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aacenc: Move local encoder specific tables to a separate file
This commit moves any tables specific to the encoder from aacenc and aaccoder to a separate file called 'aacenctab.c/.h'. This was done as a clean up attempt as the encoder was filled with tables pasted in between functions which made it confusing to follow and track where each table and definition had been used. This commit solves this by simply exporting the smaller tables out to the aacenctab.h while the larger ones are compiled using aacenctab.c and are referenced from the header file. Signed-off-by: Claudio Freire <klaussfreire@gmail.com>
This commit is contained in:
Rostislav Pehlivanov
Claudio Freire
parent
9c0407e856
commit
c47c781e83
@ -130,7 +130,7 @@ OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o aacps_float
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OBJS-$(CONFIG_AAC_FIXED_DECODER) += aacdec_fixed.o aactab.o aacsbr_fixed.o aacps_fixed.o \
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aacadtsdec.o mpeg4audio.o kbdwin.o \
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sbrdsp_fixed.o aacpsdsp_fixed.o
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OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o \
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OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o aacenctab.o \
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aacpsy.o aactab.o \
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psymodel.o mpeg4audio.o kbdwin.o
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OBJS-$(CONFIG_AASC_DECODER) += aasc.o msrledec.o
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@ -39,6 +39,7 @@
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#include "aac.h"
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#include "aacenc.h"
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#include "aactab.h"
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#include "aacenctab.h"
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#include "aac_tablegen_decl.h"
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/** Frequency in Hz for lower limit of noise substitution **/
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@ -56,37 +57,9 @@
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/** Frequency in Hz for lower limit of intensity stereo **/
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#define INT_STEREO_LOW_LIMIT 6100
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/** Total number of usable codebooks **/
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#define CB_TOT 12
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/** Total number of codebooks, including special ones **/
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#define CB_TOT_ALL 15
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/** bits needed to code codebook run value for long windows */
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static const uint8_t run_value_bits_long[64] = {
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5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
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5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 10,
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10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
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10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
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};
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/** bits needed to code codebook run value for short windows */
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static const uint8_t run_value_bits_short[16] = {
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3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
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};
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static const uint8_t * const run_value_bits[2] = {
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run_value_bits_long, run_value_bits_short
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};
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#define ROUND_STANDARD 0.4054f
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#define ROUND_TO_ZERO 0.1054f
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/** Map to convert values from BandCodingPath index to a codebook index **/
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static const uint8_t aac_cb_out_map[CB_TOT_ALL] = {0,1,2,3,4,5,6,7,8,9,10,11,13,14,15};
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/** Inverse map to convert from codebooks to BandCodingPath indices **/
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static const uint8_t aac_cb_in_map[CB_TOT_ALL+1] = {0,1,2,3,4,5,6,7,8,9,10,11,0,12,13,14};
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/**
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* Quantize one coefficient.
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* @return absolute value of the quantized coefficient
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@ -123,9 +96,6 @@ static void abs_pow34_v(float *out, const float *in, const int size)
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#endif /* USE_REALLY_FULL_SEARCH */
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}
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static const uint8_t aac_cb_range [12] = {0, 3, 3, 3, 3, 9, 9, 8, 8, 13, 13, 17};
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static const uint8_t aac_cb_maxval[12] = {0, 1, 1, 2, 2, 4, 4, 7, 7, 12, 12, 16};
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/**
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* Calculate rate distortion cost for quantizing with given codebook
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*
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@ -42,11 +42,10 @@
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#include "aac.h"
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#include "aactab.h"
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#include "aacenc.h"
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#include "aacenctab.h"
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#include "psymodel.h"
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#define AAC_MAX_CHANNELS 6
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#define ERROR_IF(cond, ...) \
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if (cond) { \
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av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
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@ -58,110 +57,6 @@
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av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \
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}
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static const uint8_t swb_size_1024_96[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
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64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
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};
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static const uint8_t swb_size_1024_64[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
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12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
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40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
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};
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static const uint8_t swb_size_1024_48[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
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32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
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96
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};
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static const uint8_t swb_size_1024_32[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
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32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
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};
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static const uint8_t swb_size_1024_24[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
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32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
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};
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static const uint8_t swb_size_1024_16[] = {
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8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
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32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
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};
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static const uint8_t swb_size_1024_8[] = {
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12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
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16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
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32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
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};
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static const uint8_t *swb_size_1024[] = {
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swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
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swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
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swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
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swb_size_1024_16, swb_size_1024_16, swb_size_1024_8,
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swb_size_1024_8
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};
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static const uint8_t swb_size_128_96[] = {
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4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
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};
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static const uint8_t swb_size_128_48[] = {
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4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
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};
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static const uint8_t swb_size_128_24[] = {
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4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
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};
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static const uint8_t swb_size_128_16[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
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};
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static const uint8_t swb_size_128_8[] = {
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4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
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};
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static const uint8_t *swb_size_128[] = {
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/* the last entry on the following row is swb_size_128_64 but is a
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duplicate of swb_size_128_96 */
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swb_size_128_96, swb_size_128_96, swb_size_128_96,
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swb_size_128_48, swb_size_128_48, swb_size_128_48,
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swb_size_128_24, swb_size_128_24, swb_size_128_16,
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swb_size_128_16, swb_size_128_16, swb_size_128_8,
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swb_size_128_8
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};
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/** default channel configurations */
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static const uint8_t aac_chan_configs[6][5] = {
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{1, TYPE_SCE}, // 1 channel - single channel element
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{1, TYPE_CPE}, // 2 channels - channel pair
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{2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
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{3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
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{3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
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{4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
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};
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/**
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* Table to remap channels from libavcodec's default order to AAC order.
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*/
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static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
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{ 0 },
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{ 0, 1 },
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{ 2, 0, 1 },
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{ 2, 0, 1, 3 },
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{ 2, 0, 1, 3, 4 },
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{ 2, 0, 1, 4, 5, 3 },
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};
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/**
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* Make AAC audio config object.
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* @see 1.6.2.1 "Syntax - AudioSpecificConfig"
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@ -843,9 +738,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
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s->channels = avctx->channels;
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ERROR_IF(i == 16
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|| i >= (sizeof(swb_size_1024) / sizeof(*swb_size_1024))
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|| i >= (sizeof(swb_size_128) / sizeof(*swb_size_128)),
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ERROR_IF(i == 16 || i >= swb_size_1024_len || i >= swb_size_128_len,
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"Unsupported sample rate %d\n", avctx->sample_rate);
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ERROR_IF(s->channels > AAC_MAX_CHANNELS,
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"Unsupported number of channels: %d\n", s->channels);
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@ -926,13 +819,6 @@ static const AVClass aacenc_class = {
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LIBAVUTIL_VERSION_INT,
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};
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/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
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* failures */
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static const int mpeg4audio_sample_rates[16] = {
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96000, 88200, 64000, 48000, 44100, 32000,
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24000, 22050, 16000, 12000, 11025, 8000, 7350
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};
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AVCodec ff_aac_encoder = {
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.name = "aac",
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.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
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108
libavcodec/aacenctab.c
Normal file
108
libavcodec/aacenctab.c
Normal file
@ -0,0 +1,108 @@
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/*
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* AAC encoder data
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* Copyright (c) 2015 Rostislav Pehlivanov ( atomnuker gmail com )
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "aacenctab.h"
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static const uint8_t swb_size_128_96[] = {
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4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
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};
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static const uint8_t swb_size_128_64[] = {
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4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
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};
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static const uint8_t swb_size_128_48[] = {
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4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
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};
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static const uint8_t swb_size_128_24[] = {
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4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
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};
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static const uint8_t swb_size_128_16[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
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};
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static const uint8_t swb_size_128_8[] = {
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4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
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};
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static const uint8_t swb_size_1024_96[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
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64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
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};
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static const uint8_t swb_size_1024_64[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
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12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
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40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
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};
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static const uint8_t swb_size_1024_48[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
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32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
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96
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};
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static const uint8_t swb_size_1024_32[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
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32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
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};
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static const uint8_t swb_size_1024_24[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
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32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
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};
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static const uint8_t swb_size_1024_16[] = {
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8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
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32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
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};
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static const uint8_t swb_size_1024_8[] = {
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12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
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16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
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32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
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};
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const uint8_t *swb_size_128[] = {
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swb_size_128_96, swb_size_128_96, swb_size_128_64,
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swb_size_128_48, swb_size_128_48, swb_size_128_48,
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swb_size_128_24, swb_size_128_24, swb_size_128_16,
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swb_size_128_16, swb_size_128_16, swb_size_128_8,
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swb_size_128_8
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};
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const uint8_t *swb_size_1024[] = {
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swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
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swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
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swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
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swb_size_1024_16, swb_size_1024_16, swb_size_1024_8,
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swb_size_1024_8
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};
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const int swb_size_128_len = sizeof(swb_size_128)/sizeof(*swb_size_128);
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const int swb_size_1024_len = sizeof(swb_size_1024)/sizeof(*swb_size_1024);
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100
libavcodec/aacenctab.h
Normal file
100
libavcodec/aacenctab.h
Normal file
@ -0,0 +1,100 @@
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/*
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* AAC encoder data
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* Copyright (c) 2015 Rostislav Pehlivanov ( atomnuker gmail com )
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
|
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* License as published by the Free Software Foundation; either
|
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* version 2.1 of the License, or (at your option) any later version.
|
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
* Lesser General Public License for more details.
|
||||
*
|
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
|
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* AAC encoder data
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* @author Rostislav Pehlivanov ( atomnuker gmail com )
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*/
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|
||||
#ifndef AVCODEC_AACENCTAB_H
|
||||
#define AVCODEC_AACENCTAB_H
|
||||
|
||||
#include "aac.h"
|
||||
|
||||
/** Total number of usable codebooks **/
|
||||
#define CB_TOT 12
|
||||
|
||||
/** Total number of codebooks, including special ones **/
|
||||
#define CB_TOT_ALL 15
|
||||
|
||||
#define AAC_MAX_CHANNELS 6
|
||||
|
||||
extern const uint8_t *swb_size_1024[];
|
||||
extern const int swb_size_1024_len;
|
||||
extern const uint8_t *swb_size_128[];
|
||||
extern const int swb_size_128_len;
|
||||
|
||||
/** default channel configurations */
|
||||
static const uint8_t aac_chan_configs[6][5] = {
|
||||
{1, TYPE_SCE}, // 1 channel - single channel element
|
||||
{1, TYPE_CPE}, // 2 channels - channel pair
|
||||
{2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
|
||||
{3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
|
||||
{3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
|
||||
{4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
|
||||
};
|
||||
|
||||
/**
|
||||
* Table to remap channels from libavcodec's default order to AAC order.
|
||||
*/
|
||||
static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
|
||||
{ 0 },
|
||||
{ 0, 1 },
|
||||
{ 2, 0, 1 },
|
||||
{ 2, 0, 1, 3 },
|
||||
{ 2, 0, 1, 3, 4 },
|
||||
{ 2, 0, 1, 4, 5, 3 },
|
||||
};
|
||||
|
||||
/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
|
||||
* failures */
|
||||
static const int mpeg4audio_sample_rates[16] = {
|
||||
96000, 88200, 64000, 48000, 44100, 32000,
|
||||
24000, 22050, 16000, 12000, 11025, 8000, 7350
|
||||
};
|
||||
|
||||
/** bits needed to code codebook run value for long windows */
|
||||
static const uint8_t run_value_bits_long[64] = {
|
||||
5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
|
||||
5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 10,
|
||||
10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
|
||||
10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
|
||||
};
|
||||
|
||||
/** bits needed to code codebook run value for short windows */
|
||||
static const uint8_t run_value_bits_short[16] = {
|
||||
3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
|
||||
};
|
||||
|
||||
static const uint8_t * const run_value_bits[2] = {
|
||||
run_value_bits_long, run_value_bits_short
|
||||
};
|
||||
|
||||
/** Map to convert values from BandCodingPath index to a codebook index **/
|
||||
static const uint8_t aac_cb_out_map[CB_TOT_ALL] = {0,1,2,3,4,5,6,7,8,9,10,11,13,14,15};
|
||||
/** Inverse map to convert from codebooks to BandCodingPath indices **/
|
||||
static const uint8_t aac_cb_in_map[CB_TOT_ALL+1] = {0,1,2,3,4,5,6,7,8,9,10,11,0,12,13,14};
|
||||
|
||||
static const uint8_t aac_cb_range [12] = {0, 3, 3, 3, 3, 9, 9, 8, 8, 13, 13, 17};
|
||||
static const uint8_t aac_cb_maxval[12] = {0, 1, 1, 2, 2, 4, 4, 7, 7, 12, 12, 16};
|
||||
|
||||
#endif /* AVCODEC_AACENCTAB_H */
|
||||
Loading…
Reference in New Issue
Block a user