WO2021244409A1 - Procédé et dispositif de décodage de signal d'onde sonore - Google Patents

Procédé et dispositif de décodage de signal d'onde sonore Download PDF

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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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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 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
    • G10L19/032—Quantisation or dequantisation of spectral components

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  • 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.

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Abstract

L'invention concerne un procédé et un dispositif de décodage d'informations d'onde sonore. Ledit procédé de décodage comprend : la sélection, à partir de données de sous-bande d'un flux de données audio compressées sur la base d'un codage en sous-bande, de données de sous-bande ayant une fréquence audio similaire à celle d'un signal d'onde sonore (S10), le signal d'onde sonore étant un signal d'identification pré-superposé dans un fichier audio d'origine, et le fichier audio d'origine étant traité par quantification audio pour générer le flux de données audio compressées ; le rétablissement des données de sous-bande sélectionnées en un flux de signaux audio numériques d'origine d'une bande de fréquences locale (S11), la bande de fréquences locale étant une bande de fréquences qui est identique à la fréquence audio du signal d'onde sonore ou maintenue dans une certaine plage de différence par rapport à celle-ci ; la réalisation d'un traitement par transformation de Fourier sur le flux de signal audio numérique d'origine (S12) ; et la réalisation d'une analyse de signal d'onde sonore sur le flux de signal audio soumis au traitement de transformation de Fourier pour obtenir un signal d'onde sonore correspondant (S13). Ledit procédé et ledit dispositif peuvent améliorer significativement la vitesse de l'opération d'exécution d'un décodage d'ondes sonores sur un flux de données audio quantifiées compressées par un langage interprété.
PCT/CN2021/096619 2020-05-30 2021-05-28 Procédé et dispositif de décodage de signal d'onde sonore WO2021244409A1 (fr)

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