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qdm2: Fix alignment of local array.

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Fixes ticket270

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
pull/2/head
Michael Niedermayer 14 years ago
parent 0c6cf3fa7c
commit 44d1b4088f
1 changed files with 3 additions and 3 deletions
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  1. 6
      libavcodec/qdm2.c

6
libavcodec/qdm2.c
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@ -175,6 +175,7 @@ typedef struct {
DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2]; DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
int synth_buf_offset[MPA_MAX_CHANNELS]; int synth_buf_offset[MPA_MAX_CHANNELS];
DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; 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 /// Mixed temporary data used in decoding
float tone_level[MPA_MAX_CHANNELS][30][64]; 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) 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; int i, k, ch, sb_used, sub_sampling, dither_state = 0;
/* copy sb_samples */ /* 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; q->sb_samples[ch][(8 * index) + i][k] = 0;
for (ch = 0; ch < q->nb_channels; ch++) { 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++) { for (i = 0; i < 8; i++) {
ff_mpa_synth_filter_float(&q->mpadsp, 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 (ch = 0; ch < q->channels; ch++)
for (i = 0; i < q->frame_size; i++) 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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