X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fbinkaudio.c;h=381fc4cd1509a25fa1e05aafe0ecedbccd5c81f7;hb=e15371061d23457554d241a80dc471515ac13ad4;hp=96cf968c66e6e2bb4eccc2986eaab865d7279cd6;hpb=409e684e79b6ee0c511292326f09b13fe230e58e;p=ffmpeg

diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c
index 96cf968c66e..381fc4cd150 100644
--- a/libavcodec/binkaudio.c
+++ b/libavcodec/binkaudio.c
@@ -30,6 +30,7 @@
 
 #include "libavutil/channel_layout.h"
 #include "libavutil/intfloat.h"
+#include "libavutil/mem_internal.h"
 
 #define BITSTREAM_READER_LE
 #include "avcodec.h"
@@ -40,8 +41,6 @@
 #include "rdft.h"
 #include "wma_freqs.h"
 
-static float quant_table[96];
-
 #define MAX_CHANNELS 2
 #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
 
@@ -54,10 +53,10 @@ typedef struct BinkAudioContext {
     int overlap_len;        ///< overlap size (samples)
     int block_size;
     int num_bands;
-    unsigned int *bands;
     float root;
-    DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
+    unsigned int bands[26];
     float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16];  ///< coeffs from previous audio block
+    float quant_table[96];
     AVPacket *pkt;
     union {
         RDFTContext rdft;
@@ -95,6 +94,8 @@ static av_cold int decode_init(AVCodecContext *avctx)
     if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {
         // audio is already interleaved for the RDFT format variant
         avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+        if (sample_rate > INT_MAX / avctx->channels)
+            return AVERROR_INVALIDDATA;
         sample_rate  *= avctx->channels;
         s->channels = 1;
         if (!s->version_b)
@@ -107,14 +108,14 @@ static av_cold int decode_init(AVCodecContext *avctx)
     s->frame_len     = 1 << frame_len_bits;
     s->overlap_len   = s->frame_len / 16;
     s->block_size    = (s->frame_len - s->overlap_len) * s->channels;
-    sample_rate_half = (sample_rate + 1) / 2;
+    sample_rate_half = (sample_rate + 1LL) / 2;
     if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
         s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
     else
         s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);
     for (i = 0; i < 96; i++) {
         /* constant is result of 0.066399999/log10(M_E) */
-        quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
+        s->quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
     }
 
     /* calculate number of bands */
@@ -122,10 +123,6 @@ static av_cold int decode_init(AVCodecContext *avctx)
         if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
             break;
 
-    s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
-    if (!s->bands)
-        return AVERROR(ENOMEM);
-
     /* populate bands data */
     s->bands[0] = 2;
     for (i = 1; i < s->num_bands; i++)
@@ -151,7 +148,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
 static float get_float(GetBitContext *gb)
 {
     int power = get_bits(gb, 5);
-    float f = ldexpf(get_bits_long(gb, 23), power - 23);
+    float f = ldexpf(get_bits(gb, 23), power - 23);
     if (get_bits1(gb))
         f = -f;
     return f;
@@ -195,7 +192,7 @@ static int decode_block(BinkAudioContext *s, float **out, int use_dct)
             return AVERROR_INVALIDDATA;
         for (i = 0; i < s->num_bands; i++) {
             int value = get_bits(gb, 8);
-            quant[i]  = quant_table[FFMIN(value, 95)];
+            quant[i]  = s->quant_table[FFMIN(value, 95)];
         }
 
         k = 0;
@@ -273,7 +270,6 @@ static int decode_block(BinkAudioContext *s, float **out, int use_dct)
 static av_cold int decode_end(AVCodecContext *avctx)
 {
     BinkAudioContext * s = avctx->priv_data;
-    av_freep(&s->bands);
     if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
         ff_rdft_end(&s->trans.rdft);
     else if (CONFIG_BINKAUDIO_DCT_DECODER)