WO2021244409A1 - Sound wave signal decoding method and device - Google Patents

Sound wave signal decoding method and device Download PDF

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Publication number
WO2021244409A1
WO2021244409A1 PCT/CN2021/096619 CN2021096619W WO2021244409A1 WO 2021244409 A1 WO2021244409 A1 WO 2021244409A1 CN 2021096619 W CN2021096619 W CN 2021096619W WO 2021244409 A1 WO2021244409 A1 WO 2021244409A1
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sub
band
audio
data
sound wave
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PCT/CN2021/096619
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French (fr)
Chinese (zh)
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唐鸿
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北京声连网信息科技有限公司
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; 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 OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; 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
    • G10L19/032—Quantisation or dequantisation of spectral components

Definitions

  • the present invention relates to the technical field of communication coding, in particular to a method and device for decoding acoustic wave signals.
  • Audio quantization compression is an audio compression technology that uses audio quantization processing. Quantization refers to the process of approximating the continuous value of the signal (or a large number of possible discrete values) to a finite number of (or fewer) discrete values, that is, converting the sampled analog signal into a digital signal by rounding
  • audio compression is the application of appropriate digital signal processing technology to the original digital audio signal stream (PCM encoding) to reduce (compress) its bit rate without loss of useful information or negligible loss.
  • PCM encoding digital signal processing technology
  • compression coding where the audio signal may introduce a lot of noise and certain distortion after passing through a codec system.
  • the sound wave signal is a communication signal or identification signal superimposed on the sound wave or audio.
  • the existing sound wave decoding technology is:
  • the main technical problem to be solved by the present invention is to provide a method and device for decoding sound wave information, which can significantly improve the calculation speed of sound wave decoding of an audio quantized compressed data stream by an interpreted language.
  • a technical solution adopted by the present invention is to provide a sound wave signal decoding method, the method includes: The subband data of the compressed data stream is selected from the subband data close to the audio frequency of the sound wave signal; wherein, the sound wave signal is an identification signal superimposed in the original audio file in advance, and the original audio file is generated by audio quantization processing The audio compression data stream; restore the selected sub-band data to the original digital audio signal stream of the local frequency band; wherein, the local frequency band is the same as the audio frequency of the sound wave signal or the frequency band that remains within a certain range of difference;
  • the original digital audio signal stream is subjected to Fourier transform processing; and the audio signal stream subjected to the Fourier transform processing is subjected to sound wave signal analysis to obtain the corresponding sound wave signal.
  • the selecting sub-band data close to the audio frequency of the sound wave signal from the sub-band data of the audio compression data stream based on sub-band encoding specifically includes: when receiving the audio compression data stream based on sub-band encoding, The audio compressed data stream is divided into sub-band data corresponding to the original sub-band signals; it is determined whether the audio frequency of the sound wave signal falls within the audio frequency range of each sub-band data; if so, the sub-band data is selected; Otherwise, discard the subband data.
  • restoring the selected sub-band data into the original digital audio signal stream of the local frequency band specifically includes: subjecting each selected sub-band data to a quantization restoration process to obtain the numerical sequence A 0 , A 1 of each sub-band data, ..., A n-1 ; where n is a positive integer preset according to the original audio compression data; the numerical sequence A 0 , A 1 , ..., A of each sub-band data using the polyphase synthesis filter after the precision adjustment is adjusted n-1 to restore; wherein, the accuracy of the polyphase synthesis filter is adjusted in advance to 1/m of the standard accuracy.
  • the use of the precision-adjusted polyphase synthesis filter to restore the numerical sequence of each subband data A 0 , A 1 , ..., An -1 specifically includes: selecting according to the accuracy of the polyphase synthesis filter The adjacent m numerical values in the numerical sequence of each subband data A 0 , A 1 ,..., A n-1 are divided into one group to divide the numerical sequence of each subband data into n/m groups; use precision adjustment The latter polyphase synthesis filter performs a standard windowing operation on the x-th value in each group, and multiplies the operation result by m to obtain the corresponding restored sub-band data to obtain the original digital audio of the sub-band data Signal flow; among them, 1 ⁇ x ⁇ m, and m is a natural number.
  • a sound wave signal decoding device includes: a sub-band filtering module for compressing sub-band data from an audio data stream based on sub-band encoding Select sub-band data close to the audio frequency of the sound wave signal; wherein the sound wave signal is an identification signal superimposed in an original audio file in advance, and the original audio file is processed by audio quantization to generate the audio compressed data stream;
  • the restoration module is used to restore the selected subband data into the original digital audio signal stream of the local frequency band; wherein the local frequency band is a frequency band that is the same as the audio frequency of the sound wave signal or kept within a certain difference range; the predetermined range It is pre-set according to the frequency characteristics of the original digital audio signal;
  • the Fourier transform module is used to perform Fourier transform processing on the original digital audio signal stream;
  • the decoding module is used to perform Fourier transform processing on the The audio signal stream analyzes the sound wave signal to obtain the corresponding sound wave signal.
  • the sub-band filtering module is further configured to: when receiving an audio compressed data stream based on sub-band encoding, divide the audio compressed data stream into sub-band data corresponding to the original sub-band signals; and determine Whether the audio frequency of the sound wave signal falls within the audio frequency range of each sub-band data; if so, the sub-band filtering module selects the sub-band data; otherwise, the sub-band filtering module discards the sub-band data.
  • the restoration module is also used to: subject each selected sub-band data to a quantization restoration process to obtain the numerical sequence A 0 , A 1 ,..., A n-1 of each sub-band data; where n is based on the original audio The positive integer preset by the compressed data; and the numerical sequence A 0 , A 1 ,..., A n-1 of each subband data is restored by the polyphase synthesis filter after the precision adjustment; wherein, the polyphase synthesis The accuracy of the filter is adjusted in advance to 1/m of the standard accuracy.
  • the restoration module is also used to: according to the accuracy of the polyphase synthesis filter, select the numerical sequence of each subband data A 0 , A 1 , ..., m adjacent values in A n-1 are divided into a group , To divide the numerical sequence of each subband data into n/m groups; and use the precision-adjusted polyphase synthesis filter to perform a standard windowing operation on the x-th value in each group, and multiply the operation result by m To obtain the corresponding restored sub-band data, to obtain the original digital audio signal stream of the sub-band data; wherein, 1 ⁇ x ⁇ m, and m is a natural number.
  • a sound wave signal decoding method includes: when receiving an audio compression data stream based on sub-band coding, dividing the audio compression data stream into Sub-band data corresponding to the original sub-band signals; determine whether the audio frequency of the sound wave signal falls within the audio frequency range of each sub-band data; if so, select the sub-band data; otherwise, discard the sub-band Data; wherein the sound wave signal is a pre-superimposed identification signal in the original audio file, the original audio file is processed by audio quantization to generate the audio compression data stream; the selected sub-band data is restored to the original local frequency band A digital audio signal stream; wherein the local frequency band is a frequency band that is the same as the audio frequency of the sound wave signal or remains within a certain range of difference; the original digital audio signal stream is subjected to Fourier transform processing; The converted audio signal stream is analyzed for the sound wave signal to obtain the corresponding sound wave signal.
  • restoring the selected sub-band data into the original digital audio signal stream of the local frequency band specifically includes: subjecting each selected sub-band data to a quantization restoration process to obtain the numerical sequence A 0 , A 1 of each sub-band data, ..., A n-1 ; where n is a positive integer preset according to the original audio compression data; the numerical sequence A 0 , A 1 , ..., A of each sub-band data using the polyphase synthesis filter after the precision adjustment is adjusted n-1 to restore; wherein, the accuracy of the polyphase synthesis filter is adjusted in advance to 1/m of the standard accuracy.
  • the use of the precision-adjusted polyphase synthesis filter to restore the numerical sequence of each subband data A 0 , A 1 , ..., An -1 specifically includes: selecting according to the accuracy of the polyphase synthesis filter The adjacent m numerical values in the numerical sequence of each subband data A 0 , A 1 ,..., A n-1 are divided into one group to divide the numerical sequence of each subband data into n/m groups; use precision adjustment The latter polyphase synthesis filter performs a standard windowing operation on the x-th value in each group, and multiplies the operation result by m to obtain the corresponding restored sub-band data to obtain the original digital audio of the sub-band data Signal flow; among them, 1 ⁇ x ⁇ m, and m is a natural number.
  • a sound wave signal decoding device the device includes: a sub-band filtering module, when receiving the audio compression data stream based on sub-band encoding, The audio compressed data stream is divided into sub-band data corresponding to the original sub-band signals; and it is determined whether the audio frequency of the sound wave signal falls within the audio frequency range of each sub-band data; if so, the sub-band filtering module Select the sub-band data; otherwise, the sub-band filtering module discards the sub-band data; wherein, the sound wave signal is an identification signal superimposed in an original audio file in advance, and the original audio file is processed by audio quantization Generate the audio compressed data stream; a restoration module for restoring the selected sub-band data into the original digital audio signal stream of the local frequency band; wherein the local frequency band is the same as the audio frequency of the sound wave signal or kept within a certain range of difference
  • the predetermined range is preset according to the frequency characteristics of the original digital audio signal; the Four
  • the restoration module is also used to: subject each selected sub-band data to a quantization restoration process to obtain the numerical sequence A 0 , A 1 ,..., A n-1 of each sub-band data; where n is based on the original audio The positive integer preset by the compressed data; and the numerical sequence A 0 , A 1 ,..., A n-1 of each subband data is restored by the polyphase synthesis filter after the precision adjustment; wherein, the polyphase synthesis The accuracy of the filter is adjusted in advance to 1/m of the standard accuracy.
  • the restoration module is also used to: according to the accuracy of the polyphase synthesis filter, select the numerical sequence of each subband data A 0 , A 1 , ..., m adjacent values in A n-1 are divided into a group , To divide the numerical sequence of each subband data into n/m groups; and use the precision-adjusted polyphase synthesis filter to perform a standard windowing operation on the x-th value in each group, and multiply the operation result by m To obtain the corresponding restored sub-band data, to obtain the original digital audio signal stream of the sub-band data; wherein, 1 ⁇ x ⁇ m, and m is a natural number.
  • the method and device for decoding a sound wave signal determine the sub-band data in the audio compression data stream related to the sound wave signal through the audio frequency of the sound wave signal, and perform restoration processing on the selected sub-band data; further,
  • the precision-adjusted polyphase synthesis filter is used for audio restoration of the sub-band data that is the same as the audio frequency of the sound wave signal or the difference is kept within a certain range, and the quantization restoration, reordering, and anti-aliasing of the sub-band data restoration can be omitted , Windowing, synthesis, filtering, phase correction and other calculation processes, thereby reducing the amount of calculations and improving the speed of interpretive speech decoding sound waves.
  • Fig. 1 is an execution flow chart of an acoustic signal decoding method in the first embodiment of the present invention
  • Figure 2 is a flow chart of the implementation method of "restore the selected subband data into the original digital audio signal stream of the local frequency band";
  • Fig. 3 is an execution flow chart of an acoustic wave signal decoding method in the second embodiment of the present invention.
  • FIG. 4 is an execution flow chart of the implementation method of "reducing the numerical sequence of each subband data using the polyphase synthesis filter after accuracy adjustment";
  • Fig. 5 is a structural block diagram of a sound wave signal decoding device in an embodiment of the present invention.
  • FIG. 1 is an execution flowchart of an acoustic signal decoding method in an embodiment of the present invention.
  • the method includes:
  • Step S10 Select sub-band data similar to the audio frequency of the sound wave signal from the sub-band data of the audio compression data stream based on the sub-band encoding.
  • the sound wave signal is an identification signal superimposed in an original audio file in advance, and the original audio file is processed through audio quantization to generate the audio compressed data stream.
  • Subband Coding (Subband Coding) referred to as SBC, is a coding method based on the signal spectrum, that is, the signal is decomposed into several subband signals on different frequency bands through a set of band pass filters to remove the signal Correlation; frequency shift the subband signals into baseband signals, and then sample them separately.
  • the sampled signal is quantized, coded, and combined into a total code stream and sent to the receiving end.
  • the code stream is first divided into sub-band code streams corresponding to the original sub-band signals, and then decoded, the spectrum is moved to the original position, and finally the reconstructed signal is obtained through band-pass filtering and addition.
  • step S10 selecting sub-band data similar to the audio frequency of the sound wave signal from the sub-band data of the audio compression data stream based on sub-band encoding, which is specifically implemented by the following steps:
  • Step S101 when receiving an audio compressed data stream based on sub-band coding, divide the audio compressed data stream into sub-band data corresponding to the original sub-band signals;
  • Step S102 judging whether the audio frequency of the sound wave signal falls within the audio frequency range of each subband data; if yes, go to step S103; otherwise, go to step S104;
  • Step S103 select the subband data; then return to step S102;
  • Step S104 abandon the subband data; then return to step S102;
  • the audio frequency of the sound wave signal is 18kHz-20kHz
  • the audio frequency range of each subband data of the audio compression data stream is: 0kHz-5kHz, 5kHz-10kHz, 5kHz-10kHz, 10kHz-15kHz, 15kHz-20kHz, according to
  • the audio frequency of the sound wave signal is 18kHz-20kHz to determine the need to select the subband data of the audio frequency range of 15kHz-20kHz.
  • Step S11 Restore the selected sub-band data into the original digital audio signal stream of the local frequency band.
  • the local frequency band is a frequency band that is the same as the audio frequency of the sound wave signal or kept within a certain range of difference. Further, the predetermined range is preset according to the frequency characteristics of the original digital audio signal.
  • the sub-band data selected in step S10 is only for and The subband data of the same audio frequency range of the sonic signal is restored, and the subband data different from the audio frequency range of the sonic signal is not restored.
  • the subband data selected in step S10 is subband data with an audio frequency range of 15kHz-20kHz
  • step S11 only restores the subband data with a frequency range of 18kHz-20kHz in the subband data to obtain the original digital audio signal Stream, and the sub-band data in the frequency range of 15kHz-18kHz in the sub-band data does not undergo restoration processing.
  • an audio compression data stream is divided into dozens of sub-band data, and the audio frequency range of the sound wave signal only occupies a part of the audio frequency range of a few sub-band data. Therefore, through the processing of steps S10 and S11, only the subband data whose audio frequency range is the same as the audio frequency of the sound wave signal is restored, and unnecessary calculations are omitted.
  • the subband data whose audio frequency range and the audio frequency of the sound wave signal have a difference within a certain range may be restored, and the difference in a certain range may be preset according to the audio frequency of the sound wave signal.
  • Step S12 Perform Fourier transform processing on the original digital audio signal stream.
  • Step S13 Perform sound wave signal analysis on the audio signal stream processed by the Fourier transform to obtain a corresponding sound wave signal.
  • the sub-band data whose audio frequency of the sound wave signal is the same or the difference is kept within a certain range is used to perform audio restoration on the selected sub-band data in the audio compression data stream, so as to obtain only the sound wave signal
  • the original digital audio signal stream of the frequency band whose audio frequency is the same or the difference remains within a certain range, thereby reducing the amount of calculation in the restoration process.
  • each sub-band data needs to be quantized, restored, reordered, aliased, windowed, synthesized, filtered, and phase correction is processed to obtain A sequence of values within the audio frequency range of each sub-band data, and then polyphase synthesis filtering is performed on these value sequences to obtain the original audio data.
  • the function of the polyphase synthesis filtering is to synthesize each subband data (after quantization restoration, reordering, anti-aliasing, windowing synthesis filtering, and phase correction) into the corresponding original signal.
  • Polyphase synthesis and filtering certain subband data can obtain the original signal of this subband data
  • polyphase synthesis and filtering certain two subband data can obtain the original signal of these two subband data.
  • step S11 namely, restoring the selected sub-band data into the original digital audio signal stream of the local frequency band, which is specifically implemented by the following steps:
  • Step S111 quantize and restore each selected sub-band data to obtain the numerical sequence A 0 , A 1 ,..., A n-1 of each sub-band data; where n is a preset positive value based on the original audio compression data. Integer.
  • step S112 the numerical sequence A 0 , A 1 ,..., A n-1 of each subband data is restored by using the polyphase synthesis filter after the accuracy adjustment.
  • the accuracy of the polyphase synthesis filter is adjusted in advance to 1/m of the standard accuracy.
  • m 4.
  • Step S1121 according to the accuracy of the polyphase synthesis filter, select the adjacent m numerical values in each sub-band data sequence A 0 , A 1 ,..., A n-1 into a group to divide each sub-band data
  • the numerical sequence of is divided into n/m groups;
  • Step S1122 Perform a standard windowing operation on the x-th value in each group using the polyphase synthesis filter after the accuracy adjustment, and multiply the operation result by m to obtain the corresponding restored subband data to obtain the subband
  • the original digital audio signal stream of the data 1 ⁇ x ⁇ m, and m is a natural number.
  • the above-mentioned standard windowing operation is sequentially performed on the n/m groups of numerical sequences until all numerical sequences of the sub-band data are restored, so as to obtain the original digital audio signal stream of the sub-band data.
  • each numerical sequence contains 4 numerical values: the first group, A 0 , A 1 , A 2 , and A 3 ; Two groups, A 4 , A 5 , A 6 , A 7 ;...the n/4th group, A n-4 , A n-3 , A n-2 , A n-1 .
  • the first set of numerical sequences A 0 , A 1 , A 2 , and A 3 are restored using a polyphase synthesis filter whose precision is adjusted to 1/4 of the standard precision, that is, the polyphase synthesis filter only selects the value A 0 for processing Standard windowing operation, and the result of the operation is multiplied by 4 to obtain the restored subband data of the first set of numerical sequences.
  • the remaining n/4-1 sets of numerical sequences are subjected to the standard windowing operation as described above, The calculation result is multiplied by 4 until all n/4 sets of numerical sequences complete the standard windowing calculation, so as to obtain the original digital audio signal stream after the subband data restoration process.
  • polyphase synthesis filtering The purpose of polyphase synthesis filtering is to convert frequency domain signals into time domain signals for output. It is one of the steps in the process of compressed audio restoration. This step requires a large number of floating-point addition and multiplication operations; through the accuracy of the polyphase synthesis filter Make settings, and use the polyphase synthesis filter after the accuracy setting to restore the selected subband data to the original digital audio signal of the local frequency band, thereby further reducing the amount of calculation in the audio restoration process.
  • FIG. 5 is a schematic structural diagram of a sound wave signal decoding apparatus in an embodiment of the present invention.
  • the device 20 includes a subband screening module 21, a restoration module 22, a Fourier transform module 23, and a decoding module 24.
  • the sub-band filtering module 21 is used to select sub-band data similar to the audio frequency of the sound wave signal from the sub-band data of the audio compression data stream based on sub-band encoding; wherein, the sound wave signal is pre-superimposed in the original audio file To identify the signal, the original audio file is processed by audio quantization to generate the audio compressed data stream.
  • the sub-band filtering module 21 is used to: when receiving an audio compressed data stream based on sub-band encoding, divide the audio compressed data stream into sub-band data corresponding to the original sub-band signals; and determine the sound wave Whether the audio frequency of the signal falls within the audio frequency range of each sub-band data; if so, the sub-band filtering module 21 selects the sub-band data; otherwise, the sub-band filtering module 21 discards the sub-band data.
  • the restoration module 22 is used for restoring the selected sub-band data into an original digital audio signal stream of a local frequency band; wherein the local frequency band is a frequency band that is the same as the audio frequency of the sound wave signal or kept within a certain difference range. Further, the predetermined range is preset according to the frequency characteristics of the original digital audio signal.
  • the Fourier transform module 23 is used to perform Fourier transform processing on the original digital audio signal stream.
  • the decoding module 24 is used to analyze the sound wave signal on the audio signal stream processed by the Fourier transform to obtain the corresponding sound wave signal.
  • restoration module 22 is specifically used for:
  • restoration module 22 is also used for:
  • the method and device for decoding a sound wave signal determine the sub-band data in the audio compression data stream related to the sound wave signal through the audio frequency of the sound wave signal, and perform restoration processing on the selected sub-band data; further,
  • the precision-adjusted polyphase synthesis filter is used for audio restoration of the sub-band data that is the same as the audio frequency of the sound wave signal or the difference is kept within a certain range, and the quantization restoration, reordering, and anti-aliasing of the sub-band data restoration can be omitted , Windowing, synthesis, filtering, phase correction and other calculation processes, thereby reducing the amount of calculations and improving the speed of interpretive speech decoding sound waves.
  • the disclosed system, device, and method may be implemented in other ways.
  • the device embodiments described above are merely illustrative.
  • the division of the modules or units is only a logical function division.
  • there may be other division methods for example, multiple units or components may be It can be combined or integrated into another system, or some features can be ignored or not implemented.
  • the mutual coupling or direct coupling or communication connection may be indirect coupling or communication connection through some interfaces, devices or units, and may be in electrical or other forms.
  • the units described as separate components may or may not be physically separated, and the components displayed as units may or may not be physical units, that is, they may be located in one place, or they may be distributed on multiple network units. Some or all of the units may be selected according to actual needs to achieve the objectives of the solutions of the embodiments.
  • the functional units in the various embodiments of the present invention may be integrated into one processing unit, or each unit may exist alone physically, or two or more units may be integrated into one unit.
  • the above-mentioned integrated unit can be implemented in the form of hardware or software functional unit.
  • the integrated unit is implemented in the form of a software functional unit and sold or used as an independent product, it can be stored in a computer readable storage medium.
  • the computer software product is stored in a storage medium and includes several instructions to enable a computer device (which can be a personal computer, The management server, or network device, etc.) or the processor executes all or part of the steps of the method described in each embodiment of the present invention.
  • the aforementioned storage media include: U disk, mobile hard disk, read-only memory (English: read-only memory, abbreviation: ROM), random access memory (English: Random Access Memory, abbreviation: RAM), magnetic disk or optical disk, etc.

Abstract

A sound wave information decoding method and device. Said decoding method comprises: selecting, from sub-band data of an audio compressed data stream based on sub-band coding, sub-band data having an audio frequency similar to that of a sound wave signal (S10), wherein the sound wave signal is an identification signal pre-superimposed in an original audio file, and the original audio file is processed by audio quantization to generate the audio compressed data stream; restoring the selected sub-band data to an original digital audio signal stream of a local frequency band (S11), wherein the local frequency band is a frequency band which is the same as or kept within a certain difference range from the audio frequency of the sound wave signal; performing Fourier transform processing on the original digital audio signal stream (S12); and performing sound wave signal analysis on the audio signal stream subjected to the Fourier transform processing to obtain a corresponding sound wave signal (S13). Said method and device can significantly improve the operation speed of performing sound wave decoding on an audio quantized compressed data stream by an interpreted language.

Description

一种声波信号解码的方法及装置Method and device for decoding sound wave signal 【技术领域】【Technical Field】
本发明涉及通信编码技术领域，特别涉及一种声波信号解码的方法及装置。The present invention relates to the technical field of communication coding, in particular to a method and device for decoding acoustic wave signals.
【背景技术】【Background technique】
音频量化压缩是采用音频量化处理的音频压缩技术。量化是指将信号的连续取值(或者大量可能的离散取值)近似为有限多个(或较少的)离散值的过程，即，通过四舍五入的方法将采样后的模拟信号转换成一种数字信号的过程；音频压缩是对原始数字音频信号流(PCM编码)运用适当的数字信号处理技术，在不损失有用信息量，或所引入损失可忽略的条件下，降低(压缩)其码率，也称为压缩编码，其中，音频信号在通过一个编解码系统后可能引入大量的噪声和一定的失真。Audio quantization compression is an audio compression technology that uses audio quantization processing. Quantization refers to the process of approximating the continuous value of the signal (or a large number of possible discrete values) to a finite number of (or fewer) discrete values, that is, converting the sampled analog signal into a digital signal by rounding The process of signal; audio compression is the application of appropriate digital signal processing technology to the original digital audio signal stream (PCM encoding) to reduce (compress) its bit rate without loss of useful information or negligible loss. Also called compression coding, where the audio signal may introduce a lot of noise and certain distortion after passing through a codec system.
声波信号是一种叠加在声波或音频中的通讯信号或标识信号，现有的声波解码技术为：The sound wave signal is a communication signal or identification signal superimposed on the sound wave or audio. The existing sound wave decoding technology is:
一、对原始数字音频信号流直接进行声波信号解码以得到声波信号；1. Directly decode the original digital audio signal stream to obtain the sound wave signal;
二、将音频压缩数据流还原成原始数字音频信号流，并对该原始数字音频信号流进行傅里叶变换，然后对经过傅里叶变换的音频信号进行声波信号解码，以获得声波信号。将经过音频压缩的数据流还原成原始数字音频信号流时，需要经过一系列的复杂运算。2. Restore the compressed audio data stream to the original digital audio signal stream, perform Fourier transform on the original digital audio signal stream, and then perform sound wave signal decoding on the Fourier transformed audio signal to obtain the sound wave signal. When the audio compressed data stream is restored to the original digital audio signal stream, a series of complex operations are required.
在运用解释型语言(如Python/JavaScript/Perl/Shell等)对音频量化压缩的数据流进行声波解码时，因程序在运行时，要先翻译成中间代码，再由解释器对中间代码进行解释运行，每执行一次都要翻译一次，运算速度较低，耗时较长。When using an interpreted language (such as Python/JavaScript/Perl/Shell, etc.) to decode audio quantized and compressed data streams, because the program is running, it must first be translated into intermediate code, and then the intermediate code will be interpreted by the interpreter Run, translate once every time it is executed, the calculation speed is low, and it takes a long time.
【发明内容】[Summary of the invention]
本发明主要解决的技术问题是提供一种声波信息的解码方法及装置，能够显著提高解释型语言对音频量化压缩数据流进行声波解码的运算速度。The main technical problem to be solved by the present invention is to provide a method and device for decoding sound wave information, which can significantly improve the calculation speed of sound wave decoding of an audio quantized compressed data stream by an interpreted language.
为解决上述技术问题，本发明采用的一个技术方案是：为解决上述技术问题，本发明采用的一个技术方案是：提供一种声波信号解码方法，所述方法包括：从基于子带编码的音频压缩数据流的子带数据中选取与声波信号的音频 频率相近的子带数据；其中，所述声波信号是预先叠加在原始音频文件中的标识信号，所述原始音频文件通过音频量化处理以生成所述音频压缩数据流；将选取的子带数据还原成局部频带的原始数字音频信号流；其中，所述局部频带是与声波信号的音频频率相同或保持在一定差异范围内的频带；对所述原始数字音频信号流进行傅里叶变换处理；以及对经过傅里叶变换处理的音频信号流进行声波信号的解析，以获得对应的声波信号。In order to solve the above technical problems, a technical solution adopted by the present invention is: to solve the above technical problems, a technical solution adopted by the present invention is to provide a sound wave signal decoding method, the method includes: The subband data of the compressed data stream is selected from the subband data close to the audio frequency of the sound wave signal; wherein, the sound wave signal is an identification signal superimposed in the original audio file in advance, and the original audio file is generated by audio quantization processing The audio compression data stream; restore the selected sub-band data to the original digital audio signal stream of the local frequency band; wherein, the local frequency band is the same as the audio frequency of the sound wave signal or the frequency band that remains within a certain range of difference; The original digital audio signal stream is subjected to Fourier transform processing; and the audio signal stream subjected to the Fourier transform processing is subjected to sound wave signal analysis to obtain the corresponding sound wave signal.
其中，所述从基于子带编码的音频压缩数据流的子带数据中选取与声波信号的音频频率相近的子带数据，具体包括：当接收到基于子带编码的音频压缩数据流时，将所述音频压缩数据流分成与原来的各子带信号相对应的子带数据；判断声波信号的音频频率是否落入各子带数据的音频频率范围内；若是，则选取所述子带数据；否则，放弃所述子带数据。Wherein, the selecting sub-band data close to the audio frequency of the sound wave signal from the sub-band data of the audio compression data stream based on sub-band encoding specifically includes: when receiving the audio compression data stream based on sub-band encoding, The audio compressed data stream is divided into sub-band data corresponding to the original sub-band signals; it is determined whether the audio frequency of the sound wave signal falls within the audio frequency range of each sub-band data; if so, the sub-band data is selected; Otherwise, discard the subband data.
其中，所述将选取的子带数据还原成局部频带的原始数字音频信号流，具体包括：将选取的每个子带数据经过量化还原处理以得到每个子带数据的数值序列A 0，A 1，…，A n-1；其中，n是根据原始音频压缩数据所预设的正整数；利用精度调整后的多相合成滤波器对每个子带数据的数值序列A 0，A 1，…，A n-1进行还原；其中，所述多相合成滤波器的精度被预先调整至标准精度的1/m。 Wherein, restoring the selected sub-band data into the original digital audio signal stream of the local frequency band specifically includes: subjecting each selected sub-band data to a quantization restoration process to obtain the numerical sequence A 0 , A 1 of each sub-band data, …, A n-1 ; where n is a positive integer preset according to the original audio compression data; the numerical sequence A 0 , A 1 , …, A of each sub-band data using the polyphase synthesis filter after the precision adjustment is adjusted n-1 to restore; wherein, the accuracy of the polyphase synthesis filter is adjusted in advance to 1/m of the standard accuracy.
其中，所述利用精度调整后的多相合成滤波器对每个子带数据的数值序列A 0，A 1，…，A n-1进行还原，具体包括：根据多相合成滤波器的精度，选取每个子带数据的数值序列A 0，A 1，…，A n-1中相邻的m个数值划分为一组，以将每个子带数据的数值序列划分为n/m组；利用精度调整后的多相合成滤波器对每组中的第x个数值进行标准加窗运算，并将运算结果乘以m以得到对应的还原后子带数据，以得到所述子带数据的原始数字音频信号流；其中，1≤x≤m，且m为自然数。 Wherein, the use of the precision-adjusted polyphase synthesis filter to restore the numerical sequence of each subband data A 0 , A 1 , ..., An -1 specifically includes: selecting according to the accuracy of the polyphase synthesis filter The adjacent m numerical values in the numerical sequence of each subband data A 0 , A 1 ,..., A n-1 are divided into one group to divide the numerical sequence of each subband data into n/m groups; use precision adjustment The latter polyphase synthesis filter performs a standard windowing operation on the x-th value in each group, and multiplies the operation result by m to obtain the corresponding restored sub-band data to obtain the original digital audio of the sub-band data Signal flow; among them, 1≤x≤m, and m is a natural number.
其中，m＝4，x＝1。Among them, m=4 and x=1.
为解决上述技术问题，本发明采用的另一个技术方案是：提供一种声波信号解码装置，所述装置包括：子带筛选模块，用于从基于子带编码的音频压缩数据流的子带数据中选取与声波信号的音频频率相近的子带数据；其中，所述声波信号是预先叠加在原始音频文件中的标识信号，所述原始音频文件通过音频量化处理以生成所述音频压缩数据流；还原模块，用于将选取的子带数据还原成局部频带的原始数字音频信号流；其中，所述局部频带是与声波信号的音频频率相同或保持在一定差异范围内的频带；所述预定范围是根据原始数字音 频信号的频率特征进行预先设定；傅里叶变换模块，用于对所述原始数字音频信号流进行傅里叶变换处理；解码模块，用于对经过傅里叶变换处理的音频信号流进行声波信号的解析，以获得对应的声波信号。In order to solve the above technical problems, another technical solution adopted by the present invention is to provide a sound wave signal decoding device, the device includes: a sub-band filtering module for compressing sub-band data from an audio data stream based on sub-band encoding Select sub-band data close to the audio frequency of the sound wave signal; wherein the sound wave signal is an identification signal superimposed in an original audio file in advance, and the original audio file is processed by audio quantization to generate the audio compressed data stream; The restoration module is used to restore the selected subband data into the original digital audio signal stream of the local frequency band; wherein the local frequency band is a frequency band that is the same as the audio frequency of the sound wave signal or kept within a certain difference range; the predetermined range It is pre-set according to the frequency characteristics of the original digital audio signal; the Fourier transform module is used to perform Fourier transform processing on the original digital audio signal stream; the decoding module is used to perform Fourier transform processing on the The audio signal stream analyzes the sound wave signal to obtain the corresponding sound wave signal.
其中，所述子带筛选模块还用于：当接收到基于子带编码的音频压缩数据流时，将所述音频压缩数据流分成与原来的各子带信号相对应的子带数据；以及判断声波信号的音频频率是否落入各子带数据的音频频率范围内；若是，则所述子带筛选模块选取所述子带数据；否则，所述子带筛选模块放弃所述子带数据。Wherein, the sub-band filtering module is further configured to: when receiving an audio compressed data stream based on sub-band encoding, divide the audio compressed data stream into sub-band data corresponding to the original sub-band signals; and determine Whether the audio frequency of the sound wave signal falls within the audio frequency range of each sub-band data; if so, the sub-band filtering module selects the sub-band data; otherwise, the sub-band filtering module discards the sub-band data.
其中，所述还原模块还用于：将选取的每个子带数据经过量化还原处理以得到每个子带数据的数值序列A 0，A 1，…，A n-1；其中，n是根据原始音频压缩数据所预设的正整数；以及利用精度调整后的多相合成滤波器对每个子带数据的数值序列A 0，A 1，…，A n-1进行还原；其中，所述多相合成滤波器的精度被预先调整至标准精度的1/m。 Wherein, the restoration module is also used to: subject each selected sub-band data to a quantization restoration process to obtain the numerical sequence A 0 , A 1 ,..., A n-1 of each sub-band data; where n is based on the original audio The positive integer preset by the compressed data; and the numerical sequence A 0 , A 1 ,..., A n-1 of each subband data is restored by the polyphase synthesis filter after the precision adjustment; wherein, the polyphase synthesis The accuracy of the filter is adjusted in advance to 1/m of the standard accuracy.
其中，所述还原模块还用于：根据多相合成滤波器的精度，选取每个子带数据的数值序列A 0，A 1，…，A n-1中相邻的m个数值划分为一组，以将每个子带数据的数值序列划分为n/m组；以及利用精度调整后的多相合成滤波器对每组中的第x个数值进行标准加窗运算，并将运算结果乘以m以得到对应的还原后子带数据，以得到所述子带数据的原始数字音频信号流；其中，1≤x≤m，且m为自然数。 Wherein, the restoration module is also used to: according to the accuracy of the polyphase synthesis filter, select the numerical sequence of each subband data A 0 , A 1 , ..., m adjacent values in A n-1 are divided into a group , To divide the numerical sequence of each subband data into n/m groups; and use the precision-adjusted polyphase synthesis filter to perform a standard windowing operation on the x-th value in each group, and multiply the operation result by m To obtain the corresponding restored sub-band data, to obtain the original digital audio signal stream of the sub-band data; wherein, 1≤x≤m, and m is a natural number.
其中，m＝4，x＝1。Among them, m=4 and x=1.
为解决上述技术问题，本发明采用的另一个技术方案是：一种声波信号解码方法，所述方法包括：当接收到基于子带编码的音频压缩数据流时，将所述音频压缩数据流分成与原来的各子带信号相对应的子带数据；判断声波信号的音频频率是否落入各子带数据的音频频率范围内；若是，则选取所述子带数据；否则，放弃所述子带数据；其中，所述声波信号是预先叠加在原始音频文件中的标识信号，所述原始音频文件通过音频量化处理以生成所述音频压缩数据流；将选取的子带数据还原成局部频带的原始数字音频信号流；其中，所述局部频带是与声波信号的音频频率相同或保持在一定差异范围内的频带；对所述原始数字音频信号流进行傅里叶变换处理；以及对经过傅里叶变换处理的音频信号流进行声波信号的解析，以获得对应的声波信号。In order to solve the above technical problems, another technical solution adopted by the present invention is: a sound wave signal decoding method, the method includes: when receiving an audio compression data stream based on sub-band coding, dividing the audio compression data stream into Sub-band data corresponding to the original sub-band signals; determine whether the audio frequency of the sound wave signal falls within the audio frequency range of each sub-band data; if so, select the sub-band data; otherwise, discard the sub-band Data; wherein the sound wave signal is a pre-superimposed identification signal in the original audio file, the original audio file is processed by audio quantization to generate the audio compression data stream; the selected sub-band data is restored to the original local frequency band A digital audio signal stream; wherein the local frequency band is a frequency band that is the same as the audio frequency of the sound wave signal or remains within a certain range of difference; the original digital audio signal stream is subjected to Fourier transform processing; The converted audio signal stream is analyzed for the sound wave signal to obtain the corresponding sound wave signal.
其中，所述将选取的子带数据还原成局部频带的原始数字音频信号流，具 体包括：将选取的每个子带数据经过量化还原处理以得到每个子带数据的数值序列A 0，A 1，…，A n-1；其中，n是根据原始音频压缩数据所预设的正整数；利用精度调整后的多相合成滤波器对每个子带数据的数值序列A 0，A 1，…，A n-1进行还原；其中，所述多相合成滤波器的精度被预先调整至标准精度的1/m。 Wherein, restoring the selected sub-band data into the original digital audio signal stream of the local frequency band specifically includes: subjecting each selected sub-band data to a quantization restoration process to obtain the numerical sequence A 0 , A 1 of each sub-band data, …, A n-1 ; where n is a positive integer preset according to the original audio compression data; the numerical sequence A 0 , A 1 , …, A of each sub-band data using the polyphase synthesis filter after the precision adjustment is adjusted n-1 to restore; wherein, the accuracy of the polyphase synthesis filter is adjusted in advance to 1/m of the standard accuracy.
其中，所述利用精度调整后的多相合成滤波器对每个子带数据的数值序列A 0，A 1，…，A n-1进行还原，具体包括：根据多相合成滤波器的精度，选取每个子带数据的数值序列A 0，A 1，…，A n-1中相邻的m个数值划分为一组，以将每个子带数据的数值序列划分为n/m组；利用精度调整后的多相合成滤波器对每组中的第x个数值进行标准加窗运算，并将运算结果乘以m以得到对应的还原后子带数据，以得到所述子带数据的原始数字音频信号流；其中，1≤x≤m，且m为自然数。 Wherein, the use of the precision-adjusted polyphase synthesis filter to restore the numerical sequence of each subband data A 0 , A 1 , ..., An -1 specifically includes: selecting according to the accuracy of the polyphase synthesis filter The adjacent m numerical values in the numerical sequence of each subband data A 0 , A 1 ,..., A n-1 are divided into one group to divide the numerical sequence of each subband data into n/m groups; use precision adjustment The latter polyphase synthesis filter performs a standard windowing operation on the x-th value in each group, and multiplies the operation result by m to obtain the corresponding restored sub-band data to obtain the original digital audio of the sub-band data Signal flow; among them, 1≤x≤m, and m is a natural number.
其中，m＝4，x＝1。Among them, m=4 and x=1.
为解决上述技术问题，本发明采用的另一个技术方案是：一种声波信号解码装置，所述装置包括：子带筛选模块，用于当接收到基于子带编码的音频压缩数据流时，将所述音频压缩数据流分成与原来的各子带信号相对应的子带数据；以及判断声波信号的音频频率是否落入各子带数据的音频频率范围内；若是，则所述子带筛选模块选取所述子带数据；否则，所述子带筛选模块放弃所述子带数据；其中，所述声波信号是预先叠加在原始音频文件中的标识信号，所述原始音频文件通过音频量化处理以生成所述音频压缩数据流；还原模块，用于将选取的子带数据还原成局部频带的原始数字音频信号流；其中，所述局部频带是与声波信号的音频频率相同或保持在一定差异范围内的频带；所述预定范围是根据原始数字音频信号的频率特征进行预先设定；傅里叶变换模块，用于对所述原始数字音频信号流进行傅里叶变换处理；解码模块，用于对经过傅里叶变换处理的音频信号流进行声波信号的解析，以获得对应的声波信号。In order to solve the above technical problems, another technical solution adopted by the present invention is: a sound wave signal decoding device, the device includes: a sub-band filtering module, when receiving the audio compression data stream based on sub-band encoding, The audio compressed data stream is divided into sub-band data corresponding to the original sub-band signals; and it is determined whether the audio frequency of the sound wave signal falls within the audio frequency range of each sub-band data; if so, the sub-band filtering module Select the sub-band data; otherwise, the sub-band filtering module discards the sub-band data; wherein, the sound wave signal is an identification signal superimposed in an original audio file in advance, and the original audio file is processed by audio quantization Generate the audio compressed data stream; a restoration module for restoring the selected sub-band data into the original digital audio signal stream of the local frequency band; wherein the local frequency band is the same as the audio frequency of the sound wave signal or kept within a certain range of difference The predetermined range is preset according to the frequency characteristics of the original digital audio signal; the Fourier transform module is used to perform Fourier transform processing on the original digital audio signal stream; the decoding module is used to The sound wave signal analysis is performed on the audio signal stream processed by the Fourier transform to obtain the corresponding sound wave signal.
其中，所述还原模块还用于：将选取的每个子带数据经过量化还原处理以得到每个子带数据的数值序列A 0，A 1，…，A n-1；其中，n是根据原始音频压缩数据所预设的正整数；以及利用精度调整后的多相合成滤波器对每个子带数据的数值序列A 0，A 1，…，A n-1进行还原；其中，所述多相合成滤波器的精度被预先调整至标准精度的1/m。 Wherein, the restoration module is also used to: subject each selected sub-band data to a quantization restoration process to obtain the numerical sequence A 0 , A 1 ,..., A n-1 of each sub-band data; where n is based on the original audio The positive integer preset by the compressed data; and the numerical sequence A 0 , A 1 ,..., A n-1 of each subband data is restored by the polyphase synthesis filter after the precision adjustment; wherein, the polyphase synthesis The accuracy of the filter is adjusted in advance to 1/m of the standard accuracy.
其中，所述还原模块还用于：根据多相合成滤波器的精度，选取每个子带数据的数值序列A 0，A 1，…，A n-1中相邻的m个数值划分为一组，以将每个子 带数据的数值序列划分为n/m组；以及利用精度调整后的多相合成滤波器对每组中的第x个数值进行标准加窗运算，并将运算结果乘以m以得到对应的还原后子带数据，以得到所述子带数据的原始数字音频信号流；其中，1≤x≤m，且m为自然数。 Wherein, the restoration module is also used to: according to the accuracy of the polyphase synthesis filter, select the numerical sequence of each subband data A 0 , A 1 , ..., m adjacent values in A n-1 are divided into a group , To divide the numerical sequence of each subband data into n/m groups; and use the precision-adjusted polyphase synthesis filter to perform a standard windowing operation on the x-th value in each group, and multiply the operation result by m To obtain the corresponding restored sub-band data, to obtain the original digital audio signal stream of the sub-band data; wherein, 1≤x≤m, and m is a natural number.
其中，m＝4，x＝1。Among them, m=4 and x=1.
本发明实施方式提供的一种声波信号解码的方法及装置，通过声波信号的音频频率确定与其相关的音频压缩数据流中的子带数据，并针对选取的子带数据进行还原处理；进一步地，利用精度调整后的多相合成滤波器针对与声波信号音频频率相同或差异保持在一定范围内的子带数据进行音频还原，可省略子带数据还原所进行的量化还原、重新排序、消除混叠、加窗合成滤波、相位修正等运算过程，从而减少运算量，提升解释型语音对声波解码的速度。The method and device for decoding a sound wave signal provided by the embodiment of the present invention determine the sub-band data in the audio compression data stream related to the sound wave signal through the audio frequency of the sound wave signal, and perform restoration processing on the selected sub-band data; further, The precision-adjusted polyphase synthesis filter is used for audio restoration of the sub-band data that is the same as the audio frequency of the sound wave signal or the difference is kept within a certain range, and the quantization restoration, reordering, and anti-aliasing of the sub-band data restoration can be omitted , Windowing, synthesis, filtering, phase correction and other calculation processes, thereby reducing the amount of calculations and improving the speed of interpretive speech decoding sound waves.
【附图说明】【Explanation of the drawings】
图1是本发明第一实施方式中一种声波信号解码方法的执行流程图；Fig. 1 is an execution flow chart of an acoustic signal decoding method in the first embodiment of the present invention;
图2是“将选取的子带数据还原成局部频带的原始数字音频信号流”的实现方法执行流程图；Figure 2 is a flow chart of the implementation method of "restore the selected subband data into the original digital audio signal stream of the local frequency band";
图3是本发明第二实施方式中一种声波信号解码方法的执行流程图；Fig. 3 is an execution flow chart of an acoustic wave signal decoding method in the second embodiment of the present invention;
图4是“利用精度调整后的多相合成滤波器对每个子带数据的数值序列进行还原”的实现方法的执行流程图；FIG. 4 is an execution flow chart of the implementation method of "reducing the numerical sequence of each subband data using the polyphase synthesis filter after accuracy adjustment";
图5是本发明一实施方式中一种声波信号解码装置的结构框图。Fig. 5 is a structural block diagram of a sound wave signal decoding device in an embodiment of the present invention.
【具体实施方式】【detailed description】
为详细说明本发明的技术内容、构造特征、所实现目的及效果，以下结合附图和实施例对本发明进行详细说明。In order to describe in detail the technical content, structural features, achieved objectives and effects of the present invention, the present invention will be described in detail below with reference to the accompanying drawings and embodiments.
请参阅图1，为本发明实施方式中一种声波信号解码方法的执行流程图，该方法包括：Please refer to FIG. 1, which is an execution flowchart of an acoustic signal decoding method in an embodiment of the present invention. The method includes:
步骤S10，从基于子带编码的音频压缩数据流的子带数据中选取与声波信号的音频频率相近的子带数据。其中，该声波信号是预先叠加在原始音频文件中的标识信号，该原始音频文件通过音频量化处理以生成该音频压缩数据流。Step S10: Select sub-band data similar to the audio frequency of the sound wave signal from the sub-band data of the audio compression data stream based on the sub-band encoding. Wherein, the sound wave signal is an identification signal superimposed in an original audio file in advance, and the original audio file is processed through audio quantization to generate the audio compressed data stream.
子带编码：(Subband Coding)简称SBC，是一种以信号频谱为依据的编码方法，即，通过一组带通滤波器将信号分解成若干个在不同频段上的子带信号， 以去除信号相关性；将子带信号分别进行频率搬移转变成基带信号，再对它们分别取样。取样后的信号经过量化、编码，并合成一个总的码流传送给接收端。在接收端，首先把码流分成与原来的各子带信号相对应的子带码流，然后解码、将频谱搬至原来的位置，最后经带通滤波、相加得到重建的信号。Subband Coding: (Subband Coding) referred to as SBC, is a coding method based on the signal spectrum, that is, the signal is decomposed into several subband signals on different frequency bands through a set of band pass filters to remove the signal Correlation; frequency shift the subband signals into baseband signals, and then sample them separately. The sampled signal is quantized, coded, and combined into a total code stream and sent to the receiving end. At the receiving end, the code stream is first divided into sub-band code streams corresponding to the original sub-band signals, and then decoded, the spectrum is moved to the original position, and finally the reconstructed signal is obtained through band-pass filtering and addition.
具体地，请参阅图2，步骤S10，从基于子带编码的音频压缩数据流的子带数据中选取与声波信号的音频频率相近的子带数据，具体通过如下步骤实现：Specifically, please refer to FIG. 2, step S10, selecting sub-band data similar to the audio frequency of the sound wave signal from the sub-band data of the audio compression data stream based on sub-band encoding, which is specifically implemented by the following steps:
步骤S101，当接收到基于子带编码的音频压缩数据流时，将该音频压缩数据流分成与原来的各子带信号相对应的子带数据；Step S101, when receiving an audio compressed data stream based on sub-band coding, divide the audio compressed data stream into sub-band data corresponding to the original sub-band signals;
步骤S102，判断声波信号的音频频率是否落入各子带数据的音频频率范围内；若是，则进入步骤S103；否则，进入步骤S104；Step S102, judging whether the audio frequency of the sound wave signal falls within the audio frequency range of each subband data; if yes, go to step S103; otherwise, go to step S104;
步骤S103，选取该子带数据；然后返回步骤S102；Step S103, select the subband data; then return to step S102;
步骤S104，放弃该子带数据；然后返回步骤S102；Step S104, abandon the subband data; then return to step S102;
具体地，当声波信号的音频频率仅落入一个子带数据的音频频率范围内时，则只要选取这一个子带数据；当声波信号的音频频率落入多个子带数据的音频频率范围内时，则选取这多个子带数据。Specifically, when the audio frequency of the sound wave signal only falls within the audio frequency range of one subband data, only this subband data is selected; when the audio frequency of the sound wave signal falls within the audio frequency range of multiple subband data , Select these multiple subband data.
例如，声波信号的音频频率是18kHz-20kHz，音频压缩数据流的各个子带数据的音频频率范围分别是：0kHz-5kHz、5kHz-10kHz、5kHz-10kHz、10kHz-15kHz、15kHz-20kHz，则根据声波信号的音频频率18kHz-20kHz确定需要选取音频频率范围为15kHz-20kHz的子带数据。For example, the audio frequency of the sound wave signal is 18kHz-20kHz, and the audio frequency range of each subband data of the audio compression data stream is: 0kHz-5kHz, 5kHz-10kHz, 5kHz-10kHz, 10kHz-15kHz, 15kHz-20kHz, according to The audio frequency of the sound wave signal is 18kHz-20kHz to determine the need to select the subband data of the audio frequency range of 15kHz-20kHz.
步骤S11，将选取的子带数据还原成局部频带的原始数字音频信号流。Step S11: Restore the selected sub-band data into the original digital audio signal stream of the local frequency band.
其中，该局部频带是与声波信号的音频频率相同或保持在一定差异范围内的频带。进一步地，该预定范围是根据原始数字音频信号的频率特征进行预先设定。Among them, the local frequency band is a frequency band that is the same as the audio frequency of the sound wave signal or kept within a certain range of difference. Further, the predetermined range is preset according to the frequency characteristics of the original digital audio signal.
具体地，对所有选取的子带数据，按照该音频压缩数据的标准还原流程和方法，还原成局部频带的原始数字音频信号流；也就是说，通过步骤S10选取的子带数据，仅对与声波信号音频频率范围相同的子带数据进行还原处理，而与声波信号音频频率范围不同的子带数据不进行还原处理。例如，步骤S10选取的子带数据为音频频率范围为15kHz-20kHz的子带数据，则步骤S11仅对该子带数据中频率范围为18kHz-20kHz的子带数据进行还原以获得原始数字音频信号流，而对该子带数据中频率范围为15kHz-18kHz的子带数据不进行还原处理。Specifically, for all selected sub-band data, according to the standard restoration process and method of the audio compression data, it is restored to the original digital audio signal stream of the local frequency band; that is, the sub-band data selected in step S10 is only for and The subband data of the same audio frequency range of the sonic signal is restored, and the subband data different from the audio frequency range of the sonic signal is not restored. For example, the subband data selected in step S10 is subband data with an audio frequency range of 15kHz-20kHz, then step S11 only restores the subband data with a frequency range of 18kHz-20kHz in the subband data to obtain the original digital audio signal Stream, and the sub-band data in the frequency range of 15kHz-18kHz in the sub-band data does not undergo restoration processing.
由于现实状况下，通常一个音频压缩数据流会被划分成数十个子带数据，而声波信号音频频率范围仅会占中其中几个子带数据的部分音频频率范围。所以，通过步骤S10、S11的处理，仅对音频频率范围与声波信号音频频率相同的子带数据进行还原处理，省略不必要的运算。Due to actual conditions, usually an audio compression data stream is divided into dozens of sub-band data, and the audio frequency range of the sound wave signal only occupies a part of the audio frequency range of a few sub-band data. Therefore, through the processing of steps S10 and S11, only the subband data whose audio frequency range is the same as the audio frequency of the sound wave signal is restored, and unnecessary calculations are omitted.
在其他实施方式中，还可以对音频频率范围与声波信号音频频率的差异保持在一定范围内的子带数据进行还原处理，该一定范围的差异可以根据声波信号音频频率进行预先设置。In other embodiments, the subband data whose audio frequency range and the audio frequency of the sound wave signal have a difference within a certain range may be restored, and the difference in a certain range may be preset according to the audio frequency of the sound wave signal.
步骤S12，对原始数字音频信号流进行傅里叶变换处理。Step S12: Perform Fourier transform processing on the original digital audio signal stream.
步骤S13，对经过傅里叶变换处理的音频信号流进行声波信号的解析，以获得对应的声波信号。Step S13: Perform sound wave signal analysis on the audio signal stream processed by the Fourier transform to obtain a corresponding sound wave signal.
在本发明中，利用选取声波信号音频频率相同或差异保持在一定范围内的子带数据，对音频压缩数据流中所选取的这些子带数据进行音频还原，从而得到只包含了具备与声波信号音频频率相同或差异保持在一定范围内的频带的原始数字音频信号流，从而降低了还原过程中的运算量。In the present invention, the sub-band data whose audio frequency of the sound wave signal is the same or the difference is kept within a certain range is used to perform audio restoration on the selected sub-band data in the audio compression data stream, so as to obtain only the sound wave signal The original digital audio signal stream of the frequency band whose audio frequency is the same or the difference remains within a certain range, thereby reducing the amount of calculation in the restoration process.
对于基于(或部分基于)子带编码生成的音频压缩数据流，需要对每个子带数据进行量化还原、重新排序、消除混叠、加窗合成滤波、以及相位修正等运算步骤进行处理，以得到每个子带数据音频频率范围内的数值序列，然后对这些数值序列进行多相合成滤波以得到原始音频数据。其中，该多相合成滤波的作用，就是将各个子带数据(经过量化还原、重新排序、消除混叠、加窗合成滤波、以及相位修正)合成为对应的原始信号。多相合成滤波某一个子带数据，就能够得到这个子带数据的原始信号，多相合成滤波某两个子带数据，就能够得到这两个子带数据的原始信号。For audio compression data streams generated based on (or partly based on) sub-band encoding, each sub-band data needs to be quantized, restored, reordered, aliased, windowed, synthesized, filtered, and phase correction is processed to obtain A sequence of values within the audio frequency range of each sub-band data, and then polyphase synthesis filtering is performed on these value sequences to obtain the original audio data. Among them, the function of the polyphase synthesis filtering is to synthesize each subband data (after quantization restoration, reordering, anti-aliasing, windowing synthesis filtering, and phase correction) into the corresponding original signal. Polyphase synthesis and filtering certain subband data can obtain the original signal of this subband data, and polyphase synthesis and filtering certain two subband data can obtain the original signal of these two subband data.
请同时参阅图3，步骤S11，即，将选取的子带数据还原成局部频带的原始数字音频信号流，具体通过如下步骤实现：Please refer to FIG. 3 at the same time, step S11, namely, restoring the selected sub-band data into the original digital audio signal stream of the local frequency band, which is specifically implemented by the following steps:
步骤S111，将选取的每个子带数据经过量化还原处理以得到每个子带数据的数值序列A 0，A 1，…，A n-1；其中，n是根据原始音频压缩数据所预设的正整数。 Step S111, quantize and restore each selected sub-band data to obtain the numerical sequence A 0 , A 1 ,..., A n-1 of each sub-band data; where n is a preset positive value based on the original audio compression data. Integer.
步骤S112，利用精度调整后的多相合成滤波器对每个子带数据的数值序列A 0，A 1，…，A n-1进行还原。 In step S112, the numerical sequence A 0 , A 1 ,..., A n-1 of each subband data is restored by using the polyphase synthesis filter after the accuracy adjustment.
其中，该多相合成滤波器的精度被预先调整至标准精度的1/m。在本实施方式中，m＝4。Among them, the accuracy of the polyphase synthesis filter is adjusted in advance to 1/m of the standard accuracy. In this embodiment, m=4.
具体地，请同时参阅图4，利用精度调整后的多相合成滤波器对每个子带数据的数值序列A 0，A 1，…，A n-1进行还原，具体通过如下步骤实现： Specifically, referring to Fig. 4 at the same time, the numerical sequence A 0 , A 1 ,..., A n-1 of each sub-band data is restored by using the polyphase synthesis filter after precision adjustment, which is specifically realized by the following steps:
步骤S1121，根据多相合成滤波器的精度，选取每个子带数据的数值序列A 0，A 1，…，A n-1中相邻的m个数值划分为一组，以将每个子带数据的数值序列划分为n/m组； Step S1121, according to the accuracy of the polyphase synthesis filter, select the adjacent m numerical values in each sub-band data sequence A 0 , A 1 ,..., A n-1 into a group to divide each sub-band data The numerical sequence of is divided into n/m groups;
步骤S1122，利用精度调整后的多相合成滤波器对每组中的第x个数值进行标准加窗运算，并将运算结果乘以m以得到对应的还原后子带数据，以得到该子带数据的原始数字音频信号流。其中，1≤x≤m，且m为自然数。Step S1122: Perform a standard windowing operation on the x-th value in each group using the polyphase synthesis filter after the accuracy adjustment, and multiply the operation result by m to obtain the corresponding restored subband data to obtain the subband The original digital audio signal stream of the data. Among them, 1≤x≤m, and m is a natural number.
具体地，依次对n/m组的数值序列进行如上所述的标准加窗运算，直至该子带数据的全部数值序列被还原处理完成，从而得到该子带数据的原始数字音频信号流。Specifically, the above-mentioned standard windowing operation is sequentially performed on the n/m groups of numerical sequences until all numerical sequences of the sub-band data are restored, so as to obtain the original digital audio signal stream of the sub-band data.
例如，当m＝4，x＝1时，即，每个子带数据的数值序列分组后，每组数值序列包含4个数值：第一组，A 0、A 1、A 2、A 3；第二组，A 4、A 5、A 6、A 7；……第n/4组，A n-4、A n-3、A n-2、A n-1。利用精度调整为标准精度1/4的多相合成滤波器对第一组数值序列A 0、A 1、A 2、A 3进行还原处理，即，该多相合成滤波器仅选取数值A 0进行标准加窗运算，并将运算结果乘以4以得到第一组数值序列还原后的子带数据，同样地，对其余的n/4-1组数值序列进行如上所述的标准加窗运算，并将运算结果乘以4，直至n/4组数值序列全部完成标准加窗运算，从而得到该子带数据还原处理后的原始数字音频信号流。 For example, when m=4 and x=1, that is, after the numerical sequence of each subband data is grouped, each numerical sequence contains 4 numerical values: the first group, A 0 , A 1 , A 2 , and A 3 ; Two groups, A 4 , A 5 , A 6 , A 7 ;...the n/4th group, A n-4 , A n-3 , A n-2 , A n-1 . The first set of numerical sequences A 0 , A 1 , A 2 , and A 3 are restored using a polyphase synthesis filter whose precision is adjusted to 1/4 of the standard precision, that is, the polyphase synthesis filter only selects the value A 0 for processing Standard windowing operation, and the result of the operation is multiplied by 4 to obtain the restored subband data of the first set of numerical sequences. Similarly, the remaining n/4-1 sets of numerical sequences are subjected to the standard windowing operation as described above, The calculation result is multiplied by 4 until all n/4 sets of numerical sequences complete the standard windowing calculation, so as to obtain the original digital audio signal stream after the subband data restoration process.
如上所述，对子带数据的数值序列中的第1个数值进行标准加窗运算后的结果乘以4，忽略第2、3、4个数值，对第5个数据进行标准加窗运算后的结果乘以4，忽略第6、7、8个数值，直到该子带数据处理完成。As mentioned above, multiply the result of the standard windowing operation on the first value in the numerical sequence of the subband data by 4, ignoring the second, third, and fourth values, and perform the standard windowing operation on the fifth data The result of is multiplied by 4, and the 6th, 7th, and 8th values are ignored until the subband data processing is completed.
在本发明中，通过如上步骤S21、S22所述，假设每个子带数据中的m个连续的数值在进行标准加窗运算后的结果是相同的，因此仅处理每组其中1/m的数据，以损失精确度的代价，来达到降低运算量的目的。In the present invention, as described in the above steps S21 and S22, it is assumed that the m consecutive values in each subband data have the same result after the standard windowing operation, so only 1/m of the data in each group is processed , At the cost of loss of accuracy, to achieve the purpose of reducing the amount of calculation.
多相合成滤波的目的是将频域信号转化为时域信号输出，是压缩音频还原过程中的步骤之一，这一步需要进行大量浮点加法和乘法运算；通过对多相合成滤波器的精度进行设置，并利用精度设置后的多相合成滤波器将选取的子带数据还原成局部频带的原始数字音频信号，从而进一步地减少音频还原过程中的运算量。The purpose of polyphase synthesis filtering is to convert frequency domain signals into time domain signals for output. It is one of the steps in the process of compressed audio restoration. This step requires a large number of floating-point addition and multiplication operations; through the accuracy of the polyphase synthesis filter Make settings, and use the polyphase synthesis filter after the accuracy setting to restore the selected subband data to the original digital audio signal of the local frequency band, thereby further reducing the amount of calculation in the audio restoration process.
请参阅图5，为本发明实施方式中的一种声波信号的解码装置的结构示意图。 该装置20包括子带筛选模块21、还原模块22、傅里叶变换模块23以及解码模块24。Please refer to FIG. 5, which is a schematic structural diagram of a sound wave signal decoding apparatus in an embodiment of the present invention. The device 20 includes a subband screening module 21, a restoration module 22, a Fourier transform module 23, and a decoding module 24.
子带筛选模块21，用于从基于子带编码的音频压缩数据流的子带数据中选取与声波信号的音频频率相近的子带数据；其中，该声波信号是预先叠加在原始音频文件中的标识信号，该原始音频文件通过音频量化处理以生成该音频压缩数据流。The sub-band filtering module 21 is used to select sub-band data similar to the audio frequency of the sound wave signal from the sub-band data of the audio compression data stream based on sub-band encoding; wherein, the sound wave signal is pre-superimposed in the original audio file To identify the signal, the original audio file is processed by audio quantization to generate the audio compressed data stream.
具体地，该子带筛选模块21用于：当接收到基于子带编码的音频压缩数据流时，将该音频压缩数据流分成与原来的各子带信号相对应的子带数据；以及判断声波信号的音频频率是否落入各子带数据的音频频率范围内；若是，则该子带筛选模块21选取该子带数据；否则，该子带筛选模块21放弃该子带数据。Specifically, the sub-band filtering module 21 is used to: when receiving an audio compressed data stream based on sub-band encoding, divide the audio compressed data stream into sub-band data corresponding to the original sub-band signals; and determine the sound wave Whether the audio frequency of the signal falls within the audio frequency range of each sub-band data; if so, the sub-band filtering module 21 selects the sub-band data; otherwise, the sub-band filtering module 21 discards the sub-band data.
还原模块22，用于将选取的子带数据还原成局部频带的原始数字音频信号流；其中，该局部频带是与声波信号的音频频率相同或保持在一定差异范围内的频带。进一步地，该预定范围是根据原始数字音频信号的频率特征进行预先设定。The restoration module 22 is used for restoring the selected sub-band data into an original digital audio signal stream of a local frequency band; wherein the local frequency band is a frequency band that is the same as the audio frequency of the sound wave signal or kept within a certain difference range. Further, the predetermined range is preset according to the frequency characteristics of the original digital audio signal.
傅里叶变换模块23，用于对原始数字音频信号流进行傅里叶变换处理。The Fourier transform module 23 is used to perform Fourier transform processing on the original digital audio signal stream.
解码模块24，用于对经过傅里叶变换处理的音频信号流进行声波信号的解析，以获得对应的声波信号。The decoding module 24 is used to analyze the sound wave signal on the audio signal stream processed by the Fourier transform to obtain the corresponding sound wave signal.
进一步地，该还原模块22具体还用于：Further, the restoration module 22 is specifically used for:
将选取的每个子带数据经过量化还原处理以得到每个子带数据的数值序列A 0，A 1，…，A n-1；其中，n是根据原始音频压缩数据所预设的正整数；以及 Quantize and restore each selected sub-band data to obtain the numerical sequence A 0 , A 1 ,..., A n-1 of each sub-band data; where n is a positive integer preset according to the original audio compression data; and
利用精度调整后的多相合成滤波器对每个子带数据的数值序列A 0，A 1，…，A n-1进行还原；其中，该多相合成滤波器的精度被预先调整至标准精度的1/m。在本实施方式中，m＝4。 Use the adjusted polyphase synthesis filter to restore the numerical sequence A 0 , A 1 ,..., A n-1 of each subband data; wherein, the accuracy of the polyphase synthesis filter is adjusted to the standard accuracy in advance 1/m. In this embodiment, m=4.
更进一步地，该还原模块22还用于：Furthermore, the restoration module 22 is also used for:
根据多相合成滤波器的精度，选取每个子带数据的数值序列A 0，A 1，…，A n-1中相邻的m个数值划分为一组，以将每个子带数据的数值序列划分为n/m组；以及 According to the accuracy of the polyphase synthesis filter, select the numerical sequence of each sub-band data A 0 , A 1 ,..., A n-1 adjacent m numerical values are divided into a group to divide the numerical sequence of each sub-band data Divide into n/m groups; and
利用精度调整后的多相合成滤波器对每组中的第x个数值进行标准加窗运算，并将运算结果乘以m以得到对应的还原后子带数据，以得到该子带数据的原始数字音频信号流。其中，1≤x≤m，且m为自然数。Use the precision-adjusted polyphase synthesis filter to perform a standard windowing operation on the xth value in each group, and multiply the operation result by m to obtain the corresponding restored subband data to obtain the original subband data Digital audio signal flow. Among them, 1≤x≤m, and m is a natural number.
本发明实施方式提供的一种声波信号解码的方法及装置，通过声波信号的 音频频率确定与其相关的音频压缩数据流中的子带数据，并针对选取的子带数据进行还原处理；进一步地，利用精度调整后的多相合成滤波器针对与声波信号音频频率相同或差异保持在一定范围内的子带数据进行音频还原，可省略子带数据还原所进行的量化还原、重新排序、消除混叠、加窗合成滤波、相位修正等运算过程，从而减少运算量，提升解释型语音对声波解码的速度。The method and device for decoding a sound wave signal provided by the embodiment of the present invention determine the sub-band data in the audio compression data stream related to the sound wave signal through the audio frequency of the sound wave signal, and perform restoration processing on the selected sub-band data; further, The precision-adjusted polyphase synthesis filter is used for audio restoration of the sub-band data that is the same as the audio frequency of the sound wave signal or the difference is kept within a certain range, and the quantization restoration, reordering, and anti-aliasing of the sub-band data restoration can be omitted , Windowing, synthesis, filtering, phase correction and other calculation processes, thereby reducing the amount of calculations and improving the speed of interpretive speech decoding sound waves.
在本发明所提供的几个实施例中，应该理解到，所揭露的系统，装置和方法，可以通过其它的方式实现。例如，以上所描述的装置实施例仅仅是示意性的，例如，所述模块或单元的划分，仅仅为一种逻辑功能划分，实际实现时可以有另外的划分方式，例如多个单元或组件可以结合或者可以集成到另一个系统，或一些特征可以忽略，或不执行。另一点，相互之间的耦合或直接耦合或通讯连接可以是通过一些接口，装置或单元的间接耦合或通讯连接，可以是电性或其它的形式。In the several embodiments provided by the present invention, it should be understood that the disclosed system, device, and method may be implemented in other ways. For example, the device embodiments described above are merely illustrative. For example, the division of the modules or units is only a logical function division. In actual implementation, there may be other division methods, for example, multiple units or components may be It can be combined or integrated into another system, or some features can be ignored or not implemented. In addition, the mutual coupling or direct coupling or communication connection may be indirect coupling or communication connection through some interfaces, devices or units, and may be in electrical or other forms.
所述作为分离部件说明的单元可以是或者也可以不是物理上分开的，作为单元显示的部件可以是或者也可以不是物理单元，即可以位于一个地方，或者也可以分布到多个网络单元上。可以根据实际的需要选择其中的部分或者全部单元来实现本实施例方案的目的。另外，在本发明各个实施例中的各功能单元可以集成在一个处理单元中，也可以是各个单元单独物理存在，也可以两个或两个以上单元集成在一个单元中。上述集成的单元既可以采用硬件的形式实现，也可以采用软件功能单元的形式实现。The units described as separate components may or may not be physically separated, and the components displayed as units may or may not be physical units, that is, they may be located in one place, or they may be distributed on multiple network units. Some or all of the units may be selected according to actual needs to achieve the objectives of the solutions of the embodiments. In addition, the functional units in the various embodiments of the present invention may be integrated into one processing unit, or each unit may exist alone physically, or two or more units may be integrated into one unit. The above-mentioned integrated unit can be implemented in the form of hardware or software functional unit.
所述集成的单元如果以软件功能单元的形式实现并作为独立的产品销售或使用时，可以存储在一个计算机可读取存储介质中。基于这样的理解，本发明的技术方案的全部或部分可以以软件产品的形式体现出来，该计算机软件产品存储在一个存储介质中，包括若干指令用以使得一台计算机设备(可以是个人计算机，管理服务器，或者网络设备等)或处理器执行本发明各个实施例所述方法的全部或部分步骤。而前述的存储介质包括：U盘、移动硬盘、只读存储器(英文：read-only memory，缩写：ROM)、随机存取存储器(英文：Random Access Memory，缩写：RAM)、磁碟或者光盘等各种可以存储程序代码的介质。If the integrated unit is implemented in the form of a software functional unit and sold or used as an independent product, it can be stored in a computer readable storage medium. Based on this understanding, all or part of the technical solution of the present invention can be embodied in the form of a software product. The computer software product is stored in a storage medium and includes several instructions to enable a computer device (which can be a personal computer, The management server, or network device, etc.) or the processor executes all or part of the steps of the method described in each embodiment of the present invention. The aforementioned storage media include: U disk, mobile hard disk, read-only memory (English: read-only memory, abbreviation: ROM), random access memory (English: Random Access Memory, abbreviation: RAM), magnetic disk or optical disk, etc. Various media that can store program codes.
以上所述仅为本发明的实施例，并非因此限制本发明的专利范围，凡是利用本发明说明书及附图内容所作的等效结构或等效流程变换，或直接或间接运用在其他相关的技术领域，均同理包括在本发明的专利保护范围内。The above are only the embodiments of the present invention and do not limit the patent scope of the present invention. Any equivalent structure or equivalent process transformation made by using the content of the description and drawings of the present invention, or directly or indirectly applied to other related technologies In the same way, all fields are included in the scope of patent protection of the present invention.

Claims (18)

  1. 一种声波信号解码方法，其特征在于，所述方法包括：An acoustic signal decoding method, characterized in that the method includes:
    从基于子带编码的音频压缩数据流的子带数据中选取与声波信号的音频频率相近的子带数据；其中，所述声波信号是预先叠加在原始音频文件中的标识信号，所述原始音频文件通过音频量化处理以生成所述音频压缩数据流；Select sub-band data similar to the audio frequency of the sound wave signal from the sub-band data of the audio compression data stream based on sub-band encoding; wherein, the sound wave signal is an identification signal superimposed in the original audio file in advance, and the original audio The file is processed by audio quantization to generate the audio compressed data stream;
    将选取的子带数据还原成局部频带的原始数字音频信号流；其中，所述局部频带是与声波信号的音频频率相同或保持在一定差异范围内的频带；Restoring the selected sub-band data into the original digital audio signal stream of the local frequency band; wherein the local frequency band is a frequency band that is the same as the audio frequency of the sound wave signal or kept within a certain range of difference;
    对所述原始数字音频信号流进行傅里叶变换处理；以及Perform Fourier transform processing on the original digital audio signal stream; and
    对经过傅里叶变换处理的音频信号流进行声波信号的解析，以获得对应的声波信号。The sound wave signal analysis is performed on the audio signal stream processed by the Fourier transform to obtain the corresponding sound wave signal.
  2. 根据权利要求1所述的声波信号解码方法，其特征在于，所述从基于子带编码的音频压缩数据流的子带数据中选取与声波信号的音频频率相近的子带数据，具体包括：The sound wave signal decoding method according to claim 1, wherein said selecting sub-band data similar to the audio frequency of the sound wave signal from the sub-band data of the audio compressed data stream based on sub-band coding specifically comprises:
    当接收到基于子带编码的音频压缩数据流时，将所述音频压缩数据流分成与原来的各子带信号相对应的子带数据；When receiving an audio compressed data stream based on sub-band coding, dividing the audio compressed data stream into sub-band data corresponding to the original sub-band signals;
    判断声波信号的音频频率是否落入各子带数据的音频频率范围内；若是，则选取所述子带数据；否则，放弃所述子带数据。Determine whether the audio frequency of the sound wave signal falls within the audio frequency range of each subband data; if so, select the subband data; otherwise, discard the subband data.
  3. 根据权利要求2所述的声波信号解码方法，其特征在于，所述将选取的子带数据还原成局部频带的原始数字音频信号流，具体包括：The sound wave signal decoding method according to claim 2, wherein said restoring the selected sub-band data into the original digital audio signal stream of the local frequency band specifically comprises:
    将选取的每个子带数据经过量化还原处理以得到每个子带数据的数值序列A 0，A 1，…，A n-1；其中，n是根据原始音频压缩数据所预设的正整数； Quantize and restore each selected sub-band data to obtain the numerical sequence A 0 , A 1 , ..., A n-1 of each sub-band data; where n is a positive integer preset according to the original audio compression data;
    利用精度调整后的多相合成滤波器对每个子带数据的数值序列A 0，A 1，…，A n-1进行还原；其中，所述多相合成滤波器的精度被预先调整至标准精度的1/m。 The precision-adjusted polyphase synthesis filter is used to restore the numerical sequence A 0 , A 1 ,..., A n-1 of each subband data; wherein, the precision of the polyphase synthesis filter is adjusted to the standard precision in advance Of 1/m.
  4. 根据权利要求3所述的声波信号解码方法，其特征在于，所述利用精度调整后的多相合成滤波器对每个子带数据的数值序列A 0，A 1，…，A n-1进行还原，具体包括： The sound wave signal decoding method according to claim 3, characterized in that, the numerical sequence A 0 , A 1 ,..., An -1 of each subband data is restored by the polyphase synthesis filter after the accuracy adjustment , Specifically including:
    根据多相合成滤波器的精度，选取每个子带数据的数值序列A 0，A 1，…，A n-1中相邻的m个数值划分为一组，以将每个子带数据的数值序列划分为n/m组； According to the accuracy of the polyphase synthesis filter, select the numerical sequence of each sub-band data A 0 , A 1 ,..., A n-1 adjacent m numerical values are divided into a group to divide the numerical sequence of each sub-band data Divided into n/m groups;
    利用精度调整后的多相合成滤波器对每组中的第x个数值进行标准加窗运算，并将运算结果乘以m以得到对应的还原后子带数据，以得到所述子带数据 的原始数字音频信号流；其中，1≤x≤m，且m为自然数。Use the polyphase synthesis filter after precision adjustment to perform standard windowing operation on the x-th value in each group, and multiply the operation result by m to obtain the corresponding restored sub-band data to obtain the sub-band data Original digital audio signal stream; where 1≤x≤m, and m is a natural number.
  5. 根据权利要求4所述的声波信号解码方法，其特征在于，m＝4，x＝1。The sound wave signal decoding method according to claim 4, wherein m=4 and x=1.
  6. 一种声波信号解码装置，其特征在于，所述装置包括：A sound wave signal decoding device, characterized in that the device includes:
    子带筛选模块，用于从基于子带编码的音频压缩数据流的子带数据中选取与声波信号的音频频率相近的子带数据；其中，所述声波信号是预先叠加在原始音频文件中的标识信号，所述原始音频文件通过音频量化处理以生成所述音频压缩数据流；The sub-band filtering module is used to select sub-band data similar to the audio frequency of the sound wave signal from the sub-band data of the audio compression data stream based on sub-band encoding; wherein, the sound wave signal is pre-superimposed in the original audio file An identification signal, the original audio file is processed by audio quantization to generate the audio compressed data stream;
    还原模块，用于将选取的子带数据还原成局部频带的原始数字音频信号流；其中，所述局部频带是与声波信号的音频频率相同或保持在一定差异范围内的频带；所述预定范围是根据原始数字音频信号的频率特征进行预先设定；The restoration module is used to restore the selected subband data into the original digital audio signal stream of the local frequency band; wherein the local frequency band is a frequency band that is the same as the audio frequency of the sound wave signal or kept within a certain difference range; the predetermined range It is preset according to the frequency characteristics of the original digital audio signal;
    傅里叶变换模块，用于对所述原始数字音频信号流进行傅里叶变换处理；A Fourier transform module, configured to perform Fourier transform processing on the original digital audio signal stream;
    解码模块，用于对经过傅里叶变换处理的音频信号流进行声波信号的解析，以获得对应的声波信号。The decoding module is used to analyze the sound wave signal on the audio signal stream processed by the Fourier transform to obtain the corresponding sound wave signal.
  7. 根据权利要求6所述的声波信号解码装置，其特征在于，所述子带筛选模块还用于：The sound wave signal decoding device according to claim 6, wherein the sub-band filtering module is further configured to:
    当接收到基于子带编码的音频压缩数据流时，将所述音频压缩数据流分成与原来的各子带信号相对应的子带数据；以及When receiving an audio compressed data stream based on sub-band coding, dividing the audio compressed data stream into sub-band data corresponding to the original sub-band signals; and
    判断声波信号的音频频率是否落入各子带数据的音频频率范围内；若是，则所述子带筛选模块选取所述子带数据；否则，所述子带筛选模块放弃所述子带数据。Determine whether the audio frequency of the sound wave signal falls within the audio frequency range of each sub-band data; if so, the sub-band filtering module selects the sub-band data; otherwise, the sub-band filtering module discards the sub-band data.
  8. 根据权利要求7所述的声波信号解码装置，其特征在于，所述还原模块还用于：The sound wave signal decoding device according to claim 7, wherein the restoration module is further used for:
    将选取的每个子带数据经过量化还原处理以得到每个子带数据的数值序列A 0，A 1，…，A n-1；其中，n是根据原始音频压缩数据所预设的正整数；以及 Quantize and restore each selected sub-band data to obtain the numerical sequence A 0 , A 1 ,..., A n-1 of each sub-band data; where n is a positive integer preset according to the original audio compression data; and
    利用精度调整后的多相合成滤波器对每个子带数据的数值序列A 0，A 1，…，A n-1进行还原；其中，所述多相合成滤波器的精度被预先调整至标准精度的1/m。 The precision-adjusted polyphase synthesis filter is used to restore the numerical sequence A 0 , A 1 ,..., A n-1 of each subband data; wherein, the precision of the polyphase synthesis filter is adjusted to the standard precision in advance Of 1/m.
  9. 根据权利要求8所述的声波信号解码装置，其特征在于，所述还原模块还用于：The sound wave signal decoding device according to claim 8, wherein the restoration module is further configured to:
    根据多相合成滤波器的精度，选取每个子带数据的数值序列A 0，A 1，…，A n-1中相邻的m个数值划分为一组，以将每个子带数据的数值序列划分为n/m组；以及 According to the accuracy of the polyphase synthesis filter, select the numerical sequence of each sub-band data A 0 , A 1 ,..., A n-1 adjacent m numerical values are divided into a group to divide the numerical sequence of each sub-band data Divide into n/m groups; and
    利用精度调整后的多相合成滤波器对每组中的第x个数值进行标准加窗运算，并将运算结果乘以m以得到对应的还原后子带数据，以得到所述子带数据的原始数字音频信号流；其中，1≤x≤m，且m为自然数。Use the polyphase synthesis filter after precision adjustment to perform standard windowing operation on the x-th value in each group, and multiply the operation result by m to obtain the corresponding restored sub-band data to obtain the sub-band data Original digital audio signal stream; where 1≤x≤m, and m is a natural number.
  10. 根据权利要求9所述的声波信号解码装置，其特征在于，m＝4，x＝1。The sound wave signal decoding device according to claim 9, wherein m=4 and x=1.
  11. 一种声波信号解码方法，其特征在于，所述方法包括：An acoustic signal decoding method, characterized in that the method includes:
    当接收到基于子带编码的音频压缩数据流时，将所述音频压缩数据流分成与原来的各子带信号相对应的子带数据；When receiving an audio compressed data stream based on sub-band coding, dividing the audio compressed data stream into sub-band data corresponding to the original sub-band signals;
    判断声波信号的音频频率是否落入各子带数据的音频频率范围内；若是，则选取所述子带数据；否则，放弃所述子带数据；其中，所述声波信号是预先叠加在原始音频文件中的标识信号，所述原始音频文件通过音频量化处理以生成所述音频压缩数据流；Determine whether the audio frequency of the sound wave signal falls within the audio frequency range of each sub-band data; if so, select the sub-band data; otherwise, discard the sub-band data; wherein the sound wave signal is superimposed on the original audio frequency in advance An identification signal in a file, the original audio file is processed by audio quantization to generate the audio compressed data stream;
    将选取的子带数据还原成局部频带的原始数字音频信号流；其中，所述局部频带是与声波信号的音频频率相同或保持在一定差异范围内的频带；Restoring the selected sub-band data into an original digital audio signal stream of a local frequency band; wherein the local frequency band is a frequency band that is the same as the audio frequency of the sound wave signal or kept within a certain range of difference;
    对所述原始数字音频信号流进行傅里叶变换处理；以及Perform Fourier transform processing on the original digital audio signal stream; and
    对经过傅里叶变换处理的音频信号流进行声波信号的解析，以获得对应的声波信号。The sound wave signal analysis is performed on the audio signal stream processed by the Fourier transform to obtain the corresponding sound wave signal.
  12. 根据权利要求11所述的声波信号解码方法，其特征在于，所述将选取的子带数据还原成局部频带的原始数字音频信号流，具体包括：The sound wave signal decoding method according to claim 11, wherein said restoring the selected sub-band data into the original digital audio signal stream of the local frequency band specifically comprises:
    将选取的每个子带数据经过量化还原处理以得到每个子带数据的数值序列A 0，A 1，…，A n-1；其中，n是根据原始音频压缩数据所预设的正整数； Quantize and restore each selected sub-band data to obtain the numerical sequence A 0 , A 1 , ..., A n-1 of each sub-band data; where n is a positive integer preset according to the original audio compression data;
    利用精度调整后的多相合成滤波器对每个子带数据的数值序列A 0，A 1，…，A n-1进行还原；其中，所述多相合成滤波器的精度被预先调整至标准精度的1/m。 The precision-adjusted polyphase synthesis filter is used to restore the numerical sequence A 0 , A 1 ,..., A n-1 of each subband data; wherein, the precision of the polyphase synthesis filter is adjusted to the standard precision in advance Of 1/m.
  13. 根据权利要求12所述的声波信号解码方法，其特征在于，所述利用精度调整后的多相合成滤波器对每个子带数据的数值序列A 0，A 1，…，A n-1进行还原，具体包括： The sound wave signal decoding method according to claim 12, characterized in that, the numerical sequence A 0 , A 1 ,..., An -1 of each subband data is restored by the polyphase synthesis filter after the accuracy adjustment , Specifically including:
    根据多相合成滤波器的精度，选取每个子带数据的数值序列A 0，A 1，…，A n-1中相邻的m个数值划分为一组，以将每个子带数据的数值序列划分为n/m组； According to the accuracy of the polyphase synthesis filter, select the numerical sequence of each sub-band data A 0 , A 1 ,..., A n-1 adjacent m numerical values are divided into a group to divide the numerical sequence of each sub-band data Divided into n/m groups;
    利用精度调整后的多相合成滤波器对每组中的第x个数值进行标准加窗运算，并将运算结果乘以m以得到对应的还原后子带数据，以得到所述子带数据的原始数字音频信号流；其中，1≤x≤m，且m为自然数。Use the polyphase synthesis filter after precision adjustment to perform standard windowing operation on the x-th value in each group, and multiply the operation result by m to obtain the corresponding restored sub-band data to obtain the sub-band data Original digital audio signal stream; where 1≤x≤m, and m is a natural number.
  14. 根据权利要求13所述的声波信号解码方法，其特征在于，m＝4，x＝1。The sound wave signal decoding method according to claim 13, wherein m=4 and x=1.
  15. 一种声波信号解码装置，其特征在于，所述装置包括：A sound wave signal decoding device, characterized in that the device comprises:
    子带筛选模块，用于当接收到基于子带编码的音频压缩数据流时，将所述音频压缩数据流分成与原来的各子带信号相对应的子带数据；以及判断声波信号的音频频率是否落入各子带数据的音频频率范围内；若是，则所述子带筛选模块选取所述子带数据；否则，所述子带筛选模块放弃所述子带数据；其中，所述声波信号是预先叠加在原始音频文件中的标识信号，所述原始音频文件通过音频量化处理以生成所述音频压缩数据流；The sub-band filtering module is used to divide the audio compressed data stream into sub-band data corresponding to the original sub-band signals when the audio compressed data stream based on sub-band coding is received; and to determine the audio frequency of the sound wave signal Whether it falls within the audio frequency range of each sub-band data; if so, the sub-band filtering module selects the sub-band data; otherwise, the sub-band filtering module discards the sub-band data; wherein, the sound wave signal Is an identification signal superimposed in an original audio file in advance, and the original audio file is processed by audio quantization to generate the audio compressed data stream;
    还原模块，用于将选取的子带数据还原成局部频带的原始数字音频信号流；其中，所述局部频带是与声波信号的音频频率相同或保持在一定差异范围内的频带；所述预定范围是根据原始数字音频信号的频率特征进行预先设定；The restoration module is used to restore the selected subband data into the original digital audio signal stream of the local frequency band; wherein the local frequency band is a frequency band that is the same as the audio frequency of the sound wave signal or kept within a certain difference range; the predetermined range It is preset according to the frequency characteristics of the original digital audio signal;
    傅里叶变换模块，用于对所述原始数字音频信号流进行傅里叶变换处理；A Fourier transform module, configured to perform Fourier transform processing on the original digital audio signal stream;
    解码模块，用于对经过傅里叶变换处理的音频信号流进行声波信号的解析，以获得对应的声波信号。The decoding module is used to analyze the sound wave signal on the audio signal stream processed by the Fourier transform to obtain the corresponding sound wave signal.
  16. 根据权利要求15所述的声波信号解码装置，其特征在于，所述还原模块还用于：The sound wave signal decoding device according to claim 15, wherein the restoration module is further configured to:
    将选取的每个子带数据经过量化还原处理以得到每个子带数据的数值序列A 0，A 1，…，A n-1；其中，n是根据原始音频压缩数据所预设的正整数；以及 Quantize and restore each selected sub-band data to obtain the numerical sequence A 0 , A 1 ,..., A n-1 of each sub-band data; where n is a positive integer preset according to the original audio compression data; and
    利用精度调整后的多相合成滤波器对每个子带数据的数值序列A 0，A 1，…，A n-1进行还原；其中，所述多相合成滤波器的精度被预先调整至标准精度的1/m。 The precision-adjusted polyphase synthesis filter is used to restore the numerical sequence A 0 , A 1 ,..., A n-1 of each subband data; wherein, the precision of the polyphase synthesis filter is adjusted to the standard precision in advance Of 1/m.
  17. 根据权利要求16所述的声波信号解码装置，其特征在于，所述还原模块还用于：The sound wave signal decoding device according to claim 16, wherein the restoration module is further configured to:
    根据多相合成滤波器的精度，选取每个子带数据的数值序列A 0，A 1，…，A n-1中相邻的m个数值划分为一组，以将每个子带数据的数值序列划分为n/m组；以及 According to the accuracy of the polyphase synthesis filter, select the numerical sequence of each sub-band data A 0 , A 1 ,..., A n-1 adjacent m numerical values are divided into a group to divide the numerical sequence of each sub-band data Divide into n/m groups; and
    利用精度调整后的多相合成滤波器对每组中的第x个数值进行标准加窗运算，并将运算结果乘以m以得到对应的还原后子带数据，以得到所述子带数据的原始数字音频信号流；其中，1≤x≤m，且m为自然数。Use the polyphase synthesis filter after precision adjustment to perform standard windowing operation on the x-th value in each group, and multiply the operation result by m to obtain the corresponding restored sub-band data to obtain the sub-band data Original digital audio signal stream; where 1≤x≤m, and m is a natural number.
  18. 根据权利要求17所述的声波信号解码装置，其特征在于，m＝4，x＝1。The sound wave signal decoding device according to claim 17, wherein m=4 and x=1.
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