CA2572052A1 - Audio encoding device, audio decoding device, and method thereof - Google Patents

Abstract

There is disclosed an audio encoding device capable of realizing effective encoding while using audio encoding of the CELP method in an extended layer when hierarchically encoding an audio signal. In this device, a first encoding unit (115) subjects an input signal (S11) to audio encoding processing of the CELP method and outputs the obtained first encoded information (S12) to a parameter decoding unit (120). The parameter decoding unit (120) acquires a first quantization LSP code (L1), a first adaptive sound source lag code (A1), and the like from the first encoded information (S12), obtains a first parameter group (S13) from these codes, and outputs it to a second encoding unit (130). The second encoding unit (130) subjects the input signal (S11) to a second encoding processing by using the first parameter group (S13) and obtains second encoded information (S14). A multiplexing unit (154) multiplexes the first encoded information (S12) with the second encoded information (S14) and outputs them via a transmission path N to a decoding device (150).

Description

DESCRIPTION
AUDIO ENCODING DEVICE, AUDIO DECODING DEVICE, AND METHOD THEREOF

Technical Field [0001] The present invention relates to a speech encoding apparatus that hierarchically encodes a speech signal, a speech decoding apparatus that hierarchically decodes encoded information generated by the speech encoding apparatus, and a method thereof.

Background Art

[0002] In communication systems handling digitized speech/sound signals, such as mobile communication or the Internet communication, speech/sound signal encoding/decodingtechniquesareessentialfor effective use of a communication line that is a limited resource, and many encoding/decoding schemes have so far been developed.

[0003] Among these, particularly a CELP encoding and decoding scheme is put in practical use as a mainstream scheme (see, for example, Non-Patent Document 1) . The CELPschemespeechencoding apparatusencodesinputspeech based on a speech generation model. Specifically, a digital speech signal is separated into frames of approximately 20 ms, linear prediction analysis of the speech signals is performed per frame, and the obtained linear prediction coefficients and linear prediction residual vectors are encoded individually.

5[0004] In communication systems where packets are transmitted, such as Internet communication, packet loss may occur depending on the network state, and a function is desired where speech and sound can be decoded using the remaining encoded information, even if part of encoded information is lost. Similarly, also in variable rate communication systems where a bit rate varies depending on line capacity, when the line capacity decreases, it is desirable to reduce the burden on communication system by transmitting a part of encoded information. As a technique of capable of decoding the original data using all or part of encoded information, a scalable encoding technique has lately attracted attention. Several scalable encoding schemes have been conventionally disclosed (see, for example, Patent Document 1).

[0005] A scalable encoding scheme generally consists of a base layer and a plurality of enhancement layers, and these layers form a hierarchical structure in which the base layer is the lowest layer. Encoding of each layer is performed by taking a residual signal, which is a signal representing a difference between an input signal of the lower layer and a decoded signal, as a target forencoding, andusingencodedinformationat lower layers.

This configuration enables the original data decoding using encoded information of all layers or only encoded information at lower layers.

PatentDocumentl: JapanesePatentApplication Laid-Open No. HE110-97295 Non-Patent Document 1: Manfred R. Schroeder, Bishnu S.
Atal, "CODE-EXCITED LINER PREDICTION (CELP):
HIGH-QUALITY SPEECH AT VERY LOW BIT RAYES, " IEEE proc. , ICASSP'85 pp.937-940 Disclosure of Invention Problems to be Solved by the Invention [0006] However, when scalable encoding on aspeechsignal is considered, in the conventional method, the target for encoding in an enhancement layer is a residual signal.
This residual signal is a differential signal between the input signal of the speech encoding apparatus (or a residual signal obtained at the subsequent lower layer) and the decoded signal at the subsequent lower layer, and therefore is a signal where many speech components arelostandmanynoisecomponentsareincluded. Therefore, in the enhancement layer in the conventional scalable encoding, when an encoding scheme specific to speech encoding such as a CELP scheme for encoding based on a speech generation model is applied, encoding has to be performed based on the speech generation model on the

4 residual signal where many speech components are lost, and it is impossible to encode this signal efficiently.
Moreover, encoding the residual signal using an encoding scheme other than CELP abandons an advantage of the CELP

scheme capable of obtaining a high-quality decodedsignal with lesser bits, and it is not effective.

[ 00071 It is therefore an obj ect of the present invention to provide a speech encoding apparatus capable of implementing, when a speech signal is hierarchically encoded, efficientencodingwhileusingCELPschemespeech encoding in an enhancement layer and obtaining a high-quality decoded signal, a speech decoding apparatus that decodes encoded inf ormation generated by this speech encoding apparatus, and a method thereof.

Means for Solving the Problem [0008] A speech encoding apparatus of the present invention adopts a configuration including a first encoding section that generates, from a speech signal, encoded information by CELP scheme speech encoding, a generating section that generates, from the encoded information, a parameter representing a feature of a generationmodelofthespeechsignal, andasecondencoding section that takes the speech signal as an input and encodes the inputted speech signal by CELP scheme speech encoding using the parameter.

[0009] Here, the above parameter means a parameter unique to the CELP scheme used in CELP scheme speech encoding, namely a quantized LSP(LineSpectralPairs), an adaptive excitation lag, a fixed excitation vector, a quantized adaptiveexcitation gain,ora quantizedfixedexcitation

5 gain.

[0010] Forexample, intheaboveconfiguration, thesecond encodingsection adoptsaconfiguration wherea difference between an LSP obtained by linear prediction analysis on the speech signal that is an input of the speech encoding apparatus, and a quantized LSPgenerated by the generating section is encoded using CELP scheme speech encoding.
That is, the second encoding section takes the different at the stage of the LSP parameter, andperforms CELP scheme speech encoding on this difference, thereby achieving CELP scheme speech encoding that does not take a residual signal as an input.

[0011] Here, in the above configuration, the first encoding section and the second encoding section do not restrictively mean first layer (base layer) encoding section and second layer encoding section, respectively, and may mean, for example, second layer encoding section and third layer encoding section, respectively. Also, these sections do not necessarily mean encoding sections for adjacent layers, and may mean, for example, first encoding means as f irst layer encoding section and second encoding means as third layer encoding section.

6 Advantageous Effect of the Invention [0012] According to the present invention, when a speech signal is encoded hierarchically, it is possible to implement efficient encoding while using CELP scheme speech encoding in an enhancement layer, and obtain a high-quality decoded signal.

Brief Description of Drawings [0013] FIG.1 is a block diagram showing the main conf igurations of a speech encoding apparatusand aspeech decoding apparatus according to Embodiment 1;

FIG.2 shows a flow of each parameter in a speech encoding apparatus according to Embodiment 1;

FIG.3 is a block diagram showing an internal configuration of a first encoding section according to Embodiment 1;

FIG.4 is a block diagram showing an internal configuration of a parameter decoding section according to Embodiment 1;

FIG.5 is a block diagram showing an internal configuration of a second encoding section according to Embodiment 1;

FIG.6 outlines processing of determining a second adaptive excitation lag;

FIG.7 outlines processing of determining a second fixed excitation vector;

FIG.8 outlines processing of determining a first

7 adaptive excitation lag;

FIG.9 outlines processing of determining a first fixed excitation vector;

FIG.10 is a block diagram showing an internal configuration of a first decoding section according to Embodiment 1;

FIG.11 is a block diagram showing an internal configuration of a second decoding section according to Embodiment 1;

FIG.12A is a block diagram showing a configuration of a speech/sound transmitting apparatus according to Embodiment 2;

FIG.12B is a block diagram showing a configuration of a speech/sound receiving apparatus according to Embodiment 2; and FIG.13 is a block diagram showing the main conf igurations of a speech encoding apparatus and a speech decoding apparatus according to Embodiment 3.

Best Mode for Carrying Out the Invention [0014] Embodiments of the present invention will be described in detail below with reference to the accompanying drawings.

[0015] (Embodiment 1) FIG.1 is a block diagram showing the main configurationsofspeechencoding apparatus 100 andspeech decoding apparatus 150 according to Embodiment 1 of the

8 present invention.

[0016] In this figure, speech encoding apparatus 100 hierarchically encodes input signal S11 in accordance with an encoding method according to this embodiment, multiplexes obtained hierarchical encoded information S12andS14,andtransmitsmultiplexedencodedinformation (multiplexed information) to speech decoding apparatus 150 via transmission path N. On the other hand, speech decoding apparatus 150 demultiplexes the multiplexed information from speech encoding apparatus 100 toencoded information S12 and S14, decodes the encoded information after demultiplexing in accordance with a decoding method according to this embodiment, and outputs output signal S54.

[0017] First, speech encoding apparatus 100 will be described in detail.

[0018] Speech encoding apparatus 100 is mainly composed of f irst encoding section 115, parameter decoding section 120,secondencodingsection130,and multiplexingsection 154,andsectionsperformthefollowingoperations. Here, FIG.2 shows a flow of each parameter in speech encoding apparatus 100 according to Embodiment 1.

[0019] First encoding section 115 perf orms a CELP scheme speech encoding (first encoding) processing on speech signal S11 inputted to speech encoding apparatus 100, and outputs encoded information (first encoded information) S12 representing each parameter obtained

9 based on a generation model of the speech signal to multiplexing section 154. Also, first encoding section 115a1sooutputsfirstencodedinformationSl2toparameter decoding section 120 to perform hierarchical encoding.

The parameters obtained by the f irst encoding processing are hereinafter referred to as a first parameter group.
Specifically, the first parameter group includes a first quantized LSP (Line Spectral Pairs), a first adaptive excitation lag, a first fixed excitation vector, a first quantized adaptiveexcitation gain,and afirstquantized fixed excitation gain.

[00201 Parameterdecodingsection120performsparameter decoding on first encoded information S12 outputted from first encoding section 115, and generates parameters representing a feature of the generation mode of the speech signal. In this parameter decoding, encoded information is not completely decoded, but partially decoded, thereby obtaining the above-described first parameter group.
That is, while it is an obj ect of the conventional decoding processing to obtain the original signal before encoding by decoding encoded information, it is an object of the parameter decoding processing to obtain the first parameter group. Specifically, the parameter decoding section120 demultiplexes first encoded information S12, andobtainsafirstquantizedLSPcode(L1),afirstadaptive excitation lag code (Al), a first quantized adaptive excitation gain code (G1) , and a first fixed excitation vector gain (F1), and obtains a first parameter group S13 from each of the obtained codes. This first parameter group S13 is outputted to second encoding section 130.
[0021] Second encoding section 130 obtains a second 5 parameter group by performingsecond encoding processing which will be described later, using the input signal S1lofspeechencodingapparatus100andthefirstparameter group S13 outputted from parameter decoding section 12 0, and outputs encoded information (second encoded

10 information) S14 representing this secondparameter group to multiplexing section 154. Here, the second parameter group includes a second quantized LSP, a second adaptive excitationlag, a second fixed excitation vector, asecond quantized adaptiveexcitation gain, andasecond quantized fixed excitation gain each corresponding to those of the first parameter group.

[0022] The first encoded information S12 is inputted to multiplexing section 154 from first encoding section 115,andalsothesecondencodedinformationSl4isinputted from second encoding section 130. Multiplexing section 154 selects necessary encoded information in accordance with mode information of the speech signal inputted to speech encoding apparatus 100, multiplexes the selected encoded information and the mode information, and generates the multiplexed encoded information (multiplexed information) . Here, the mode information is information that indicates encoded information to be

11 multiplexed and transmitted. For example, when the mode informationis"0",multiplexingsection154multiplexes thefirstencodedinformationSl2andthemodeinformation, and whenthemodeinformationis"1",multiplexingsection 154 multiplexes the first encoded information S12, the secondencodedinformationSl4,andthemodeinformation.
As described above, by changing a value of the mode information, a combination of encoded information to be transmittedto speechdecodingapparatus 150 canbe changed.

Next, multiplexing section 154 outputs the multiplexed information after multiplexing to speech decoding apparatus 150 via the transmission path N.

[0023] As described above, this embodiment is characterized by the operations of parameter decoding section 120 and second encoding section 130. For convenience of description, processing of sections will be described in detail in the order of first encoding section 115, parameter decoding section 120, and then s.econd encoding section 130.

[0024] FIG.3 is a block diagram showing an internal configuration of first encoding section 115.

[0025] Preprocessing section 101 performs, onthespeech signal S11 inputted to speech encoding apparatus 100, high-pass filter processing of removing DC components and waveform shaping processing and pre-emphasis processing which help to improve the performance of subsequentencodingprocessing,andoutputsthe processed

12 signal (Xin) to LSP analyzing section 102 and adder 105.
[0026] LSP analyzing section 102 performs linear prediction analysis using the Xin, converts LPC (Linear Prediction Coefficients)resultingfromtheanalysisinto LSP, and outputs the conversion result as a first LPC
to LSP quantizing section 103.

[0027] LSP quantizing section 103 quantizes the first LSP outputted from LSP analyzing section 102 using quantizing processing which will be described later, and outputs a quantized first LSP (first quantized LSP) to synthesis filter 104. Also, LSP quantizing section 103 outputs a first quantized LSP code (L1) representing the first quantized LSP to multiplexing section 114.

[0028] Synthesis filter 104 performs filer synthesis of a driving excitation outputted from adder 111 using a filter coefficient based on the first quantized LSP, and generates a synthesis signal. The synthesis signal is outputted to adder 105.

[0029] Adder 105 reverses the polarity of the synthesis signal, adds this signal to Xin, thereby calculating an error signal, and outputs this calculated error signal to auditory weighting section 112.

[0030] Adaptive excitation codebook 106 has a buffer storing driving excitations which have been previously outputted from adder 111. Also, based on an extraction position specified by a signal outputted from parameter determining section 113, adaptive excitation codebook

13 106 extracts a set of samples for one frame from the buffer at the extraction position, and outputs the sample set as a first adaptive excitation vector to multiplier 109.
Further, adaptive excitation codebook 106 updates the above buffer, each time a driving excitation is inputted from adder 111.

[0031] Quantized gain generatingsection107determines, basedon aninstructionfrom parameterdeterminingsection 113, a first quantized adaptive excitation gain and a first quantized fixed excitation gain, and outputs the first quantized adaptive excitation gain to multiplier 109 and the first quantized fixed excitation gain to multiplier 110.

[0032] Fixed excitation codebook 108 outputs a vector having a form specified by the instruction from parameter determining section 113 as a first fixed excitation vector to multiplier 110.

[0033] Multiplier 109 multiplies the first quantized adaptive excitation gain outputted from quantized gain generating section 107 by the first adaptive excitation vector outputted from adaptive excitation codebook 106, and outputs the result to adder 111. Multiplier 110 multiplies the first quantized fixed excitation gain output from quantized gain generating section 107 by the first fixed excitation vector outputted from fixed excitation codebook 108 and outputs the result to adder 111. Adder 111 adds the f irst adaptive excitation vector

14 multiplied by the gain at multiplier 109 and the first fixed excitation vector multiplied by the gain at multiplier110,andoutputsa drivingexcitationresulting from the addition to synthesis filter 104 and adaptive excitationcodebook106. Thedriving excitationinputted to adaptive excitation codebook 106 is stored into the buffer.

[0034] Auditory weightingsection112appliesan auditory weight to the error signal outputted from adder 105 and outputs a result as an encoding distortion to parameter determining section 113.

[0035] Parameter determining section 113 selectsafirst adaptive excitation lag that minimizes the encoding distortionoutputtedfrom auditory weightingsection112, and outputs a first adaptive excitation lag code (Al) indicating a selected lag to multiplexing section 114.
Also, parameter determining section 113 selects a first fixed excitation vector that minimizes the encoding distortionoutputtedfrom auditory weightingsection112, and outputs a first fixed excitation vector code (Fl) indicating a selected vector to multiplexing section 114.
Further,parameterdeterminingsection113selectsafirst quantized adaptive excitation gain and afirst quantized fixed excitation gain that minimize the encoding distortionoutputtedfrom auditory weightingsection112, and outputs a first quantized excitation gain code (G1) indicating selected gains to multiplexing section 114.

[0036] Multiplexing section 114 multiplexes the first quantized LSP code (L1) outputted from LSP quantizing section 103 and the first adaptive excitation lag code (Al) , the first fixed excitation vector code (Fl) , and 5 the first quantized excitation gain code (Gl) outputted from parameter determining section 113, and outputs the result as the first encoded information S12.

[0037] FIG.4 is a block diagram showing an internal configuration of parameter decoding section 120.

10 [0038] Demultiplexing section 121 demultiplexes the first encoded information S12 outputted from first encoding section 115 into individual codes ( L1 , Al, G1 , andF1),andoutputcodestoeachcomponent. Specifically, the first quantized LSP code (L1) demultiplexedfrom the

15 firstencodedinformationSl2isoutputtedtoLSPdecoding section 122, the first adaptive excitation lag code (Al) demultiplexed as well is outputted to adaptive excitation codebook 123, the first quantized excitation gain code (G1) demultiplexed as well is outputted to quantized gain generating section 124, and the first fixed excitation vector code (Fl) demultiplexed as well is outputted to fixed excitation codebook 125.

[0039] LSP decoding section 122 decodes the first quantized LSP code (L1) outputted from demultiplexing section 121 to a first quantized LSP, and outputs the decoded first quantized LSP to second encoding section 130.

16 [0040] Adaptive excitation codebook 123 decodes an extraction position specified by the first adaptive excitation lag code (Al) as a first adaptive excitation lag. Then, adaptive excitation codebook 123 outputs the obtained f irst adaptive excitation lag to second encoding section 130.

[0041] Quantized gain generating section 124 decodes thefirstquantized adaptive excitation gain andthefirst quantized fixed excitation gain specified by the first quantized excitation gain code (G1) outputted from demultiplexing section 121. Then, quantized gain generating section 124 outputs the obtained first quantized adaptive excitation gain to second encoding section130,and alsothefirstquantizedfixedexcitation gain to second encoding section 130.

[0042] Fixed excitation codebook 125 generates a first fixed excitation vector specified by the first fixed excitation vectorcode(F1)outputtedfrom demultiplexing section 121, and outputs the vector to second encoding section 130.

[0043] The above-described first quantized LSP, first adaptive excitation lag, first fixed excitation vector, first quantized adaptive excitation gain, and first quantizedfixedexcitation gain areoutputted as the first parameter group S13 to second encoding section 130.
[0044] FIG.5 is a block diagram showing an internal configuration of second encoding section 130.

17 [0045] Preprocessing section 131 performs, onthespeech signal S11 inputted to speech encoding apparatus 100, high-pass filter processing of removing DC components and waveform shaping processing and pre-emphasis processing which help to improve the performance of subsequent encoding processing, andoutputs theprocessed signal (Xin) to LSP analyzing section 132 and adder 135.
[0046] LSP analyzing section 132 performs linear prediction analysis using the Xin, converts LPC (Linear PredictionCoefficients)resultingfromtheanalysisinto LSP (Line Spectral Pairs), and outputs the conversion result as a second LSP to LSP quantizing section 133.
[0047] LSP quantizing section 133 reverses the polarity of the first quantized LSP outputted from parameter decoding section 120, adds the first quantized LSP after polarity reversion to the second LSP outputted from LSP
analyzing section 132 and thereby calculating a residual LSP. Next, LSP quantizing section 133 quantizes the calculatedresidualLSPusing quantizing processing which will be described later, adds the quantized residual LSP
(quantized residual LSP) and the first quantized LSP
outputtedfrom parameterdecodingsection120,andthereby calculatingasecondquantizedLSP. Thissecond quantized LSP is outputted to synthesis filter 134, while the second quantized LSP code (L2) representing the quantized residual LSP is outputted to multiplexing section 144.
[0048] Synthesis filter 134 performs filter synthesis

18 of a driving excitation, outputted from adder 141, by a filter coefficient based on the second quantized LSP, and thereby generates a synthesis signal. The synthesis signal is outputted to adder 135.

5[0049] Adder 135 reverses the polarity of the synthesis signal, adds this signal to Xin, thereby calculating an error signal, and outputs this calculated error signal to auditory weighting section 142.

[0050] Adaptive excitation codebook 136 has a buffer storing driving excitations which have been previously outputted from adder 141. Also, based on an extraction position specified by the first adaptive excitation lag and asignaloutputtedfrom parameterdeterminingsection 143, adaptive excitation codebook 136 extracts a set of samples for one frame from the buffer at the extraction position, and outputs the sample set as a second adaptive excitation vector to multiplier 139. Further, adaptive excitation codebook 136 updates the above buffer, each time a driving excitation is inputted from adder 141.

[0051] Quantized gain generating section 137 obtains, basedon aninstructionfrom parameterdeterminingsection 143, a second quantized adaptive excitation gain and a second quantized fixed excitation gain using the first quantizedadaptiveexcitation gain andthefirstquantized fixed excitation gain outputted from parameter decoding section 120. The second quantized adaptive excitation gain is outputted to multiplier 139, and the second .. =

19 quantizedfixedexcitation gainisoutputtedtomultiplier 140.

[0052] Fixed excitation codebook 138 obtains a second fixed excitation vector by adding a vector having a form specified by the indication from parameter determining section143andthefirstfixedexcitationvectoroutputted fromparameterdecodingsection120, andoutputs theresult to multiplier 140.

[0053] Multiplier 139 multiplies the second adaptive excitation vector outputted from adaptive excitation codebook 136 by the second quantized adaptive excitation gain outputted from quantized gain generating section 137, and outputs the result to adder 141. Multiplier 140 multiplies the second fixed excitation vector outputted from f ixed excitation codebook 138 by the second quantizedfixedexcitation gain outputted f rom quantized gain generating section 137, and outputs the result to adder 141. Adder 141 adds the second adaptive excitation vector multiplied by the gain at multiplier 139 and the second fixed excitation vector multiplied by the gain at multiplier 140, and outputs a driving excitation resulting from the addition to synthesis filter 134 and adaptiveexcitationcodebook136. Thedrivingexcitation inputted to adaptive excitation codebook 136 is stored into the buffer.

[0054] Auditory weightingsection142appliesan auditory weighting to the error signal outputted from adder 135, and outputs a result as encoding distortion to parameter determining section 143.

[0055] Parameterdeterminingsection143selectsasecond adaptive excitation lag that minimizes the encoding 5 distortion output from auditory weighting section 142, and outputs a second adaptive excitation lag code (A2) indicating the a selected lag to multiplexing section 144. Also, parameter determining section 143 selects a second fixed excitation vector that minimizes the 10 encoding distortion outputted from auditory weighting section 142 using the first adaptive excitation lag outputtedfrom parameterdecodingsection120,and outputs a second fixed excitation vector code (F2) indicating aselected vector to multiplexing section 144. Further, 15 parameter determining section 143 selects a second quantized adaptive excitation gain and a second quantized fixed excitation gain that minimizes the encoding distortionoutputtedfrom auditory weightingsection142, and outputs a second quantized excitation gain code (G2)

20 indicating a selected gain to multiplexing section 144.
[0056] Multiplexing section 144 multiplexes the second quantized LSP code (L2) outputted from LSP quantizing section 133 and the second adaptive excitation lag code (A2 ), the second fixed excitation vector code (F2 ), and the second quantized excitation gain code (G2) outputted from parameterdeterminingsection143,outputstheresult as the second encoded information S14.

21 [0057] Next, processing will be described where LSP
quantizing section 13 3 shown in FIG. 5 determinesasecond quantized LSP. Here, an example will be described where the number of bits assigned to the second quantized LSP

code (L2) is "8" and the residual LSP is vector-quantized.
[0058] LSP quantizing section 133 is providedwith second LSP codebook in which 256 variants of second LSP code vectors [1Spres(L21 ) (i)] created in advance are stored.
Here, L2' is an index attached to the second LDP code vector, and takes anyvalue of 0 to 255 . Also, 1Spres(L2 ) ( i) is an N-dimensional vector, and i takes a value from 0 to N-1.

[0059] A second LSP [aZ (1)] is inputted to LSP quantizing section 133 from LSP analyzing section 132 , Here, a2 (i) is an N-dimensional vector, and i takes a value from 0 to N-1. A first quantized LSP [lSp1(L1'min) (i) ] is also inputted to LSP quantizing section 133 from parameter decoding section 120. Here, lspl(L1'min) (i) is an N-dimensional vector, and i takes a value from 0 to N-1.

[0060] LSP quantizing section 133 obtains a residual LSP [res(i)] by the following (Equation 1).
[Equation 1]

res(i)=a2W-lspi''''"" ) (i) ~~~(~ N-1) ... (Equation 1) Next, LSP quantizing section 133 obtains squared error er2 between the residual LSP [ res ( i)] and the second

22 LSP code vector [ 1Spres (L2 )( i)] by the following (Equation 2).

[Equation 2]

er2 (res(i) -lsp~s2')(i)~Z ... (Equation 2) r=o Then, LSP quantizing section 133 obtains a squared error er2 for al l L2 ' and determines a value of L2 '( L2 ' min ) that minimizes the squared error er2. The determined L2'minis outputtedtomultiplexingsection144 as asecond quantized LSP code (L2).

[0061] Next, LSP quantizing section 133 obtains a second quantized LSP [lsp2(i)] by the following (Equation 3).
[Equation 3]

lspZW =lsp;Ll'min)(i) +lsp~s2min)~i~~(~~(~ ... (Equation 3) LSP quantizing section 133 outputs this second quantized LSP [lsp2(i)] to synthesis filter 134.

[0062] As described above, [lsp2(i)] obtained by LSP
quantizing section 133 is the second quantized LSP, and 1Spres(L21min) (i) that minimizes the squared error er2 is a quantized residual LSP.

[0063] FIG. 6 outlines processing of determining asecond adaptive excitation lag by parameter determiningsection 143 shown in FIG. 5.

23 [0064] In this figure, a buffer B2 is a buffer provided by adaptive excitation codebook 136, a position P2 is an extraction position of the second adaptive excitation vector, and a vector V2 is extracted second adaptive excitation vector. Also, t represents a first adaptive excitation lag, and values 41 and 296 correspond to a lowerlimitandanupperlimitoftherangeinwhichparameter determining section 143 searches for the first adaptive excitation lag. Further, t-16 and t+15 correspond to a lower limit and an upper limit of the range in which the extraction position of the second adaptive excitation vector is shifted.

[0065] The range in which the extraction position P2 is shifted is set at a range of a length of 32 (=25) (for example, t-16 to t+15 ), when 5 bits are assigned to the code (A2) representing the second adaptive excitation lag. However, the range in which the extraction position P2 is shifted can be arbitrarily set.

[0066] Parameter determining section 143 sets the range in which the extraction position P2 is shifted at t-16 to t+15 with reference to the first adaptive excitation lag t inputted from parameterdecodingsection120. Next, parameter determining section 143 shifts the extraction position P2 within the above range and sequentially specifies the extraction position P2 to adaptive excitation codebook 136.

[0067] Adaptive excitation codebook 136 extracts the

24 second adaptive excitation vector V2 for the length of the frame from the extraction position P2 specified by parameter determining section 143, and outputs the extracted second adaptive excitation vector V2 to multiplier 139.

[0068] Parameter determining section 143 obtains an encoding distortion outputted from auditory weighting section 142 for all second adaptive excitation vectors V2 extracted from all extraction positions P2, and determines an extraction position P2 that minimizes this encoding distortion. The buffer extraction position P2 obtained by the parameter determining section 143 is the second adaptive excitation lag. Parameter determining section 143 encodes a difference (in the example of FIG. 6, -16 to +15) between the first adaptive excitation lag and the second adaptive excitation lag, and outputs the code obtained through encoding to multiplexing section 144 as the second adaptive excitation lag code (A2).
[0069] In this manner, with the difference between the first adaptive excitation lag and the second adaptive excitation lag being encoded in second encoding section 130, second decoding section 180 adds the first adaptive excitation lag (t) obtained through the first adaptive excitation lag code and the difference from the second adaptiveexcitationlagcode (-16to+15) , therebydecoding the second adaptive excitation lag (t-16 to t+15).
[0070] Asdescribedabove,parameterdeterminingsection 143 receives the first adaptive excitation lag t from parameter decoding section 120, and searches for a range around this t in search for the second adaptive excitation lag, thereby making it possible to quickly find an optimum 5 second adaptive excitation lag.

[0071] FIG. 7 outlines processing of determining asecond fixedexcitation vectorbytheaboveparameter determining section 143. This figure indicates the process of generating asecondfixedexcitation vectorfrom algebraic 10 fixed excitation codebook 138.

[00721 Track 1, track 2, and track 3 each generate one unit pulse (701, 702, and 703) with an amplitude value of 1(solid lines in the figure) . Each track has different positions where a unit pulse can be generated. In the 15 example of the figure, the tracks are configured such that track 1 raises a unit pulse at any of eight positions {0, 3, 6, 9, 12, 15, 18, 21}, track 2 raises a unit pulse at any of eight positions {1, 4, 7, 10, 13, 16, 19, 22}, and track 3 raises a unit pulse at any of eight positions 20 {2, 5, 8, 11, 14, 17, 20, 23}.

[0073] Multiplier704appliespolaritytotheunitpulse generated in track 1. Multiplier 705 applies polarity to the unit pulse generated in track 2. Multiplier 706 applies polarity to the unit pulse generated in track

25 3. Adder707addsthegeneratedthreeunitpulsestogether.
Multiplier 708 multiplies the added three unit pulses byapredeterminedconstant R. The constant R is a constant .. . .

26 for changing the magnitude of the pulse, and it has been experimentally known that an excellent performance can be obtained when the constant R is set at a value in the order of 0 to 1. Also, the value of the constant R may be set so as to obtain a performance suitable according to the speech encoding apparatus. Adder 711 adds residual fixed excitation vector 709 composed of three pulses and a f irst f ixed excitation vector 710 together, and obtains secondfixed excitation vector712. Here, residual f ixed excitation vector 709 is multiplied by the constant R
in a range from 0 to 1 and is then added to first fixed excitation vector 7 10, andasaresult, weighting addition withthe first fixed excitation vector 710 beingweighted is applied.

[0074] In this example, unit pulse has eight patterns of positions and two patterns of positions, positive and negative, and three bits for position information and one bit for polarity information are used to represent eachunitpulse. Therefore,thefixedexcitationcodebook has 12 bits in total.

[0075] In order to shift generation position of three unit pulses andpolarities, parameter determiningsection 143 sequentially indicates the generation position and polarity to fixed excitation codebook 138.

[0076] Fixedexcitationcodebook138configuresresidual fixed excitation vector 709 using the generation position and polarity indicated by parameter determining section

27 143, adds the configured residual fixed excitation vector 709 and first fixed excitation vector 710 outputted from parameter decoding section 120 together, and outputs second fixed excitation vector 712 resulting from the addition to multiplier 140.

[0077] Parameter determining section 143 obtains an encoding distortion outputted from auditory weighting section 142 for the second fixed excitation vector with respect to all combinations of the generation position and polarity, and determines a combination of the generation position and polarity that minimizes the encoding distortion. Next, parameter determining section 143 outputs the second fixed excitation vector code (F2) representing the determined combination of the generation position and the polarity to multiplexing section 144.

[0078] Next,processing willbedescribed where the above parameter determining section 143 carries out an instruction to quantized gain generating section 137, and determines a second quantized adaptive excitation gain and a second quantized fixed excitation gain. Here, a case will be described as an example where 8 bits are assigned to the second quantized excitation gain code (G2).

[0079] Quantized gain generating section 137 isprovided with a residual excitation gain codebook in which 256 variants of residual excitation gain code vectors .~ =

28 [gain2 ( K 2 ' )( i)] created in advance are s tored . Here, K2 ' is an index attached to the residual excitation gain code vector, and takes any value of 0 to 255. Also, gain2 (K2' )( i) is a two-dimensional vector, and i takes a value from 0 to 1.

[0080] Parameter determining section 143 indicates a value of K2' from 0 to 255 to quantized gain generating section 137. Quantized gain generating section 137 selects a residual excitation gain code vector [gainZ M2' (i)] from the residual excitation gain codebook using K2'indicated by parameter determining section 143, obtains a second quantized adaptive excitation gain [gainq(0)] from the following (Equation 4) , and outputs the obtained gainq(0) to multiplier 139.

[Equation 4]

gainq(0) = gain;"''"" )(O) +gain212~)(0) ... (Equation 4) Also, quantized gain generating section 137 obtains a second quantized f ixed excitation gain [gainq(1)] from the following (Equation 5), and outputs the obtained gainq(1) to multiplier 140.

[Equation 5]

gainq(1) = gain,(K''"" ) (1) + gain2K2')(1) ( Equa t i on 5) Here, gain1(xi'min) (0) represents a first quantized

29 adaptive excitation gain, and gainl (k1 'min) ( 1) represents a first quantized fixed excitation gain, each being outputted from parameter decoding section 120.

[0081] Asdescribed above,gainq(0)obtainedby quantized gain generatingsection137representsasecond quantized adaptiveexcitationgain,andgainq(1)isasecondquantized fixed excitation gain.

[0082] Parameter determining section 143 obtains an encoding distortion outputted from auditory weighting section 142 for all K2', and determines a value of K2' (K2'min) that minimizes the encoding distortion. Next, parameter determining section 143 outputs the determined K2'min to multiplexing section 144 as a second quantized excitation gain code (G2).

[0083] As described above, according to the speech encoding apparatus of this embodiment, by taking the input signal of the speech encoding apparatus as a target for encodingbysecondencodingsection130, CELPschemespeech encoding suitable for encoding a speech signal can be effectively applied, thereby obtaining decoded signal with good quality. Also, second encoding section 130 encodes the input signal using the first parameter group and generates a second parameter group, the decoding apparatus side can generate a second decoded signal using two parameter groups (the first parameter group and the second parameter group).

[0084] Also, in the above configuration, parameter .~ >

decoding section 120 partially decodes the first encoded information S12 inputted from first encoding section 115 and outputs each obtained parameter to second encoding section 130 corresponding to an upper layer of first 5 encoding section 115, and second encoding section 130 performs second encoding using each of these parameters and the input signal of speech encoding apparatus 100.
Byadoptingtheaboveconfiguration, whenthespeechsignal is hierarchically encoded, the speech encoding apparatus 10 according to the present embodiment can achieve efficient encoding while using CELP scheme speech encoding in an enhancement layer, and can obtain decoded signal with good quality. Further, it is not necessary for the first encoded information to be completely decoded, so that 15 it is possible to reduce the amount of process operations in encoding.

[0085] Moreover, in the above configuration, second encoding section 130 encodes, by CELP scheme speech encoding, a di f ference between an LSP obtained by a linear 20 prediction analysis on the speech signal that is the input of speech encoding apparatus 100 and a quantized LSP
generated by parameter decoding section 120. That is, second encoding section 130 takes a difference at the stage of the LSP parameter, andperforms CELP scheme speech 25 encodingonthisdifference,thereby achievingCELPscheme speech encoding that does not take a residual signal as an input.

[0086] Furthermore, in the above configuration, the second encoded information S14 outputted from (second encoding section 130 of) speech encoding apparatus 100 is a totally new signal not generated from any conventional speech encoding apparatus.

[0087] Next, supplemental description will be given to the operation of f irst encoding section 115 shown in FIG. 3.
[0088] Thefollowing describesprocessingofdetermining a first quantized LSP by LSP quantizing section 103 in first encoding section 115.

[0089] Here, description will be made with an example where 8 bits are assigned to the first quantized LSP code (L1), and the first LSP is vector-quantized.

[0090] LSP quantizing section 103 is provided with a first LSP codebook in which 256 variants of first LSP
code vectors [ lspl (L1 ')( i)] created in advance are s tored .
Here, L1' is an index attached to the first LDP code vector, and takes any value of 0 to 255. Also, lsp1(L1')(i) is an N-dimensional vector, and i takes a value from 0 to N-1.

[0091] A first LSP [a1 (i) I is inputted to LSP quantizing section 103 from LSP analyzing section 102. Here, a1 (i) is an N-dimensional vector, and i takes a value from 0 to N-1.

[0092] LSPquantizingsection103obtainsasquarederror erl between the first LSP [a1 (i) ] and the first LSP code vector [ 1sp1 ( L l 'min) ( i)] from the following (Equation 6).

. =

[Equation 61 er, =I (a,(i) -lspl''Y)(i)y ... (Equation 6) i=o Next, LSP quantizing section 103 obtains a squared error erl for all L1' to determine a value of L1' (L1'min) thatminimizes thesquarederrorerl. Then, LSPquantizing section103outputsthisdetermined L1'mintomultiplexing section 114 as a first quantized LSP code (L1) , and also outputs lspl(L1I min) (i) to synthesis filter 104 as a first quantized LSP.

[0093] As described above, lspl(L1'min) (i) obtained by LSP
quantizing section 103 is the first quantized LSP.
[0094] FIG.8 outlines processing of determining a first adaptive excitation lag by parameterdeterminingsection 113 in first encoding section 115.

[0095] In this figure, a buffer B1 is a buffer provided by adaptive excitation codebook 106, a position P1 is an extraction position of the first adaptive excitation vector, and a vector Vi is an extracted first adaptive excitation vector. Also, values 41 and 296 correspond tolowerandupperlimitsoftherangeofshiftingextraction position P1.

[0096] Assuming that that 8 bits are assigned to the code (Al) indicating the first adaptive excitation lag, the range of shifting the extraction position P1 is set in a range of length of 256 (=28) (for example, 41 to 296). However, the range of shifting the extraction position P1 can be arbitrarily set.

[0097] Parameter determining section 113 shifts the extraction position P1 within the set range, and sequentially indicates the extraction position P1 to adaptive excitation codebook 106.

[0098] Adaptive excitation codebook 106 extracts the first adaptive excitation vector Vl with a length of the frame by the extraction position P1 indicated from parameter determining section 113, and outputs the extracted f irst adaptive excitation vector to multiplier 109.

[0099] Parameter determining section 113 obtains the encoding distortion outputted from auditory weighting section 112 for all first adaptive excitation vectors Vl extracted from all extraction positions P1, and determines an extraction position P1 that minimizes the encoding distortion. Extraction position Pl f rom buffer obtained by parameter determining section 113 is the f irst adaptive excitation lag. Parameter determining section 113 outputs the first adaptive excitation lag code (Al) indicating the first adaptive excitation lag to multiplexing section 114.

[0100] FIG.9 outlines processing of determining afirst fixedexcitation vector by parameter determining section 113 in first encoding section 115. This f igure indicates the process of generating afirstfixed excitation vector from an algebraic fixed excitation codebook.

[0101] Track 1, track 2, and track 3 each generate one unit pulse (having an amplitude value of 1) . Also, multiplier 404, multiplier 405, andmultiplier 406assign polarity to the unit pulse generated by tracks 1 to 3.
Adder 407 adds the generated three unit pulses together, andvector408isafirstfixedexcitationvectorconsisting of three unit pulses.

[0102] Each track has different position where a unit pulse can be generated, and in this figure, the tracks are configured such that track 1 raises a unit pulse at any of eight positions {0, 3, 6, 9, 12, 15, 18, 21}, track 2 raises a unit pulse at any of eight positions {1, 4, 7, 10, 13, 16, 19, 22), and track 3 raises a unit pulse at any of eight positions {2, 5, 8, 11, 14, 17, 20, 2-3} .
[0103] Polarity is assigned to the generated unit pulse in each track by multipliers 404 to 406, respectively, the three unit pulses are added at adder 407, and first fixed excitation vector 408 resulting from the addition is formed.

[0104] In this example, unit pulse has eight patterns of positions and two patterns of position, positive and negative, and three bits for position information and one bit for polarity information are used to represent eachunitpulse. Therefore,thefixedexcitationcodebook has 12 bits in total.

[0105] Parameter determining section 113 shifts the generation positionofthethreeunitpulsesand polarity, and sequentially indicates the generation position and polarity to fixed excitation codebook 108.

5 [0106] Fixed excitation codebook 108 configures first fixed excitation vector 408 using the generation position and polarity indicated by parameter determining section 113, and outputs the configured first fixed excitation vector 408 to multiplier 110.

10 [0107] Parameter determining section 113 obtains an encoding distortion outputted from auditory weighting section 112 for all combinations of the generation positions and polarity, and determines a combination of the generation positions and polarity that minimizes the 15 encoding distortion. Next, parameter determining section 113 outputs the first fixed excitation vector code (Fl) indicating the combination of the generation positions. and polarity that minimizes the encoding distortion to multiplexing section 114.

20 [0108] Next,processingwillbedescribedwhereparameter determining section 113 in first encoding section 115 indicates quantized gain generating section 107 and determines a first quantized adaptive excitation gain and a first quantized fixed excitation gain. Here, 25 description will be made with an example where 8 bits are assigned to the first quantized excitation gain code (G1) .

[0109] Quantized gain generating section 107 is provided with a first excitation gain codebook inwhich 256 variants of first excitation gain code vectors [gain1(K1) (i)]
created in advance are stored. Here, K1' is an index attached to the first excitation gain code vector, and takes any value of 0 to 255. Also, gain1 (K1 ') ( i) is a two-dimensional vector, and i takes a value from 0 to 1.

[0110] Parameter determining section 113 sequentially indicates a value of K1' from 0 to 255 to quantized gain generating section 107. Quantized gain generating section 107 selects a first excitation gain code vector [gain1(K1) (i) ] from the first excitation gain codebook using K1'indicated by parameter determining section 113, outputs gain1 (K1 ' ) ( 0) to multiplier 109 as a first quantized adaptive excitation gain and gain1 (xl ' ) ( 1) to multiplier 110 as a first quantized fixed excitation gain.
[0111] As described above, gain1(K1' ) (0) obtained by quantized gain generatingsection107representsthefirst quantized adaptive excitation gain, and gain1(K1) (1) represents the first quantized fixed excitation gain.
[0112] Parameter determining section 113 obtains an encoding distortion outputted from auditory weighting section 112 for all K1' and determines a value of K1' (K1'min) that minimizes the encoding distortion. Next, parameter determining section 113 outputs K1'min to multiplexing section 114 as a first quantized excitation gain code (G1).

[0113] In the above, speech encoding apparatus 100 according to this embodiment has been described in detail.
[0114] Next, speech decoding apparatus 150 according to this embodiment will be described where the encoded information S12 and S14 transmitted from the above-configured speech encoding apparatus 100 are decoded.

[0115] As already shown in FIG.l,the mainconfigurations of speech decoding apparatus 150 are provided by first decoding section 160, second decoding section 180, signal control section 195, and demultiplexing section 155.
Sections of speech decoding apparatus 150 perform the following operations.

[0116] Demultiplexingsection155demultiplexesthe mode information and the encoded information multiplexed and outputted from speech encoding apparatus 100, and outputs thefirstencodedinformationSl2tofirstdecodingsection 160 when the mode information is "0" and 11111, the second encoded information S14 to second decoding section 180 when the mode information is "1". Also, demultiplexing section 155 outputs themode information to signal control section 195.

[0117] First decoding section 160 decodes the first encoded information S12 outputted from demultiplexing section 155 by using a CELP scheme speech decoding method (first decoding) , and outputs a first decoded signal S52 obtained by decodingtosignalcontrolsection195. Also, first decoding section 160 outputs the first parameter group S51 obtained in the decoding to second decoding section 180.

5[0118] Second decoding section 180 performs a second decoding process using the first parameter group S51 outputted from first decoding section 160, which will bedescribedlater, performsdecodingonthesecondencoded information S14 outputted from demultiplexing section 155, generates a second decoded signal S53 ancl.outputs the result to signal control section 195 [0119] Signalcontrolsection195inputsthefirstdecoded signal S52 outputted from first decoding section 160 and the second decoded signal S53 outputted from second decoding section 180, and outputs a decoded signal in accordance with the mode information outputted from demultiplexingsection155. Specifically, firstdecoded signal S5.2 is outputted as an output signal when the mode information is "0" and the second decoded signal S53 is outputted as an output signal when the mode information is "1".

[0120] FIG.10 is a block diagram showing an internal configuration of first decoding section 160.

[0121] Demultiplexing section 161 demultiplexes the first encoded information S12 inputted to first decoding section 160 into individual codes (L1, Al, G1, and Fl) , and outputs codes to each component. Specifically, the firstquantized LSP code (L1) demultiplexed f rom the f irst encoded information S12 is outputted to LSP decoding section 162, the first adaptive excitation lag code (Al) demultiplexed as well is outputted to adaptive excitation codebook 165, the first quantized excitation gain code (G1) demultiplexed as well is outputted to quantized gain generating section 166, and f irst f ixed excitation vector code (Fl) demultiplexed as well is outputted to fixed excitation codebook 167.

[0122] LSP decoding section 162 decodes the first quantized LSP code (Li) outputted from demultiplexing section 161 to a first quantized LSP, and outputs the decoded first quantized LSP to synthesis filter 163 and second encoding section 180.

[0123] Adaptive excitation codebook 165 extracts a set of samples for one frame from the buffer at an extraction position specified by the first adaptive excitation lag code (Al) outputted from demultiplexing section 161, and outputs the extracted vector to multiplier 168 as a first adaptive excitation vector. Also, adaptive excitation codebook 165 outputs the extraction position specified by the first adaptive excitation lag code (A1) to second decoding section 180 as a first adaptive excitation lag.
[0124] Quantized gain generating section 166 decodes the first quantized adaptiveexcitation gain andthefirst quantized fixed excitation gain specified by the first quantized excitation gain code (Gl) outputted from demultiplexing section 161. Then, quantized gain generating section 166 outputs the obtained first quantized adaptive excitation gain to multiplier 168 and second decoding section 180, and also the f irst quantized 5 fixedexcitation gaintomultiplier169andseconddecoding section 180.

[0125] Fixed excitation codebook 167 generates a first fixed excitation vector specified by the first fixed excitation vectorcode(F1)outputtedfrom demultiplexing 10 section 161, and outputs the vector to multiplier 169 and second decoding section 180.

[0126] Multiplier 168 multiplies the first adaptive excitation vector by the first quantized adaptive excitation gain, and outputs the result to adder 170.

15 Multiplier 169 multiplies the first fixed excitation vector by the first quantized fixed excitation gain, and outputs the result to adder 170. Adder 170 adds the first adaptive excitation vector and the first fixed excitation vector after gain multiplication outputted from 20 multipliers 168 and 169, generates a driving excitation, and outputs the generated driving excitation to synthesis filter 163 and adaptive excitation codebook 165.
[0127] Synthesis filter 163 performs filer synthesis using the driving excitation outputted from adder 170 25 andthefiltercoefficientdecoded by LSP decoding section 162, and outputs a synthesis signal to postprocessing section 164.

[0128] Postprocessing section 164 processes the synthesis signal outputted from synthesis filter 163 by performing processing for improving a subjective speech quality,such asformantemphasizingorpitch emphasizing, and by performing processing for improving a subjective stationary, noisequality,andoutputstheprocessedresult as a first decoded signal S52.

[0129] Here, the reproduced parameters are outputted to second decoding section 180 as the first parameter group S51.

[0130] FIG.11 is a block diagram showing an internal configuration of second decoding section 180.

[0131] Demultiplexing section 181 demultiplexes the secondencodedinformationSl4inputtedtosecond decoding section 180 into individual codes (L2, A2, G2, and F2) , and outputs codes to each component. Specifically, the second quantized LSP code (L2) demultiplexed from the secondencodedinformationSl4isoutputtedtoLSPdecoding section 182, the second adaptive excitation lag code (A2) demultiplexed as well is outputted to adaptive excitationcodebook185,the second quantized excitation gain code (G2) demultiplexed as well is outputted to quantized gain generating section 186, and the second fixed excitation vector code (F2) demultiplexed as well is outputted to fixed excitation codebook 187.

[0132] LSP decoding section 182 decodes the second quantized LSP code (L2) outputted from demultiplexing section 181 to a quantized residual LSP, adds the quantized residual LSP and the first quantized LSP outputted from first decoding section 160, and outputs a second quantized LSP resulting from the addition to synthesis filter 183.

[0133] Adaptive excitation codebook 185 extracts a set of samples for one frame from.the buffer at an extraction position specified by the first adaptive excitation lag outputted from first decoding section 160 and the second adaptive excitation lag code (Al) outputted from demultiplexing section 181, and outputs the extracted vector to multiplier 188 as a second adaptive excitation vector.

[0134] Quantized gain generating section 186 obtains a second quantized adaptive excitation gain and a second quantizedfixedexcitation gain using the f irst quantized adaptive excitation gain and the first quantized fixed excitation gain outputted from first decoding section 160 and the second quantized excitation gain code (G2) outputted from demultiplexing section 181, and outputs the second quantized adaptive excitation gain to multiplier 188 and the second quantized fixed excitation gain to multiplier 189.

[0135] Fixedexcitationcodebook187generatesa residual fixed excitation vector specified by the second fixed excitation vector code (F2) outputted f romdemultiplexing section181,addsthegeneratedresidualfixedexcitation vector and the first fixed excitation vector outputted from first decoding section 160, and outputs a second fixed excitation vector resulted from the addition to multiplier 189.

[0136] Multiplier 188 multiplies the second adaptive excitation vector by the second quantized adaptive excitation gain, and outputs the result to adder 190.
Multiplier 189 multiplies the second fixed excitation vector by the second quantized fixed excitation gain, and outputs the result to adder 190. Adder 190 generates a driving excitation by adding the second adaptive excitation vector gain multiplied by multiplier 188 and the second fixed excitation vector gain multiplied by multiplier 189, and outputs the generated driving excitationtosynthe.sis filter183 andadaptiveexcitation codebook 185.

[0137] Synthesis filter 183 performs filer synthesis using the driving excitation outputted from adder 190 and a filter coefficient decoded by LSP decoding section 182, and outputs a synthesis signal to postprocessing section 184.

[0138] Postprocessing section 184 processes the synthesis signal outputted from synthesis filter 183 by performing processing for improving a subjective speech quality, suchasformantemphasizingorpitchemphasizing, and by performing for improving a subjective stationary noise quality, andoutputs theprocessed signal as a second decoded signal S53.

[0139] In the above, speech decoding apparatus 150 has been described in detail.

[0140] As described above, according to the speech decoding apparatus of this embodiment, the first decoded signalisgeneratedfromthefirstparametergroupobtained by decoding the first encoded information, the second decoded signal is generated from the second parameter group obtained by decoding the second encoded information and the first parameter group, and thereby these signals can be obtained as output signals. Also, when only the f irst encoded information is used, by generating the f irst decoded signal from the first parameter group obtained by decoding the first encoded information, this signal can be obtained as an output signal. That is, by adopting a configuration capable of obtaining an output signal using all or part of the encoded information, a function capable of decoding speech/sound even f rompart ofencoded information (hierarchical encoding) can be implemented.
[0141] Also, in the above configuration, first decoding section 160 performs decoding on the first encoded informationS12and also outputs the f irst parameter group S51 obtained in this decoding to second decoding section 180, and second decoding section 180 decode the second encoded information S14 using this first parameter group S51. By adoptingthisconfiguration,thespeech decoding apparatus according to this embodiment can decode a signal hierarchically encoded by the speech encoding apparatus according to the present invention.

[0142] Here, in this embodiment, a case has been described as an example where parameter decoding section 120 demultiplexes individual codes (L1, Al, G1, and Fl) from 5 the first encoded information S12 outputted from first encoding section 115, but the multiplexing and demultiplexing procedure may be omitted by directly inputting each of the codes from first encoding section 115 to parameter decoding section 120.

10 [0143] Also, in this embodiment, a case has been described as an example where, in speech encoding apparatus 100, the first fixed excitation vector generated by fixed excitation codebook 108 and the second fixed excitation vector generated by fixed excitation codebook 138 are 15 formed by pulses, but vectors may be formed by spread pulses.

[0144] Further, in this embodiment, a case has been described with an example of hierarchical encoding of two layers, but the number of layers is not restricted 20 to this, and the number of layers may be three or more.
[0145] (Embodiment 2) FIG.12A is a block diagram showing a configuration of speech/sound transmitting apparatus according to 25 Embodiment 2 having incorporated therein speech encoding apparatus 100 described in Embodiment 1.

[0146] Speech/sound signal 1001 is converted by input apparatus 1002 into an electrical signal, and outputted to A/D converting apparatus 1003. A/D converting apparatus 1003 converts a (analog) signal outputted from input apparatus 1002 into a digital signal, and outputs the digital signal to speech/sound encoding apparatus 1004. Speech/soundencoding apparatus 10 04 incorporates speech encoding apparatus 100 shown in FIG.1, encodes the digital speech/sound signal outputted from A/D
converting apparatus 1003 and outputs the encoded information to RF modulating apparatus 1005. RF
modulatingapparatus1005convertstheencodedinformation outputted from speech/sound encoding apparatus 1004 to a signal to transmit on a propagation medium, such as aradiowave, andoutputs the signal to transmissionantenna 1006. Transmission antenna 1006 transmits the output signal outputted from RF modulating apparatus 1005 as a radio wave (RF signal). RF signal 107 in the figure represents a radio wave (RF signal) sent f rom transmission antenna 1006.

[0147] Theaboveoutlinestheconfiguration andoperation of the speech/sound signal transmitting apparatus.
[0148] FIG.12B is a block diagram showing the configuration of a speech/sound receiving apparatus according to Embodiment 2 having incorporated therein speech decoding apparatus 150 described in Embodiment 1.

[0149] RF signal 1008 is received by reception antenna 1009 and output to RF demodulating apparatus 1010. In the figure, RF signal 1008 represents the radio wave received by reception antenna 1009, and is identical to RF signal 1007, unless the signal is attenuated or noise is superimposed on it in a propagation path.

[0150] RF demodulating apparatus 1010 demodulates the RFsignaloutputtedfromreception antenna1009intoencode information, and outputs the encoded information to speech/sound decoding apparatus 1011. Speech/sound decoding apparatus 1011 incorporates speech decoding apparatus 150 shown in FIG.1, decodes the speech/sound signal from the encoded information outputted from RF
demodulating apparatus 1010, and outputs the encoded information to D/A converting apparatus 1012. D/A

converting apparatus 1012 converts the digital speech/soundsignaloutputtedfromspeech/sound decoding apparatus 1011 into an analog electrical signal, and outputs the signal to output apparatus 1013. Output apparatus 1013 converts the electrical signal into air vibration, and outputs it as acoustic waves that can be heard by human ears. In the figure, reference numeral 1014 indicates outputted acoustic wave.

[0151] Theaboveoutlinestheconfiguration andoperation of the speech/sound signal receiving apparatus.

[0152] By providing the above speech/sound signal transmitting apparatus and speech/ sound signal receiving apparatus in a base station apparatus and a communication terminal apparatus in a wireless communication system, high quality output signal can be obtained.

[0153] As described above, according to this embodiment, the speech encoding apparatus and speech decoding apparatus according to the present invention can be implemented in the speech/sound signal transmitting apparatusandthespeech/soundsignalreceivingapparatus.
[0154] (Embodiment 3) In Embodiment 1, a case has been described as an example in which the speech encoding method according to the present invention, that is, processing mainly performed by parameter decoding section 120 and second encoding section 130, is performed at the second layer.

However, the speech encoding method according to the present invention can be performed not only at the second layer butalsoatanotherenhancementlayer. Forexample, in the case of hierarchical encoding of three layers, the speech encoding method of the present invention may be performed at both the second layer and the third layer.
Such embodiment will be described below in detail.
[0155] FIG.13 is a block diagram showing the main configurationsofspeechencoding apparatus 300 andspeech decoding apparatus 350 according to Embodiment 3. Here, these speech encoding apparatus 300 and speech decoding apparatus 350 have a basic configuration similar to that of speech encoding apparatus 100 and speech decoding apparatus 150 shown in Embodiment 1. The same components are assigned the same reference numerals and the description thereof will be omitted.

[0156] First, speech encoding apparatus 300 will be described. The speech encoding apparatus 300 is further provided with second parameter decoding section 310 and third encoding section 3 2 0 in addi tion to the conf iguration of speech encoding apparatus 100 shown in Embodiment 1.
[0157] First parameter de.codingsection120 outputs the firstparameter group S13 obtained by parameter decoding to second encoding section 13 0 and third encoding section 320.

[0158] Second encoding section 130 obtains a second parameter group by a second encoding process, and outputs second encoded information S14 representing this second parameter group to multiplexing section 154 and second parameter decoding section 310.

[0159] Second parameter decoding section 310 performs parameter decoding, which is similar to that of first parameter decoding section 120, on the second encoded information S14 outputted from second encoding section 130. Specifically, second parameter decoding section 310 demultiplexes the second decoded information S14, and obtains a second quantized LSP code (L2 ), a second adaptive excitation lag code (A2), a second quantized excitation gain code (G2) , and a second fixed excitation vector code (F2), and obtains a second parameter group S21 from each of the obtained codes. The second parameter group S21 is outputted to third encoding section 320.
[0160] Third encoding section 320 performs a third encoding process using the input signal S11 of speech 5 encoding apparatus 300, the first parameter group S13 outputted from first parameter decoding section 120, and the second parameter group S21 outputted from second parameter decoding section 310, thereby obtaining athird parameter group, and outputs encoded information (third 10 encodedinformation)S22representingthisthirdparameter group to multiplexing section 154. The third parameter group is composed of, correspondingly to the first and second parameter groups, a third quantized LSP, a third adaptive excitation lag, a third f ixed excitation vector, 15 a third quantized adaptive excitation gain, and a third quantized fixed excitation gain.

[0161] The first encoded information is inputted to multiplexingsection154fromfirstencodingsection115, the second encoded information is inputted from second 20 encoding section 130, and the third encoded information is inputted from third encoding section 320. According to the mode information inputted to speech encoding apparatus 300, multiplexing section 154 multiplexes each piece of encoded information and mode information, and 25 generates multiplexed encoded information (multiplexed information) . For example, when the mode information is "0", multiplexing section 154 multiplexes the first encoded information and the mode information. When the mode information is "1", multiplexing section 154 multiplexes the first encoded information, the second encoded information, and the mode information. When the mode information is "2", multiplexing section 154 multiplexes the first encoded information, the second encoded information, the third encoded information, and the mode information. Next, multiplexing section 154 outputs the multiplexed information after multiplexing to speech decoding apparatus 350 via the transmission path N.

[0162] Next, speech decoding apparatus 350 will be described. The speech decoding apparatus 350 is further provided with third decoding section 360 in addition to the configuration of speech decoding apparatus 150 shown in Embodiment 1.

[0163] Demultiplexingsection155demultiplexesthe mode information and the encoded information outputted from speech encoding apparatus 300 after multiplexing, and outputsthefirstencodedinformationSl2tofirstdecoding section 160 when the mode information is "0", "1", or "2",thesecondencodedinformationSl4tosecond decoding section 180 when the mode information is "1" or 11211, and thethirdencodedinformationS22tothird decodingsection 360 when the mode information indicates "2".

[0164] First decoding section 160 outputs the first parameter group S51 obtained in the first decoding to second decoding section 180 and third decoding section 360.

[0165] Second decoding section 180 outputs the second parameter group S71 obtained in the second decoding to third decoding section 360.

[0166] Third decoding section 360 performs a third decoding process on the third encoded information S22 outputted fromdemultiplexingsection155usingthefirst parameter group S51 outputted f rom f irst decoding section 160 and the second parameter group S71 outputted from second decoding section 180. Third decoding section 360 outputs a third decoded signal S72 generated by this third decoding process to signal control section 195.

[0167] According to the mode inf ormation outputted f rom demultiplexing section 155, signal control section 195 outputs the first decoded signal S52, the second decoded signal S53, or the third decoded signal S72 as a decoded signal. Specifically, when the mode inf ormation is "0", the first decoded signal S52 is outputted. When the mode information is "111, the second decoded signal S53 is outputted. When the mode information is "2 ,,, the third decoded signal S72 is outputted.

[0168] As describedabove,accordingtothisembodiment, in hierarchical encoding with three layers, the speech encoding method according to the present invention can be implemented in both of the second layer and the third layer.

[0169] Here, thi s embodiment shows that, in hierarchical encoding with three layers, the speech encoding method according to the present invention is implemented in both of the second layer and the third layer, but the speech encoding method according to the present invention may be implemented only in the third layer.

[0170] The speech encoding apparatus and the speech decoding apparatus according to the present invention are not limited to the above Embodiments 1 to 3, and can be changed and implemented in various ways.

[0171] The speech encoding apparatus and the speech decoding apparatus according to the present invention can be incorporated in a communication terminal apparatus or a basestation apparatusin mobilecommunicationsystem or the like, thereby providing a communication terminal apparatus or a base station apparatus having operation effects similar to those described above.

[0172] Here, a case has been described as an example where the present invention is implemented with hardware.
However, the present invention can also be realized by software.

[0173] ThepresentapplicationisbasedonJapanesePatent Application No.2004-188755 filed on June 25, 2004, the entire contents of which is incorporated herein by reference.

Industrial Applicability [0174] Thespeechencodingapparatus,thespeech decoding apparatus, and the method thereof according to the present invention can be applied to a communication system or the like where a packet loss occurs depending on the state of a network, or a variable-rate communication system where a bit rate is varied according to the communication state, such as line capacity.

Claims (12)

1.

A speech encoding apparatus, comprising:

a first encoding section that generates, from a speech signal, encoded information by CELP scheme speech encoding;

a generating section that generates a parameter representing a feature of a generation model of the speech signal, which parameter is any of a quantized LSP (Line Spectral Pairs), an adaptive excitation lag, a fixed excitation vector,a quantized adaptive excitation gain, and a quantized fixed excitation gain from the encoded information; and a second encoding section that takes the speech signal as an input and encodes the input speech signal by CELP
scheme speech encoding using the parameter.

2. (Canceled)

3. The speech encoding apparatus according to claim 2, wherein the second encoding section sets a search range of an adaptive excitation codebook based on an adaptive excitation lag generated by the generating section.

4. The speech encoding apparatus according to claim 3, wherein the second encoding section encodes a difference between an adaptive excitation lag obtained by a search of the adaptive excitation codebook and the adaptive excitation lag generated by the generating section.

5. The speech encoding apparatus according to claim 2, wherein the second encoding section adds a fixed excitation vector generated by the generating section to a fixed excitation vector generated from a fixed excitation codebook and encodes a fixed excitation vector obtained by the addition.

6. The speech encoding apparatus according to claim 5, wherein the second encoding section performs the addition by weighting the fixed excitation vector generated by the generating section more than the fixed excitation vector generated from the fixed excitation codebook.

7. The speech encoding apparatus according to claim 2, wherein the second encoding section encodes a difference between an LSP obtained by a linear prediction analysis on the speech signal and a quantized LSP generated by the generating section.

8. The speech encoding apparatus according to claim 1, further comprising a multiplexing section that multiplexes one or both of the encoded information generated by the first section and the encoded information generated by the second section with the mode information, and outputs information.

9. A speech decoding apparatus corresponding to the speech encoding apparatus according to claim 1, comprising:

a first decoding section that decodes the encoded information generated by the first encoding section; and a second decoding section that decodes the encoded information generated in the second encoding step using the parameter generated in the first decoding step, which parameter represents the feature of the generation model of the speech signal.

10. A speech decoding apparatus corresponding to the speech encoding apparatus according to claim 8, comprising:

a first decoding section that decodes the encoded information generated by the first encoding section;
a second decoding section that decodes the encoded information generated in the second encoding step using the parameter generated in the first decoding step, which parameter represents the feature of the generation model of the speech signal; and an output section that outputs a signal obtained by decoding by either one of the first and second decoding sections according to the mode information.

11. A speech encoding method comprising:

a first encoding step of generating, from a speech signal, encoded information by CELP scheme speech encoding;

a generating step of generating, from the encoded information, a parameter representing a feature of a generation model of the speech signal; and a second encoding step of encoding the speech signal by CELP scheme speech encoding using the parameter.

12. A speech decoding method corresponding to the speech encoding method according to claim 11, comprising:

a first decoding step of decoding the encoded information generated in the first encoding step;

a second decoding step of decoding the encoded information generated in the second encoding step using the parameter generated in the first decoding step, which parameter represents the feature of the generation model of the speech signal.