CN102074242A - Extraction system and method of core layer residual in speech audio hybrid scalable coding - Google Patents
Extraction system and method of core layer residual in speech audio hybrid scalable coding Download PDFInfo
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Abstract
The invention relates to the technical field of audio coding, in particular to an extraction system and method of a core layer residual in speech audio hybrid scalable coding. The system comprises a pre-processing module (1), a speech coding module (2), an audio coding module (3), a mode selection module (4), a speech decoding synthesis module (5), an audio decoding synthesis module (6) and a residual generation module (7). The method for obtaining the accurate residual between the scalable core layer and original signals in a speed audio hybrid coding mode is the main content of the invention. Synthetic signals generated by a speed/audio coder at a coding end are used for selecting the optimal coding mode, and the synthetic signals consistent to the coding end are obtained by coding parameters output by the speed/audio coder, thus the accurate residual is obtained to be used for a scalable enhanced layer. The extraction system and method overcome the defect that the scalable core layer residual signals can not be accurately extracted in the traditional speed audio hybrid coding mode.
Description
Technical field
The present invention relates to technical field of audio, relate in particular to core layer residual error extraction system and method in a kind of speech audio mixing-classifying coding.
Background technology
In the scalable enhancement layer coding method, coding side receives the output parameter of core layer simultaneously by calculating the residual error of original signal and core layer composite signal, and residual error territory signal is done hierarchical coding.Decoding end decodes residual error territory signal, with the core layer signal addition, and the composite signal that is restored.The residual error territory signal of each enhancement layer is added on the core layer signal step by step, thereby improves reconstruction quality gradually.The tonequality gain that the accurate extraction of residual error territory signal and enhancement layer coding can provide has direct relation, the residual signals that the residual signals that the decoding end decoding obtains calculates near coding side more, and then composite signal is more near original signal, and decoding tonequality is high more.
Because the technology and the method for voice coding and audio coding there are differences, the hybrid coder of existing speech audio adopts different patterns respectively voice/audio to be encoded.Which kind of pattern is system can adopt according to the type selecting of current voice signal, perhaps adopts two kinds of patterns to encode respectively, and the composite signal that generates according to coding is selected the effective final coding mode of conduct of coding then.Because the generation of this composite signal just is used for model selection, be not equal to all operations that decoding end generates composite signal, add the overlapping of two kinds of coding modes, the composite signal that tends to cause the coding and decoding end to obtain is inconsistent, thereby the residual signals of the original signal that obtains of coding side and composite signal is not a residual signals accurately.For the accurate extraction of core layer coded residual under the speech audio hybrid coding pattern, become the key that improves speech audio hybrid coder hierarchical coding quality.
Summary of the invention
At the technical matters of above-mentioned existence, the purpose of this invention is to provide core layer residual error extraction system and method in a kind of speech audio mixing-classifying coding, to solve the demand of speech audio mixing-classifying coding.
For achieving the above object, the present invention adopts following technical scheme:
Core layer residual error extraction system in a kind of speech audio mixing-classifying coding framework comprises:
Pretreatment module: carry out pre-service from the monophony of audio input device or the input signal of multichannel, obtain the sound signal of present frame, and the sound signal of the present frame that obtained done three tunnel outputs: the one tunnel exports to the voice coding module, one the tunnel exports to the audio coding module, and one the tunnel exports to the residual error generation module;
Voice coding module: adopt the coding audio signal of general speech coding algorithm to the present frame of input, do two-way output behind the coding, wherein one the tunnel be output as the synthetic code stream of tone decoding, be used for mode adjudging, another road is output as all kinds of coding parameters that coding extracts, and passes to the tone decoding synthesis module;
Audio coding module: adopt the coding audio signal of General Audio encryption algorithm to the present frame of input, do two-way output behind the coding, wherein one the tunnel be output as the synthetic code stream of audio decoder, be used for mode adjudging, another road is output as all kinds of coding parameters that coding extracts, and passes to the audio decoder synthesis module;
Pattern is chosen module: choose optimum coding mode, the pattern bit-identify of obtaining divides two-way output, and one the tunnel enters the tone decoding synthesis module, and one the tunnel enters the audio decoder synthesis module;
The tone decoding synthesis module: if pattern is chosen the output of module is the voice coding pattern, then enters the tone decoding synthesis module, and all kinds of coding parameters that utilize the output of voice coding module are exported final core layer tone decoding composite signal as input;
The audio decoder synthesis module: if pattern is chosen the output of module is the audio coding pattern, then enters the audio decoder synthesis module, and all kinds of coding parameters that utilize the output of audio coding module are exported final core layer audio decoder composite signal as input;
The residual error generation module: composite signal is the two-way input of this module after the decoding of the sound signal of the pretreated present frame of pretreatment module output and the output of voice/audio decoding synthesis module, according to pattern position difference, select tone decoding composite signal or audio decoder composite signal, calculate residual signals, this module is output as residual signals.
Core layer residual error extracting method in a kind of speech audio mixing-classifying coding comprises:
1. import the voice/audio signal and at first pass through pre-service, input signal can be monophony or multi-channel signal, and pre-service can comprise high-pass filtering, divide processing such as frame, pre-emphasis, obtains pretreated signal s (n);
2. by the 1. pretreated sound signal of gained, carry out after the voice coding one the tunnel and be output as the synthetic code stream x of voice coding
1(n), another road is output as speech coding parameters;
3. by the 1. pretreated sound signal of gained, carry out behind the audio coding one the tunnel and be output as the synthetic code stream x of audio coding
2(n), another road is output as audio coding parameters;
2. and the synthetic code stream x that 3. obtains 4. by
1(n) and x
2(n), carry out pattern and choose, select optimum coded system, the output mode bit-identify;
5. by the 4. pattern position of gained, if what select is the voice coding pattern, then change step over to 6., it is synthetic to carry out tone decoding; If what select is the audio coding pattern, then change step over to 7., it is synthetic to carry out audio decoder;
6. the coding parameter of 2. being exported by step carries out the synthetic tone decoding composite signal that obtains of tone decoding
7. the coding parameter of 3. being exported by step carries out the synthetic audio decoder composite signal that obtains of audio decoder
6. or the decoded composite signal that 7. obtains 8. by 1. obtaining pretreated signal s (n) and by
Or
Obtain residual signals r (n).
6. described step further comprises following substep:
Decoding LP filter parameter, by the synthetic ISP vector that has quantized of the ISP quantization index that receives, the ISP vector after the interpolation is switched to LP filter coefficient territory, is used for the composite filter reconstructed speech;
Self-adaption of decoding codebook vectors and fixed codebook vector and both gains, synthetic speech;
Carry out the aftertreatment that white noise character strengthens and fundamental tone strengthens;
Obtain final synthetic audio signal and upgrade public buffer memory.
7. described step further comprises following substep:
Read the sampling frequency sample value, carry out anti-vector quantization based on splitting table;
Gain balance, the influence of removing the different zoom factor;
The contrary shaping of peak value;
Inverse time conversion frequently, to time domain, the time-domain signal and the global gain that obtain multiply each other signal by frequency domain transform;
Overlap-add in windowing and the TVC;
Obtain synthetic audio signal by contrary perceptual weighting filter;
If what former frame adopted is the ACELP pattern-coding, so that present frame is initial crossover part and last subframe of previous frame ACELP composite signal are done the windowing crossover, obtain final synthetic audio signal and upgrade public buffer memory.
The present invention has the following advantages and good effect:
The present invention has overcome the problem that gradable core layer residual signals can't accurately extract under the existing voice audio mix coding mode.
Description of drawings
Fig. 1 is the framework synoptic diagram of core layer residual error extraction system in the speech audio mixing-classifying coding provided by the invention.
Fig. 2 is the process flow diagram of core layer residual error extracting method in the speech audio mixing-classifying coding provided by the invention.
The 1-pretreatment module, 2-voice coding module, 3-audio coding module, the 4-pattern is chosen module, 5-tone decoding synthesis module, 6-audio decoder synthesis module, 7-residual error generation module, 8-audio frequency original signal, signal after the 9-pre-service, 10-voice coding composite signal, 11-audio coding composite signal, 12-coding mode flag, 13-tone decoding composite signal, 14-audio decoder composite signal, 15-speech coding parameters, the 16-audio coding parameters, the 17-residual signals.
Embodiment
The invention will be further described in conjunction with the accompanying drawings with specific embodiment below:
Core layer residual error extraction system is divided into two parts in the speech audio mixing-classifying coding provided by the invention, a part is the core encoder layer, another part is the residual error enhancement layer, wherein core layer adopts general voice/audio mixed coding technology, the residual error enhancement layer can be divided into a plurality of enhancement layers, as shown in Figure 1, comprising:
Pretreatment module 1: input signal is carried out pre-service, it is input as the audio frequency original signal, is output as pretreated signal, and pretreated signal is done three tunnel outputs: the one tunnel exports to the voice coding module, one the tunnel exports to the audio coding module, and one the tunnel exports to the residual error generation module;
Voice coding module 2: this module is the core layer coding module in the graduated encoding framework, adopt general speech coding algorithm that the input signal of pretreatment module is encoded, do two-way output behind the coding, wherein one the tunnel be output as the synthetic code stream of voice coding, be used for mode adjudging, another road is output as all kinds of coding parameters that coding extracts, and passes to the tone decoding synthesis module, is used to calculate composite signal accurately;
Audio coding module 3: this module is the core layer coding module in the graduated encoding framework, adopt the General Audio encryption algorithm that the input signal of pretreatment module is encoded, do two-way output behind the coding, wherein one the tunnel be output as the synthetic code stream of audio coding, be used for mode adjudging, another road is output as all kinds of coding parameters that coding extracts, and passes to the audio decoder synthesis module, is used to calculate composite signal accurately;
Pattern is chosen module 4: for the two-way output of voice coding module and audio coding module, choose optimum pattern as scrambler, be output as the pattern bit-identify.If selected the voice coding pattern, then enter the tone decoding synthesis module, if selected the audio coding pattern, then enter the audio decoder synthesis module;
Tone decoding synthesis module 5: if pattern is chosen the output of module is the voice coding pattern, then enters the tone decoding synthesis module, and all kinds of coding parameters that utilize the output of voice coding module are exported final core layer voice coding composite signal as input;
Audio decoder synthesis module 6: if pattern is chosen the output of module is the audio coding pattern, then enters the audio decoder synthesis module, and all kinds of coding parameters that utilize the output of audio coding module are exported final core layer audio coding composite signal as input;
Residual error generation module 7: composite signal is the two-way input of this module after the decoding of the pretreated signal of pretreatment module output and the output of voice/audio decoding synthesis module, obtains residual signals, and this module is output as residual signals.
Core layer residual error extracting method may further comprise the steps in the speech audio mixing-classifying coding provided by the invention:
Step 1: sampling rate is carried out pre-service for the 16kHz input signal, and pre-service specifically comprises high-pass filtering and two processes of perceptual weighting, and output signal is s (n);
Input signal is sent into Hi-pass filter, the low frequency signal that filtering 50Hz is following;
Signal after the high-pass filtering is sent into perceptual weighting filter W
LB(z), while γ '
1, γ '
2And γ '
3(0＜γ '
1, γ '
2, γ '
3＜1) three also corresponding adjustment of coefficient are to relax quantization noise spectrum:
γ ' wherein
1, γ '
2, γ '
3For adjusting parameter, a
iBe the linear prediction analysis coefficient, i is the exponent number of linear prediction,
Step 2: to carry out the ACELP encoder encodes of 12kbps pattern through the signal after pretreated, the back one tunnel of encoding is output as the synthetic code stream x of voice coding
1(n), another road is output as speech coding parameters, comprises that ISF index value, VQ gain index value, code book index value, filtering index value, gene postpone index value;
Step 3: to carry out TVC scrambler (transform domain audio coder) coding of 12kbps pattern through the signal after pretreated, the back one tunnel of encoding is output as the synthetic code stream x of audio coding
2(n), another road is output as audio coding parameters, comprises sampling frequency sample value, zoom factor, global gain;
Step 4: the synthetic code stream x that obtains by step 2 and step 3
1(n) and x
2(n), calculate perceptual weighting segmental signal-to-noise ratio SNR1 and SNR2 with the output signal s (n) of step 1 respectively, if SNR1＞SNR2, then pattern bit-identify mod puts 0, select the ACELP encoder encodes, otherwise mod puts 1, selects the TVC encoder encodes;
Step 5: if mod is 0, then change step 6 over to, carry out the tone decoding synthesis module; If mod is 1, then change step 7 over to, carry out the audio decoder synthesis module;
Step 6: by the coding parameter of step 2 output, by the synthetic composite signal of decoding that obtains of ACELP demoder
This step embodiment comprises following substep:
1.. decoding LP filter parameter, by the synthetic ISP vector that has quantized of the ISP quantization index that receives, the ISP vector after the interpolation is switched to LP filter coefficient territory, is used for the composite filter reconstructed speech;
2.. self-adaption of decoding codebook vectors and fixed codebook vector and both gains, synthetic speech;
3.. aftertreatment (white noise character strengthens and fundamental tone strengthens);
4.. obtain final synthetic audio signal and upgrade public buffer memory.
Step 7: by the coding parameter of step 3 output, by the synthetic audio decoder composite signal that obtains of TVC demoder
This step embodiment comprises following substep:
1.. read the sampling frequency sample value, carry out anti-vector quantization based on splitting table;
2.. gain balance, the influence of removing the different zoom factor;
3.. the contrary shaping of peak value;
4.. inverse time conversion frequently, to time domain, the time-domain signal and the global gain that obtain multiply each other signal by frequency domain transform;
5.. the overlap-add in windowing and the TVC;
6.. obtain synthetic audio signal by contrary perceptual weighting filter;
7.. what if former frame adopted is the ACELP pattern-coding, and so that present frame is initial crossover part and last subframe of previous frame ACELP composite signal are done the windowing crossover, obtain final synthetic audio signal and upgrade public buffer memory.
Step 8: obtain pretreated signal s (n) by step 1 and deduct decoded composite signal, if adopt the voice coding pattern, then with
Subtract each other, as adopt the audio coding pattern then with
Subtract each other, obtain residual signals r (n).
Classification core layer speech audio of the present invention coding method is general encryption algorithm, the method of the accurate residual error of classification core layer and original signal is main contents of the present invention under the acquisition speech audio hybrid coding pattern, the composite signal that the voice/audio scrambler of coding side generates is used to carry out the selection of optimum code pattern, utilize the coding parameter of voice/audio scrambler output to obtain the composite signal consistent with decoding end, residual error is used for the classification enhancement layer thereby obtain accurately.
Above embodiment is only for the usefulness that the present invention is described, but not limitation of the present invention, person skilled in the relevant technique; under the situation that does not break away from the spirit and scope of the present invention; can also make various conversion or modification, so all technical schemes that are equal to, all fall into protection scope of the present invention.
Claims (4)
1. core layer residual error extraction system in the speech audio mixing-classifying coding framework is characterized in that, comprising:
Pretreatment module (1): carry out pre-service from the monophony of audio input device or the input signal of multichannel (8), obtain the sound signal (9) of present frame, and the sound signal (9) of the present frame that obtained done three tunnel outputs: the one tunnel exports to voice coding module (2), one the tunnel exports to audio coding module (3), and one the tunnel exports to residual error generation module (7);
Voice coding module (2): adopt general speech coding algorithm that the sound signal (9) of the present frame of input is encoded, do two-way output behind the coding, wherein one the tunnel be output as the synthetic code stream (10) of tone decoding, be used for mode adjudging, another road is output as all kinds of coding parameters (15) that coding extracts, and passes to tone decoding synthesis module (5);
Audio coding module (3): adopt the General Audio encryption algorithm that the sound signal (9) of the present frame of input is encoded, do two-way output behind the coding, wherein one the tunnel be output as the synthetic code stream (11) of audio decoder, be used for mode adjudging, another road is output as all kinds of coding parameters (16) that coding extracts, and passes to audio decoder synthesis module (6);
Pattern is chosen module (4): choose optimum coding mode, obtain pattern bit-identify (12) and divide two-way output, the one tunnel enters tone decoding synthesis module (5), and one the tunnel enters audio decoder synthesis module (6);
Tone decoding synthesis module (5): if pattern is chosen the output (12) of module is the voice coding pattern, then enter tone decoding synthesis module (5), all kinds of coding parameters (15) that utilize the output of voice coding module are exported final core layer tone decoding composite signal (13) as input;
Audio decoder synthesis module (6): if pattern is chosen the output (12) of module is the audio coding pattern, then enter audio decoder synthesis module (6), all kinds of coding parameters (16) that utilize the output of audio coding module are exported final core layer audio decoder composite signal (14) as input;
Residual error generation module (7): composite signal is the two-way input of this module after the decoding of the sound signal (9) of the pretreated present frame of pretreatment module output and the output of voice/audio decoding synthesis module, according to pattern position difference, select tone decoding composite signal (13) or audio decoder composite signal (14), calculate residual signals (17), this module is output as residual signals (17).
2. core layer residual error extracting method during a speech audio mixing-classifying is encoded is characterized in that, may further comprise the steps:
1. import the voice/audio signal and at first pass through pre-service, input signal can be monophony or multi-channel signal, and pre-service can comprise high-pass filtering, divide processing such as frame, pre-emphasis, obtains pretreated signal s (n);
2. by the 1. pretreated sound signal of gained, carry out after the voice coding one the tunnel and be output as the synthetic code stream x of voice coding
1(n), another road is output as speech coding parameters;
3. by the 1. pretreated sound signal of gained, carry out behind the audio coding one the tunnel and be output as the synthetic code stream x of audio coding
2(n), another road is output as audio coding parameters;
2. and the synthetic code stream x that 3. obtains 4. by
1(n) and x
2(n), carry out pattern and choose, select optimum coded system, the output mode bit-identify;
5. by the 4. pattern position of gained, if what select is the voice coding pattern, then change step over to 6., it is synthetic to carry out tone decoding; If what select is the audio coding pattern, then change step over to 7., it is synthetic to carry out audio decoder;
6. the coding parameter of 2. being exported by step carries out the synthetic tone decoding composite signal that obtains of tone decoding
7. the coding parameter of 3. being exported by step carries out the synthetic audio decoder composite signal that obtains of audio decoder
6. or the decoded composite signal that 7. obtains 8. by 1. obtaining pretreated signal s (n) and by
Or
Obtain residual signals r (n).
3. core layer residual error extracting method in the speech audio mixing-classifying coding according to claim 2 is characterized in that:
6. described step further comprises following substep:
Decoding LP filter parameter, by the synthetic ISP vector that has quantized of the ISP quantization index that receives, the ISP vector after the interpolation is switched to LP filter coefficient territory, is used for the composite filter reconstructed speech;
Self-adaption of decoding codebook vectors and fixed codebook vector and both gains, synthetic speech;
Carry out the aftertreatment that white noise character strengthens and fundamental tone strengthens;
Obtain final synthetic audio signal and upgrade public buffer memory.
4. according to core layer residual error extracting method in claim 2 or the 3 described speech audio mixing-classifying codings, it is characterized in that:
7. described step further comprises following substep:
Read the sampling frequency sample value, carry out anti-vector quantization based on splitting table;
Gain balance, the influence of removing the different zoom factor;
The contrary shaping of peak value;
Inverse time conversion frequently, to time domain, the time-domain signal and the global gain that obtain multiply each other signal by frequency domain transform;
Overlap-add in windowing and the TVC;
Obtain synthetic audio signal by contrary perceptual weighting filter;
If what former frame adopted is the ACELP pattern-coding, so that present frame is initial crossover part and last subframe of previous frame ACELP composite signal are done the windowing crossover, obtain final synthetic audio signal and upgrade public buffer memory.
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