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authorMichael Niedermayer <michaelni@gmx.at>2011-06-18 02:44:17 +0200
committerAnton Khirnov <anton@khirnov.net>2011-06-20 21:03:59 +0200
commit3a07f5a47a16bef86faab99bc02d2fd0f396afe8 (patch)
tree4613a6a7406fe037d59854a918b50563daac72a5
parent5a0a6ae639ac791ddebce64e2c316186d1db575c (diff)
downloadffmpeg-streaming-3a07f5a47a16bef86faab99bc02d2fd0f396afe8.zip
ffmpeg-streaming-3a07f5a47a16bef86faab99bc02d2fd0f396afe8.tar.gz
qdm2: Fix alignment of local array.
Fixes ticket270 Signed-off-by: Michael Niedermayer <michaelni@gmx.at> Signed-off-by: Anton Khirnov <anton@khirnov.net>
-rw-r--r--libavcodec/qdm2.c6
1 files changed, 3 insertions, 3 deletions
diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c
index 53ee304..86847ad 100644
--- a/libavcodec/qdm2.c
+++ b/libavcodec/qdm2.c
@@ -175,6 +175,7 @@ typedef struct {
DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
int synth_buf_offset[MPA_MAX_CHANNELS];
DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
+ DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
/// Mixed temporary data used in decoding
float tone_level[MPA_MAX_CHANNELS][30][64];
@@ -1598,7 +1599,6 @@ static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
*/
static void qdm2_synthesis_filter (QDM2Context *q, int index)
{
- float samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
int i, k, ch, sb_used, sub_sampling, dither_state = 0;
/* copy sb_samples */
@@ -1610,7 +1610,7 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index)
q->sb_samples[ch][(8 * index) + i][k] = 0;
for (ch = 0; ch < q->nb_channels; ch++) {
- float *samples_ptr = samples + ch;
+ float *samples_ptr = q->samples + ch;
for (i = 0; i < 8; i++) {
ff_mpa_synth_filter_float(&q->mpadsp,
@@ -1627,7 +1627,7 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index)
for (ch = 0; ch < q->channels; ch++)
for (i = 0; i < q->frame_size; i++)
- q->output_buffer[q->channels * i + ch] += (1 << 23) * samples[q->nb_channels * sub_sampling * i + ch];
+ q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
}
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