Abstract: A audio signal encoding method and apparatus includes: obtaining an audio signal comprising a plurality of sub-bands, wherein each sub-band has an index; obtaining a spectrum energy of each sub-band of at least a part of the plurality of sub-bands; obtaining a highest index of a sub-band to be allocated bits according to the spectrum energy and a ratio factor, wherein the ratio factor is greater than 0 and less than 1; allocating at least one bit for a sub-band having an index no greater than the highest index; and encoding a spectrum coefficient of the sub-band having the index no greater than the highest index with the allocated at least one bit. In this manner, the signal bandwidth is effectively coded and decoded by centralizing the bits.

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: This application discloses an IPD parameter encoding method, including: obtaining a reference parameter used to determine an IPD parameter encoding scheme of a current frame of a multi-channel signal; determining the IPD parameter encoding scheme of the current frame based on the reference parameter, where the determined IPD parameter encoding scheme of the current frame is one of at least two preset IPD parameter encoding schemes; and processing an IPD parameter of the current frame based on the determined IPD parameter encoding scheme of the current frame. The technical solutions provided in this application can improve encoding quality of the multi-channel signal.

Abstract: This application provides an audio rendering method, including: obtaining a to-be-rendered BRIR signal, where an elevation angle corresponding to the to-be-rendered BRIR signal is 0 degrees; obtaining a direct sound signal based on the to-be-rendered BRIR signal; correcting, based on a target elevation angle, a frequency-domain signal corresponding to the direct sound signal, to obtain a frequency-domain signal corresponding to the target elevation angle; obtaining a time-domain signal based on the frequency-domain signal of the target elevation angle; and superposing the time-domain signal on a signal that is in the to-be-rendered BRIR signal and that is in a second time period after a first time period, to obtain a BRIR signal of the target elevation angle.

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: An audio signal decoding device includes a non-transitory memory storage stores audio data in a form of a bitstream; and an audio decoder, by which a first spectral coefficient of a first sub-band of a current frame of an audio signal by decoding the bitstream is obtained; a first average quantity of allocated bits per spectral coefficient of the first sub-band is obtained; a first noise filling gain for the first sub-band is obtained when the first average quantity is less than a threshold; a second spectral coefficient is reconstructed according to the first noise filling gain; a frequency domain audio signal is obtained according to the first spectral coefficient and the second spectral coefficient; and a time domain audio signal is generated according to the frequency domain signal.

Abstract: An audio processing method includes: M first audio signals are obtained by processing a to-be-processed audio signal by M first virtual speakers; N second audio signals are obtained by processing the to-be-processed audio signal by N second virtual speakers; M first head-related transfer functions (HRTFs) centered at a left ear position and N second HRTFs centered at a right ear position are obtained; a first target audio signal is obtained based on the M first audio signals and the M first HRTFs; and a second target audio signal is obtained based on the N second audio signals and the N second HRTFs.

Abstract: An audio processing method includes: M audio signals are obtained by processing an audio signal by M virtual speakers; M first HRTFs and M second HRTFs are obtained, where the M first HRTFs corresponding to a left ear position, and the M second HRTFs corresponding to a right ear position; high-band impulse responses of some of the M first HRTFs are modified to obtain modified first target HRTFs, and high-band impulse responses of some of the M second HRTFs are modified to obtain modified second target HRTFs; a first target audio signal corresponding to the left ear position is obtained based on the modified first target HRTFs and un-modified first HRTFs, and the M audio signals; and a second target audio signal corresponding to the right ear position is obtained based on the modified second HRTFs, un-modified second target HRTFs, and the M audio signals.

Abstract: A speech/audio bitstream decoding method includes 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 using the post-processed speech/audio decoding parameter of the foregoing current speech/audio frame. The technical solutions of the speech/audio bitstream decoding method help improve quality of an output speech/audio signal.

Abstract: This application discloses an IPD parameter encoding method, including: obtaining a reference parameter used to determine an IPD parameter encoding scheme of a current frame of a multi-channel signal; determining the IPD parameter encoding scheme of the current frame based on the reference parameter, where the determined IPD parameter encoding scheme of the current frame is one of at least two preset IPD parameter encoding schemes; and processing an IPD parameter of the current frame based on the determined IPD parameter encoding scheme of the current frame. The technical solutions provided in this application can improve encoding quality of the multi-channel signal.

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.

Abstract: Various embodiments provide a method and an apparatus for determining a weighting factor during stereo signal encoding. In those embodiments, a parameter value corresponding to the encoding mode of the to-be-encoded signal is determining based on an encoding mode of a to-be-encoded signal in a stereo signal and a correspondence between an encoding mode and a parameter value. Based on the determined parameter value and an energy spectrum of a linear prediction filter corresponding to an original line spectral frequency parameter of the to-be-encoded signal is a weighting factor for calculating a distance between the original line spectral frequency parameter and a target original line spectral frequency parameter is calculated.

Abstract: A method and device for decoding a signal, where the method includes: obtaining an average quantity of allocated bits per spectral coefficient of a sub-band of a current frame of the audio signal, wherein the sub-band includes a plurality of spectral coefficients; obtaining a noise filling gain for the sub-band when the average quantity of allocated bits per spectral coefficient is less than a classification threshold; reconstructing, according to the noise filling gain, at least some of the spectral coefficients to generate reconstructed spectral coefficients when the average quantity of allocated bits per spectral coefficient is less than a classification threshold; obtaining a frequency domain signal according to the reconstructed spectral coefficients; and generating a time domain signal based on the frequency domain signal. 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 method and an apparatus for calculating a downmixed signal and a residual signal are provided. According to the method, if a first target frame (a current frame or a previous frame of the current frame) is a switching frame, a to-be-encoded downmixed signal and a to-be-encoded residual signal of the subband corresponding to the preset frequency band in the current frame is calculated based on a switch fade-in/fade-out factor of a second target frame, an initial downmixed signal and an initial residual signal of the preset frequency band.

Abstract: A stereo signal encoding method includes obtaining indication information of an encoding mode of a residual signal of a current frame, where the indication information includes at least one of an encoding status of a residual signal of a previous frame, a value of an updating manner flag for a long-term smooth parameter of the current frame, or a value of a status change parameter of the current frame relative to a stereo signal of the previous frame, and determining the encoding mode of the residual signal of the current frame based on the indication information of the encoding mode of the residual signal of the current frame, where the encoding mode indicates whether to encode the residual signal of the current frame.

Abstract: A stereo signal encoding method includes obtaining a residual signal encoding parameter of a current frame of a stereo signal based on downmixed signal energy and residual signal energy of each of M sub-bands of the current frame, where the residual signal encoding parameter indicates whether to encode residual signals of the M sub-bands, determining whether to encode the residual signals based on the residual signal encoding parameter, and encoding the residual signals when it is determined that the residual signals need to be encoded.

Abstract: This application discloses a downmixed signal calculation method and apparatus. The method includes: when a current frame or a previous frame of the current frame of a stereo signal is not a switching frame and a residual signal in the current frame or the previous frame does not need to be encoded, obtaining a second downmixed signal in the current frame and a downmix compensation factor of the current frame, correcting the second downmixed signal in the current frame based on the downmix compensation factor of the current frame, to obtain the first downmixed signal in the current frame and determining the first downmixed signal in the current frame as a downmixed signal in the current frame in a preset frequency band.

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.

Abstract: A multi-channel signal decoding method includes obtaining a downmixed signal of a first channel signal and a second channel signal in a multi-channel signal, obtaining parameter indication information, obtaining a reverberation gain parameter corresponding to each of the part of subbands according to the parameter indication information; and obtaining a reconstructed first channel signal and a reconstructed second channel signal according to the downmixed signal and the reverberation gain parameter.

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.