---
While at it.
libavcodec/dcadec.c | 10 ++++++----
1 file changed, 6 insertions(+), 4 deletions(-)
diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c
index af26dceafd..fa2a2400fe 100644
--- a/libavcodec/dcadec.c
+++ b/libavcodec/dcadec.c
@@ -947,15 +947,16 @@ static int dca_filter_channels(DCAContext *s, int
block_index, int upsample)
/* 64 subbands QMF */
for (k = 0; k < s->audio_header.prim_channels; k++) {
+ int channel = s->channel_order_tab[k];
int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
s->dca_chan[k].subband_samples[block_index];
s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
DCA_SUBBANDS_X96K *
SAMPLES_PER_SUBBAND);
- if (s->channel_order_tab[k] >= 0)
+ if (channel >= 0)
qmf_64_subbands(s, k, samples,
- s->samples_chanptr[s->channel_order_tab[k]],
+ s->samples_chanptr[channel],
/* Upsampling needs a factor 2 here. */
M_SQRT2 / 32768.0);
}
@@ -964,15 +965,16 @@ static int dca_filter_channels(DCAContext *s, int
block_index, int upsample)
LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS],
[SAMPLES_PER_SUBBAND]);
for (k = 0; k < s->audio_header.prim_channels; k++) {
+ int channel = s->channel_order_tab[k];
int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
s->dca_chan[k].subband_samples[block_index];
s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
DCA_SUBBANDS * SAMPLES_PER_SUBBAND);
- if (s->channel_order_tab[k] >= 0)
+ if (channel >= 0)
qmf_32_subbands(s, k, samples,
- s->samples_chanptr[s->channel_order_tab[k]],
+ s->samples_chanptr[channel],
M_SQRT1_2 / 32768.0);
}
}
--
2.11.1
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