#include "get_bits.h"
#include "internal.h"
#include "mathops.h"
+#include "profiles.h"
#include "put_bits.h"
#include "synth_filter.h"
#define DCA_NSYNCAUX 0x9A1105A0
-#define SAMPLES_PER_SUBBAND 8 // number of samples per subband per subsubframe
-
/** Bit allocation */
typedef struct BitAlloc {
int offset; ///< code values offset
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 uint8_t adj_table[4] = { 16, 18, 20, 23 };
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->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
- s->prim_channels = s->total_channels;
+ s->audio_header.total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
+ s->audio_header.prim_channels = s->audio_header.total_channels;
- if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
- s->prim_channels = DCA_PRIM_CHANNELS_MAX;
+ if (s->audio_header.prim_channels > DCA_PRIM_CHANNELS_MAX)
+ s->audio_header.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->audio_header.prim_channels; i++) {
+ s->audio_header.subband_activity[i] = get_bits(&s->gb, 5) + 2;
+ if (s->audio_header.subband_activity[i] > DCA_SUBBANDS)
+ s->audio_header.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;
+ for (i = base_channel; i < s->audio_header.prim_channels; i++) {
+ s->audio_header.vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
+ if (s->audio_header.vq_start_subband[i] > DCA_SUBBANDS)
+ s->audio_header.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_array(&s->gb, s->audio_header.joint_intensity + base_channel,
+ s->audio_header.prim_channels - base_channel, 3);
+ get_array(&s->gb, s->audio_header.transient_huffman + base_channel,
+ s->audio_header.prim_channels - base_channel, 2);
+ get_array(&s->gb, s->audio_header.scalefactor_huffman + base_channel,
+ s->audio_header.prim_channels - base_channel, 3);
+ get_array(&s->gb, s->audio_header.bitalloc_huffman + base_channel,
+ s->audio_header.prim_channels - base_channel, 3);
/* Get codebooks quantization indexes */
if (!base_channel)
- memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
+ memset(s->audio_header.quant_index_huffman, 0, sizeof(s->audio_header.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]);
+ for (i = base_channel; i < s->audio_header.prim_channels; i++)
+ s->audio_header.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 (i = base_channel; i < s->audio_header.prim_channels; i++)
+ s->audio_header.scalefactor_adj[i][j] = 16;
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)];
+ for (i = base_channel; i < s->audio_header.prim_channels; i++)
+ if (s->audio_header.quant_index_huffman[i][j] < thr[j])
+ s->audio_header.scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
if (s->crc_present) {
/* Audio header CRC check */
s->output |= DCA_LFE;
/* Primary audio coding header */
- s->subframes = get_bits(&s->gb, 4) + 1;
+ s->audio_header.subframes = get_bits(&s->gb, 4) + 1;
return dca_parse_audio_coding_header(s, 0);
}
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);
+ for (j = base_channel; j < s->audio_header.prim_channels; j++) {
+ for (k = 0; k < s->audio_header.subband_activity[j]; k++)
+ s->dca_chan[j].prediction_mode[k] = get_bits(&s->gb, 1);
}
/* Get prediction codebook */
- 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) {
+ for (j = base_channel; j < s->audio_header.prim_channels; j++) {
+ for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
+ if (s->dca_chan[j].prediction_mode[k] > 0) {
/* (Prediction coefficient VQ address) */
- s->prediction_vq[j][k] = get_bits(&s->gb, 12);
+ s->dca_chan[j].prediction_vq[k] = get_bits(&s->gb, 12);
}
}
}
/* Bit allocation index */
- 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 if (s->bitalloc_huffman[j] == 7) {
+ for (j = base_channel; j < s->audio_header.prim_channels; j++) {
+ for (k = 0; k < s->audio_header.vq_start_subband[j]; k++) {
+ if (s->audio_header.bitalloc_huffman[j] == 6)
+ s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 5);
+ else if (s->audio_header.bitalloc_huffman[j] == 5)
+ s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 4);
+ else if (s->audio_header.bitalloc_huffman[j] == 7) {
av_log(s->avctx, AV_LOG_ERROR,
"Invalid bit allocation index\n");
return AVERROR_INVALIDDATA;
} else {
- s->bitalloc[j][k] =
- get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
+ s->dca_chan[j].bitalloc[k] =
+ get_bitalloc(&s->gb, &dca_bitalloc_index, s->audio_header.bitalloc_huffman[j]);
}
- if (s->bitalloc[j][k] > 26) {
+ if (s->dca_chan[j].bitalloc[k] > 26) {
ff_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
- j, k, s->bitalloc[j][k]);
+ j, k, s->dca_chan[j].bitalloc[k]);
return AVERROR_INVALIDDATA;
}
}
}
/* Transition mode */
- for (j = base_channel; j < s->prim_channels; j++) {
- for (k = 0; k < s->subband_activity[j]; k++) {
- s->transition_mode[j][k] = 0;
+ for (j = base_channel; j < s->audio_header.prim_channels; j++) {
+ for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
+ s->dca_chan[j].transition_mode[k] = 0;
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]);
+ k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].bitalloc[k] > 0) {
+ s->dca_chan[j].transition_mode[k] =
+ get_bitalloc(&s->gb, &dca_tmode, s->audio_header.transient_huffman[j]);
}
}
}
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
- for (j = base_channel; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->audio_header.prim_channels; j++) {
const uint32_t *scale_table;
int scale_sum, log_size;
- memset(s->scale_factor[j], 0,
- s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
+ memset(s->dca_chan[j].scale_factor, 0,
+ s->audio_header.subband_activity[j] * sizeof(s->dca_chan[j].scale_factor[0][0]) * 2);
- if (s->scalefactor_huffman[j] == 6) {
+ if (s->audio_header.scalefactor_huffman[j] == 6) {
scale_table = ff_dca_scale_factor_quant7;
log_size = 7;
} else {
/* 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], scale_sum, log_size);
- s->scale_factor[j][k][0] = scale_table[scale_sum];
+ for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
+ if (k >= s->audio_header.vq_start_subband[j] || s->dca_chan[j].bitalloc[k] > 0) {
+ scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size);
+ s->dca_chan[j].scale_factor[k][0] = scale_table[scale_sum];
}
- if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
+ if (k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].transition_mode[k]) {
/* Get second scale factor */
- scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);
- s->scale_factor[j][k][1] = scale_table[scale_sum];
+ scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size);
+ s->dca_chan[j].scale_factor[k][1] = scale_table[scale_sum];
}
}
}
/* Joint subband scale factor codebook select */
- for (j = base_channel; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->audio_header.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);
+ if (s->audio_header.joint_intensity[j] > 0)
+ s->dca_chan[j].joint_huff = get_bits(&s->gb, 3);
}
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
/* Scale factors for joint subband coding */
- for (j = base_channel; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->audio_header.prim_channels; j++) {
int source_channel;
/* Transmitted only if joint subband coding enabled */
- if (s->joint_intensity[j] > 0) {
+ if (s->audio_header.joint_intensity[j] > 0) {
int scale = 0;
- source_channel = s->joint_intensity[j] - 1;
+ source_channel = s->audio_header.joint_intensity[j] - 1;
/* When huffman coded, only the difference is encoded
* (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], 64 /* bias */, 7);
- s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */
+ for (k = s->audio_header.subband_activity[j];
+ k < s->audio_header.subband_activity[source_channel]; k++) {
+ scale = get_scale(&s->gb, s->dca_chan[j].joint_huff, 64 /* bias */, 7);
+ s->dca_chan[j].joint_scale_factor[k] = scale; /*joint_scale_table[scale]; */
}
if (!(s->debug_flag & 0x02)) {
*/
/* VQ encoded high frequency subbands */
- for (j = base_channel; j < s->prim_channels; j++)
- for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
+ for (j = base_channel; j < s->audio_header.prim_channels; j++)
+ for (k = s->audio_header.vq_start_subband[j]; k < s->audio_header.subband_activity[j]; k++)
/* 1 vector -> 32 samples */
- s->high_freq_vq[j][k] = get_bits(&s->gb, 10);
+ s->dca_chan[j].high_freq_vq[k] = get_bits(&s->gb, 10);
/* Low frequency effect data */
if (!base_channel && s->lfe) {
}
static void qmf_32_subbands(DCAContext *s, int chans,
- float samples_in[32][SAMPLES_PER_SUBBAND], float *samples_out,
+ float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], float *samples_out,
float scale)
{
const float *prCoeff;
- int sb_act = s->subband_activity[chans];
+ int sb_act = s->audio_header.subband_activity[chans];
scale *= sqrt(1 / 8.0);
prCoeff = ff_dca_fir_32bands_perfect;
s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,
- s->subband_fir_hist[chans],
- &s->hist_index[chans],
- s->subband_fir_noidea[chans], prCoeff,
+ s->dca_chan[chans].subband_fir_hist,
+ &s->dca_chan[chans].hist_index,
+ s->dca_chan[chans].subband_fir_noidea, prCoeff,
samples_out, s->raXin, scale);
}
/* FIXME: Totally unoptimized. Based on the reference code and
* http://multimedia.cx/mirror/dca-transform.pdf, with guessed tweaks
* for doubling the size. */
-static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][SAMPLES_PER_SUBBAND],
+static void qmf_64_subbands(DCAContext *s, int chans,
+ float samples_in[DCA_SUBBANDS_X96K][SAMPLES_PER_SUBBAND],
float *samples_out, float scale)
{
float raXin[64];
float A[32], B[32];
- float *raX = s->subband_fir_hist[chans];
- float *raZ = s->subband_fir_noidea[chans];
+ float *raX = s->dca_chan[chans].subband_fir_hist;
+ float *raZ = s->dca_chan[chans].subband_fir_noidea;
unsigned i, j, k, subindex;
- for (i = s->subband_activity[chans]; i < 64; i++)
+ for (i = s->audio_header.subband_activity[chans]; i < DCA_SUBBANDS_X96K; i++)
raXin[i] = 0.0;
for (subindex = 0; subindex < SAMPLES_PER_SUBBAND; subindex++) {
- for (i = 0; i < s->subband_activity[chans]; i++)
+ for (i = 0; i < s->audio_header.subband_activity[chans]; i++)
raXin[i] = samples_in[i][subindex];
for (k = 0; k < 32; k++) {
raX[63 - k] = s->qmf64_table->rsin[k] * (A[k] - B[k]);
}
- for (i = 0; i < 64; i++) {
+ for (i = 0; i < DCA_SUBBANDS_X96K; i++) {
float out = raZ[i];
for (j = 0; j < 1024; j += 128)
out += ff_dca_fir_64bands[j + i] * (raX[j + i] - raX[j + 63 - i]);
*samples_out++ = out * scale;
}
- for (i = 0; i < 64; i++) {
+ for (i = 0; i < DCA_SUBBANDS_X96K; i++) {
float hist = 0.0;
for (j = 0; j < 1024; j += 128)
hist += ff_dca_fir_64bands[64 + j + i] * (-raX[i + j] - raX[j + 63 - i]);
{
int k, l;
int subsubframe = s->current_subsubframe;
-
- const float *quant_step_table;
-
- /* FIXME */
- float (*subband_samples)[DCA_SUBBANDS][SAMPLES_PER_SUBBAND] = s->subband_samples[block_index];
- LOCAL_ALIGNED_16(int32_t, block, [SAMPLES_PER_SUBBAND * DCA_SUBBANDS]);
+ const uint32_t *quant_step_table;
/*
* Audio data
/* Select quantization step size table */
if (s->bit_rate_index == 0x1f)
- quant_step_table = ff_dca_lossless_quant_d;
+ quant_step_table = ff_dca_lossless_quant;
else
- quant_step_table = ff_dca_lossy_quant_d;
+ quant_step_table = ff_dca_lossy_quant;
- for (k = base_channel; k < s->prim_channels; k++) {
- float rscale[DCA_SUBBANDS];
+ for (k = base_channel; k < s->audio_header.prim_channels; k++) {
+ int32_t (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index];
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
- for (l = 0; l < s->vq_start_subband[k]; l++) {
+ for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
int m;
/* Select the mid-tread linear quantizer */
- int abits = s->bitalloc[k][l];
-
- float quant_step_size = quant_step_table[abits];
-
- /*
- * Determine quantization index code book and its type
- */
+ int abits = s->dca_chan[k].bitalloc[l];
- /* Select quantization index code book */
- int sel = s->quant_index_huffman[k][abits];
+ uint32_t quant_step_size = quant_step_table[abits];
/*
* Extract bits from the bit stream
*/
- if (!abits) {
- rscale[l] = 0;
- memset(block + SAMPLES_PER_SUBBAND * l, 0, SAMPLES_PER_SUBBAND * sizeof(block[0]));
- } else {
+ if (!abits)
+ memset(subband_samples[l], 0, SAMPLES_PER_SUBBAND *
+ sizeof(subband_samples[l][0]));
+ else {
+ uint32_t rscale;
/* Deal with transients */
- int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
- rscale[l] = quant_step_size * s->scale_factor[k][l][sfi] *
- s->scalefactor_adj[k][sel];
+ int sfi = s->dca_chan[k].transition_mode[l] &&
+ subsubframe >= s->dca_chan[k].transition_mode[l];
+ /* Determine quantization index code book and its type.
+ Select quantization index code book */
+ int sel = s->audio_header.quant_index_huffman[k][abits];
+
+ rscale = (s->dca_chan[k].scale_factor[l][sfi] *
+ s->audio_header.scalefactor_adj[k][sel] + 8) >> 4;
if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
if (abits <= 7) {
block_code1 = get_bits(&s->gb, size);
block_code2 = get_bits(&s->gb, size);
err = decode_blockcodes(block_code1, block_code2,
- levels, block + SAMPLES_PER_SUBBAND * l);
+ levels, subband_samples[l]);
if (err) {
av_log(s->avctx, AV_LOG_ERROR,
"ERROR: block code look-up failed\n");
} else {
/* no coding */
for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
- block[SAMPLES_PER_SUBBAND * l + m] = get_sbits(&s->gb, abits - 3);
+ subband_samples[l][m] = get_sbits(&s->gb, abits - 3);
}
} else {
/* Huffman coded */
for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
- block[SAMPLES_PER_SUBBAND * l + m] = get_bitalloc(&s->gb,
- &dca_smpl_bitalloc[abits], sel);
+ subband_samples[l][m] = get_bitalloc(&s->gb,
+ &dca_smpl_bitalloc[abits], sel);
}
+ s->dcadsp.dequantize(subband_samples[l], quant_step_size, rscale);
}
}
- s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0],
- block, rscale, SAMPLES_PER_SUBBAND * s->vq_start_subband[k]);
-
- for (l = 0; l < s->vq_start_subband[k]; l++) {
+ for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
int m;
/*
* Inverse ADPCM if in prediction mode
*/
- if (s->prediction_mode[k][l]) {
+ if (s->dca_chan[k].prediction_mode[l]) {
int n;
if (s->predictor_history)
- subband_samples[k][l][0] += (ff_dca_adpcm_vb[s->prediction_vq[k][l]][0] *
- s->subband_samples_hist[k][l][3] +
- ff_dca_adpcm_vb[s->prediction_vq[k][l]][1] *
- s->subband_samples_hist[k][l][2] +
- ff_dca_adpcm_vb[s->prediction_vq[k][l]][2] *
- s->subband_samples_hist[k][l][1] +
- ff_dca_adpcm_vb[s->prediction_vq[k][l]][3] *
- s->subband_samples_hist[k][l][0]) *
- (1.0f / 8192);
+ subband_samples[l][0] += (ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
+ (int64_t)s->dca_chan[k].subband_samples_hist[l][3] +
+ ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] *
+ (int64_t)s->dca_chan[k].subband_samples_hist[l][2] +
+ ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] *
+ (int64_t)s->dca_chan[k].subband_samples_hist[l][1] +
+ ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] *
+ (int64_t)s->dca_chan[k].subband_samples_hist[l][0]) +
+ (1 << 12) >> 13;
for (m = 1; m < SAMPLES_PER_SUBBAND; m++) {
- float sum = ff_dca_adpcm_vb[s->prediction_vq[k][l]][0] *
- subband_samples[k][l][m - 1];
+ int64_t sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
+ (int64_t)subband_samples[l][m - 1];
for (n = 2; n <= 4; n++)
if (m >= n)
- sum += ff_dca_adpcm_vb[s->prediction_vq[k][l]][n - 1] *
- subband_samples[k][l][m - n];
+ sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
+ (int64_t)subband_samples[l][m - n];
else if (s->predictor_history)
- sum += ff_dca_adpcm_vb[s->prediction_vq[k][l]][n - 1] *
- s->subband_samples_hist[k][l][m - n + 4];
- subband_samples[k][l][m] += sum * 1.0f / 8192;
+ sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
+ (int64_t)s->dca_chan[k].subband_samples_hist[l][m - n + 4];
+ subband_samples[l][m] += (int32_t)(sum + (1 << 12) >> 13);
}
}
+
}
+ /* Backup predictor history for adpcm */
+ for (l = 0; l < DCA_SUBBANDS; l++)
+ AV_COPY128(s->dca_chan[k].subband_samples_hist[l], &subband_samples[l][4]);
+
/*
* Decode VQ encoded high frequencies
*/
- if (s->subband_activity[k] > s->vq_start_subband[k]) {
+ if (s->audio_header.subband_activity[k] > s->audio_header.vq_start_subband[k]) {
if (!s->debug_flag & 0x01) {
av_log(s->avctx, AV_LOG_DEBUG,
"Stream with high frequencies VQ coding\n");
s->debug_flag |= 0x01;
}
- s->dcadsp.decode_hf(subband_samples[k], s->high_freq_vq[k],
- ff_dca_high_freq_vq, subsubframe * SAMPLES_PER_SUBBAND,
- s->scale_factor[k], s->vq_start_subband[k],
- s->subband_activity[k]);
+
+ s->dcadsp.decode_hf(subband_samples, s->dca_chan[k].high_freq_vq,
+ ff_dca_high_freq_vq,
+ subsubframe * SAMPLES_PER_SUBBAND,
+ s->dca_chan[k].scale_factor,
+ s->audio_header.vq_start_subband[k],
+ s->audio_header.subband_activity[k]);
}
}
}
}
- /* Backup predictor history for adpcm */
- for (k = base_channel; k < s->prim_channels; k++)
- for (l = 0; l < s->vq_start_subband[k]; l++)
- AV_COPY128(s->subband_samples_hist[k][l], &subband_samples[k][l][4]);
-
return 0;
}
static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
{
- float (*subband_samples)[DCA_SUBBANDS][SAMPLES_PER_SUBBAND] = s->subband_samples[block_index];
int k;
if (upsample) {
+ LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS_X96K], [SAMPLES_PER_SUBBAND]);
+
if (!s->qmf64_table) {
s->qmf64_table = qmf64_precompute();
if (!s->qmf64_table)
}
/* 64 subbands QMF */
- for (k = 0; k < s->prim_channels; k++) {
+ for (k = 0; k < s->audio_header.prim_channels; k++) {
+ int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
+ s->dca_chan[k].subband_samples[block_index];
+
+ s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
+ DCA_SUBBANDS_X96K * SAMPLES_PER_SUBBAND);
+
if (s->channel_order_tab[k] >= 0)
- qmf_64_subbands(s, k, subband_samples[k],
+ qmf_64_subbands(s, k, samples,
s->samples_chanptr[s->channel_order_tab[k]],
/* Upsampling needs a factor 2 here. */
M_SQRT2 / 32768.0);
}
} else {
/* 32 subbands QMF */
- for (k = 0; k < s->prim_channels; k++) {
+ LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS], [SAMPLES_PER_SUBBAND]);
+
+ for (k = 0; k < s->audio_header.prim_channels; k++) {
+ int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
+ s->dca_chan[k].subband_samples[block_index];
+
+ s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
+ DCA_SUBBANDS * SAMPLES_PER_SUBBAND);
+
if (s->channel_order_tab[k] >= 0)
- qmf_32_subbands(s, k, subband_samples[k],
+ qmf_32_subbands(s, k, samples,
s->samples_chanptr[s->channel_order_tab[k]],
M_SQRT1_2 / 32768.0);
}
/* FIXME: This downmixing is probably broken with upsample.
* Probably totally broken also with XLL in general. */
/* Downmixing to Stereo */
- if (s->prim_channels + !!s->lfe > 2 &&
+ if (s->audio_header.prim_channels + !!s->lfe > 2 &&
s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,
s->channel_order_tab);
return AVERROR_INVALIDDATA;
}
for (out = 0; out < ff_dca_channels[s->core_downmix_amode]; out++) {
- for (in = 0; in < s->prim_channels + !!s->lfe; in++) {
+ for (in = 0; in < s->audio_header.prim_channels + !!s->lfe; in++) {
uint16_t tmp = get_bits(&s->gb, 9);
if ((tmp & 0xFF) > 241) {
av_log(s->avctx, AV_LOG_ERROR,
align_get_bits(&s->gb); // byte align
skip_bits(&s->gb, 16); // nAUXCRC16
- // additional data (reserved, cf. ETSI TS 102 114 V1.4.1)
- if ((reserved = (aux_data_end - get_bits_count(&s->gb))) < 0) {
- av_log(s->avctx, AV_LOG_ERROR,
- "Overread auxiliary data by %d bits\n", -reserved);
- return AVERROR_INVALIDDATA;
- } else if (reserved) {
+ /*
+ * additional data (reserved, cf. ETSI TS 102 114 V1.4.1)
+ *
+ * Note: don't check for overreads, aux_data_count can't be trusted.
+ */
+ if ((reserved = (aux_data_end - get_bits_count(&s->gb))) > 0) {
avpriv_request_sample(s->avctx,
"Core auxiliary data reserved content");
skip_bits_long(&s->gb, reserved);
int ret;
/* Sanity check */
- if (s->current_subframe >= s->subframes) {
+ if (s->current_subframe >= s->audio_header.subframes) {
av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
- s->current_subframe, s->subframes);
+ s->current_subframe, s->audio_header.subframes);
return AVERROR_INVALIDDATA;
}
s->current_subsubframe = 0;
s->current_subframe++;
}
- if (s->current_subframe >= s->subframes) {
+ if (s->current_subframe >= s->audio_header.subframes) {
/* Read subframe footer */
if ((ret = dca_subframe_footer(s, base_channel)))
return ret;
case DCA_SYNCWORD_XCH: {
int ext_amode, xch_fsize;
- s->xch_base_channel = s->prim_channels;
+ s->xch_base_channel = s->audio_header.prim_channels;
/* validate sync word using XCHFSIZE field */
xch_fsize = show_bits(&s->gb, 10);
if (s->amode < 16) {
avctx->channel_layout = dca_core_channel_layout[s->amode];
- if (s->prim_channels + !!s->lfe > 2 &&
+ if (s->audio_header.prim_channels + !!s->lfe > 2 &&
avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
/*
* Neither the core's auxiliary data nor our default tables contain
if (num_core_channels + !!s->lfe > 2 &&
avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
channels = 2;
- s->output = s->prim_channels == 2 ? s->amode : DCA_STEREO;
+ s->output = s->audio_header.prim_channels == 2 ? s->amode : DCA_STEREO;
avctx->channel_layout = AV_CH_LAYOUT_STEREO;
/* Stereo downmix coefficients
if (num_core_channels + !!s->lfe >
FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) {
avpriv_request_sample(s->avctx, "Downmixing %d channels",
- s->prim_channels + !!s->lfe);
+ s->audio_header.prim_channels + !!s->lfe);
return AVERROR_PATCHWELCOME;
}
for (i = 0; i < num_core_channels + !!s->lfe; i++) {
ff_dlog(s->avctx, "\n");
}
} else {
- av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);
+ av_log(avctx, AV_LOG_ERROR, "Nonstandard configuration %d !\n", s->amode);
return AVERROR_INVALIDDATA;
}
}
/* record number of core channels incase less than max channels are requested */
- num_core_channels = s->prim_channels;
+ num_core_channels = s->audio_header.prim_channels;
if (s->ext_coding)
s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];
avctx->profile = s->profile;
- full_channels = channels = s->prim_channels + !!s->lfe;
+ full_channels = channels = s->audio_header.prim_channels + !!s->lfe;
ret = set_channel_layout(avctx, channels, num_core_channels);
if (ret < 0)
return 0;
}
-static const AVProfile profiles[] = {
- { FF_PROFILE_DTS, "DTS" },
- { FF_PROFILE_DTS_ES, "DTS-ES" },
- { FF_PROFILE_DTS_96_24, "DTS 96/24" },
- { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },
- { FF_PROFILE_DTS_HD_MA, "DTS-HD MA" },
- { FF_PROFILE_UNKNOWN },
-};
-
static const AVOption options[] = {
{ "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
{ "disable_xll", "disable decoding of the XLL extension", offsetof(DCAContext, xll_disable), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
- .profiles = NULL_IF_CONFIG_SMALL(profiles),
+ .profiles = NULL_IF_CONFIG_SMALL(ff_dca_profiles),
.priv_class = &dca_decoder_class,
};
projects / ffmpeg / blobdiff