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Update avcodec to 20080825

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git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@27541 a95241bf-73f2-0310-859d-f6bbb57e9c96
This commit is contained in:
David McPaul
2008-09-15 14:00:39 +00:00
parent 70a0cbbb07
commit a191927369
6 changed files with 3395 additions and 504 deletions
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src/add-ons/media/plugins/avcodec/libavcodec/ac3.c
+241
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@@ -0,0 +1,241 @@
/*
* Common code between the AC-3 encoder and decoder
* Copyright (c) 2000 Fabrice Bellard.
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* 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.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file ac3.c
* Common code between the AC-3 encoder and decoder.
*/
#include "avcodec.h"
#include "ac3.h"
#include "bitstream.h"
static uint8_t band_start_tab[51];
static uint8_t bin_to_band_tab[253];
static inline int calc_lowcomp1(int a, int b0, int b1, int c)
{
if ((b0 + 256) == b1) {
a = c;
} else if (b0 > b1) {
a = FFMAX(a - 64, 0);
}
return a;
}
static inline int calc_lowcomp(int a, int b0, int b1, int bin)
{
if (bin < 7) {
return calc_lowcomp1(a, b0, b1, 384);
} else if (bin < 20) {
return calc_lowcomp1(a, b0, b1, 320);
} else {
return FFMAX(a - 128, 0);
}
}
void ff_ac3_bit_alloc_calc_psd(int8_t *exp, int start, int end, int16_t *psd,
int16_t *band_psd)
{
int bin, i, j, k, end1, v;
/* exponent mapping to PSD */
for(bin=start;bin<end;bin++) {
psd[bin]=(3072 - (exp[bin] << 7));
}
/* PSD integration */
j=start;
k=bin_to_band_tab[start];
do {
v=psd[j];
j++;
end1 = FFMIN(band_start_tab[k+1], end);
for(i=j;i<end1;i++) {
/* logadd */
int adr = FFMIN(FFABS(v - psd[j]) >> 1, 255);
v = FFMAX(v, psd[j]) + ff_ac3_log_add_tab[adr];
j++;
}
band_psd[k]=v;
k++;
} while (end > band_start_tab[k]);
}
void ff_ac3_bit_alloc_calc_mask(AC3BitAllocParameters *s, int16_t *band_psd,
int start, int end, int fast_gain, int is_lfe,
int dba_mode, int dba_nsegs, uint8_t *dba_offsets,
uint8_t *dba_lengths, uint8_t *dba_values,
int16_t *mask)
{
int16_t excite[50]; /* excitation */
int bin, k;
int bndstrt, bndend, begin, end1, tmp;
int lowcomp, fastleak, slowleak;
/* excitation function */
bndstrt = bin_to_band_tab[start];
bndend = bin_to_band_tab[end-1] + 1;
if (bndstrt == 0) {
lowcomp = 0;
lowcomp = calc_lowcomp1(lowcomp, band_psd[0], band_psd[1], 384);
excite[0] = band_psd[0] - fast_gain - lowcomp;
lowcomp = calc_lowcomp1(lowcomp, band_psd[1], band_psd[2], 384);
excite[1] = band_psd[1] - fast_gain - lowcomp;
begin = 7;
for (bin = 2; bin < 7; bin++) {
if (!(is_lfe && bin == 6))
lowcomp = calc_lowcomp1(lowcomp, band_psd[bin], band_psd[bin+1], 384);
fastleak = band_psd[bin] - fast_gain;
slowleak = band_psd[bin] - s->slow_gain;
excite[bin] = fastleak - lowcomp;
if (!(is_lfe && bin == 6)) {
if (band_psd[bin] <= band_psd[bin+1]) {
begin = bin + 1;
break;
}
}
}
end1=bndend;
if (end1 > 22) end1=22;
for (bin = begin; bin < end1; bin++) {
if (!(is_lfe && bin == 6))
lowcomp = calc_lowcomp(lowcomp, band_psd[bin], band_psd[bin+1], bin);
fastleak = FFMAX(fastleak - s->fast_decay, band_psd[bin] - fast_gain);
slowleak = FFMAX(slowleak - s->slow_decay, band_psd[bin] - s->slow_gain);
excite[bin] = FFMAX(fastleak - lowcomp, slowleak);
}
begin = 22;
} else {
/* coupling channel */
begin = bndstrt;
fastleak = (s->cpl_fast_leak << 8) + 768;
slowleak = (s->cpl_slow_leak << 8) + 768;
}
for (bin = begin; bin < bndend; bin++) {
fastleak = FFMAX(fastleak - s->fast_decay, band_psd[bin] - fast_gain);
slowleak = FFMAX(slowleak - s->slow_decay, band_psd[bin] - s->slow_gain);
excite[bin] = FFMAX(fastleak, slowleak);
}
/* compute masking curve */
for (bin = bndstrt; bin < bndend; bin++) {
tmp = s->db_per_bit - band_psd[bin];
if (tmp > 0) {
excite[bin] += tmp >> 2;
}
mask[bin] = FFMAX(ff_ac3_hearing_threshold_tab[bin >> s->sr_shift][s->sr_code], excite[bin]);
}
/* delta bit allocation */
if (dba_mode == DBA_REUSE || dba_mode == DBA_NEW) {
int band, seg, delta;
band = 0;
for (seg = 0; seg < FFMIN(8, dba_nsegs); seg++) {
band = FFMIN(49, band + dba_offsets[seg]);
if (dba_values[seg] >= 4) {
delta = (dba_values[seg] - 3) << 7;
} else {
delta = (dba_values[seg] - 4) << 7;
}
for (k = 0; k < dba_lengths[seg]; k++) {
mask[band] += delta;
band++;
}
}
}
}
void ff_ac3_bit_alloc_calc_bap(int16_t *mask, int16_t *psd, int start, int end,
int snr_offset, int floor,
const uint8_t *bap_tab, uint8_t *bap)
{
int i, j, k, end1, v, address;
/* special case, if snr offset is -960, set all bap's to zero */
if(snr_offset == -960) {
memset(bap, 0, 256);
return;
}
i = start;
j = bin_to_band_tab[start];
do {
v = (FFMAX(mask[j] - snr_offset - floor, 0) & 0x1FE0) + floor;
end1 = FFMIN(band_start_tab[j] + ff_ac3_critical_band_size_tab[j], end);
for (k = i; k < end1; k++) {
address = av_clip((psd[i] - v) >> 5, 0, 63);
bap[i] = bap_tab[address];
i++;
}
} while (end > band_start_tab[j++]);
}
/* AC-3 bit allocation. The algorithm is the one described in the AC-3
spec. */
void ac3_parametric_bit_allocation(AC3BitAllocParameters *s, uint8_t *bap,
int8_t *exp, int start, int end,
int snr_offset, int fast_gain, int is_lfe,
int dba_mode, int dba_nsegs,
uint8_t *dba_offsets, uint8_t *dba_lengths,
uint8_t *dba_values)
{
int16_t psd[256]; /* scaled exponents */
int16_t band_psd[50]; /* interpolated exponents */
int16_t mask[50]; /* masking value */
ff_ac3_bit_alloc_calc_psd(exp, start, end, psd, band_psd);
ff_ac3_bit_alloc_calc_mask(s, band_psd, start, end, fast_gain, is_lfe,
dba_mode, dba_nsegs, dba_offsets, dba_lengths, dba_values,
mask);
ff_ac3_bit_alloc_calc_bap(mask, psd, start, end, snr_offset, s->floor,
ff_ac3_bap_tab, bap);
}
/**
* Initializes some tables.
* note: This function must remain thread safe because it is called by the
* AVParser init code.
*/
av_cold void ac3_common_init(void)
{
int i, j, k, l, v;
/* compute bndtab and masktab from bandsz */
k = 0;
l = 0;
for(i=0;i<50;i++) {
band_start_tab[i] = l;
v = ff_ac3_critical_band_size_tab[i];
for(j=0;j<v;j++) bin_to_band_tab[k++]=i;
l += v;
}
band_start_tab[50] = l;
}
src/add-ons/media/plugins/avcodec/libavcodec/ac3_parser.c
+196
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@@ -0,0 +1,196 @@
/*
* AC-3 parser
* Copyright (c) 2003 Fabrice Bellard.
* Copyright (c) 2003 Michael Niedermayer.
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* 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.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "parser.h"
#include "ac3_parser.h"
#include "aac_ac3_parser.h"
#include "bitstream.h"
#define AC3_HEADER_SIZE 7
static const uint8_t eac3_blocks[4] = {
1, 2, 3, 6
};
int ff_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr)
{
int frame_size_code;
memset(hdr, 0, sizeof(*hdr));
hdr->sync_word = get_bits(gbc, 16);
if(hdr->sync_word != 0x0B77)
return AC3_PARSE_ERROR_SYNC;
/* read ahead to bsid to distinguish between AC-3 and E-AC-3 */
hdr->bitstream_id = show_bits_long(gbc, 29) & 0x1F;
if(hdr->bitstream_id > 16)
return AC3_PARSE_ERROR_BSID;
hdr->num_blocks = 6;
/* set default mix levels */
hdr->center_mix_level = 1; // -4.5dB
hdr->surround_mix_level = 1; // -6.0dB
if(hdr->bitstream_id <= 10) {
/* Normal AC-3 */
hdr->crc1 = get_bits(gbc, 16);
hdr->sr_code = get_bits(gbc, 2);
if(hdr->sr_code == 3)
return AC3_PARSE_ERROR_SAMPLE_RATE;
frame_size_code = get_bits(gbc, 6);
if(frame_size_code > 37)
return AC3_PARSE_ERROR_FRAME_SIZE;
skip_bits(gbc, 5); // skip bsid, already got it
skip_bits(gbc, 3); // skip bitstream mode
hdr->channel_mode = get_bits(gbc, 3);
if(hdr->channel_mode == AC3_CHMODE_STEREO) {
skip_bits(gbc, 2); // skip dsurmod
} else {
if((hdr->channel_mode & 1) && hdr->channel_mode != AC3_CHMODE_MONO)
hdr->center_mix_level = get_bits(gbc, 2);
if(hdr->channel_mode & 4)
hdr->surround_mix_level = get_bits(gbc, 2);
}
hdr->lfe_on = get_bits1(gbc);
hdr->sr_shift = FFMAX(hdr->bitstream_id, 8) - 8;
hdr->sample_rate = ff_ac3_sample_rate_tab[hdr->sr_code] >> hdr->sr_shift;
hdr->bit_rate = (ff_ac3_bitrate_tab[frame_size_code>>1] * 1000) >> hdr->sr_shift;
hdr->channels = ff_ac3_channels_tab[hdr->channel_mode] + hdr->lfe_on;
hdr->frame_size = ff_ac3_frame_size_tab[frame_size_code][hdr->sr_code] * 2;
hdr->frame_type = EAC3_FRAME_TYPE_AC3_CONVERT; //EAC3_FRAME_TYPE_INDEPENDENT;
hdr->substreamid = 0;
} else {
/* Enhanced AC-3 */
hdr->crc1 = 0;
hdr->frame_type = get_bits(gbc, 2);
if(hdr->frame_type == EAC3_FRAME_TYPE_RESERVED)
return AC3_PARSE_ERROR_FRAME_TYPE;
hdr->substreamid = get_bits(gbc, 3);
hdr->frame_size = (get_bits(gbc, 11) + 1) << 1;
if(hdr->frame_size < AC3_HEADER_SIZE)
return AC3_PARSE_ERROR_FRAME_SIZE;
hdr->sr_code = get_bits(gbc, 2);
if (hdr->sr_code == 3) {
int sr_code2 = get_bits(gbc, 2);
if(sr_code2 == 3)
return AC3_PARSE_ERROR_SAMPLE_RATE;
hdr->sample_rate = ff_ac3_sample_rate_tab[sr_code2] / 2;
hdr->sr_shift = 1;
} else {
hdr->num_blocks = eac3_blocks[get_bits(gbc, 2)];
hdr->sample_rate = ff_ac3_sample_rate_tab[hdr->sr_code];
hdr->sr_shift = 0;
}
hdr->channel_mode = get_bits(gbc, 3);
hdr->lfe_on = get_bits1(gbc);
hdr->bit_rate = (uint32_t)(8.0 * hdr->frame_size * hdr->sample_rate /
(hdr->num_blocks * 256.0));
hdr->channels = ff_ac3_channels_tab[hdr->channel_mode] + hdr->lfe_on;
}
return 0;
}
int ff_ac3_parse_header_full(GetBitContext *gbc, AC3HeaderInfo *hdr){
int ret, i;
ret = ff_ac3_parse_header(gbc, hdr);
if(!ret){
if(hdr->bitstream_id>10){
/* Enhanced AC-3 */
skip_bits(gbc, 5); // skip bitstream id
/* skip dialog normalization and compression gain */
for (i = 0; i < (hdr->channel_mode ? 1 : 2); i++) {
skip_bits(gbc, 5); // skip dialog normalization
if (get_bits1(gbc)) {
skip_bits(gbc, 8); //skip Compression gain word
}
}
/* dependent stream channel map */
if (hdr->frame_type == EAC3_FRAME_TYPE_DEPENDENT && get_bits1(gbc)) {
hdr->channel_map = get_bits(gbc, 16); //custom channel map
return 0;
}
}
//default channel map based on acmod and lfeon
hdr->channel_map = ff_eac3_default_chmap[hdr->channel_mode];
if(hdr->lfe_on)
hdr->channel_map |= AC3_CHMAP_LFE;
}
return ret;
}
static int ac3_sync(uint64_t state, AACAC3ParseContext *hdr_info,
int *need_next_header, int *new_frame_start)
{
int err;
uint64_t tmp = be2me_64(state);
AC3HeaderInfo hdr;
GetBitContext gbc;
init_get_bits(&gbc, ((uint8_t *)&tmp)+8-AC3_HEADER_SIZE, 54);
err = ff_ac3_parse_header(&gbc, &hdr);
if(err < 0)
return 0;
hdr_info->sample_rate = hdr.sample_rate;
hdr_info->bit_rate = hdr.bit_rate;
hdr_info->channels = hdr.channels;
hdr_info->samples = AC3_FRAME_SIZE;
*need_next_header = (hdr.frame_type != EAC3_FRAME_TYPE_AC3_CONVERT);
*new_frame_start = (hdr.frame_type != EAC3_FRAME_TYPE_DEPENDENT);
return hdr.frame_size;
}
static av_cold int ac3_parse_init(AVCodecParserContext *s1)
{
AACAC3ParseContext *s = s1->priv_data;
s->header_size = AC3_HEADER_SIZE;
s->sync = ac3_sync;
return 0;
}
AVCodecParser ac3_parser = {
{ CODEC_ID_AC3 },
sizeof(AACAC3ParseContext),
ac3_parse_init,
ff_aac_ac3_parse,
ff_parse_close,
};
src/add-ons/media/plugins/avcodec/libavcodec/ac3dec.c
+1268
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File diff suppressed because it is too large Load Diff
src/add-ons/media/plugins/avcodec/libavcodec/ac3dec_data.c
+1134
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File diff suppressed because it is too large Load Diff
src/add-ons/media/plugins/avcodec/libavcodec/ac3enc.c
+204 -414
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@@ -1,30 +1,33 @@
/*
* The simplest AC3 encoder
* The simplest AC-3 encoder
* Copyright (c) 2000 Fabrice Bellard.
*
* This library is free software; you can redistribute it and/or
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
* version 2.1 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* FFmpeg is distributed in the hope that it will be useful,
* 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.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file ac3enc.c
* The simplest AC3 encoder.
* The simplest AC-3 encoder.
*/
//#define DEBUG
//#define DEBUG_BITALLOC
#include "libavutil/crc.h"
#include "avcodec.h"
#include "bitstream.h"
#include "ac3.h"
typedef struct AC3EncodeContext {
@@ -34,30 +37,35 @@ typedef struct AC3EncodeContext {
int lfe_channel;
int bit_rate;
unsigned int sample_rate;
unsigned int bsid;
unsigned int bitstream_id;
unsigned int frame_size_min; /* minimum frame size in case rounding is necessary */
unsigned int frame_size; /* current frame size in words */
int halfratecod;
unsigned int frmsizecod;
unsigned int fscod; /* frequency */
unsigned int acmod;
unsigned int bits_written;
unsigned int samples_written;
int sr_shift;
unsigned int frame_size_code;
unsigned int sr_code; /* frequency */
unsigned int channel_mode;
int lfe;
unsigned int bsmod;
unsigned int bitstream_mode;
short last_samples[AC3_MAX_CHANNELS][256];
unsigned int chbwcod[AC3_MAX_CHANNELS];
int nb_coefs[AC3_MAX_CHANNELS];
/* bitrate allocation control */
int sgaincod, sdecaycod, fdecaycod, dbkneecod, floorcod;
int slow_gain_code, slow_decay_code, fast_decay_code, db_per_bit_code, floor_code;
AC3BitAllocParameters bit_alloc;
int csnroffst;
int fgaincod[AC3_MAX_CHANNELS];
int fsnroffst[AC3_MAX_CHANNELS];
int coarse_snr_offset;
int fast_gain_code[AC3_MAX_CHANNELS];
int fine_snr_offset[AC3_MAX_CHANNELS];
/* mantissa encoding */
int mant1_cnt, mant2_cnt, mant4_cnt;
} AC3EncodeContext;
#include "ac3tab.h"
static int16_t costab[64];
static int16_t sintab[64];
static int16_t xcos1[128];
static int16_t xsin1[128];
#define MDCT_NBITS 9
#define N (1 << MDCT_NBITS)
@@ -65,9 +73,6 @@ typedef struct AC3EncodeContext {
/* new exponents are sent if their Norm 1 exceed this number */
#define EXP_DIFF_THRESHOLD 1000
static void fft_init(int ln);
static void ac3_crc_init(void);
static inline int16_t fix15(float a)
{
int v;
@@ -79,219 +84,13 @@ static inline int16_t fix15(float a)
return v;
}
static inline int calc_lowcomp1(int a, int b0, int b1)
{
if ((b0 + 256) == b1) {
a = 384 ;
} else if (b0 > b1) {
a = a - 64;
if (a < 0) a=0;
}
return a;
}
static inline int calc_lowcomp(int a, int b0, int b1, int bin)
{
if (bin < 7) {
if ((b0 + 256) == b1) {
a = 384 ;
} else if (b0 > b1) {
a = a - 64;
if (a < 0) a=0;
}
} else if (bin < 20) {
if ((b0 + 256) == b1) {
a = 320 ;
} else if (b0 > b1) {
a= a - 64;
if (a < 0) a=0;
}
} else {
a = a - 128;
if (a < 0) a=0;
}
return a;
}
/* AC3 bit allocation. The algorithm is the one described in the AC3
spec. */
void ac3_parametric_bit_allocation(AC3BitAllocParameters *s, uint8_t *bap,
int8_t *exp, int start, int end,
int snroffset, int fgain, int is_lfe,
int deltbae,int deltnseg,
uint8_t *deltoffst, uint8_t *deltlen, uint8_t *deltba)
{
int bin,i,j,k,end1,v,v1,bndstrt,bndend,lowcomp,begin;
int fastleak,slowleak,address,tmp;
int16_t psd[256]; /* scaled exponents */
int16_t bndpsd[50]; /* interpolated exponents */
int16_t excite[50]; /* excitation */
int16_t mask[50]; /* masking value */
/* exponent mapping to PSD */
for(bin=start;bin<end;bin++) {
psd[bin]=(3072 - (exp[bin] << 7));
}
/* PSD integration */
j=start;
k=masktab[start];
do {
v=psd[j];
j++;
end1=bndtab[k+1];
if (end1 > end) end1=end;
for(i=j;i<end1;i++) {
int c,adr;
/* logadd */
v1=psd[j];
c=v-v1;
if (c >= 0) {
adr=c >> 1;
if (adr > 255) adr=255;
v=v + latab[adr];
} else {
adr=(-c) >> 1;
if (adr > 255) adr=255;
v=v1 + latab[adr];
}
j++;
}
bndpsd[k]=v;
k++;
} while (end > bndtab[k]);
/* excitation function */
bndstrt = masktab[start];
bndend = masktab[end-1] + 1;
if (bndstrt == 0) {
lowcomp = 0;
lowcomp = calc_lowcomp1(lowcomp, bndpsd[0], bndpsd[1]) ;
excite[0] = bndpsd[0] - fgain - lowcomp ;
lowcomp = calc_lowcomp1(lowcomp, bndpsd[1], bndpsd[2]) ;
excite[1] = bndpsd[1] - fgain - lowcomp ;
begin = 7 ;
for (bin = 2; bin < 7; bin++) {
if (!(is_lfe && bin == 6))
lowcomp = calc_lowcomp1(lowcomp, bndpsd[bin], bndpsd[bin+1]) ;
fastleak = bndpsd[bin] - fgain ;
slowleak = bndpsd[bin] - s->sgain ;
excite[bin] = fastleak - lowcomp ;
if (!(is_lfe && bin == 6)) {
if (bndpsd[bin] <= bndpsd[bin+1]) {
begin = bin + 1 ;
break ;
}
}
}
end1=bndend;
if (end1 > 22) end1=22;
for (bin = begin; bin < end1; bin++) {
if (!(is_lfe && bin == 6))
lowcomp = calc_lowcomp(lowcomp, bndpsd[bin], bndpsd[bin+1], bin) ;
fastleak -= s->fdecay ;
v = bndpsd[bin] - fgain;
if (fastleak < v) fastleak = v;
slowleak -= s->sdecay ;
v = bndpsd[bin] - s->sgain;
if (slowleak < v) slowleak = v;
v=fastleak - lowcomp;
if (slowleak > v) v=slowleak;
excite[bin] = v;
}
begin = 22;
} else {
/* coupling channel */
begin = bndstrt;
fastleak = (s->cplfleak << 8) + 768;
slowleak = (s->cplsleak << 8) + 768;
}
for (bin = begin; bin < bndend; bin++) {
fastleak -= s->fdecay ;
v = bndpsd[bin] - fgain;
if (fastleak < v) fastleak = v;
slowleak -= s->sdecay ;
v = bndpsd[bin] - s->sgain;
if (slowleak < v) slowleak = v;
v=fastleak;
if (slowleak > v) v = slowleak;
excite[bin] = v;
}
/* compute masking curve */
for (bin = bndstrt; bin < bndend; bin++) {
v1 = excite[bin];
tmp = s->dbknee - bndpsd[bin];
if (tmp > 0) {
v1 += tmp >> 2;
}
v=hth[bin >> s->halfratecod][s->fscod];
if (v1 > v) v=v1;
mask[bin] = v;
}
/* delta bit allocation */
if (deltbae == 0 || deltbae == 1) {
int band, seg, delta;
band = 0 ;
for (seg = 0; seg < deltnseg; seg++) {
band += deltoffst[seg] ;
if (deltba[seg] >= 4) {
delta = (deltba[seg] - 3) << 7;
} else {
delta = (deltba[seg] - 4) << 7;
}
for (k = 0; k < deltlen[seg]; k++) {
mask[band] += delta ;
band++ ;
}
}
}
/* compute bit allocation */
i = start ;
j = masktab[start] ;
do {
v=mask[j];
v -= snroffset ;
v -= s->floor ;
if (v < 0) v = 0;
v &= 0x1fe0 ;
v += s->floor ;
end1=bndtab[j] + bndsz[j];
if (end1 > end) end1=end;
for (k = i; k < end1; k++) {
address = (psd[i] - v) >> 5 ;
if (address < 0) address=0;
else if (address > 63) address=63;
bap[i] = baptab[address];
i++;
}
} while (end > bndtab[j++]) ;
}
typedef struct IComplex {
short re,im;
} IComplex;
static void fft_init(int ln)
{
int i, j, m, n;
int i, n;
float alpha;
n = 1 << ln;
@@ -301,14 +100,6 @@ static void fft_init(int ln)
costab[i] = fix15(cos(alpha));
sintab[i] = fix15(sin(alpha));
}
for(i=0;i<n;i++) {
m=0;
for(j=0;j<ln;j++) {
m |= ((i >> j) & 1) << (ln-j-1);
}
fft_rev[i]=m;
}
}
/* butter fly op */
@@ -346,14 +137,9 @@ static void fft(IComplex *z, int ln)
/* reverse */
for(j=0;j<np;j++) {
int k;
IComplex tmp;
k = fft_rev[j];
if (k < j) {
tmp = z[k];
z[k] = z[j];
z[j] = tmp;
}
int k = ff_reverse[j] >> (8 - ln);
if (k < j)
FFSWAP(IComplex, z[k], z[j]);
}
/* pass 0 */
@@ -515,7 +301,7 @@ static int encode_exp(uint8_t encoded_exp[N/2],
int nb_exps,
int exp_strategy)
{
int group_size, nb_groups, i, j, k, recurse, exp_min, delta;
int group_size, nb_groups, i, j, k, exp_min;
uint8_t exp1[N/2];
switch(exp_strategy) {
@@ -550,24 +336,12 @@ static int encode_exp(uint8_t encoded_exp[N/2],
if (exp1[0] > 15)
exp1[0] = 15;
/* Iterate until the delta constraints between each groups are
satisfyed. I'm sure it is possible to find a better algorithm,
but I am lazy */
do {
recurse = 0;
for(i=1;i<=nb_groups;i++) {
delta = exp1[i] - exp1[i-1];
if (delta > 2) {
/* if delta too big, we encode a smaller exponent */
exp1[i] = exp1[i-1] + 2;
} else if (delta < -2) {
/* if delta is too small, we must decrease the previous
exponent, which means we must recurse */
recurse = 1;
exp1[i-1] = exp1[i] + 2;
}
}
} while (recurse);
/* Decrease the delta between each groups to within 2
* so that they can be differentially encoded */
for (i=1;i<=nb_groups;i++)
exp1[i] = FFMIN(exp1[i], exp1[i-1] + 2);
for (i=nb_groups-1;i>=0;i--)
exp1[i] = FFMIN(exp1[i], exp1[i+1] + 2);
/* now we have the exponent values the decoder will see */
encoded_exp[0] = exp1[0];
@@ -641,13 +415,45 @@ static int compute_mantissa_size(AC3EncodeContext *s, uint8_t *m, int nb_coefs)
}
static int bit_alloc(AC3EncodeContext *s,
uint8_t bap[NB_BLOCKS][AC3_MAX_CHANNELS][N/2],
static void bit_alloc_masking(AC3EncodeContext *s,
uint8_t encoded_exp[NB_BLOCKS][AC3_MAX_CHANNELS][N/2],
uint8_t exp_strategy[NB_BLOCKS][AC3_MAX_CHANNELS],
int frame_bits, int csnroffst, int fsnroffst)
int16_t psd[NB_BLOCKS][AC3_MAX_CHANNELS][N/2],
int16_t mask[NB_BLOCKS][AC3_MAX_CHANNELS][50])
{
int blk, ch;
int16_t band_psd[NB_BLOCKS][AC3_MAX_CHANNELS][50];
for(blk=0; blk<NB_BLOCKS; blk++) {
for(ch=0;ch<s->nb_all_channels;ch++) {
if(exp_strategy[blk][ch] == EXP_REUSE) {
memcpy(psd[blk][ch], psd[blk-1][ch], (N/2)*sizeof(int16_t));
memcpy(mask[blk][ch], mask[blk-1][ch], 50*sizeof(int16_t));
} else {
ff_ac3_bit_alloc_calc_psd(encoded_exp[blk][ch], 0,
s->nb_coefs[ch],
psd[blk][ch], band_psd[blk][ch]);
ff_ac3_bit_alloc_calc_mask(&s->bit_alloc, band_psd[blk][ch],
0, s->nb_coefs[ch],
ff_ac3_fast_gain_tab[s->fast_gain_code[ch]],
ch == s->lfe_channel,
DBA_NONE, 0, NULL, NULL, NULL,
mask[blk][ch]);
}
}
}
}
static int bit_alloc(AC3EncodeContext *s,
int16_t mask[NB_BLOCKS][AC3_MAX_CHANNELS][50],
int16_t psd[NB_BLOCKS][AC3_MAX_CHANNELS][N/2],
uint8_t bap[NB_BLOCKS][AC3_MAX_CHANNELS][N/2],
int frame_bits, int coarse_snr_offset, int fine_snr_offset)
{
int i, ch;
int snr_offset;
snr_offset = (((coarse_snr_offset - 15) << 4) + fine_snr_offset) << 2;
/* compute size */
for(i=0;i<NB_BLOCKS;i++) {
@@ -655,21 +461,17 @@ static int bit_alloc(AC3EncodeContext *s,
s->mant2_cnt = 0;
s->mant4_cnt = 0;
for(ch=0;ch<s->nb_all_channels;ch++) {
ac3_parametric_bit_allocation(&s->bit_alloc,
bap[i][ch], (int8_t *)encoded_exp[i][ch],
0, s->nb_coefs[ch],
(((csnroffst-15) << 4) +
fsnroffst) << 2,
fgaintab[s->fgaincod[ch]],
ch == s->lfe_channel,
2, 0, NULL, NULL, NULL);
ff_ac3_bit_alloc_calc_bap(mask[i][ch], psd[i][ch], 0,
s->nb_coefs[ch], snr_offset,
s->bit_alloc.floor, ff_ac3_bap_tab,
bap[i][ch]);
frame_bits += compute_mantissa_size(s, bap[i][ch],
s->nb_coefs[ch]);
}
}
#if 0
printf("csnr=%d fsnr=%d frame_bits=%d diff=%d\n",
csnroffst, fsnroffst, frame_bits,
coarse_snr_offset, fine_snr_offset, frame_bits,
16 * s->frame_size - ((frame_bits + 7) & ~7));
#endif
return 16 * s->frame_size - frame_bits;
@@ -684,39 +486,43 @@ static int compute_bit_allocation(AC3EncodeContext *s,
int frame_bits)
{
int i, ch;
int csnroffst, fsnroffst;
int coarse_snr_offset, fine_snr_offset;
uint8_t bap1[NB_BLOCKS][AC3_MAX_CHANNELS][N/2];
static int frame_bits_inc[8] = { 0, 0, 2, 2, 2, 4, 2, 4 };
int16_t psd[NB_BLOCKS][AC3_MAX_CHANNELS][N/2];
int16_t mask[NB_BLOCKS][AC3_MAX_CHANNELS][50];
static const int frame_bits_inc[8] = { 0, 0, 2, 2, 2, 4, 2, 4 };
/* init default parameters */
s->sdecaycod = 2;
s->fdecaycod = 1;
s->sgaincod = 1;
s->dbkneecod = 2;
s->floorcod = 4;
s->slow_decay_code = 2;
s->fast_decay_code = 1;
s->slow_gain_code = 1;
s->db_per_bit_code = 2;
s->floor_code = 4;
for(ch=0;ch<s->nb_all_channels;ch++)
s->fgaincod[ch] = 4;
s->fast_gain_code[ch] = 4;
/* compute real values */
s->bit_alloc.fscod = s->fscod;
s->bit_alloc.halfratecod = s->halfratecod;
s->bit_alloc.sdecay = sdecaytab[s->sdecaycod] >> s->halfratecod;
s->bit_alloc.fdecay = fdecaytab[s->fdecaycod] >> s->halfratecod;
s->bit_alloc.sgain = sgaintab[s->sgaincod];
s->bit_alloc.dbknee = dbkneetab[s->dbkneecod];
s->bit_alloc.floor = floortab[s->floorcod];
s->bit_alloc.sr_code = s->sr_code;
s->bit_alloc.sr_shift = s->sr_shift;
s->bit_alloc.slow_decay = ff_ac3_slow_decay_tab[s->slow_decay_code] >> s->sr_shift;
s->bit_alloc.fast_decay = ff_ac3_fast_decay_tab[s->fast_decay_code] >> s->sr_shift;
s->bit_alloc.slow_gain = ff_ac3_slow_gain_tab[s->slow_gain_code];
s->bit_alloc.db_per_bit = ff_ac3_db_per_bit_tab[s->db_per_bit_code];
s->bit_alloc.floor = ff_ac3_floor_tab[s->floor_code];
/* header size */
frame_bits += 65;
// if (s->acmod == 2)
// if (s->channel_mode == 2)
// frame_bits += 2;
frame_bits += frame_bits_inc[s->acmod];
frame_bits += frame_bits_inc[s->channel_mode];
/* audio blocks */
for(i=0;i<NB_BLOCKS;i++) {
frame_bits += s->nb_channels * 2 + 2; /* blksw * c, dithflag * c, dynrnge, cplstre */
if (s->acmod == 2)
if (s->channel_mode == AC3_CHMODE_STEREO) {
frame_bits++; /* rematstr */
if(i==0) frame_bits += 4;
}
frame_bits += 2 * s->nb_channels; /* chexpstr[2] * c */
if (s->lfe)
frame_bits++; /* lfeexpstr */
@@ -735,49 +541,55 @@ static int compute_bit_allocation(AC3EncodeContext *s,
/* (fsnoffset[4] + fgaincod[4]) * c */
frame_bits += 2*4 + 3 + 6 + s->nb_all_channels * (4 + 3);
/* auxdatae, crcrsv */
frame_bits += 2;
/* CRC */
frame_bits += 16;
/* calculate psd and masking curve before doing bit allocation */
bit_alloc_masking(s, encoded_exp, exp_strategy, psd, mask);
/* now the big work begins : do the bit allocation. Modify the snr
offset until we can pack everything in the requested frame size */
csnroffst = s->csnroffst;
while (csnroffst >= 0 &&
bit_alloc(s, bap, encoded_exp, exp_strategy, frame_bits, csnroffst, 0) < 0)
csnroffst -= SNR_INC1;
if (csnroffst < 0) {
av_log(NULL, AV_LOG_ERROR, "Yack, Error !!!\n");
coarse_snr_offset = s->coarse_snr_offset;
while (coarse_snr_offset >= 0 &&
bit_alloc(s, mask, psd, bap, frame_bits, coarse_snr_offset, 0) < 0)
coarse_snr_offset -= SNR_INC1;
if (coarse_snr_offset < 0) {
av_log(NULL, AV_LOG_ERROR, "Bit allocation failed. Try increasing the bitrate.\n");
return -1;
}
while ((csnroffst + SNR_INC1) <= 63 &&
bit_alloc(s, bap1, encoded_exp, exp_strategy, frame_bits,
csnroffst + SNR_INC1, 0) >= 0) {
csnroffst += SNR_INC1;
while ((coarse_snr_offset + SNR_INC1) <= 63 &&
bit_alloc(s, mask, psd, bap1, frame_bits,
coarse_snr_offset + SNR_INC1, 0) >= 0) {
coarse_snr_offset += SNR_INC1;
memcpy(bap, bap1, sizeof(bap1));
}
while ((csnroffst + 1) <= 63 &&
bit_alloc(s, bap1, encoded_exp, exp_strategy, frame_bits, csnroffst + 1, 0) >= 0) {
csnroffst++;
while ((coarse_snr_offset + 1) <= 63 &&
bit_alloc(s, mask, psd, bap1, frame_bits, coarse_snr_offset + 1, 0) >= 0) {
coarse_snr_offset++;
memcpy(bap, bap1, sizeof(bap1));
}
fsnroffst = 0;
while ((fsnroffst + SNR_INC1) <= 15 &&
bit_alloc(s, bap1, encoded_exp, exp_strategy, frame_bits,
csnroffst, fsnroffst + SNR_INC1) >= 0) {
fsnroffst += SNR_INC1;
fine_snr_offset = 0;
while ((fine_snr_offset + SNR_INC1) <= 15 &&
bit_alloc(s, mask, psd, bap1, frame_bits,
coarse_snr_offset, fine_snr_offset + SNR_INC1) >= 0) {
fine_snr_offset += SNR_INC1;
memcpy(bap, bap1, sizeof(bap1));
}
while ((fsnroffst + 1) <= 15 &&
bit_alloc(s, bap1, encoded_exp, exp_strategy, frame_bits,
csnroffst, fsnroffst + 1) >= 0) {
fsnroffst++;
while ((fine_snr_offset + 1) <= 15 &&
bit_alloc(s, mask, psd, bap1, frame_bits,
coarse_snr_offset, fine_snr_offset + 1) >= 0) {
fine_snr_offset++;
memcpy(bap, bap1, sizeof(bap1));
}
s->csnroffst = csnroffst;
s->coarse_snr_offset = coarse_snr_offset;
for(ch=0;ch<s->nb_all_channels;ch++)
s->fsnroffst[ch] = fsnroffst;
s->fine_snr_offset[ch] = fine_snr_offset;
#if defined(DEBUG_BITALLOC)
{
int j;
@@ -797,23 +609,7 @@ static int compute_bit_allocation(AC3EncodeContext *s,
return 0;
}
void ac3_common_init(void)
{
int i, j, k, l, v;
/* compute bndtab and masktab from bandsz */
k = 0;
l = 0;
for(i=0;i<50;i++) {
bndtab[i] = l;
v = bndsz[i];
for(j=0;j<v;j++) masktab[k++]=i;
l += v;
}
bndtab[50] = 0;
}
static int AC3_encode_init(AVCodecContext *avctx)
static av_cold int AC3_encode_init(AVCodecContext *avctx)
{
int freq = avctx->sample_rate;
int bitrate = avctx->bit_rate;
@@ -821,7 +617,8 @@ static int AC3_encode_init(AVCodecContext *avctx)
AC3EncodeContext *s = avctx->priv_data;
int i, j, ch;
float alpha;
static const uint8_t acmod_defs[6] = {
int bw_code;
static const uint8_t channel_mode_defs[6] = {
0x01, /* C */
0x02, /* L R */
0x03, /* L C R */
@@ -832,10 +629,12 @@ static int AC3_encode_init(AVCodecContext *avctx)
avctx->frame_size = AC3_FRAME_SIZE;
ac3_common_init();
/* number of channels */
if (channels < 1 || channels > 6)
return -1;
s->acmod = acmod_defs[channels - 1];
s->channel_mode = channel_mode_defs[channels - 1];
s->lfe = (channels == 6) ? 1 : 0;
s->nb_all_channels = channels;
s->nb_channels = channels > 5 ? 5 : channels;
@@ -844,46 +643,53 @@ static int AC3_encode_init(AVCodecContext *avctx)
/* frequency */
for(i=0;i<3;i++) {
for(j=0;j<3;j++)
if ((ac3_freqs[j] >> i) == freq)
if ((ff_ac3_sample_rate_tab[j] >> i) == freq)
goto found;
}
return -1;
found:
s->sample_rate = freq;
s->halfratecod = i;
s->fscod = j;
s->bsid = 8 + s->halfratecod;
s->bsmod = 0; /* complete main audio service */
s->sr_shift = i;
s->sr_code = j;
s->bitstream_id = 8 + s->sr_shift;
s->bitstream_mode = 0; /* complete main audio service */
/* bitrate & frame size */
bitrate /= 1000;
for(i=0;i<19;i++) {
if ((ac3_bitratetab[i] >> s->halfratecod) == bitrate)
if ((ff_ac3_bitrate_tab[i] >> s->sr_shift)*1000 == bitrate)
break;
}
if (i == 19)
return -1;
s->bit_rate = bitrate;
s->frmsizecod = i << 1;
s->frame_size_min = (bitrate * 1000 * AC3_FRAME_SIZE) / (freq * 16);
/* for now we do not handle fractional sizes */
s->frame_size_code = i << 1;
s->frame_size_min = ff_ac3_frame_size_tab[s->frame_size_code][s->sr_code];
s->bits_written = 0;
s->samples_written = 0;
s->frame_size = s->frame_size_min;
/* bit allocation init */
if(avctx->cutoff) {
/* calculate bandwidth based on user-specified cutoff frequency */
int cutoff = av_clip(avctx->cutoff, 1, s->sample_rate >> 1);
int fbw_coeffs = cutoff * 512 / s->sample_rate;
bw_code = av_clip((fbw_coeffs - 73) / 3, 0, 60);
} else {
/* use default bandwidth setting */
/* XXX: should compute the bandwidth according to the frame
size, so that we avoid annoying high frequency artifacts */
bw_code = 50;
}
for(ch=0;ch<s->nb_channels;ch++) {
/* bandwidth for each channel */
/* XXX: should compute the bandwidth according to the frame
size, so that we avoid anoying high freq artefacts */
s->chbwcod[ch] = 50; /* sample bandwidth as mpeg audio layer 2 table 0 */
s->nb_coefs[ch] = ((s->chbwcod[ch] + 12) * 3) + 37;
s->chbwcod[ch] = bw_code;
s->nb_coefs[ch] = bw_code * 3 + 73;
}
if (s->lfe) {
s->nb_coefs[s->lfe_channel] = 7; /* fixed */
}
/* initial snr offset */
s->csnroffst = 40;
ac3_common_init();
s->coarse_snr_offset = 40;
/* mdct init */
fft_init(MDCT_NBITS - 2);
@@ -893,31 +699,29 @@ static int AC3_encode_init(AVCodecContext *avctx)
xsin1[i] = fix15(-sin(alpha));
}
ac3_crc_init();
avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1;
return 0;
}
/* output the AC3 frame header */
/* output the AC-3 frame header */
static void output_frame_header(AC3EncodeContext *s, unsigned char *frame)
{
init_put_bits(&s->pb, frame, AC3_MAX_CODED_FRAME_SIZE);
put_bits(&s->pb, 16, 0x0b77); /* frame header */
put_bits(&s->pb, 16, 0); /* crc1: will be filled later */
put_bits(&s->pb, 2, s->fscod);
put_bits(&s->pb, 6, s->frmsizecod + (s->frame_size - s->frame_size_min));
put_bits(&s->pb, 5, s->bsid);
put_bits(&s->pb, 3, s->bsmod);
put_bits(&s->pb, 3, s->acmod);
if ((s->acmod & 0x01) && s->acmod != 0x01)
put_bits(&s->pb, 2, s->sr_code);
put_bits(&s->pb, 6, s->frame_size_code + (s->frame_size - s->frame_size_min));
put_bits(&s->pb, 5, s->bitstream_id);
put_bits(&s->pb, 3, s->bitstream_mode);
put_bits(&s->pb, 3, s->channel_mode);
if ((s->channel_mode & 0x01) && s->channel_mode != AC3_CHMODE_MONO)
put_bits(&s->pb, 2, 1); /* XXX -4.5 dB */
if (s->acmod & 0x04)
if (s->channel_mode & 0x04)
put_bits(&s->pb, 2, 1); /* XXX -6 dB */
if (s->acmod == 0x02)
if (s->channel_mode == AC3_CHMODE_STEREO)
put_bits(&s->pb, 2, 0); /* surround not indicated */
put_bits(&s->pb, 1, s->lfe); /* LFE */
put_bits(&s->pb, 5, 31); /* dialog norm: -31 db */
@@ -928,7 +732,7 @@ static void output_frame_header(AC3EncodeContext *s, unsigned char *frame)
put_bits(&s->pb, 1, 1); /* original bitstream */
put_bits(&s->pb, 1, 0); /* no time code 1 */
put_bits(&s->pb, 1, 0); /* no time code 2 */
put_bits(&s->pb, 1, 0); /* no addtional bit stream info */
put_bits(&s->pb, 1, 0); /* no additional bit stream info */
}
/* symetric quantization on 'levels' levels */
@@ -968,7 +772,7 @@ static inline int asym_quant(int c, int e, int qbits)
return v & ((1 << qbits)-1);
}
/* Output one audio block. There are NB_BLOCKS audio blocks in one AC3
/* Output one audio block. There are NB_BLOCKS audio blocks in one AC-3
frame */
static void output_audio_block(AC3EncodeContext *s,
uint8_t exp_strategy[AC3_MAX_CHANNELS],
@@ -1000,7 +804,7 @@ static void output_audio_block(AC3EncodeContext *s,
put_bits(&s->pb, 1, 0); /* no new coupling strategy */
}
if (s->acmod == 2)
if (s->channel_mode == AC3_CHMODE_STEREO)
{
if(block_num==0)
{
@@ -1090,20 +894,20 @@ static void output_audio_block(AC3EncodeContext *s,
baie = (block_num == 0);
put_bits(&s->pb, 1, baie);
if (baie) {
put_bits(&s->pb, 2, s->sdecaycod);
put_bits(&s->pb, 2, s->fdecaycod);
put_bits(&s->pb, 2, s->sgaincod);
put_bits(&s->pb, 2, s->dbkneecod);
put_bits(&s->pb, 3, s->floorcod);
put_bits(&s->pb, 2, s->slow_decay_code);
put_bits(&s->pb, 2, s->fast_decay_code);
put_bits(&s->pb, 2, s->slow_gain_code);
put_bits(&s->pb, 2, s->db_per_bit_code);
put_bits(&s->pb, 3, s->floor_code);
}
/* snr offset */
put_bits(&s->pb, 1, baie); /* always present with bai */
if (baie) {
put_bits(&s->pb, 6, s->csnroffst);
put_bits(&s->pb, 6, s->coarse_snr_offset);
for(ch=0;ch<s->nb_all_channels;ch++) {
put_bits(&s->pb, 4, s->fsnroffst[ch]);
put_bits(&s->pb, 3, s->fgaincod[ch]);
put_bits(&s->pb, 4, s->fine_snr_offset[ch]);
put_bits(&s->pb, 3, s->fast_gain_code[ch]);
}
}
@@ -1243,35 +1047,8 @@ static void output_audio_block(AC3EncodeContext *s,
}
}
/* compute the ac3 crc */
#define CRC16_POLY ((1 << 0) | (1 << 2) | (1 << 15) | (1 << 16))
static void ac3_crc_init(void)
{
unsigned int c, n, k;
for(n=0;n<256;n++) {
c = n << 8;
for (k = 0; k < 8; k++) {
if (c & (1 << 15))
c = ((c << 1) & 0xffff) ^ (CRC16_POLY & 0xffff);
else
c = c << 1;
}
crc_table[n] = c;
}
}
static unsigned int ac3_crc(uint8_t *data, int n, unsigned int crc)
{
int i;
for(i=0;i<n;i++) {
crc = (crc_table[data[i] ^ (crc >> 8)] ^ (crc << 8)) & 0xffff;
}
return crc;
}
static unsigned int mul_poly(unsigned int a, unsigned int b, unsigned int poly)
{
unsigned int c;
@@ -1349,16 +1126,17 @@ static int output_frame_end(AC3EncodeContext *s)
/* Now we must compute both crcs : this is not so easy for crc1
because it is at the beginning of the data... */
frame_size_58 = (frame_size >> 1) + (frame_size >> 3);
crc1 = ac3_crc(frame + 4, (2 * frame_size_58) - 4, 0);
crc1 = bswap_16(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0,
frame + 4, 2 * frame_size_58 - 4));
/* XXX: could precompute crc_inv */
crc_inv = pow_poly((CRC16_POLY >> 1), (16 * frame_size_58) - 16, CRC16_POLY);
crc1 = mul_poly(crc_inv, crc1, CRC16_POLY);
frame[2] = crc1 >> 8;
frame[3] = crc1;
AV_WB16(frame+2,crc1);
crc2 = ac3_crc(frame + 2 * frame_size_58, (frame_size - frame_size_58) * 2 - 2, 0);
frame[2*frame_size - 2] = crc2 >> 8;
frame[2*frame_size - 1] = crc2;
crc2 = bswap_16(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0,
frame + 2 * frame_size_58,
(frame_size - frame_size_58) * 2 - 2));
AV_WB16(frame+2*frame_size-2,crc2);
// printf("n=%d frame_size=%d\n", n, frame_size);
return frame_size * 2;
@@ -1368,7 +1146,7 @@ static int AC3_encode_frame(AVCodecContext *avctx,
unsigned char *frame, int buf_size, void *data)
{
AC3EncodeContext *s = avctx->priv_data;
short *samples = data;
int16_t *samples = data;
int i, j, k, v, ch;
int16_t input_samples[N];
int32_t mdct_coef[NB_BLOCKS][AC3_MAX_CHANNELS][N/2];
@@ -1400,9 +1178,9 @@ static int AC3_encode_frame(AVCodecContext *avctx,
/* apply the MDCT window */
for(j=0;j<N/2;j++) {
input_samples[j] = MUL16(input_samples[j],
ac3_window[j]) >> 15;
ff_ac3_window[j]) >> 15;
input_samples[N-j-1] = MUL16(input_samples[N-j-1],
ac3_window[j]) >> 15;
ff_ac3_window[j]) >> 15;
}
/* Normalize the samples to use the maximum available
@@ -1410,7 +1188,7 @@ static int AC3_encode_frame(AVCodecContext *avctx,
v = 14 - log2_tab(input_samples, N);
if (v < 0)
v = 0;
exp_samples[i][ch] = v - 8;
exp_samples[i][ch] = v - 9;
lshift_tab(input_samples, N, v);
/* do the MDCT */
@@ -1458,6 +1236,15 @@ static int AC3_encode_frame(AVCodecContext *avctx,
}
}
/* adjust for fractional frame sizes */
while(s->bits_written >= s->bit_rate && s->samples_written >= s->sample_rate) {
s->bits_written -= s->bit_rate;
s->samples_written -= s->sample_rate;
}
s->frame_size = s->frame_size_min + (s->bits_written * s->sample_rate < s->samples_written * s->bit_rate);
s->bits_written += s->frame_size * 16;
s->samples_written += AC3_FRAME_SIZE;
compute_bit_allocation(s, bap, encoded_exp, exp_strategy, frame_bits);
/* everything is known... let's output the frame */
output_frame_header(s, frame);
@@ -1469,7 +1256,7 @@ static int AC3_encode_frame(AVCodecContext *avctx,
return output_frame_end(s);
}
static int AC3_encode_close(AVCodecContext *avctx)
static av_cold int AC3_encode_close(AVCodecContext *avctx)
{
av_freep(&avctx->coded_frame);
return 0;
@@ -1479,6 +1266,7 @@ static int AC3_encode_close(AVCodecContext *avctx)
/*************************************************************************/
/* TEST */
#undef random
#define FN (N/4)
void fft_test(void)
@@ -1576,4 +1364,6 @@ AVCodec ac3_encoder = {
AC3_encode_frame,
AC3_encode_close,
NULL,
.sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52 (AC-3, E-AC-3)"),
};
src/add-ons/media/plugins/avcodec/libavcodec/ac3tab.c
+262
View File
@@ -0,0 +1,262 @@
/*
* AC-3 tables
* copyright (c) 2001 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* 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.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file ac3tab.c
* tables taken directly from the AC-3 spec.
*/
#include "ac3tab.h"
/**
* Possible frame sizes.
* from ATSC A/52 Table 5.18 Frame Size Code Table.
*/
const uint16_t ff_ac3_frame_size_tab[38][3] = {
{ 64, 69, 96 },
{ 64, 70, 96 },
{ 80, 87, 120 },
{ 80, 88, 120 },
{ 96, 104, 144 },
{ 96, 105, 144 },
{ 112, 121, 168 },
{ 112, 122, 168 },
{ 128, 139, 192 },
{ 128, 140, 192 },
{ 160, 174, 240 },
{ 160, 175, 240 },
{ 192, 208, 288 },
{ 192, 209, 288 },
{ 224, 243, 336 },
{ 224, 244, 336 },
{ 256, 278, 384 },
{ 256, 279, 384 },
{ 320, 348, 480 },
{ 320, 349, 480 },
{ 384, 417, 576 },
{ 384, 418, 576 },
{ 448, 487, 672 },
{ 448, 488, 672 },
{ 512, 557, 768 },
{ 512, 558, 768 },
{ 640, 696, 960 },
{ 640, 697, 960 },
{ 768, 835, 1152 },
{ 768, 836, 1152 },
{ 896, 975, 1344 },
{ 896, 976, 1344 },
{ 1024, 1114, 1536 },
{ 1024, 1115, 1536 },
{ 1152, 1253, 1728 },
{ 1152, 1254, 1728 },
{ 1280, 1393, 1920 },
{ 1280, 1394, 1920 },
};
/**
* Maps audio coding mode (acmod) to number of full-bandwidth channels.
* from ATSC A/52 Table 5.8 Audio Coding Mode
*/
const uint8_t ff_ac3_channels_tab[8] = {
2, 1, 2, 3, 3, 4, 4, 5
};
/* possible frequencies */
const uint16_t ff_ac3_sample_rate_tab[3] = { 48000, 44100, 32000 };
/* possible bitrates */
const uint16_t ff_ac3_bitrate_tab[19] = {
32, 40, 48, 56, 64, 80, 96, 112, 128,
160, 192, 224, 256, 320, 384, 448, 512, 576, 640
};
/* AC-3 MDCT window */
/* MDCT window */
const int16_t ff_ac3_window[256] = {
4, 7, 12, 16, 21, 28, 34, 42,
51, 61, 72, 84, 97, 111, 127, 145,
164, 184, 207, 231, 257, 285, 315, 347,
382, 419, 458, 500, 544, 591, 641, 694,
750, 810, 872, 937, 1007, 1079, 1155, 1235,
1318, 1406, 1497, 1593, 1692, 1796, 1903, 2016,
2132, 2253, 2379, 2509, 2644, 2783, 2927, 3076,
3230, 3389, 3552, 3721, 3894, 4072, 4255, 4444,
4637, 4835, 5038, 5246, 5459, 5677, 5899, 6127,
6359, 6596, 6837, 7083, 7334, 7589, 7848, 8112,
8380, 8652, 8927, 9207, 9491, 9778,10069,10363,
10660,10960,11264,11570,11879,12190,12504,12820,
13138,13458,13780,14103,14427,14753,15079,15407,
15735,16063,16392,16720,17049,17377,17705,18032,
18358,18683,19007,19330,19651,19970,20287,20602,
20914,21225,21532,21837,22139,22438,22733,23025,
23314,23599,23880,24157,24430,24699,24964,25225,
25481,25732,25979,26221,26459,26691,26919,27142,
27359,27572,27780,27983,28180,28373,28560,28742,
28919,29091,29258,29420,29577,29729,29876,30018,
30155,30288,30415,30538,30657,30771,30880,30985,
31086,31182,31274,31363,31447,31528,31605,31678,
31747,31814,31877,31936,31993,32046,32097,32145,
32190,32232,32272,32310,32345,32378,32409,32438,
32465,32490,32513,32535,32556,32574,32592,32608,
32623,32636,32649,32661,32671,32681,32690,32698,
32705,32712,32718,32724,32729,32733,32737,32741,
32744,32747,32750,32752,32754,32756,32757,32759,
32760,32761,32762,32763,32764,32764,32765,32765,
32766,32766,32766,32766,32767,32767,32767,32767,
32767,32767,32767,32767,32767,32767,32767,32767,
32767,32767,32767,32767,32767,32767,32767,32767,
};
const uint8_t ff_ac3_log_add_tab[260]= {
0x40,0x3f,0x3e,0x3d,0x3c,0x3b,0x3a,0x39,0x38,0x37,
0x36,0x35,0x34,0x34,0x33,0x32,0x31,0x30,0x2f,0x2f,
0x2e,0x2d,0x2c,0x2c,0x2b,0x2a,0x29,0x29,0x28,0x27,
0x26,0x26,0x25,0x24,0x24,0x23,0x23,0x22,0x21,0x21,
0x20,0x20,0x1f,0x1e,0x1e,0x1d,0x1d,0x1c,0x1c,0x1b,
0x1b,0x1a,0x1a,0x19,0x19,0x18,0x18,0x17,0x17,0x16,
0x16,0x15,0x15,0x15,0x14,0x14,0x13,0x13,0x13,0x12,
0x12,0x12,0x11,0x11,0x11,0x10,0x10,0x10,0x0f,0x0f,
0x0f,0x0e,0x0e,0x0e,0x0d,0x0d,0x0d,0x0d,0x0c,0x0c,
0x0c,0x0c,0x0b,0x0b,0x0b,0x0b,0x0a,0x0a,0x0a,0x0a,
0x0a,0x09,0x09,0x09,0x09,0x09,0x08,0x08,0x08,0x08,
0x08,0x08,0x07,0x07,0x07,0x07,0x07,0x07,0x06,0x06,
0x06,0x06,0x06,0x06,0x06,0x06,0x05,0x05,0x05,0x05,
0x05,0x05,0x05,0x05,0x04,0x04,0x04,0x04,0x04,0x04,
0x04,0x04,0x04,0x04,0x04,0x03,0x03,0x03,0x03,0x03,
0x03,0x03,0x03,0x03,0x03,0x03,0x03,0x03,0x03,0x02,
0x02,0x02,0x02,0x02,0x02,0x02,0x02,0x02,0x02,0x02,
0x02,0x02,0x02,0x02,0x02,0x02,0x02,0x02,0x01,0x01,
0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,
0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,
0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,
0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
};
const uint16_t ff_ac3_hearing_threshold_tab[50][3]= {
{ 0x04d0,0x04f0,0x0580 },
{ 0x04d0,0x04f0,0x0580 },
{ 0x0440,0x0460,0x04b0 },
{ 0x0400,0x0410,0x0450 },
{ 0x03e0,0x03e0,0x0420 },
{ 0x03c0,0x03d0,0x03f0 },
{ 0x03b0,0x03c0,0x03e0 },
{ 0x03b0,0x03b0,0x03d0 },
{ 0x03a0,0x03b0,0x03c0 },
{ 0x03a0,0x03a0,0x03b0 },
{ 0x03a0,0x03a0,0x03b0 },
{ 0x03a0,0x03a0,0x03b0 },
{ 0x03a0,0x03a0,0x03a0 },
{ 0x0390,0x03a0,0x03a0 },
{ 0x0390,0x0390,0x03a0 },
{ 0x0390,0x0390,0x03a0 },
{ 0x0380,0x0390,0x03a0 },
{ 0x0380,0x0380,0x03a0 },
{ 0x0370,0x0380,0x03a0 },
{ 0x0370,0x0380,0x03a0 },
{ 0x0360,0x0370,0x0390 },
{ 0x0360,0x0370,0x0390 },
{ 0x0350,0x0360,0x0390 },
{ 0x0350,0x0360,0x0390 },
{ 0x0340,0x0350,0x0380 },
{ 0x0340,0x0350,0x0380 },
{ 0x0330,0x0340,0x0380 },
{ 0x0320,0x0340,0x0370 },
{ 0x0310,0x0320,0x0360 },
{ 0x0300,0x0310,0x0350 },
{ 0x02f0,0x0300,0x0340 },
{ 0x02f0,0x02f0,0x0330 },
{ 0x02f0,0x02f0,0x0320 },
{ 0x02f0,0x02f0,0x0310 },
{ 0x0300,0x02f0,0x0300 },
{ 0x0310,0x0300,0x02f0 },
{ 0x0340,0x0320,0x02f0 },
{ 0x0390,0x0350,0x02f0 },
{ 0x03e0,0x0390,0x0300 },
{ 0x0420,0x03e0,0x0310 },
{ 0x0460,0x0420,0x0330 },
{ 0x0490,0x0450,0x0350 },
{ 0x04a0,0x04a0,0x03c0 },
{ 0x0460,0x0490,0x0410 },
{ 0x0440,0x0460,0x0470 },
{ 0x0440,0x0440,0x04a0 },
{ 0x0520,0x0480,0x0460 },
{ 0x0800,0x0630,0x0440 },
{ 0x0840,0x0840,0x0450 },
{ 0x0840,0x0840,0x04e0 },
};
const uint8_t ff_ac3_bap_tab[64]= {
0, 1, 1, 1, 1, 1, 2, 2, 3, 3,
3, 4, 4, 5, 5, 6, 6, 6, 6, 7,
7, 7, 7, 8, 8, 8, 8, 9, 9, 9,
9, 10, 10, 10, 10, 11, 11, 11, 11, 12,
12, 12, 12, 13, 13, 13, 13, 14, 14, 14,
14, 14, 14, 14, 14, 15, 15, 15, 15, 15,
15, 15, 15, 15,
};
const uint8_t ff_ac3_slow_decay_tab[4]={
0x0f, 0x11, 0x13, 0x15,
};
const uint8_t ff_ac3_fast_decay_tab[4]={
0x3f, 0x53, 0x67, 0x7b,
};
const uint16_t ff_ac3_slow_gain_tab[4]= {
0x540, 0x4d8, 0x478, 0x410,
};
const uint16_t ff_ac3_db_per_bit_tab[4]= {
0x000, 0x700, 0x900, 0xb00,
};
const int16_t ff_ac3_floor_tab[8]= {
0x2f0, 0x2b0, 0x270, 0x230, 0x1f0, 0x170, 0x0f0, 0xf800,
};
const uint16_t ff_ac3_fast_gain_tab[8]= {
0x080, 0x100, 0x180, 0x200, 0x280, 0x300, 0x380, 0x400,
};
const uint8_t ff_ac3_critical_band_size_tab[50]={
1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 3, 3, 3, 3, 3, 3,
3, 6, 6, 6, 6, 6, 6, 12, 12, 12, 12, 24, 24, 24, 24, 24
};
/**
* Default channel map for a dependent substream defined by acmod
*/
const uint16_t ff_eac3_default_chmap[8] = {
AC3_CHMAP_L | AC3_CHMAP_R, // FIXME Ch1+Ch2
AC3_CHMAP_C,
AC3_CHMAP_L | AC3_CHMAP_R,
AC3_CHMAP_L | AC3_CHMAP_C | AC3_CHMAP_R,
AC3_CHMAP_L | AC3_CHMAP_R | AC3_CHMAP_C_SUR,
AC3_CHMAP_L | AC3_CHMAP_C | AC3_CHMAP_R | AC3_CHMAP_C_SUR,
AC3_CHMAP_L | AC3_CHMAP_R | AC3_CHMAP_L_SUR | AC3_CHMAP_R_SUR,
AC3_CHMAP_L | AC3_CHMAP_C | AC3_CHMAP_R | AC3_CHMAP_L_SUR | AC3_CHMAP_R_SUR
};