* RealAudio 2.0 (28.8K)
* Copyright (c) 2003 the ffmpeg project
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#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 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) */
+ /// 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) */
+ /// recursive part of the gain autocorrelation (spec: REXPLG)
float gain_rec[11];
-
- float sp_block[41]; ///< Speech data of four blocks (spec: STTMP)
- float gain_block[10]; ///< Gain data of four blocks (spec: GSTATE)
} RA288Context;
-static inline float scalar_product_float(const float * v1, const float * v2,
- int size)
+static av_cold int ra288_decode_init(AVCodecContext *avctx)
{
- float res = 0.;
-
- while (size--)
- res += *v1++ * *v2++;
-
- return res;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ return 0;
}
-static void colmult(float *tgt, const float *m1, const float *m2, int n)
+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 x, y;
+ 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_block + 5, ractx->sp_block, 36*sizeof(*ractx->sp_block));
-
- for (x=4; x >= 0; x--)
- ractx->sp_block[x] = -scalar_product_float(ractx->sp_block + x + 1,
- ractx->sp_lpc, 36);
+ memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
/* block 46 of G.728 spec */
- sum = 32. - scalar_product_float(ractx->gain_lpc, ractx->gain_block, 10);
+ 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 */
- 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 (x=0; x < 5; x++)
- buffer[x] = codetable[cb_coef][x] * sumsum;
+ for (i=0; i < 5; i++)
+ 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,
- 10 * sizeof(*ractx->gain_block));
-
- *ractx->gain_block = 10 * log10(sum) - 32;
-
- for (x=1; x < 5; x++)
- for (y=x-1; y >= 0; y--)
- buffer[x] -= ractx->sp_lpc[x-y-1] * buffer[y];
-
- /* output */
- for (x=0; x < 5; x++)
- ractx->sp_block[4-x] =
- av_clipf(ractx->sp_block[4-x] + buffer[x], -4095, 4095);
-}
-
-/**
- * Converts autocorrelation coefficients to LPC coefficients using the
- * Levinson-Durbin algorithm. See blocks 37 and 50 of the G.728 specification.
- *
- * @return 0 if success, -1 if fail
- */
-static int eval_lpc_coeffs(const float *in, float *tgt, int n)
-{
- int x, y;
- double f0, f1, f2;
-
- if (in[n] == 0)
- return -1;
+ memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
- if ((f0 = *in) <= 0)
- return -1;
-
- in--; // To avoid a -1 subtraction in the inner loop
-
- for (x=1; x <= n; x++) {
- f1 = in[x+1];
-
- for (y=0; y < x - 1; y++)
- f1 += in[x-y]*tgt[y];
-
- tgt[x-1] = f2 = -f1/f0;
- for (y=0; y < x >> 1; y++) {
- float temp = tgt[y] + tgt[x-y-2]*f2;
- tgt[x-y-2] += tgt[y]*f2;
- tgt[y] = temp;
- }
- if ((f0 += f1*f2) < 0)
- return -1;
- }
-
- return 0;
-}
-
-static void prodsum(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);
}
/**
- * Hybrid window filtering. See blocks 36 and 49 of the G.728 specification.
- *
- * @note This function is slightly different from that described in the spec.
- * It expects in[0] to be the newest sample and in[n-1] to be the oldest
- * one stored. The spec has in the more ordinary way (in[0] the oldest
- * and in[n-1] the newest).
+ * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
*
- * @param order the order of the filter
- * @param n the length of the input
- * @param non_rec the number of non-recursive samples
- * @param out the 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 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, 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)
{
- unsigned int x;
- float buffer1[order + 1];
- float buffer2[order + 1];
- float work[order + n + non_rec];
+ int i;
+ 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];
- /* update history */
- memmove(hist, hist + n, (order + non_rec)*sizeof(*hist));
+ apply_window(work, window, hist, order + n + non_rec);
- for (x=0; x < n; x++)
- hist[order + non_rec + x] = in[n-x-1];
+ convolve(buffer1, work + order , n , order);
+ convolve(buffer2, work + order + n, non_rec, order);
- colmult(work, window, hist, order + n + non_rec);
-
- prodsum(buffer1, work + order , n , order);
- prodsum(buffer2, work + order + n, non_rec, order);
-
- for (x=0; x <= order; x++) {
- out2[x] = out2[x] * 0.5625 + buffer1[x];
- out [x] = out2[x] + buffer2[x];
+ for (i=0; i <= order; i++) {
+ out2[i] = out2[i] * 0.5625 + buffer1[i];
+ out [i] = out2[i] + buffer2[i];
}
- /* Multiply by the white noise correcting factor (WNCF) */
+ /* Multiply by the white noise correcting factor (WNCF). */
*out *= 257./256.;
}
/**
- * Backward synthesis filter. Find the LPC coefficients from past speech data.
+ * 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, temp1, ractx->sp_hist,
- ractx->sp_rec, syn_window);
+ float temp[MAX_BACKWARD_FILTER_ORDER+1];
- if (!eval_lpc_coeffs(temp1, ractx->sp_lpc, 36))
- colmult(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, 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 (!eval_lpc_coeffs(temp2, ractx->gain_lpc, 10))
- colmult(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;
- int x, y;
+ 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*4)
+ return -1;
+
init_get_bits(&gb, buf, avctx->block_align * 8);
- for (x=0; x < 32; x++) {
+ for (i=0; i < 32; i++) {
float gain = amptable[get_bits(&gb, 3)];
- int cb_coef = get_bits(&gb, 6 + (x&1));
+ int cb_coef = get_bits(&gb, 6 + (i&1));
decode(ractx, gain, cb_coef);
- for (y=0; y < 5; y++)
- *(out++) = 8 * ractx->sp_block[4 - y];
+ 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);
- if ((x & 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),
- NULL,
+ ra288_decode_init,
NULL,
NULL,
ra288_decode_frame,
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