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| author | Michael Niedermayer <michaelni@gmx.at> | 2011-06-18 02:44:17 +0200 |
|---|---|---|
| committer | Anton Khirnov <anton@khirnov.net> | 2011-06-20 21:03:59 +0200 |
| commit | 3a07f5a47a16bef86faab99bc02d2fd0f396afe8 (patch) | |
| tree | 4613a6a7406fe037d59854a918b50563daac72a5 | |
| parent | 5a0a6ae639ac791ddebce64e2c316186d1db575c (diff) | |
| download | ffmpeg-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.c | 6 |
1 files changed, 3 insertions, 3 deletions
diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index 53ee304a28..86847adc10 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]; } |