CN108109629A - A kind of more description voice decoding methods and system based on linear predictive residual classification quantitative - Google Patents

A kind of more description voice decoding methods and system based on linear predictive residual classification quantitative Download PDF

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Publication number
CN108109629A
CN108109629A CN201611033175.8A CN201611033175A CN108109629A CN 108109629 A CN108109629 A CN 108109629A CN 201611033175 A CN201611033175 A CN 201611033175A CN 108109629 A CN108109629 A CN 108109629A
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parameter
bit
coding
term prediction
coded
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林志斌
邱小军
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Nanjing University
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Nanjing University
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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
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
    • G10L19/07—Line spectrum pair [LSP] vocoders
    • 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
    • G10L19/0204—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 using subband decomposition
    • 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
    • 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
    • G10L19/038—Vector quantisation, e.g. TwinVQ audio

Abstract

A kind of more description voice codings, coding/decoding method and the system based on linear predictive residual classification quantitative, voice flow to be encoded is obtained into short-term prediction parameter, long-term prediction parameter and temporal prediction residue signal by short-term prediction analysis, long-term prediction analysis, and residual signals are divided into multiple multidimensional coding vectors, it calculates each coded vector energy envelope and carries out bit distribution, and Two-way Cycle vector quantization and coding are carried out to the time-domain coefficients of each coded vector；Short-term prediction parameter, long-term prediction parameter are interweaved and describe grouping packing more, each coded-bit of the vector quantization value of prediction residual quantification index and each quantization encoding signal carries out multiplexing packing.

Description

A kind of more description voice decoding methods based on linear predictive residual classification quantitative and System
Technical field
The present invention relates to encoding and decoding speech and transmission fields more particularly to a kind of based on linear predictive residual classification quantitative More description voice codings, coding/decoding method and system.
Background technology
In many application scenarios, voice communication all has higher requirements to real-time and continuity, but present network The voice data for reaching receiving terminal is usually caused to lose due to network delay, network congestion and error of transmission etc., causes to talk about Sound quality degradation；On the other hand, in order to obtain preferably communication voice quality, and require the voice signal of compression that should have Higher clarity.Therefore it is directed to for VoIP, encoded voice quality problems and the loss of data of insecure packet network Compensation is the community of contradiction.And the method for traditional processing packet loss is to retransmit, but when packet loss rate is higher, weight The environment of more congestion can be caused by passing, and cannot meet the requirement of real-time.Different from re-transmission, multiple description coded (MDC) can To significantly improve the stability of transmission, and apparent time delay is not introduced, be a kind of effective method for solving packet loss.It is early More description waveform speech coder algorithms that phase proposes are simple, can improve the transmission stability of system, but this kind of volume well The compression ratio of code device is not high, affects encoded voice quality.The subsequently linear prediction based on code excited (CELP) voice multi-description Encoder, these encoders have sufficiently high compression efficiency, but have very strong dependence between their parameter, and description is decomposed Method underaction, the raising of stability are to be greatly reduced with performance as cost, simultaneously as the use of CELP codings is adaptive Code book and random code book codebook excitation mode is answered to carry out voice coding, coding quality reaches one on certain middle low bit- rate The limit can not further promote the encoding and decoding speech quality under the conditions of normal non-frame losing.The present invention provides one kind based on linear The multiple description coded device of prediction residual classification quantitative, voice quality when can improve normally without packet loss, while again have compared with The codec of good anti-grouping loss recovery ability.
The content of the invention
The technical problems to be solved by the invention are to overcome the deficiencies of the prior art and provide one kind based on linear prediction residual More description voice codings, coding/decoding method and the system of difference class quantization.
In order to solve the above technical problem, the present invention provides a kind of more descriptions based on linear predictive residual classification quantitative Sound coding method, this method include：
Voice flow to be encoded is subjected to signal framing, mute detection, high-pass filtering pretreatment, and using asymmetric window from Correlation method calculates the short-term prediction coefficient of voice signal and is converted into Line Spectral Pair coefficients (LSP), and subframe is obtained using interpolation method LSP coefficients；
Long-term prediction parameter (LTP) is calculated using pitch evaluation mode, pitch evaluation estimates mode using open-loop pitch, two A same fundamental tone of predicting subframes, a frame are made of two fundamental tones, and LTP coefficient is chosen for 3 ranks；
By short-term prediction coefficient and long-term prediction coefficient, linear prediction residual difference signal is obtained；
Further, the time-domain coefficients of linear prediction residual difference signal to be encoded are divided into L coding subband, calculate each volume The amplitude envelope quantification index of numeral band；
Further, according to coding bit rate, determine envelope energy quantization step, quantify sub-belt energy envelope, according to can For coded-bit and its sub-belt energy envelope, each coding sub-band coding bit is determined, each coded-bit corresponds to a coding Quantization step；
Further, according to the distribution relation of sub-belt energy envelope, multiple coding sub-band coding bit (i.e. subband amounts are determined Change step-length) candidate code bit code book relation, the coding of 1 coding subband is only existed between multiple coded-bit candidate codebooks Bit correction；
Further, subband data is quantified using vector quantization mode according to coded-bit, and quantifies initially coding first Bit carries out subband data coding, and difference of the sub-band coding result by local synthesis and original input signal is filtered by perceptual weighting Ripple device obtains quantifies perceptual weighting error for the first time；Multiple and different coding sub-band coding bits are calculated using the same manner and are obtained Corresponding perceptual weighting error；The quantization bit method of salary distribution of weighted error minimum is obtained, is passed as final bit quantization mode Defeated decoding end；
Further, it is LSP that n-th frame, which encodes parameter to be transmitted,n, subframe_paran[1 ..., M], subframe_ Para is subframe parameter, includes gain, pitch delay and its quantization parameter.And it is packaged as LSP by code stream is quantifiedn, subframe_ paran[odd], LSPn-1, subframe_paran-1[even], redundancy_paran, wherein n-th frame signal includes current Frame frame parameter and its odd numbered sub-frames parameter and former frame frame parameter and its even subframe parameter, while include n-th frame nuisance parameter.
By intertexture Prediction Parameters signal and the coding coded-bit of amplitude envelope of subband and the coded-bit of time-domain coefficients After multiplexing is packaged, decoding end is sent to.
Description of the drawings
To describe the technical solutions in the embodiments of the present invention more clearly, make required in being described below to embodiment Attached drawing is briefly described, it should be apparent that, the accompanying drawings in the following description is only some embodiments of the present invention, for For those of ordinary skill in the art, without creative efforts, other are can also be obtained according to these attached drawings Attached drawing.
Fig. 1 is a kind of more description voice coding schematic diagrams based on linear predictive residual classification quantitative of the present invention；
The time domain that Fig. 2 is the present invention quantifies block diagram；
Fig. 3 is the subframe parameter intertexture description schematic diagram of the present invention.
Specific embodiment
Below in conjunction with the attached drawing in the embodiment of the present invention, the technical solution in the embodiment of the present invention is described, is shown So, described embodiment is only part of the embodiment of the present invention, instead of all the embodiments.Based on the reality in the present invention Apply example, those of ordinary skill in the art's all other embodiments obtained without making creative work all belong to In the scope of protection of the invention.
The main idea of the present invention is invention provides a kind of more description voice coders based on linear predictive residual classification quantitative Code method, detailed process are as follows：
Voice flow to be encoded is carried out signal framing by step 1, is carried out mute detection using short-time average zero-crossing rate, is performed 101, and output mute detection identifier flag to 111；
Step 2, Signal Pretreatment use the high-pass filter structure of 140Hz, perform 102；
Step 3, the short-term prediction coefficient that voice signal is calculated using the correlation method of asymmetric window, are performed 103, performed 104, Line Spectral Pair coefficients (LSP) are converted into, and subframe LSP coefficients are obtained using interpolation method；Short-time analysis filtering is obtained simultaneously Device 105；
Step 4 calculates long-term prediction parameter (LTP) using pitch evaluation mode, performs 106, pitch evaluation uses open loop Pitch evaluation mode, two same fundamental tones of predicting subframes, a frame are made of two fundamental tones, and LTP coefficient is chosen in the present embodiment For 3 ranks；
Step 5, by short-term prediction coefficient and long-term prediction coefficient, construction in short-term composite filter 107 and it is long when synthesize Wave filter 108 obtains linear prediction residual difference signal according to 110 module of error minimum principle, performs 109；Step 6,109 modules will The time-domain coefficients of linear prediction residual difference signal to be encoded are divided into L coding subband, in the present embodiment, L values 20；Calculate each volume The amplitude envelope quantification index of numeral band：
A) residual signals pre-process：Echo signal is divided into 20 8 n dimensional vector ns, calculates 8 n dimensional vector n subband envelope energy respectively Amount；According to coding bit rate, envelope energy quantization step is determined, quantify sub-belt energy envelope；
B) according to for coded-bit and its sub-belt energy envelope, primarily determine that each coding sub-band coding bit is (i.e. sub Band quantization step), in the present embodiment, 20 sub-band coding bits of original of some energy are respectively [2 34145621 0233214423 3] etc., each coded-bit corresponds to a coded quantization step-length；
C) 201 are performed, whether calculation code subband bit consumption and quantization error meet pre-provisioning request；Meet predetermined want It asks, then performs 207, otherwise time domain residual classification quantitative end-of-encode, performs 202；
D) whole envelope quantization step is adjusted；And it is transferred to 203；
E) 203 actual coding bit and calculation code consumption bit are calculated, and are transferred to 204,；
F) whether 204 calculation code bits meet actual coding code check requirement, if meeting the requirements, perform 207, coding Band coded quantization terminates；Otherwise, 205 are performed, and according to the distribution relation of sub-belt energy envelope, determines multiple coding sub-band codings Bit (i.e. quantized subband step-length) candidate code bit code book relation only exists 1 between multiple coded-bit candidate codebooks The coded-bit amendment of subband is encoded, is existed such as in the present embodiment：
Primarily determine that coded-bit is：[2 3 4 1 4 5 5 2 1 0 2 3 3 2 1 4 4 2 3 3]；
Then basic extended coding sub-band coding bit herein：[2 3 4 1 4 5 6 2 1 0 2 3 3 2 1 4 4 2 3 3]；
Remaining coding sub-band coding bit equally extends；If coded vector step-length can be adjusted, into 206；
Otherwise 202 are returned；
G) 206 modules adjustment coded vector quantization step, and perform 203；Until quantizing process terminates, 207 modules are performed；
H) subband data is quantified using lattice vector quantization mode according to coded-bit, and quantifies initial coded-bit first Subband data coding is carried out, sub-band coding result passes through LTP/LPC, and locally the difference of synthesis and original input signal adds by perceiving It weighs wave filter and obtains and quantify perceptual weighting error for the first time；Multiple and different coding sub-band coding bits are calculated simultaneously using the same manner Obtain corresponding perceptual weighting error；The quantization bit method of salary distribution of weighted error minimum is obtained, as final bit quantization side Formula transmits decoding end.
It is LSP that step 7, n-th frame, which encode parameter to be transmitted,n, subframe_paran[1 ..., M], subframe_para For subframe parameter, gain, pitch delay and its quantization parameter are included.And it is packaged as LSP by code stream is quantifiedn, subframe_paran [odd], LSPn-1, subframe_paran-1[even], redundancy_paran, wherein n-th frame signal include present frame frame Parameter and its odd numbered sub-frames parameter and former frame frame parameter and its even subframe parameter, while comprising n-th frame nuisance parameter, retouch more Code stream is stated to be packaged as shown in Figure 3.
By intertexture Prediction Parameters signal and the coding coded-bit of amplitude envelope of subband and the coded-bit of time-domain coefficients After multiplexing is packaged, decoding end is sent to.

Claims (3)

1. a kind of more description voice codings, coding/decoding method and system based on linear predictive residual classification quantitative, which is characterized in that This method includes：
Voice signal is obtained into linear prediction residual difference signal after short-term prediction analysis and long-term prediction analysis, according to energy point For different coding subbands, and amplitude envelope values of these coding subbands are quantified and encoded.
Short-term prediction parameter and long-term prediction parameter are described to be packaged more using odd-even mode.
The coded-bit of the coded-bit and time-domain coefficients of intertexture Prediction Parameters signal and the amplitude envelope of coding subband is multiplexed After packing, decoding end is sent to.
2. the method as described in claim 1, which is characterized in that
The classification quantitative of linear predictive residual coefficient determines multiple coding sub-band codings according to the distribution relation of sub-belt energy envelope Bit (i.e. quantized subband step-length) candidate code bit code book relation only exists 1 between multiple coded-bit candidate codebooks Encode the coded-bit amendment of subband.
3. the method as described in claim 1, which is characterized in that
The more describing modes of the odd-even of short-term prediction parameter and long-term prediction parameter are that wherein n-th frame signal includes present frame Frame parameter and its odd numbered sub-frames parameter and former frame frame parameter and its even subframe parameter, while include n-th frame nuisance parameter.
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