Merge remote-tracking branch 'qatar/master'

[coroid/ffmpeg_saccubus.git] / libavcodec / ac3dec.c

diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c
index 42b62ef..ca6f18f 100644 (file)
--- a/libavcodec/ac3dec.c
+++ b/libavcodec/ac3dec.c
@@ -7,20 +7,20 @@
  * Copyright (c) 2007-2008 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
  * Copyright (c) 2007 Justin Ruggles <justin.ruggles@gmail.com>
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * 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.
  *
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -37,9 +37,6 @@
 #include "ac3dec_data.h"
 #include "kbdwin.h"
 
-/** Large enough for maximum possible frame size when the specification limit is ignored */
-#define AC3_FRAME_BUFFER_SIZE 32768
-
 /**
  * table for ungrouping 3 values in 7 bits.
  * used for exponents and bap=2 mantissas
@@ -205,11 +202,6 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
     }
     s->downmixed = 1;
 
-    /* allocate context input buffer */
-        s->input_buffer = av_mallocz(AC3_FRAME_BUFFER_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
-        if (!s->input_buffer)
-            return AVERROR(ENOMEM);
-
     return 0;
 }
 
@@ -1307,6 +1299,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
     float   *out_samples_flt = data;
     int16_t *out_samples_s16 = data;
     int blk, ch, err;
+    int data_size_orig, data_size_tmp;
     const uint8_t *channel_map;
     const float *output[AC3_MAX_CHANNELS];
 
@@ -1323,6 +1316,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
     init_get_bits(&s->gbc, buf, buf_size * 8);
 
     /* parse the syncinfo */
+    data_size_orig = *data_size;
     *data_size = 0;
     err = parse_frame_header(s);
 
@@ -1406,11 +1400,17 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
     channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on];
     for (ch = 0; ch < s->out_channels; ch++)
         output[ch] = s->output[channel_map[ch]];
+    data_size_tmp = s->num_blocks * 256 * avctx->channels;
+    data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(*out_samples_flt) : sizeof(*out_samples_s16);
+    if (data_size_orig < data_size_tmp)
+        return -1;
+    *data_size = data_size_tmp;
     for (blk = 0; blk < s->num_blocks; blk++) {
         if (!err && decode_audio_block(s, blk)) {
             av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
             err = 1;
         }
+
         if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
             s->fmt_conv.float_interleave(out_samples_flt, output, 256,
                                          s->out_channels);
@@ -1435,8 +1435,6 @@ static av_cold int ac3_decode_end(AVCodecContext *avctx)
     ff_mdct_end(&s->imdct_512);
     ff_mdct_end(&s->imdct_256);
 
-    av_freep(&s->input_buffer);
-
     return 0;
 }