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/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