dca: change the core to work with integer coefficients.

The DCA core decoder converts integer coefficients read from the
bitstream to floats just after reading them (along with dequantization).
All the other steps of the audio reconstruction are done with floats
which makes the output for the DTS lossless extension (XLL)
actually lossy.
This patch changes the DCA core to work with integer coefficients
until QMF. At this point the integer coefficients are converted to floats.
The coefficients for the LFE channel (lfe_data) are not touched.
This is the first step for the really lossless XLL decoding.

1 files changed, 34 insertions, 0 deletions

diff --git a/libavcodec/dcadsp.c b/libavcodec/dcadsp.c
index 9105a4cb4b..c333f4a44c 100644
--- a/libavcodec/dcadsp.c
+++ b/libavcodec/dcadsp.c
@@ -25,6 +25,7 @@
 #include "libavutil/intreadwrite.h"
 
 #include "dcadsp.h"
+#include "dcamath.h"
 
 static void decode_hf_c(float dst[DCA_SUBBANDS][8],
                         const int32_t vq_num[DCA_SUBBANDS],
@@ -44,6 +45,21 @@ static void decode_hf_c(float dst[DCA_SUBBANDS][8],
     }
 }
 
+static void decode_hf_int_c(int32_t dst[DCA_SUBBANDS][8],
+                            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, j;
+
+    for (j = start; j < end; j++) {
+        const int8_t *ptr = &hf_vq[vq_num[j]][vq_offset];
+        for (i = 0; i < 8; i++)
+            dst[j][i] = ptr[i] * scale[j][0] + 8 >> 4;
+    }
+}
+
 static inline void dca_lfe_fir(float *out, const float *in, const float *coefs,
                                int decifactor)
 {
@@ -93,6 +109,22 @@ static void dca_qmf_32_subbands(float samples_in[32][8], int sb_act,
     }
 }
 
+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 < 8; 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);
@@ -109,6 +141,8 @@ av_cold void ff_dcadsp_init(DCADSPContext *s)
     s->lfe_fir[1]      = dca_lfe_fir1_c;
     s->qmf_32_subbands = dca_qmf_32_subbands;
     s->decode_hf       = decode_hf_c;
+    s->decode_hf_int   = decode_hf_int_c;
+    s->dequantize      = dequantize_c;
 
     if (ARCH_AARCH64)
         ff_dcadsp_init_aarch64(s);