a time-domain audio signal, to obtain spectral coefficients of the current audio frame; obtaining one or more reference coding parameters of the current frame; and determining whether the reference coding parameters satisfy a set of parameter conditions. If any one of the parameter conditions is satisfied, the spectral coefficients of the current frame are encoded by using a transform coded excitation (TCX) algorithm. If none of the parameter conditions is satisfied, the spectral coefficients of the current audio frame are encoded using a high quality transform coding (HQ) algorithm. The audio encoding method and the related apparatus help improve encoding quality or encoding efficiency in audio signal encoding.

Abstract: Embodiments of the present invention provide a coding/decoding method, apparatus, and system. According to the coding method, de-emphasis processing is performed on a full band signal by using a de-emphasis parameter determined according to a characteristic factor of an input audio signal, and then the full band signal is coded and sent to a decoder, so that the decoder performs corresponding de-emphasis decoding processing on the full band signal according to the characteristic factor of the input audio signal and restores the input audio signal. This resolves a prior-art problem that an audio signal restored by a decoder is apt to have signal distortion, and implements adaptive de-emphasis processing on the full band signal according to the characteristic factor of the audio signal to enhance coding performance, so that the input audio signal restored by the decoder has relatively high fidelity and is closer to an original signal.

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: Embodiments of the present application provide a method and an apparatus for recovering a lost frame in a received audio signal. The method for recovering a lost frame includes: determining an initial high-frequency band signal of a current lost frame; determining a gain of the current lost frame; determining gain adjustment information of the current lost frame; adjusting the gain of the current lost frame according to the gain adjustment information, to obtain an adjusted gain of the current lost frame; and adjusting the initial high-band signal according to the adjusted gain, to obtain a high-frequency band signal of the current lost frame. The method and the apparatus for recovering a lost frame provided in the embodiments of the present application can be used in an audio signal decoding process for low-loss recovery of a lost frame of the audio signal, resulting in improved performance of an audio signal decoder.

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, M 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: Method and apparatus are provided for reconstructing a noise component of a speech/audio signal. A bitstream, is received and decoded to obtain a speech/audio signal. A first speech/audio signal is determined according to the speech/audio signal. A symbol of each sample value in the first speech/audio signal and an amplitude value of each sample value in the first speech/audio signal is determined. An adaptive normalization length and an adjusted amplitude value of each sample value are determined according to the adaptive normalization length and the amplitude value of each sample value. A second speech/audio signal is determined according to the symbol of each sample value and the adjusted amplitude value of each sample value.

Abstract: In a signal coding method, bits for coding allocated to different bands of a frequency domain signal obtained from an input signal are adjusted to improve the coding quality. The total available bits for coding are first allocated to the bands of the frequency domain signal according to a predetermined allocation rule. The numbers of bits allocated to the respective bands of the frequency domain signal are then adjusted when a highest frequency of the frequency domain signal to which bits are allocated is greater than a predetermined value. The frequency domain signal is coded according to the adjusted bit allocation for the bands of the frequency domain signal.

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 method and an apparatus for allocating bits of an audio signal. The method includes dividing a frequency band of an audio signal into multiple sub-bands, and quantizing a sub-band normalization factor of each sub-band; classifying the multiple sub-bands into multiple groups, and acquiring a sum of intra-group sub-band normalization factors of each group; performing initial inter-group bit allocation to determine the initial number of bits of each group; performing secondary inter-group bit allocation to allocate coding bits of the audio signal to at least one group; and allocating the bits of the audio signal to sub-bands in the group. The present disclosure can, by means of grouping, ensure relatively stable allocation in a previous frame and a next frame and reduce an impact of global allocation on local discontinuity in a case of low and medium bit rates.

Abstract: The present disclosure relates to an audio signal coding method and apparatus. The method includes categorizing audio signals into high-frequency audio signals and low-frequency audio signals, coding the low-frequency audio signals using a corresponding low-frequency coding manner according to characteristics of low-frequency audio signals, and selecting a bandwidth extension mode to code the high-frequency audio signals according to the low-frequency coding manner and/or characteristics of the audio signals.

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: 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: In a signal coding method, bits for coding allocated to different bands of a frequency domain signal obtained from an input signal are adjusted to improve the coding quality. The total available bits for coding are first allocated to the bands of the frequency domain signal according to a predetermined allocation rule. The numbers of bits allocated to the respective bands of the frequency domain signal are then adjusted when a highest frequency of the frequency domain signal to which bits are allocated is greater than a predetermined value. The frequency domain signal is coded according to the adjusted bit allocation for the bands of the frequency domain signal.

Abstract: A method and an apparatus for allocating bits of an audio signal. The method includes dividing a frequency band of an audio signal into multiple sub-bands, and quantizing a sub-band normalization factor of each sub-band; classifying the multiple sub-bands into multiple groups, and acquiring a sum of intra-group sub-band normalization factors of each group; performing initial inter-group bit allocation to determine the initial number of bits of each group; performing secondary inter-group bit allocation to allocate coding bits of the audio signal to at least one group; and allocating the bits of the audio signal to sub-bands in the group. The present invention can, by means of grouping, ensure relatively stable allocation in a previous frame and a next frame and reduce an impact of global allocation on local discontinuity in a case of low and medium bit rates.

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: The present invention relates to an audio signal coding method and apparatus. The method includes: categorizing audio signals into high-frequency audio signals and low-frequency audio signals; coding the low-frequency audio signals by using a corresponding low-frequency coding manner according to characteristics of low-frequency audio signals; and selecting a bandwidth extension mode to code the high-frequency audio signals according to the low-frequency coding manner and/or characteristics of the audio signals.

Abstract: A method and an apparatus for decoding a speech/audio bitstream are disclosed, where the method for decoding a speech/audio bitstream includes determining whether a current frame is a normal decoding frame or a redundancy decoding frame, obtaining a decoded parameter of the current frame by means of parsing when the current frame is a normal decoding frame or a redundancy decoding frame, performing post-processing on the decoded parameter of the current frame to obtain a post-processed decoded parameter of the current frame, and using the post-processed decoded parameter of the current frame to reconstruct a speech/audio signal.

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: A method and an apparatus for estimating an interchannel delay of a sound signal are, related to the communication field and capable of realizing a stable sound field in a crosstalk. The method includes calculating an error between an actual interchannel phase difference and a predicted interchannel phase difference of a sound signal. The predicted interchannel phase difference is predicted according to a predetermined interchannel delay of the sound signal. The method also includes determining whether the sound signal is a sound signal in a crosstalk according to the error. The method further includes, if the sound signal is a sound signal in the crosstalk, setting an interchannel delay corresponding to the sound signal to a fixed value. The apparatus is configured to perform the method.

Abstract: A method for predicting a bandwidth extension frequency band signal includes demultiplexing a received bitstream to obtain a frequency domain signal; determining whether a highest frequency bin, to which a bit is allocated, of the frequency domain signal is less than a preset start frequency bin of a bandwidth extension frequency band; predicting an excitation signal of the bandwidth extension frequency band according to the determination; and predicting the bandwidth extension frequency band signal according to the predicted excitation signal of the bandwidth extension frequency band and a frequency envelope of the bandwidth extension frequency band.