CN101281749A - Apparatus for encoding and decoding hierarchical voice and musical sound together - Google Patents
Apparatus for encoding and decoding hierarchical voice and musical sound together Download PDFInfo
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Abstract
A gradable combined voice and music CODEC device in the technical domain of digital audio encoding is provided. In the encoding unit in the present invention, a voice and music grading unit carries out gradation for audio signals, a graded voice encoding unit and a graded music residual encoding unit carry out encoding for graded voice signals, a graded music encoding unit and a graded voice residual encoding unit encoding unit carry out encoding for graded music signal, and provide output via a combined output module; the decoding unit comprises an encoding mode obtaining unit, an encoding parameter obtaining unit, a graded voice decoding unit, a graded music decoding unit, a combined output unit, and is designed to accomplish the process reverse to the process of the encoding unit. The CODEC device provided in the present invention is applicable to all voice and music signals, and can reduce data loss in the encoding process, improve encoding quality; especially, the CODEC device can maintain high acoustic quality at higher code rates.
Description
Technical field
The present invention relates to a kind of device of digital audio encoding technical field, specifically is a kind of gradable voice and musical sound combined coding device and decoding device.
Background technology
The rapid growth of digital communication service and stored digital business has proposed more and more higher requirement to Digital Signal Processing.Because the continuous expansion of multimedia service not only needs higher code efficiency and real-time, and is also increasing to the general adaptive demand of voice and note signal.Simultaneously, in mobile voice applications, because the instability of channel, need be according to the adaptive adjustment code check of channel situation, this just requires audio codec can satisfy the self-adaptation multi code Rate of Chinese character, thereby the graduated encoding technology of mobile audio frequency also becomes the gordian technique that audio digital signals of future generation is handled needs solution.In common audio coding, use the mode of time domain and frequency domain hybrid coding to solve the problem of encoding at voice and note signal different qualities.
Through the prior art literature search is found, third generation partner program (The 3rd GenerationPartnership Project, 3GPP) AMR-WB+ (the Adaptive MultiRate WideBandplus of tissue recommendation, AMR-WB) [the InternationalTelecommunication Union of audio coding decoding standard and International Telecommunications Union (ITU), ITU-T] the G.729.1 audio coding decoding standard of recommending, these two kinds of standards are main can satisfy low code check, low complex degree, gradable voice and musical sound combined coding technology.
AMR-WB+ is by taking two kinds of core encoder of ACELP (Algebraic Code Excited Linear Prediction) and TCX (transform domain excitation coding) to satisfy generally adaptive to voice and note signal, is a kind ofly to expand a kind of hybrid coding mode that forms by the audio coding to low code check.G.729.1 adopt the graduated encoding mode, 12 embedded layers have been comprised, every layer of all corresponding different code check and different coding techniquess, main core encoder technology have three kinds of CELP (Code Excited Linear Prediction), TDBE (time domain band spreading technique) and TDAC (time domain aliasing counteracting predictive transformation coding techniques).G.729.1 because the layering complexity, the also corresponding increase of computation complexity.In addition, owing to AMR-WB+ and G.729.1 all at low Bit Rate Audio Coding, the range of code rates that AMR-WB+ adapts to is: monophony 6～36kbps, and stereo 8～48kbps, G.729.1 the range of code rates of Shi Yinging is 8～32kbps.When code check is high to a certain degree the time, such as when code check reaches 64kbps, the effect of these two kinds of encoding schemes is not just obviously as the AAC under the code check (Advance Audio Coding, advanced audio volume).The advantage of these two kinds of coded systems just can not embodied under high code check.For this reason, be necessary to propose a kind of encoding scheme that not only is adapted to hang down code check but also can still keeps high tone quality along with the raising of code check.
In addition, ACELP that in AMR-WB+, proposes and TCX mixed coding technology, every frame signal is encoded with ACELP and two kinds of coded systems of TCX respectively, respectively behind local decode, choose the higher coded system coding of signal to noise ratio (S/N ratio), the encoder complexity height, and, in cataloged procedure, lost many quantity of information because every frame signal has finally only been used a kind of coding method coding.
Summary of the invention
The present invention is directed to above-mentioned the deficiencies in the prior art, a kind of gradable voice and musical sound combined coding device and decoding device have been proposed, make it utilize voice and musical sound detection technique, speech-like signal coding techniques, class note signal coding techniques, residual coding technology etc. under the prerequisite that guarantees high tone quality and low computation complexity, raising is satisfied the adaptivity to code check simultaneously to the general adaptability of voice signal and note signal.
The present invention is achieved by the following technical solutions:
The present invention relates to a kind of gradable voice and musical sound combined coding device, comprise: voice and musical sound taxon, class voice coding unit, local class tone decoding unit, class musical sound coding unit, local class musical sound decoding unit, residual error acquiring unit, packing output unit, wherein:
Voice and musical sound taxon are classified to the sound signal of input, according to classification results sound signal are imported class voice coding unit or class musical sound coding unit;
Being responsible for voice signal is encoded in class voice coding unit, obtains coding parameter, and export local class tone decoding unit, residual error acquiring unit and packing output unit to;
Local class tone decoding unit carries out local decode with the parameter that class voice coding cell encoding obtains, and obtain local reconstruction signal, and the output decoder data is to the residual error acquiring unit;
Class musical sound coding unit is responsible for note signal is encoded, and obtains coding parameter, and exports local class musical sound decoding unit, residual error acquiring unit and packing output unit to;
Local class musical sound decoding unit carries out local decode with the parameter that class musical sound coding unit coding obtains, and obtain local reconstruction signal, and the output decoder data is to the residual error acquiring unit;
The residual error acquiring unit is obtained the residual error of input audio signal and local reconstruction signal, if this frame signal is judged as speech-like signal, then residual error is input to class musical sound coding unit; If this frame signal is judged as the class note signal, then residual error is input to class voice coding unit;
The parameter packing output that the packing output unit obtains class voice coding unit and class musical sound coding unit coding.
Described class voice coding unit, every frame of voice signal of input is carried out linear prediction (LP) to be analyzed and obtains linear predictor coefficient, by the linear prediction residual difference signal is calculated echo signal by perceptual weighting filter, calculate the impulse response of weighted synthesis filter simultaneously, adopt echo signal and impulse response by searching for closed loop pitch cycle and gain near the cycle at open-loop pitch, echo signal is upgraded by the contribution of removing adaptive codebook, the new echo signal that obtains is used for fixing the code word searching analysis, calculate the gain of self-adaptation and fixed codebook vector, at last with the code vector address and the gain of random code resultant fixed code vector, and packing output behind the filter parameter quantization encoding.
Described class musical sound coding unit, the class note signal is passed through analysis filterbank, signal is divided into high-frequency band signals and low band signal two-way, when two paths of signals is carried out respectively-the frequency conversion, convert the signal into frequency domain, obtain the non-quantization spectral coefficient of two frequency bands, the non-quantization spectral coefficient of low-frequency band is carried out vector coding based on lattice vector quantization, the non-quantization spectral coefficient of high frequency band is carried out band spread based on the low-frequency band lattice vector quantization, at last with low-frequency band vector coding code stream and packing of high frequency band extended coding code stream and output.
Described residual error acquiring unit, when it carries out residual coding to speech-like signal, cataloged procedure is identical with class voice coding unit, the bit number of class voice residual coding distributes according to channel and code check situation to be decided, the bit number that the good code check height of channel then distributes is many, otherwise then the bit number of Fen Peiing is less, has gradability and to the adaptivity of channel and code check.
Described residual error acquiring unit, when it carries out residual coding to the class note signal, cataloged procedure is identical with class musical sound coding unit, the bit number of class musical sound residual coding distributes and will decide according to channel and code check situation, the bit number that the good code check height of channel then distributes is many, otherwise then the bit number of Fen Peiing is less, and it has the adaptivity to channel and code check.
Described class musical sound coding unit, its vector coding based on lattice vector quantization also comprises: scrambler is adjusted the quality that quantizes according to the Bit Allocation in Discrete of Rate Control quantized segment, and quantification is gradable.
Described class musical sound coding unit, its high frequency band expansion also comprises: scrambler is adjusted the quality of band spread according to the Bit Allocation in Discrete of Rate Control high frequency band expansion, and band spread is gradable.
The invention still further relates to a kind of gradable voice and musical sound combined decoding device, comprising: coding mode acquiring unit, coding parameter acquiring unit, class tone decoding unit, class musical sound decoding unit, synthetic output unit, wherein:
The coding mode acquiring unit is tentatively resolved code stream, and the coding mode that this frame signal adopted is inputed to the coding parameter acquiring unit;
The coding parameter acquiring unit is resolved code stream according to different coding modes, obtains each coding parameter and inputs to class tone decoding unit and class musical sound decoding unit;
Class tone decoding unit by using class speech coding parameters is decoded, and obtains class tone decoding data, and the output decoder data are to synthetic output unit;
Class musical sound decoding unit utilizes class musical sound coding parameter to decode, and obtains class musical sound decoded data, and the output decoder data are to synthetic output unit;
Synthetic output unit synthesizes the sound signal that obtains rebuilding with the decoded data of class tone decoding unit and the output of class musical sound decoding unit.
Described class tone decoding unit, it is by resolving code stream, corresponding code vector is found in address according to fixed code vector and self-adaptation code vector, fixed code vector and self-adaptation code vector are multiplied each other with its gain respectively, fixed code vector that obtains reducing and self-adaptation code vector, and with the reduction fixed code vector and the self-adaptation code vector respectively by the excitation composite filter, obtain synthetic speech, at last synthetic speech is exported.
Described class musical sound decoding unit, it is by resolving code stream, low-frequency band quantization spectral coefficient in the code stream is carried out the lattice type vector inverse quantization, obtain non-quantization spectral coefficient, high frequency band spreading parameter in the code stream is carried out high-frequency band signals reconstruct based on low frequency lattice vector quantization coefficient, obtain the high-frequency band signals of reconstruct, simultaneously the height frequency band is carried out the time-frequency inverse transformation respectively, obtain the low band signal and the high-frequency band signals of time domain reduction, and with the reduction low band signal and high-frequency band signals by the synthesis filter group, the class note signal that obtains reducing is exported the class note signal of reduction at last.
Compared with prior art, the present invention has following beneficial effect
1. the present invention adopts the mode of voice and musical sound combined coding, employing is encoded to speech-like signal and class note signal respectively based on the class voice coding unit of CELP with based on the class musical sound coding unit of lattice vector quantization, has the general adaptability to voice signal and note signal;
2. the present invention encodes to the residual error of local reconstruction signal and input signal, has reduced the loss of information in the coding, has improved coding tonequality, especially when code check improves, can keep high-quality tonequality;
3. realize the self-adaptation multi code Rate of Chinese character characteristic of coding by the dynamic bit distribution of lattice vector quantization, the graded properties that variable bit distributes and high frequency band is expanded of residual coding, thereby feasible coding has adaptivity to channel condition.
4. adopt voice and musical sound taxon that input signal is detected and judge that using class voice coding unit still is class musical sound coding unit, with respect to encoding and local decode with speech coder and musical sound scrambler respectively in this locality, compare the scheme which kind of coding mode is both effects select finally to use again, scheme among the present invention only need be twice of local coder, decode once, so computation complexity is lower.
Description of drawings
Fig. 1 is the structured flowchart of code device of the present invention;
Figure is the workflow diagram of class musical sound coding unit of the present invention;
Fig. 3 is the structured flowchart of decoding device of the present invention;
Fig. 4 is the workflow diagram of class musical sound decoding unit of the present invention.
Embodiment
Below in conjunction with accompanying drawing embodiments of the invention are elaborated: present embodiment is being to implement under the prerequisite with the technical solution of the present invention, provided detailed embodiment and concrete operating process, but protection scope of the present invention is not limited to following embodiment.
As shown in Figure 1, present embodiment relates to a kind of gradable voice and musical sound combined coding device, comprise: voice and musical sound taxon, class voice coding unit, local class tone decoding unit, class musical sound coding unit, local class musical sound decoding unit, residual error acquiring unit, packing output unit, wherein:
Voice and musical sound taxon are classified to the sound signal of input, according to classification results sound signal is imported class voice coding unit or class musical sound coding unit, the sound signal of importing in the present embodiment is that a frame signal is got 20ms sampling input, 256 sampled points of every frame.
Being responsible for voice signal is encoded in class voice coding unit, obtains coding parameter, and export local class tone decoding unit, residual error acquiring unit and packing output unit to;
Local class tone decoding unit carries out local decode with the parameter that class voice coding cell encoding obtains, and obtain local reconstruction signal, and the output decoder data is to the residual error acquiring unit;
Class musical sound coding unit is responsible for note signal is encoded, and obtains coding parameter, and exports local class musical sound decoding unit, residual error acquiring unit and packing output unit to;
Local class musical sound decoding unit carries out local decode with the parameter that class musical sound coding unit coding obtains, and obtain local reconstruction signal, and the output decoder data is to the residual error acquiring unit;
The residual error acquiring unit is obtained the residual error of input audio signal and local reconstruction signal, if this frame signal is judged as speech-like signal, then residual error is input to class musical sound coding unit; If this frame signal is judged as the class note signal, then residual error is input to class voice coding unit;
The parameter packing output that the packing output unit obtains class voice coding unit and class musical sound coding unit coding.
Described voice and musical sound taxon, it adopts the sound classifier that is adopted among the AVS-M (the digital audio/video encoding and decoding technique is towards the application standard that moves) to carry out the classification and Detection of voice and musical sound.According to parameters such as linear spectral coefficient and sub belt energy information and open-loop pitch parameters, the judgement input signal types comprises speech-like signal and class note signal two classes.
Described class voice coding unit, it adopts CELP (Code Excited Linear Prediction) to carry out Qualcomm Code Excited Linear Prediction (QCELP) on the time domain,, every frame is carried out LP analyze and obtain linear predictor coefficient a by high pass and pre-emphasis through pretreated voice signal
i, each speech frame is divided into 4 subframes, and each subframe is 64 samples, and 5ms is long, and following operation is carried out according to each subframe:
1. by the LP residual signals is calculated echo signal x (n) by perceptual weighting filter W (z) (its original state is upgraded by the difference of LP residual signals and pumping signal);
2. calculate the impulse response h (n) of weighted synthesis filter;
3. adopt echo signal x (n) and impulse response h (n) by searching for closed loop pitch cycle and gain near the cycle at open-loop pitch;
4. echo signal x (n) upgrades by the contribution (filtered adaptive codebook vector) of removing adaptive codebook.This new echo signal is x
2(n), be used for fixing the code word searching analysis;
5. the gain of self-adaptation and fixed codebook vector quantizes with 6 or 7 bits;
6. last, adopt selected Optimum Excitation signal that wave filter is upgraded, be used to seek the echo signal of next subframe;
7. with LP parameter, self-adaptation code vector address and the gain, fixed code vector address and the gain packing output that quantize.
Described local class musical sound decoding unit, it adopts the CELP method to carry out decoding based on Code Excited Linear Prediction, obtain parameters such as LP parameter, adaptive codebook vector, adaptive codebook gain, fixed codebook vector, fixed codebook gain, adopt the code book identical then with scrambler, find this code vector and be multiplied by gain according to the code vector address, the excitation composite filter obtains synthetic speech.
As shown in Figure 2, described class musical sound coding unit, its operational analysis bank of filters is divided into high-frequency band signals and low band signal two classes with the class note signal, when sampling rate is Fs, the low band signal frequency range is 0～Fs/4, and the high-frequency band signals frequency range is Fs/4～Fs/2, respectively these two band signals is carried out FFT (time-frequency) conversion, convert the signal into frequency domain, obtain the non-quantization spectral coefficient of two frequency bands of height;
To the non-quantization spectral coefficient of low-frequency band, adopt the lattice vector quantization method to encode.Here select 8 Wei Gaosige RE8 for use, the lattice type vector code book is made up of low order basis code book and high-order expansion code book, quantizes gain according to channel situation and code check adjustment during quantification, thereby realizes the self-adaptation of quantizing bit number is distributed, and control quantizes quality;
And high-frequency band signals is carried out encoding based on the band spread of low frequency vector quantization.The high frequency extended mode is a grading extension, and the progression according to current channel situation and code check decision expansion distributes the expansion bit number, the quality of control band spread;
To the encoding code stream format of above-mentioned two frequency bands, packing output.
Described local class musical sound decoding unit, it passes the code stream of coming to class musical sound coding unit and resolves, low-frequency band is partly carried out the lattice type vector inverse quantization, obtain the spectral coefficient of non-quantification, HFS is by band spread parameter reconstructed high frequency band spectrum envelope on the basis of the non-quantization spectral coefficient of low frequency; The spectral coefficient of two frequency bands is carried out FFT (time-frequency) inverse transformation respectively, convert the signal into time domain; Time-domain signal with two frequency bands obtains output sample by the synthesis filter group, reconstruction signal at last.
Described residual error acquiring unit, when it carries out residual coding to speech-like signal, cataloged procedure is identical with class voice coding unit, the bit number of class voice residual coding distributes according to channel and code check situation to be decided, the bit number that the good code check height of channel then distributes is many, otherwise then the bit number of Fen Peiing is less, has gradability and to the adaptivity of channel and code check.
Described residual error acquiring unit, when it carries out residual coding to the class note signal, cataloged procedure is identical with class musical sound coding unit, the bit number of class musical sound residual coding distributes and will decide according to channel and code check situation, the bit number that the good code check height of channel then distributes is many, otherwise then the bit number of Fen Peiing is less, and it has the adaptivity to channel and code check.
Described class musical sound coding unit, its vector coding based on lattice vector quantization also comprises: scrambler is adjusted the quality that quantizes according to the Bit Allocation in Discrete of Rate Control quantized segment, and quantification is gradable.
Described class musical sound coding unit, its high frequency band expansion also comprises: scrambler is adjusted the quality of band spread according to the Bit Allocation in Discrete of Rate Control high frequency band expansion, and band spread is gradable.
As shown in Figure 3, present embodiment also relates to a kind of gradable voice and musical sound combined decoding device, comprising: coding mode acquiring unit, coding parameter acquiring unit, class tone decoding unit, class musical sound decoding unit, synthetic output unit, wherein:
The coding mode acquiring unit is tentatively resolved code stream, and the coding mode that this frame signal adopted is inputed to the coding parameter acquiring unit;
The coding parameter acquiring unit is resolved code stream according to different coding modes, obtains each coding parameter and inputs to class tone decoding unit and class musical sound decoding unit;
Class tone decoding unit by using class speech coding parameters is decoded, and obtains class tone decoding data, and the output decoder data are to synthetic output unit;
Class musical sound decoding unit utilizes class musical sound coding parameter to decode, and obtains class musical sound decoded data, and the output decoder data are to synthetic output unit;
Synthetic output unit synthesizes the sound signal that obtains rebuilding with the decoded data of class tone decoding unit and the output of class musical sound decoding unit.
Described class tone decoding unit, it is by resolving code stream, corresponding code vector is found in address according to fixed code vector and self-adaptation code vector, fixed code vector and self-adaptation code vector are multiplied each other with its gain respectively, fixed code vector that obtains reducing and self-adaptation code vector, and with the reduction fixed code vector and the self-adaptation code vector respectively by the excitation composite filter, obtain synthetic speech, at last synthetic speech is exported.
As shown in Figure 4, described class musical sound decoding unit, it is by resolving code stream, low-frequency band quantization spectral coefficient in the code stream is carried out the lattice type vector inverse quantization, obtain non-quantization spectral coefficient, high frequency band spreading parameter in the code stream is carried out high-frequency band signals reconstruct based on low frequency lattice vector quantization coefficient, obtain the high-frequency band signals of reconstruct, simultaneously the height frequency band is carried out the time-frequency inverse transformation respectively, obtain the low band signal and the high-frequency band signals of time domain reduction, and with low band signal and the high-frequency band signals of reduction by the synthesis filter group, the class note signal that obtains reducing is exported the class note signal of reduction at last.
In the present embodiment, adopt the mode of voice and musical sound combined coding, employing is encoded to speech-like signal and class note signal respectively based on the class voice coding unit of CELP with based on the class musical sound coding unit of lattice vector quantization, has general adaptability to voice signal and note signal, reduce the loss of information in the coding, improved coding tonequality.Especially when code check improves, can keep high-quality tonequality.Simultaneously, only need decode once, so computation complexity is lower in the present embodiment at local coder twice.
Claims (10)
1, a kind of gradable voice and musical sound combined coding device, it is characterized in that, comprise: voice and musical sound taxon, class voice coding unit, local class tone decoding unit, class musical sound coding unit, local class musical sound decoding unit, residual error acquiring unit, packing output unit, wherein:
Voice and musical sound taxon are classified to the sound signal of input, according to classification results sound signal are imported class voice coding unit or class musical sound coding unit;
Being responsible for voice signal is encoded in class voice coding unit, obtains coding parameter, and export local class tone decoding unit, residual error acquiring unit and packing output unit to;
Local class tone decoding unit carries out local decode with the parameter that class voice coding cell encoding obtains, and obtain local reconstruction signal, and the output decoder data is to the residual error acquiring unit;
Class musical sound coding unit is responsible for note signal is encoded, and obtains coding parameter, and exports local class musical sound decoding unit, residual error acquiring unit and packing output unit to;
Local class musical sound decoding unit carries out local decode with the parameter that class musical sound coding unit coding obtains, and obtain local reconstruction signal, and the output decoder data is to the residual error acquiring unit;
The residual error acquiring unit is obtained the residual error of input audio signal and local reconstruction signal, if this frame signal is judged as speech-like signal, then residual error is input to class musical sound coding unit; If this frame signal is judged as the class note signal, then residual error is input to class voice coding unit;
The parameter packing output that the packing output unit obtains class voice coding unit and class musical sound coding unit coding.
2, gradable voice according to claim 1 and musical sound combined coding device, it is characterized in that, described class voice coding unit, every frame of voice signal of input is carried out linear prediction analysis obtain linear predictor coefficient, by the linear prediction residual difference signal is calculated echo signal by perceptual weighting filter, calculate the impulse response of weighted synthesis filter simultaneously, adopt echo signal and impulse response by searching for closed loop pitch cycle and gain near the cycle at open-loop pitch, echo signal is upgraded by the contribution of removing adaptive codebook, the new echo signal that obtains is used for fixing the code word searching analysis, calculate the gain of self-adaptation and fixed codebook vector, at last with the code vector address and the gain of random code resultant fixed code vector, and packing output behind the filter parameter quantization encoding.
3, gradable voice according to claim 1 and musical sound combined coding device, it is characterized in that, described class musical sound coding unit, the class note signal is passed through analysis filterbank, signal is divided into high-frequency band signals and low band signal two-way, when two paths of signals is carried out respectively-the frequency conversion, convert the signal into frequency domain, obtain the non-quantization spectral coefficient of two frequency bands, the non-quantization spectral coefficient of low-frequency band is carried out vector coding based on lattice vector quantization, the non-quantization spectral coefficient of high frequency band is carried out band spread based on the low-frequency band lattice vector quantization, at last with low-frequency band vector coding code stream and packing of high frequency band extended coding code stream and output.
4, according to claim 1 or 3 described gradable voice and musical sound combined coding device, it is characterized in that, described class musical sound coding unit, its vector coding based on lattice vector quantization also comprises: scrambler is adjusted the quality that quantizes according to the Bit Allocation in Discrete of Rate Control quantized segment.
5, according to claim 1 or 3 described gradable voice and musical sound combined coding device, it is characterized in that, described class musical sound coding unit, its high frequency band expansion also comprises: scrambler is adjusted the quality of band spread according to the Bit Allocation in Discrete of Rate Control high frequency band expansion.
6, gradable voice according to claim 1 and musical sound combined coding device, it is characterized in that, described residual error acquiring unit, when it carries out residual coding to speech-like signal, cataloged procedure is identical with class voice coding unit, and the bit number of class voice residual coding distributes according to channel and code check situation to be decided, and the bit number that the good code check height of channel then distributes is many, otherwise then the bit number of Fen Peiing is less, has gradability and to the adaptivity of channel and code check.
7, according to claim 1 or 6 described gradable voice and musical sound combined coding device, it is characterized in that, described residual error acquiring unit, when it carries out residual coding to the class note signal, cataloged procedure is identical with class musical sound coding unit, and the bit number of class musical sound residual coding distributes and will decide according to channel and code check situation, and the bit number that the good code check height of channel then distributes is many, otherwise then the bit number of Fen Peiing is less, and it has the adaptivity to channel and code check.
8, a kind of gradable voice and musical sound combined decoding device is characterized in that, comprising: coding mode acquiring unit, coding parameter acquiring unit, class tone decoding unit, class musical sound decoding unit, synthetic output unit, wherein:
The coding mode acquiring unit is tentatively resolved code stream, and the coding mode that this frame signal adopted is inputed to the coding parameter acquiring unit;
The coding parameter acquiring unit is resolved code stream according to coding mode, obtains each coding parameter and inputs to class tone decoding unit and class musical sound decoding unit;
Class tone decoding unit by using class speech coding parameters is decoded, and obtains class tone decoding data, and the output decoder data are to synthetic output unit;
Class musical sound decoding unit utilizes class musical sound coding parameter to decode, and obtains class musical sound decoded data, and the output decoder data are to synthetic output unit;
Synthetic output unit synthesizes the sound signal that obtains rebuilding with the decoded data of class tone decoding unit and the output of class musical sound decoding unit.
9, gradable voice according to claim 8 and musical sound combined decoding device, it is characterized in that, described class tone decoding unit, it finds corresponding code vector by resolving code stream according to the address of fixed code vector and self-adaptation code vector, and fixed code vector and self-adaptation code vector are multiplied each other with its gain respectively, fixed code vector that obtains reducing and self-adaptation code vector, and with the reduction fixed code vector and the self-adaptation code vector respectively by the excitation composite filter, obtain synthetic speech, at last synthetic speech is exported.
10, gradable voice according to claim 8 and musical sound combined decoding device, it is characterized in that, described class musical sound decoding unit, it is by resolving code stream, low-frequency band quantization spectral coefficient in the code stream is carried out the lattice type vector inverse quantization, obtain non-quantization spectral coefficient, high frequency band spreading parameter in the code stream is carried out high-frequency band signals reconstruct based on low frequency lattice vector quantization coefficient, obtain the high-frequency band signals of reconstruct, simultaneously the height frequency band is carried out the time-frequency inverse transformation respectively, obtain the low band signal and the high-frequency band signals of time domain reduction, and with the reduction low band signal and high-frequency band signals by the synthesis filter group, the class note signal that obtains reducing is exported the class note signal of reduction at last.
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