Code Rle Rpl

8/4/2019 Code Rle Rpl

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function R=AutoCorrelate(y,lenth1,Max)

% Function: AutoCorrelate

% This subprogram is for the autocorrelate function

% y: sample point of speech

R=zeros(1,lenth1);

[R(1),V,X]=MiddleShap(y,lenth1,Max); % middleshap

for j1=21:880

for i1=j1:lenth1

if (i1+j1)


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6,206,188,172,158,144,132,120,110,102,92,84,78,70,64,60,54,50,46,42,38,34,32,30,2

6,24,22,20,18,17,16,15,14,13,12,11,10,9,8,7,6,5,4,3,2,1,0];

ivtab

=[24960,24960,24960,24960,25480,25480,25483,25480,16640,1560,1560,1560,1664

0,1816,1563,1560,24960,24960,24859,24856,26001,25881,25915,25913,1560,1560,7800,3640,1561,1561,3643,3641];

corth=[32767.0, 32767.0, 32.0, 32.0, 32.0, 32.0, 16.0, 16.0;10.0, 8.0, 6.4, 6.4, 11.2,

11.2, 5.6, 5.6;5.0, 4.0, 3.2, 3.2, 6.4, 6.4, 3.2, 3.2;0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0];

detau=[0,0,0,3,0,3,3,31, 0,3,3,21,3,3,29,30,0,3,3,20,3,25,27,26,

3,23,58,22,3,24,28,3,0,3,3,3,3,39,33,32, 3,37,35,36,3,38,34,3,3,42,46,44,50,40,48,3,

54,3,56,3,52,3,3,1,0,3,3,108,3,78,100,104,

3,84,92,88,156,80,96,3,3,74,70,72,66,76,68,3,

62,3,60,3,64,3,3,1,3,116,132,112,148,152,3,3,

140,3,136,3,144,3,3,1,124,120,128,3,3,3,3,1, 3,3,3,1,3,1,1,1];

detab7=[4,11,18,25,32,39,46,53,60,66,72,77,82,87,92,96,101,104,108,111,114,115,11

7,119,121,122,123,124,125,126,127,127];

descl=[0.6953,0.6250,0.5781,0.5469,0.5312,0.5391,0.4688,0.3828];

deadd=[1152,-2816,-1536,-3584,-1280,-2432,768,-1920];

qb=[511,511,1023,1023,1023,1023,2047,4095];

nbit=[8,8,5,5,4,4,4,4,3,2];

zrc=[0,0,0,0,0,3,0,2,0,0];

abit=[2,4,8,16,32];

ivp2h=0;erate=0;

iovoic=0;

iavgp=60;

ethrs=2048;

ethrs1=128;

ethrs2=1024;

ethrs3=2048;

fut=1;

pres=1;

past=2;

first=1;

rcn=length(irc);

pitcha=0;

% If no error correction, do pitch and voicing then jump to decode

% i4 = detau(ipitch);

i4=ipitch;

% Do error correction pitch and voicing

if(i4>4)


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dpit(fut) = i4;

ivoic = 2;

iavgp =fix((15*iavgp+i4+8)*0.0625);

else

ivoic = i4;dpit(fut) = iavgp;

end

drms(fut)=irms;

for n1=1:rcn

drc(fut,n1) = irc(n1);

end

% Determine index to IVTAB from V/UV decision

% If error rate is high then use alternate table

index = 16*ivp2h + 4*iovoic + ivoic + 1;

i1 = ivtab(index);

ipit = bitand(i1,3);

icorf = fix(i1*0.125);

if erate


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irc(n1)= drc(pres,n1);

end

if ipit==1

dpit(pres) = dpit(past);

endif ipit==3

dpit(pres) = dpit(fut);

end

pitcha= dpit(pres);

% If bit 2 of ICORF is set then smooth RMS and RC's

if(bitand(icorf,abit(1))~=0)

if (abs(drms(pres)-drms(fut))>=corth(ixcor-1,1))&(abs(drms(res)-

drms(past))>= corth(ixcor-1,1))

temp1=[drms(past),drms(pres),drms(fut)];

irms=median(temp1);

end

for n1=1:6

if( abs(drc(pres,n1-1)-drc(fut,n1-1))>=corth(ixcor-

1,n1+1)&abs(drc(pres,n1-1)-drc(past,n1-1))>= corth(ixcor-1,n1+1))

temp1=[drc(past,n1-1),drc(pres,n1-1),drc(fut,n1-1)];

irc(n1)=median(temp1);

endend

end

% If bit 3 of ICORF is set then smooth pitch


if(abs(dpit(pres)-dpit(fut)) >= corth(ixcor-1,0)&abs(dpit(pres)-

dpit(past))>= corth(ixcor-1,0))

temp1=[dpit(past), dpit(pres), dpit(fut)];

pitch = median(temp1);

end

end

end

% If bit 5 of ICORF is set then RC(5) - RC(10) are loaded with

% values so that after quantization bias is removed in decode

% the values will be zero.

% end if first


for(n1=5:rcn)

irc(n1)=zrc(n1);end


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end

% House keeping - one frame delay

iovoic = ivoic;

ivp2h = voice(2);

dpit(past) = dpit(pres);dpit(pres) = dpit(fut);

drms(past) = drms(pres);

drms(pres) = drms(fut);

for(n1=1:rcn)

drc(past,n1) = drc(pres,n1);

drc(pres,n1) = drc(fut,n1);

end

% Decode RMS

irms = rmst((31-irms)*2);

% Decode RC(1) and RC(2) from log-area-ratios

% Protect from illegal coded value (-16) caused by bit errors

for(n1=1:2)

i2 = irc(n1);

i1 = 0;if(i215)

i2 = 0;

end

end

i2 = detab7(2*i2+1);

if(i1==1)

i2 = -i2;

end

ishift = 15 - nbit(n1);

irc(n1) = i2*bitshift((ishift-1),2);

end

% Decode RC(3)-RC(10) to sign plus 14 bits

for(n1=3:rcn)

i2 = irc(n1);ishift = 15 - nbit(n1-1);


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i2 = i2*bitshift((ishift-1),2);

i2 = i2+qb(n1-2);

ftemp = i2*descl(n1-2) + deadd(n1-2); % rounding problem

if(ftemp < 0)

ftemp = - ftemp;itemp = fix(ftemp);

irc(n1) = -itemp;

else

irc(n1) = fix(ftemp);

end

end

% Scale RMS and RC's to reals

rms=double(irms);

for(n1=1:rcn)

rc(n1)= double(irc(n1)* 6.103515625e-05); %16,384 = 2^14

end

function [ipitch,irms,irc]=encode1(pitcha,rms,rc)

% Parameters Encoding

% Quantize LPC parameters for transmission

% INPUTS:% VOICE - Half frame voicing decisions

% PITCH - Pitch

% RMS - Energy

% RC - Reflection coefficients

% CORRP - Error Correction: TRUE = yes, FALSE = none

% OUTPUTS:

% IPITCH - Coded pitch and voicing

% IRMS - Quantized energy

% IRC - Quantized reflection coefficients

enctab=[0,7,11,12,13,10,6,1,14,9,5,2,3,4,8,15];

entau=[19,11,27,25,29,21,23,22,30,14,15,7,39,38,46,42,43,41,45,37,53,49,51,50,

54,52,60,56,58,26,90,88,92,84,86,82,83,81,85,60,77,73,75,74,78,70,71,67,99,97,113,

112,114,98,106,104,108,100,101,76];

enadd=[1920,-768,2432,1280,3584,1536,2816,-1152];

enscl=[0.0204,0.0167,0.0145,0.0147,0.0143,0.0135,0.0125,0.0112];

enbits=[6,5,4,4,4,4,3,3];

entab6=[0,0,0,0,0,0,1,1,1,1,1,1,1,2,2,2,2,2,2,2,3,3,3,3,3,3,3,4,4,4,4,4,4,4,5,5,5,5,

5,6,6,6,6,6,7,7,7,7,7,8,8,8,8,9,9,9,10,10,11,11,12,13,14,15];rmst=[1024,936,856,784,718,656,600,550,502,460,420,384,352,328,294,270,24


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6,226,206,188,172,158,144,132,120,110,102,92,84,78,70,64,60,54,50,46,42,38,34,32,

30,26,24,22,20,18,17,16,15,14,13,12,11,10,9,8,7,6,5,4,3,2,1,0];

% Scale RMS and RC's to integers

rcn=length(rc);

irms=fix(rms);for n1=1:rcn

irc(n1)=fix(rc(n1)*32768);

end

% Unvoice, voice and pitch period encoding:

% Unvoice is 1and it will change to 0 after encoding.

% Voice is the value of the pitch periodusing Gray code to encode.

if pitcha~=1

ipitch=pitcha;

else

ipitch=0;

end

% RMS encoding

n2=32;

idel=16;

irms=min(irms,1023);

while idel>0

if irms>rmst(n2)n2=n2-idel;

end

if irmsrmst(n2)

n2=n2-1;

end

irms=fix(31-n2*0.5);

% Reflection coefficient encoding

% Use RC(1),RC(2) as an area of logarithm to encode

for n1=1:2

i2=irc(n1);

mrk=0;

if i2


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end

i2=fix(bitshift(i2,-9));

i2=min(i2,63);

i2=entab6(i2+1);

if mrk~=0;i2=-i2;

end

irc(n1)=i2;

end

% RC(3)-RC(10) linear encoding

for n1=3:rcn

i2=bitshift(irc(n1),-1);

if enadd(rcn-n1+1)


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% replacing RC(5) - RC(10).

if length(ipitch)==0|length(ipitch)==127

irc(5)=enctab(bitshift(bitand(irc(1),30),-1));


irc(7)=enctab(bitshift(bitand(irc(3),30),-1));irc(8)=enctab(bitshift(bitand(irms,30),-1));

irc(9)=bitshift(enctab(bitshift(bitand(irc(4),30),-1)),-1);


end

function [a,e]=lpc(x,N);

%LPC Linear Predictor Coefficients.

% A = LPC(X,N) finds the coefficients, A=[ 1 A(2) ... A(N+1) ],

% of an Nth order forward linear predictor

%

% Xp(n) = -A(2)*X(n-1) - A(3)*X(n-2) - ... - A(N+1)*X(n-N)

%

% such that the sum of the squares of the errors

%

% err(n) = X(n) - Xp(n)

%

% is minimized. X can be a vector or a matrix. If X is a matrix

% containing a separate signal in each column, LPC returns a model% estimate for each column in the rows of A. N specifies the order

% of the polynomial A(z).

%

% If you do not specify a value for N, LPC uses a default N = length(X)-1.

%

% [A,E] = LPC(X,N) returns the variance (power) of the prediction error.

%

% LPC uses the Levinson-Durbin recursion to solve the normal equations

% that arise from the least-squares formulation. This computation

% of the linear prediction coefficients is often referred to as the

% autocorrelation method.

%

% See also LEVINSON, ARYULE, PRONY, STMCB.

% Author(s): T. Krauss, 9-21-93

% Modified: T. Bryan 11-14-97

% Copyright 1988-2002 The MathWorks, Inc.

% $Revision: 1.11 $ $Date: 2002/03/28 17:28:47 $

error(nargchk(1,2,nargin))


11/15

[m,n] = size(x);

if (n>1)&(m==1)

x = x(:);

[m,n] = size(x);end

if nargin < 2, N = m-1; end

if (N > m-1),

% disp('Warning: zero-padding short input sequence')

x(N+1,:)=zeros(1,n);

end

% Compute autocorrelation vector or matrix

X = fft(x,2^nextpow2(2*size(x,1)-1));

R = ifft(abs(X).^2);

R = R./(m-1); % Biased autocorrelation estimate

[a,e] = levinson(R,N);

%-----------------------------------------------------------------

% FINAL PROJECT%

% Digital Signal Processing of Audio Signals for Speech Coding

%

%

% Date: April 2004

%-----------------------------------------------------------------

% Original Speech Analysis

[wavebuf,fs,nbits]=wavread('speech'); % read speech file

wavplay(wavebuf,fs) % produce speech

plot(wavebuf); % plot waveform

lenth=length(wavebuf); % time domain analysis

t=(0:lenth-1)/fs;

figure,

plot(t,wavebuf); % produce waveform with correct scale in

time domain

title('time domain analysis');

xlabel('time (s)');ylabel('amplitude');


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a=fft(wavebuf,2*lenth); % frequency domain analysis

b=20*log10(abs(a));

f=(0:lenth-1)/(lenth-1)*(fs/2);

figure,plot(f,b(1:lenth)); % produce waveform with correct scale in

frequency domain

title('frequency domain analysis');

xlabel('frequency (Hz)');

ylabel('magnitude (dB)');

w=kaiser(256,5); % Mixed time-frequency domain analysis

figure,

specgram(wavebuf,512,fs,w,220); % produce the specgram

% Pitch Detect

pitch=pitchesdetect(wavebuf,fs);

% Linear Predictive System (10 orders)

pitchk=length(pitch);

for k1=1:pitchk

x=wavebuf((k1-1)*300+1:k1*300+600);

x=filter(1,[1 1/2 1/3 1/4 1/5, 1/6 1/7 1/8 1/9 1/10],x); %filter

rms=0; % calculate root mean squared

for k2=1:130

temp=1-0.9375*(x(k2+385+1))^(-1);

rms=rms+(temp^2);

end

rmsn(k1)=(rms/130)^(1/2);

lpcn(k1,1:11)=lpc(x,10); % linear predictive coefficients

end

lpcn=real(lpcn);

% Parameters encoding

pitchk=length(pitch);

for k1=1:pitchk

[kk,r0]=ac2rc(lpcn(k1,1:11));

pitcha=pitch(k1);

[ipitch,irms,irc]=encode1(pitcha,rms,kk);

% Parameters decodingrcn=length(kk);


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[voice,pitcha,rms,kk]=decode1(ipitch,irms,irc,rcn);

pitch(k1)=pitcha;

lpca=rc2ac(kk,r0);

lpcn(k1,1:11)=lpca';

rms(k1)=rms;end

waveoutbuf=filter(1,[1 1/2 1/3 1/4 1/5, 1/6 1/7 1/8 1/9 1/10],wavebuf); % filter

wavwrite(waveoutbuf,fs,nbits,'newspeech');

[wavebuf,fs,nbits]=wavread('newspeech'); % read speech file

wavplay(wavebuf); % produce speech

plot(wavebuf); % plot waveform

function [R0,V,X]=MiddleShap(y,lenth2,Max);

% Function: MiddleShap

% This subprogram is used to middleshap.

RO=0;

for i1=1:lenth2

if abs(y(i1))0

V(i1)=1;

X(i1)=y(i1)-Max;

RO=RO+X(i1);

else

V(i1)=-1;

X(i1)=y(i1)+Max;

RO=RO-X(i1);

end

end

function y=myFilter(wavebuf,i1,lngth)

% Filter Function Design

% This subprogram is to design the filter function.

% wavebuf: sample point of speech


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y=zeros(1,900); % produce a matrix

H=[0.0035270585,-0.0075853243,-0.022130724,-0.037701912,-0.040792551,-

0.017618544,0.037134223,0.1139423,0.18955371,0.23657782,0.23657782,0.189537

1,0.1139423,0.037134223,-0.017618544,-0.040792551,-0.037701912,-0.022130724,-

0.0075853243,0.0035270585]; % filter parametersfor ii1=10:(lngth-11)

for j1=1:20

y(ii1)=y(ii1)+wavebuf(ii1+i1-j1+11)*H(j1); % filter

end

end

function pitchgray=pitchcode(pitcha)

% Gray code

% This program is for Gray code.

for aa=1:20

pitchm(aa)=aa+19;

end

for aa=21:40

pitchm(aa)=(aa-1)*2;

end

for aa=41:60pitchm(aa)=(aa-41)*4+80;

end

graym=[19,11,27,25,29,21,23,22,30,14,15,7,39,38,46,42,43,41,45,37,53,49,51,5

0,54,52,60,56,58,26,90,88,92,84,86,82,83,81,85,69,77,73,75,74,78,70,71,67,99,97,11

3,112,114,98,106,104,108,100,101,76];

for aa=1:60

if pitcha==pitchm(aa);

pitchgray=graym(aa);

else pitchgray=0;

end

end

function n=PitchDetect(y)

% Function: PitchDetect

% This subprogram is also for detecting the pitch.% y: sample point of speech


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% Determined the threshold of middleshap

Maxhead=max(abs(y(1:300)));

Maxtail=max(abs(y(301:900)));

Maxy=0.68*min(Maxhead,Maxtail);

% Autocorrelate function

R=AutoCorrelate(y,900,Maxy);

[Maxy,n]=max(R(20:880)); % calculate the maximum and subscription of

array

n=n+20;

if Maxy

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