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authorJustin Ruggles <justin.ruggles@gmail.com>2011-10-25 13:47:50 -0400
committerMichael Niedermayer <michaelni@gmx.at>2011-11-04 13:31:15 +0100
commit8491677ab632f89f31bb26e48c8d1c429be07928 (patch)
tree4b91f046f64321efe78f8dd0544948c7cfc9af16
parentf98bb0d3ec73b047121bcdf172cc809ac3c47013 (diff)
downloadffmpeg-8491677ab632f89f31bb26e48c8d1c429be07928.tar.gz
binkaudio: add some buffer overread checks.
This stops decoding before overreads instead of after. (cherry picked from commit 101ef19ef4dc9f5c3d536aee8fcc10fff2af4d9e) Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
-rw-r--r--libavcodec/binkaudio.c47
1 files changed, 37 insertions, 10 deletions
diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c
index ff36458c7e..3b65a19261 100644
--- a/libavcodec/binkaudio.c
+++ b/libavcodec/binkaudio.c
@@ -153,11 +153,18 @@ static const uint8_t rle_length_tab[16] = {
2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
};
+#define GET_BITS_SAFE(out, nbits) do { \
+ if (get_bits_left(gb) < nbits) \
+ return AVERROR_INVALIDDATA; \
+ out = get_bits(gb, nbits); \
+} while (0)
+
/**
* Decode Bink Audio block
* @param[out] out Output buffer (must contain s->block_size elements)
+ * @return 0 on success, negative error code on failure
*/
-static void decode_block(BinkAudioContext *s, short *out, int use_dct)
+static int decode_block(BinkAudioContext *s, short *out, int use_dct)
{
int ch, i, j, k;
float q, quant[25];
@@ -170,13 +177,19 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
for (ch = 0; ch < s->channels; ch++) {
FFTSample *coeffs = s->coeffs_ptr[ch];
if (s->version_b) {
+ if (get_bits_left(gb) < 64)
+ return AVERROR_INVALIDDATA;
coeffs[0] = av_int2flt(get_bits(gb, 32)) * s->root;
coeffs[1] = av_int2flt(get_bits(gb, 32)) * s->root;
} else {
+ if (get_bits_left(gb) < 58)
+ return AVERROR_INVALIDDATA;
coeffs[0] = get_float(gb) * s->root;
coeffs[1] = get_float(gb) * s->root;
}
+ if (get_bits_left(gb) < s->num_bands * 8)
+ return AVERROR_INVALIDDATA;
for (i = 0; i < s->num_bands; i++) {
/* constant is result of 0.066399999/log10(M_E) */
int value = get_bits(gb, 8);
@@ -191,15 +204,20 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
while (i < s->frame_len) {
if (s->version_b) {
j = i + 16;
- } else if (get_bits1(gb)) {
- j = i + rle_length_tab[get_bits(gb, 4)] * 8;
} else {
- j = i + 8;
+ int v;
+ GET_BITS_SAFE(v, 1);
+ if (v) {
+ GET_BITS_SAFE(v, 4);
+ j = i + rle_length_tab[v] * 8;
+ } else {
+ j = i + 8;
+ }
}
j = FFMIN(j, s->frame_len);
- width = get_bits(gb, 4);
+ GET_BITS_SAFE(width, 4);
if (width == 0) {
memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
i = j;
@@ -209,9 +227,11 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
while (i < j) {
if (s->bands[k] == i)
q = quant[k++];
- coeff = get_bits(gb, width);
+ GET_BITS_SAFE(coeff, width);
if (coeff) {
- if (get_bits1(gb))
+ int v;
+ GET_BITS_SAFE(v, 1);
+ if (v)
coeffs[i] = -q * coeff;
else
coeffs[i] = q * coeff;
@@ -247,6 +267,8 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
s->overlap_len * s->channels * sizeof(*out));
s->first = 0;
+
+ return 0;
}
static av_cold int decode_end(AVCodecContext *avctx)
@@ -278,12 +300,17 @@ static int decode_frame(AVCodecContext *avctx,
int reported_size;
GetBitContext *gb = &s->gb;
+ if (buf_size < 4) {
+ av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
+ return AVERROR_INVALIDDATA;
+ }
+
init_get_bits(gb, buf, buf_size * 8);
reported_size = get_bits_long(gb, 32);
- while (get_bits_count(gb) / 8 < buf_size &&
- samples + s->block_size <= samples_end) {
- decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT);
+ while (samples + s->block_size <= samples_end) {
+ if (decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT))
+ break;
samples += s->block_size;
get_bits_align32(gb);
}