dcadec: reorganise context data

place primary audio coding header data into DCAAudioHeader
structure to make DCAContext clearer
and move channel related data to DCAChan structure to make
them easier to use by extensions
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>

libavcodec/dca.h

@@ -130,6 +130,47 @@ typedef struct QMF64_table {
     float rsin[32];
 } QMF64_table;
 
+/* Primary audio coding header */
+typedef struct DCAAudioHeader {
+    int subband_activity[DCA_PRIM_CHANNELS_MAX];    ///< subband activity count
+    int vq_start_subband[DCA_PRIM_CHANNELS_MAX];    ///< high frequency vq start subband
+    int joint_intensity[DCA_PRIM_CHANNELS_MAX];     ///< joint intensity coding index
+    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 subframes;              ///< number of subframes
+    int total_channels;         ///< number of channels including extensions
+    int prim_channels;          ///< number of primary audio channels
+} DCAAudioHeader;
+
+typedef struct DCAChan {
+    DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_SUBBANDS][8];
+
+    /* Subband samples history (for ADPCM) */
+    DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_SUBBANDS][4];
+    int hist_index;
+
+    /* Half size is sufficient for core decoding, but for 96 kHz data
+     * we need QMF with 64 subbands and 1024 samples. */
+    DECLARE_ALIGNED(32, float, subband_fir_hist)[1024];
+    DECLARE_ALIGNED(32, float, subband_fir_noidea)[64];
+
+    /* Primary audio coding side information */
+    int prediction_mode[DCA_SUBBANDS];    ///< prediction mode (ADPCM used or not)
+    int prediction_vq[DCA_SUBBANDS];      ///< prediction VQ coefs
+    int bitalloc[DCA_SUBBANDS];           ///< bit allocation index
+    int transition_mode[DCA_SUBBANDS];    ///< transition mode (transients)
+    int32_t scale_factor[DCA_SUBBANDS][2];///< scale factors (2 if transient)
+    int joint_huff;                       ///< joint subband scale factors codebook
+    int joint_scale_factor[DCA_SUBBANDS]; ///< joint subband scale factors
+
+    int32_t  high_freq_vq[DCA_SUBBANDS];  ///< VQ encoded high frequency subbands
+} DCAChan;
+
+
 typedef struct DCAContext {
     AVClass *class;             ///< class for AVOptions
     AVCodecContext *avctx;
@@ -163,28 +204,11 @@ typedef struct DCAContext {
     int dialog_norm;            ///< dialog normalisation parameter
 
     /* Primary audio coding header */
-    int subframes;              ///< number of subframes
-    int total_channels;         ///< number of channels including extensions
-    int prim_channels;          ///< number of primary audio channels
-    int subband_activity[DCA_PRIM_CHANNELS_MAX];    ///< subband activity count
-    int vq_start_subband[DCA_PRIM_CHANNELS_MAX];    ///< high frequency vq start subband
-    int joint_intensity[DCA_PRIM_CHANNELS_MAX];     ///< joint intensity coding index
-    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
+    DCAAudioHeader audio_header;
 
     /* Primary audio coding side information */
     int subsubframes[DCA_SUBFRAMES_MAX];                         ///< number of subsubframes
     int partial_samples[DCA_SUBFRAMES_MAX];                      ///< partial subsubframe samples count
-    int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];    ///< prediction mode (ADPCM used or not)
-    int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];      ///< prediction VQ coefs
-    int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];           ///< bit allocation index
-    int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];    ///< transition mode (transients)
-    int32_t scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2];///< scale factors (2 if transient)
-    int joint_huff[DCA_PRIM_CHANNELS_MAX];                       ///< joint subband scale factors codebook
-    int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
     float downmix_coef[DCA_PRIM_CHANNELS_MAX + 1][2];            ///< stereo downmix coefficients
     int dynrange_coef;                                           ///< dynamic range coefficient
 
@@ -195,23 +219,17 @@ typedef struct DCAContext {
     uint8_t  core_downmix_amode;                                 ///< audio channel arrangement of embedded downmix
     uint16_t core_downmix_codes[DCA_PRIM_CHANNELS_MAX + 1][4];   ///< embedded downmix coefficients (9-bit codes)
 
-    int32_t  high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];  ///< VQ encoded high frequency subbands
 
     float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)];      ///< Low frequency effect data
     int lfe_scale_factor;
 
     /* Subband samples history (for ADPCM) */
-    DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
-    /* Half size is sufficient for core decoding, but for 96 kHz data
-     * we need QMF with 64 subbands and 1024 samples. */
-    DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][1024];
-    DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][64];
-    int hist_index[DCA_PRIM_CHANNELS_MAX];
     DECLARE_ALIGNED(32, float, raXin)[32];
 
+    DCAChan dca_chan[DCA_PRIM_CHANNELS_MAX];
+
     int output;                 ///< type of output
 
-    DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
     float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];
     float *extra_channels[DCA_PRIM_CHANNELS_MAX + 1];
     uint8_t *extra_channels_buffer;