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dcaenc: Implementation of Huffman codes for DCA encoder

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Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
pull/247/head
Daniil Cherednik 8 years ago committed by Rostislav Pehlivanov
parent a6191d098a
commit c2500d62c6
9 changed files with 217 additions and 103 deletions
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  1. 2
      libavcodec/Makefile
  2. 16
      libavcodec/dca_core.c
  3. 8
      libavcodec/dcadata.c
  4. 5
      libavcodec/dcadata.h
  5. 241
      libavcodec/dcaenc.c
  6. 1
      libavcodec/dcaenc.h
  7. 22
      libavcodec/dcahuff.c
  8. 3
      libavcodec/dcahuff.h
  9. 2
      tests/fate/acodec.mak

2
libavcodec/Makefile
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@ -235,7 +235,7 @@ OBJS-$(CONFIG_CYUV_DECODER) += cyuv.o
OBJS-$(CONFIG_DCA_DECODER) += dcadec.o dca.o dcadata.o dcahuff.o \
dca_core.o dca_exss.o dca_xll.o dca_lbr.o \
dcadsp.o dcadct.o synth_filter.o
OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o dca.o dcadata.o
OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o dca.o dcadata.o dcahuff.o
OBJS-$(CONFIG_DDS_DECODER) += dds.o
OBJS-$(CONFIG_DIRAC_DECODER) += diracdec.o dirac.o diracdsp.o diractab.o \
dirac_arith.o dirac_dwt.o dirac_vlc.o

16
libavcodec/dca_core.c
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@ -92,14 +92,6 @@ static const uint8_t block_code_nbits[7] = {
7, 10, 12, 13, 15, 17, 19
};
static const uint8_t quant_index_sel_nbits[DCA_CODE_BOOKS] = {
1, 2, 2, 2, 2, 3, 3, 3, 3, 3
};
static const uint8_t quant_index_group_size[DCA_CODE_BOOKS] = {
1, 3, 3, 3, 3, 7, 7, 7, 7, 7
};
static int dca_get_vlc(GetBitContext *s, DCAVLC *v, int i)
{
return get_vlc2(s, v->vlc[i].table, v->vlc[i].bits, v->max_depth) + v->offset;
@ -400,12 +392,12 @@ static int parse_coding_header(DCACoreDecoder *s, enum HeaderType header, int xc
// Quantization index codebook select
for (n = 0; n < DCA_CODE_BOOKS; n++)
for (ch = xch_base; ch < s->nchannels; ch++)
s->quant_index_sel[ch][n] = get_bits(&s->gb, quant_index_sel_nbits[n]);
s->quant_index_sel[ch][n] = get_bits(&s->gb, ff_dca_quant_index_sel_nbits[n]);
// Scale factor adjustment index
for (n = 0; n < DCA_CODE_BOOKS; n++)
for (ch = xch_base; ch < s->nchannels; ch++)
if (s->quant_index_sel[ch][n] < quant_index_group_size[n])
if (s->quant_index_sel[ch][n] < ff_dca_quant_index_group_size[n])
s->scale_factor_adj[ch][n] = ff_dca_scale_factor_adj[get_bits(&s->gb, 2)];
if (header == HEADER_XXCH) {
@ -663,7 +655,7 @@ static inline int extract_audio(DCACoreDecoder *s, int32_t *audio, int abits, in
if (abits <= DCA_CODE_BOOKS) {
int sel = s->quant_index_sel[ch][abits - 1];
if (sel < quant_index_group_size[abits - 1]) {
if (sel < ff_dca_quant_index_group_size[abits - 1]) {
// Huffman codes
return parse_huffman_codes(s, audio, abits, sel);
}
@ -1562,7 +1554,7 @@ static int parse_x96_coding_header(DCACoreDecoder *s, int exss, int xch_base)
// Quantization index codebook select
for (n = 0; n < 6 + 4 * s->x96_high_res; n++)
for (ch = xch_base; ch < s->x96_nchannels; ch++)
s->quant_index_sel[ch][n] = get_bits(&s->gb, quant_index_sel_nbits[n]);
s->quant_index_sel[ch][n] = get_bits(&s->gb, ff_dca_quant_index_sel_nbits[n]);
if (exss) {
// Reserved

8
libavcodec/dcadata.c
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@ -50,6 +50,14 @@ const uint8_t ff_dca_dmix_primary_nch[8] = {
1, 2, 2, 3, 3, 4, 4, 0
};
const uint8_t ff_dca_quant_index_sel_nbits[DCA_CODE_BOOKS] = {
1, 2, 2, 2, 2, 3, 3, 3, 3, 3
};
const uint8_t ff_dca_quant_index_group_size[DCA_CODE_BOOKS] = {
1, 3, 3, 3, 3, 7, 7, 7, 7, 7
};
/* ADPCM data */
/* 16 bits signed fractional Q13 binary codes */

5
libavcodec/dcadata.h
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@ -23,6 +23,8 @@
#include <stdint.h>
#include "dcahuff.h"
extern const uint32_t ff_dca_bit_rates[32];
extern const uint8_t ff_dca_channels[16];
@ -31,6 +33,9 @@ extern const uint8_t ff_dca_bits_per_sample[8];
extern const uint8_t ff_dca_dmix_primary_nch[8];
extern const uint8_t ff_dca_quant_index_sel_nbits[DCA_CODE_BOOKS];
extern const uint8_t ff_dca_quant_index_group_size[DCA_CODE_BOOKS];
extern const int16_t ff_dca_adpcm_vb[4096][4];
extern const uint32_t ff_dca_scale_factor_quant6[64];

241
libavcodec/dcaenc.c
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@ -70,6 +70,7 @@ typedef struct DCAEncContext {
int abits[MAX_CHANNELS][DCAENC_SUBBANDS];
int scale_factor[MAX_CHANNELS][DCAENC_SUBBANDS];
softfloat quant[MAX_CHANNELS][DCAENC_SUBBANDS];
int32_t quant_index_sel[MAX_CHANNELS][DCA_CODE_BOOKS];
int32_t eff_masking_curve_cb[256];
int32_t band_masking_cb[32];
int32_t worst_quantization_noise;
@ -109,7 +110,7 @@ static int encode_init(AVCodecContext *avctx)
{
DCAEncContext *c = avctx->priv_data;
uint64_t layout = avctx->channel_layout;
int i, min_frame_bits;
int i, j, min_frame_bits;
c->fullband_channels = c->channels = avctx->channels;
c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6);
@ -142,6 +143,12 @@ static int encode_init(AVCodecContext *avctx)
c->channel_order_tab = channel_reorder_nolfe[c->channel_config];
}
for (i = 0; i < MAX_CHANNELS; i++) {
for (j = 0; j < DCA_CODE_BOOKS; j++) {
c->quant_index_sel[i][j] = ff_dca_quant_index_group_size[j];
}
}
for (i = 0; i < 9; i++) {
if (sample_rates[i] == avctx->sample_rate)
break;
@ -568,9 +575,109 @@ static const int snr_fudge = 128;
#define USED_NABITS 2
#define USED_26ABITS 4
static int32_t quantize_value(int32_t value, softfloat quant)
{
int32_t offset = 1 << (quant.e - 1);
value = mul32(value, quant.m) + offset;
value = value >> quant.e;
return value;
}
static int calc_one_scale(int32_t peak_cb, int abits, softfloat *quant)
{
int32_t peak;
int our_nscale, try_remove;
softfloat our_quant;
av_assert0(peak_cb <= 0);
av_assert0(peak_cb >= -2047);
our_nscale = 127;
peak = cb_to_level[-peak_cb];
for (try_remove = 64; try_remove > 0; try_remove >>= 1) {
if (scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e <= 17)
continue;
our_quant.m = mul32(scalefactor_inv[our_nscale - try_remove].m, stepsize_inv[abits].m);
our_quant.e = scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e - 17;
if ((ff_dca_quant_levels[abits] - 1) / 2 < quantize_value(peak, our_quant))
continue;
our_nscale -= try_remove;
}
if (our_nscale >= 125)
our_nscale = 124;
quant->m = mul32(scalefactor_inv[our_nscale].m, stepsize_inv[abits].m);
quant->e = scalefactor_inv[our_nscale].e + stepsize_inv[abits].e - 17;
av_assert0((ff_dca_quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant));
return our_nscale;
}
static void quantize_all(DCAEncContext *c)
{
int sample, band, ch;
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < 32; band++)
for (sample = 0; sample < SUBBAND_SAMPLES; sample++)
c->quantized[ch][band][sample] = quantize_value(c->subband[ch][band][sample], c->quant[ch][band]);
}
static void accumulate_huff_bit_consumption(int abits, int32_t *quantized, uint32_t *result)
{
uint8_t sel, id = abits - 1;
for (sel = 0; sel < ff_dca_quant_index_group_size[id]; sel++)
result[sel] += ff_dca_vlc_calc_quant_bits(quantized, SUBBAND_SAMPLES, sel, id);
}
static uint32_t set_best_code(uint32_t vlc_bits[DCA_CODE_BOOKS][7], uint32_t clc_bits[DCA_CODE_BOOKS], int32_t res[DCA_CODE_BOOKS])
{
uint8_t i, sel;
uint32_t best_sel_bits[DCA_CODE_BOOKS];
int32_t best_sel_id[DCA_CODE_BOOKS];
uint32_t t, bits = 0;
for (i = 0; i < DCA_CODE_BOOKS; i++) {
av_assert0(!((!!vlc_bits[i][0]) ^ (!!clc_bits[i])));
if (vlc_bits[i][0] == 0) {
/* do not transmit adjustment index for empty codebooks */
res[i] = ff_dca_quant_index_group_size[i];
/* and skip it */
continue;
}
best_sel_bits[i] = vlc_bits[i][0];
best_sel_id[i] = 0;
for (sel = 0; sel < ff_dca_quant_index_group_size[i]; sel++) {
if (best_sel_bits[i] > vlc_bits[i][sel] && vlc_bits[i][sel]) {
best_sel_bits[i] = vlc_bits[i][sel];
best_sel_id[i] = sel;
}
}
/* 2 bits to transmit scale factor adjustment index */
t = best_sel_bits[i] + 2;
if (t < clc_bits[i]) {
res[i] = best_sel_id[i];
bits += t;
} else {
res[i] = ff_dca_quant_index_group_size[i];
bits += clc_bits[i];
}
}
return bits;
}
static int init_quantization_noise(DCAEncContext *c, int noise)
{
int ch, band, ret = 0;
uint32_t huff_bit_count_accum[MAX_CHANNELS][DCA_CODE_BOOKS][7];
uint32_t clc_bit_count_accum[MAX_CHANNELS][DCA_CODE_BOOKS];
uint32_t bits_counter = 0;
c->consumed_bits = 132 + 493 * c->fullband_channels;
if (c->lfe_channel)
@ -597,11 +704,37 @@ static int init_quantization_noise(DCAEncContext *c, int noise)
}
}
for (ch = 0; ch < c->fullband_channels; ch++)
/* Recalc scale_factor each time to get bits consumption in case of Huffman coding.
It is suboptimal solution */
/* TODO: May be cache scaled values */
for (ch = 0; ch < c->fullband_channels; ch++) {
for (band = 0; band < 32; band++) {
c->scale_factor[ch][band] = calc_one_scale(c->peak_cb[ch][band],
c->abits[ch][band],
&c->quant[ch][band]);
}
}
quantize_all(c);
memset(huff_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * 7 * sizeof(uint32_t));
memset(clc_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * sizeof(uint32_t));
for (ch = 0; ch < c->fullband_channels; ch++) {
for (band = 0; band < 32; band++) {
c->consumed_bits += bit_consumption[c->abits[ch][band]];
if (c->abits[ch][band] && c->abits[ch][band] <= DCA_CODE_BOOKS) {
accumulate_huff_bit_consumption(c->abits[ch][band], c->quantized[ch][band], huff_bit_count_accum[ch][c->abits[ch][band] - 1]);
clc_bit_count_accum[ch][c->abits[ch][band] - 1] += bit_consumption[c->abits[ch][band]];
} else {
bits_counter += bit_consumption[c->abits[ch][band]];
}
}
}
for (ch = 0; ch < c->fullband_channels; ch++) {
bits_counter += set_best_code(huff_bit_count_accum[ch], clc_bit_count_accum[ch], c->quant_index_sel[ch]);
}
c->consumed_bits += bits_counter;
return ret;
}
@ -655,71 +788,12 @@ static void shift_history(DCAEncContext *c, const int32_t *input)
}
}
static int32_t quantize_value(int32_t value, softfloat quant)
{
int32_t offset = 1 << (quant.e - 1);
value = mul32(value, quant.m) + offset;
value = value >> quant.e;
return value;
}
static int calc_one_scale(int32_t peak_cb, int abits, softfloat *quant)
static void calc_lfe_scales(DCAEncContext *c)
{
int32_t peak;
int our_nscale, try_remove;
softfloat our_quant;
av_assert0(peak_cb <= 0);
av_assert0(peak_cb >= -2047);
our_nscale = 127;
peak = cb_to_level[-peak_cb];
for (try_remove = 64; try_remove > 0; try_remove >>= 1) {
if (scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e <= 17)
continue;
our_quant.m = mul32(scalefactor_inv[our_nscale - try_remove].m, stepsize_inv[abits].m);
our_quant.e = scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e - 17;
if ((ff_dca_quant_levels[abits] - 1) / 2 < quantize_value(peak, our_quant))
continue;
our_nscale -= try_remove;
}
if (our_nscale >= 125)
our_nscale = 124;
quant->m = mul32(scalefactor_inv[our_nscale].m, stepsize_inv[abits].m);
quant->e = scalefactor_inv[our_nscale].e + stepsize_inv[abits].e - 17;
av_assert0((ff_dca_quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant));
return our_nscale;
}
static void calc_scales(DCAEncContext *c)
{
int band, ch;
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < 32; band++)
c->scale_factor[ch][band] = calc_one_scale(c->peak_cb[ch][band],
c->abits[ch][band],
&c->quant[ch][band]);
if (c->lfe_channel)
c->lfe_scale_factor = calc_one_scale(c->lfe_peak_cb, 11, &c->lfe_quant);
}
static void quantize_all(DCAEncContext *c)
{
int sample, band, ch;
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < 32; band++)
for (sample = 0; sample < SUBBAND_SAMPLES; sample++)
c->quantized[ch][band][sample] = quantize_value(c->subband[ch][band][sample], c->quant[ch][band]);
}
static void put_frame_header(DCAEncContext *c)
{
/* SYNC */
@ -805,9 +879,6 @@ static void put_frame_header(DCAEncContext *c)
static void put_primary_audio_header(DCAEncContext *c)
{
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 ch, i;
/* Number of subframes */
put_bits(&c->pb, 4, SUBFRAMES - 1);
@ -839,20 +910,34 @@ static void put_primary_audio_header(DCAEncContext *c)
for (ch = 0; ch < c->fullband_channels; ch++)
put_bits(&c->pb, 3, 6);
/* Quantization index codebook select: dummy data
to avoid transmission of scale factor adjustment */
for (i = 1; i < 11; i++)
/* Quantization index codebook select */
for (i = 0; i < DCA_CODE_BOOKS; i++)
for (ch = 0; ch < c->fullband_channels; ch++)
put_bits(&c->pb, bitlen[i], thr[i]);
put_bits(&c->pb, ff_dca_quant_index_sel_nbits[i], c->quant_index_sel[ch][i]);
/* Scale factor adjustment index: transmitted in case of Huffman coding */
for (i = 0; i < DCA_CODE_BOOKS; i++)
for (ch = 0; ch < c->fullband_channels; ch++)
if (c->quant_index_sel[ch][i] < ff_dca_quant_index_group_size[i])
put_bits(&c->pb, 2, 0);
/* Scale factor adjustment index: not transmitted */
/* Audio header CRC check word: not transmitted */
}
static void put_subframe_samples(DCAEncContext *c, int ss, int band, int ch)
{
int i, j, sum, bits, sel;
if (c->abits[ch][band] <= DCA_CODE_BOOKS) {
av_assert0(c->abits[ch][band] > 0);
sel = c->quant_index_sel[ch][c->abits[ch][band] - 1];
// Huffman codes
if (sel < ff_dca_quant_index_group_size[c->abits[ch][band] - 1]) {
ff_dca_vlc_enc_quant(&c->pb, &c->quantized[ch][band][ss * 8], 8, sel, c->abits[ch][band] - 1);
return;
}
// Block codes
if (c->abits[ch][band] <= 7) {
int sum, i, j;
for (i = 0; i < 8; i += 4) {
sum = 0;
for (j = 3; j >= 0; j--) {
@ -862,13 +947,14 @@ static void put_subframe_samples(DCAEncContext *c, int ss, int band, int ch)
}
put_bits(&c->pb, bit_consumption[c->abits[ch][band]] / 4, sum);
}
} else {
int i;
return;
}
}
for (i = 0; i < 8; i++) {
int bits = bit_consumption[c->abits[ch][band]] / 16;
bits = bit_consumption[c->abits[ch][band]] / 16;
put_sbits(&c->pb, bits, c->quantized[ch][band][ss * 8 + i]);
}
}
}
static void put_subframe(DCAEncContext *c, int subframe)
@ -947,8 +1033,7 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
calc_masking(c, samples);
find_peaks(c);
assign_bits(c);
calc_scales(c);
quantize_all(c);
calc_lfe_scales(c);
shift_history(c, samples);
init_put_bits(&c->pb, avpkt->data, avpkt->size);

1
libavcodec/dcaenc.h
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@ -95,7 +95,6 @@ static const softfloat scalefactor_inv[128] = {
/* manually derived from
* Table B.5: Selection of quantization levels and codebooks
* FIXME: will become invalid when Huffman codes are introduced.
*/
static const int bit_consumption[27] = {
-8, 28, 40, 48, 52, 60, 68, 76, 80, 96,

22
libavcodec/dcahuff.c
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@ -1335,3 +1335,25 @@ av_cold void ff_dca_init_vlcs(void)
vlcs_initialized = 1;
}
uint32_t ff_dca_vlc_calc_quant_bits(int *values, uint8_t n, uint8_t sel, uint8_t table)
{
uint8_t i, id;
uint32_t sum = 0;
for (i = 0; i < n; i++) {
id = values[i] - bitalloc_offsets[table];
av_assert0(id < bitalloc_sizes[table]);
sum += bitalloc_bits[table][sel][id];
}
return sum;
}
void ff_dca_vlc_enc_quant(PutBitContext *pb, int *values, uint8_t n, uint8_t sel, uint8_t table)
{
uint8_t i, id;
for (i = 0; i < n; i++) {
id = values[i] - bitalloc_offsets[table];
av_assert0(id < bitalloc_sizes[table]);
put_bits(pb, bitalloc_bits[table][sel][id], bitalloc_codes[table][sel][id]);
}
}

3
libavcodec/dcahuff.h
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@ -27,6 +27,7 @@
#include "avcodec.h"
#include "get_bits.h"
#include "put_bits.h"
#define DCA_CODE_BOOKS 10
@ -55,5 +56,7 @@ extern VLC ff_dca_vlc_grid_3;
extern VLC ff_dca_vlc_rsd;
av_cold void ff_dca_init_vlcs(void);
uint32_t ff_dca_vlc_calc_quant_bits(int *values, uint8_t n, uint8_t sel, uint8_t abits);
void ff_dca_vlc_enc_quant(PutBitContext *pb, int *values, uint8_t n, uint8_t sel, uint8_t abits);
#endif /* AVCODEC_DCAHUFF_H */

2
tests/fate/acodec.mak
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@ -104,7 +104,7 @@ fate-acodec-dca: tests/data/asynth-44100-2.wav
fate-acodec-dca: SRC = tests/data/asynth-44100-2.wav
fate-acodec-dca: CMD = md5 -i $(TARGET_PATH)/$(SRC) -c:a dca -strict -2 -f dts -flags +bitexact
fate-acodec-dca: CMP = oneline
fate-acodec-dca: REF = 7ffdefdf47069289990755c79387cc90
fate-acodec-dca: REF = 7cd79a3717943a06b217f1130223a86f
FATE_ACODEC-$(call ENCDEC, DCA, WAV) += fate-acodec-dca2
fate-acodec-dca2: CMD = enc_dec_pcm dts wav s16le $(SRC) -c:a dca -strict -2 -flags +bitexact

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