CN103137135A - LPC coefficient quantization method and device and multi-coding-core audio coding method and device - Google Patents
LPC coefficient quantization method and device and multi-coding-core audio coding method and device Download PDFInfo
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Abstract
The invention relates to a method and a device for quantizing LPC coefficients for audio signal coding. The method comprises the following steps: s1, determining the type of the input audio signal based on a predetermined rule; s2, linear prediction processing is carried out on the input audio signal, and an LPC coefficient is calculated; and S3, for different audio signal types, applying a vector quantization code book matched with the audio signal type to carry out vector quantization on the LPC coefficients. The invention also relates to a multi-coding-core audio coding method and a device adopting the LPC coefficient quantization method and device. The invention quantizes LPC coefficients based on audio signal classification, is applied to a multi-coding-core coding algorithm which has at least one coding core and uses linear prediction LPC to code at least two types of audio signals, and can further improve the quantization precision of prediction parameters of an internal linear prediction coding module, thereby improving the efficiency of the whole digital audio coding algorithm and the subjective sound quality of a coder.
Description
Technical field
The present invention relates to Digital Audio Coding Technology, more particularly, relate to a kind of LPC coefficient quantization method and apparatus for audio-frequency signal coding and a kind of odd encoder core audio coding method and equipment.
Background technology
In digital audio encoding, because sound signal is very complicated, generally comprise music class signal, voice class signal and mix class signal etc., some audio coding algorithms such as MPEG-1, MPEG-2, MPEG-4, Dolby AC-3 and DTS etc. operate mainly in high code check high-quality, when the code efficiency for the voice class signal under low code check lower; And other ITU G series standard encryption algorithm is mainly for low code check voice signal, for code efficiency decline of broadband signal.In order unanimously to obtain higher code efficiency to all types of sound signals, generally need to adopt have the hybrid coding structure of Multi-encoding kernel, as the AMR-WB+(of 3GPP referring to 3GPP TS26.290: " Audio codec processing functions; Extended AMR Wideband codec; Transcoding functions ") and MPEG-D USAC(are referring to ISO/IEC DIS23003-3-Information technology--MPEG audio technologies--Part3:Unified speech and audio coding ") etc.In these hybrid coding algorithms, there is different compression algorithms to process to each sound signal type, expectation integrated encode performance is improved.
In AMR-WB+, adopt ACELP(Algebraic Code Excited Linear Prediction for voice signal, the algebraic code excited linear prediction) coding core, generally adopt TCX(Transform Coded Excitation for mixing class and music class signal, change the code excitation) coding core, two kinds of coding cores are all used LPC(Linear Predictive Coding, linear predictive coding)) technology is described the short-time spectrum envelope of voice, thereby is a critical problem in voice coding to the high effective quantization of LPC coefficient.Because the dynamic range of LPC coefficient is larger, consideration for composite filter stability and quantitative efficiency, the LPC coefficient is converted into re-quantization after the parameter of other form of equivalent on mathematics usually, common representation is ISF(Immittance Spectral Frequency, the immittance spectral frequencies coefficient) or LSF(Line Spectral Frequency, line spectral frequencies parameter).LSF is as a kind of frequency domain parameter of LPC coefficient, because it has better quantification and interpolation characteristic, the voice coding end is often the LSF parameter with the LPC coefficients conversion, and then the LSF parameter is quantized (generally adopting vector quantization technology), the tone decoding end carries out the LSF parameter after re-quantization is quantized, and the LSF parameter is converted to the LPC coefficient again, so LSF is widely used in based on the LPC voice coding.
At MPEG-D USAC(Unified Speech and Audio Coding, unified voice/audio coding) in coding, for music class signal, adopts efficient AAC(AdvancedAudio coding, Advanced Audio Coding) encode; For voice signal, generally adopt ACELP class coding core; For mixing the class signal, generally adopt TCX class coding core.As AMR-WB+, in MPEG-D USAC hybrid coding structure, ACELP and TCX coding core can share the LPC coding techniques.
Although the odd encoder core audio coding algorithms such as AMR-WB+ and MPEG-D USAC begin to have the type of pair input audio signal to analyze, for dissimilar, adopt different coding core, obtain comprehensive optimum coding efficient.In AMR-WB+, the voice signal class adopts ACELP coding core, and music class and mixing class signal adopt TCX coding core; In MPEG-D USAC, the voice class signal adopts ACELP coding core, mixes class and adopts TCX coding core, and the music class adopts AAC coding core.ACELP and TCX have all been adopted in these two kinds of odd encoder core audio coding algorithms of AMR-WB+ and MPEG-D USAC, and these two coding cores can share a linear prediction LPC technology, and be all generally after the LPC coefficients conversion is LSF spectrum parameter, carry out again vector quantization coding, and adoptable vector quantization method has a variety of, for example, the applying date is on July 17th, 2012, application number is 201210246780.9, the Chinese patent application that name is called " method and system that is used for voice signal LPC coefficient is carried out multi-stage vector quantization " just discloses a kind of multilevel vector quantization method, but the code book that these vector quantization methods generate does not rely on the type of the digital audio and video signals of input, namely to all sound signals, all only generate a cover vector quantization code book, thereby the quantified precision of LPC coefficient is not still very desirable, thereby affect the code efficiency of overall digital audio coding algorithm and the subjective sound quality of scrambler.
Summary of the invention
First technical matters that the present invention will solve is, for the defects of prior art, provides a kind of LPC coefficient quantization method and apparatus that is used for audio-frequency signal coding that can further improve quantified precision.
Second technical matters that the present invention will solve is, for the defects of prior art, provide a kind of and can improve the quantified precision of intra-prediction parameter and then improve the efficient of overall digital audio coding algorithm and odd encoder core audio coding method and the encoding device of subjective sound quality.
The present invention solves the technical scheme that its first technical matters adopts: propose a kind of LPC coefficient quantization method for audio-frequency signal coding, comprise the steps:
S1, determine the type of input audio signal based on predetermined rule;
S2, input audio signal is carried out linear prediction process, calculate the LPC coefficient;
S3, for different sound signal types, the vector quantization code book of using with this sound signal type matching carries out vector quantization to described LPC coefficient.
The above-mentioned LPC coefficient quantization method for audio-frequency signal coding of the present invention,
Described step S2 further comprises:
It is the LSF parameter of equivalence with described LPC coefficients conversion;
Described step S3 further comprises:
For different sound signal types, the vector quantization code book of application and this sound signal type matching carries out vector quantization to described LSF parameter.
The above-mentioned LPC coefficient quantization method for audio-frequency signal coding of the present invention, in described step S3, vector quantization adopts multilevel vector quantization method.
The above-mentioned LPC coefficient quantization method for audio-frequency signal coding of the present invention, described method also comprised before step S1:
The required vector quantization code book of coding that will build for the signal model of different audio signals type is stored in this locality.
The above-mentioned LPC coefficient quantization method for audio-frequency signal coding of the present invention further comprises:
S4, send the coding parameter of vector quantization to multiplexer and be multiplexed in total audio coding frame.
The present invention also proposes a kind of LPC coefficient quantization device for audio-frequency signal coding for solving its first technical matters, comprising:
The audio types determination module is for determine the type of input audio signal based on predetermined rule;
The linear prediction processing module is used for that input audio signal is carried out linear prediction and processes, and calculates the LPC coefficient;
Spectrum parameter quantification module is used for for different sound signal types, and the vector quantization code book of application and this sound signal type matching carries out vector quantization to described LPC coefficient.
The above-mentioned LPC coefficient quantization device for audio-frequency signal coding of the present invention, described linear prediction processing module further comprises:
The LPC coefficients calculation block is used for input audio signal and carries out the linear prediction processing, calculates the LPC coefficient;
The equivalence modular converter, being used for described LPC coefficients conversion is the LSF parameter of equivalence.
The above-mentioned LPC coefficient quantization device for audio-frequency signal coding of the present invention, described spectrum parameter quantification module is further used for for different sound signal types, and the vector quantization code book of application and this sound signal type matching carries out vector quantization to described LSF parameter.
The above-mentioned LPC coefficient quantization device for audio-frequency signal coding of the present invention also comprises:
Memory module is used for storage for the required vector quantization code book of coding of the signal model structure of different audio signals type.
The above-mentioned LPC coefficient quantization device for audio-frequency signal coding of the present invention also sends the coding parameter of vector quantization to multiplexer and is multiplexed into total audio coding frame.
The present invention solves the technical scheme that its second technical matters adopt: propose a kind of odd encoder core audio coding method, comprise the steps:
A, the type of input audio signal is analyzed;
B, for a plurality of sound signal types, adopt corresponding a plurality of coding core to encode, wherein, at least one coding is checked the sound signal of at least two sound signal types and is carried out linear predictive coding;
Wherein, described linear predictive coding quantizes the LPC coefficient by the above-mentioned LPC coefficient quantization method that is used for audio-frequency signal coding.
The present invention also proposes a kind of odd encoder core audio coding equipment for solving its second technical matters, comprising:
The audio signal classification processing module is used for the type of input audio signal is analyzed;
A plurality of coding cores are used for based on the sound signal type corresponding coding audio signal, and wherein, at least one coding is checked the sound signal of at least two sound signal types and carried out linear predictive coding;
Wherein, described linear predictive coding quantizes the LPC coefficient by the above-mentioned LPC coefficient quantization device that is used for audio-frequency signal coding.
The LPC coefficient quantization method and apparatus that is used for audio-frequency signal coding by the present invention, when the LPC parameter quantification in encryption algorithm is encoded, the vector quantization code book that mates most separately is provided respectively for the different audio signals type, in the situation that do not need extra audio frequency signal type indication bit expense, can be further to LPC spectrum parameter improvement quantified precision.And then the code efficiency that the odd encoder core audio coding method of this LPC coefficient quantization method and apparatus of employing of the present invention and equipment can improve the binary encoding algorithm perhaps reduces coding bit rate under same quality, improves the subjective sound quality of scrambler.
Description of drawings
The invention will be further described below in conjunction with drawings and Examples, in accompanying drawing:
Fig. 1 is the cataloged procedure schematic block diagram of MPEG-D USAC encryption algorithm;
Fig. 2 is the process flow diagram of the LPC coefficient quantization method that is used for audio-frequency signal coding of one embodiment of the invention;
Fig. 3 is the process flow diagram of the LPC coefficient quantization method that is used for audio-frequency signal coding of another embodiment of the present invention;
Fig. 4 is the logic diagram of the LPC coefficient quantization device that is used for audio-frequency signal coding of one embodiment of the invention.
Embodiment
In order to make purpose of the present invention, technical scheme and advantage clearer, below in conjunction with drawings and Examples, the present invention is further elaborated.Should be appreciated that specific embodiment described herein only in order to explain the present invention, is not intended to limit the present invention.
The signal type of sound signal can be divided into 2 classes, 3 classes or polymorphic type more.Can be voice signal, non-speech audio when being divided into 2 class; Can be voice signal, music signal, voice and music mix class signal when being divided into 3 class.LPC coefficient quantization method and apparatus for audio-frequency signal coding of the present invention just adopts respectively different match vector quantization code books to carry out vector quantization to the LPC spectral coefficient based on audio signal classification, thereby can further improve the quantified precision of spectrum parameter, can reduce in other words the needed bit number of spectrum parameter coding under equal accuracy.
For example, AMR-WB+ odd encoder core hybrid coding algorithm becomes voice class signal, music class signal and voice music to mix the class signal audio signal classification.When the present invention is applied to AMR-WB+, for the voice class signal, use the one group vector quantization code book relevant to this signal model to carry out vector quantization, and then carry out the ACELP coding; And mix the class signal for music class and voice music, and quantize with two groups of different vector quantization code books separately respectively, then complete TCX and process.
Again for example, MPEG-D USAC odd encoder core hybrid coding algorithm also becomes audio signal classification voice class signal, music class signal and voice music to mix the class signal.When the present invention is applied to MPEG-D USAC, because only voice class and music voice mix the signal demand LPC processing of two types of classes, therefore provide respectively the vector quantization code book of coupling to quantize for this two classes signal, and then correspondence is carried out ACELP coding or TCX coding.Below will introduce in detail the present invention as an example of MPEG-D USAC encryption algorithm example.
Fig. 1 shows the cataloged procedure of MPEG-D USAC encryption algorithm.As shown in Figure 1, the MPEG-DUSAC encryption algorithm mainly comprises three phases.Pretreatment stage: in step 110 to input PCM(Pulse Code Modulation, pulse code modulation (PCM)) sound signal resamples, its objective is in input sampling rate and coded sample rate not simultaneously, adjust input sampling rate to the optimum sampling rate that is fit to coding and processes; In step 130, input pcm audio signal is carried out the signal type analysis, process in order to carry out different coding for signal with different type; To carry out the SBR(Spectral band replication around mpeg encoded (MPEG Surround) and enhancing through the sound signal that resamples, frequency range copies in step 120) process.Dissimilar based on sound signal is admitted to two coding cores through pretreated sound signal, i.e. first the 140 and second coding branch (time domain coding core) 150 of coding branch (Frequency Domain Coding core).As front introduction, in MPEG-D USAC, the voice class signal adopts ACELP coding core, mixes the class signal and adopts TCX coding core, and music class signal adopts AAC coding core.Also namely, enter the first coding branch 140 through pretreated music class signal, through tone estimation, piece switching controls, psychoacoustic model control, filtering, TNS(Temporal Noise Shaping, time domain noise shaped), the processing such as M/S coding; Enter the second coding branch 150 through pretreated voice class signal, carry out carrying out again the ACELP coding after LPC spectrum parameter quantification processes 151; Enter the second coding branch 150 through pretreated mixing class signal, carry out carrying out again the TCX coding after LPC spectrum parameter quantification processes 151.Two branches signals out are through after aftertreatments 160, and are through multiplexer, that all coding parameters are multiplexing 170, export total audio coding frame.
Technical scheme of the present invention is mainly reflected in processes 151 improvement to LPC spectrum parameter quantification, proposes a kind of LPC coefficient quantization method and apparatus based on audio signal classification, below will provide in detail explanation.About other functional module and the step of MPEG-D USAC encryption algorithm shown in Figure 1, the prior art that is well known to those skilled in the art is not therefore repeat them here.
Fig. 2 is used for the process flow diagram of the LPC coefficient quantization method 200 of audio-frequency signal coding according to an embodiment of the invention.As shown in Figure 2, the method 200 comprises the steps:
In step 210, determine the type of input audio signal based on predetermined rule.As previously mentioned, the odd encoder core audio coding algorithms such as MPEG-D USAC can adopt different coding core based on different signal types, will inevitably be at first carry out the signal type analysis to the pcm audio signal of input, and with the classification type parameter coding in compressed bit stream (shown in label in Fig. 1 130).For example, in MPEG-USAC, be divided into voice class signal, music class signal, music voice mixing class signal three types.Thereby LPC coefficient quantization method of the present invention does not need additionally to increase signal type processing module and signal type indication information again, can parse based on predetermined rule the type of input audio signal from compressed bit stream.Therefore, when LPC coefficient quantization method of the present invention is applied to odd encoder core audio coding algorithm, need to not increase any overhead in coded frame.
In later step 220, input audio signal is carried out linear prediction process (LPC), calculate the LPC coefficient.To inputting the PCM signal, the present invention calculates the LPC coefficient by equitable subsection (256 sampling points calculate once as the ACELP coding, and TCX coding possibility 256,512 or 1024 sampling points calculate once).
In later step 230, for different sound signal types, the vector quantization code book of application and this sound signal type matching carries out vector quantization to the LPC coefficient.The present invention can be in advance for encode the respectively design of required vector quantization code book of the signal model of different audio signals type, construct a vector quantization code book that mates most with each sound signal type, and be stored in this locality.Due to the vector quantization code book of the corresponding coupling of different audio signals type, therefore use the present invention and carry out audio-frequency signal coding and need to these vector quantization code books of storage be arranged at coding side and decoding end, can increase certain memory space requirements.
In step 230, the LPC coefficient is carried out vector quantization, can adopt known in the art and feasible various vector quantization methods, for example, the applying date is on July 17th, 2012, application number is 201210246780.9, and name is called the disclosed multilevel vector quantization method of Chinese patent application of " method and system that is used for voice signal LPC coefficient is carried out multi-stage vector quantization ".
Fig. 3 is the process flow diagram of the LPC coefficient quantization method 300 that is used for audio-frequency signal coding of another specific embodiment according to the present invention.As shown in Figure 3, the method 300 comprises the steps:
In step 310, determine the type of input audio signal based on predetermined rule.
In later step 320, input audio signal is carried out linear prediction process, calculate the LPC coefficient.
In later step 330, be the LSF parameter of equivalence with the LPC coefficients conversion.
In later step 340, for different sound signal types, the vector quantization code book with this sound signal type matching of using local storage carries out vector quantization to the LSF parameter.As previously mentioned, the present invention can carry out respectively the design of vector quantization code book for the LSF parameter of different audio signals type in advance, constructs the vector quantization code book that mates most of coding needs and is stored in this locality.
In later step 350, send the coding parameter of vector quantization to multiplexer and be multiplexed in total audio coding frame, in order to send receiving end (demoder) to.
The LPC coefficient quantization method that is used for audio-frequency signal coding by the present invention, when the LPC parameter quantification in encryption algorithm is encoded, the vector quantization code book that mates most separately is provided respectively for the different audio signals type, in the situation that do not need extra audio frequency signal type indication bit expense, can be further to LPC spectrum parameter improvement quantified precision, thereby improve the code efficiency of binary encoding algorithm, perhaps reduce coding bit rate under same quality.
Fig. 4 is used for the logic diagram of the LPC coefficient quantization device 400 of audio-frequency signal coding according to an embodiment of the invention.The LPC coefficient quantization device 400 that is used for audio-frequency signal coding comprises audio types determination module 410, linear prediction processing module 420, spectrum parameter quantification module 430 and memory module 440.Wherein, audio types determination module 410 is used for determining based on predetermined rule the type of input audio signal.As previously mentioned, the odd encoder core audio coding algorithms such as MPEG-D USAC can adopt different coding core based on different signal types, must be provided with the audio signal classification processing module at first to the input the pcm audio signal carry out the signal type analysis, and with the classification type parameter coding in compressed bit stream (shown in label in Fig. 1 130).Thereby, audio types determination module 410 can parse based on predetermined rule the type of input audio signal from compressed bit stream, do not need additionally to increase the sound signal type analysis and process and indication information, therefore need to not increase any overhead in coded frame.Linear prediction processing module 420 is used for that input audio signal is carried out linear prediction to be processed, and calculates the LPC coefficient.Memory module 440 is used in this locality storage for the required vector quantization code book of the constructed coding of the signal model of different audio signals type.Spectrum parameter quantification module 430 is used for for different sound signal types, and the vector quantization code book with this sound signal type matching of using local storage carries out vector quantization to the LPC coefficient that linear prediction processing module 420 calculates.
In specific embodiment, as shown in Figure 4, linear prediction processing module 420 further comprises LPC coefficients calculation block 421 and equivalent modular converter 422.421 pairs of input audio signals of LCP coefficients calculation block calculate the LPC coefficient by equitable subsection.Equivalence modular converter 422 becomes the LCP coefficients conversion LSF parameter of equivalence.Further, general parameter quantification module 430 is for different sound signal types, and the vector quantization code book with this sound signal type matching of using local storage carries out vector quantization to the LSF parameter.At last, the coding parameter of vector quantization is transmitted to multiplexer and is multiplexed in total audio coding frame.
The present invention is based on the LPC coefficient quantization method and apparatus of audio signal classification, be applied to the odd encoder core encryption algorithm that at least one coding core uses linear prediction LPC that the coding audio signal of at least two types is processed, can further improve the quantified precision of the Prediction Parameters of inner linear predictive coding module, thereby improve the efficient of overall digital audio coding algorithm and the subjective sound quality of scrambler.
Take AMR-WB+ multinuclear encryption algorithm as example, use the many vector quantization code books based on audio classification of the present invention, LPC coefficient wherein (the perhaps LSF parameter of conversion) is carried out vector quantization, wherein the concrete vector quantization scheme employing applying date is on July 17th, 2012, application number is 201210246780.9, and name is called the disclosed multilevel vector quantization method of Chinese patent application of " method and system that is used for voice signal LPC coefficient is carried out multi-stage vector quantization ".
The characteristic of following match stop and non-categorical situation:
(a) complexity
Due to the processing that has existed in AMR-WB+ audio signal classification, carry out respectively ACELP(voice class signal) and TCX(mixing class signal) coding, therefore the present invention mainly can increase the storage space (approximately 98k byte) of 1 times, and other complexities are suitable.
(b) performance
Adopt the multilevel vector quantization method of 201210246780.9 " being used for voice signal LPC coefficient is carried out the method and system of multi-stage vector quantization ", a plurality of vector quantization code books of match stop situation and the single vector quantization code book of non-categorical situation, respectively 12 MPEG typical case cycle testss are carried out precision comparative result after the LPC parameter vector quantizes, as table 1 to as shown in table 12.
From 12 tables, observe the averaging spectrum distortion that represents LPC parameter quantification precision, can think: sorting algorithm unanimously is better than the non-categorical algorithm, and this indicates that also the many vector quantization code books algorithm based on audio classification can further improve code efficiency.
Table 1 algorithm performance (cycle tests: es01)
| Algorithm | Averaging spectrum distortion (dB) | 2～4dB ratio (%) | ＞4dB ratio (%) |
| Sorting algorithm not | 0.697600 | 0.000000 | 0.000000 |
| Sorting algorithm | 0.502818 | 0.000000 | 0.000000 |
Table 2 algorithm performance (cycle tests: es02)
| Algorithm | Averaging spectrum distortion (dB) | 2～4dB ratio (%) | ＞4dB ratio (%) |
| Sorting algorithm not | 0.662532 | 0.000000 | 0.000000 |
| Sorting algorithm | 0.506807 | 0.000000 | 0.000000 |
Table 3 algorithm performance (cycle tests: es03)
| Algorithm | Averaging spectrum distortion (dB) | 2～4dB ratio (%) | ＞4dB ratio (%) |
| Sorting algorithm not | 0.662490 | 0.000000 | 0.000000 |
| Sorting algorithm | 0.597712 | 0.000000 | 0.000000 |
Table 4 algorithm performance (cycle tests: sc01)
| Algorithm | Averaging spectrum distortion (dB) | 2～4dB ratio (%) | ＞4dB ratio (%) |
| Sorting algorithm not | 0.679964 | 0.000000 | 0.000000 |
| Sorting algorithm | 0.568026 | 0.000000 | 0.000000 |
Table 5 algorithm performance (cycle tests: sc02)
| Algorithm | Averaging spectrum distortion (dB) | 2～4dB ratio (%) | ＞4dB ratio (%) |
| Sorting algorithm not | 0.624548 | 0.000000 | 0.000000 |
| Sorting algorithm | 0.600093 | 0.000000 | 0.000000 |
Table 6 algorithm performance (cycle tests: sc03)
| Algorithm | Averaging spectrum distortion (dB) | 2～4dB ratio (%) | ＞4dB ratio (%) |
| Sorting algorithm not | 0.620681 | 0.000000 | 0.000000 |
| Sorting algorithm | 0.483082 | 0.000000 | 0.000000 |
Table 7 algorithm performance (cycle tests: si01)
| Algorithm | Averaging spectrum distortion (dB) | 2～4dB ratio (%) | ＞4dB ratio (%) |
| Sorting algorithm not | 0.657625 | 0.000000 | 0.000000 |
| Sorting algorithm | 0.530154 | 0.000000 | 0.000000 |
Table 8 algorithm performance (cycle tests: si02)
| Algorithm | Averaging spectrum distortion (dB) | 2～4dB ratio (%) | ＞4dB ratio (%) |
| Sorting algorithm not | 0.735683 | 0.000000 | 0.000000 |
| Sorting algorithm | 0.701430 | 0.000000 | 0.000000 |
Table 9 algorithm performance (cycle tests: si03)
| Algorithm | Averaging spectrum distortion (dB) | 2～4dB ratio (%) | ＞4dB ratio (%) |
| Sorting algorithm not | 0.612262 | 0.000000 | 0.000000 |
| Sorting algorithm | 0.366940 | 0.000000 | 0.000000 |
Table 10 algorithm performance (cycle tests: sm01)
| Algorithm | Averaging spectrum distortion (dB) | 2～4dB ratio (%) | ＞4dB ratio (%) |
| Sorting algorithm not | 0.731752 | 0.000000 | 0.000000 |
| Sorting algorithm | 0.475733 | 0.000000 | 0.000000 |
Table 11 algorithm performance (cycle tests: sm02)
| Algorithm | Averaging spectrum distortion (dB) | 2～4dB ratio (%) | ＞4dB ratio (%) |
| Sorting algorithm not | 1.051757 | 0.423729 | 0.000000 |
| Sorting algorithm | 0.800497 | 0.847458 | 0.000000 |
Table 12 algorithm performance (cycle tests: sm03)
| Algorithm | Averaging spectrum distortion (dB) | 2～4dB ratio (%) | ＞4dB ratio (%) |
| Sorting algorithm not | 0.643514 | 0.423729 | 0.000000 |
| Sorting algorithm | 0.626824 | 0.847458 | 0.000000 |
The above is only preferred embodiment of the present invention, not in order to limiting the present invention, all any modifications of doing within the spirit and principles in the present invention, is equal to and replaces and improvement etc., within all should being included in protection scope of the present invention.
Claims (12)
1. a LPC coefficient quantization method that is used for audio-frequency signal coding, is characterized in that, comprises the steps:
S1, determine the type of input audio signal based on predetermined rule;
S2, input audio signal is carried out linear prediction process, calculate the LPC coefficient;
S3, for different sound signal types, the vector quantization code book of using with this sound signal type matching carries out vector quantization to described LPC coefficient.
2. method according to claim 1, is characterized in that,
Described step S2 further comprises:
It is the LSF parameter of equivalence with described LPC coefficients conversion;
Described step S3 further comprises:
For different sound signal types, the vector quantization code book of application and this sound signal type matching carries out vector quantization to described LSF parameter.
3. method according to claim 2, is characterized in that, in described step S3, vector quantization adopts multilevel vector quantization method.
4. method according to claim 1, is characterized in that, described method also comprised before step S1:
The required vector quantization code book of coding that will build for the signal model of different audio signals type is stored in this locality.
5. method according to claim 1, is characterized in that, described method further comprises:
S4, send the coding parameter of vector quantization to multiplexer and be multiplexed in total audio coding frame.
6. a LPC coefficient quantization device that is used for audio-frequency signal coding, is characterized in that, comprising:
The audio types determination module is for determine the type of input audio signal based on predetermined rule;
The linear prediction processing module is used for that input audio signal is carried out linear prediction and processes, and calculates the LPC coefficient;
Spectrum parameter quantification module is used for for different sound signal types, and the vector quantization code book of application and this sound signal type matching carries out vector quantization to described LPC coefficient.
7. device according to claim 6, is characterized in that, described linear prediction processing module further comprises:
The LPC coefficients calculation block is used for input audio signal and carries out the linear prediction processing, calculates the LPC coefficient;
The equivalence modular converter, being used for described LPC coefficients conversion is the LSF parameter of equivalence.
8. device according to claim 7, is characterized in that, described spectrum parameter quantification module is further used for for different sound signal types, and the vector quantization code book of application and this sound signal type matching carries out vector quantization to described LSF parameter.
9. device according to claim 6, is characterized in that, described device also comprises:
Memory module is used for storage for the required vector quantization code book of coding of the signal model structure of different audio signals type.
10. device according to claim 6, is characterized in that, described device sends the coding parameter of vector quantization to multiplexer and is multiplexed in total audio coding frame.
11. an odd encoder core audio coding method is characterized in that, comprises the steps:
A, the type of input audio signal is analyzed;
B, for a plurality of sound signal types, adopt corresponding a plurality of coding core to encode, wherein, at least one coding is checked the sound signal of at least two sound signal types and is carried out linear predictive coding;
It is characterized in that,
Described linear predictive coding quantizes the LPC coefficient by the described method of any one according to claim 1-5.
12. an odd encoder core audio coding equipment is characterized in that, comprising:
The audio signal classification processing module is used for the type of input audio signal is analyzed;
A plurality of coding cores are used for based on the sound signal type corresponding coding audio signal, and wherein, at least one coding is checked the sound signal of at least two sound signal types and carried out linear predictive coding;
It is characterized in that,
Described linear predictive coding quantizes the LPC coefficient by the described device of any one according to claim 6-10.
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| CN111105807A (en) * | 2014-01-15 | 2020-05-05 | 三星电子株式会社 | Weight function determination apparatus and method for quantizing linear predictive coding coefficients |
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| CN111105807A (en) * | 2014-01-15 | 2020-05-05 | 三星电子株式会社 | Weight function determination apparatus and method for quantizing linear predictive coding coefficients |
| CN111105807B (en) * | 2014-01-15 | 2023-09-15 | 三星电子株式会社 | Weighting function determining apparatus and method for quantizing linear predictive coding coefficient |
| CN110444219A (en) * | 2014-07-28 | 2019-11-12 | 弗劳恩霍夫应用研究促进协会 | The apparatus and method of the first coding algorithm of selection or the second coding algorithm |
| CN110444219B (en) * | 2014-07-28 | 2023-06-13 | 弗劳恩霍夫应用研究促进协会 | Apparatus and method for selecting a first encoding algorithm or a second encoding algorithm |
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