#include "get_bits.h"
#include "internal.h"
#include "mathops.h"
+#include "profiles.h"
#include "synth_filter.h"
#if ARCH_ARM
int xxch)
{
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 };
int hdr_pos = 0, hdr_size = 0;
/* Get scale factor adjustment */
for (j = 0; j < 11; j++)
for (i = base_channel; i < s->audio_header.prim_channels; i++)
- s->audio_header.scalefactor_adj[i][j] = 1;
+ s->audio_header.scalefactor_adj[i][j] = 16;
for (j = 1; j < 11; j++)
for (i = base_channel; i < s->audio_header.prim_channels; i++)
{
int k, l;
int subsubframe = s->current_subsubframe;
-
- const float *quant_step_table;
-
- 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->audio_header.prim_channels; k++) {
- float (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index];
- float rscale[DCA_SUBBANDS];
+ int32_t (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index];
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
/* Select the mid-tread linear quantizer */
int abits = s->dca_chan[k].bitalloc[l];
- float quant_step_size = quant_step_table[abits];
-
- /*
- * Determine quantization index code book and its type
- */
-
- /* Select quantization index code book */
- int sel = s->audio_header.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->dca_chan[k].transition_mode[l] &&
subsubframe >= s->dca_chan[k].transition_mode[l];
- rscale[l] = quant_step_size * s->dca_chan[k].scale_factor[l][sfi] *
- s->audio_header.scalefactor_adj[k][sel];
+ /* 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[0],
- block, rscale, SAMPLES_PER_SUBBAND * s->audio_header.vq_start_subband[k]);
-
for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
int m;
/*
int n;
if (s->predictor_history)
subband_samples[l][0] += (ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
- s->dca_chan[k].subband_samples_hist[l][3] +
- ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] *
- s->dca_chan[k].subband_samples_hist[l][2] +
- ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] *
- s->dca_chan[k].subband_samples_hist[l][1] +
- ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] *
- s->dca_chan[k].subband_samples_hist[l][0]) *
- (1.0f / 8192);
+ (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->dca_chan[k].prediction_vq[l]][0] *
- subband_samples[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->dca_chan[k].prediction_vq[l]][n - 1] *
- subband_samples[l][m - n];
+ (int64_t)subband_samples[l][m - n];
else if (s->predictor_history)
sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
- s->dca_chan[k].subband_samples_hist[l][m - n + 4];
- subband_samples[l][m] += sum * (1.0f / 8192);
+ (int64_t)s->dca_chan[k].subband_samples_hist[l][m - n + 4];
+ subband_samples[l][m] += (int32_t)(sum + (1 << 12) >> 13);
}
}
}
s->dcadsp.decode_hf(subband_samples, s->dca_chan[k].high_freq_vq,
- ff_dca_high_freq_vq, subsubframe * SAMPLES_PER_SUBBAND,
+ 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]);
int k;
if (upsample) {
+ LOCAL_ALIGNED(32, float, samples, [64], [SAMPLES_PER_SUBBAND]);
+
if (!s->qmf64_table) {
s->qmf64_table = qmf64_precompute();
if (!s->qmf64_table)
/* 64 subbands QMF */
for (k = 0; k < s->audio_header.prim_channels; k++) {
- float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index];
+ 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],
+ 64 * SAMPLES_PER_SUBBAND);
if (s->channel_order_tab[k] >= 0)
- qmf_64_subbands(s, k, subband_samples,
+ 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 */
+ LOCAL_ALIGNED(32, float, samples, [32], [SAMPLES_PER_SUBBAND]);
+
for (k = 0; k < s->audio_header.prim_channels; k++) {
- float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index];
+ 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],
+ 32 * SAMPLES_PER_SUBBAND);
if (s->channel_order_tab[k] >= 0)
- qmf_32_subbands(s, k, subband_samples,
+ qmf_32_subbands(s, k, samples,
s->samples_chanptr[s->channel_order_tab[k]],
M_SQRT1_2 / 32768.0);
}
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_BOOL, { .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_BOOL, { .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