ffmpeg roll of source to mar 9 version...

patched-ffmpeg-mt/libavcodec/ra144.c

 /*
  * Real Audio 1.0 (14.4K)
  *
  * Copyright (c) 2008 Vitor Sessak
  * Copyright (c) 2003 Nick Kurshev
  *     Based on public domain decoder at http://www.honeypot.net/audio
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg 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,
  * 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
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include "libavutil/intmath.h"
 #include "avcodec.h"
 #include "get_bits.h"
 #include "ra144.h"
 #include "celp_filters.h"
 
 #define NBLOCKS         4       ///< number of subblocks within a block
 #define BLOCKSIZE       40      ///< subblock size in 16-bit words
 #define BUFFERSIZE      146     ///< the size of the adaptive codebook
 
 
@@ skipping 177 matching lines @@
 
     if (ff_celp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs,
                                     block, BLOCKSIZE, 10, 1, 0xfff))
         memset(ractx->curr_sblock, 0, 50*sizeof(*ractx->curr_sblock));
 }
 
 static void int_to_int16(int16_t *out, const int *inp)
 {
     int i;
 
-    for (i=0; i < 30; i++)
+    for (i=0; i < 10; i++)
         *out++ = *inp++;
 }
 
 /**
  * Evaluate the reflection coefficients from the filter coefficients.
  * Does the inverse of the eval_coefs() function.
  *
  * @return 1 if one of the reflection coefficients is greater than
  *         4095, 0 if not.
  */
@@ skipping 36 matching lines @@
 
 static int interp(RA144Context *ractx, int16_t *out, int a,
                   int copyold, int energy)
 {
     int work[10];
     int b = NBLOCKS - a;
     int i;
 
     // Interpolate block coefficients from the this frame's forth block and
     // last frame's forth block.
-    for (i=0; i<30; i++)
+    for (i=0; i<10; i++)
         out[i] = (a * ractx->lpc_coef[0][i] + b * ractx->lpc_coef[1][i])>> 2;
 
     if (eval_refl(work, out, ractx->avctx)) {
         // The interpolated coefficients are unstable, copy either new or old
         // coefficients.
         int_to_int16(out, ractx->lpc_coef[copyold]);
         return rescale_rms(ractx->lpc_refl_rms[copyold], energy);
     } else {
         return rescale_rms(rms(work), energy);
     }
 }
 
 /** Uncompress one block (20 bytes -> 160*2 bytes). */
 static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
                               int *data_size, AVPacket *avpkt)
 {
     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;
     static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
     unsigned int refl_rms[4];    // RMS of the reflection coefficients
-    uint16_t block_coefs[4][30]; // LPC coefficients of each sub-block
+    uint16_t block_coefs[4][10]; // LPC coefficients of each sub-block
     unsigned int lpc_refl[10];   // LPC reflection coefficients of the frame
     int i, j;
     int16_t *data = vdata;
     unsigned int energy;
 
     RA144Context *ractx = avctx->priv_data;
     GetBitContext gb;
 
     if (*data_size < 2*160)
         return -1;
@@ skipping 43 matching lines @@
     "real_144",
     CODEC_TYPE_AUDIO,
     CODEC_ID_RA_144,
     sizeof(RA144Context),
     ra144_decode_init,
     NULL,
     NULL,
     ra144_decode_frame,
     .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
 };