*/
/**
- * @file dca.c
+ * @file libavcodec/dca.c
*/
#include <math.h>
#include "avcodec.h"
#include "dsputil.h"
-#include "bitstream.h"
+#include "get_bits.h"
+#include "put_bits.h"
#include "dcadata.h"
#include "dcahuff.h"
-#include "parser.h"
-
-/** DCA syncwords, also used for bitstream type detection */
-//@{
-#define DCA_MARKER_RAW_BE 0x7FFE8001
-#define DCA_MARKER_RAW_LE 0xFE7F0180
-#define DCA_MARKER_14B_BE 0x1FFFE800
-#define DCA_MARKER_14B_LE 0xFF1F00E8
-//@}
+#include "dca.h"
+#include "synth_filter.h"
//#define TRACE
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[][8] = {
+ { 0, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 4, -1, -1, -1, -1},
+ { 0, 1, 3, 4, -1, -1, -1, -1},
+ { 2, 0, 1, 4, 5, -1, -1, -1},
+ { 3, 4, 0, 1, 5, 6, -1, -1},
+ { 2, 0, 1, 4, 5, 6, -1, -1},
+ { 0, 6, 4, 5, 2, 3, -1, -1},
+ { 4, 2, 5, 0, 1, 6, 7, -1},
+ { 5, 6, 0, 1, 7, 3, 8, 4},
+ { 4, 2, 5, 0, 1, 6, 8, 7},
+};
+
+static const int8_t dca_channel_reorder_nolfe[][8] = {
+ { 0, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 3, -1, -1, -1, -1},
+ { 0, 1, 2, 3, -1, -1, -1, -1},
+ { 2, 0, 1, 3, 4, -1, -1, -1},
+ { 2, 3, 0, 1, 4, 5, -1, -1},
+ { 2, 0, 1, 3, 4, 5, -1, -1},
+ { 0, 5, 3, 4, 1, 2, -1, -1},
+ { 3, 2, 4, 0, 1, 5, 6, -1},
+ { 4, 5, 0, 1, 6, 2, 7, 3},
+ { 3, 2, 4, 0, 1, 5, 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 int av_always_inline get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx)
+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
/* 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];
+ 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, samples)[1536]; /* 6 * 256 = 1536, might only need 5 */
+ const float *samples_chanptr[6];
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 debug_flag; ///< used for suppressing repeated error messages output
DSPContext dsp;
+ FFTContext imdct;
} DCAContext;
-static void dca_init_vlcs()
+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 = 1;
- for (i = 0; i < 5; i++)
+ dca_bitalloc_index.wrap = 2;
+ 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;
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)
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;
-
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;
s->dialog_norm = get_bits(&s->gb, 4);
/* FIXME: channels mixing levels */
- s->output = DCA_STEREO;
+ s->output = s->amode;
+ 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;
+ s->total_channels = get_bits(&s->gb, 3) + 1;
+ s->prim_channels = s->total_channels;
+ if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
+ s->prim_channels = DCA_PRIM_CHANNELS_MAX; /* We only support DTS core */
for (i = 0; i < s->prim_channels; i++) {
}
-static inline int get_scale(GetBitContext *gb, int level, int index, int value)
+static inline int get_scale(GetBitContext *gb, int level, int value)
{
if (level < 5) {
/* huffman encoded */
- value += get_bitalloc(gb, &dca_scalefactor, index);
+ value += get_bitalloc(gb, &dca_scalefactor, level);
} else if(level < 8)
value = get_bits(gb, level + 1);
return value;
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, j);
+ get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
}
if (s->bitalloc[j][k] > 26) {
}
for (j = 0; j < s->prim_channels; j++) {
- uint32_t *scale_table;
+ 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;
for (k = 0; k < s->subband_activity[j]; k++) {
if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
- scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], j, scale_sum);
+ scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
s->scale_factor[j][k][0] = scale_table[scale_sum];
}
if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
/* Get second scale factor */
- scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], j, scale_sum);
+ scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
s->scale_factor[j][k][1] = scale_table[scale_sum];
}
}
* (is this valid as well for joint scales ???) */
for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {
- scale = get_scale(&s->gb, s->joint_huff[j], j, 0);
+ scale = get_scale(&s->gb, s->joint_huff[j], 0);
scale += 64; /* bias */
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 && s->downmix) {
- for (j = 0; 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);
+ if (s->prim_channels > 2) {
+ if(s->downmix) {
+ for (j = 0; 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++) {
+ s->downmix_coef[j][0] = dca_default_coeffs[am][j][0];
+ s->downmix_coef[j][1] = dca_default_coeffs[am][j][1];
+ }
}
}
}
for (j = 0; 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");
for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++)
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]);
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;
av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
for (j = lfe_samples; j < lfe_samples * 2; j++)
av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
float samples_in[32][8], float *samples_out,
float scale, float bias)
{
- 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];
+ const float *prCoeff;
+ int i;
- int chindex = 0, subindex;
+ int subindex;
- praXin[0] = 0.0;
+ scale *= sqrt(1/8.0);
/* Select filter */
if (!s->multirate_inter) /* Non-perfect reconstruction */
- prCoeff = (float *) fir_32bands_nonperfect;
+ prCoeff = fir_32bands_nonperfect;
else /* Perfect reconstruction */
- prCoeff = (float *) fir_32bands_perfect;
+ prCoeff = fir_32bands_perfect;
/* 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 < s->subband_activity[chans]; i++){
+ if((i-1)&2) s->raXin[i] = -samples_in[i][subindex];
+ else s->raXin[i] = samples_in[i][subindex];
}
+ 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;
+ ff_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));
}
}
//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;
+ samples_out[interp_index++] = (rTmp * scale) + bias;
}
}
}
/* downmixing routines */
-#define MIX_REAR1(samples, si1) \
- samples[i] += samples[si1]; \
- samples[i+256] += samples[si1];
+#define MIX_REAR1(samples, si1, rs, coef) \
+ samples[i] += samples[si1] * coef[rs][0]; \
+ samples[i+256] += samples[si1] * coef[rs][1];
-#define MIX_REAR2(samples, si1, si2) \
- samples[i] += samples[si1]; \
- samples[i+256] += samples[si2];
+#define MIX_REAR2(samples, si1, si2, rs, coef) \
+ samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \
+ samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1];
-#define MIX_FRONT3(samples) \
+#define MIX_FRONT3(samples, coef) \
t = samples[i]; \
- samples[i] += samples[i+256]; \
- samples[i+256] = samples[i+512] + t;
+ samples[i] = t * coef[0][0] + samples[i+256] * coef[1][0] + samples[i+512] * coef[2][0]; \
+ 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++){ \
op2 \
}
-static void dca_downmix(float *samples, int srcfmt)
+static void dca_downmix(float *samples, int srcfmt,
+ int downmix_coef[DCA_PRIM_CHANNELS_MAX][2])
{
int i;
float t;
+ float coef[DCA_PRIM_CHANNELS_MAX][2];
+
+ 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]];
+ }
switch (srcfmt) {
case DCA_MONO:
case DCA_STEREO:
break;
case DCA_3F:
- DOWNMIX_TO_STEREO(MIX_FRONT3(samples),);
+ DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);
break;
case DCA_2F1R:
- DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512),);
+ DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512, 2, coef),);
break;
case DCA_3F1R:
- DOWNMIX_TO_STEREO(MIX_FRONT3(samples),
- MIX_REAR1(samples, i + 768));
+ DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
+ MIX_REAR1(samples, i + 768, 3, coef));
break;
case DCA_2F2R:
- DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768),);
+ DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768, 2, coef),);
break;
case DCA_3F2R:
- DOWNMIX_TO_STEREO(MIX_FRONT3(samples),
- MIX_REAR2(samples, i + 768, i + 1024));
+ DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
+ MIX_REAR2(samples, i + 768, i + 1024, 3, coef));
break;
}
}
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];
*/
/* 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 (l = 0; l < s->vq_start_subband[k]; l++) {
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]) {
- dca_downmix(s->samples, s->amode);
+ 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],
- 8388608.0, s->bias);
+ &s->samples[256 * dca_lfe_index[s->amode]],
+ (1.0/256.0)*s->scale_bias, s->add_bias);
/* Outputs 20bits pcm samples */
}
/**
* 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;
+ 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++)
+ for (i = 0; i < (src_size + 1) >> 1; i++)
*sdst++ = bswap_16(*ssrc++);
- return FFMIN(src_size, max_size);
+ 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 i;
int16_t *samples = data;
DCAContext *s = avctx->priv_data;
int channels;
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 DCA frame\n");
+ av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
return -1;
}
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
if (dca_parse_frame_header(s) < 0) {
//seems like the frame is corrupt, try with the next one
+ *data_size=0;
return buf_size;
}
//set AVCodec values with parsed data
avctx->sample_rate = s->sample_rate;
- avctx->channels = 2; //FIXME
avctx->bit_rate = s->bit_rate;
- channels = dca_channels[s->output];
+ channels = s->prim_channels + !!s->lfe;
+
+ if (s->amode<16) {
+ avctx->channel_layout = dca_core_channel_layout[s->amode];
+
+ 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(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;
+ }
+
+
+ /* 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;
+
if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
return -1;
- *data_size = 0;
+ *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;
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;
- }
- *data_size += 256 * sizeof(int16_t) * channels;
+ s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
+ samples += 256 * channels;
}
return buf_size;
-/**
- * Build the cosine modulation tables for the QMF
- *
- * @param s pointer to the DCAContext
- */
-
-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);
-
- for (k = 0; k < 16; k++)
- for (i = 0; i < 16; i++)
- cos_mod[j++] = cos((i) * (2 * k + 1) * M_PI / 32);
-
- for (k = 0; k < 16; k++)
- cos_mod[j++] = 0.25 / (2 * cos((2 * k + 1) * M_PI / 128));
-
- for (k = 0; k < 16; k++)
- cos_mod[j++] = -0.25 / (2.0 * sin((2 * k + 1) * M_PI / 128));
-
- cosmod_inited = 1;
-}
-
-
/**
* 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);
- return 0;
-}
-
-
-AVCodec dca_decoder = {
- .name = "dca",
- .type = CODEC_TYPE_AUDIO,
- .id = CODEC_ID_DTS,
- .priv_data_size = sizeof(DCAContext),
- .init = dca_decode_init,
- .decode = dca_decode_frame,
-};
+ ff_mdct_init(&s->imdct, 6, 1, 1.0);
-#ifdef CONFIG_DCA_PARSER
+ for(i = 0; i < 6; i++)
+ s->samples_chanptr[i] = s->samples + i * 256;
+ avctx->sample_fmt = SAMPLE_FMT_S16;
-typedef struct DCAParseContext {
- ParseContext pc;
- uint32_t lastmarker;
-} DCAParseContext;
-
-#define IS_MARKER(state, i, buf, buf_size) \
- ((state == DCA_MARKER_14B_LE && (i < buf_size-2) && (buf[i+1] & 0xF0) == 0xF0 && buf[i+2] == 0x07) \
- || (state == DCA_MARKER_14B_BE && (i < buf_size-2) && buf[i+1] == 0x07 && (buf[i+2] & 0xF0) == 0xF0) \
- || state == DCA_MARKER_RAW_LE || state == DCA_MARKER_RAW_BE)
+ 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;
-/**
- * finds the end of the current frame in the bitstream.
- * @return the position of the first byte of the next frame, or -1
- */
-static int dca_find_frame_end(DCAParseContext * pc1, const uint8_t * buf,
- int buf_size)
-{
- int start_found, i;
- uint32_t state;
- ParseContext *pc = &pc1->pc;
-
- start_found = pc->frame_start_found;
- state = pc->state;
-
- i = 0;
- if (!start_found) {
- for (i = 0; i < buf_size; i++) {
- state = (state << 8) | buf[i];
- if (IS_MARKER(state, i, buf, buf_size)) {
- if (pc1->lastmarker && state == pc1->lastmarker) {
- start_found = 1;
- break;
- } else if (!pc1->lastmarker) {
- start_found = 1;
- pc1->lastmarker = state;
- break;
- }
- }
+ /* allow downmixing to stereo */
+ if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
+ avctx->request_channels == 2) {
+ avctx->channels = avctx->request_channels;
}
}
- if (start_found) {
- for (; i < buf_size; i++) {
- state = (state << 8) | buf[i];
- if (state == pc1->lastmarker && IS_MARKER(state, i, buf, buf_size)) {
- pc->frame_start_found = 0;
- pc->state = -1;
- return i - 3;
- }
- }
- }
- pc->frame_start_found = start_found;
- pc->state = state;
- return END_NOT_FOUND;
-}
-static int dca_parse_init(AVCodecParserContext * s)
-{
- DCAParseContext *pc1 = s->priv_data;
- pc1->lastmarker = 0;
return 0;
}
-static int dca_parse(AVCodecParserContext * s,
- AVCodecContext * avctx,
- uint8_t ** poutbuf, int *poutbuf_size,
- const uint8_t * buf, int buf_size)
+static av_cold int dca_decode_end(AVCodecContext * avctx)
{
- DCAParseContext *pc1 = s->priv_data;
- ParseContext *pc = &pc1->pc;
- int next;
-
- if (s->flags & PARSER_FLAG_COMPLETE_FRAMES) {
- next = buf_size;
- } else {
- next = dca_find_frame_end(pc1, buf, buf_size);
-
- if (ff_combine_frame(pc, next, (uint8_t **) & buf, &buf_size) < 0) {
- *poutbuf = NULL;
- *poutbuf_size = 0;
- return buf_size;
- }
- }
- *poutbuf = (uint8_t *) buf;
- *poutbuf_size = buf_size;
- return next;
+ DCAContext *s = avctx->priv_data;
+ ff_mdct_end(&s->imdct);
+ return 0;
}
-AVCodecParser dca_parser = {
- {CODEC_ID_DTS},
- sizeof(DCAParseContext),
- dca_parse_init,
- dca_parse,
- ff_parse_close,
+AVCodec dca_decoder = {
+ .name = "dca",
+ .type = CODEC_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)"),
};
-#endif /* CONFIG_DCA_PARSER */
projects / ffmpeg / blobdiff