#include "libavutil/intreadwrite.h"
#include "dcadsp.h"
+#include "dcamath.h"
-static void decode_hf_c(float dst[DCA_SUBBANDS][8],
+static void decode_hf_c(int32_t dst[DCA_SUBBANDS][SAMPLES_PER_SUBBAND],
const int32_t vq_num[DCA_SUBBANDS],
const int8_t hf_vq[1024][32], intptr_t vq_offset,
int32_t scale[DCA_SUBBANDS][2],
intptr_t start, intptr_t end)
{
- int i, l;
+ int i, j;
- for (l = start; l < end; l++) {
- /* 1 vector -> 32 samples but we only need the 8 samples
- * for this subsubframe. */
- const int8_t *ptr = &hf_vq[vq_num[l]][vq_offset];
- float fscale = scale[l][0] * (1 / 16.0);
+ for (j = start; j < end; j++) {
+ const int8_t *ptr = &hf_vq[vq_num[j]][vq_offset];
for (i = 0; i < 8; i++)
- dst[l][i] = ptr[i] * fscale;
+ dst[j][i] = ptr[i] * scale[j][0] + 8 >> 4;
}
}
}
}
-static void dca_qmf_32_subbands(float samples_in[32][8], int sb_act,
+static void dca_qmf_32_subbands(float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], int sb_act,
SynthFilterContext *synth, FFTContext *imdct,
float synth_buf_ptr[512],
int *synth_buf_offset, float synth_buf2[32],
}
}
+static void dequantize_c(int32_t *samples, uint32_t step_size, uint32_t scale)
+{
+ int64_t step = (int64_t)step_size * scale;
+ int shift, i;
+ int32_t step_scale;
+
+ if (step > (1 << 23))
+ shift = av_log2(step >> 23) + 1;
+ else
+ shift = 0;
+ step_scale = (int32_t)(step >> shift);
+
+ for (i = 0; i < SAMPLES_PER_SUBBAND; i++)
+ samples[i] = dca_clip23(dca_norm((int64_t)samples[i] * step_scale, 22 - shift));
+}
+
static void dca_lfe_fir0_c(float *out, const float *in, const float *coefs)
{
dca_lfe_fir(out, in, coefs, 32);
s->lfe_fir[1] = dca_lfe_fir1_c;
s->qmf_32_subbands = dca_qmf_32_subbands;
s->decode_hf = decode_hf_c;
+ s->dequantize = dequantize_c;
if (ARCH_AARCH64)
ff_dcadsp_init_aarch64(s);