* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-/**
- * @file libavcodec/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
/* Subband samples history (for ADPCM) */
float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
- DECLARE_ALIGNED_16(float, subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512]);
- float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][32];
+ 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
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, float, samples)[1536]; /* 6 * 256 = 1536, might only need 5 */
const float *samples_chanptr[6];
uint8_t dca_buffer[DCA_MAX_FRAME_SIZE];
int debug_flag; ///< used for suppressing repeated error messages output
DSPContext dsp;
- MDCTContext imdct;
+ FFTContext imdct;
+ SynthFilterContext synth;
+ DCADSPContext dcadsp;
} DCAContext;
+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_initialized = 0;
- int i, j;
+ int i, j, c = 14;
+ static VLC_TYPE dca_table[23622][2];
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;
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_initialized = 1;
}
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;
float scale, float bias)
{
const float *prCoeff;
- int i, j;
- DECLARE_ALIGNED_16(float, raXin[32]);
-
- int hist_index= s->hist_index[chans];
- float *subband_fir_hist2 = s->subband_fir_noidea[chans];
+ int i;
+ int sb_act = s->subband_activity[chans];
int subindex;
scale *= sqrt(1/8.0);
/* Reconstructed channel sample index */
for (subindex = 0; subindex < 8; subindex++) {
- float *subband_fir_hist = s->subband_fir_hist[chans] + hist_index;
/* Load in one sample from each subband and clear inactive subbands */
- for (i = 0; i < s->subband_activity[chans]; i++){
- if((i-1)&2) raXin[i] = -samples_in[i][subindex];
- else raXin[i] = samples_in[i][subindex];
+ 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++)
- raXin[i] = 0.0;
-
- ff_imdct_half(&s->imdct, subband_fir_hist, raXin);
-
- /* Multiply by filter coefficients */
- for (i = 0; i < 16; i++){
- float a= subband_fir_hist2[i ];
- float b= subband_fir_hist2[i+16];
- float c= 0;
- float d= 0;
- for (j = 0; j < 512-hist_index; j += 64){
- a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j]);
- b += prCoeff[i+j+16]*( subband_fir_hist[ i+j]);
- c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j]);
- d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j]);
- }
- for ( ; j < 512; j += 64){
- a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j-512]);
- b += prCoeff[i+j+16]*( subband_fir_hist[ i+j-512]);
- c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j-512]);
- d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j-512]);
- }
- samples_out[i ] = a * scale + bias;
- samples_out[i+16] = b * scale + bias;
- subband_fir_hist2[i ] = c;
- subband_fir_hist2[i+16] = d;
- }
+ s->raXin[i] = 0.0;
+
+ 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;
- hist_index = (hist_index-32)&511;
}
- s->hist_index[chans]= hist_index;
}
-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;
}
}
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)
const float *quant_step_table;
/* FIXME */
- float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
+ LOCAL_ALIGNED_16(float, subband_samples, [DCA_PRIM_CHANNELS_MAX], [DCA_SUBBANDS][8]);
+ LOCAL_ALIGNED_16(int, block, [8]);
/*
* Audio data
int abits = s->bitalloc[k][l];
float quant_step_size = quant_step_table[abits];
- float rscale;
/*
* Determine quantization index code book and its type
*/
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
if (s->output & DCA_LFE) {
int lfe_samples = 2 * s->lfe * s->subsubframes;
- lfe_interpolation_fir(s->lfe, 2 * s->lfe,
+ lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
s->lfe_data + lfe_samples +
2 * s->lfe * subsubframe,
&s->samples[256 * dca_lfe_index[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;
+
if(avctx->request_channels == 2 && s->prim_channels > 2) {
channels = 2;
s->output = DCA_STEREO;
dca_init_vlcs();
dsputil_init(&s->dsp, avctx);
- ff_mdct_init(&s->imdct, 6, 1);
+ ff_mdct_init(&s->imdct, 6, 1, 1.0);
+ ff_synth_filter_init(&s->synth);
+ ff_dcadsp_init(&s->dcadsp);
for(i = 0; i < 6; i++)
s->samples_chanptr[i] = s->samples + i * 256;
avctx->sample_fmt = SAMPLE_FMT_S16;
- if(s->dsp.float_to_int16 == ff_float_to_int16_c) {
+ 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 {
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,
projects / ffmpeg / blobdiff