* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-/**
- * @file dca.c
- */
-
#include <math.h>
#include <stddef.h>
#include <stdio.h>
+#include "libavutil/intmath.h"
+#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "dsputil.h"
-#include "bitstream.h"
+#include "fft.h"
+#include "get_bits.h"
+#include "put_bits.h"
#include "dcadata.h"
#include "dcahuff.h"
#include "dca.h"
+#include "synth_filter.h"
+#include "dcadsp.h"
//#define TRACE
-#define DCA_PRIM_CHANNELS_MAX (5)
+#define DCA_PRIM_CHANNELS_MAX (7)
#define DCA_SUBBANDS (32)
#define DCA_ABITS_MAX (32) /* Should be 28 */
-#define DCA_SUBSUBFAMES_MAX (4)
+#define DCA_SUBSUBFRAMES_MAX (4)
+#define DCA_SUBFRAMES_MAX (16)
+#define DCA_BLOCKS_MAX (16)
#define DCA_LFE_MAX (3)
enum DCAMode {
DCA_4F2R
};
+/* Tables for mapping dts channel configurations to libavcodec multichannel api.
+ * Some compromises have been made for special configurations. Most configurations
+ * are never used so complete accuracy is not needed.
+ *
+ * L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead.
+ * S -> side, when both rear and back are configured move one of them to the side channel
+ * OV -> center back
+ * All 2 channel configurations -> CH_LAYOUT_STEREO
+ */
+
+static const int64_t dca_core_channel_layout[] = {
+ CH_FRONT_CENTER, ///< 1, A
+ CH_LAYOUT_STEREO, ///< 2, A + B (dual mono)
+ CH_LAYOUT_STEREO, ///< 2, L + R (stereo)
+ CH_LAYOUT_STEREO, ///< 2, (L+R) + (L-R) (sum-difference)
+ CH_LAYOUT_STEREO, ///< 2, LT +RT (left and right total)
+ CH_LAYOUT_STEREO|CH_FRONT_CENTER, ///< 3, C+L+R
+ CH_LAYOUT_STEREO|CH_BACK_CENTER, ///< 3, L+R+S
+ CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_BACK_CENTER, ///< 4, C + L + R+ S
+ CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT, ///< 4, L + R +SL+ SR
+ CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_SIDE_LEFT|CH_SIDE_RIGHT, ///< 5, C + L + R+ SL+SR
+ CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT|CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR
+ CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_RIGHT|CH_FRONT_CENTER|CH_BACK_CENTER, ///< 6, C + L + R+ LR + RR + OV
+ CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_FRONT_LEFT_OF_CENTER|CH_BACK_CENTER|CH_BACK_LEFT|CH_BACK_RIGHT, ///< 6, CF+ CR+LF+ RF+LR + RR
+ CH_FRONT_LEFT_OF_CENTER|CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR
+ CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT|CH_BACK_LEFT|CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2+ SR1 + SR2
+ CH_FRONT_LEFT_OF_CENTER|CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_BACK_CENTER|CH_SIDE_RIGHT, ///< 8, CL + C+ CR + L + R + SL + S+ SR
+};
+
+static const int8_t dca_lfe_index[] = {
+ 1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
+};
+
+static const int8_t dca_channel_reorder_lfe[][9] = {
+ { 0, -1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 4, -1, -1, -1, -1, -1},
+ { 0, 1, 3, 4, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 4, 5, -1, -1, -1, -1},
+ { 3, 4, 0, 1, 5, 6, -1, -1, -1},
+ { 2, 0, 1, 4, 5, 6, -1, -1, -1},
+ { 0, 6, 4, 5, 2, 3, -1, -1, -1},
+ { 4, 2, 5, 0, 1, 6, 7, -1, -1},
+ { 5, 6, 0, 1, 7, 3, 8, 4, -1},
+ { 4, 2, 5, 0, 1, 6, 8, 7, -1},
+};
+
+static const int8_t dca_channel_reorder_lfe_xch[][9] = {
+ { 0, 2, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 4, -1, -1, -1, -1, -1},
+ { 0, 1, 3, 4, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 4, 5, -1, -1, -1, -1},
+ { 0, 1, 4, 5, 3, -1, -1, -1, -1},
+ { 2, 0, 1, 5, 6, 4, -1, -1, -1},
+ { 3, 4, 0, 1, 6, 7, 5, -1, -1},
+ { 2, 0, 1, 4, 5, 6, 7, -1, -1},
+ { 0, 6, 4, 5, 2, 3, 7, -1, -1},
+ { 4, 2, 5, 0, 1, 7, 8, 6, -1},
+ { 5, 6, 0, 1, 8, 3, 9, 4, 7},
+ { 4, 2, 5, 0, 1, 6, 9, 8, 7},
+};
+
+static const int8_t dca_channel_reorder_nolfe[][9] = {
+ { 0, -1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 3, -1, -1, -1, -1, -1},
+ { 0, 1, 2, 3, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 3, 4, -1, -1, -1, -1},
+ { 2, 3, 0, 1, 4, 5, -1, -1, -1},
+ { 2, 0, 1, 3, 4, 5, -1, -1, -1},
+ { 0, 5, 3, 4, 1, 2, -1, -1, -1},
+ { 3, 2, 4, 0, 1, 5, 6, -1, -1},
+ { 4, 5, 0, 1, 6, 2, 7, 3, -1},
+ { 3, 2, 4, 0, 1, 5, 7, 6, -1},
+};
+
+static const int8_t dca_channel_reorder_nolfe_xch[][9] = {
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 3, -1, -1, -1, -1, -1},
+ { 0, 1, 2, 3, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 3, 4, -1, -1, -1, -1},
+ { 0, 1, 3, 4, 2, -1, -1, -1, -1},
+ { 2, 0, 1, 4, 5, 3, -1, -1, -1},
+ { 2, 3, 0, 1, 5, 6, 4, -1, -1},
+ { 2, 0, 1, 3, 4, 5, 6, -1, -1},
+ { 0, 5, 3, 4, 1, 2, 6, -1, -1},
+ { 3, 2, 4, 0, 1, 6, 7, 5, -1},
+ { 4, 5, 0, 1, 7, 2, 8, 3, 6},
+ { 3, 2, 4, 0, 1, 5, 8, 7, 6},
+};
+
#define DCA_DOLBY 101 /* FIXME */
#define DCA_CHANNEL_BITS 6
#define DCA_LFE 0x80
#define HEADER_SIZE 14
-#define CONVERT_BIAS 384
-#define DCA_MAX_FRAME_SIZE 16383
+#define DCA_MAX_FRAME_SIZE 16384
/** Bit allocation */
typedef struct {
static BitAlloc dca_scalefactor; ///< scalefactor VLCs
static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
-/** Pre-calculated cosine modulation coefs for the QMF */
-static float cos_mod[544];
-
static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx)
{
return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset;
int amode; ///< audio channels arrangement
int sample_rate; ///< audio sampling rate
int bit_rate; ///< transmission bit rate
+ int bit_rate_index; ///< transmission bit rate index
int downmix; ///< embedded downmix enabled
int dynrange; ///< embedded dynamic range flag
/* 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
float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
/* Primary audio coding side information */
- int subsubframes; ///< number of subsubframes
- int partial_samples; ///< partial subsubframe samples count
+ 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 high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
- float lfe_data[2 * DCA_SUBSUBFAMES_MAX * DCA_LFE_MAX *
- 2 /*history */ ]; ///< Low frequency effect data
+ float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)]; ///< Low frequency effect data
int lfe_scale_factor;
/* Subband samples history (for ADPCM) */
float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
- float subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512];
- float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][64];
+ DECLARE_ALIGNED(16, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
+ DECLARE_ALIGNED(16, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
+ int hist_index[DCA_PRIM_CHANNELS_MAX];
+ DECLARE_ALIGNED(16, float, raXin)[32];
int output; ///< type of output
- int bias; ///< output bias
+ float add_bias; ///< output bias
+ float scale_bias; ///< output scale
- DECLARE_ALIGNED_16(float, samples[1536]); /* 6 * 256 = 1536, might only need 5 */
- DECLARE_ALIGNED_16(int16_t, tsamples[1536]);
+ DECLARE_ALIGNED(16, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
+ DECLARE_ALIGNED(16, float, samples)[(DCA_PRIM_CHANNELS_MAX+1)*256];
+ const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX+1];
uint8_t dca_buffer[DCA_MAX_FRAME_SIZE];
int dca_buffer_size; ///< how much data is in the dca_buffer
+ const int8_t* channel_order_tab; ///< channel reordering table, lfe and non lfe
GetBitContext gb;
/* Current position in DCA frame */
int current_subframe;
int current_subsubframe;
+ /* XCh extension information */
+ int xch_present;
+ int xch_base_channel; ///< index of first (only) channel containing XCH data
+
int debug_flag; ///< used for suppressing repeated error messages output
DSPContext dsp;
+ FFTContext imdct;
+ SynthFilterContext synth;
+ DCADSPContext dcadsp;
} DCAContext;
-static void dca_init_vlcs(void)
+static const uint16_t dca_vlc_offs[] = {
+ 0, 512, 640, 768, 1282, 1794, 2436, 3080, 3770, 4454, 5364,
+ 5372, 5380, 5388, 5392, 5396, 5412, 5420, 5428, 5460, 5492, 5508,
+ 5572, 5604, 5668, 5796, 5860, 5892, 6412, 6668, 6796, 7308, 7564,
+ 7820, 8076, 8620, 9132, 9388, 9910, 10166, 10680, 11196, 11726, 12240,
+ 12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,
+ 18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,
+};
+
+static av_cold void dca_init_vlcs(void)
{
- static int vlcs_inited = 0;
- int i, j;
+ static int vlcs_initialized = 0;
+ int i, j, c = 14;
+ static VLC_TYPE dca_table[23622][2];
- if (vlcs_inited)
+ if (vlcs_initialized)
return;
dca_bitalloc_index.offset = 1;
dca_bitalloc_index.wrap = 2;
- for (i = 0; i < 5; i++)
+ for (i = 0; i < 5; i++) {
+ dca_bitalloc_index.vlc[i].table = &dca_table[dca_vlc_offs[i]];
+ dca_bitalloc_index.vlc[i].table_allocated = dca_vlc_offs[i + 1] - dca_vlc_offs[i];
init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
bitalloc_12_bits[i], 1, 1,
- bitalloc_12_codes[i], 2, 2, 1);
+ bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
+ }
dca_scalefactor.offset = -64;
dca_scalefactor.wrap = 2;
- for (i = 0; i < 5; i++)
+ for (i = 0; i < 5; i++) {
+ dca_scalefactor.vlc[i].table = &dca_table[dca_vlc_offs[i + 5]];
+ dca_scalefactor.vlc[i].table_allocated = dca_vlc_offs[i + 6] - dca_vlc_offs[i + 5];
init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
scales_bits[i], 1, 1,
- scales_codes[i], 2, 2, 1);
+ scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
+ }
dca_tmode.offset = 0;
dca_tmode.wrap = 1;
- for (i = 0; i < 4; i++)
+ for (i = 0; i < 4; i++) {
+ dca_tmode.vlc[i].table = &dca_table[dca_vlc_offs[i + 10]];
+ dca_tmode.vlc[i].table_allocated = dca_vlc_offs[i + 11] - dca_vlc_offs[i + 10];
init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
tmode_bits[i], 1, 1,
- tmode_codes[i], 2, 2, 1);
+ tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
+ }
- for(i = 0; i < 10; i++)
- for(j = 0; j < 7; j++){
- if(!bitalloc_codes[i][j]) break;
+ for (i = 0; i < 10; i++)
+ for (j = 0; j < 7; j++){
+ if (!bitalloc_codes[i][j]) break;
dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i];
dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4);
+ dca_smpl_bitalloc[i+1].vlc[j].table = &dca_table[dca_vlc_offs[c]];
+ dca_smpl_bitalloc[i+1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];
init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j],
bitalloc_sizes[i],
bitalloc_bits[i][j], 1, 1,
- bitalloc_codes[i][j], 2, 2, 1);
+ bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
+ c++;
}
- vlcs_inited = 1;
+ vlcs_initialized = 1;
}
static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
*dst++ = get_bits(gb, bits);
}
-static int dca_parse_frame_header(DCAContext * s)
+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 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 };
- s->bias = CONVERT_BIAS;
+ s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
+ s->prim_channels = s->total_channels;
+
+ if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
+ s->prim_channels = DCA_PRIM_CHANNELS_MAX;
+
+
+ for (i = base_channel; i < s->prim_channels; i++) {
+ s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
+ if (s->subband_activity[i] > DCA_SUBBANDS)
+ s->subband_activity[i] = DCA_SUBBANDS;
+ }
+ for (i = base_channel; i < s->prim_channels; i++) {
+ s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
+ if (s->vq_start_subband[i] > DCA_SUBBANDS)
+ s->vq_start_subband[i] = DCA_SUBBANDS;
+ }
+ get_array(&s->gb, s->joint_intensity + base_channel, s->prim_channels - base_channel, 3);
+ get_array(&s->gb, s->transient_huffman + base_channel, s->prim_channels - base_channel, 2);
+ get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3);
+ get_array(&s->gb, s->bitalloc_huffman + base_channel, s->prim_channels - base_channel, 3);
+
+ /* Get codebooks quantization indexes */
+ if (!base_channel)
+ memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
+ for (j = 1; j < 11; j++)
+ for (i = base_channel; i < s->prim_channels; i++)
+ s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
+
+ /* Get scale factor adjustment */
+ for (j = 0; j < 11; j++)
+ for (i = base_channel; i < s->prim_channels; i++)
+ s->scalefactor_adj[i][j] = 1;
+
+ for (j = 1; j < 11; j++)
+ for (i = base_channel; i < s->prim_channels; i++)
+ if (s->quant_index_huffman[i][j] < thr[j])
+ s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
+
+ if (s->crc_present) {
+ /* Audio header CRC check */
+ get_bits(&s->gb, 16);
+ }
+ s->current_subframe = 0;
+ s->current_subsubframe = 0;
+
+#ifdef TRACE
+ av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
+ av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
+ for (i = base_channel; i < s->prim_channels; i++){
+ av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
+ for (j = 0; j < 11; j++)
+ av_log(s->avctx, AV_LOG_DEBUG, " %i",
+ s->quant_index_huffman[i][j]);
+ av_log(s->avctx, AV_LOG_DEBUG, "\n");
+ av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
+ for (j = 0; j < 11; j++)
+ av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
+ av_log(s->avctx, AV_LOG_DEBUG, "\n");
+ }
+#endif
+
+ return 0;
+}
+
+static int dca_parse_frame_header(DCAContext * s)
+{
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
/* Sync code */
s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)];
if (!s->sample_rate)
return -1;
- s->bit_rate = dca_bit_rates[get_bits(&s->gb, 5)];
+ s->bit_rate_index = get_bits(&s->gb, 5);
+ s->bit_rate = dca_bit_rates[s->bit_rate_index];
if (!s->bit_rate)
return -1;
/* FIXME: channels mixing levels */
s->output = s->amode;
- if(s->lfe) s->output |= DCA_LFE;
+ if (s->lfe) s->output |= DCA_LFE;
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size);
av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n",
s->amode, dca_channels[s->amode]);
- av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i (%i Hz)\n",
- s->sample_rate, dca_sample_rates[s->sample_rate]);
- av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i (%i bits/s)\n",
- s->bit_rate, dca_bit_rates[s->bit_rate]);
+ av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i Hz\n",
+ s->sample_rate);
+ av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i bits/s\n",
+ s->bit_rate);
av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix);
av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange);
av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp);
/* Primary audio coding header */
s->subframes = get_bits(&s->gb, 4) + 1;
- s->prim_channels = get_bits(&s->gb, 3) + 1;
-
-
- for (i = 0; i < s->prim_channels; i++) {
- s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
- if (s->subband_activity[i] > DCA_SUBBANDS)
- s->subband_activity[i] = DCA_SUBBANDS;
- }
- for (i = 0; i < s->prim_channels; i++) {
- s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
- if (s->vq_start_subband[i] > DCA_SUBBANDS)
- s->vq_start_subband[i] = DCA_SUBBANDS;
- }
- get_array(&s->gb, s->joint_intensity, s->prim_channels, 3);
- get_array(&s->gb, s->transient_huffman, s->prim_channels, 2);
- get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3);
- get_array(&s->gb, s->bitalloc_huffman, s->prim_channels, 3);
-
- /* Get codebooks quantization indexes */
- memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
- for (j = 1; j < 11; j++)
- for (i = 0; i < s->prim_channels; i++)
- s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
-
- /* Get scale factor adjustment */
- for (j = 0; j < 11; j++)
- for (i = 0; i < s->prim_channels; i++)
- s->scalefactor_adj[i][j] = 1;
-
- for (j = 1; j < 11; j++)
- for (i = 0; i < s->prim_channels; i++)
- if (s->quant_index_huffman[i][j] < thr[j])
- s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
-
- if (s->crc_present) {
- /* Audio header CRC check */
- get_bits(&s->gb, 16);
- }
- s->current_subframe = 0;
- s->current_subsubframe = 0;
-
-#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
- av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
- for(i = 0; i < s->prim_channels; i++){
- av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
- for (j = 0; j < 11; j++)
- av_log(s->avctx, AV_LOG_DEBUG, " %i",
- s->quant_index_huffman[i][j]);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
- for (j = 0; j < 11; j++)
- av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
-#endif
-
- return 0;
+ return dca_parse_audio_coding_header(s, 0);
}
if (level < 5) {
/* huffman encoded */
value += get_bitalloc(gb, &dca_scalefactor, level);
- } else if(level < 8)
+ } else if (level < 8)
value = get_bits(gb, level + 1);
return value;
}
-static int dca_subframe_header(DCAContext * s)
+static int dca_subframe_header(DCAContext * s, int base_channel, int block_index)
{
/* Primary audio coding side information */
int j, k;
- s->subsubframes = get_bits(&s->gb, 2) + 1;
- s->partial_samples = get_bits(&s->gb, 3);
- for (j = 0; j < s->prim_channels; j++) {
+ if (!base_channel) {
+ s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
+ s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
+ }
+
+ for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++)
s->prediction_mode[j][k] = get_bits(&s->gb, 1);
}
/* Get prediction codebook */
- for (j = 0; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++) {
if (s->prediction_mode[j][k] > 0) {
/* (Prediction coefficient VQ address) */
}
/* Bit allocation index */
- for (j = 0; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->vq_start_subband[j]; k++) {
if (s->bitalloc_huffman[j] == 6)
s->bitalloc[j][k] = get_bits(&s->gb, 5);
else if (s->bitalloc_huffman[j] == 5)
s->bitalloc[j][k] = get_bits(&s->gb, 4);
- else {
+ else if (s->bitalloc_huffman[j] == 7) {
+ av_log(s->avctx, AV_LOG_ERROR,
+ "Invalid bit allocation index\n");
+ return -1;
+ } else {
s->bitalloc[j][k] =
get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
}
}
/* Transition mode */
- for (j = 0; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++) {
s->transition_mode[j][k] = 0;
- if (s->subsubframes > 1 &&
+ if (s->subsubframes[s->current_subframe] > 1 &&
k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {
s->transition_mode[j][k] =
get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);
}
}
- for (j = 0; j < s->prim_channels; j++) {
- uint32_t *scale_table;
+ for (j = base_channel; j < s->prim_channels; j++) {
+ const uint32_t *scale_table;
int scale_sum;
memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
if (s->scalefactor_huffman[j] == 6)
- scale_table = (uint32_t *) scale_factor_quant7;
+ scale_table = scale_factor_quant7;
else
- scale_table = (uint32_t *) scale_factor_quant6;
+ scale_table = scale_factor_quant6;
/* When huffman coded, only the difference is encoded */
scale_sum = 0;
}
/* Joint subband scale factor codebook select */
- for (j = 0; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->prim_channels; j++) {
/* Transmitted only if joint subband coding enabled */
if (s->joint_intensity[j] > 0)
s->joint_huff[j] = get_bits(&s->gb, 3);
}
/* Scale factors for joint subband coding */
- for (j = 0; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->prim_channels; j++) {
int source_channel;
/* Transmitted only if joint subband coding enabled */
s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */
}
- if (!s->debug_flag & 0x02) {
+ if (!(s->debug_flag & 0x02)) {
av_log(s->avctx, AV_LOG_DEBUG,
"Joint stereo coding not supported\n");
s->debug_flag |= 0x02;
}
/* Stereo downmix coefficients */
- if (s->prim_channels > 2) {
- if(s->downmix) {
- for (j = 0; j < s->prim_channels; j++) {
+ if (!base_channel && s->prim_channels > 2) {
+ if (s->downmix) {
+ for (j = base_channel; j < s->prim_channels; j++) {
s->downmix_coef[j][0] = get_bits(&s->gb, 7);
s->downmix_coef[j][1] = get_bits(&s->gb, 7);
}
} else {
int am = s->amode & DCA_CHANNEL_MASK;
- for (j = 0; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->prim_channels; j++) {
s->downmix_coef[j][0] = dca_default_coeffs[am][j][0];
s->downmix_coef[j][1] = dca_default_coeffs[am][j][1];
}
*/
/* VQ encoded high frequency subbands */
- for (j = 0; j < s->prim_channels; j++)
+ for (j = base_channel; j < s->prim_channels; j++)
for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
/* 1 vector -> 32 samples */
s->high_freq_vq[j][k] = get_bits(&s->gb, 10);
/* Low frequency effect data */
- if (s->lfe) {
+ if (!base_channel && s->lfe) {
/* LFE samples */
- int lfe_samples = 2 * s->lfe * s->subsubframes;
+ int lfe_samples = 2 * s->lfe * (4 + block_index);
+ int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
float lfe_scale;
- for (j = lfe_samples; j < lfe_samples * 2; j++) {
+ for (j = lfe_samples; j < lfe_end_sample; j++) {
/* Signed 8 bits int */
s->lfe_data[j] = get_sbits(&s->gb, 8);
}
/* Quantization step size * scale factor */
lfe_scale = 0.035 * s->lfe_scale_factor;
- for (j = lfe_samples; j < lfe_samples * 2; j++)
+ for (j = lfe_samples; j < lfe_end_sample; j++)
s->lfe_data[j] *= lfe_scale;
}
#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes);
+ av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes[s->current_subframe]);
av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",
- s->partial_samples);
- for (j = 0; j < s->prim_channels; j++) {
+ s->partial_samples[s->current_subframe]);
+ for (j = base_channel; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");
for (k = 0; k < s->subband_activity[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
- for (j = 0; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG,
"prediction coefs: %f, %f, %f, %f\n",
(float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,
(float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);
}
- for (j = 0; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");
for (k = 0; k < s->vq_start_subband[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
- for (j = 0; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:");
for (k = 0; k < s->subband_activity[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
- for (j = 0; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:");
for (k = 0; k < s->subband_activity[j]; k++) {
if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0)
}
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
- for (j = 0; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->prim_channels; j++) {
if (s->joint_intensity[j] > 0) {
int source_channel = s->joint_intensity[j] - 1;
av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n");
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
}
- if (s->prim_channels > 2 && s->downmix) {
+ if (!base_channel && s->prim_channels > 2 && s->downmix) {
av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n");
for (j = 0; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]);
}
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
- for (j = 0; j < s->prim_channels; j++)
+ for (j = base_channel; j < s->prim_channels; j++)
for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);
- if(s->lfe){
- int lfe_samples = 2 * s->lfe * s->subsubframes;
+ if (!base_channel && s->lfe) {
+ int lfe_samples = 2 * s->lfe * (4 + block_index);
+ int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
+
av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
- for (j = lfe_samples; j < lfe_samples * 2; j++)
+ for (j = lfe_samples; j < lfe_end_sample; j++)
av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
float scale, float bias)
{
const float *prCoeff;
- int i, j, k;
- float praXin[33], *raXin = &praXin[1];
-
- float *subband_fir_hist = s->subband_fir_hist[chans];
- float *subband_fir_hist2 = s->subband_fir_noidea[chans];
+ int i;
- int chindex = 0, subindex;
+ int sb_act = s->subband_activity[chans];
+ int subindex;
- praXin[0] = 0.0;
+ scale *= sqrt(1/8.0);
/* Select filter */
if (!s->multirate_inter) /* Non-perfect reconstruction */
/* Reconstructed channel sample index */
for (subindex = 0; subindex < 8; subindex++) {
- float t1, t2, sum[16], diff[16];
-
/* Load in one sample from each subband and clear inactive subbands */
- for (i = 0; i < s->subband_activity[chans]; i++)
- raXin[i] = samples_in[i][subindex];
- for (; i < 32; i++)
- raXin[i] = 0.0;
-
- /* Multiply by cosine modulation coefficients and
- * create temporary arrays SUM and DIFF */
- for (j = 0, k = 0; k < 16; k++) {
- t1 = 0.0;
- t2 = 0.0;
- for (i = 0; i < 16; i++, j++){
- t1 += (raXin[2 * i] + raXin[2 * i + 1]) * cos_mod[j];
- t2 += (raXin[2 * i] + raXin[2 * i - 1]) * cos_mod[j + 256];
- }
- sum[k] = t1 + t2;
- diff[k] = t1 - t2;
+ for (i = 0; i < sb_act; i++){
+ uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ ((i-1)&2)<<30;
+ AV_WN32A(&s->raXin[i], v);
}
+ for (; i < 32; i++)
+ s->raXin[i] = 0.0;
- j = 512;
- /* Store history */
- for (k = 0; k < 16; k++)
- subband_fir_hist[k] = cos_mod[j++] * sum[k];
- for (k = 0; k < 16; k++)
- subband_fir_hist[32-k-1] = cos_mod[j++] * diff[k];
-
- /* Multiply by filter coefficients */
- for (k = 31, i = 0; i < 32; i++, k--)
- for (j = 0; j < 512; j += 64){
- subband_fir_hist2[i] += prCoeff[i+j] * ( subband_fir_hist[i+j] - subband_fir_hist[j+k]);
- subband_fir_hist2[i+32] += prCoeff[i+j+32]*(-subband_fir_hist[i+j] - subband_fir_hist[j+k]);
- }
-
- /* Create 32 PCM output samples */
- for (i = 0; i < 32; i++)
- samples_out[chindex++] = subband_fir_hist2[i] * scale + bias;
+ s->synth.synth_filter_float(&s->imdct,
+ s->subband_fir_hist[chans], &s->hist_index[chans],
+ s->subband_fir_noidea[chans], prCoeff,
+ samples_out, s->raXin, scale, bias);
+ samples_out+= 32;
- /* Update working arrays */
- memmove(&subband_fir_hist[32], &subband_fir_hist[0], (512 - 32) * sizeof(float));
- memmove(&subband_fir_hist2[0], &subband_fir_hist2[32], 32 * sizeof(float));
- memset(&subband_fir_hist2[32], 0, 32 * sizeof(float));
}
}
-static void lfe_interpolation_fir(int decimation_select,
+static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
int num_deci_sample, float *samples_in,
float *samples_out, float scale,
float bias)
* samples_out: An array holding interpolated samples
*/
- int decifactor, k, j;
+ int decifactor;
const float *prCoeff;
-
- int interp_index = 0; /* Index to the interpolated samples */
int deciindex;
/* Select decimation filter */
if (decimation_select == 1) {
- decifactor = 128;
+ decifactor = 64;
prCoeff = lfe_fir_128;
} else {
- decifactor = 64;
+ decifactor = 32;
prCoeff = lfe_fir_64;
}
/* Interpolation */
for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
- /* One decimated sample generates decifactor interpolated ones */
- for (k = 0; k < decifactor; k++) {
- float rTmp = 0.0;
- //FIXME the coeffs are symetric, fix that
- for (j = 0; j < 512 / decifactor; j++)
- rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor];
- samples_out[interp_index++] = rTmp / scale + bias;
- }
+ s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor,
+ scale, bias);
+ samples_in++;
+ samples_out += 2 * decifactor;
}
}
samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * coef[2][1];
#define DOWNMIX_TO_STEREO(op1, op2) \
- for(i = 0; i < 256; i++){ \
+ for (i = 0; i < 256; i++){ \
op1 \
op2 \
}
float t;
float coef[DCA_PRIM_CHANNELS_MAX][2];
- for(i=0; i<DCA_PRIM_CHANNELS_MAX; i++) {
+ for (i=0; i<DCA_PRIM_CHANNELS_MAX; i++) {
coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];
coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];
}
int offset = (levels - 1) >> 1;
for (i = 0; i < 4; i++) {
- values[i] = (code % levels) - offset;
- code /= levels;
+ int div = FASTDIV(code, levels);
+ values[i] = code - offset - div*levels;
+ code = div;
}
if (code == 0)
static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
-static int dca_subsubframe(DCAContext * s)
+static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
{
int k, l;
int subsubframe = s->current_subsubframe;
- float *quant_step_table;
+ const float *quant_step_table;
/* FIXME */
- float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
+ float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
+ LOCAL_ALIGNED_16(int, block, [8]);
/*
* Audio data
*/
/* Select quantization step size table */
- if (s->bit_rate == 0x1f)
- quant_step_table = (float *) lossless_quant_d;
+ if (s->bit_rate_index == 0x1f)
+ quant_step_table = lossless_quant_d;
else
- quant_step_table = (float *) lossy_quant_d;
+ quant_step_table = lossy_quant_d;
- for (k = 0; k < s->prim_channels; k++) {
+ for (k = base_channel; k < s->prim_channels; k++) {
for (l = 0; l < s->vq_start_subband[k]; l++) {
int m;
int abits = s->bitalloc[k][l];
float quant_step_size = quant_step_table[abits];
- float rscale;
/*
* Determine quantization index code book and its type
/*
* Extract bits from the bit stream
*/
- if(!abits){
+ if (!abits){
memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
- }else if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
- if(abits <= 7){
- /* Block code */
- int block_code1, block_code2, size, levels;
- int block[8];
-
- size = abits_sizes[abits-1];
- levels = abits_levels[abits-1];
-
- block_code1 = get_bits(&s->gb, size);
- /* FIXME Should test return value */
- decode_blockcode(block_code1, levels, block);
- block_code2 = get_bits(&s->gb, size);
- decode_blockcode(block_code2, levels, &block[4]);
- for (m = 0; m < 8; m++)
- subband_samples[k][l][m] = block[m];
+ } else {
+ /* Deal with transients */
+ int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
+ float rscale = quant_step_size * s->scale_factor[k][l][sfi] * s->scalefactor_adj[k][sel];
+
+ if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
+ if (abits <= 7){
+ /* Block code */
+ int block_code1, block_code2, size, levels;
+
+ size = abits_sizes[abits-1];
+ levels = abits_levels[abits-1];
+
+ block_code1 = get_bits(&s->gb, size);
+ /* FIXME Should test return value */
+ decode_blockcode(block_code1, levels, block);
+ block_code2 = get_bits(&s->gb, size);
+ decode_blockcode(block_code2, levels, &block[4]);
+ }else{
+ /* no coding */
+ for (m = 0; m < 8; m++)
+ block[m] = get_sbits(&s->gb, abits - 3);
+ }
}else{
- /* no coding */
+ /* Huffman coded */
for (m = 0; m < 8; m++)
- subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3);
+ block[m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
}
- }else{
- /* Huffman coded */
- for (m = 0; m < 8; m++)
- subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
- }
-
- /* Deal with transients */
- if (s->transition_mode[k][l] &&
- subsubframe >= s->transition_mode[k][l])
- rscale = quant_step_size * s->scale_factor[k][l][1];
- else
- rscale = quant_step_size * s->scale_factor[k][l][0];
-
- rscale *= s->scalefactor_adj[k][sel];
- for (m = 0; m < 8; m++)
- subband_samples[k][l][m] *= rscale;
+ s->dsp.int32_to_float_fmul_scalar(subband_samples[k][l],
+ block, rscale, 8);
+ }
/*
* Inverse ADPCM if in prediction mode
}
/* Check for DSYNC after subsubframe */
- if (s->aspf || subsubframe == s->subsubframes - 1) {
+ if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {
if (0xFFFF == get_bits(&s->gb, 16)) { /* 0xFFFF */
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");
}
/* Backup predictor history for adpcm */
- for (k = 0; k < s->prim_channels; k++)
+ for (k = base_channel; k < s->prim_channels; k++)
for (l = 0; l < s->vq_start_subband[k]; l++)
memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],
4 * sizeof(subband_samples[0][0][0]));
+ return 0;
+}
+
+static int dca_filter_channels(DCAContext * s, int block_index)
+{
+ float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
+ int k;
+
/* 32 subbands QMF */
for (k = 0; k < s->prim_channels; k++) {
/* static float pcm_to_double[8] =
{32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
- qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * k],
- 2.0 / 3 /*pcm_to_double[s->source_pcm_res] */ ,
- 0 /*s->bias */ );
+ qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]],
+ M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ ,
+ s->add_bias );
}
/* Down mixing */
-
- if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) {
+ if (s->avctx->request_channels == 2 && s->prim_channels > 2) {
dca_downmix(s->samples, s->amode, s->downmix_coef);
}
/* Generate LFE samples for this subsubframe FIXME!!! */
if (s->output & DCA_LFE) {
- int lfe_samples = 2 * s->lfe * s->subsubframes;
- int i_channels = dca_channels[s->output & DCA_CHANNEL_MASK];
-
- lfe_interpolation_fir(s->lfe, 2 * s->lfe,
- s->lfe_data + lfe_samples +
- 2 * s->lfe * subsubframe,
- &s->samples[256 * i_channels],
- 256.0, 0 /* s->bias */);
+ lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
+ s->lfe_data + 2 * s->lfe * (block_index + 4),
+ &s->samples[256 * dca_lfe_index[s->amode]],
+ (1.0/256.0)*s->scale_bias, s->add_bias);
/* Outputs 20bits pcm samples */
}
}
-static int dca_subframe_footer(DCAContext * s)
+static int dca_subframe_footer(DCAContext * s, int base_channel)
{
int aux_data_count = 0, i;
- int lfe_samples;
/*
* Unpack optional information
*/
- if (s->timestamp)
- get_bits(&s->gb, 32);
-
- if (s->aux_data)
- aux_data_count = get_bits(&s->gb, 6);
+ /* presumably optional information only appears in the core? */
+ if (!base_channel) {
+ if (s->timestamp)
+ get_bits(&s->gb, 32);
- for (i = 0; i < aux_data_count; i++)
- get_bits(&s->gb, 8);
+ if (s->aux_data)
+ aux_data_count = get_bits(&s->gb, 6);
- if (s->crc_present && (s->downmix || s->dynrange))
- get_bits(&s->gb, 16);
+ for (i = 0; i < aux_data_count; i++)
+ get_bits(&s->gb, 8);
- lfe_samples = 2 * s->lfe * s->subsubframes;
- for (i = 0; i < lfe_samples; i++) {
- s->lfe_data[i] = s->lfe_data[i + lfe_samples];
+ if (s->crc_present && (s->downmix || s->dynrange))
+ get_bits(&s->gb, 16);
}
return 0;
* @param s pointer to the DCAContext
*/
-static int dca_decode_block(DCAContext * s)
+static int dca_decode_block(DCAContext * s, int base_channel, int block_index)
{
/* Sanity check */
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
#endif
/* Read subframe header */
- if (dca_subframe_header(s))
+ if (dca_subframe_header(s, base_channel, block_index))
return -1;
}
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
#endif
- if (dca_subsubframe(s))
+ if (dca_subsubframe(s, base_channel, block_index))
return -1;
/* Update state */
s->current_subsubframe++;
- if (s->current_subsubframe >= s->subsubframes) {
+ if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {
s->current_subsubframe = 0;
s->current_subframe++;
}
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
#endif
/* Read subframe footer */
- if (dca_subframe_footer(s))
+ if (dca_subframe_footer(s, base_channel))
return -1;
}
/**
* Convert bitstream to one representation based on sync marker
*/
-static int dca_convert_bitstream(uint8_t * src, int src_size, uint8_t * dst,
+static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst,
int max_size)
{
uint32_t mrk;
int i, tmp;
- uint16_t *ssrc = (uint16_t *) src, *sdst = (uint16_t *) dst;
+ const uint16_t *ssrc = (const uint16_t *) src;
+ uint16_t *sdst = (uint16_t *) dst;
PutBitContext pb;
- if((unsigned)src_size > (unsigned)max_size) {
- av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n");
- return -1;
+ if ((unsigned)src_size > (unsigned)max_size) {
+// av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n");
+// return -1;
+ src_size = max_size;
}
mrk = AV_RB32(src);
switch (mrk) {
case DCA_MARKER_RAW_BE:
- memcpy(dst, src, FFMIN(src_size, max_size));
- return FFMIN(src_size, max_size);
+ memcpy(dst, src, src_size);
+ return src_size;
case DCA_MARKER_RAW_LE:
- for (i = 0; i < (FFMIN(src_size, max_size) + 1) >> 1; i++)
- *sdst++ = bswap_16(*ssrc++);
- return FFMIN(src_size, max_size);
+ for (i = 0; i < (src_size + 1) >> 1; i++)
+ *sdst++ = av_bswap16(*ssrc++);
+ return src_size;
case DCA_MARKER_14B_BE:
case DCA_MARKER_14B_LE:
init_put_bits(&pb, dst, max_size);
*/
static int dca_decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
- uint8_t * buf, int buf_size)
+ AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
- int i, j, k;
+ int lfe_samples;
+ int num_core_channels = 0;
+ int i;
int16_t *samples = data;
DCAContext *s = avctx->priv_data;
int channels;
+ s->xch_present = 0;
s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer, DCA_MAX_FRAME_SIZE);
if (s->dca_buffer_size == -1) {
av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
avctx->sample_rate = s->sample_rate;
avctx->bit_rate = s->bit_rate;
- channels = s->prim_channels + !!s->lfe;
- if(avctx->request_channels == 2 && s->prim_channels > 2) {
- channels = 2;
- s->output = DCA_STEREO;
+ for (i = 0; i < (s->sample_blocks / 8); i++) {
+ dca_decode_block(s, 0, i);
}
- avctx->channels = channels;
- if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
- return -1;
- *data_size = 0;
- for (i = 0; i < (s->sample_blocks / 8); i++) {
- dca_decode_block(s);
- s->dsp.float_to_int16(s->tsamples, s->samples, 256 * channels);
- /* interleave samples */
- for (j = 0; j < 256; j++) {
- for (k = 0; k < channels; k++)
- samples[k] = s->tsamples[j + k * 256];
- samples += channels;
+ /* record number of core channels incase less than max channels are requested */
+ num_core_channels = s->prim_channels;
+
+ /* extensions start at 32-bit boundaries into bitstream */
+ skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
+
+ while(get_bits_left(&s->gb) >= 32) {
+ uint32_t bits = get_bits_long(&s->gb, 32);
+
+ switch(bits) {
+ case 0x5a5a5a5a: {
+ int ext_amode, xch_fsize;
+
+ s->xch_base_channel = s->prim_channels;
+
+ /* validate sync word using XCHFSIZE field */
+ xch_fsize = show_bits(&s->gb, 10);
+ if((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&
+ (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))
+ continue;
+
+ /* skip length-to-end-of-frame field for the moment */
+ skip_bits(&s->gb, 10);
+
+ /* extension amode should == 1, number of channels in extension */
+ /* AFAIK XCh is not used for more channels */
+ if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
+ av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"
+ " supported!\n",ext_amode);
+ continue;
+ }
+
+ /* much like core primary audio coding header */
+ dca_parse_audio_coding_header(s, s->xch_base_channel);
+
+ for (i = 0; i < (s->sample_blocks / 8); i++) {
+ dca_decode_block(s, s->xch_base_channel, i);
+ }
+
+ s->xch_present = 1;
+ break;
+ }
+ case 0x1d95f262:
+ av_log(avctx, AV_LOG_DEBUG, "Possible X96 extension found at %d bits\n", get_bits_count(&s->gb));
+ av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", get_bits(&s->gb, 12)+1);
+ av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
+ break;
}
- *data_size += 256 * sizeof(int16_t) * channels;
+
+ skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
}
- return buf_size;
-}
+ channels = s->prim_channels + !!s->lfe;
+ if (s->amode<16) {
+ avctx->channel_layout = dca_core_channel_layout[s->amode];
+
+ if (s->xch_present && (!avctx->request_channels ||
+ avctx->request_channels > num_core_channels)) {
+ avctx->channel_layout |= CH_BACK_CENTER;
+ if (s->lfe) {
+ avctx->channel_layout |= CH_LOW_FREQUENCY;
+ s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];
+ } else {
+ s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];
+ }
+ } else {
+ if (s->lfe) {
+ avctx->channel_layout |= CH_LOW_FREQUENCY;
+ s->channel_order_tab = dca_channel_reorder_lfe[s->amode];
+ } else
+ s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];
+ }
+ if (s->prim_channels > 0 &&
+ s->channel_order_tab[s->prim_channels - 1] < 0)
+ return -1;
-/**
- * Build the cosine modulation tables for the QMF
- *
- * @param s pointer to the DCAContext
- */
+ if (avctx->request_channels == 2 && s->prim_channels > 2) {
+ channels = 2;
+ s->output = DCA_STEREO;
+ avctx->channel_layout = CH_LAYOUT_STEREO;
+ }
+ } else {
+ av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode);
+ return -1;
+ }
-static void pre_calc_cosmod(DCAContext * s)
-{
- int i, j, k;
- static int cosmod_inited = 0;
- if(cosmod_inited) return;
- for (j = 0, k = 0; k < 16; k++)
- for (i = 0; i < 16; i++)
- cos_mod[j++] = cos((2 * i + 1) * (2 * k + 1) * M_PI / 64);
+ /* There is nothing that prevents a dts frame to change channel configuration
+ but FFmpeg doesn't support that so only set the channels if it is previously
+ unset. Ideally during the first probe for channels the crc should be checked
+ and only set avctx->channels when the crc is ok. Right now the decoder could
+ set the channels based on a broken first frame.*/
+ if (!avctx->channels)
+ avctx->channels = channels;
- for (k = 0; k < 16; k++)
- for (i = 0; i < 16; i++)
- cos_mod[j++] = cos((i) * (2 * k + 1) * M_PI / 32);
+ if (*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
+ return -1;
+ *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;
- for (k = 0; k < 16; k++)
- cos_mod[j++] = 0.25 / (2 * cos((2 * k + 1) * M_PI / 128));
+ /* filter to get final output */
+ for (i = 0; i < (s->sample_blocks / 8); i++) {
+ dca_filter_channels(s, i);
+ s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
+ samples += 256 * channels;
+ }
- for (k = 0; k < 16; k++)
- cos_mod[j++] = -0.25 / (2.0 * sin((2 * k + 1) * M_PI / 128));
+ /* update lfe history */
+ lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);
+ for (i = 0; i < 2 * s->lfe * 4; i++) {
+ s->lfe_data[i] = s->lfe_data[i + lfe_samples];
+ }
- cosmod_inited = 1;
+ return buf_size;
}
+
/**
* DCA initialization
*
* @param avctx pointer to the AVCodecContext
*/
-static int dca_decode_init(AVCodecContext * avctx)
+static av_cold int dca_decode_init(AVCodecContext * avctx)
{
DCAContext *s = avctx->priv_data;
+ int i;
s->avctx = avctx;
dca_init_vlcs();
- pre_calc_cosmod(s);
dsputil_init(&s->dsp, avctx);
+ ff_mdct_init(&s->imdct, 6, 1, 1.0);
+ ff_synth_filter_init(&s->synth);
+ ff_dcadsp_init(&s->dcadsp);
+
+ for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
+ s->samples_chanptr[i] = s->samples + i * 256;
+ avctx->sample_fmt = SAMPLE_FMT_S16;
+
+ if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
+ s->add_bias = 385.0f;
+ s->scale_bias = 1.0 / 32768.0;
+ } else {
+ s->add_bias = 0.0f;
+ s->scale_bias = 1.0;
- /* allow downmixing to stereo */
- if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
- avctx->request_channels == 2) {
- avctx->channels = avctx->request_channels;
+ /* allow downmixing to stereo */
+ if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
+ avctx->request_channels == 2) {
+ avctx->channels = avctx->request_channels;
+ }
}
+
return 0;
}
+static av_cold int dca_decode_end(AVCodecContext * avctx)
+{
+ DCAContext *s = avctx->priv_data;
+ ff_mdct_end(&s->imdct);
+ return 0;
+}
AVCodec dca_decoder = {
.name = "dca",
- .type = CODEC_TYPE_AUDIO,
+ .type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_DTS,
.priv_data_size = sizeof(DCAContext),
.init = dca_decode_init,
.decode = dca_decode_frame,
+ .close = dca_decode_end,
+ .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
};
projects / ffmpeg / blobdiff