decode.c

MPEG 2的音频编码软件。喜欢多媒体的开发人员可以看看。

        else {                  /* for no sample transmitted */
            for (k=0;k<3;k++) sample[j][k][i] = 0;
        }
        if(stereo == 2 && i>= jsbound) /* joint stereo : copy L to R */
            for (k=0;k<3;k++) sample[1][k][i] = sample[0][k][i];
    }
    for (i=sblimit;i<SBLIMIT;i++) for (j=0;j<stereo;j++) for (k=0;k<3;k++)
        sample[j][k][i] = 0;
}      

/**************************************************************
/*
/*   Restore the compressed sample to a factional number.
/*   first complement the MSB of the sample
/*    for layer I :
/*    Use s = (s' + 2^(-nb+1) ) * 2^nb / (2^nb-1)
/*   for Layer II :
/*   Use the formula s = s' * c + d
/*
/**************************************************************/

static double c[17] = { 1.33333333333, 1.60000000000, 1.14285714286,
                        1.77777777777, 1.06666666666, 1.03225806452,
                        1.01587301587, 1.00787401575, 1.00392156863,
                        1.00195694716, 1.00097751711, 1.00048851979,
                        1.00024420024, 1.00012208522, 1.00006103888,
                        1.00003051851, 1.00001525902 };

static double d[17] = { 0.500000000, 0.500000000, 0.250000000, 0.500000000,
                        0.125000000, 0.062500000, 0.031250000, 0.015625000,
                        0.007812500, 0.003906250, 0.001953125, 0.0009765625,
                        0.00048828125, 0.00024414063, 0.00012207031,
                        0.00006103516, 0.00003051758 };

/************************** Layer II stuff ************************/

void II_dequantize_sample(sample, bit_alloc, fraction, fr_ps)
unsigned int FAR sample[2][3][SBLIMIT];
unsigned int bit_alloc[2][SBLIMIT];
double FAR fraction[2][3][SBLIMIT];
frame_params *fr_ps;
{
    int i, j, k, x;
    int stereo = fr_ps->stereo;
    int sblimit = fr_ps->sblimit;
    al_table *alloc = fr_ps->alloc;

    for (i=0;i<sblimit;i++)  for (j=0;j<3;j++) for (k=0;k<stereo;k++)
        if (bit_alloc[k][i]) {

            /* locate MSB in the sample */
            x = 0;
#ifndef MS_DOS
            while ((1L<<x) < (*alloc)[i][bit_alloc[k][i]].steps) x++;
#else
            /* microsoft C thinks an int is a short */
            while (( (unsigned long) (1L<<(long)x) <
                    (unsigned long)( (*alloc)[i][bit_alloc[k][i]].steps)
                    ) && ( x < 16) ) x++;
#endif

            /* MSB inversion */
            if (((sample[k][j][i] >> x-1) & 1) == 1)
                fraction[k][j][i] = 0.0;
            else  fraction[k][j][i] = -1.0;

            /* Form a 2's complement sample */
            fraction[k][j][i] += (double) (sample[k][j][i] & ((1<<x-1)-1)) /
                (double) (1L<<x-1);

            /* Dequantize the sample */
            fraction[k][j][i] += d[(*alloc)[i][bit_alloc[k][i]].quant];
            fraction[k][j][i] *= c[(*alloc)[i][bit_alloc[k][i]].quant];
        }
        else fraction[k][j][i] = 0.0;   
   
    for (i=sblimit;i<SBLIMIT;i++) for (j=0;j<3;j++) for(k=0;k<stereo;k++)
        fraction[k][j][i] = 0.0;
}

/***************************** Layer I stuff ***********************/

void I_dequantize_sample(sample, fraction, bit_alloc, fr_ps)
unsigned int FAR sample[2][3][SBLIMIT];
unsigned int bit_alloc[2][SBLIMIT];
double FAR fraction[2][3][SBLIMIT];
frame_params *fr_ps;
{
    int i, nb, k;
    int stereo = fr_ps->stereo;
    int sblimit = fr_ps->sblimit;

    for (i=0;i<SBLIMIT;i++)
        for (k=0;k<stereo;k++)
            if (bit_alloc[k][i]) {
                nb = bit_alloc[k][i] + 1;
                if (((sample[k][0][i] >> nb-1) & 1) == 1) fraction[k][0][i] = 0.0;
                else fraction[k][0][i] = -1.0;
                fraction[k][0][i] += (double) (sample[k][0][i] & ((1<<nb-1)-1)) /
                    (double) (1L<<nb-1);

                fraction[k][0][i] =
                    (double) (fraction[k][0][i]+1.0/(double)(1L<<nb-1)) *
                        (double) (1L<<nb) / (double) ((1L<<nb)-1);
            }
            else fraction[k][0][i] = 0.0;
}

/************************************************************
/*
/*   Restore the original value of the sample ie multiply
/*    the fraction value by its scalefactor.
/*
/************************************************************/

/************************* Layer II Stuff **********************/

void II_denormalize_sample(fraction, scale_index,fr_ps,x)
double FAR fraction[2][3][SBLIMIT];
unsigned int scale_index[2][3][SBLIMIT];
frame_params *fr_ps;
int x;
{
    int i,j,k;
    int stereo = fr_ps->stereo;
    int sblimit = fr_ps->sblimit;

    for (i=0;i<sblimit;i++) for (j=0;j<stereo;j++) {
        fraction[j][0][i] *= multiple[scale_index[j][x][i]];
        fraction[j][1][i] *= multiple[scale_index[j][x][i]];
        fraction[j][2][i] *= multiple[scale_index[j][x][i]];
    }
}

/**************************** Layer I stuff ******************************/

void I_denormalize_sample(fraction,scale_index,fr_ps)
double FAR fraction[2][3][SBLIMIT];
unsigned int scale_index[2][3][SBLIMIT];
frame_params *fr_ps;
{
    int i,j,k;
    int stereo = fr_ps->stereo;
    int sblimit = fr_ps->sblimit;

    for (i=0;i<SBLIMIT;i++) for (j=0;j<stereo;j++)
        fraction[j][0][i] *= multiple[scale_index[j][0][i]];
}

/*****************************************************************
/*
/* The following are the subband synthesis routines. They apply
/* to both layer I and layer II stereo or mono. The user has to
/* decide what parameters are to be passed to the routines.
/*
/***************************************************************/

/*************************************************************
/*
/*   Pass the subband sample through the synthesis window
/*
/**************************************************************/

/* create in synthesis filter */

void create_syn_filter(filter)
double FAR filter[64][SBLIMIT];
{
    register int i,k;

    for (i=0; i<64; i++)
        for (k=0; k<32; k++) {
            if ((filter[i][k] = 1e9*cos((double)((PI64*i+PI4)*(2*k+1)))) >= 0)
                modf(filter[i][k]+0.5, &filter[i][k]);
            else
                modf(filter[i][k]-0.5, &filter[i][k]);
            filter[i][k] *= 1e-9;
        }
}

/***************************************************************
/*
/*   Window the restored sample
/*
/***************************************************************/

/* read in synthesis window */

void read_syn_window(window)
double FAR window[HAN_SIZE];
{
    int i,j[4];
    FILE *fp;
    double f[4];
    char t[150];

    if (!(fp = OpenTableFile("dewindow") )) {
        printf("Please check synthesis window table 'dewindow'\n");
        exit(1);
    }
    for (i=0;i<512;i+=4) {
        fgets(t, 150, fp);
        sscanf(t,"D[%d] = %lf D[%d] = %lf D[%d] = %lf D[%d] = %lf\n",
               j, f,j+1,f+1,j+2,f+2,j+3,f+3);
        if (i==j[0]) {
            window[i] = f[0];
            window[i+1] = f[1];
            window[i+2] = f[2];
            window[i+3] = f[3];
        }
        else {
            printf("Check index in synthesis window table\n");
            exit(1);
        }
        fgets(t,150,fp);
    }
    fclose(fp);
}

int SubBandSynthesis (bandPtr, channel, samples)
double *bandPtr;
int channel;
short *samples;
{
    register int i,j,k;
    register double *bufOffsetPtr, sum;
    register long foo;
    static int init = 1;
    typedef double NN[64][32];
    static NN FAR *filter;
    typedef double BB[2][2*HAN_SIZE];
    static BB FAR *buf;
    static int bufOffset[2] = {64,64};
    static double FAR *window;
    int clip = 0;               /* count & return how many samples clipped */

    if (init) {
        buf = (BB FAR *) mem_alloc(sizeof(BB),"BB");
        filter = (NN FAR *) mem_alloc(sizeof(NN), "NN");
        create_syn_filter(*filter);
        window = (double FAR *) mem_alloc(sizeof(double) * HAN_SIZE, "WIN");
        read_syn_window(window);
        init = 0;
    }
/*    if (channel == 0) */
    bufOffset[channel] = (bufOffset[channel] - 64) & 0x3ff;
    bufOffsetPtr = &((*buf)[channel][bufOffset[channel]]);

    for (i=0; i<64; i++) {
        sum = 0;
        for (k=0; k<32; k++)
            sum += bandPtr[k] * (*filter)[i][k];
        bufOffsetPtr[i] = sum;
    }
    /*  S(i,j) = D(j+32i) * U(j+32i+((i+1)>>1)*64)  */
    /*  samples(i,j) = MWindow(j+32i) * bufPtr(j+32i+((i+1)>>1)*64)  */
    for (j=0; j<32; j++) {
        sum = 0;
        for (i=0; i<16; i++) {
            k = j + (i<<5);
            sum += window[k] * (*buf) [channel] [( (k + ( ((i+1)>>1) <<6) ) +
                                                  bufOffset[channel]) & 0x3ff];
        }

/*      Casting truncates towards zero for both positive and negative numbers,
	the result is cross-over distortion,  1995-07-12 shn */

        if(sum > 0)
        {
          foo = (long)(sum * (double) SCALE + (double)0.5);
        }
        else
        {
           foo = (long)(sum * (double)SCALE -(double)0.5);
        } 

     if (foo >= (long) SCALE)      {samples[j] = (short)(SCALE-1); ++clip;}
     else if (foo < (long) -SCALE) {samples[j] = (short)(-SCALE);  ++clip;}
     else                           samples[j] =(short)foo;
    }
    return(clip);
}

void out_fifo(pcm_sample, num, fr_ps, done, outFile, psampFrames)
short FAR pcm_sample[2][SSLIMIT][SBLIMIT];
int num;
frame_params *fr_ps;
int done;
FILE *outFile;
unsigned long *psampFrames;
{
    int i,j,l;
    int stereo = fr_ps->stereo;
    int sblimit = fr_ps->sblimit;
    static short int outsamp[1600];
    static long k = 0;

    if (!done)
        for (i=0;i<num;i++) for (j=0;j<SBLIMIT;j++) {
            (*psampFrames)++;
            for (l=0;l<stereo;l++) {
                if (!(k%1600) && k) {
		    /*
		      Samples are big-endian. If this is a little-endian machine
		      we must swap
		      */
		    if ( NativeByteOrder == order_unknown )
		    {
			NativeByteOrder = DetermineByteOrder();
			if ( NativeByteOrder == order_unknown )
			{
			    fprintf( stderr, "byte order not determined\n" );
			    exit( 1 );
			}
		    }
		    if ( NativeByteOrder == order_littleEndian )
			SwapBytesInWords( outsamp, 1600 );
		    
                    fwrite(outsamp,2,1600,outFile);
                    k = 0;
                }
                outsamp[k++] = pcm_sample[l][i][j];
            }
        }
    else {
        fwrite(outsamp,2,(int)k,outFile);
        k = 0;
    }
}

void  buffer_CRC(bs, old_crc)
Bit_stream_struc  *bs;
unsigned int  *old_crc;
{
    *old_crc = getbits(bs, 16);
}

void  recover_CRC_error(pcm_sample, error_count, fr_ps, outFile, psampFrames)
short FAR pcm_sample[2][SSLIMIT][SBLIMIT];
int error_count;
frame_params *fr_ps;
FILE *outFile;
unsigned long *psampFrames;
{
    int  stereo = fr_ps->stereo;
    int  num, done, i;
    int  samplesPerFrame, samplesPerSlot;
    layer  *hdr = fr_ps->header;
    long  offset;
    short  *temp;

    num = 3;
    if (hdr->lay == 1) num = 1;

    samplesPerSlot = SBLIMIT * num * stereo;
    samplesPerFrame = samplesPerSlot * 32;

    if (error_count == 1) {     /* replicate previous error_free frame */
        done = 1;
        /* flush out fifo */
        out_fifo(pcm_sample, num, fr_ps, done, outFile, psampFrames);
        /* go back to the beginning of the previous frame */
        offset = sizeof(short int) * samplesPerFrame;