Merge commit 'aebf0707'

* commit 'aebf0707':
  dca: change the core to work with integer coefficients.
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>

libavcodec/dca.h

@@ -140,8 +140,8 @@ typedef struct DCAAudioHeader {
     int transient_huffman[DCA_PRIM_CHANNELS_MAX];   ///< transient mode code book
     int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
     int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX];    ///< bit allocation quantizer select
-    int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
-    float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX];   ///< scale factor adjustment
+    int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX];  ///< quantization index codebook select
+    uint32_t scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
 
     int subframes;              ///< number of subframes
     int total_channels;         ///< number of channels including extensions
@@ -149,10 +149,10 @@ typedef struct DCAAudioHeader {
 } DCAAudioHeader;
 
 typedef struct DCAChan {
-    DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_SUBBANDS][8];
+    DECLARE_ALIGNED(32, int32_t, subband_samples)[DCA_BLOCKS_MAX][DCA_SUBBANDS][8];
 
     /* Subband samples history (for ADPCM) */
-    DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_SUBBANDS][4];
+    DECLARE_ALIGNED(32, int32_t, subband_samples_hist)[DCA_SUBBANDS][4];
     int hist_index;
 
     /* Half size is sufficient for core decoding, but for 96 kHz data

libavcodec/dcadec.c

@@ -214,7 +214,7 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel,
                                          int xxch)
 {
     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 };
     int hdr_pos = 0, hdr_size = 0;
@@ -327,7 +327,7 @@ 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++)
@@ -869,10 +869,7 @@ 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
@@ -880,13 +877,12 @@ 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;
@@ -897,27 +893,25 @@ 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) {
@@ -930,7 +924,7 @@ 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");
@@ -939,20 +933,18 @@ 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;
             /*
@@ -962,25 +954,25 @@ 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);
                 }
             }
 
@@ -1000,11 +992,12 @@ 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]);
+
         }
     }
 
@@ -1024,6 +1017,8 @@ 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)
@@ -1032,21 +1027,31 @@ 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);
         }

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);

libavcodec/dcadsp.h

@@ -37,6 +37,12 @@ typedef struct DCADSPContext {
                       const int8_t hf_vq[1024][32], intptr_t vq_offset,
                       int32_t scale[DCA_SUBBANDS][2],
                       intptr_t start, intptr_t end);
+    void (*decode_hf_int)(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);
+    void (*dequantize)(int32_t *samples, uint32_t step_size, uint32_t scale);
 } DCADSPContext;
 
 void ff_dcadsp_init(DCADSPContext *s);

libavcodec/fmtconvert.c

@@ -32,6 +32,14 @@ static void int32_to_float_fmul_scalar_c(float *dst, const int32_t *src,
         dst[i] = src[i] * mul;
 }
 
+static void int32_to_float_c(float *dst, const int32_t *src, intptr_t len)
+{
+    int i;
+
+    for (i = 0; i < len; i++)
+        dst[i] = (float)src[i];
+}
+
 static void int32_to_float_fmul_array8_c(FmtConvertContext *c, float *dst,
                                          const int32_t *src, const float *mul,
                                          int len)
@@ -43,6 +51,7 @@ static void int32_to_float_fmul_array8_c(FmtConvertContext *c, float *dst,
 
 av_cold void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx)
 {
+    c->int32_to_float             = int32_to_float_c;
     c->int32_to_float_fmul_scalar = int32_to_float_fmul_scalar_c;
     c->int32_to_float_fmul_array8 = int32_to_float_fmul_array8_c;
 

libavcodec/fmtconvert.h

@@ -37,6 +37,16 @@ typedef struct FmtConvertContext {
      */
     void (*int32_to_float_fmul_scalar)(float *dst, const int32_t *src,
                                        float mul, int len);
+    /**
+     * Convert an array of int32_t to float.
+     * @param dst destination array of float.
+     *            constraints: 32-byte aligned
+     * @param src source array of int32_t.
+     *            constraints: 32-byte aligned
+     * @param len number of elements to convert.
+     *            constraints: multiple of 8
+     */
+    void (*int32_to_float)(float *dst, const int32_t *src, intptr_t len);
 
     /**
      * Convert an array of int32_t to float and multiply by a float value from another array,

tests/fate/audio.mak

@@ -24,7 +24,7 @@ fate-dca-core: REF = $(SAMPLES)/dts/dts.pcm
 FATE_DCA-$(CONFIG_DTS_DEMUXER) += fate-dca-xll
 fate-dca-xll: CMD = pcm -disable_xll 0 -i $(TARGET_SAMPLES)/dts/master_audio_7.1_24bit.dts
 fate-dca-xll: CMP = oneoff
-fate-dca-xll: REF = $(SAMPLES)/dts/master_audio_7.1_24bit.pcm
+fate-dca-xll: REF = $(SAMPLES)/dts/master_audio_7.1_24bit_2.pcm
 
 FATE_SAMPLES_AUDIO-$(CONFIG_DCA_DECODER) += $(FATE_DCA-yes)
 fate-dca: $(FATE_DCA-yes)
@@ -39,7 +39,7 @@ fate-dss-sp: CMD = framecrc -i $(TARGET_SAMPLES)/dss/sp.dss -frames 30
 FATE_SAMPLES_AUDIO-$(call DEMDEC, DTS, DCA) += fate-dts_es
 fate-dts_es: CMD = pcm -i $(TARGET_SAMPLES)/dts/dts_es.dts
 fate-dts_es: CMP = oneoff
-fate-dts_es: REF = $(SAMPLES)/dts/dts_es.pcm
+fate-dts_es: REF = $(SAMPLES)/dts/dts_es_2.pcm
 
 FATE_SAMPLES_AUDIO-$(call DEMDEC, AVI, IMC) += fate-imc
 fate-imc: CMD = pcm -i $(TARGET_SAMPLES)/imc/imc.avi