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: 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: Embodiments of the present disclosure provide signal encoding and decoding methods and devices. The method includes: determining, a quantity k of subbands to be encoded, where i is a positive number, and k is a positive integer; selecting, according to quantized envelopes of all subbands, k subbands from all the subbands, or selecting k subbands from all subbands according to a psychoacoustic model; and performing a first-time encoding operation on spectral coefficients of the k subbands. In some embodiments of the present disclosure, the quantity k of subbands to be encoded is determined according to the quantity of available bits and the first saturation threshold, and encoding is performed on the k subbands that are selected from all the subbands, instead of on an entire frequency band.

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.

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: 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 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: A method for encoding a multi-channel signal and an encoder, where the encoding method includes obtaining a multi-channel signal of a current frame, determining an initial inter-channel time difference (ITD) value of the current frame, controlling, based on characteristic information of the multi-channel signal, a quantity of target frames that are allowed to appear continuously, where the characteristic information includes at least one of a signal-to-noise ratio of the multi-channel signal or a peak feature of cross correlation coefficients of the multi-channel signal, and an ITD value of a previous frame of the target frame is reused as an ITD value of the target frame, determining an ITD value of the current frame based on the initial ITD value and the quantity of target frames allowed to appear continuously, and encoding the multi-channel signal based on the ITD value of the current frame.

Abstract: Example signal reconstructing method and apparatus are described. One example method includes obtaining a reference sound channel and a target sound channel. An adaptive length of a transition segment is obtained based on an inter-channel time difference in the current frame and an initial length of the transition segment. A transition window in the current frame is obtained based on the adaptive length of the transition segment. A gain modification factor of a reconstructed signal is obtained. A transition segment signal on the target sound channel is obtained based on the inter-channel time difference, the adaptive length of the transition segment, the transition window, the gain modification factor, and a reference sound channel signal and a target sound channel signal.

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 for encoding a multi-channel signal and an encoder, where the encoding method includes obtaining a multi-channel signal of a current frame, determining an initial inter-channel time difference (ITD) value of the current frame, controlling, based on characteristic information of the multi-channel signal, a quantity of target frames that are allowed to appear continuously, where the characteristic information includes at least one of a signal-to-noise ratio of the multi-channel signal or a peak feature of cross correlation coefficients of the multi-channel signal, and an ITD value of a previous frame of the target frame is reused as an ITD value of the target frame, determining an ITD value of the current frame based on the initial ITD value and the quantity of target frames allowed to appear continuously, and encoding the multi-channel signal based on the ITD value of the current frame.

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: 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 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: 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 a non-harmonic type 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 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: 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: 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.