Abstract: A method for recovering lost frame in a media bitstream, where when a frame loss event occurs, a decoder obtains a synthesized high frequency band signal of a current lost frame, and recovery information related to the current lost frame, determines a global gain gradient of the current lost frame, and determines a global gain of the current lost frame according to the global gain gradient and a global gain of each frame in previous M frames of the current lost frame. A high frequency band signal of the current lost frame is obtained by adjusting the synthesized high frequency band signal of the current lost frame according to the global gain and a subframe gain of the current lost frame. Hence, the method enables natural and smooth transitions of the high frequency band signal between the frames, and attenuates noises in the high frequency band 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: 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: 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 not a harmonic type 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: 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: A audio signal encoding method includes: dividing a frequency band of an audio signal into a plurality of sub-bands, and quantifying a sub-band normalization factor of each sub-band; determining signal bandwidth of bit allocation according to the quantified sub-band normalization factor, or according to the quantified sub-band normalization factor and bit rate information; allocating bits for a sub-band within the determined signal bandwidth; and coding a spectrum coefficient of the audio signal according to the bits allocated for each sub-band. According to embodiments of the present disclosure, during coding and decoding, signal bandwidth of bit allocation is determined according to the quantified sub-band normalization factor and bit rate information. In this manner, the determined signal bandwidth is effectively coded and decoded by centralizing the bits, and audio quality is improved.

Abstract: A method for recovering lost frame in a media bitstream is provided. When a frame loss event occurs, a decoder obtains a synthesized high frequency band signal of a current lost frame, and recovery information related to the current lost frame. The decoder determines a global gain gradient of the current lost frame, and further determines a global gain of the current lost frame according to the global gain gradient and a global gain of each frame in previous M frames of the current lost frame. A high frequency band signal of the current lost frame is obtained by adjusting the synthesized high frequency band signal of the current lost frame according to the global gain and a subframe gain of the current lost frame. The process enables natural and smooth transitions of the high frequency band signal between the frames, and attenuates noises in the high frequency band 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: 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 application proposes a frequency envelope vector quantization method and apparatus, where the method includes: dividing N frequency envelopes in one frame into N1 vectors; quantizing a first vector in the N1 vectors by using a first codebook, to obtain a code word corresponding to the quantized first vector, where the first codebook is divided into 2B1 portions; determining, according to the code word corresponding to the quantized first vector; determining a second codebook according to the codebook of the ith portion; and quantizing a second vector in the N1 vectors based on the second codebook. In the embodiments of the present application, vector quantization can be performed on frequency envelope vectors by using a codebook with a smaller quantity of bits. Therefore, complexity of vector quantization can be reduced, and an effect of vector quantization can also be ensured.

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

Abstract: A method, an apparatus, and a system for encoding and decoding multi-channel signals are disclosed. The method for encoding multi-channel signals includes: determining the category of an index corresponding to a channel level difference (CLD) which needs to be quantized in a current frame; quantizing the CLD of at least one frequency band whose index category is the same as the determined category of the index in the current frame, and obtaining quantized data.

Abstract: A signal decoding method and device, where the method includes decoding a bit stream of a voice signal or an audio signal to acquire a decoded signal, predicting an excitation signal of an extension band according to the decoded signal, where the extension band is adjacent to a band of the decoded signal, and the band of the decoded signal is lower than the extension band; selecting a first band and a second band from the decoded signal, and predicting a spectral envelope of the extension band according to a spectral coefficient of the first band and a spectral coefficient of the second band; and determining a frequency-domain signal of the extension band according to the spectral envelope of the extension band and the excitation signal of the extension band.

Abstract: An audio signal encoding method is provided. The method includes: dividing a frequency band of an audio signal into a plurality of sub-bands, and quantifying a sub-band normalization factor of each sub-band; determining signal bandwidth of bit allocation according to the quantified sub-band normalization factor, or according to the quantified sub-band normalization factor and bit rate information; allocating bits for a sub-band within the determined signal bandwidth; and coding a spectrum coefficient of the audio signal according to the bits allocated for each sub-band. According to embodiments of the present invention, during coding and decoding, signal bandwidth of bit allocation is determined according to the quantified sub-band normalization factor and bit rate information. In this manner, the determined signal bandwidth is effectively coded and decoded by centralizing the bits, and audio quality is improved.

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: 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: In an audio signal decoding method, a decoded frequency domain signal of a current frame of the audio signal is obtained by decoding a received bitstream; a predicted frequency domain signal of the current frame is obtained according to the decoded frequency domain signal the current frame when the decoded frequency domain signal meets anyone of two given conditions; and a time domain signal of the current frame is obtained according to the decoded frequency domain signal and the predicted frequency domain 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.

Abstract: A method for recovering a lost frame in a received audio signal includes: obtaining an initial high-frequency band signal of a current lost frame in the received audio signal; calculating a ratio R, wherein the ratio R is a ratio of a high frequency excitation energy of a previous frame of the current lost frame to a high frequency excitation energy of the current lost frame; obtaining a global gain of the current lost frame according to the ratio R and a global gain of the previous frame of the current lost frame; and recovering a high-frequency band signal of the current lost frame according to the initial high-frequency band signal of the current lost frame and the global gain of the current lost frame. The method can be used in an audio signal decoding process for low-loss recovery of lost frames of the audio signal.