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, 58 insertions, 53 deletions

diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c
index aca6ed325f..399b1e5d32 100644
--- a/libavcodec/dcadec.c
+++ b/libavcodec/dcadec.c
@@ -226,7 +226,7 @@ static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
 static int dca_parse_audio_coding_header(DCAContext *s, int base_channel)
 {
     int i, j;
-    static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
+    static const uint8_t adj_table[4] = { 16, 18, 20, 23 };
     static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
     static const int thr[11]    = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
 
@@ -265,7 +265,7 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel)
     /* Get scale factor adjustment */
     for (j = 0; j < 11; j++)
         for (i = base_channel; i < s->audio_header.prim_channels; i++)
-            s->audio_header.scalefactor_adj[i][j] = 1;
+            s->audio_header.scalefactor_adj[i][j] = 16;
 
     for (j = 1; j < 11; j++)
         for (i = base_channel; i < s->audio_header.prim_channels; i++)
@@ -790,10 +790,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
 {
     int k, l;
     int subsubframe = s->current_subsubframe;
-
-    const float *quant_step_table;
-
-    LOCAL_ALIGNED_16(int32_t, block, [SAMPLES_PER_SUBBAND * DCA_SUBBANDS]);
+    const uint32_t *quant_step_table;
 
     /*
      * Audio data
@@ -801,13 +798,12 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
 
     /* Select quantization step size table */
     if (s->bit_rate_index == 0x1f)
-        quant_step_table = ff_dca_lossless_quant_d;
+        quant_step_table = ff_dca_lossless_quant;
     else
-        quant_step_table = ff_dca_lossy_quant_d;
+        quant_step_table = ff_dca_lossy_quant;
 
     for (k = base_channel; k < s->audio_header.prim_channels; k++) {
-        float (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index];
-        float rscale[DCA_SUBBANDS];
+        int32_t (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index];
 
         if (get_bits_left(&s->gb) < 0)
             return AVERROR_INVALIDDATA;
@@ -818,27 +814,25 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
             /* Select the mid-tread linear quantizer */
             int abits = s->dca_chan[k].bitalloc[l];
 
-            float quant_step_size = quant_step_table[abits];
-
-            /*
-             * Determine quantization index code book and its type
-             */
-
-            /* Select quantization index code book */
-            int sel = s->audio_header.quant_index_huffman[k][abits];
+            uint32_t quant_step_size = quant_step_table[abits];
 
             /*
              * Extract bits from the bit stream
              */
-            if (!abits) {
-                rscale[l] = 0;
-                memset(block + SAMPLES_PER_SUBBAND * l, 0, SAMPLES_PER_SUBBAND * sizeof(block[0]));
-            } else {
+            if (!abits)
+                memset(subband_samples[l], 0, SAMPLES_PER_SUBBAND *
+                       sizeof(subband_samples[l][0]));
+            else {
+                uint32_t rscale;
                 /* Deal with transients */
                 int sfi = s->dca_chan[k].transition_mode[l] &&
                     subsubframe >= s->dca_chan[k].transition_mode[l];
-                rscale[l] = quant_step_size * s->dca_chan[k].scale_factor[l][sfi] *
-                            s->audio_header.scalefactor_adj[k][sel];
+                /* Determine quantization index code book and its type.
+                   Select quantization index code book */
+                int sel = s->audio_header.quant_index_huffman[k][abits];
+
+                rscale = (s->dca_chan[k].scale_factor[l][sfi] *
+                          s->audio_header.scalefactor_adj[k][sel] + 8) >> 4;
 
                 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
                     if (abits <= 7) {
@@ -851,7 +845,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
                         block_code1 = get_bits(&s->gb, size);
                         block_code2 = get_bits(&s->gb, size);
                         err         = decode_blockcodes(block_code1, block_code2,
-                                                        levels, block + SAMPLES_PER_SUBBAND * l);
+                                                        levels, subband_samples[l]);
                         if (err) {
                             av_log(s->avctx, AV_LOG_ERROR,
                                    "ERROR: block code look-up failed\n");
@@ -860,20 +854,18 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
                     } else {
                         /* no coding */
                         for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
-                            block[SAMPLES_PER_SUBBAND * l + m] = get_sbits(&s->gb, abits - 3);
+                            subband_samples[l][m] = get_sbits(&s->gb, abits - 3);
                     }
                 } else {
                     /* Huffman coded */
                     for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
-                        block[SAMPLES_PER_SUBBAND * l + m] = get_bitalloc(&s->gb,
-                                                        &dca_smpl_bitalloc[abits], sel);
+                        subband_samples[l][m] = get_bitalloc(&s->gb,
+                                                             &dca_smpl_bitalloc[abits], sel);
                 }
+                s->dcadsp.dequantize(subband_samples[l], quant_step_size, rscale);
             }
         }
 
-        s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[0],
-                                               block, rscale, SAMPLES_PER_SUBBAND * s->audio_header.vq_start_subband[k]);
-
         for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
             int m;
             /*
@@ -883,25 +875,25 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
                 int n;
                 if (s->predictor_history)
                     subband_samples[l][0] += (ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
-                                                 s->dca_chan[k].subband_samples_hist[l][3] +
-                                                 ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] *
-                                                 s->dca_chan[k].subband_samples_hist[l][2] +
-                                                 ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] *
-                                                 s->dca_chan[k].subband_samples_hist[l][1] +
-                                                 ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] *
-                                                 s->dca_chan[k].subband_samples_hist[l][0]) *
-                                                (1.0f / 8192);
+                                              (int64_t)s->dca_chan[k].subband_samples_hist[l][3] +
+                                              ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] *
+                                              (int64_t)s->dca_chan[k].subband_samples_hist[l][2] +
+                                              ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] *
+                                              (int64_t)s->dca_chan[k].subband_samples_hist[l][1] +
+                                              ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] *
+                                              (int64_t)s->dca_chan[k].subband_samples_hist[l][0]) +
+                                              (1 << 12) >> 13;
                 for (m = 1; m < SAMPLES_PER_SUBBAND; m++) {
-                    float sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
-                                subband_samples[l][m - 1];
+                    int64_t sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
+                                  (int64_t)subband_samples[l][m - 1];
                     for (n = 2; n <= 4; n++)
                         if (m >= n)
                             sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
-                                   subband_samples[l][m - n];
+                                   (int64_t)subband_samples[l][m - n];
                         else if (s->predictor_history)
                             sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
-                                   s->dca_chan[k].subband_samples_hist[l][m - n + 4];
-                    subband_samples[l][m] += sum * 1.0f / 8192;
+                                   (int64_t)s->dca_chan[k].subband_samples_hist[l][m - n + 4];
+                    subband_samples[l][m] += (int32_t)(sum + (1 << 12) >> 13);
                 }
             }
 
@@ -921,11 +913,12 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
                 s->debug_flag |= 0x01;
             }
 
-            s->dcadsp.decode_hf(subband_samples, s->dca_chan[k].high_freq_vq,
-                                ff_dca_high_freq_vq, subsubframe * SAMPLES_PER_SUBBAND,
-                                s->dca_chan[k].scale_factor,
-                                s->audio_header.vq_start_subband[k],
-                                s->audio_header.subband_activity[k]);
+            s->dcadsp.decode_hf_int(subband_samples, s->dca_chan[k].high_freq_vq,
+                                    ff_dca_high_freq_vq, subsubframe * SAMPLES_PER_SUBBAND,
+                                    s->dca_chan[k].scale_factor,
+                                    s->audio_header.vq_start_subband[k],
+                                    s->audio_header.subband_activity[k]);
+
         }
     }
 
@@ -945,6 +938,8 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
     int k;
 
     if (upsample) {
+        LOCAL_ALIGNED(32, float, samples, [64], [SAMPLES_PER_SUBBAND]);
+
         if (!s->qmf64_table) {
             s->qmf64_table = qmf64_precompute();
             if (!s->qmf64_table)
@@ -953,21 +948,31 @@ 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++) {
-            float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index];
+            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],
+                                       64 * SAMPLES_PER_SUBBAND);
 
             if (s->channel_order_tab[k] >= 0)
-                qmf_64_subbands(s, k, subband_samples,
+                qmf_64_subbands(s, k, samples,
                                 s->samples_chanptr[s->channel_order_tab[k]],
                                 /* Upsampling needs a factor 2 here. */
                                 M_SQRT2 / 32768.0);
         }
     } else {
         /* 32 subbands QMF */
+        LOCAL_ALIGNED(32, float, samples, [32], [SAMPLES_PER_SUBBAND]);
+
         for (k = 0; k < s->audio_header.prim_channels; k++) {
-            float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index];
+            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],
+                                       32 * SAMPLES_PER_SUBBAND);
 
             if (s->channel_order_tab[k] >= 0)
-                qmf_32_subbands(s, k, subband_samples,
+                qmf_32_subbands(s, k, samples,
                                 s->samples_chanptr[s->channel_order_tab[k]],
                                 M_SQRT1_2 / 32768.0);
         }