#include "avcodec.h"
#define ALT_BITSTREAM_READER_LE
-#include "bitstream.h"
+#include "get_bits.h"
#include "ra288.h"
#include "lpc.h"
+#include "celp_math.h"
+#include "celp_filters.h"
+
+#define MAX_BACKWARD_FILTER_ORDER 36
+#define MAX_BACKWARD_FILTER_LEN 40
+#define MAX_BACKWARD_FILTER_NONREC 35
typedef struct {
float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A)
float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB)
- float sp_hist[111]; ///< speech data history (spec: SB)
+ /** 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];
- float gain_hist[38]; ///< log-gain history (spec: SBLG)
+ /** 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];
-
- float sp_block[41]; ///< four blocks of speech data (spec: STTMP)
- float gain_block[10]; ///< four blocks of gain data (spec: GSTATE)
} RA288Context;
static av_cold int ra288_decode_init(AVCodecContext *avctx)
{
- avctx->sample_fmt = SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
return 0;
}
-static inline float scalar_product_float(const float * v1, const float * v2,
- int size)
-{
- float res = 0.;
-
- while (size--)
- res += *v1++ * *v2++;
-
- return res;
-}
-
static void apply_window(float *tgt, const float *m1, const float *m2, int n)
{
while (n--)
*tgt++ = *m1++ * *m2++;
}
+static void convolve(float *tgt, const float *src, int len, int n)
+{
+ for (; n >= 0; n--)
+ tgt[n] = ff_dot_productf(src, src - n, len);
+
+}
+
static void decode(RA288Context *ractx, float gain, int cb_coef)
{
- int i, j;
+ int i;
double sumsum;
float sum, buffer[5];
- float *block = ractx->sp_block + 36; // current block
-
- memmove(ractx->sp_block, ractx->sp_block + 5, 36*sizeof(*ractx->sp_block));
+ float *block = ractx->sp_hist + 70 + 36; // current block
+ float *gain_block = ractx->gain_hist + 28;
- for (i=0; i < 5; i++) {
- block[i] = 0.;
- for (j=0; j < 36; j++)
- block[i] -= block[i-1-j]*ractx->sp_lpc[j];
- }
+ 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 -= ractx->gain_block[9-i] * ractx->gain_lpc[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 */
- sumsum = exp(sum * 0.1151292546497) * gain; /* pow(10.0,sum/20)*gain */
+ /* 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 * (1./2048.);
+ buffer[i] = codetable[cb_coef][i] * sumsum;
- sum = scalar_product_float(buffer, buffer, 5) / 5;
+ sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.);
sum = FFMAX(sum, 1);
/* shift and store */
- memmove(ractx->gain_block, ractx->gain_block + 1,
- 9 * sizeof(*ractx->gain_block));
-
- ractx->gain_block[9] = 10 * log10(sum) - 32;
-
- for (i=1; i < 5; i++)
- for (j=i-1; j >= 0; j--)
- buffer[i] -= ractx->sp_lpc[i-j-1] * buffer[j];
-
- /* output */
- for (i=0; i < 5; i++)
- block[i] = av_clipf(block[i] + buffer[i], -4095, 4095);
-}
+ memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
-static void convolve(float *tgt, const float *src, int len, int n)
-{
- for (; n >= 0; n--)
- tgt[n] = scalar_product_float(src, src - n, len);
+ gain_block[9] = 10 * log10(sum) - 32;
+ ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
}
/**
* @param n input length
* @param non_rec number of non-recursive samples
* @param out filter output
- * @param in pointer to the input of the filter
- * @param hist Pointer to the input history of the filter, it is updated by
- * this function.
+ * @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, const float *in,
- float *out, float *hist, float *out2,
- const float *window)
+static void do_hybrid_window(int order, int n, int non_rec, float *out,
+ float *hist, float *out2, const float *window)
{
int i;
- float buffer1[order + 1];
- float buffer2[order + 1];
- float work[order + n + non_rec];
-
- /* update history */
- memmove(hist , hist + n, (order + non_rec)*sizeof(*hist));
- memcpy (hist + order + non_rec, in , n *sizeof(*hist));
+ float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
+ float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
+ float work[MAX_BACKWARD_FILTER_ORDER + MAX_BACKWARD_FILTER_LEN + MAX_BACKWARD_FILTER_NONREC];
apply_window(work, window, hist, order + n + non_rec);
/**
* Backward synthesis filter, find the LPC coefficients from past speech data.
*/
-static void backward_filter(RA288Context *ractx)
+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)
{
- float temp1[37]; // RTMP in the spec
- float temp2[11]; // GPTPMP in the spec
-
- do_hybrid_window(36, 40, 35, ractx->sp_block+1, temp1, ractx->sp_hist,
- ractx->sp_rec, syn_window);
+ float temp[MAX_BACKWARD_FILTER_ORDER+1];
- if (!compute_lpc_coefs(temp1, 36, ractx->sp_lpc, 0, 1, 1))
- apply_window(ractx->sp_lpc, ractx->sp_lpc, syn_bw_tab, 36);
+ do_hybrid_window(order, n, non_rec, temp, hist, rec, window);
- do_hybrid_window(10, 8, 20, ractx->gain_block+2, temp2, ractx->gain_hist,
- ractx->gain_rec, gain_window);
+ if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
+ apply_window(lpc, lpc, tab, order);
- if (!compute_lpc_coefs(temp2, 10, ractx->gain_lpc, 0, 1, 1))
- apply_window(ractx->gain_lpc, ractx->gain_lpc, gain_bw_tab, 10);
+ memmove(hist, hist + n, move_size*sizeof(*hist));
}
static int ra288_decode_frame(AVCodecContext * avctx, void *data,
- int *data_size, const uint8_t * buf,
- int buf_size)
+ int *data_size, AVPacket *avpkt)
{
- int16_t *out = data;
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+ float *out = data;
int i, j;
RA288Context *ractx = avctx->priv_data;
GetBitContext gb;
return 0;
}
- if (*data_size < 32*5*2)
+ if (*data_size < 32*5*4)
return -1;
init_get_bits(&gb, buf, avctx->block_align * 8);
decode(ractx, gain, cb_coef);
for (j=0; j < 5; j++)
- *(out++) = 8 * ractx->sp_block[36 + 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);
- if ((i & 7) == 3)
- backward_filter(ractx);
+ backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window,
+ ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
+ }
}
*data_size = (char *)out - (char *)data;
return avctx->block_align;
}
-AVCodec ra_288_decoder =
+AVCodec ff_ra_288_decoder =
{
"real_288",
- CODEC_TYPE_AUDIO,
+ AVMEDIA_TYPE_AUDIO,
CODEC_ID_RA_288,
sizeof(RA288Context),
ra288_decode_init,
projects / ffmpeg / blobdiff