WO2024094006A1 - 一种音频信号的编码、解码方法及装置 - Google Patents

一种音频信号的编码、解码方法及装置 Download PDF

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WO2024094006A1
WO2024094006A1 PCT/CN2023/128523 CN2023128523W WO2024094006A1 WO 2024094006 A1 WO2024094006 A1 WO 2024094006A1 CN 2023128523 W CN2023128523 W CN 2023128523W WO 2024094006 A1 WO2024094006 A1 WO 2024094006A1
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signal
frequency
low
target
residual signal
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PCT/CN2023/128523
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English (en)
French (fr)
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林坤鹏
张德军
伍子谦
蒋佳为
王鹤
肖益剑
丁飘
宋慎义
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抖音视界有限公司
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/26—Pre-filtering or post-filtering

Definitions

  • the present application relates to the field of data processing technology, and in particular to a method and device for encoding and decoding an audio signal.
  • the bandwidth extension algorithm can use most of the bit rate to encode low-frequency signals that the human ear is more sensitive to, within the bit rate limit, while transmitting high-frequency signals that the human ear pays less attention to using a lower bit rate, or only rely on restoring high-frequency signals based on decoded low-frequency signals at the decoding end, thereby improving the overall quality of encoded speech at a fixed bit rate.
  • the spectrum of the high-frequency signal is generally generated by folding the spectrum of the low-frequency signal. Therefore, the recovered audio frame signal will lack some harmonic components.
  • the high-frequency energy is attenuated when the high-frequency signal energy is restored at the decoding end, resulting in low high-frequency energy after recovery and poor listening quality of the overall audio frame.
  • an embodiment of the present application provides an audio signal encoding and decoding method and device for improving the audio quality of audio signals after processing.
  • an embodiment of the present application provides a method for encoding an audio signal, comprising:
  • the center frequency of the target frequency range is the fundamental frequency of the low-frequency residual signal
  • Encoded data of the target audio frame is generated according to the high frequency energy gain.
  • suppressing the frequency components within the target frequency range in the low-frequency residual signal to obtain a coding suppression signal includes:
  • the low-frequency residual signal is pre-emphasized based on a high-pass filter to suppress frequency components within a target frequency range in the low-frequency residual signal to obtain a coding suppression signal.
  • suppressing the frequency components within the target frequency range in the low-frequency residual signal to obtain a coding suppression signal includes:
  • the low-frequency residual signal is filtered based on a shelving filter to suppress frequency components within a target frequency range in the low-frequency residual signal, thereby obtaining a coding suppression signal.
  • suppressing the frequency components within the target frequency range in the low-frequency residual signal to obtain a coding suppression signal includes:
  • the coding notch signal is subjected to whitening processing to obtain the coding suppression signal.
  • performing spectrum inversion on the coding suppression signal to obtain a spectrum inversion signal includes:
  • the amplitude of the sampling point with an odd index in the coding suppression signal is modified to the opposite number to obtain a spectrum inverted signal.
  • the obtaining of the high-frequency residual signal and the low-frequency residual signal of the target audio signal includes:
  • the low-frequency signal is encoded to obtain low-frequency encoding information and the low-frequency residual signal.
  • generating the encoded data of the target audio frame according to the high-frequency energy gain includes:
  • the low-frequency coding information, the line spectrum pair coefficients, and the high-frequency energy gain are encoded to generate encoded data of the target audio frame.
  • an embodiment of the present application provides a method for decoding an audio signal, comprising:
  • the center frequency of the target frequency range is the fundamental frequency of the low-frequency residual signal
  • An audio signal of the target audio frame is generated according to the low-frequency signal and the high-frequency signal.
  • the low-frequency residual The frequency components within the target frequency range in the signal are suppressed to obtain a decoded suppressed signal, including:
  • suppressing the frequency components within the target frequency range in the low-frequency residual signal to obtain a decoded suppressed signal includes:
  • the low-frequency residual signal is filtered based on a shelving filter to suppress frequency components within a target frequency range in the low-frequency residual signal, thereby obtaining a decoded suppressed signal.
  • suppressing the frequency components within the target frequency range in the low-frequency residual signal to obtain a decoded suppressed signal includes:
  • the decoded notch signal is whitened to obtain the decoded suppression signal.
  • the performing spectrum inversion on the decoded suppression signal to obtain the low-frequency excitation signal includes:
  • the amplitudes of sampling points with odd indexes in the decoded suppression signal are modified to opposite numbers to obtain a spectrum inversion signal.
  • the encoded data of the target audio frame further includes: LSP coefficients and high-frequency energy gains;
  • a signal is reconstructed according to the low-frequency excitation signal, the LSP coefficient and the high-frequency energy gain to obtain the high-frequency signal.
  • the signal is reconstructed according to the low-frequency excitation signal, the LSP coefficient and the high-frequency energy gain.
  • Acquire high frequency signals including:
  • the high-frequency signal is generated according to each sub-signal.
  • an audio signal encoding device including:
  • An acquisition unit used to acquire a high-frequency residual signal and a low-frequency residual signal of a target audio frame
  • a suppression unit configured to suppress frequency components within a target frequency range in the low-frequency residual signal to obtain a coded suppression signal; the center frequency of the target frequency range is the fundamental frequency of the low-frequency residual signal;
  • an inversion unit configured to perform spectrum inversion on the coding suppression signal to obtain a spectrum inversion signal
  • a processing unit configured to obtain a high-frequency energy gain of the target audio signal according to the spectrum inversion signal and the high-frequency residual signal
  • a generating unit is used to generate the encoding data of the target audio frame according to the high-frequency energy gain.
  • the suppression unit is specifically used to perform pre-emphasis processing on the low-frequency residual signal based on a high-pass filter to suppress frequency components within a target frequency range in the low-frequency residual signal to obtain a coded suppression signal.
  • the suppression unit is specifically used to filter the low-frequency residual signal based on a ramp filter to suppress frequency components within a target frequency range in the low-frequency residual signal to obtain a coded suppression signal.
  • the suppression unit is specifically used to perform notch processing on the frequency components within the target frequency range based on a second-order notch filter to obtain a coded notch signal, and to perform whitening processing on the coded notch signal to obtain the coded suppression signal.
  • the inversion unit is specifically used to modify the amplitude of the sampling point with an odd index in the coding suppression signal to an opposite number to obtain a spectrum inversion signal.
  • the acquisition unit is specifically used to:
  • the low-frequency signal is encoded to obtain low-frequency encoding information and the low-frequency residual signal.
  • the generating unit is specifically used to encode the low-frequency coding information, the line spectrum pair coefficients and the high-frequency energy gain to generate the coding data of the target audio frame.
  • an embodiment of the present application provides a decoding device for an audio signal, including:
  • An acquisition unit parsing the coded data of the target audio frame to obtain low-frequency coding information
  • a decoding unit decoding the low-frequency coded information to obtain a low-frequency signal and a low-frequency residual signal
  • a suppression unit suppresses frequency components within a target frequency range in the low-frequency residual signal to obtain a decoded suppression signal; the center frequency of the target frequency range is the fundamental frequency of the low-frequency residual signal;
  • an inversion unit performing spectrum inversion on the decoded suppression signal to obtain a low-frequency excitation signal
  • a reconstruction unit reconstructing a signal according to the low-frequency excitation signal to obtain a high-frequency signal
  • a generating unit generates an audio signal of the target audio frame according to the low-frequency signal and the high-frequency signal.
  • the suppression unit is specifically used to: perform pre-emphasis processing on the low-frequency residual signal based on a high-pass filter to suppress frequency components within a target frequency range in the low-frequency residual signal to obtain a decoded suppression signal.
  • the suppression unit is specifically used to filter the low-frequency residual signal based on a ramp filter to suppress frequency components within a target frequency range in the low-frequency residual signal to obtain a decoded suppression signal.
  • the suppression unit is specifically used to perform notch processing on the frequency components within the target frequency range based on a second-order notch filter to obtain a decoded notch signal, and to perform whitening processing on the decoded notch signal to obtain the decoded suppression signal.
  • the inversion unit is specifically used to modify the amplitude of the sampling point with an odd index in the decoded suppression signal to an opposite number to obtain a low-frequency excitation signal.
  • the encoded data of the target audio frame also includes: LSP coefficients and high-frequency energy gains, and the reconstruction unit is specifically used to reconstruct the signal according to the low-frequency excitation signal, the LSP coefficients and the high-frequency energy gain to obtain the high-frequency signal.
  • the reconstruction unit is specifically used to obtain the energy gain corresponding to each sub-signal in the high-frequency energy gain; obtain the residual signal of each sub-signal according to the low-frequency excitation signal and the energy gain of each sub-signal; restore the LSP coefficient to the LPC coefficient; obtain each sub-signal according to the LPC coefficient predicting sub-signals; generating each sub-signal according to each predicted sub-signal and a residual signal of each sub-signal; and generating the high-frequency signal according to each sub-signal.
  • an embodiment of the present application provides an electronic device, comprising: a memory and a processor, wherein the memory is used to store a computer program; and the processor is used to enable the electronic device to implement the audio signal encoding method or the audio signal decoding method described in any of the above-mentioned embodiments when executing the computer program.
  • an embodiment of the present application provides a computer-readable storage medium.
  • the computing device implements the audio signal encoding method or the audio signal decoding method described in any of the above embodiments.
  • an embodiment of the present application provides a computer program product.
  • the computer program product When the computer program product is run on a computer, the computer implements the audio signal encoding method or the audio signal decoding method described in any of the above embodiments.
  • the audio signal encoding method provided in the embodiment of the present application obtains the high-frequency residual signal and the low-frequency residual signal of the target audio frame, and then suppresses the frequency components within the target frequency range in the low-frequency residual signal to obtain the encoding suppression signal; performs spectrum inversion on the encoding suppression signal to obtain the spectrum inversion signal; then, obtains the high-frequency energy gain of the target audio signal according to the spectrum inversion signal and the high-frequency residual signal; and finally generates the encoding data of the target audio frame according to the high-frequency energy gain.
  • the embodiment of the present application suppresses and inverts the frequency components of the acquired low-frequency residual signal, and then combines the high-frequency residual signal and the high-frequency energy gain to obtain the encoding data of the target audio frame, thereby ensuring that the reconstructed high-frequency signal will not lack harmonic components and have low energy. In this way, the problem of poor audio quality when obtaining the code stream data of the target audio frame can be avoided, thereby improving the user experience. Therefore, the embodiment of the present application can improve the quality of audio during the encoding and decoding process.
  • FIG1 is a flowchart of a method for encoding an audio signal according to an embodiment of the present application
  • FIG2 is a second flowchart of the audio signal encoding method provided in an embodiment of the present application.
  • FIG3 is a third flowchart of the audio signal encoding method provided in an embodiment of the present application.
  • FIG4 is a fourth flowchart of the audio signal encoding method provided in an embodiment of the present application.
  • FIG5 is a hardware block diagram of an audio signal encoding device provided in an embodiment of the present application.
  • FIG6 is a flowchart of a method for decoding an audio signal according to an embodiment of the present application.
  • FIG7 is a second flowchart of the audio signal decoding method provided in an embodiment of the present application.
  • FIG8 is a third flowchart of the audio signal decoding method provided in an embodiment of the present application.
  • FIG9 is a fourth flowchart of the audio signal decoding method provided in an embodiment of the present application.
  • FIG10 is a hardware block diagram of an audio signal decoding device provided in an embodiment of the present application.
  • FIG11 is a schematic diagram of the structure of an audio signal encoding device provided in an embodiment of the present application.
  • FIG12 is a schematic diagram of the structure of an audio signal decoding device provided in an embodiment of the present application.
  • FIG. 13 is a schematic diagram of the structure of an electronic device provided in an embodiment of the present application.
  • words such as “exemplary” or “for example” are used to indicate examples, illustrations or descriptions. Any embodiment or design scheme described as “exemplary” or “for example” in the embodiments of the present application should not be interpreted as being superior to other embodiments or designs. Specifically, the use of words such as “exemplary” or “for example” is intended to present the related concepts in a specific way. In addition, in the description of the embodiments of the present application, unless otherwise specified, the meaning of "multiple” refers to two or more.
  • the embodiment of the present application provides a method for encoding an audio signal. As shown in FIG1 , the method for encoding an audio signal includes the following steps:
  • S101 Acquire a high-frequency residual signal and a low-frequency residual signal of a target audio frame.
  • the high-frequency residual signal refers to the difference between the value of each sample point of the high-frequency signal of the audio signal and the corresponding prediction value, and the prediction value corresponding to each sample point is the product of the linear prediction coefficient (Linear Prediction Coefficient, LPC) and the low-frequency signal of the historical audio signal;
  • the low-frequency residual signal refers to the difference between the value of each sample point of the low-frequency signal of the audio signal and the corresponding prediction value, and the prediction value corresponding to each sample point is the product of the linear prediction coefficient and the low-frequency signal of the historical audio signal.
  • the linear prediction coefficient means that the sample value of the audio signal can be approximated by a linear combination of the sample value of the historical audio data multiplied by the coefficient and the sum of the products. For example: if the LPC order is 10, there are 10 coefficients. Using these 10 coefficients to multiply the 10 sample values of the historical audio data respectively and summing the products can approximate the current sample value, and these coefficients are the linear prediction coefficients.
  • the method for obtaining the high-frequency residual signal and the low-frequency residual signal can be the same as the prior art.
  • the implementation method of obtaining the high-frequency residual signal and the low-frequency residual signal of the target audio frame in the embodiment of the present application is not limited, and the main thing is to be able to obtain the high-frequency residual signal and the low-frequency residual signal of the target audio frame.
  • S102 Suppress frequency components within a target frequency range in the low-frequency residual signal to obtain a coded suppressed signal.
  • the center frequency of the target frequency range is the fundamental frequency of the low-frequency residual signal.
  • Fundamental frequency is also called baseband or baseband, which refers to the frequency of the fundamental tone in a complex tone.
  • baseband refers to the frequency of the fundamental tone in a complex tone.
  • the fundamental tone has the lowest frequency and the greatest intensity.
  • the height of the fundamental frequency determines the height of a tone.
  • the target frequency range when the center frequency of the target frequency range is 20 kilohertz (Khz), the target frequency range may be: ⁇ 10Khz, 30Khz ⁇ ; when the center frequency of the target frequency range is If the frequency is 40Khz, the target frequency range can be: ⁇ 20Khz, 60Khz ⁇ .
  • the sampled baseband spectrum (center frequency is near 0 Hz) obtained according to the formula of the bandpass sampling theorem is exactly opposite to the shape of the positive and negative spectra of the original signal. Therefore, in the embodiment of the present application, the spectrum of the coding suppression signal is inverted, that is, the spectrum inversion signal is opposite to the shape of the positive and negative spectra of the coding suppression signal.
  • S104 Acquire a high-frequency energy gain of the target audio signal according to the spectrum inversion signal and the high-frequency residual signal.
  • the high-frequency energy gain refers to the energy gain of the high-frequency residual signal, specifically, the energy ratio of the high-frequency residual signal to the low-frequency residual signal.
  • the gain value refers to the energy offset between the high-frequency signal and the low-frequency signal.
  • the implementation method of obtaining the high-frequency energy gain of the target audio signal may include:
  • the energy value of the spectrum inversion signal and the energy value of the high-frequency residual signal are obtained, and the ratio of the energy value of the spectrum inversion signal to the energy value of the high-frequency residual signal is calculated to obtain the high-frequency energy gain of the target audio signal.
  • S105 Generate encoding data of the target audio frame according to the high-frequency energy gain.
  • the audio signal encoding method provided in the embodiment of the present application obtains the high-frequency residual signal and the low-frequency residual signal of the target audio frame, and then suppresses the frequency components within the target frequency range in the low-frequency residual signal to obtain the encoding suppression signal; inverts the frequency components that meet the preset conditions in the encoding suppression signal to obtain the spectrum inversion signal; then, according to the spectrum inversion signal and the high-frequency residual signal, obtains the high-frequency energy gain of the target audio signal; and finally generates the encoding data of the target audio frame according to the high-frequency energy gain.
  • the embodiment of the present application suppresses and inverts the frequency components of the acquired low-frequency residual signal, thereby ensuring that the reconstructed high-frequency signal will not lack harmonic components and have low energy. In this way, the problem of poor audio quality when obtaining the code stream data of the target audio frame can be avoided, thereby improving the user experience. Therefore, the embodiment of the present application can improve the audio quality during the encoding process.
  • the method for encoding an audio signal comprises the following steps:
  • the target audio frame can be divided into a low-frequency signal and a high-frequency signal by a quadrature mirror filter (QMF), wherein the frequency range of the low-frequency signal can be [0kHz-4kHz], and the frequency range of the high-frequency signal can be [4kHz-8kHz].
  • QMF quadrature mirror filter
  • the high frequency signal may be subjected to linear prediction analysis by a Burg algorithm to obtain a first linear prediction coefficient.
  • the Burg algorithm is a recursive algorithm that directly calculates a power spectrum estimate from a known time signal sequence.
  • the implementation method of converting the first linear prediction coefficient into the line spectrum pair coefficient may be the same as the implementation method of converting the LPC coefficient into the LSP coefficient in the prior art, and the embodiment of the present application does not limit this.
  • the method for restoring the line spectrum pair coefficients to the second linear prediction coefficients may be the same as the method for restoring the LSP coefficients to the LPC coefficients in the prior art, and the embodiments of the present application do not limit this.
  • the high-frequency signal can be evenly divided into any number of sub-signals according to the actual encoding process.
  • the high-frequency signal can be divided into 4 sub-signals of equal length, or for another example, the high-frequency signal can be divided into 8 sub-signals of equal length.
  • S206 Perform filtering processing on each sub-signal based on the second linear prediction coefficient to obtain a residual signal of each sub-signal to obtain the high-frequency residual signal.
  • the transfer function of the linear prediction filter that performs filtering processing on each sub-signal based on the second linear prediction coefficient may be:
  • the residual signal of the sub-signal obtained by this transfer function is:
  • i is the index of the sub-signal
  • x hb represents the original sub-signal
  • a i is the linear prediction coefficient of the sub-signal with index i
  • res hb is the residual signal of the sub-signal with index i.
  • S207 Encode the low-frequency signal to obtain low-frequency encoding information and the low-frequency residual signal.
  • the low-frequency signal may be encoded by an encoder SILK encoder to obtain the low-frequency encoding information and the low-frequency residual signal.
  • a high-pass filter will be used to perform pre-emphasis processing on each sub-signal based on the second linear prediction coefficient.
  • the high-pass filter is used to suppress prominent frequency components near the fundamental frequency.
  • ⁇ is the pre-filtering coefficient
  • ⁇ determines the degree of suppression of lower frequency components in the low-frequency residual signal and the degree of emphasis on higher frequency components. The larger the ⁇ value, the higher the degree of suppression of lower frequency components and the higher the degree of emphasis on higher frequency components.
  • S209 Modify the amplitudes of the sampling points with odd indexes in the coding suppression signal to opposite numbers to obtain a spectrum inversion signal.
  • the coded suppression signal needs to be spectrum inverted.
  • the sampling points with odd indexes in the coding suppression signal are negated to obtain a spectrum inversion signal.
  • i is the index of the sampling point in the coding suppression signal.
  • the spectrum inversion signal obtained after calculation by the above formula is: ⁇ -a 1 , a 2 , -a 3 , ... a 64 ⁇ .
  • S210 Acquire a high-frequency energy gain of the target audio signal according to the spectrum inversion signal and the high-frequency residual signal.
  • the high-frequency energy gain includes the energy gain of each sub-signal.
  • the energy gain value of the sub-signal of index i is:
  • gain i is the energy gain value of the sub-signal with index i
  • the low-frequency coding information, the LSP coefficient and the high-frequency energy gain are encapsulated into an audio signal packet to obtain the coding data of the target audio frame.
  • the audio signal encoding method provided by the embodiment of the present invention further includes: performing double codebook quantization on the LSP coefficient.
  • the LSP coefficients are first quantized using dual codebooks, and then the corresponding codebook index is encoded into the main code stream using 12 bits.
  • the dual codebook quantization is to retrieve the obtained LSP coefficient through two different codebooks to obtain the LSP coefficient and the subscript code of the corresponding codebook, and synthesize the new LSP coefficient subscript code through the two codebook retrieval.
  • the data volume of the LSP coefficients can be reduced, thereby improving the efficiency of audio signal transmission.
  • the audio signal encoding method provided by the embodiment of the present invention further includes: performing codebook quantization on the high frequency energy gain.
  • the high-frequency energy gain may be quantified and the corresponding index may be encoded into the main stream using 5 bits.
  • the encoding data of the high-frequency energy gain consumes 20 bits in total.
  • the data volume of the high-frequency energy gain can be reduced, thereby improving the efficiency of audio signal transmission.
  • the embodiment of the present application provides another method for encoding an audio signal. As shown in FIG. 3 , the method for encoding an audio signal includes the following steps:
  • S301 Divide the target audio frame into a low-frequency signal and a high-frequency signal.
  • S302 Perform linear prediction analysis on the high-frequency signal to obtain a first linear prediction coefficient.
  • S306 Perform filtering processing on each sub-signal based on the second linear prediction coefficient to obtain a residual signal of each sub-signal to obtain the high-frequency residual signal.
  • S307 Encode the low-frequency signal to obtain low-frequency encoding information and the low-frequency Residual signal.
  • steps S301 to S307 may be the same as the implementation of steps S201 to S207 in the embodiment shown in FIG. 2 , and will not be described in detail here to avoid redundancy.
  • S308 Filter the low-frequency residual signal based on a shelving filter to suppress frequency components within a target frequency range in the low-frequency residual signal to obtain a coding suppression signal.
  • the low-frequency residual signal is filtered by a shelving filter, thereby suppressing the frequency components within the target frequency range whose center frequency is the fundamental frequency of the low-frequency residual signal.
  • the shelving filter transfer function may be as follows:
  • fc represents the frequency to be adjusted
  • the filter suppression frequency range and suppression degree can be specified according to the degree of spectral tilt to reduce the spectral tilt of the low-frequency residual.
  • the spectrum of the obtained high-frequency signal is inverted.
  • the low-frequency residual signal needs to be spectrum inverted.
  • the implementation method of spectrum inversion of the coding suppression signal is the same as that described in step S209 above, and will not be repeated here.
  • step S310 may be the same as the implementation method of step S210 in the embodiment shown in FIG. 2 , and will not be described in detail here to avoid redundancy.
  • the embodiment of the present application provides another method for encoding an audio signal.
  • the method for encoding an audio signal includes the following steps:
  • S401 Divide the target audio frame into a low-frequency signal and a high-frequency signal.
  • S402 Perform linear prediction analysis on the high-frequency signal to obtain a first linear prediction coefficient.
  • S405 Divide the high-frequency signal into a preset number of sub-signals on average.
  • S406 Perform filtering processing on each sub-signal based on the second linear prediction coefficient to obtain a residual signal of each sub-signal to obtain the high-frequency residual signal.
  • S407 Encode the low-frequency signal to obtain low-frequency encoding information and the low-frequency residual signal.
  • steps S401 to S407 may be the same as the implementation of steps S201 to S207 in the embodiment shown in FIG. 2 , and will not be described in detail here to avoid redundancy.
  • S408 Perform notch processing on the frequency components within the target frequency range based on a second-order notch filter to obtain a coded notch signal.
  • the filter performs notch processing on the frequency components within the target frequency range to obtain a coded notch signal.
  • the spectrum inversion signal mainly has higher frequency components near the fundamental frequency (within the target frequency range)
  • the spectrum inversion signal is passed through a second-order notch filter to perform notch processing on the frequency components within the target frequency range.
  • the transfer function of the second-order notch filter is as follows:
  • bw represents the notch bandwidth of the filter
  • ⁇ 0 represents the center frequency point of the notch filter
  • G represents the notch gain value at the specified frequency
  • S409 Perform whitening processing on the coded notch signal to obtain the coded suppression signal.
  • the implementation of whitening the coded notch signal includes:
  • the LPC coefficients of the low-frequency residual signal are obtained through the Berg algorithm.
  • the coding notch signal obtained by the above steps is subjected to high-order LPC filtering using the LPC coefficient to obtain a coding suppression signal.
  • the high-order LPC filter is 8th order, it can be calculated by the following formula:
  • S410 Modify the amplitudes of sampling points with odd indexes in the coding suppression signal to opposite numbers to obtain a spectrum inversion signal.
  • S412 Encode the low-frequency coding information, the line spectrum pair coefficients, and the high-frequency energy gain to generate coding data of the target audio frame.
  • the audio signal encoding device includes: an orthogonal mirror filter 501, an encoder 502, a suppression module 503, an inversion module 504, a splitting module 505, a linear prediction analyzer 506, a parameter quantizer 507, a restoration module 508, a high-frequency residual generator 509, a gain calculator 510, and a packager 511.
  • the orthogonal mirror filter 501 is used to divide a single frame audio signal into a low frequency (Low Band, LB) signal and a high frequency (High Band, HB) signal.
  • the encoder 502 is used to encode the low-frequency signal to generate low-frequency coding information and a low-frequency residual signal.
  • the suppression module 503 is used to suppress the frequency components within the target frequency range whose center frequency is the fundamental frequency of the low-frequency residual signal to obtain a coded suppression signal.
  • the inversion module 504 is used to perform spectrum inversion on the coding suppression signal to obtain a spectrum inversion signal.
  • the splitting module 505 is used to evenly divide the high-frequency signal of a single frame into a preset number of sub-signals.
  • the linear prediction analyzer 506 is used to perform linear prediction analysis on the high frequency signal to obtain a first LPC coefficient of the high frequency signal.
  • the parameter quantizer 507 is used to convert the first linear prediction coefficient into an LSP coefficient.
  • the restoration module 508 is used to restore the LSP coefficients to the second linear prediction coefficients.
  • the high frequency residual generator 509 generates a residual signal of each sub-signal according to the second linear prediction coefficient and each sub-signal to obtain a high frequency residual signal.
  • the gain calculator 510 calculates a high energy gain value according to the spectrum inverted signal and the high frequency residual signal.
  • the encapsulator 511 is used to encapsulate the low-frequency coding information, the LSP coefficients and the high-frequency energy gain to generate coding data of the audio signal.
  • Another embodiment of the present application provides a method for decoding an audio signal. As shown in FIG. 6 , the method for decoding an audio signal includes the following steps:
  • S601 parse the coded data of the target audio frame to obtain low-frequency coding information.
  • the coded data of the received audio frame is decapsulated to obtain the low-frequency coding information carried in the coded data.
  • S602 Decode the low-frequency coding information to obtain a low-frequency signal and a low-frequency residual signal.
  • the low-frequency coding information may be decoded by a decoder to obtain the low-frequency signal and the low-frequency residual signal.
  • S603 Suppress frequency components within a target frequency range in the low-frequency residual signal to obtain a decoded suppressed signal.
  • the center frequency of the target frequency range is the fundamental frequency of the low-frequency residual signal.
  • S604 Perform spectrum inversion on the decoded suppression signal to obtain a low-frequency excitation signal.
  • S605 Reconstruct a signal according to the low-frequency excitation signal to obtain a high-frequency signal.
  • S606 Generate an audio signal of the target audio frame according to the low-frequency signal and the high-frequency signal.
  • the audio signal decoding method provided in the embodiment of the present application obtains low-frequency coding information by parsing the coding data of the target audio frame, then decodes the low-frequency coding information to obtain a low-frequency signal and a low-frequency residual signal, then suppresses the frequency components within the target frequency range in the low-frequency residual signal, then performs spectrum inversion on the obtained decoded suppression signal to obtain a low-frequency excitation signal, and reconstructs the signal according to the low-frequency excitation signal to obtain a high-frequency signal. Finally, the audio signal of the target audio frame is generated according to the low-frequency signal and the high-frequency signal.
  • the embodiment of the present application suppresses the spectrum of the low-frequency excitation signal without attenuating the high-frequency signal, the problem of low energy of the high-frequency signal is avoided, and because the embodiment of the present application also reverses the spectrum value of the sampling point that meets the preset conditions when reconstructing the high-frequency signal, the problem of lack of harmonic components of the high-frequency signal is avoided.
  • the embodiment of the present application can avoid low high-frequency energy and lack of high-frequency harmonics when reconstructing the high-frequency signal at the decoding end, so the embodiment of the present application can improve the audio quality.
  • the embodiment of the present application provides another method for decoding an audio signal.
  • the method for decoding an audio signal includes the following steps:
  • S702 Decode the low-frequency coding information to obtain a low-frequency signal and a low-frequency residual signal.
  • S703 Perform pre-emphasis processing on the low-frequency residual signal based on a high-pass filter to suppress frequency components within a target frequency range in the low-frequency residual signal to obtain a coded suppression signal.
  • S704 Modify the amplitudes of sampling points with odd indexes in the decoded suppression signal to opposite numbers to obtain a low-frequency excitation signal.
  • the step of reconstructing a signal according to the low-frequency excitation signal, the LSP coefficient, and the high-frequency energy gain to obtain a high-frequency signal includes the following steps 1 to 6:
  • Step 1 Obtain the energy gain corresponding to each sub-signal in the high-frequency energy gain.
  • Step 2 Obtain a residual signal of each sub-signal according to the low-frequency excitation signal and the energy gain of each sub-signal.
  • Step 3 Restore the LSP coefficients to LPC coefficients.
  • Step 4 Obtain each prediction sub-signal according to the LPC coefficient.
  • Step 5 Generate each sub-signal according to each predicted sub-signal and the residual signal of each sub-signal.
  • Step 6 Generate the high-frequency signal according to each sub-signal.
  • S706 Generate an audio signal of the target audio frame according to the low-frequency signal and the high-frequency signal.
  • the low-frequency signal and the high-frequency signal may be synthesized by using an orthogonal mirror filter to generate an audio signal of the target audio frame.
  • the embodiment of the present application provides another method for decoding an audio signal. As shown in FIG. 8 , the frequency components within a target frequency range in the low-frequency residual signal are suppressed to obtain a decoded suppressed signal.
  • the method for decoding an audio signal includes the following steps:
  • S802 Decode the low-frequency coding information to obtain a low-frequency signal and a low-frequency residual signal.
  • S803 Filter the low-frequency residual signal based on a shelving filter to suppress frequency components within a target frequency range in the low-frequency residual signal, and obtain a decoded suppressed signal.
  • S804 Modify the amplitudes of the sampling points with odd indexes in the decoded suppression signal to opposite numbers to obtain a low-frequency excitation signal.
  • S806 Generate an audio signal of the target audio frame according to the low-frequency signal and the high-frequency signal.
  • the embodiment of the present application provides another method for decoding an audio signal. As shown in FIG. 9, the method suppresses the frequency components within the target frequency range in the low-frequency residual signal.
  • the decoding suppression signal is obtained by the decoding method of the audio signal, and the decoding method of the audio signal comprises the following steps:
  • S901 parse the coded data of the target audio frame to obtain low-frequency coding information, LSP coefficients, and high-frequency energy gain.
  • S902 Decode the low-frequency coding information to obtain a low-frequency signal and a low-frequency residual signal.
  • S904 Perform whitening processing on the decoded notch signal to obtain the decoded suppression signal.
  • S907 Generate an audio signal of the target audio frame according to the low-frequency signal and the high-frequency signal.
  • Figure 10 is a hardware block diagram of a decoding device for an audio signal provided in an embodiment of the present application, and the decoding device includes: a decapsulator 101, a decoder 102, a suppression module 103, an inversion module 104, a residual generator 105, a restoration module 106, a prediction module 107, a reconstruction module 108, a splicing module 109 and an orthogonal mirror filter 1010.
  • the decapsulator 101 is used to parse and obtain low-frequency coding information, LSP coefficients and high-frequency energy gains.
  • the decoder 102 is used to decode the low-frequency coded information to obtain a low-frequency signal and a low-frequency residual signal.
  • the suppression module 103 is used to suppress the frequency components within the target frequency range whose center frequency is the fundamental frequency of the low-frequency residual signal to obtain a decoded suppression signal.
  • the inversion module 104 is used to perform spectrum inversion on the decoded suppression signal to obtain a low-frequency excitation signal.
  • the residual generator 105 is used to generate a residual error according to the low-frequency excitation signal and the high-frequency energy gain.
  • the energy gain corresponding to each sub-signal in the gain is obtained to obtain the residual signal of each sub-signal.
  • the restoration module 106 is used to restore the LSP coefficients to LPC coefficients.
  • the prediction module 107 is used to obtain each high frequency sub-signal according to the LPC coefficient.
  • the reconstruction module 108 is used to generate each sub-signal according to each predicted sub-signal and a residual signal of each sub-signal.
  • the splicing module 109 is used to splice the sub-signals into a high-frequency signal.
  • the orthogonal mirror filter 1010 is used to synthesize the high-frequency signal and the low-frequency signal into an audio signal.
  • the embodiment of the present application further provides an audio signal encoding device and an audio signal decoding device, which corresponds to the above method embodiment.
  • an audio signal encoding device and an audio signal decoding device which corresponds to the above method embodiment.
  • the embodiment of the present application no longer repeats the details of the above method embodiment one by one, but it should be clear that the audio signal decoding device and the audio signal decoding device in the embodiment of the present application can correspond to the implementation of all the contents in the above method embodiment.
  • FIG11 is a structural schematic diagram of the audio signal encoding device. As shown in FIG11, the audio signal encoding device 1100 includes:
  • the acquisition unit 1101 is used to acquire a high-frequency residual signal and a low-frequency residual signal of a target audio frame.
  • the suppression unit 1102 is used to suppress the frequency components within the target frequency range in the low-frequency residual signal to obtain a coded suppression signal; the center frequency of the target frequency range is the fundamental frequency of the low-frequency residual signal.
  • the inversion unit 1103 is used to perform spectrum inversion on the coding suppression signal to obtain a spectrum inversion signal.
  • the processing unit 1104 is configured to obtain a high-frequency energy gain of the target audio signal according to the spectrum inversion signal and the high-frequency residual signal.
  • the generating unit 1105 is configured to generate the encoding data of the target audio frame according to the high frequency energy gain.
  • the suppression unit 1102 Specifically used for:
  • the low-frequency residual signal is pre-emphasized based on a high-pass filter to suppress frequency components within a target frequency range in the low-frequency residual signal to obtain a coding suppression signal.
  • the suppression unit 1102 is specifically used to filter the low-frequency residual signal based on a slope filter to suppress the frequency components within the target frequency range in the low-frequency residual signal to obtain a coded suppression signal.
  • the suppression unit 1102 is specifically used to perform notch processing on the frequency components within the target frequency range based on a second-order notch filter to obtain a coded notch signal, and to perform whitening processing on the coded notch signal to obtain the coded suppression signal.
  • the inversion unit 1103 is specifically configured to modify the amplitude of the sampling point with an odd index in the coding suppression signal to an opposite number to obtain a spectrum inversion signal.
  • the processing unit 1104 is specifically used to divide the target audio frame into a low-frequency signal and a high-frequency signal; perform linear prediction analysis on the high-frequency signal to obtain a first linear prediction LPC coefficient; convert the first linear prediction coefficient into a line spectrum pair LSP coefficient; restore the line spectrum pair coefficient to a second linear prediction coefficient; divide the high-frequency signal into a preset number of sub-signals on average; filter each sub-signal based on the second linear prediction coefficient to obtain a residual signal of each sub-signal to obtain the high-frequency residual signal; encode the low-frequency signal to obtain low-frequency coding information and the low-frequency residual signal.
  • the generating unit 1105 is specifically used to encode the low-frequency coding information, the line spectrum pair coefficients and the high-frequency energy gain to generate the coding data of the target audio frame.
  • the audio signal encoding device provided in the embodiment of the present application can execute the audio signal encoding method provided in the above method embodiment, and its implementation principle and technical effect are similar, which will not be repeated here.
  • FIG12 is a structural schematic diagram of the decoding device for an audio signal.
  • the audio signal decoding device 1200 includes:
  • the acquisition unit 1201 is used to parse the coded data of the target audio frame and obtain the low-frequency coding information
  • the decoding unit 1202 is used to decode the low-frequency coding information to obtain a low-frequency signal and a low-frequency residual signal.
  • the suppression unit 1203 is used to suppress the frequency components within the target frequency range in the low-frequency residual signal to obtain a decoded suppression signal.
  • the center frequency of the target frequency range is the fundamental frequency of the low-frequency residual signal.
  • the inversion unit 1204 is used to perform spectrum inversion on the decoded suppression signal to obtain a low-frequency excitation signal.
  • the reconstruction unit 1205 is used to reconstruct the signal according to the low-frequency excitation signal to obtain a high-frequency signal.
  • the generating unit 1206 is configured to generate an audio signal of the target audio frame according to the low-frequency signal and the high-frequency signal.
  • the suppression unit 1203 is specifically used to perform pre-emphasis processing on the low-frequency residual signal based on a high-pass filter to suppress frequency components within a target frequency range in the low-frequency residual signal to obtain a decoded suppression signal.
  • the suppression unit 1203 is specifically used to filter the low-frequency residual signal based on a ramp filter to suppress the frequency components within the target frequency range in the low-frequency residual signal to obtain a decoded suppression signal.
  • the suppression unit 1203 is specifically used to perform notch processing on the frequency components within the target frequency range based on a second-order notch filter to obtain a decoded notch signal, and to perform whitening processing on the decoded notch signal to obtain the decoded suppression signal.
  • the inversion unit 1204 is specifically used to modify the amplitude of the sampling point with an odd index in the decoded suppression signal to an opposite number to obtain a low-frequency excitation signal.
  • the encoded data of the target audio frame also includes: LSP coefficients and high-frequency energy gains, and the reconstruction unit 1205 is specifically used to reconstruct the signal according to the low-frequency excitation signal, the LSP coefficients and the high-frequency energy gain to obtain the high-frequency signal.
  • the reconstruction unit 1205 is specifically used to obtain the energy gain corresponding to each sub-signal in the high-frequency energy gain; obtain the residual signal of each sub-signal according to the low-frequency excitation signal and the energy gain of each sub-signal; restore the LSP coefficient to the LPC coefficient; obtain each predicted sub-signal according to the LPC coefficient; generate each sub-signal according to each predicted sub-signal and the residual signal of each sub-signal; generate the high-frequency signal according to each sub-signal.
  • the audio signal decoding device provided in the embodiment of the present application can execute the audio signal decoding method provided in the above method embodiment, and its implementation principle and technical effect are similar, which will not be repeated here.
  • FIG13 is a schematic diagram of the structure of an electronic device provided by an embodiment of the present application.
  • the electronic device provided by an embodiment of the present application includes: a memory 131 and a processor 132, wherein the memory 131 is used to store a computer program; and the processor 132 is used to execute the audio signal encoding method or the audio signal decoding method provided by the above embodiment when executing the computer program.
  • an embodiment of the present application further provides a computer-readable storage medium, on which a computer program is stored.
  • the computer program is executed by a processor, the computing device implements the audio signal encoding method or audio signal decoding method provided in the above embodiment.
  • the embodiment of the present application further provides a computer program product, which, when executed on a computer, enables the computing device to implement the audio signal encoding method or audio signal decoding method provided in the above embodiment. Law.
  • the embodiments of the present application may be provided as methods, systems, or computer program products. Therefore, the present application may adopt the form of a complete hardware embodiment, a complete software embodiment, or an embodiment combining software and hardware. Moreover, the present application may adopt the form of a computer program product implemented on one or more computer-usable storage media that include computer-usable program code.
  • the processor may be a central processing unit (CPU), other general-purpose processors, digital signal processors (DSP), application-specific integrated circuits (ASIC), field-programmable gate arrays (FPGA) or other programmable logic devices, discrete gate or transistor logic devices, discrete hardware components, etc.
  • CPU central processing unit
  • DSP digital signal processors
  • ASIC application-specific integrated circuits
  • FPGA field-programmable gate arrays
  • a general-purpose processor may be a microprocessor or the processor may be any conventional processor, etc.
  • Memory may include non-permanent storage in a computer-readable medium, random access memory (RAM) and/or non-volatile memory in the form of read-only memory (ROM) or flash memory (flash RAM). Memory is an example of a computer-readable medium.
  • RAM random access memory
  • ROM read-only memory
  • flash RAM flash memory
  • Computer readable media include permanent and non-permanent, removable and non-removable storage media. Storage media can be implemented by any method or technology to store information, which can be computer readable instructions, data structures, program modules or other data. Examples of computer storage media include, but are not limited to, phase change memory (PRAM), static random access memory (SRAM), dynamic random access memory (DRAM), other types of random access memory (RAM), read-only memory (ROM), electrically erasable programmable read-only memory (EEPROM), flash memory or other memory technology, compact disk read-only memory (CD-ROM), digital versatile disk (DVD) or other optical storage, magnetic cassettes, magnetic disk storage or other magnetic storage devices or any other non-transmission media that can be used to store information that can be accessed by a computing device. As defined in this article, computer readable media does not include temporary computer readable media (transitory media), such as modulated data signals and carrier waves.
  • PRAM phase change memory
  • SRAM static random access memory
  • DRAM dynamic random access memory
  • RAM random access memory

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Abstract

一种音频信号的编码、解码方法及装置，该方法包括：获取目标音频帧的高频残差信号和低频残差信号（S101）；对低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号（S102）；目标频率范围的中心频率为低频残差信号的基音频率；对编码抑制信号进行频谱反转，获取频谱反转信号（S103）；根据频谱反转信号和高频残差信号，获取目标音频信号的高频能量增益（S104）；根据高频能量增益生成目标音频帧的编码数据（S105）。

Description

一种音频信号的编码、解码方法及装置
本申请要求于2022年11月01日提交中国专利局、申请号为202211357728.0、发明名称为“一种音频信号的编码、编码方法及装置”的中国专利申请的优先权，其全部内容通过引用结合在本申请中。
技术领域
本申请涉及数据处理技术领域，尤其涉及一种音频信号的编码、解码方法及装置。
背景技术
在音频信号处理时，带宽扩展算法可以在码率的限制下，将大部分码率来编码人耳更加敏感的低频信号，而人耳关注较少的高频信号则使用较少的码率传输，或者仅依靠在解码端根据解码后的低频信号来还原高频信号，以此来在固定码率下提升整体编码语音的质量。
在现有技术中，音频信号处理时高频信号的频谱普遍为对低频信号的频谱进行折叠生成的，因此恢复的音频帧信号会缺少部分谐波分量，且为了抑制折叠到高频的基音分量，在解码端恢复高频信号能量时对高频能量进行衰减，导致恢复后的高频能量偏低，整体音频帧听感不佳。
发明内容
有鉴于此，本申请实施例提供了一种音频信号编码、解码方法及装置，用于提高音频信号处理后的音频质量。
为了实现上述目的，本申请实施例提供技术方案如下：
第一方面，本申请的实施例提供了一种音频信号的编码方法，包括：
获取目标音频帧的高频残差信号和低频残差信号；
对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号；所述目标频率范围的中心频率为所述低频残差信号的基音频率；
对所述编码抑制信号进行频谱反转，获取频谱反转信号；
根据所述频谱反转信号和所述高频残差信号，获取所述目标音频信号的高频能量增益；
根据所述高频能量增益生成所述目标音频帧的编码数据。
作为本申请实施例一种可选的实施方式，所述对所述低频残差信号中目标频率范围内的频率分量进行抑制，以获取编码抑制信号，包括：
基于高通滤波器对所述低频残差信号进行预加重处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号。
作为本申请实施例一种可选的实施方式，所述对所述低频残差信号中目标频率范围内的频率分量进行抑制，以获取编码抑制信号，包括：
基于斜坡式滤波器对所述低频残差信号进行滤波处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号。
作为本申请实施例一种可选的实施方式，所述对所述低频残差信号中目标频率范围内的频率分量进行抑制，以获取编码抑制信号，包括：
基于二阶陷波滤波器对所述目标频率范围内的频率分量进行陷波处理，获取编码陷波信号；
对所述编码陷波信号进行白化处理，以获取所述编码抑制信号。
作为本申请实施例一种可选的实施方式，所述对所述编码抑制信号进行频谱反转，获取频谱反转信号，包括：
将所述编码抑制信号中索引为奇数的采样点的幅值修改为相反 数，以获取频谱反转信号。
作为本申请实施例一种可选的实施方式，所述获取目标音频信号的高频残差信号和低频残差信号，包括：
将所述目标音频帧分频为低频信号和高频信号；
对所述高频信号进行线性预测分析获取第一线性预测LPC系数；
将所述第一线性预测系数转换为线谱对LSP系数；
将所述线谱对系数还原为第二线性预测系数；
将所述高频信号平均分为预设数量个子信号；
基于所述第二线性预测系数分别对各个子信号进行滤波处理，获取各个子信号的残差信号，以获取所述高频残差信号；
对所述低频信号进行编码获取低频编码信息和所述低频残差信号。
作为本申请实施例一种可选的实施方式，所述根据所述高频能量增益生成所述目标音频帧的编码数据，包括：
对所述低频编码信息、所述线谱对系数以及所述高频能量增益进行编码，生成所述目标音频帧的编码数据。
第二方面，本申请的实施例提供了一种音频信号的解码方法，包括：
解析目标音频帧的编码数据，获取低频编码信息；
对所述低频编码信息进行解码，获取低频信号和低频残差信号；
对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号；所述目标频率范围的中心频率为所述低频残差信号的基音频率；
对所述解码抑制信号进行频谱反转，获取低频激励信号；
根据所述低频激励信号进行信号的重建，获取高频信号；
根据所述低频信号和所述高频信号生成所述目标音频帧的音频信号。
作为本申请实施例一种可选的实施方式，所述对所述低频残差 信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号，包括：
基于高通滤波器对所述低频残差信号进行预加重处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号；
作为本申请实施例一种可选的实施方式，所述对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号，包括：
基于斜坡式滤波器对所述低频残差信号进行滤波处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号。
作为本申请实施例一种可选的实施方式，所述对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号，包括：
基于二阶陷波滤波器对所述目标频率范围内的频率分量进行陷波处理，获取解码陷波信号；
对所述解码陷波信号进行白化处理，以获取所述解码抑制信号。
作为本申请实施例一种可选的实施方式，所述对所述解码抑制信号进行频谱反转，获取低频激励信号，包括：
将所述解码抑制信号中索引为奇数的采样点的幅值修改为相反数，以获取频谱反转信号。
作为本申请实施例一种可选的实施方式，所述目标音频帧的编码数据还包括：LSP系数和高频能量增益；
所述根据所述低频激励信号进行信号的重建，以获取高频信号，包括：
根据所述低频激励信号、所述LSP系数以及所述高频能量增益进行信号的重建，以获取所述高频信号。
作为本申请实施例一种可选的实施方式，所述根据所述低频激励信号、所述LSP系数以及所述高频能量增益进行信号的重建，以 获取高频信号，包括：
获取所述高频能量增益中各个子信号对应的能量增益；
根据所述低频激励信号和各个子信号的能量增益，获取各个子信号的残差信号；
将所述LSP系数还原为LPC系数；
根据所述LPC系数获取各个预测子信号；
根据各个预测子信号和各个子信号的残差信号，生成各个子信号；
根据各个子信号生成所述高频信号。
第三方面，本申请实施例提供一种音频信号的编码装置，包括：
获取单元，用于获取目标音频帧的高频残差信号和低频残差信号；
抑制单元，用于对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号；所述目标频率范围的中心频率为所述低频残差信号的基音频率；
反转单元，用于对所述编码抑制信号进行频谱反转，获取频谱反转信号；
处理单元，用于根据所述频谱反转信号和所述高频残差信号，获取所述目标音频信号的高频能量增益；
生成单元，用于根据所述高频能量增益生成所述目标音频帧的编码数据。
作为本申请实施例一种可选的实施方式，所述抑制单元，具体用于基于高通滤波器对所述低频残差信号进行预加重处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号。
作为本申请实施例一种可选的实施方式，所述抑制单元，具体用于基于斜坡式滤波器对所述低频残差信号进行滤波处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号。
作为本申请实施例一种可选的实施方式，所述抑制单元，具体用于基于二阶陷波滤波器对所述目标频率范围内的频率分量进行陷波处理，获取编码陷波信号，以及对所述编码陷波信号进行白化处理，以获取所述编码抑制信号。
作为本申请实施例一种可选的实施方式，所述反转单元，具体用于将所述编码抑制信号中索引为奇数的采样点的幅值修改为相反数，以获取频谱反转信号。
作为本申请实施例一种可选的实施方式，所述获取单元，具体用于：
将所述目标音频帧分频为低频信号和高频信号；
对所述高频信号进行线性预测分析获取第一线性预测LPC系数；
将所述第一线性预测系数转换为线谱对LSP系数；
将所述线谱对系数还原为第二线性预测系数；
将所述高频信号平均分为预设数量个子信号；
基于所述第二线性预测系数分别对各个子信号进行滤波处理，获取各个子信号的残差信号，以获取所述高频残差信号；
对所述低频信号进行编码获取低频编码信息和所述低频残差信号。
作为本申请实施例一种可选的实施方式，所述生成单元，具体用于对所述低频编码信息、所述线谱对系数以及所述高频能量增益进行编码，生成所述目标音频帧的编码数据。
第四方面，本申请实施例提供一种音频信号的解码装置，包括：
获取单元，解析目标音频帧的编码数据，获取低频编码信息；
解码单元，对所述低频编码信息进行解码，获取低频信号和低频残差信号；
抑制单元，对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号；所述目标频率范围的中心频率为所述低频残差信号的基音频率；
反转单元，对所述解码抑制信号进行频谱反转，获取低频激励信号；
重建单元，根据所述低频激励信号进行信号的重建，获取高频信号；
生成单元，根据所述低频信号和所述高频信号生成所述目标音频帧的音频信号。
作为本申请实施例一种可选的实施方式，所述抑制单元，具体用于：基于高通滤波器对所述低频残差信号进行预加重处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号。
作为本申请实施例一种可选的实施方式，所述抑制单元，具体用于基于斜坡式滤波器对所述低频残差信号进行滤波处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号。
作为本申请实施例一种可选的实施方式，所述抑制单元，具体用于基于二阶陷波滤波器对所述目标频率范围内的频率分量进行陷波处理，获取解码陷波信号，以及对所述解码陷波信号进行白化处理，以获取所述解码抑制信号。
作为本申请实施例一种可选的实施方式，所述反转单元，具体用于将所述解码抑制信号中索引为奇数的采样点的幅值修改为相反数，以获取低频激励信号。
作为本申请实施例一种可选的实施方式，所述目标音频帧的编码数据还包括：LSP系数和高频能量增益，所述重建单元，具体用于根据所述低频激励信号、所述LSP系数以及所述高频能量增益进行信号的重建，以获取所述高频信号。
作为本申请实施例一种可选的实施方式，所述重建单元，具体用于获取所述高频能量增益中各个子信号对应的能量增益；根据所述低频激励信号和各个子信号的能量增益，获取各个子信号的残差信号；将所述LSP系数还原为LPC系数；根据所述LPC系数获取各 个预测子信号；根据各个预测子信号和各个子信号的残差信号，生成各个子信号；根据各个子信号生成所述高频信号。
第五方面，本申请实施例提供一种电子设备，包括：存储器和处理器，所述存储器用于存储计算机程序；所述处理器用于在执行计算机程序时，使得所述电子设备实现上述任一实施方式所述的音频信号的编码方法或音频信号的解码方法。
第六方面，本申请实施例提供一种计算机可读存储介质，当所述计算机程序被计算设备执行时，使得所述计算设备实现上述任一实施方式所述的音频信号的编码方法或音频信号的解码方法。
第七方面，本申请实施例提供一种计算机程序产品，当所述计算机程序产品在计算机上运行时，使得所述计算机实现上述任一实施方式所述的音频信号的编码方法或音频信号的解码方法。
本申请实施例所提供的音频信号的编码方法，通过获取目标音频帧的高频残差信号和低频残差信号，再对低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号；对编码抑制信号进行频谱反转，获取频谱反转信号；然后，根据频谱反转信号和所述高频残差信号，获取目标音频信号的高频能量增益；最终根据所述高频能量增益生成目标音频帧的编码数据。本申请实施例通过对获取的低频残差信号的频率分量进行抑制、反转，再结合高频残差信号以及高频能量增益或得目标音频帧的编码数据，从而保证重建后的高频信号不会出现缺少谐波分量以及能量偏低的问题。这样，就可以避免在获取目标音频帧的码流数据时出现音频质量不佳的问题，提高用户体验。因此本申请实施例可以在编码、解码的过程中，提高音频的质量。
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为了更清楚地说明本申请实施例或现有技术中的技术方案，下 面将对实施例或现有技术描述中所需要调用的附图作简单地介绍，显而易见地，对于本领域普通技术人员而言，在不付出创造性劳动性的前提下，还可以根据这些附图获得其他的附图。
图1为本申请实施例提供的音频信号的编码方法的流程图之一；
图2为本申请实施例提供的音频信号的编码方法的流程图之二；
图3为本申请实施例提供的音频信号的编码方法的流程图之三；
图4为本申请实施例提供的音频信号的编码方法的流程图之四；
图5为本申请实施例提供的音频信号的编码设备的硬件框图；
图6为本申请实施例提供的音频信号的解码方法的流程图之一；
图7为本申请实施例提供的音频信号的解码方法的流程图之二；
图8为本申请实施例提供的音频信号的解码方法的流程图之三；
图9为本申请实施例提供的音频信号的解码方法的流程图之四；
图10为本申请实施例提供的音频信号的解码设备的硬件框图；
图11为本申请实施例提供的音频信号的编码装置的结构示意图；
图12为本申请实施例提供的音频信号的解码装置的结构示意图；
图13为本申请实施例提供的电子设备结构示意图。
具体实施方式
为了能够更清楚地理解本申请的上述目的、特征和优点，下面将对本申请的方案进行进一步描述。需要说明的是，在不冲突的情况下，本申请的实施例及实施例中的特征可以相互组合。
在下面的描述中阐述了很多具体细节以便于充分理解本申请，但本申请还可以采用其他不同于在此描述的方式来实施；显然，说明书中的实施例只是本申请的一部分实施例，而不是全部的实施例。
在本申请实施例中，“示例性的”或者“例如”等词用于表示作例子、例证或说明。本申请实施例中被描述为“示例性的”或者“例如”的任何实施例或设计方案不应被解释为比其它实施例或设 计方案更优选或更具优势。确切而言，调用“示例性的”或者“例如”等词旨在以具体方式呈现相关概念。此外，在本申请实施例的描述中，除非另有说明，“多个”的含义是指两个或两个以上。
本申请实施例提供了一种音频信号的编码方法，参照图1所示，该音频信号的编码方法包括以下步骤：
S101：获取目标音频帧的高频残差信号和低频残差信号。
其中，高频残差信号是指音频信号的高频信号的各个样点的值与对应的预测值的差值，各个样点对应的预测值为线性预测系数(Linear Prediction Coefficient，LPC)乘与历史音频信号的低频信号的乘积；低频残差信号是指音频信号的低频信号的各个样点的值与对应的预测值的差值，各个样点对应的预测值为线性预测系数乘与历史音频信号的低频信号的乘积。其中，线性预测系数是指音频信号的样点值可以通过历史音频数据的样点值乘以系数，并对乘积求和的线形组合来逼近，例如：LPC阶数为10，则具有10个系数，使用该10个系数分别乘以历史音频数据的10个样点值，并对乘积求和可以逼近当前样点值，而这些系数即为线性预测系数。
获得高频残差信号和低频残差信号的方式可以与现有技术相同，本申请实施例获取目标音频帧的高频残差信号和低频残差信号的实现方式不做限定，以能够获取目标音频帧的高频残差信号和低频残差信号为主。
S102：对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号。
其中，所述目标频率范围的中心频率为所述低频残差信号的基音频率。
基音频率又叫基频(Baseband)或基带，是指一个复音中基音的频率。在构成一个复音的若干个音中，基音的频率最低，强度最大。基频的高低决定一个音的高低。
示例性的，当目标频率范围的中心频率为20千赫兹(Khz)，则目标频率范围可以为：{10Khz，30Khz}；当目标频率范围的中心 频率为40Khz，则目标频率范围可以为：{20Khz，60Khz}。
S103：对所述编码抑制信号进行频谱反转，获取频谱反转信号。
根据带通采样定理的公式得到的采样基带频谱(中心频率在0Hz附近)，与原始信号正、负频谱的形状刚好相反，因此本申请实施例中对所述编码抑制信号进行频谱反转即为使频谱反转信号与编码抑制信号正、负频谱的形状相反。
S104：根据所述频谱反转信号和所述高频残差信号，获取所述目标音频信号的高频能量增益。
高频能量增益是指高频残差信号的能量增益，具体为高频残差信号和低频残差信号的能量之比增益值指高频信号与低频信号之间的能量抵值。
在一些实施例中，根据所述频谱反转信号和所述高频残差信号，获取所述目标音频信号的高频能量增益的实现方式可以包括：
获取所述频谱反转信号的能量值和所述高频残差信号的能量值，以及计算所述频谱反转信号的能量值和所述高频残差信号的能量值的比值，以获取所述目标音频信号的高频能量增益。
S105：根据所述高频能量增益生成所述目标音频帧的编码数据。
本申请实施例所提供的音频信号的编码方法，通过获取目标音频帧的高频残差信号和低频残差信号，再对低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号；对编码抑制信号中符合预设条件的频率分量进行反转，获取频谱反转信号；然后，根据频谱反转信号和所述高频残差信号，获取目标音频信号的高频能量增益；最终根据所述高频能量增益生成目标音频帧的编码数据。本申请实施例通过对获取的低频残差信号的频率分量进行抑制、反转，从而保证重建后的高频信号不会出现缺少谐波分量以及能量偏低的问题。这样，就可以避免在获取目标音频帧的码流数据时出现音频质量不佳的问题，提高用户体验。因此本申请实施例可以在编码过程中，提高音频质量。
作为对上实施例的扩展和细化，本申请实施例提供了另一种音 频信号的编码方法，参照图2所示，该音频信号的编码方法包括以下步骤：
S201、将所述目标音频帧分频为低频信号和高频信号。
在一些实施例中，可以通过正交镜像滤波器(Quadrature Mirror Filter,QMF)将目标音频帧分频为低频信号和高频信号。其中，低频信号的频率范围可以为[0kHz-4kHz]，高频信号的频率范围可以为[4kHz-8kHz]。
S202、对所述高频信号进行线性预测分析获取第一线性预测系数。
在一些实施例中，可以通过伯格(burg algorithm)算法对所述高频信号进行线性预测分析,以获取第一线性预测系数。其中，伯格算法是一种直接由已知的时间信号序列计算功率谱估计值的递推算法。
S203、将所述第一线性预测系数转换为线谱对(Line Spectrum Pair，LSP)系数。
将第一线性预测系数转换为线谱对系数的实现方式可以现有技术中将LPC系数转换为LSP系数的实现方式相同，本申请实施例对此不做限定。
S204、将所述线谱对系数还原为第二线性预测系数。
同样，将所述线谱对系数还原为第二线性预测系数的实现方式可以现有技术中将LSP系数还原为LPC系数的实现方式相同，本申请实施例对此不做限定。
S205、将所述高频信号平均分为预设数量个子信号。
本申请实施例中不对预设数量进行限制，实际编码过程中可以根据需求将高频信号平均分为任意数量个子信号。例如：可以将所述高频信号分为4个等长度的子信号，再例如：将所述高频信号分为8个等长度的子信号。
S206、基于所述第二线性预测系数分别对各个子信号进行滤波处理，获取各个子信号的残差信号，以获取所述高频残差信号。
具体的，基于所述第二线性预测系数分别对各个子信号进行滤波处理的线性预测滤波器的传递函数可以为：
通过该传递函数获得的子信号的残差信号为：
其中，i为子信号的索引，xhb表示原始的子信号，ai为索引为i的子信号的线性预测系数，reshb为索引为i的子信号的残差信号。
S207、对所述低频信号进行编码获取低频编码信息和所述低频残差信号。
在一些实施例中，可以通过编码器SILK编码器对所述低频信号进行编码，以获取所述低频编码信息和所述低频残差信号。
S208、基于高通滤波器对所述低频残差信号进行预加重处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号。
具体的，由于预加重处理也是一种滤波处理，所以基于所述第二线性预测系数分别对各个子信号进行预加重处理会采用高通滤波器进行预加重处理高通滤波器用来抑制基音频率附近突出的频率分量，则高通滤波器的传递函数为：H(z)＝1-μz-1，μ为预设滤波系数。
使用差分方程表示为：
其中表示经过处理后的低频残差信号，μ为预滤波系数，其中，μ决定了对低频残差信号中频率较低的频率分量的抑制以及对频率较高的频率分量的加重程度，μ值越大对频率较低的频率分量的抑制程度越高，对频率较高的频率分量的加重程度越高。
S209、将所述编码抑制信号中索引为奇数的采样点的幅值修改为相反数，以获取频谱反转信号。
由于目标音频帧的音频信号经过正交镜像滤波器分频后，得到的高频信号的频谱发生了反转，因此为了保证频谱反转信号和原始高频信号的频谱对应，需要对编码抑制信号进行频谱反转。
对所述编码抑制信号中索引为奇数的采样点进行取相反数，以获取频谱反转信号，具体可以为通过如下公式和所述编码抑制信号，获取频谱反转信号：
reslb(i)＝reslb(i)*(-1)i
其中，i为编码抑制信号中的采样点的索引。例如，当编码抑制信号中的采样点的索引为1时，即i＝1时，该公式记为：reslb(1)＝reslb(1)*(-1)1，得到reslb(1)＝-reslb(1)表示当i＝1时，得到的频谱反转信号对应的索引为1采样点为编码抑制信号索引为1采样点的相反数；当编码抑制信号中的采样点的索引为2时，即i＝2时，该公式记为：reslb(2)＝reslb(2)*(-1)2，得到reslb(2)＝reslb(2)，表示当i＝2时，得到的频谱反转信号对应的索引为2采样点等于为编码抑制信号索引为2采样点。
示例性的，若编码抑制信号为：{a1、a2、a3、……a64}，则经过上述公式计算后得到的频谱反转信号为：{-a1、a2、-a3、……a64}。
S210、根据所述频谱反转信号和所述高频残差信号，获取所述目标音频信号的高频能量增益。
其中，所述高频能量增益包括各个子信号的能量增益。
在一些实施例中，索引i的子信号的能量增益值为:
其中，N为子信号的长度，gaini为索引为i的子信号的能量增益值，为频谱反转信号的能量，为索引为i的子信号的能量。
S211、对所述低频编码信息、所述LSP系数以及所述高频能量增益进行编码，生成所述目标音频帧的编码数据。
即，对所述低频编码信息、所述LSP系数以及所述高频能量增益进行音频信号包的封装，以获取所述目标音频帧的编码数据。
在一些实施例中，在根据所述低频编码信息、所述LSP系数以及所述高频能量增益生成所述目标音频帧的编码数据之前，本发明实施例提供的音频信号的编码方法还包括：对所述LSP系数进行双码本量化。
例如：先对LSP系数进行双码本量化，然后再使用12bit将对应码本下标编码到主码流。
其中，双码本量化是将得到的LSP系数通过两个不同码本进行系数检索得到LSP系数与对应码本的下标编码，经过两个码本检索合成新的LSP系数下标编码。
双码本编码下标与LSP系数对应关系可以如下表一所示：
表一
当双码本编码下标与LSP系数对应关系如上表一所示，则当LSP系数为{1111，1112，1113，1115，1117，1118，1119}时，经过双码本量化，得到对应的码本下标编码{C1、C2、C3、C4、C5、C6、C7、C8}。
通过对LSP系数进行双码本量化可减少LSP系数的数据量，进而提升音频信号传输的效率。
在一些实施例中，在根据所述低频编码信息、所述LSP系数以 及所述高频能量增益生成所述目标音频帧的编码数据之前，本发明实施例提供的音频信号的编码方法还包括：对所述高频能量增益进行码本量化。
例如：可以将高频能量增益量化后将对应下标使用5bit编码到主码流，当包括4个子信号时，高频能量增益的编码数据共计消耗20bit。
编码下标与高频能量增益对应关系可以如下表二所示：
表二
通过对高频能量增益进行码本量化可减少高频能量增益的数据量，进而提升音频信号传输的效率。
本申请实施例提供了另一种音频信号的编码方法，参照图3所示，该音频信号的编码方法包括以下步骤：
S301、将所述目标音频帧分频为低频信号和高频信号。
S302、对所述高频信号进行线性预测分析获取第一线性预测系数。
S303、将所述第一线性预测系数转换为线谱对系数。
S304、将所述线谱对系数还原为第二线性预测系数。
S305、将所述高频信号平均分为预设数量个子信号。
S306、基于所述第二线性预测系数分别对各个子信号进行滤波处理，获取各个子信号的残差信号，以获取所述高频残差信号。
S307、对所述低频信号进行编码获取低频编码信息和所述低频 残差信号。
上述步骤S301至S307的实现方式可以与图2所示实施例中的步骤S201至S207的实现方式相同，为避免赘述，此处不再详细说明。
S308、基于斜坡式滤波器对所述低频残差信号进行滤波处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号。
即，通过一个斜坡式滤波器对低频残差信号进行滤波处理，从而对中心频率为所述低频残差信号的基音频率的目标频率范围内的频率分量进行抑制。
在一些实施例中，斜坡滤波器传递函数可以如下：
使用差分方程表示为：




其中，fc表示需要调整的频率，G＝1+B0表示相应频率下的增益值,可以根据谱倾斜的程度，指定滤波器抑制频率范围的和抑制程度，降低低频残差的频谱倾斜程度。
S309、对所述编码抑制信号中符合预设条件的频率分量进行反转，获取频谱反转信号。
同样，音频信号经过正交镜像滤波器分频后，得到的高频信号的频谱发生了反转，为了保证频谱反转信号和原始高频频谱对应，需要对低频残差信号进行频谱反转。对所述编码抑制信号进行频谱反转的实现方式与上述步骤S209所述相同，此处不再赘述。
S310、根据所述频谱反转信号和所述高频残差信号，获取所述目标音频信号的高频能量增益。
上述步骤S310的实现方式可以与图2所示实施例中的步骤S210的实现方式相同，为避免赘述，此处不再详细说明。
S311、对所述低频编码信息、所述LSP系数以及所述高频能量增益进行编码，生成所述目标音频帧的编码数据。
本申请实施例提供了另一种音频信号的编码方法，参照图4所示，所该音频信号的编码方法包括以下步骤：
S401、将所述目标音频帧分频为低频信号和高频信号。
S402、对所述高频信号进行线性预测分析获取第一线性预测系数。
S403、将所述第一线性预测系数转换为线谱对系数。
S404、将所述线谱对系数还原为第二线性预测系数。
S405、将所述高频信号平均分为预设数量个子信号。
S406、基于所述第二线性预测系数分别对各个子信号进行滤波处理，获取各个子信号的残差信号，以获取所述高频残差信号。
S407、对所述低频信号进行编码获取低频编码信息和所述低频残差信号。
上述步骤S401至S407的实现方式可以与图2所示实施例中的步骤S201至S207的实现方式相同，为避免赘述，此处不再详细说明。
S408：基于二阶陷波滤波器对所述目标频率范围内的频率分量进行陷波处理，获取编码陷波信号。
即，先获取所述低频残差信号的基因频率，然后根据所述低频残差信号的基因频率确定所述目标频率范围，以及使用二阶陷波滤 波器对所述目标频率范围内的频率分量进行陷波处理，以获取编码陷波信号。
由于频谱反转信号主要在基音频率附近(目标频率范围内)有较高的频率分量，因此将频谱反转信号通入二阶陷波器，对目标频率范围内的频率分量做陷波处理。
在一些实施例中，二阶陷波器的传递函数如下：
使用差分方程表示为：
其中，




γ＝G*tanf(bw/2)
其中，表示经过二阶陷波处理后的低频残差信号，bw表示滤波器的陷波带宽，Ω0表示陷波器的中心频率点，G表示在指定频率下的陷波增益值。
S409、对所述编码陷波信号进行白化处理，以获取所述编码抑制信号。
即，将低频残差信号做完陷波处理后，对处理结果进一步进行白化处理。
在一些实施例中，对所述编码陷波信号进行白化处理的实现方式包括：
首先，通过伯格算法得到低频残差信号的LPC系数。
其次，使用该LPC系数对上述步骤中处理得到的编码陷波信号做高阶的LPC滤波得到编码抑制信号。
例如，高阶的LPC滤波为8阶，则可以通过如下公式进行计算：
S410、将所述编码抑制信号中索引为奇数的采样点的幅值修改为相反数，以获取频谱反转信号。
S411、根据所述频谱反转信号和所述高频残差信号，获取所述目标音频信号的高频能量增益。
S412、对所述低频编码信息、所述线谱对系数以及所述高频能量增益进行编码，生成所述目标音频帧的编码数据。
参照图5所示，图5为本申请实施例提供的音频信号的编码设备的硬件框图，音频信号的编码设备包括：正交镜像滤波器501、编码器502、抑制模块503、反转模块504、拆分模块505、线性预测分析器506、参数量化器507、还原模块508、高频残差生成器509、增益计算器510以及封装器511。
其中，正交镜像滤波器501用于将单一帧音频信号分频成低频(Low Band，LB)信号和高频(High Band，HB)信号。
编码器502用于对低频信号进行编码，以生成低频编码信息和低频残差信号。
抑制模块503用于对中心频率为所述低频残差信号的基音频率的目标频率范围内的频率分量进行抑制，以获取编码抑制信号。
反转模块504用于对编码抑制信号进行频谱反转，以获取频谱反转信号。
拆分模块505用于将单帧的高频信号平均分为预设数量个子信号。
线性预测分析器506用于对高频信号进行线性预测分析，以获取高频信号的第一LPC系数。
参数量化器507用于将第一线性预测系数转换为LSP系数。
还原模块508用于将LSP系数还原为第二线性预测系数。
高频残差生成器509根据第二线性预测系数和各个子信号生成各个子信号的残差信号，以获取高频残差信号。
增益计算器510根据频谱反转信号和高频残差信号计算高能量增益值。
封装器511用于封装所述低频编码信息、所述LSP系数以及所述高频能量增益生成音频信号的编码数据。
本申请另一实施例提供了一种音频信号的解码方法，参照图6所示，该音频信号的解码方法，包括如下步骤：
S601、解析目标音频帧的编码数据，获取低频编码信息。
即，对接收到的音频帧的编码数据进行解封装，以获取编码数据中携带的低频编码信息。
S602、对所述低频编码信息进行解码，获取低频信号和低频残差信号。
在一些实施例中，可以用解码器对所述低频编码信息进行解码，以获取所述低频信号和所述低频残差信号。
S603、对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号。
其中，所述目标频率范围的中心频率为所述低频残差信号的基音频率。
S604、对所述解码抑制信号进行频谱反转，获取低频激励信号。
S605、根据所述低频激励信号进行信号的重建，获取高频信号。
S606、根据所述低频信号和所述高频信号生成所述目标音频帧的音频信号。
本申请实施例所提供的音频信号的解码方法，通过解析目标音频帧的编码数据获取低频编码信息，然后对低频编码信息进行解码获取低频信号和低频残差信号，再对所述低频残差信号中目标频率范围内的频率分量进行抑制，然后对获取到的解码抑制信号进行频谱反转获取低频激励信号，以及根据低频激励信号进行信号的重建，获取高频信号。最终根据低频信号和高频信号生成目标音频帧的音频信号。由于本申请实施例通过在对低频激励信号的频谱进行抑制，而无需对高频信号进行衰减，因此避免了高频信号能量偏低的问题，又因为本申请实施例还会在重建高频信号时对符合预设条件的采样点的频谱值进行反转，因此避免了高频信号缺少谐波分量的问题。综上，本申请实施例可以在解码端重建高频信号时避免高频能量偏低以及高频谐波缺失，因此，本申请实施例可以提高音频质量。
本申请实施例提供了另一种音频信号的解码方法，参照图7所示，该音频信号的解码方法包括以下步骤：
S701、解析目标音频帧的编码数据，获取低频编码信息、LSP系数以及高频能量增益。
S702、对所述低频编码信息进行解码，获取低频信号和低频残差信号。
S703、基于高通滤波器对所述低频残差信号进行预加重处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号。
S704、将所述解码抑制信号中索引为奇数的采样点的幅值修改为相反数，以获取低频激励信号。
S705、根据所述低频激励信号、所述LSP系数以及所述高频能量增益进行信号的重建，以获取所述高频信号。
其中，所述根据所述低频激励信号、所述LSP系数以及所述高频能量增益进行信号的重建，以获取高频信号，包括以下步骤1至步骤6：
步骤1、获取所述高频能量增益中各个子信号对应的能量增益。
步骤2、根据所述低频激励信号和各个子信号的能量增益，获取各个子信号的残差信号。
步骤3、将所述LSP系数还原为LPC系数。
步骤4、根据所述LPC系数获取各个预测子信号。
步骤5、根据各个预测子信号和各个子信号的残差信号，生成各个子信号。
步骤6、根据各个子信号生成所述高频信号。
S706、根据所述低频信号和所述高频信号生成所述目标音频帧的音频信号。
在一些实施例中，可以通过正交镜像滤波器对所述低频信号和所述高频信号进行合成，以生成所述目标音频帧的音频信号。
本申请实施例提供了另一种音频信号的解码方法，参照图8所示，所述对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号，该音频信号的解码方法包括以下步骤：
S801、解析目标音频帧的编码数据，获取低频编码信息、LSP系数以及高频能量增益。
S802、对所述低频编码信息进行解码，获取低频信号和低频残差信号。
S803、基于斜坡式滤波器对所述低频残差信号进行滤波处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号。
S804、将所述解码抑制信号中索引为奇数的采样点的幅值修改为相反数，以获取低频激励信号。
S805、根据所述低频激励信号、所述LSP系数以及所述高频能量增益进行信号的重建，以获取所述高频信号。
S806、根据所述低频信号和所述高频信号生成所述目标音频帧的音频信号。
本申请实施例提供了另一种音频信号的解码方法，参照图9所示，所述对所述低频残差信号中目标频率范围内的频率分量进行抑 制，获取解码抑制信号，该音频信号的解码方法包括以下步骤：
S901、解析目标音频帧的编码数据，获取低频编码信息、LSP系数以及高频能量增益。
S902、对所述低频编码信息进行解码，获取低频信号和低频残差信号。
S903、基于二阶陷波滤波器对所述目标频率范围内的频率分量进行陷波处理，获取解码陷波信号。
S904、对所述解码陷波信号进行白化处理，以获取所述解码抑制信号。
S905、将所述解码抑制信号中索引为奇数的采样点的幅值修改为相反数，以获取低频激励信号。
S906、根据所述低频激励信号、所述LSP系数以及所述高频能量增益进行信号的重建，以获取所述高频信号。
S907、根据所述低频信号和所述高频信号生成所述目标音频帧的音频信号。
结合上述实施例，参照图10所示，图10为本申请实施例提供的音频信号的解码设备的硬件框图，解码设备包括：解封器101、解码器102、抑制模块103、反转模块104、残差生成器105、还原模块106、预测模块107、重建模块108、拼接模块109以及正交镜像滤波器1010。
其中，解封器101用于解析获取低频编码信息、LSP系数以及高频能量增益。
解码器102用于对所述低频编码信息进行解码获取低频信号和低频残差信号。
抑制模块103用于对中心频率为所述低频残差信号的基音频率的目标频率范围内的频率分量进行抑制，以获取解码抑制信号。
反转模块104用于对解码抑制信号进行频谱反转，以获取低频激励信号。
残差生成器105用于根据所述低频激励信号和所述高频能量增 益中各个子信号对应的能量增益，获取各个子信号的残差信号。
还原模块106用于将LSP系数还原为LPC系数。
预测模块107用于根据LPC系数获取各个高频子信号。
重建模块108用于根据各个预测子信号和各个子信号的残差信号，生成各个子信号。
拼接模块109用于将各个子信号拼接为高频信号。
正交镜像滤波器1010用于将高频信号和低频信号合成为音频信号。
基于同一发明构思，作为对上述方法的实现，本申请实施例还提供了一种音频信号的编码装置和一种音频信号的解码装置，该实施例与前述方法实施例对应，为便于阅读，本申请实施例不再对前述方法实施例中的细节内容进行逐一赘述，但应当明确，本申请实施例中的音频信号的解码装置和音频信号的解码装置能够对应实现前述方法实施例中的全部内容。
基于同一构思，本申请实施例提供了一种音频信号的编码装置，图11该音频信号的编码装置的结构示意图，参照图11所示，该音频信号的编码装置1100包括：
获取单元1101用于获取目标音频帧的高频残差信号和低频残差信号。
抑制单元1102用于对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号；所述目标频率范围的中心频率为所述低频残差信号的基音频率。
反转单元1103用于对所述编码抑制信号进行频谱反转，获取频谱反转信号。
处理单元1104用于根据所述频谱反转信号和所述高频残差信号，获取所述目标音频信号的高频能量增益。
生成单元1105用于根据所述高频能量增益生成所述目标音频帧的编码数据。
作为本申请实施例一种可选的实施方式，所述抑制单元1102， 具体用于：
基于高通滤波器对所述低频残差信号进行预加重处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号。
作为本申请实施例一种可选的实施方式，所述抑制单元1102，具体用于基于斜坡式滤波器对所述低频残差信号进行滤波处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号。
作为本申请实施例一种可选的实施方式，所述抑制单元1102，具体用于基于二阶陷波滤波器对所述目标频率范围内的频率分量进行陷波处理，获取编码陷波信号，以及对所述编码陷波信号进行白化处理，以获取所述编码抑制信号。
作为本申请实施例一种可选的实施方式，所述反转单元1103，具体用于将所述编码抑制信号中索引为奇数的采样点的幅值修改为相反数，以获取频谱反转信号。
作为本申请实施例一种可选的实施方式，所述处理单元1104，具体用于将所述目标音频帧分频为低频信号和高频信号；对所述高频信号进行线性预测分析获取第一线性预测LPC系数；将所述第一线性预测系数转换为线谱对LSP系数；将所述线谱对系数还原为第二线性预测系数；将所述高频信号平均分为预设数量个子信号；基于所述第二线性预测系数分别对各个子信号进行滤波处理，获取各个子信号的残差信号，以获取所述高频残差信号；对所述低频信号进行编码获取低频编码信息和所述低频残差信号。
作为本申请实施例一种可选的实施方式，所述生成单元1105，具体用于对所述低频编码信息、所述线谱对系数以及所述高频能量增益进行编码，生成所述目标音频帧的编码数据。
本申请实施例提供的音频信号的编码装置可以执行上述方法实施例提供的音频信号的编码方法，其实现原理与技术效果类似，此处不再赘述。
基于同一构思，本申请实施例提供了一种音频信号的解码装置，图12该音频信号的解码装置的结构示意图，参照图12所示，该音频信号解码设备1200包括：
获取单元1201用于解析目标音频帧的编码数据，获取低频编码信息；
解码单元1202用于对所述低频编码信息进行解码，获取低频信号和低频残差信号。
抑制单元1203用于对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号。
所述目标频率范围的中心频率为所述低频残差信号的基音频率。
反转单元1204用于对所述解码抑制信号进行频谱反转，获取低频激励信号。
重建单元1205用于根据所述低频激励信号进行信号的重建，获取高频信号。
生成单元1206用于根据所述低频信号和所述高频信号生成所述目标音频帧的音频信号。
作为本申请实施例一种可选的实施方式，所述抑制单元1203，具体用于基于高通滤波器对所述低频残差信号进行预加重处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号。
作为本申请实施例一种可选的实施方式，所述抑制单元1203，具体用于基于斜坡式滤波器对所述低频残差信号进行滤波处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号。
作为本申请实施例一种可选的实施方式，所述抑制单元1203，具体用于基于二阶陷波滤波器对所述目标频率范围内的频率分量进行陷波处理，获取解码陷波信号，以及对所述解码陷波信号进行白化处理，以获取所述解码抑制信号。
作为本申请实施例一种可选的实施方式，所述反转单元1204，具体用于将所述解码抑制信号中索引为奇数的采样点的幅值修改为相反数，以获取低频激励信号。
作为本申请实施例一种可选的实施方式，所述目标音频帧的编码数据还包括：LSP系数和高频能量增益，所述重建单元1205，具体用于根据所述低频激励信号，所述LSP系数以及所述高频能量增益进行信号的重建，以获取所述高频信号。
作为本申请实施例一种可选的实施方式，所述重建单元1205，具体用于获取所述高频能量增益中各个子信号对应的能量增益；根据所述低频激励信号和各个子信号的能量增益，获取各个子信号的残差信号；将所述LSP系数还原为LPC系数；根据所述LPC系数获取各个预测子信号；根据各个预测子信号和各个子信号的残差信号，生成各个子信号；根据各个子信号生成所述高频信号。
本申请实施例提供的音频信号的解码装置可以执行上述方法实施例提供的音频信号的解码方法，其实现原理与技术效果类似，此处不再赘述。
基于同一发明构思，本申请实施例还提供了一种电子设备。图13为本申请实施例提供的电子设备的结构示意图，参照图13所示，本申请实施例提供的电子设备包括：存储器131和处理器132，所述存储器131用于存储计算机程序；所述处理器132用于在执行计算机程序时执行上述实施例提供的音频信号的编码方法或音频信号的解码方法。
基于同一发明构思，本申请实施例还提供了一种计算机可读存储介质，该计算机可读存储介质上存储有计算机程序，当计算机程序被处理器执行时，使得所述计算设备实现上述实施例提供的音频信号的编码方法或音频信号的解码方法。
基于同一发明构思，本申请实施例还提供了一种计算机程序产品，当所述计算机程序产品在计算机上运行时，使得所述计算设备实现上述实施例提供的音频信号的编码方法或音频信号的解码方 法。
本领域技术人员应明白，本申请的实施例可提供为方法，***，或计算机程序产品。因此，本申请可采用完全硬件实施例，完全软件实施例，或结合软件和硬件方面的实施例的形式。而且，本申请可采用在一个或多个其中包含有计算机可用程序代码的计算机可用存储介质上实施的计算机程序产品的形式。
处理器可以是中央渲染单元(Central Processing Unit，CPU)，还可以是其他通用处理器，数字信号处理器(Digital Signal Processor，DSP)，专用集成电路(Application Specific Integrated Circuit，ASIC)，现成可编程门阵列(Field-Programmable Gate Array，FPGA)或者其他可编程逻辑器件，分立门或者晶体管逻辑器件，分立硬件组件等。通用处理器可以是微处理器或者该处理器也可以是任何常规的处理器等。
存储器可能包括计算机可读介质中的非永久性存储器，随机存取存储器(RAM)和/或非易失性内存等形式，如只读存储器(ROM)或闪存(flash RAM)。存储器是计算机可读介质的示例。
计算机可读介质包括永久性和非永久性，可移动和非可移动存储介质。存储介质可以由任何方法或技术来实现信息存储，信息可以是计算机可读指令，数据结构，程序的模块或其他数据。计算机的存储介质的例子包括，但不限于相变内存(PRAM)，静态随机存取存储器(SRAM)，动态随机存取存储器(DRAM)，其他类型的随机存取存储器(RAM)，只读存储器(ROM)，电可擦除可编程只读存储器(EEPROM)，快闪记忆体或其他内存技术，只读光盘只读存储器(CD-ROM)，数字多功能光盘(DVD)或其他光学存储，磁盒式磁带，磁盘存储或其他磁性存储设备或任何其他非传输介质，可用于存储可以被计算设备访问的信息。根据本文中的界定，计算机可读介质不包括暂存电脑可读媒体(transitory media)，如调制的数据信号和载波。
最后应说明的是：以上各实施例仅用以说明本申请的技术方案， 而非对其限制；尽管参照前述各实施例对本申请进行了详细的说明，本领域的普通技术人员应当理解：其依然可以对前述各实施例所记载的技术方案进行修改，或者对其中部分或者全部技术特征进行等同替换；而这些修改或者替换，并不使相应技术方案的本质脱离本申请各实施例技术方案的范围。

Claims (19)

  1. 一种音频信号的编码方法，其特征在于，包括：
    获取目标音频帧的高频残差信号和低频残差信号；
    对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号；所述目标频率范围的中心频率为所述低频残差信号的基音频率；
    对所述编码抑制信号进行频谱反转，获取频谱反转信号；
    根据所述频谱反转信号和所述高频残差信号，获取所述目标音频信号的高频能量增益；
    根据所述高频能量增益生成所述目标音频帧的编码数据。
  2. 根据权利要求1所述的方法，其特征在于，所述对所述低频残差信号中目标频率范围内的频率分量进行抑制，以获取编码抑制信号，包括：
    基于高通滤波器对所述低频残差信号进行预加重处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号。
  3. 根据权利要求1所述的方法，其特征在于，所述对所述低频残差信号中目标频率范围内的频率分量进行抑制，以获取编码抑制信号，包括：
    基于斜坡式滤波器对所述低频残差信号进行滤波处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号。
  4. 根据权利要求1所述的方法，其特征在于，所述对所述低频残差信号中目标频率范围内的频率分量进行抑制，以获取编码抑制信号，包括：
    基于二阶陷波滤波器对所述目标频率范围内的频率分量进行陷波处理，获取编码陷波信号；
    对所述编码陷波信号进行白化处理，以获取所述编码抑制信号。
  5. 根据权利要求1所述的方法，其特征在于，所述对所述编码抑制信号进行频谱反转，获取频谱反转信号，包括：
    将所述编码抑制信号中索引为奇数的采样点的幅值修改为相反数，以获取频谱反转信号。
  6. 根据权利要求1-5任一项所述的方法，其特征在于，所述获取目标音频信号的高频残差信号和低频残差信号，包括：
    将所述目标音频帧分频为低频信号和高频信号；
    对所述高频信号进行线性预测分析获取第一线性预测LPC系数；
    将所述第一线性预测系数转换为线谱对LSP系数；
    将所述线谱对系数还原为第二线性预测系数；
    将所述高频信号平均分为预设数量个子信号；
    基于所述第二线性预测系数分别对各个子信号进行滤波处理，获取各个子信号的残差信号，以获取所述高频残差信号；
    对所述低频信号进行编码获取低频编码信息和所述低频残差信号。
  7. 根据权利要求6所述的方法，其特征在于，所述根据所述高频能量增益生成所述目标音频帧的编码数据，包括：
    对所述低频编码信息、所述线谱对系数以及所述高频能量增益进行编码，生成所述目标音频帧的编码数据。
  8. 一种音频信号的解码方法，其特征在于，包括：
    解析目标音频帧的编码数据，获取低频编码信息；
    对所述低频编码信息进行解码，获取低频信号和低频残差信号；
    对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号；所述目标频率范围的中心频率为所述低频残差信号的基音频率；
    对所述解码抑制信号进行频谱反转，获取低频激励信号；
    根据所述低频激励信号进行信号的重建，获取高频信号；
    根据所述低频信号和所述高频信号生成所述目标音频帧的音频 信号。
  9. 根据权利要求8所述的方法，其特征在于，所述对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号，包括：
    基于高通滤波器对所述低频残差信号进行预加重处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号。
  10. 根据权利要求8所述的方法，其特征在于，所述对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号，包括：
    基于斜坡式滤波器对所述低频残差信号进行滤波处理，以对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号。
  11. 根据权利要求8所述的方法，其特征在于，所述对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号，包括：
    基于二阶陷波滤波器对所述目标频率范围内的频率分量进行陷波处理，获取解码陷波信号；
    对所述解码陷波信号进行白化处理，以获取所述解码抑制信号。
  12. 根据权利要求8所述的方法，其特征在于，所述对所述解码抑制信号中符合预设条件的采样点的频谱值进行反转，获取频谱反转信号，包括：
    将所述解码抑制信号中索引为奇数的采样点的幅值修改为相反数，以获取低频激励信号。
  13. 根据权利要求8-12任一项所述的方法，其特征在于，所述目标音频帧的编码数据还包括：LSP系数和高频能量增益；
    所述根据所述低频激励信号进行信号的重建，以获取高频信号，包括：
    根据所述低频激励信号、所述LSP系数以及所述高频能量增益 进行信号的重建，以获取所述高频信号。
  14. 根据权利要求13所述的方法，其特征在于，所述根据所述低频激励信号、所述LSP系数以及所述高频能量增益进行信号的重建，以获取高频信号，包括：
    获取所述高频能量增益中各个子信号对应的能量增益；
    根据所述低频激励信号和各个子信号的能量增益，获取各个子信号的残差信号；
    将所述LSP系数还原为LPC系数；
    根据所述LPC系数获取各个预测子信号；
    根据各个预测子信号和各个子信号的残差信号，生成各个子信号；
    根据各个子信号生成所述高频信号。
  15. 一种音频信号的编码装置，其特征在于，包括：
    获取单元，用于获取目标音频帧的高频残差信号和低频残差信号；
    抑制单元，用于对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取编码抑制信号；所述目标频率范围的中心频率为所述低频残差信号的基音频率；
    反转单元，用于对所述编码抑制信号进行频谱反转，获取频谱反转信号；
    处理单元，用于根据所述频谱反转信号和所述高频残差信号，获取所述目标音频信号的高频能量增益；
    生成单元，用于根据所述高频能量增益生成所述目标音频帧的编码数据。
  16. 一种音频信号的解码装置，其特征在于，包括：
    获取单元，用于解析目标音频帧的编码数据，获取低频编码信息；
    解码单元，用于对所述低频编码信息进行解码，获取低频信号和低频残差信号；
    抑制单元，用于对所述低频残差信号中目标频率范围内的频率分量进行抑制，获取解码抑制信号；所述目标频率范围的中心频率为所述低频残差信号的基音频率；
    反转单元，对所述解码抑制信号进行频谱反转，获取低频激励信号；
    重建单元，用于根据所述低频激励信号进行信号的重建，获取高频信号；
    生成单元，用于根据所述低频信号和所述高频信号生成所述目标音频帧的音频信号。
  17. 一种电子设备，其特征在于，包括：存储器和处理器，所述存储器用于存储计算机程序；所述处理器用于在执行计算机程序时，使得所述电子设备实现权利要求1-7任一项所述的音频信号的编码方法或权利要求8-14任一项所述的音频信号的解码方法。
  18. 一种计算机可读存储介质，其特征在于，所述计算机可读存储介质上存储有计算机程序，当所述计算机程序被计算设备执行时，使得所述计算设备实现权利要求1-7任一项所述的音频信号的编码方法或权利要求8-14任一项所述的音频信号的解码方法。
  19. 一种计算机程序产品，其特征在于，当所述计算机程序产品在计算机上运行时，使得所述计算机实现如权利要求1-7任一项所述的音频信号的编码方法或权利要求8-14任一项所述的音频信号的解码方法。
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CN101436407A (zh) * 2008-12-22 2009-05-20 西安电子科技大学 音频编解码方法
CN101521014A (zh) * 2009-04-08 2009-09-02 武汉大学 音频带宽扩展编解码装置
CN105765655A (zh) * 2013-11-22 2016-07-13 高通股份有限公司 高频带译码中的选择性相位补偿
CN107925388A (zh) * 2016-02-17 2018-04-17 弗劳恩霍夫应用研究促进协会 用于增强瞬时处理的后置处理器、预处理器、音频编码器、音频解码器及相关方法
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CN1992533A (zh) * 2005-12-26 2007-07-04 索尼株式会社 信号编码设备和方法、信号译码设备和方法、程序及介质
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