* 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)
static av_cold int ra288_decode_init(AVCodecContext *avctx)
{
- avctx->sample_fmt = SAMPLE_FMT_FLT;
+ 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--)
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);
+ 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_hist + 70 + 36; // current block
for (i=0; i < 5; i++)
buffer[i] = codetable[cb_coef][i] * sumsum;
- sum = scalar_product_float(buffer, buffer, 5) * ((1<<24)/5.);
+ sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.);
sum = FFMAX(sum, 1);
gain_block[9] = 10 * log10(sum) - 32;
- for (i=0; i < 5; i++) {
- block[i] = buffer[i];
- for (j=0; j < 36; j++)
- block[i] -= block[i-1-j]*ractx->sp_lpc[j];
- }
-
- /* output */
- for (i=0; i < 5; i++)
- block[i] = av_clipf(block[i], -4095./4096., 4095./4096.);
+ ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
}
/**
float *hist, float *out2, const float *window)
{
int i;
- float buffer1[order + 1];
- float buffer2[order + 1];
- float work[order + n + non_rec];
+ 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);
float *lpc, const float *tab,
int order, int n, int non_rec, int move_size)
{
- float temp[order+1];
+ float temp[MAX_BACKWARD_FILTER_ORDER+1];
do_hybrid_window(order, n, non_rec, temp, hist, rec, window);
}
static int ra288_decode_frame(AVCodecContext * avctx, void *data,
- int *data_size, const uint8_t * buf,
- int buf_size)
+ int *data_size, AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
float *out = data;
int i, j;
RA288Context *ractx = avctx->priv_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,