*/
/**
- * @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 "dca.h"
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 16384
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
/* 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];
+ 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];
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 */
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;
+ MDCTContext imdct;
} 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;
}
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;
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);
float scale, float bias)
{
const float *prCoeff;
- int i, j, k;
- float praXin[33], *raXin = &praXin[1];
+ 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 chindex = 0, subindex;
+ int subindex;
- praXin[0] = 0.0;
+ scale *= sqrt(1/8.0);
/* Select filter */
if (!s->multirate_inter) /* Non-perfect reconstruction */
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++)
- raXin[i] = samples_in[i][subindex];
+ 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 < 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++) {
- float t1 = 0.0;
- float 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];
- }
- subband_fir_hist[ k ] = cos_mod[k+512 ] * (t1 + t2);
- subband_fir_hist[32-k-1] = cos_mod[k+512+16] * (t1 - t2);
- }
+ ff_imdct_half(&s->imdct, subband_fir_hist, raXin);
/* Multiply by filter coefficients */
- for (k = 31, i = 0; i < 32; i++, k--){
- float a= subband_fir_hist2[i];
- float b= 0;
+ 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[i+j] - subband_fir_hist[j+k]);
- b += prCoeff[i+j+32]*(-subband_fir_hist[i+j] - subband_fir_hist[j+k]);
+ 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[i+j-512] - subband_fir_hist[j+k-512]);
- b += prCoeff[i+j+32]*(-subband_fir_hist[i+j-512] - subband_fir_hist[j+k-512]);
+ 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[chindex++] = a * scale + bias;
- subband_fir_hist2[i] = b;
+ samples_out[i ] = a * scale + bias;
+ samples_out[i+16] = b * scale + bias;
+ subband_fir_hist2[i ] = c;
+ subband_fir_hist2[i+16] = d;
}
+ samples_out+= 32;
hist_index = (hist_index-32)&511;
}
//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;
}
}
}
*/
/* Select quantization step size table */
- if (s->bit_rate == 0x1f)
+ if (s->bit_rate_index == 0x1f)
quant_step_table = lossless_quant_d;
else
quant_step_table = lossy_quant_d;
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],
- M_SQRT1_2 /*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 */
/* 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 */);
+ &s->samples[256 * dca_lfe_index[s->amode]],
+ (1.0/256.0)*s->scale_bias, s->add_bias);
/* Outputs 20bits pcm samples */
}
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;
+// 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);
*/
static int dca_decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
- const uint8_t * buf, int buf_size)
+ AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
int i;
int16_t *samples = data;
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;
+
+ 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
-/**
- * Build the cosine modulation tables for the QMF
- *
- * @param s pointer to the DCAContext
- */
-
-static av_cold void pre_calc_cosmod(DCAContext * s)
-{
- int i, j, k;
- static int cosmod_initialized = 0;
-
- if(cosmod_initialized) 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_initialized = 1;
-}
-
-
/**
* DCA initialization
*
s->avctx = avctx;
dca_init_vlcs();
- pre_calc_cosmod(s);
dsputil_init(&s->dsp, avctx);
+ ff_mdct_init(&s->imdct, 6, 1);
- /* allow downmixing to stereo */
- if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
- avctx->request_channels == 2) {
- avctx->channels = avctx->request_channels;
- }
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) {
+ 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;
+ }
+ }
+
+
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",
.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 / frescor / ffmpeg.git / blobdiff