*/
#include "avcodec.h"
+#define ALT_BITSTREAM_READER_LE
+#include "get_bits.h"
#include "ra288.h"
+#include "lpc.h"
+#include "celp_math.h"
+#include "celp_filters.h"
typedef struct {
- float history[8];
- float output[40];
- float pr1[36];
- float pr2[10];
- int phase, phasep;
-
- float st1a[111],st1b[37],st1[37];
- float st2a[38],st2b[11],st2[11];
- float sb[41];
- float lhist[10];
-} Real288_internal;
-
-static int ra288_decode_init(AVCodecContext * avctx)
-{
- Real288_internal *glob=avctx->priv_data;
- memset(glob,0,sizeof(Real288_internal));
- return 0;
-}
+ float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A)
+ float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB)
+
+ /** speech data history (spec: SB).
+ * Its first 70 coefficients are updated only at backward filtering.
+ */
+ float sp_hist[111];
+
+ /// speech part of the gain autocorrelation (spec: REXP)
+ float sp_rec[37];
-static void prodsum(float *tgt, float *src, int len, int n);
-static void co(int n, int i, int j, float *in, float *out, float *st1, float *st2, const float *table);
-static int pred(float *in, float *tgt, int n);
-static void colmult(float *tgt, float *m1, const float *m2, int n);
+ /** log-gain history (spec: SBLG).
+ * Its first 28 coefficients are updated only at backward filtering.
+ */
+ float gain_hist[38];
+ /// recursive part of the gain autocorrelation (spec: REXPLG)
+ float gain_rec[11];
+} RA288Context;
-/* initial decode */
-static void unpack(unsigned short *tgt, unsigned char *src, unsigned int len)
+static av_cold int ra288_decode_init(AVCodecContext *avctx)
{
- int x,y,z;
- int n,temp;
- int buffer[len];
-
- for (x=0;x<len;tgt[x++]=0)
- buffer[x]=9+(x&1);
-
- for (x=y=z=0;x<len/*was 38*/;x++) {
- n=buffer[y]-z;
- temp=src[x];
- if (n<8) temp&=255>>(8-n);
- tgt[y]+=temp<<z;
- if (n<=8) {
- tgt[++y]+=src[x]>>n;
- z=8-n;
- } else z+=8;
- }
+ avctx->sample_fmt = SAMPLE_FMT_FLT;
+ return 0;
}
-static void update(Real288_internal *glob)
+static void apply_window(float *tgt, const float *m1, const float *m2, int n)
{
- int x,y;
- float buffer1[40],temp1[37];
- float buffer2[8],temp2[11];
-
- for (x=0,y=glob->phasep+5;x<40;buffer1[x++]=glob->output[(y++)%40]);
- co(36,40,35,buffer1,temp1,glob->st1a,glob->st1b,table1);
- if (pred(temp1,glob->st1,36))
- colmult(glob->pr1,glob->st1,table1a,36);
-
- for (x=0,y=glob->phase+1;x<8;buffer2[x++]=glob->history[(y++)%8]);
- co(10,8,20,buffer2,temp2,glob->st2a,glob->st2b,table2);
- if (pred(temp2,glob->st2,10))
- colmult(glob->pr2,glob->st2,table2a,10);
+ while (n--)
+ *tgt++ = *m1++ * *m2++;
}
-/* Decode and produce output */
-static void decode(Real288_internal *glob, unsigned int input)
+static void convolve(float *tgt, const float *src, int len, int n)
{
- unsigned int x,y;
- float f;
- double sum,sumsum;
- float *p1,*p2;
- float buffer[5];
- const float *table;
-
- for (x=36;x--;glob->sb[x+5]=glob->sb[x]);
- for (x=5;x--;) {
- p1=glob->sb+x;p2=glob->pr1;
- for (sum=0,y=36;y--;sum-=(*(++p1))*(*(p2++)));
- glob->sb[x]=sum;
- }
-
- f=amptable[input&7];
- table=codetable+(input>>3)*5;
-
- /* convert log and do rms */
- for (sum=32,x=10;x--;sum-=glob->pr2[x]*glob->lhist[x]);
- if (sum<0) sum=0; else if (sum>60) sum=60;
-
- sumsum=exp(sum*0.1151292546497)*f; /* pow(10.0,sum/20)*f */
- for (sum=0,x=5;x--;) { buffer[x]=table[x]*sumsum; sum+=buffer[x]*buffer[x]; }
- if ((sum/=5)<1) sum=1;
-
- /* shift and store */
- for (x=10;--x;glob->lhist[x]=glob->lhist[x-1]);
- *glob->lhist=glob->history[glob->phase]=10*log10(sum)-32;
-
- for (x=1;x<5;x++) for (y=x;y--;buffer[x]-=glob->pr1[x-y-1]*buffer[y]);
-
- /* output */
- for (x=0;x<5;x++) {
- f=glob->sb[4-x]+buffer[x];
- if (f>4095) f=4095; else if (f<-4095) f=-4095;
- glob->output[glob->phasep+x]=glob->sb[4-x]=f;
- }
+ for (; n >= 0; n--)
+ tgt[n] = ff_dot_productf(src, src - n, len);
+
}
-/* column multiply */
-static void colmult(float *tgt, float *m1, const float *m2, int n)
+static void decode(RA288Context *ractx, float gain, int cb_coef)
{
- while (n--)
- *(tgt++)=(*(m1++))*(*(m2++));
+ int i;
+ double sumsum;
+ float sum, buffer[5];
+ float *block = ractx->sp_hist + 70 + 36; // current block
+ float *gain_block = ractx->gain_hist + 28;
+
+ memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
+
+ /* block 46 of G.728 spec */
+ sum = 32.;
+ for (i=0; i < 10; i++)
+ sum -= gain_block[9-i] * ractx->gain_lpc[i];
+
+ /* block 47 of G.728 spec */
+ sum = av_clipf(sum, 0, 60);
+
+ /* block 48 of G.728 spec */
+ /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
+ sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
+
+ for (i=0; i < 5; i++)
+ buffer[i] = codetable[cb_coef][i] * sumsum;
+
+ sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.);
+
+ sum = FFMAX(sum, 1);
+
+ /* shift and store */
+ memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
+
+ gain_block[9] = 10 * log10(sum) - 32;
+
+ ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
+
+ /* output */
+ for (i=0; i < 5; i++)
+ block[i] = av_clipf(block[i], -4095./4096., 4095./4096.);
}
-static int pred(float *in, float *tgt, int n)
+/**
+ * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
+ *
+ * @param order filter order
+ * @param n input length
+ * @param non_rec number of non-recursive samples
+ * @param out filter output
+ * @param hist pointer to the input history of the filter
+ * @param out pointer to the non-recursive part of the output
+ * @param out2 pointer to the recursive part of the output
+ * @param window pointer to the windowing function table
+ */
+static void do_hybrid_window(int order, int n, int non_rec, float *out,
+ float *hist, float *out2, const float *window)
{
- int x,y;
- float *p1,*p2;
- double f0,f1,f2;
- float temp;
-
- if (in[n]==0) return 0;
- if ((f0=*in)<=0) return 0;
-
- for (x=1;;x++) {
- if (n<x) return 1;
-
- p1=in+x;
- p2=tgt;
- f1=*(p1--);
- for (y=x;--y;f1+=(*(p1--))*(*(p2++)));
-
- p1=tgt+x-1;
- p2=tgt;
- *(p1--)=f2=-f1/f0;
- for (y=x>>1;y--;) {
- temp=*p2+*p1*f2;
- *(p1--)+=*p2*f2;
- *(p2++)=temp;
+ int i;
+ float buffer1[order + 1];
+ float buffer2[order + 1];
+ float work[order + n + non_rec];
+
+ apply_window(work, window, hist, order + n + non_rec);
+
+ convolve(buffer1, work + order , n , order);
+ convolve(buffer2, work + order + n, non_rec, order);
+
+ for (i=0; i <= order; i++) {
+ out2[i] = out2[i] * 0.5625 + buffer1[i];
+ out [i] = out2[i] + buffer2[i];
}
- if ((f0+=f1*f2)<0) return 0;
- }
-}
-static void co(int n, int i, int j, float *in, float *out, float *st1, float *st2, const float *table)
-{
- int a,b,c;
- unsigned int x;
- float *fp;
- float buffer1[37];
- float buffer2[37];
- float work[111];
-
- /* rotate and multiply */
- c=(b=(a=n+i)+j)-i;
- fp=st1+i;
- for (x=0;x<b;x++) {
- if (x==c) fp=in;
- work[x]=*(table++)*(*(st1++)=*(fp++));
- }
-
- prodsum(buffer1,work+n,i,n);
- prodsum(buffer2,work+a,j,n);
-
- for (x=0;x<=n;x++) {
- *st2=*st2*(0.5625)+buffer1[x];
- out[x]=*(st2++)+buffer2[x];
- }
- *out*=1.00390625; /* to prevent clipping */
+ /* Multiply by the white noise correcting factor (WNCF). */
+ *out *= 257./256.;
}
-/* product sum (lsf) */
-static void prodsum(float *tgt, float *src, int len, int n)
+/**
+ * Backward synthesis filter, find the LPC coefficients from past speech data.
+ */
+static void backward_filter(float *hist, float *rec, const float *window,
+ float *lpc, const float *tab,
+ int order, int n, int non_rec, int move_size)
{
- unsigned int x;
- float *p1,*p2;
- double sum;
-
- while (n>=0)
- {
- p1=(p2=src)-n;
- for (sum=0,x=len;x--;sum+=(*p1++)*(*p2++));
- tgt[n--]=sum;
- }
-}
+ float temp[order+1];
-static void * decode_block(AVCodecContext * avctx, unsigned char *in, signed short int *out,unsigned len)
-{
- int x,y;
- Real288_internal *glob=avctx->priv_data;
- unsigned short int buffer[len];
-
- unpack(buffer,in,len);
- for (x=0;x<32;x++)
- {
- glob->phasep=(glob->phase=x&7)*5;
- decode(glob,buffer[x]);
- for (y=0;y<5;*(out++)=8*glob->output[glob->phasep+(y++)]);
- if (glob->phase==3) update(glob);
- }
- return out;
+ do_hybrid_window(order, n, non_rec, temp, hist, rec, window);
+
+ if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
+ apply_window(lpc, lpc, tab, order);
+
+ memmove(hist, hist + n, move_size*sizeof(*hist));
}
-/* Decode a block (celp) */
-static int ra288_decode_frame(AVCodecContext * avctx,
- void *data, int *data_size,
- uint8_t * buf, int buf_size)
+static int ra288_decode_frame(AVCodecContext * avctx, void *data,
+ int *data_size, AVPacket *avpkt)
{
- void *datao;
-
- if (buf_size < avctx->block_align)
- {
- av_log(avctx, AV_LOG_ERROR, "ffra288: Error! Input buffer is too small [%d<%d]\n",buf_size,avctx->block_align);
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+ float *out = data;
+ int i, j;
+ RA288Context *ractx = avctx->priv_data;
+ GetBitContext gb;
+
+ if (buf_size < avctx->block_align) {
+ av_log(avctx, AV_LOG_ERROR,
+ "Error! Input buffer is too small [%d<%d]\n",
+ buf_size, avctx->block_align);
return 0;
}
- datao = data;
- data = decode_block(avctx, buf, (signed short *)data, avctx->block_align);
+ if (*data_size < 32*5*4)
+ return -1;
+
+ init_get_bits(&gb, buf, avctx->block_align * 8);
+
+ for (i=0; i < 32; i++) {
+ float gain = amptable[get_bits(&gb, 3)];
+ int cb_coef = get_bits(&gb, 6 + (i&1));
+
+ decode(ractx, gain, cb_coef);
+
+ for (j=0; j < 5; j++)
+ *(out++) = ractx->sp_hist[70 + 36 + j];
+
+ if ((i & 7) == 3) {
+ backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window,
+ ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
+
+ backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window,
+ ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
+ }
+ }
- *data_size = (char *)data - (char *)datao;
+ *data_size = (char *)out - (char *)data;
return avctx->block_align;
}
"real_288",
CODEC_TYPE_AUDIO,
CODEC_ID_RA_288,
- sizeof(Real288_internal),
+ sizeof(RA288Context),
ra288_decode_init,
NULL,
NULL,
ra288_decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
};
projects / ffmpeg / blobdiff