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dca: use defines for subband related constants

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Signed-off-by: Janne Grunau <janne-libav@jannau.net>
This commit is contained in:
Alexandra Hájková 2015-12-30 15:29:03 +01:00 committed by Janne Grunau
parent 8563f98871
commit 40d9496773
4 changed files with 20 additions and 18 deletions
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2
libavcodec/dca.h
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@ -147,7 +147,7 @@ typedef struct DCAAudioHeader {
} DCAAudioHeader;
typedef struct DCAChan {
DECLARE_ALIGNED(32, int32_t, subband_samples)[DCA_BLOCKS_MAX][DCA_SUBBANDS][8];
DECLARE_ALIGNED(32, int32_t, subband_samples)[DCA_BLOCKS_MAX][DCA_SUBBANDS][SAMPLES_PER_SUBBAND];
/* Subband samples history (for ADPCM) */
DECLARE_ALIGNED(32, int32_t, subband_samples_hist)[DCA_SUBBANDS][4];

21
libavcodec/dcadec.c
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@ -140,8 +140,6 @@ 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
@ -544,7 +542,7 @@ 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][SAMPLES_PER_SUBBAND], float *samples_out,
float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], float *samples_out,
float scale)
{
const float *prCoeff;
@ -592,7 +590,8 @@ 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][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];
@ -601,7 +600,7 @@ static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][SAMPL
float *raZ = s->dca_chan[chans].subband_fir_noidea;
unsigned i, j, k, subindex;
for (i = s->audio_header.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->audio_header.subband_activity[chans]; i++)
@ -622,14 +621,14 @@ static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][SAMPL
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]);
@ -938,7 +937,7 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
int k;
if (upsample) {
LOCAL_ALIGNED(32, float, samples, [64], [SAMPLES_PER_SUBBAND]);
LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS_X96K], [SAMPLES_PER_SUBBAND]);
if (!s->qmf64_table) {
s->qmf64_table = qmf64_precompute();
@ -952,7 +951,7 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
s->dca_chan[k].subband_samples[block_index];
s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
64 * SAMPLES_PER_SUBBAND);
DCA_SUBBANDS_X96K * SAMPLES_PER_SUBBAND);
if (s->channel_order_tab[k] >= 0)
qmf_64_subbands(s, k, samples,
@ -962,14 +961,14 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
}
} else {
/* 32 subbands QMF */
LOCAL_ALIGNED(32, float, samples, [32], [SAMPLES_PER_SUBBAND]);
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],
32 * SAMPLES_PER_SUBBAND);
DCA_SUBBANDS * SAMPLES_PER_SUBBAND);
if (s->channel_order_tab[k] >= 0)
qmf_32_subbands(s, k, samples,

6
libavcodec/dcadsp.c
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@ -27,7 +27,7 @@
#include "dcadsp.h"
#include "dcamath.h"
static void decode_hf_c(int32_t dst[DCA_SUBBANDS][8],
static void decode_hf_c(int32_t dst[DCA_SUBBANDS][SAMPLES_PER_SUBBAND],
const int32_t vq_num[DCA_SUBBANDS],
const int8_t hf_vq[1024][32], intptr_t vq_offset,
int32_t scale[DCA_SUBBANDS][2],
@ -62,7 +62,7 @@ static inline void dca_lfe_fir(float *out, const float *in, const float *coefs,
}
}
static void dca_qmf_32_subbands(float samples_in[32][8], int sb_act,
static void dca_qmf_32_subbands(float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], int sb_act,
SynthFilterContext *synth, FFTContext *imdct,
float synth_buf_ptr[512],
int *synth_buf_offset, float synth_buf2[32],
@ -103,7 +103,7 @@ static void dequantize_c(int32_t *samples, uint32_t step_size, uint32_t scale)
shift = 0;
step_scale = (int32_t)(step >> shift);
for (i = 0; i < 8; i++)
for (i = 0; i < SAMPLES_PER_SUBBAND; i++)
samples[i] = dca_clip23(dca_norm((int64_t)samples[i] * step_scale, 22 - shift));
}

9
libavcodec/dcadsp.h
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@ -22,17 +22,20 @@
#include "avfft.h"
#include "synth_filter.h"
#define DCA_SUBBANDS 32
#define DCA_SUBBANDS_X96K 64
#define DCA_SUBBANDS 32
#define SAMPLES_PER_SUBBAND 8 // number of samples per subband per subsubframe
typedef struct DCADSPContext {
void (*lfe_fir[2])(float *out, const float *in, const float *coefs);
void (*qmf_32_subbands)(float samples_in[32][8], int sb_act,
void (*qmf_32_subbands)(float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], int sb_act,
SynthFilterContext *synth, FFTContext *imdct,
float synth_buf_ptr[512],
int *synth_buf_offset, float synth_buf2[32],
const float window[512], float *samples_out,
float raXin[32], float scale);
void (*decode_hf)(int32_t dst[DCA_SUBBANDS][8],
void (*decode_hf)(int32_t dst[DCA_SUBBANDS][SAMPLES_PER_SUBBAND],
const int32_t vq_num[DCA_SUBBANDS],
const int8_t hf_vq[1024][32], intptr_t vq_offset,
int32_t scale[DCA_SUBBANDS][2],