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:
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 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:
A voice signal encoding and decoding method, device, and codec system are provided. The coding method includes: encoding an input voice signal to obtain a broadband code stream, where the broadband code stream includes a core layer bit stream and an extension enhancement layer bit stream (101); compressing the core layer bit stream to obtain a compressed code stream (102); and packing the compressed code stream and the extension enhancement layer bit stream to obtain a packed code stream (103). The core layer bit stream is compressed, and the compressed code stream and the extension enhancement layer bit stream are packed, thereby reducing transmission bandwidth occupied by the input voice signal. Since the broadband voice encoding is performed on the input voice signal, a broadband voice code stream is transmitted by using narrowband transmission bandwidth, thereby improving the cost performance of voice signal transmission.
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:
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:
An embodiment of the present invention provides a method for generating a downmixed signal, including: performing a time-frequency transform on a received left sound channel signal and a received right sound channel signal to obtain a frequency domain signal, and dividing the frequency domain signal into several frequency bands; calculating a sound channel energy ratio and a sound channel phase difference of each frequency band; calculating a phase difference between the downmixed signal and a first sound channel signal in each frequency band according to the sound channel energy ratio and the sound channel phase difference; and calculating a frequency domain downmixed signal according to the left sound channel signal, the right sound channel signal, and the phase difference between the downmixed signal and the first sound channel signal in each frequency band. This method effectively improves quality of stereo encoding and decoding.
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:
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.