generic-library/ffmpeg
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

dcadec: make a number of samples per subband per subsubframe a named constant

Browse Source 
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This commit is contained in:
Alexandra Khirnova 2015-09-18 13:34:20 +02:00 committed by Luca Barbato
parent 8ae1d87a24
commit d0a3e89d41
1 changed files with 21 additions and 19 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

40
libavcodec/dcadec.c
View File

@ -139,6 +139,8 @@ static const uint64_t dca_core_channel_layout[] = {
#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
@ -536,7 +538,7 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
}
static void qmf_32_subbands(DCAContext *s, int chans,
float samples_in[32][8], float *samples_out,
float samples_in[32][SAMPLES_PER_SUBBAND], float *samples_out,
float scale)
{
const float *prCoeff;
@ -584,7 +586,7 @@ static QMF64_table *qmf64_precompute(void)
/* 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][8],
static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][SAMPLES_PER_SUBBAND],
float *samples_out, float scale)
{
float raXin[64];
@ -595,7 +597,7 @@ static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][8],
for (i = s->subband_activity[chans]; i < 64; i++)
raXin[i] = 0.0;
for (subindex = 0; subindex < 8; subindex++) {
for (subindex = 0; subindex < SAMPLES_PER_SUBBAND; subindex++) {
for (i = 0; i < s->subband_activity[chans]; i++)
raXin[i] = samples_in[i][subindex];
@ -786,8 +788,8 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
const float *quant_step_table;
/* FIXME */
float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
LOCAL_ALIGNED_16(int32_t, block, [8 * DCA_SUBBANDS]);
float (*subband_samples)[DCA_SUBBANDS][SAMPLES_PER_SUBBAND] = s->subband_samples[block_index];
LOCAL_ALIGNED_16(int32_t, block, [SAMPLES_PER_SUBBAND * DCA_SUBBANDS]);
/*
* Audio data
@ -825,7 +827,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
*/
if (!abits) {
rscale[l] = 0;
memset(block + 8 * l, 0, 8 * sizeof(block[0]));
memset(block + SAMPLES_PER_SUBBAND * l, 0, SAMPLES_PER_SUBBAND * sizeof(block[0]));
} else {
/* Deal with transients */
int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
@ -843,7 +845,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
block_code1 = get_bits(&s->gb, size);
block_code2 = get_bits(&s->gb, size);
err = decode_blockcodes(block_code1, block_code2,
levels, block + 8 * l);
levels, block + SAMPLES_PER_SUBBAND * l);
if (err) {
av_log(s->avctx, AV_LOG_ERROR,
"ERROR: block code look-up failed\n");
@ -851,20 +853,20 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
}
} else {
/* no coding */
for (m = 0; m < 8; m++)
block[8 * l + m] = get_sbits(&s->gb, abits - 3);
for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
block[SAMPLES_PER_SUBBAND * l + m] = get_sbits(&s->gb, abits - 3);
}
} else {
/* Huffman coded */
for (m = 0; m < 8; m++)
block[8 * l + m] = get_bitalloc(&s->gb,
for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
block[SAMPLES_PER_SUBBAND * l + m] = get_bitalloc(&s->gb,
&dca_smpl_bitalloc[abits], sel);
}
}
}
s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0],
block, rscale, 8 * s->vq_start_subband[k]);
block, rscale, SAMPLES_PER_SUBBAND * s->vq_start_subband[k]);
for (l = 0; l < s->vq_start_subband[k]; l++) {
int m;
@ -883,7 +885,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
ff_dca_adpcm_vb[s->prediction_vq[k][l]][3] *
s->subband_samples_hist[k][l][0]) *
(1.0f / 8192);
for (m = 1; m < 8; m++) {
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];
for (n = 2; n <= 4; n++)
@ -908,7 +910,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
s->debug_flag |= 0x01;
}
s->dcadsp.decode_hf(subband_samples[k], s->high_freq_vq[k],
ff_dca_high_freq_vq, subsubframe * 8,
ff_dca_high_freq_vq, subsubframe * SAMPLES_PER_SUBBAND,
s->scale_factor[k], s->vq_start_subband[k],
s->subband_activity[k]);
}
@ -932,7 +934,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
{
float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
float (*subband_samples)[DCA_SUBBANDS][SAMPLES_PER_SUBBAND] = s->subband_samples[block_index];
int k;
if (upsample) {
@ -1377,7 +1379,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
s->profile = FF_PROFILE_DTS;
for (i = 0; i < (s->sample_blocks / 8); i++) {
for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); i++) {
if ((ret = dca_decode_block(s, 0, i))) {
av_log(avctx, AV_LOG_ERROR, "error decoding block\n");
return ret;
@ -1404,7 +1406,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
avctx->channels = channels;
/* get output buffer */
frame->nb_samples = 256 * (s->sample_blocks / 8);
frame->nb_samples = 256 * (s->sample_blocks / SAMPLES_PER_SUBBAND);
if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
int xll_nb_samples = s->xll_segments * s->xll_smpl_in_seg;
/* Check for invalid/unsupported conditions first */
@ -1470,7 +1472,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
}
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); i++) {
int ch;
unsigned block = upsample ? 512 : 256;
for (ch = 0; ch < channels; ch++)
@ -1492,7 +1494,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
}
/* update lfe history */
lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);
lfe_samples = 2 * s->lfe * (s->sample_blocks / SAMPLES_PER_SUBBAND);
for (i = 0; i < 2 * s->lfe * 4; i++)
s->lfe_data[i] = s->lfe_data[i + lfe_samples];