Abstract: A decoder for processing an audio signal receives an audio bitstream, decodes the bitstream to obtain a set of spectral frequency parameters that are arranged in an order of frequencies, determines a minimum spectral frequency parameter difference from a plurality of calculated spectral frequency parameter differences, determines a start frequency bin for predicting a high band excitation signal according to the minimum spectral frequency parameter difference, generates the high band excitation signal by selecting a frequency band with a preset bandwidth selected from a low band excitation signal according to the start frequency bin, and synthesizes a wideband signal based on the generated high band excitation 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: 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 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: An audio signal encoding method and a mobile phone, where the audio signal encoding method includes obtaining a digital audio signal in time domain; transforming the digital audio signal in time domain to an audio signal in frequency domain, which comprises a current frame comprises a plurality of subbands; obtaining, reference parameters of the plurality of subbands; encoding, using a HQ algorithm, the current frame to obtain an encoded audio signal when the reference parameters meet a preset parameter condition; and transmitting the encoded audio signal via a network. The audio signal encoding method and the mobile phone help improve encoding quality or encoding efficiency in audio signal encoding.

Abstract: The present disclosure relates to speech/audio signal processing methods and apparatus. One example speech/audio signal processing method includes when a speech/audio signal switches bandwidth, obtaining an initial high frequency signal corresponding to a current frame of speech/audio signal, obtaining a time-domain global gain parameter of the initial high frequency signal, performing weighting processing on an energy ratio and the time-domain global gain parameter, using an obtained weighted value as a predicted global gain parameter, where the energy ratio is a ratio between energy of a historical frame of high frequency time-domain signal and energy of a current frame of initial high frequency signal, correcting the initial high frequency signal by using the predicted global gain parameter to obtain a corrected high frequency time-domain signal, and synthesizing a current frame of narrow frequency time-domain signal and the corrected high frequency time-domain signal and outputting the synthesized signal.

Abstract: A multi-channel signal encoding method includes determining a downmixed signal of a first channel signal and a second channel signal in a multi-channel signal, and reverberation gain parameters corresponding to different subbands of the first channel signal and the second channel signal, determining a target reverberation gain parameter that needs to be encoded in the reverberation gain parameters corresponding to the different subbands of the first channel signal and the second channel signal, generating parameter indication information, where the parameter indication information is used to indicate a subband corresponding to the target reverberation gain parameter, and encoding the target reverberation gain parameter, the parameter indication information, and the downmixed signal to generate a bitstream.

Abstract: A signal processing method and device includes obtaining spectral coefficients of a current frame of an audio signal, in which N sub-bands of the current frame comprises at least one of the spectral coefficients. A total energy of M successive sub-bands of the N sub-bands, a total energy of K successive sub-bands of the N sub-bands, and an energy of a first sub-band are obtained to determine whether to modify original envelope values of the M sub-bands. When the original envelope values of the M sub-bands are modified, encoding bits are allocated to each of the N sub-bands according to the modified envelope values of the M sub-bands.

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 and device for decoding a signal, where the method for decoding the signal includes obtaining spectral coefficients of sub-bands from a received bitstream by 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 decoding and is in the sub-band with unsaturated bit allocation to restore the spectral coefficient that has not been obtained by decoding, and obtaining a frequency domain signal according to the spectral coefficients obtained by 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: A multi-channel signal encoding method includes determining a downmixed signal of a first channel signal and a second channel signal, an initial reverberation gain parameter of the first channel signal and the second channel signal, determining a target reverberation gain parameter of the first channel signal and the second channel signal based on a correlation between the first channel signal and the downmixed signal, a correlation between the second channel signal and the downmixed signal, and the initial reverberation gain parameter, and quantizing the first channel signal and the second channel signal based on the downmixed signal and the target reverberation gain parameter, and writing a quantized first channel signal and a quantized second channel signal into a bitstream.

Abstract: A method comprises determining a first modification weight according to linear spectral frequency (LSF) differences of the current frame and LSF differences of a previous frame of the current frame, when a signal characteristic of the current frame meets a preset modification condition, modifying the linear predictive parameter of the current frame according to the determined first modification weight, and coding the current frame according to the modified linear predictive parameter.

Abstract: The present disclosure relates methods and apparatus for recovering a lost frame in a received audio signal. One example method includes obtaining an initial high-frequency band signal of a current lost frame in the received audio signal, calculating a ratio R, where 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.

Abstract: Present disclosure provide an audio signal encoding method and communication terminal, which relate to the communications field. The method and communication terminal are used to obtain an analog audio signal and encoding the analog audio signal to obtain a bitstream representing the analog audio signal, in which a proper bit allocation for spectral coefficients can be performed.

Abstract: An audio encoding method and a related apparatus, where the audio coding method includes obtaining an audio signal, obtaining some subband parameters of a current frame of the audio signal, and encoding the current frame using a high quality transform coding (HQ) algorithm based on the obtained subband parameters. The audio encoding method and the related apparatus help improve encoding quality or encoding efficiency in audio signal encoding.

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: This application provides an audio signal encoding and decoding method. An audio signal encoder encodes a low frequency band signal of a received audio signal, to obtain one or more low frequency encoding parameters. A voiced factor is determined according to the one or more low frequency encoding parameters. The encoder obtains a high frequency band excitation signal according to the one or more low frequency encoding parameters and the voiced factor. The encoder further obtains a synthesized excitation signal according to the high frequency band excitation signal and the voiced factor, and a high frequency encoding parameter according to the synthesized excitation signal and a high frequency band signal of the received audio signal. The encoder outputs a bitstream that includes the high frequency encoding parameter and the one or more low frequency encoding parameters. An audio signal decoder obtains the audio signal from the bitstream through reversed steps.

Abstract: Embodiments of the present application 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: 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: A method for encoding a high frequency signal includes determining a signal type of a high frequency signal of a current frame, smoothing and scaling time envelopes of the high frequency signal of the current frame and obtaining time envelopes of the high frequency signal of the current frame that require to be encoded when the high frequency signal of the current frame is a non-transient signal and a high frequency signal of the previous frame is a transient signal, and quantizing and encoding the time envelopes of the high frequency signal of the current frame that require to be encoded, and frequency information and signal type information of the high frequency signal of the current frame.