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merged ffmpeg 0.6.1 into trunk

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git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@39560 a95241bf-73f2-0310-859d-f6bbb57e9c96
This commit is contained in:
Jérôme Duval 2010-11-21 17:51:21 +00:00
parent 3bb30583a3
commit 3ae6771c5f
23 changed files with 1950 additions and 322 deletions
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3
src/add-ons/media/plugins/ffmpeg/libavcodec/Jamfile
View File

@ -25,12 +25,13 @@ StaticLibrary libavcodec.a :
4xm.c
8bps.c
8svx.c
aac.c
aac_ac3_parser.c
aac_adtstoasc_bsf.c
aac_parser.c
aaccoder.c
aacdec.c
aacenc.c
aacps.c
aacpsy.c
aacsbr.c
aactab.c

13
src/add-ons/media/plugins/ffmpeg/libavcodec/aac.h
View File

@ -38,12 +38,6 @@
#include <stdint.h>
#define AAC_INIT_VLC_STATIC(num, size) \
INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
size);
#define MAX_CHANNELS 64
#define MAX_ELEM_ID 16
@ -241,7 +235,7 @@ typedef struct {
* main AAC context
*/
typedef struct {
AVCodecContext * avccontext;
AVCodecContext *avctx;
MPEG4AudioConfig m4ac;
@ -255,8 +249,9 @@ typedef struct {
enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
* first index as the first 4 raw data block types
*/
ChannelElement * che[4][MAX_ELEM_ID];
ChannelElement * tag_che_map[4][MAX_ELEM_ID];
ChannelElement *che[4][MAX_ELEM_ID];
ChannelElement *tag_che_map[4][MAX_ELEM_ID];
uint8_t tags_seen_this_frame[4][MAX_ELEM_ID];
int tags_mapped;
/** @} */

39
src/add-ons/media/plugins/ffmpeg/libavcodec/aac_tablegen.c Normal file
View File

@ -0,0 +1,39 @@
/*
* Generate a header file for hardcoded AAC tables
*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* 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 <stdlib.h>
#define CONFIG_HARDCODED_TABLES 0
#include "aac_tablegen.h"
#include "tableprint.h"
int main(void)
{
ff_aac_tableinit();
write_fileheader();
printf("const float ff_aac_pow2sf_tab[428] = {\n");
write_float_array(ff_aac_pow2sf_tab, 428);
printf("};\n");
return 0;
}

42
src/add-ons/media/plugins/ffmpeg/libavcodec/aac_tablegen.h Normal file
View File

@ -0,0 +1,42 @@
/*
* Header file for hardcoded AAC tables
*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* 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
*/
#ifndef AAC_TABLEGEN_H
#define AAC_TABLEGEN_H
#include "aac_tablegen_decl.h"
#if CONFIG_HARDCODED_TABLES
#include "libavcodec/aac_tables.h"
#else
#include "../libavutil/mathematics.h"
float ff_aac_pow2sf_tab[428];
void ff_aac_tableinit(void)
{
int i;
for (i = 0; i < 428; i++)
ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
}
#endif /* CONFIG_HARDCODED_TABLES */
#endif /* AAC_TABLEGEN_H */

34
src/add-ons/media/plugins/ffmpeg/libavcodec/aac_tablegen_decl.h Normal file
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@ -0,0 +1,34 @@
/*
* Header file for hardcoded AAC tables
*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* 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
*/
#ifndef AAC_TABLEGEN_INIT_H
#define AAC_TABLEGEN_INIT_H
#if CONFIG_HARDCODED_TABLES
#define ff_aac_tableinit()
extern const float ff_aac_pow2sf_tab[428];
#else
void ff_aac_tableinit(void);
extern float ff_aac_pow2sf_tab[428];
#endif /* CONFIG_HARDCODED_TABLES */
#endif /* AAC_TABLEGEN_INIT_H */

174
src/add-ons/media/plugins/ffmpeg/libavcodec/aac.c → src/add-ons/media/plugins/ffmpeg/libavcodec/aacdec.c
View File

@ -67,7 +67,7 @@
* Y (not in this code) Layer-2
* Y (not in this code) Layer-3
* N SinuSoidal Coding (Transient, Sinusoid, Noise)
* N (planned) Parametric Stereo
* Y Parametric Stereo
* N Direct Stream Transfer
*
* Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
@ -113,6 +113,22 @@ static const char overread_err[] = "Input buffer exhausted before END element fo
static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
{
/* Some buggy encoders appear to set all elem_ids to zero and rely on
channels always occurring in the same order. This is expressly forbidden
by the spec but we will try to work around it.
*/
int err_printed = 0;
while (ac->tags_seen_this_frame[type][elem_id] && elem_id < MAX_ELEM_ID) {
if (ac->output_configured < OC_LOCKED && !err_printed) {
av_log(ac->avctx, AV_LOG_WARNING, "Duplicate channel tag found, attempting to remap.\n");
err_printed = 1;
}
elem_id++;
}
if (elem_id == MAX_ELEM_ID)
return NULL;
ac->tags_seen_this_frame[type][elem_id] = 1;
if (ac->tag_che_map[type][elem_id]) {
return ac->tag_che_map[type][elem_id];
}
@ -127,8 +143,8 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
}
case 6:
/* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have
encountered such a stream, transfer the LFE[0] element to SCE[1] */
instead of SCE[0] CPE[0] CPE[1] LFE[0]. If we seem to have
encountered such a stream, transfer the LFE[0] element to the SCE[1]'s mapping */
if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
ac->tags_mapped++;
return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
@ -184,7 +200,8 @@ static av_cold int che_configure(AACContext *ac,
ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr);
if (type != TYPE_CCE) {
ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
if (type == TYPE_CPE) {
if (type == TYPE_CPE ||
(type == TYPE_SCE && ac->m4ac.ps == 1)) {
ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
}
}
@ -209,9 +226,10 @@ static av_cold int output_configure(AACContext *ac,
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
int channel_config, enum OCStatus oc_type)
{
AVCodecContext *avctx = ac->avccontext;
AVCodecContext *avctx = ac->avctx;
int i, type, channels = 0, ret;
if (new_che_pos != che_pos)
memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
if (channel_config) {
@ -292,7 +310,7 @@ static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_EL
sampling_index = get_bits(gb, 4);
if (ac->m4ac.sampling_index != sampling_index)
av_log(ac->avccontext, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");
av_log(ac->avctx, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");
num_front = get_bits(gb, 4);
num_side = get_bits(gb, 4);
@ -323,7 +341,7 @@ static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_EL
/* comment field, first byte is length */
comment_len = get_bits(gb, 8) * 8;
if (get_bits_left(gb) < comment_len) {
av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
av_log(ac->avctx, AV_LOG_ERROR, overread_err);
return -1;
}
skip_bits_long(gb, comment_len);
@ -343,7 +361,7 @@ static av_cold int set_default_channel_config(AACContext *ac,
int channel_config)
{
if (channel_config < 1 || channel_config > 7) {
av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
av_log(ac->avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
channel_config);
return -1;
}
@ -388,7 +406,7 @@ static int decode_ga_specific_config(AACContext *ac, GetBitContext *gb,
int extension_flag, ret;
if (get_bits1(gb)) { // frameLengthFlag
av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1);
av_log_missing_feature(ac->avctx, "960/120 MDCT window is", 1);
return -1;
}
@ -452,9 +470,11 @@ static int decode_audio_specific_config(AACContext *ac, void *data,
if ((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
return -1;
if (ac->m4ac.sampling_index > 12) {
av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
return -1;
}
if (ac->m4ac.sbr == 1 && ac->m4ac.ps == -1)
ac->m4ac.ps = 1;
skip_bits_long(&gb, i);
@ -465,7 +485,7 @@ static int decode_audio_specific_config(AACContext *ac, void *data,
return -1;
break;
default:
av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
av_log(ac->avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
return -1;
}
@ -508,20 +528,25 @@ static void reset_predictor_group(PredictorState *ps, int group_num)
reset_predict_state(&ps[i]);
}
static av_cold int aac_decode_init(AVCodecContext *avccontext)
#define AAC_INIT_VLC_STATIC(num, size) \
INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
size);
static av_cold int aac_decode_init(AVCodecContext *avctx)
{
AACContext *ac = avccontext->priv_data;
int i;
AACContext *ac = avctx->priv_data;
ac->avccontext = avccontext;
ac->m4ac.sample_rate = avccontext->sample_rate;
ac->avctx = avctx;
ac->m4ac.sample_rate = avctx->sample_rate;
if (avccontext->extradata_size > 0) {
if (decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
if (avctx->extradata_size > 0) {
if (decode_audio_specific_config(ac, avctx->extradata, avctx->extradata_size))
return -1;
}
avccontext->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = SAMPLE_FMT_S16;
AAC_INIT_VLC_STATIC( 0, 304);
AAC_INIT_VLC_STATIC( 1, 270);
@ -537,7 +562,7 @@ static av_cold int aac_decode_init(AVCodecContext *avccontext)
ff_aac_sbr_init();
dsputil_init(&ac->dsp, avccontext);
dsputil_init(&ac->dsp, avctx);
ac->random_state = 0x1f2e3d4c;
@ -555,10 +580,7 @@ static av_cold int aac_decode_init(AVCodecContext *avccontext)
ac->sf_offset = 60;
}
#if !CONFIG_HARDCODED_TABLES
for (i = 0; i < 428; i++)
ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
#endif /* CONFIG_HARDCODED_TABLES */
ff_aac_tableinit();
INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
@ -591,7 +613,7 @@ static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
align_get_bits(gb);
if (get_bits_left(gb) < 8 * count) {
av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
av_log(ac->avctx, AV_LOG_ERROR, overread_err);
return -1;
}
skip_bits_long(gb, 8 * count);
@ -605,7 +627,7 @@ static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
if (get_bits1(gb)) {
ics->predictor_reset_group = get_bits(gb, 5);
if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
return -1;
}
}
@ -624,7 +646,7 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
GetBitContext *gb, int common_window)
{
if (get_bits1(gb)) {
av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
memset(ics, 0, sizeof(IndividualChannelStream));
return -1;
}
@ -665,11 +687,11 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
return -1;
}
} else if (ac->m4ac.object_type == AOT_AAC_LC) {
av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
memset(ics, 0, sizeof(IndividualChannelStream));
return -1;
} else {
av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1);
av_log_missing_feature(ac->avctx, "Predictor bit set but LTP is", 1);
memset(ics, 0, sizeof(IndividualChannelStream));
return -1;
}
@ -677,7 +699,7 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
}
if (ics->max_sfb > ics->num_swb) {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
ics->max_sfb, ics->num_swb);
memset(ics, 0, sizeof(IndividualChannelStream));
@ -708,18 +730,18 @@ static int decode_band_types(AACContext *ac, enum BandType band_type[120],
int sect_len_incr;
int sect_band_type = get_bits(gb, 4);
if (sect_band_type == 12) {
av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n");
av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
return -1;
}
while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
sect_end += sect_len_incr;
sect_end += sect_len_incr;
if (get_bits_left(gb) < 0) {
av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
av_log(ac->avctx, AV_LOG_ERROR, overread_err);
return -1;
}
if (sect_end > ics->max_sfb) {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"Number of bands (%d) exceeds limit (%d).\n",
sect_end, ics->max_sfb);
return -1;
@ -764,7 +786,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
for (; i < run_end; i++, idx++) {
offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
if (offset[2] > 255U) {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"%s (%d) out of range.\n", sf_str[2], offset[2]);
return -1;
}
@ -777,7 +799,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
else
offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
if (offset[1] > 255U) {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"%s (%d) out of range.\n", sf_str[1], offset[1]);
return -1;
}
@ -787,7 +809,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
for (; i < run_end; i++, idx++) {
offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
if (offset[0] > 255U) {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"%s (%d) out of range.\n", sf_str[0], offset[0]);
return -1;
}
@ -844,7 +866,7 @@ static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
tns->order[w][filt], tns_max_order);
tns->order[w][filt] = 0;
return -1;
@ -1163,7 +1185,7 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
b = 31 - av_log2(~b);
if (b > 8) {
av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
return -1;
}
@ -1216,7 +1238,7 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
return 0;
err_cb_overflow:
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n",
band_type[idx], err_idx, ff_aac_spectral_sizes[band_type[idx]]);
return -1;
@ -1337,18 +1359,18 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
if (!scale_flag) {
if ((pulse_present = get_bits1(gb))) {
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
return -1;
}
if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
return -1;
}
}
if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
return -1;
if (get_bits1(gb)) {
av_log_missing_feature(ac->avccontext, "SSR", 1);
av_log_missing_feature(ac->avctx, "SSR", 1);
return -1;
}
}
@ -1448,7 +1470,7 @@ static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
cpe->ch[1].ics.use_kb_window[1] = i;
ms_present = get_bits(gb, 2);
if (ms_present == 3) {
av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
return -1;
} else if (ms_present)
decode_mid_side_stereo(cpe, gb, ms_present);
@ -1635,16 +1657,20 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
crc_flag++;
case EXT_SBR_DATA:
if (!che) {
av_log(ac->avccontext, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
return res;
} else if (!ac->m4ac.sbr) {
av_log(ac->avccontext, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
skip_bits_long(gb, 8 * cnt - 4);
return res;
} else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) {
av_log(ac->avccontext, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
skip_bits_long(gb, 8 * cnt - 4);
return res;
} else if (ac->m4ac.ps == -1 && ac->output_configured < OC_LOCKED && ac->avctx->channels == 1) {
ac->m4ac.sbr = 1;
ac->m4ac.ps = 1;
output_configure(ac, ac->che_pos, ac->che_pos, ac->m4ac.chan_config, ac->output_configured);
} else {
ac->m4ac.sbr = 1;
}
@ -1728,7 +1754,7 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce, float
// imdct
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE)
av_log(ac->avccontext, AV_LOG_WARNING,
av_log(ac->avctx, AV_LOG_WARNING,
"Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. "
"If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n");
for (i = 0; i < 1024; i += 128)
@ -1794,7 +1820,7 @@ static void apply_dependent_coupling(AACContext *ac,
const float *src = cce->ch[0].coeffs;
int g, i, group, k, idx = 0;
if (ac->m4ac.object_type == AOT_AAC_LTP) {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"Dependent coupling is not supported together with LTP\n");
return;
}
@ -1924,58 +1950,65 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
} else if (ac->output_configured != OC_LOCKED) {
ac->output_configured = OC_NONE;
}
if (ac->output_configured != OC_LOCKED)
if (ac->output_configured != OC_LOCKED) {
ac->m4ac.sbr = -1;
ac->m4ac.ps = -1;
}
ac->m4ac.sample_rate = hdr_info.sample_rate;
ac->m4ac.sampling_index = hdr_info.sampling_index;
ac->m4ac.object_type = hdr_info.object_type;
if (!ac->avccontext->sample_rate)
ac->avccontext->sample_rate = hdr_info.sample_rate;
if (!ac->avctx->sample_rate)
ac->avctx->sample_rate = hdr_info.sample_rate;
if (hdr_info.num_aac_frames == 1) {
if (!hdr_info.crc_absent)
skip_bits(gb, 16);
} else {
av_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0);
av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0);
return -1;
}
}
return size;
}
static int aac_decode_frame(AVCodecContext *avccontext, void *data,
static int aac_decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AACContext *ac = avccontext->priv_data;
AACContext *ac = avctx->priv_data;
ChannelElement *che = NULL, *che_prev = NULL;
GetBitContext gb;
enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
int err, elem_id, data_size_tmp;
int buf_consumed;
int samples = 1024, multiplier;
int samples = 0, multiplier;
int buf_offset;
init_get_bits(&gb, buf, buf_size * 8);
if (show_bits(&gb, 12) == 0xfff) {
if (parse_adts_frame_header(ac, &gb) < 0) {
av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
return -1;
}
if (ac->m4ac.sampling_index > 12) {
av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
return -1;
}
}
memset(ac->tags_seen_this_frame, 0, sizeof(ac->tags_seen_this_frame));
// parse
while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
elem_id = get_bits(&gb, 4);
if (elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) {
av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id);
return -1;
if (elem_type < TYPE_DSE) {
if (!(che=get_che(ac, elem_type, elem_id))) {
av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
elem_type, elem_id);
return -1;
}
samples = 1024;
}
switch (elem_type) {
@ -2006,7 +2039,7 @@ static int aac_decode_frame(AVCodecContext *avccontext, void *data,
if ((err = decode_pce(ac, new_che_pos, &gb)))
break;
if (ac->output_configured > OC_TRIAL_PCE)
av_log(avccontext, AV_LOG_ERROR,
av_log(avctx, AV_LOG_ERROR,
"Not evaluating a further program_config_element as this construct is dubious at best.\n");
else
err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
@ -2017,7 +2050,7 @@ static int aac_decode_frame(AVCodecContext *avccontext, void *data,
if (elem_id == 15)
elem_id += get_bits(&gb, 8) - 1;
if (get_bits_left(&gb) < 8 * elem_id) {
av_log(avccontext, AV_LOG_ERROR, overread_err);
av_log(avctx, AV_LOG_ERROR, overread_err);
return -1;
}
while (elem_id > 0)
@ -2037,7 +2070,7 @@ static int aac_decode_frame(AVCodecContext *avccontext, void *data,
return err;
if (get_bits_left(&gb) < 3) {
av_log(avccontext, AV_LOG_ERROR, overread_err);
av_log(avctx, AV_LOG_ERROR, overread_err);
return -1;
}
}
@ -2047,20 +2080,21 @@ static int aac_decode_frame(AVCodecContext *avccontext, void *data,
multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0;
samples <<= multiplier;
if (ac->output_configured < OC_LOCKED) {
avccontext->sample_rate = ac->m4ac.sample_rate << multiplier;
avccontext->frame_size = samples;
avctx->sample_rate = ac->m4ac.sample_rate << multiplier;
avctx->frame_size = samples;
}
data_size_tmp = samples * avccontext->channels * sizeof(int16_t);
data_size_tmp = samples * avctx->channels * sizeof(int16_t);
if (*data_size < data_size_tmp) {
av_log(avccontext, AV_LOG_ERROR,
av_log(avctx, AV_LOG_ERROR,
"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
*data_size, data_size_tmp);
return -1;
}
*data_size = data_size_tmp;
ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avccontext->channels);
if (samples)
ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
if (ac->output_configured)
ac->output_configured = OC_LOCKED;
@ -2073,9 +2107,9 @@ static int aac_decode_frame(AVCodecContext *avccontext, void *data,
return buf_size > buf_offset ? buf_consumed : buf_size;
}
static av_cold int aac_decode_close(AVCodecContext *avccontext)
static av_cold int aac_decode_close(AVCodecContext *avctx)
{
AACContext *ac = avccontext->priv_data;
AACContext *ac = avctx->priv_data;
int i, type;
for (i = 0; i < MAX_ELEM_ID; i++) {

25
src/add-ons/media/plugins/ffmpeg/libavcodec/aacenc.c
View File

@ -201,13 +201,11 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
lengths[1] = ff_aac_num_swb_128[i];
ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
s->psypp = ff_psy_preprocess_init(avctx);
s->coder = &ff_aac_coders[0];
s->coder = &ff_aac_coders[2];
s->lambda = avctx->global_quality ? avctx->global_quality : 120;
#if !CONFIG_HARDCODED_TABLES
for (i = 0; i < 428; i++)
ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
#endif /* CONFIG_HARDCODED_TABLES */
ff_aac_tableinit();
if (avctx->channels > 5)
av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. "
@ -234,25 +232,21 @@ static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
s->output[i] = sce->saved[i];
}
if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
j = channel;
for (i = 0; i < 1024; i++, j += avctx->channels) {
for (i = 0, j = channel; i < 1024; i++, j += avctx->channels) {
s->output[i+1024] = audio[j] * lwindow[1024 - i - 1];
sce->saved[i] = audio[j] * lwindow[i];
}
} else {
j = channel;
for (i = 0; i < 448; i++, j += avctx->channels)
for (i = 0, j = channel; i < 448; i++, j += avctx->channels)
s->output[i+1024] = audio[j];
for (i = 448; i < 576; i++, j += avctx->channels)
for (; i < 576; i++, j += avctx->channels)
s->output[i+1024] = audio[j] * swindow[576 - i - 1];
memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
j = channel;
for (i = 0; i < 1024; i++, j += avctx->channels)
for (i = 0, j = channel; i < 1024; i++, j += avctx->channels)
sce->saved[i] = audio[j];
}
ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
} else {
j = channel;
for (k = 0; k < 1024; k += 128) {
for (i = 448 + k; i < 448 + k + 256; i++)
s->output[i - 448 - k] = (i < 1024)
@ -262,8 +256,7 @@ static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
}
j = channel;
for (i = 0; i < 1024; i++, j += avctx->channels)
for (i = 0, j = channel; i < 1024; i++, j += avctx->channels)
sce->saved[i] = audio[j];
}
}
@ -562,6 +555,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
cpe = &s->cpe[i];
for (j = 0; j < chans; j++) {
s->cur_channel = start_ch + j;
ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda);
}
cpe->common_window = 0;
@ -592,7 +586,6 @@ static int aac_encode_frame(AVCodecContext *avctx,
}
for (j = 0; j < chans; j++) {
s->cur_channel = start_ch + j;
ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
}
start_ch += chans;

2
src/add-ons/media/plugins/ffmpeg/libavcodec/aacenc.h
View File

@ -64,7 +64,7 @@ typedef struct AACEncContext {
int cur_channel;
int last_frame;
float lambda;
DECLARE_ALIGNED(16, int, qcoefs)[96][2]; ///< quantized coefficients
DECLARE_ALIGNED(16, int, qcoefs)[96]; ///< quantized coefficients
DECLARE_ALIGNED(16, float, scoefs)[1024]; ///< scaled coefficients
} AACEncContext;

1037
src/add-ons/media/plugins/ffmpeg/libavcodec/aacps.c Normal file
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File diff suppressed because it is too large Load Diff

82
src/add-ons/media/plugins/ffmpeg/libavcodec/aacps.h Normal file
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@ -0,0 +1,82 @@
/*
* MPEG-4 Parametric Stereo definitions and declarations
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* 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
*/
#ifndef AVCODEC_PS_H
#define AVCODEC_PS_H
#include <stdint.h>
#include "avcodec.h"
#include "get_bits.h"
#define PS_MAX_NUM_ENV 5
#define PS_MAX_NR_IIDICC 34
#define PS_MAX_NR_IPDOPD 17
#define PS_MAX_SSB 91
#define PS_MAX_AP_BANDS 50
#define PS_QMF_TIME_SLOTS 32
#define PS_MAX_DELAY 14
#define PS_AP_LINKS 3
#define PS_MAX_AP_DELAY 5
typedef struct {
int start;
int enable_iid;
int iid_quant;
int nr_iid_par;
int nr_ipdopd_par;
int enable_icc;
int icc_mode;
int nr_icc_par;
int enable_ext;
int frame_class;
int num_env_old;
int num_env;
int enable_ipdopd;
int border_position[PS_MAX_NUM_ENV+1];
int8_t iid_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Inter-channel Intensity Difference Parameters
int8_t icc_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Inter-Channel Coherence Parameters
/* ipd/opd is iid/icc sized so that the same functions can handle both */
int8_t ipd_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Inter-channel Phase Difference Parameters
int8_t opd_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Overall Phase Difference Parameters
int is34bands;
int is34bands_old;
float in_buf[5][44][2];
float delay[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
float ap_delay[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
float peak_decay_nrg[34];
float power_smooth[34];
float peak_decay_diff_smooth[34];
float H11[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
float H12[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
float H21[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
float H22[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
int8_t opd_hist[PS_MAX_NR_IIDICC];
int8_t ipd_hist[PS_MAX_NR_IIDICC];
} PSContext;
void ff_ps_init(void);
void ff_ps_ctx_init(PSContext *ps);
int ff_ps_read_data(AVCodecContext *avctx, GetBitContext *gb, PSContext *ps, int bits_left);
int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float R[2][38][64], int top);
#endif /* AVCODEC_PS_H */

93
src/add-ons/media/plugins/ffmpeg/libavcodec/aacps_tablegen.c Normal file
View File

@ -0,0 +1,93 @@
/*
* Generate a header file for hardcoded Parametric Stereo tables
*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* 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 <stdlib.h>
#define CONFIG_HARDCODED_TABLES 0
#include "aacps_tablegen.h"
#include "tableprint.h"
void write_float_3d_array (const void *p, int b, int c, int d)
{
int i;
const float *f = p;
for (i = 0; i < b; i++) {
printf("{\n");
write_float_2d_array(f, c, d);
printf("},\n");
f += c * d;
}
}
void write_float_4d_array (const void *p, int a, int b, int c, int d)
{
int i;
const float *f = p;
for (i = 0; i < a; i++) {
printf("{\n");
write_float_3d_array(f, b, c, d);
printf("},\n");
f += b * c * d;
}
}
int main(void)
{
ps_tableinit();
write_fileheader();
printf("static const float pd_re_smooth[8*8*8] = {\n");
write_float_array(pd_re_smooth, 8*8*8);
printf("};\n");
printf("static const float pd_im_smooth[8*8*8] = {\n");
write_float_array(pd_im_smooth, 8*8*8);
printf("};\n");
printf("static const float HA[46][8][4] = {\n");
write_float_3d_array(HA, 46, 8, 4);
printf("};\n");
printf("static const float HB[46][8][4] = {\n");
write_float_3d_array(HB, 46, 8, 4);
printf("};\n");
printf("static const float f20_0_8[8][7][2] = {\n");
write_float_3d_array(f20_0_8, 8, 7, 2);
printf("};\n");
printf("static const float f34_0_12[12][7][2] = {\n");
write_float_3d_array(f34_0_12, 12, 7, 2);
printf("};\n");
printf("static const float f34_1_8[8][7][2] = {\n");
write_float_3d_array(f34_1_8, 8, 7, 2);
printf("};\n");
printf("static const float f34_2_4[4][7][2] = {\n");
write_float_3d_array(f34_2_4, 4, 7, 2);
printf("};\n");
printf("static const float Q_fract_allpass[2][50][3][2] = {\n");
write_float_4d_array(Q_fract_allpass, 2, 50, 3, 2);
printf("};\n");
printf("static const float phi_fract[2][50][2] = {\n");
write_float_3d_array(phi_fract, 2, 50, 2);
printf("};\n");
return 0;
}

212
src/add-ons/media/plugins/ffmpeg/libavcodec/aacps_tablegen.h Normal file
View File

@ -0,0 +1,212 @@
/*
* Header file for hardcoded Parametric Stereo tables
*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* 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
*/
#ifndef AACPS_TABLEGEN_H
#define AACPS_TABLEGEN_H
#include <stdint.h>
#if CONFIG_HARDCODED_TABLES
#define ps_tableinit()
#include "libavcodec/aacps_tables.h"
#else
#include "../libavutil/common.h"
#include "../libavutil/mathematics.h"
#define NR_ALLPASS_BANDS20 30
#define NR_ALLPASS_BANDS34 50
#define PS_AP_LINKS 3
static float pd_re_smooth[8*8*8];
static float pd_im_smooth[8*8*8];
static float HA[46][8][4];
static float HB[46][8][4];
static float f20_0_8 [ 8][7][2];
static float f34_0_12[12][7][2];
static float f34_1_8 [ 8][7][2];
static float f34_2_4 [ 4][7][2];
static float Q_fract_allpass[2][50][3][2];
static float phi_fract[2][50][2];
static const float g0_Q8[] = {
0.00746082949812f, 0.02270420949825f, 0.04546865930473f, 0.07266113929591f,
0.09885108575264f, 0.11793710567217f, 0.125f
};
static const float g0_Q12[] = {
0.04081179924692f, 0.03812810994926f, 0.05144908135699f, 0.06399831151592f,
0.07428313801106f, 0.08100347892914f, 0.08333333333333f
};
static const float g1_Q8[] = {
0.01565675600122f, 0.03752716391991f, 0.05417891378782f, 0.08417044116767f,
0.10307344158036f, 0.12222452249753f, 0.125f
};
static const float g2_Q4[] = {
-0.05908211155639f, -0.04871498374946f, 0.0f, 0.07778723915851f,
0.16486303567403f, 0.23279856662996f, 0.25f
};
static void make_filters_from_proto(float (*filter)[7][2], const float *proto, int bands)
{
int q, n;
for (q = 0; q < bands; q++) {
for (n = 0; n < 7; n++) {
double theta = 2 * M_PI * (q + 0.5) * (n - 6) / bands;
filter[q][n][0] = proto[n] * cos(theta);
filter[q][n][1] = proto[n] * -sin(theta);
}
}
}
static void ps_tableinit(void)
{
static const float ipdopd_sin[] = { 0, M_SQRT1_2, 1, M_SQRT1_2, 0, -M_SQRT1_2, -1, -M_SQRT1_2 };
static const float ipdopd_cos[] = { 1, M_SQRT1_2, 0, -M_SQRT1_2, -1, -M_SQRT1_2, 0, M_SQRT1_2 };
int pd0, pd1, pd2;
static const float iid_par_dequant[] = {
//iid_par_dequant_default
0.05623413251903, 0.12589254117942, 0.19952623149689, 0.31622776601684,
0.44668359215096, 0.63095734448019, 0.79432823472428, 1,
1.25892541179417, 1.58489319246111, 2.23872113856834, 3.16227766016838,
5.01187233627272, 7.94328234724282, 17.7827941003892,
//iid_par_dequant_fine
0.00316227766017, 0.00562341325190, 0.01, 0.01778279410039,
0.03162277660168, 0.05623413251903, 0.07943282347243, 0.11220184543020,
0.15848931924611, 0.22387211385683, 0.31622776601684, 0.39810717055350,
0.50118723362727, 0.63095734448019, 0.79432823472428, 1,
1.25892541179417, 1.58489319246111, 1.99526231496888, 2.51188643150958,
3.16227766016838, 4.46683592150963, 6.30957344480193, 8.91250938133745,
12.5892541179417, 17.7827941003892, 31.6227766016838, 56.2341325190349,
100, 177.827941003892, 316.227766016837,
};
static const float icc_invq[] = {
1, 0.937, 0.84118, 0.60092, 0.36764, 0, -0.589, -1
};
static const float acos_icc_invq[] = {
0, 0.35685527, 0.57133466, 0.92614472, 1.1943263, M_PI/2, 2.2006171, M_PI
};
int iid, icc;
int k, m;
static const int8_t f_center_20[] = {
-3, -1, 1, 3, 5, 7, 10, 14, 18, 22,
};
static const int8_t f_center_34[] = {
2, 6, 10, 14, 18, 22, 26, 30,
34,-10, -6, -2, 51, 57, 15, 21,
27, 33, 39, 45, 54, 66, 78, 42,
102, 66, 78, 90,102,114,126, 90,
};
static const float fractional_delay_links[] = { 0.43f, 0.75f, 0.347f };
const float fractional_delay_gain = 0.39f;
for (pd0 = 0; pd0 < 8; pd0++) {
float pd0_re = ipdopd_cos[pd0];
float pd0_im = ipdopd_sin[pd0];
for (pd1 = 0; pd1 < 8; pd1++) {
float pd1_re = ipdopd_cos[pd1];
float pd1_im = ipdopd_sin[pd1];
for (pd2 = 0; pd2 < 8; pd2++) {
float pd2_re = ipdopd_cos[pd2];
float pd2_im = ipdopd_sin[pd2];
float re_smooth = 0.25f * pd0_re + 0.5f * pd1_re + pd2_re;
float im_smooth = 0.25f * pd0_im + 0.5f * pd1_im + pd2_im;
float pd_mag = 1 / sqrt(im_smooth * im_smooth + re_smooth * re_smooth);
pd_re_smooth[pd0*64+pd1*8+pd2] = re_smooth * pd_mag;
pd_im_smooth[pd0*64+pd1*8+pd2] = im_smooth * pd_mag;
}
}
}
for (iid = 0; iid < 46; iid++) {
float c = iid_par_dequant[iid]; //<Linear Inter-channel Intensity Difference
float c1 = (float)M_SQRT2 / sqrtf(1.0f + c*c);
float c2 = c * c1;
for (icc = 0; icc < 8; icc++) {
/*if (PS_BASELINE || ps->icc_mode < 3)*/ {
float alpha = 0.5f * acos_icc_invq[icc];
float beta = alpha * (c1 - c2) * (float)M_SQRT1_2;
HA[iid][icc][0] = c2 * cosf(beta + alpha);
HA[iid][icc][1] = c1 * cosf(beta - alpha);
HA[iid][icc][2] = c2 * sinf(beta + alpha);
HA[iid][icc][3] = c1 * sinf(beta - alpha);
} /* else */ {
float alpha, gamma, mu, rho;
float alpha_c, alpha_s, gamma_c, gamma_s;
rho = FFMAX(icc_invq[icc], 0.05f);
alpha = 0.5f * atan2f(2.0f * c * rho, c*c - 1.0f);
mu = c + 1.0f / c;
mu = sqrtf(1 + (4 * rho * rho - 4)/(mu * mu));
gamma = atanf(sqrtf((1.0f - mu)/(1.0f + mu)));
if (alpha < 0) alpha += M_PI/2;
alpha_c = cosf(alpha);
alpha_s = sinf(alpha);
gamma_c = cosf(gamma);
gamma_s = sinf(gamma);
HB[iid][icc][0] = M_SQRT2 * alpha_c * gamma_c;
HB[iid][icc][1] = M_SQRT2 * alpha_s * gamma_c;
HB[iid][icc][2] = -M_SQRT2 * alpha_s * gamma_s;
HB[iid][icc][3] = M_SQRT2 * alpha_c * gamma_s;
}
}
}
for (k = 0; k < NR_ALLPASS_BANDS20; k++) {
double f_center, theta;
if (k < FF_ARRAY_ELEMS(f_center_20))
f_center = f_center_20[k] * 0.125;
else
f_center = k - 6.5f;
for (m = 0; m < PS_AP_LINKS; m++) {
theta = -M_PI * fractional_delay_links[m] * f_center;
Q_fract_allpass[0][k][m][0] = cos(theta);
Q_fract_allpass[0][k][m][1] = sin(theta);
}
theta = -M_PI*fractional_delay_gain*f_center;
phi_fract[0][k][0] = cos(theta);
phi_fract[0][k][1] = sin(theta);
}
for (k = 0; k < NR_ALLPASS_BANDS34; k++) {
double f_center, theta;
if (k < FF_ARRAY_ELEMS(f_center_34))
f_center = f_center_34[k] / 24.;
else
f_center = k - 26.5f;
for (m = 0; m < PS_AP_LINKS; m++) {
theta = -M_PI * fractional_delay_links[m] * f_center;
Q_fract_allpass[1][k][m][0] = cos(theta);
Q_fract_allpass[1][k][m][1] = sin(theta);
}
theta = -M_PI*fractional_delay_gain*f_center;
phi_fract[1][k][0] = cos(theta);
phi_fract[1][k][1] = sin(theta);
}
make_filters_from_proto(f20_0_8, g0_Q8, 8);
make_filters_from_proto(f34_0_12, g0_Q12, 12);
make_filters_from_proto(f34_1_8, g1_Q8, 8);
make_filters_from_proto(f34_2_4, g2_Q4, 4);
}
#endif /* CONFIG_HARDCODED_TABLES */
#endif /* AACPS_TABLEGEN_H */

163
src/add-ons/media/plugins/ffmpeg/libavcodec/aacpsdata.c Normal file
View File

@ -0,0 +1,163 @@
/*
* MPEG-4 Parametric Stereo data tables
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* 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
*/
static const uint8_t huff_iid_df1_bits[] = {
18, 18, 18, 18, 18, 18, 18, 18, 18, 17, 18, 17, 17, 16, 16, 15, 14, 14,
13, 12, 12, 11, 10, 10, 8, 7, 6, 5, 4, 3, 1, 3, 4, 5, 6, 7,
8, 9, 10, 11, 11, 12, 13, 14, 14, 15, 16, 16, 17, 17, 18, 17, 18, 18,
18, 18, 18, 18, 18, 18, 18,
};
static const uint32_t huff_iid_df1_codes[] = {
0x01FEB4, 0x01FEB5, 0x01FD76, 0x01FD77, 0x01FD74, 0x01FD75, 0x01FE8A,
0x01FE8B, 0x01FE88, 0x00FE80, 0x01FEB6, 0x00FE82, 0x00FEB8, 0x007F42,
0x007FAE, 0x003FAF, 0x001FD1, 0x001FE9, 0x000FE9, 0x0007EA, 0x0007FB,
0x0003FB, 0x0001FB, 0x0001FF, 0x00007C, 0x00003C, 0x00001C, 0x00000C,
0x000000, 0x000001, 0x000001, 0x000002, 0x000001, 0x00000D, 0x00001D,
0x00003D, 0x00007D, 0x0000FC, 0x0001FC, 0x0003FC, 0x0003F4, 0x0007EB,
0x000FEA, 0x001FEA, 0x001FD6, 0x003FD0, 0x007FAF, 0x007F43, 0x00FEB9,
0x00FE83, 0x01FEB7, 0x00FE81, 0x01FE89, 0x01FE8E, 0x01FE8F, 0x01FE8C,
0x01FE8D, 0x01FEB2, 0x01FEB3, 0x01FEB0, 0x01FEB1,
};
static const uint8_t huff_iid_dt1_bits[] = {
16, 16, 16, 16, 16, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 13,
13, 13, 12, 12, 11, 10, 9, 9, 7, 6, 5, 3, 1, 2, 5, 6, 7, 8,
9, 10, 11, 11, 12, 12, 13, 13, 14, 14, 15, 15, 15, 15, 16, 16, 16, 16,
16, 16, 16, 16, 16, 16, 16,
};
static const uint16_t huff_iid_dt1_codes[] = {
0x004ED4, 0x004ED5, 0x004ECE, 0x004ECF, 0x004ECC, 0x004ED6, 0x004ED8,
0x004F46, 0x004F60, 0x002718, 0x002719, 0x002764, 0x002765, 0x00276D,
0x0027B1, 0x0013B7, 0x0013D6, 0x0009C7, 0x0009E9, 0x0009ED, 0x0004EE,
0x0004F7, 0x000278, 0x000139, 0x00009A, 0x00009F, 0x000020, 0x000011,
0x00000A, 0x000003, 0x000001, 0x000000, 0x00000B, 0x000012, 0x000021,
0x00004C, 0x00009B, 0x00013A, 0x000279, 0x000270, 0x0004EF, 0x0004E2,
0x0009EA, 0x0009D8, 0x0013D7, 0x0013D0, 0x0027B2, 0x0027A2, 0x00271A,
0x00271B, 0x004F66, 0x004F67, 0x004F61, 0x004F47, 0x004ED9, 0x004ED7,
0x004ECD, 0x004ED2, 0x004ED3, 0x004ED0, 0x004ED1,
};
static const uint8_t huff_iid_df0_bits[] = {
17, 17, 17, 17, 16, 15, 13, 10, 9, 7, 6, 5, 4, 3, 1, 3, 4, 5,
6, 6, 8, 11, 13, 14, 14, 15, 17, 18, 18,
};
static const uint32_t huff_iid_df0_codes[] = {
0x01FFFB, 0x01FFFC, 0x01FFFD, 0x01FFFA, 0x00FFFC, 0x007FFC, 0x001FFD,
0x0003FE, 0x0001FE, 0x00007E, 0x00003C, 0x00001D, 0x00000D, 0x000005,
0x000000, 0x000004, 0x00000C, 0x00001C, 0x00003D, 0x00003E, 0x0000FE,
0x0007FE, 0x001FFC, 0x003FFC, 0x003FFD, 0x007FFD, 0x01FFFE, 0x03FFFE,
0x03FFFF,
};
static const uint8_t huff_iid_dt0_bits[] = {
19, 19, 19, 20, 20, 20, 17, 15, 12, 10, 8, 6, 4, 2, 1, 3, 5, 7,
9, 11, 13, 14, 17, 19, 20, 20, 20, 20, 20,
};
static const uint32_t huff_iid_dt0_codes[] = {
0x07FFF9, 0x07FFFA, 0x07FFFB, 0x0FFFF8, 0x0FFFF9, 0x0FFFFA, 0x01FFFD,
0x007FFE, 0x000FFE, 0x0003FE, 0x0000FE, 0x00003E, 0x00000E, 0x000002,
0x000000, 0x000006, 0x00001E, 0x00007E, 0x0001FE, 0x0007FE, 0x001FFE,
0x003FFE, 0x01FFFC, 0x07FFF8, 0x0FFFFB, 0x0FFFFC, 0x0FFFFD, 0x0FFFFE,
0x0FFFFF,
};
static const uint8_t huff_icc_df_bits[] = {
14, 14, 12, 10, 7, 5, 3, 1, 2, 4, 6, 8, 9, 11, 13,
};
static const uint16_t huff_icc_df_codes[] = {
0x3FFF, 0x3FFE, 0x0FFE, 0x03FE, 0x007E, 0x001E, 0x0006, 0x0000,
0x0002, 0x000E, 0x003E, 0x00FE, 0x01FE, 0x07FE, 0x1FFE,
};
static const uint8_t huff_icc_dt_bits[] = {
14, 13, 11, 9, 7, 5, 3, 1, 2, 4, 6, 8, 10, 12, 14,
};
static const uint16_t huff_icc_dt_codes[] = {
0x3FFE, 0x1FFE, 0x07FE, 0x01FE, 0x007E, 0x001E, 0x0006, 0x0000,
0x0002, 0x000E, 0x003E, 0x00FE, 0x03FE, 0x0FFE, 0x3FFF,
};
static const uint8_t huff_ipd_df_bits[] = {
1, 3, 4, 4, 4, 4, 4, 4,
};
static const uint8_t huff_ipd_df_codes[] = {
0x01, 0x00, 0x06, 0x04, 0x02, 0x03, 0x05, 0x07,
};
static const uint8_t huff_ipd_dt_bits[] = {
1, 3, 4, 5, 5, 4, 4, 3,
};
static const uint8_t huff_ipd_dt_codes[] = {
0x01, 0x02, 0x02, 0x03, 0x02, 0x00, 0x03, 0x03,
};
static const uint8_t huff_opd_df_bits[] = {
1, 3, 4, 4, 5, 5, 4, 3,
};
static const uint8_t huff_opd_df_codes[] = {
0x01, 0x01, 0x06, 0x04, 0x0F, 0x0E, 0x05, 0x00,
};
static const uint8_t huff_opd_dt_bits[] = {
1, 3, 4, 5, 5, 4, 4, 3,
};
static const uint8_t huff_opd_dt_codes[] = {
0x01, 0x02, 0x01, 0x07, 0x06, 0x00, 0x02, 0x03,
};
static const int8_t huff_offset[] = {
30, 30,
14, 14,
7, 7,
0, 0,
0, 0,
};
///Table 8.48
static const int8_t k_to_i_20[] = {
1, 0, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 14, 15,
15, 15, 16, 16, 16, 16, 17, 17, 17, 17, 17, 18, 18, 18, 18, 18, 18, 18, 18,
18, 18, 18, 18, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19,
19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19
};
///Table 8.49
static const int8_t k_to_i_34[] = {
0, 1, 2, 3, 4, 5, 6, 6, 7, 2, 1, 0, 10, 10, 4, 5, 6, 7, 8,
9, 10, 11, 12, 9, 14, 11, 12, 13, 14, 15, 16, 13, 16, 17, 18, 19, 20, 21,
22, 22, 23, 23, 24, 24, 25, 25, 26, 26, 27, 27, 27, 28, 28, 28, 29, 29, 29,
30, 30, 30, 31, 31, 31, 31, 32, 32, 32, 32, 33, 33, 33, 33, 33, 33, 33, 33,
33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33
};
static const float g1_Q2[] = {
0.0f, 0.01899487526049f, 0.0f, -0.07293139167538f,
0.0f, 0.30596630545168f, 0.5f
};

159
src/add-ons/media/plugins/ffmpeg/libavcodec/aacsbr.c
View File

@ -31,6 +31,7 @@
#include "aacsbr.h"
#include "aacsbrdata.h"
#include "fft.h"
#include "aacps.h"
#include <stdint.h>
#include <float.h>
@ -71,9 +72,6 @@ enum {
static VLC vlc_sbr[10];
static const int8_t vlc_sbr_lav[10] =
{ 60, 60, 24, 24, 31, 31, 12, 12, 31, 12 };
static DECLARE_ALIGNED(16, float, analysis_cos_pre)[64];
static DECLARE_ALIGNED(16, float, analysis_sin_pre)[64];
static DECLARE_ALIGNED(16, float, analysis_cossin_post)[32][2];
static const DECLARE_ALIGNED(16, float, zero64)[64];
#define SBR_INIT_VLC_STATIC(num, size) \
@ -87,7 +85,7 @@ static const DECLARE_ALIGNED(16, float, zero64)[64];
av_cold void ff_aac_sbr_init(void)
{
int n, k;
int n;
static const struct {
const void *sbr_codes, *sbr_bits;
const unsigned int table_size, elem_size;
@ -116,16 +114,6 @@ av_cold void ff_aac_sbr_init(void)
SBR_INIT_VLC_STATIC(8, 592);
SBR_INIT_VLC_STATIC(9, 512);
for (n = 0; n < 64; n++) {
float pre = M_PI * n / 64;
analysis_cos_pre[n] = cosf(pre);
analysis_sin_pre[n] = sinf(pre);
}
for (k = 0; k < 32; k++) {
float post = M_PI * (k + 0.5) / 128;
analysis_cossin_post[k][0] = 4.0 * cosf(post);
analysis_cossin_post[k][1] = -4.0 * sinf(post);
}
for (n = 1; n < 320; n++)
sbr_qmf_window_us[320 + n] = sbr_qmf_window_us[320 - n];
sbr_qmf_window_us[384] = -sbr_qmf_window_us[384];
@ -133,6 +121,8 @@ av_cold void ff_aac_sbr_init(void)
for (n = 0; n < 320; n++)
sbr_qmf_window_ds[n] = sbr_qmf_window_us[2*n];
ff_ps_init();
}
av_cold void ff_aac_sbr_ctx_init(SpectralBandReplication *sbr)
@ -142,13 +132,14 @@ av_cold void ff_aac_sbr_ctx_init(SpectralBandReplication *sbr)
sbr->data[0].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
sbr->data[1].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
ff_mdct_init(&sbr->mdct, 7, 1, 1.0/64);
ff_rdft_init(&sbr->rdft, 6, IDFT_R2C);
ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0);
ff_ps_ctx_init(&sbr->ps);
}
av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
{
ff_mdct_end(&sbr->mdct);
ff_rdft_end(&sbr->rdft);
ff_mdct_end(&sbr->mdct_ana);
}
static int qsort_comparison_function_int16(const void *a, const void *b)
@ -293,15 +284,15 @@ static void make_bands(int16_t* bands, int start, int stop, int num_bands)
bands[num_bands-1] = stop - previous;
}
static int check_n_master(AVCodecContext *avccontext, int n_master, int bs_xover_band)
static int check_n_master(AVCodecContext *avctx, int n_master, int bs_xover_band)
{
// Requirements (14496-3 sp04 p205)
if (n_master <= 0) {
av_log(avccontext, AV_LOG_ERROR, "Invalid n_master: %d\n", n_master);
av_log(avctx, AV_LOG_ERROR, "Invalid n_master: %d\n", n_master);
return -1;
}
if (bs_xover_band >= n_master) {
av_log(avccontext, AV_LOG_ERROR,
av_log(avctx, AV_LOG_ERROR,
"Invalid bitstream, crossover band index beyond array bounds: %d\n",
bs_xover_band);
return -1;
@ -349,7 +340,7 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
sbr_offset_ptr = sbr_offset[5];
break;
default:
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"Unsupported sample rate for SBR: %d\n", sbr->sample_rate);
return -1;
}
@ -367,7 +358,7 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
} else if (spectrum->bs_stop_freq == 15) {
sbr->k[2] = 3*sbr->k[0];
} else {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"Invalid bs_stop_freq: %d\n", spectrum->bs_stop_freq);
return -1;
}
@ -382,18 +373,17 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
max_qmf_subbands = 32;
if (sbr->k[2] - sbr->k[0] > max_qmf_subbands) {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"Invalid bitstream, too many QMF subbands: %d\n", sbr->k[2] - sbr->k[0]);
return -1;
}
if (!spectrum->bs_freq_scale) {
unsigned int dk;
int k2diff;
int dk, k2diff;
dk = spectrum->bs_alter_scale + 1;
sbr->n_master = ((sbr->k[2] - sbr->k[0] + (dk&2)) >> dk) << 1;
if (check_n_master(ac->avccontext, sbr->n_master, sbr->spectrum_params.bs_xover_band))
if (check_n_master(ac->avctx, sbr->n_master, sbr->spectrum_params.bs_xover_band))
return -1;
for (k = 1; k <= sbr->n_master; k++)
@ -402,7 +392,7 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
k2diff = sbr->k[2] - sbr->k[0] - sbr->n_master * dk;
if (k2diff < 0) {
sbr->f_master[1]--;
sbr->f_master[2]-= (k2diff < 1);
sbr->f_master[2]-= (k2diff < -1);
} else if (k2diff) {
sbr->f_master[sbr->n_master]++;
}
@ -428,7 +418,7 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
num_bands_0 = lrintf(half_bands * log2f(sbr->k[1] / (float)sbr->k[0])) * 2;
if (num_bands_0 <= 0) { // Requirements (14496-3 sp04 p205)
av_log(ac->avccontext, AV_LOG_ERROR, "Invalid num_bands_0: %d\n", num_bands_0);
av_log(ac->avctx, AV_LOG_ERROR, "Invalid num_bands_0: %d\n", num_bands_0);
return -1;
}
@ -442,7 +432,7 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
vk0[0] = sbr->k[0];
for (k = 1; k <= num_bands_0; k++) {
if (vk0[k] <= 0) { // Requirements (14496-3 sp04 p205)
av_log(ac->avccontext, AV_LOG_ERROR, "Invalid vDk0[%d]: %d\n", k, vk0[k]);
av_log(ac->avctx, AV_LOG_ERROR, "Invalid vDk0[%d]: %d\n", k, vk0[k]);
return -1;
}
vk0[k] += vk0[k-1];
@ -472,14 +462,14 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
vk1[0] = sbr->k[1];
for (k = 1; k <= num_bands_1; k++) {
if (vk1[k] <= 0) { // Requirements (14496-3 sp04 p205)
av_log(ac->avccontext, AV_LOG_ERROR, "Invalid vDk1[%d]: %d\n", k, vk1[k]);
av_log(ac->avctx, AV_LOG_ERROR, "Invalid vDk1[%d]: %d\n", k, vk1[k]);
return -1;
}
vk1[k] += vk1[k-1];
}
sbr->n_master = num_bands_0 + num_bands_1;
if (check_n_master(ac->avccontext, sbr->n_master, sbr->spectrum_params.bs_xover_band))
if (check_n_master(ac->avctx, sbr->n_master, sbr->spectrum_params.bs_xover_band))
return -1;
memcpy(&sbr->f_master[0], vk0,
(num_bands_0 + 1) * sizeof(sbr->f_master[0]));
@ -488,7 +478,7 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
} else {
sbr->n_master = num_bands_0;
if (check_n_master(ac->avccontext, sbr->n_master, sbr->spectrum_params.bs_xover_band))
if (check_n_master(ac->avctx, sbr->n_master, sbr->spectrum_params.bs_xover_band))
return -1;
memcpy(sbr->f_master, vk0, (num_bands_0 + 1) * sizeof(sbr->f_master[0]));
}
@ -524,7 +514,7 @@ static int sbr_hf_calc_npatches(AACContext *ac, SpectralBandReplication *sbr)
// illegal however the Coding Technologies decoder check stream has a final
// count of 6 patches
if (sbr->num_patches > 5) {
av_log(ac->avccontext, AV_LOG_ERROR, "Too many patches: %d\n", sbr->num_patches);
av_log(ac->avctx, AV_LOG_ERROR, "Too many patches: %d\n", sbr->num_patches);
return -1;
}
@ -563,12 +553,12 @@ static int sbr_make_f_derived(AACContext *ac, SpectralBandReplication *sbr)
// Requirements (14496-3 sp04 p205)
if (sbr->kx[1] + sbr->m[1] > 64) {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"Stop frequency border too high: %d\n", sbr->kx[1] + sbr->m[1]);
return -1;
}
if (sbr->kx[1] > 32) {
av_log(ac->avccontext, AV_LOG_ERROR, "Start frequency border too high: %d\n", sbr->kx[1]);
av_log(ac->avctx, AV_LOG_ERROR, "Start frequency border too high: %d\n", sbr->kx[1]);
return -1;
}
@ -580,7 +570,7 @@ static int sbr_make_f_derived(AACContext *ac, SpectralBandReplication *sbr)
sbr->n_q = FFMAX(1, lrintf(sbr->spectrum_params.bs_noise_bands *
log2f(sbr->k[2] / (float)sbr->kx[1]))); // 0 <= bs_noise_bands <= 3
if (sbr->n_q > 5) {
av_log(ac->avccontext, AV_LOG_ERROR, "Too many noise floor scale factors: %d\n", sbr->n_q);
av_log(ac->avctx, AV_LOG_ERROR, "Too many noise floor scale factors: %d\n", sbr->n_q);
return -1;
}
@ -638,7 +628,7 @@ static int read_sbr_grid(AACContext *ac, SpectralBandReplication *sbr,
ch_data->bs_amp_res = 0;
if (ch_data->bs_num_env > 4) {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"Invalid bitstream, too many SBR envelopes in FIXFIX type SBR frame: %d\n",
ch_data->bs_num_env);
return -1;
@ -693,7 +683,7 @@ static int read_sbr_grid(AACContext *ac, SpectralBandReplication *sbr,
ch_data->bs_num_env = num_rel_lead + num_rel_trail + 1;
if (ch_data->bs_num_env > 5) {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"Invalid bitstream, too many SBR envelopes in VARVAR type SBR frame: %d\n",
ch_data->bs_num_env);
return -1;
@ -714,7 +704,7 @@ static int read_sbr_grid(AACContext *ac, SpectralBandReplication *sbr,
}
if (bs_pointer > ch_data->bs_num_env + 1) {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"Invalid bitstream, bs_pointer points to a middle noise border outside the time borders table: %d\n",
bs_pointer);
return -1;
@ -722,7 +712,7 @@ static int read_sbr_grid(AACContext *ac, SpectralBandReplication *sbr,
for (i = 1; i <= ch_data->bs_num_env; i++) {
if (ch_data->t_env[i-1] > ch_data->t_env[i]) {
av_log(ac->avccontext, AV_LOG_ERROR, "Non monotone time borders\n");
av_log(ac->avctx, AV_LOG_ERROR, "Non monotone time borders\n");
return -1;
}
}
@ -903,25 +893,24 @@ static void read_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
GetBitContext *gb,
int bs_extension_id, int *num_bits_left)
{
//TODO - implement ps_data for parametric stereo parsing
switch (bs_extension_id) {
case EXTENSION_ID_PS:
if (!ac->m4ac.ps) {
av_log(ac->avccontext, AV_LOG_ERROR, "Parametric Stereo signaled to be not-present but was found in the bitstream.\n");
av_log(ac->avctx, AV_LOG_ERROR, "Parametric Stereo signaled to be not-present but was found in the bitstream.\n");
skip_bits_long(gb, *num_bits_left); // bs_fill_bits
*num_bits_left = 0;
} else {
#if 0
*num_bits_left -= ff_ps_data(gb, ps);
#if 1
*num_bits_left -= ff_ps_read_data(ac->avctx, gb, &sbr->ps, *num_bits_left);
#else
av_log_missing_feature(ac->avccontext, "Parametric Stereo is", 0);
av_log_missing_feature(ac->avctx, "Parametric Stereo is", 0);
skip_bits_long(gb, *num_bits_left); // bs_fill_bits
*num_bits_left = 0;
#endif
}
break;
default:
av_log_missing_feature(ac->avccontext, "Reserved SBR extensions are", 1);
av_log_missing_feature(ac->avctx, "Reserved SBR extensions are", 1);
skip_bits_long(gb, *num_bits_left); // bs_fill_bits
*num_bits_left = 0;
break;
@ -1006,7 +995,7 @@ static unsigned int read_sbr_data(AACContext *ac, SpectralBandReplication *sbr,
return get_bits_count(gb) - cnt;
}
} else {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"Invalid bitstream - cannot apply SBR to element type %d\n", id_aac);
sbr->start = 0;
return get_bits_count(gb) - cnt;
@ -1021,6 +1010,11 @@ static unsigned int read_sbr_data(AACContext *ac, SpectralBandReplication *sbr,
num_bits_left -= 2;
read_sbr_extension(ac, sbr, gb, get_bits(gb, 2), &num_bits_left); // bs_extension_id
}
if (num_bits_left < 0) {
av_log(ac->avctx, AV_LOG_ERROR, "SBR Extension over read.\n");
}
if (num_bits_left > 0)
skip_bits(gb, num_bits_left);
}
return get_bits_count(gb) - cnt;
@ -1033,7 +1027,7 @@ static void sbr_reset(AACContext *ac, SpectralBandReplication *sbr)
if (err >= 0)
err = sbr_make_f_derived(ac, sbr);
if (err < 0) {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"SBR reset failed. Switching SBR to pure upsampling mode.\n");
sbr->start = 0;
}
@ -1085,7 +1079,7 @@ int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
bytes_read = ((num_sbr_bits + num_align_bits + 4) >> 3);
if (bytes_read > cnt) {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"Expected to read %d SBR bytes actually read %d.\n", cnt, bytes_read);
}
return cnt;
@ -1139,7 +1133,7 @@ static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
* @param x pointer to the beginning of the first sample window
* @param W array of complex-valued samples split into subbands
*/
static void sbr_qmf_analysis(DSPContext *dsp, RDFTContext *rdft, const float *in, float *x,
static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct, const float *in, float *x,
float z[320], float W[2][32][32][2],
float scale)
{
@ -1152,23 +1146,23 @@ static void sbr_qmf_analysis(DSPContext *dsp, RDFTContext *rdft, const float *in
memcpy(x+288, in, 1024*sizeof(*x));
for (i = 0; i < 32; i++) { // numTimeSlots*RATE = 16*2 as 960 sample frames
// are not supported
float re, im;
dsp->vector_fmul_reverse(z, sbr_qmf_window_ds, x, 320);
for (k = 0; k < 64; k++) {
float f = z[k] + z[k + 64] + z[k + 128] + z[k + 192] + z[k + 256];
z[k] = f * analysis_cos_pre[k];
z[k+64] = f;
z[k] = f;
}
ff_rdft_calc(rdft, z);
re = z[0] * 0.5f;
im = 0.5f * dsp->scalarproduct_float(z+64, analysis_sin_pre, 64);
W[1][i][0][0] = re * analysis_cossin_post[0][0] - im * analysis_cossin_post[0][1];
W[1][i][0][1] = re * analysis_cossin_post[0][1] + im * analysis_cossin_post[0][0];
//Shuffle to IMDCT
z[64] = z[0];
for (k = 1; k < 32; k++) {
re = z[2*k ] - re;
im = z[2*k+1] - im;
W[1][i][k][0] = re * analysis_cossin_post[k][0] - im * analysis_cossin_post[k][1];
W[1][i][k][1] = re * analysis_cossin_post[k][1] + im * analysis_cossin_post[k][0];
z[64+2*k-1] = z[ k];
z[64+2*k ] = -z[64-k];
}
z[64+63] = z[32];
ff_imdct_half(mdct, z, z+64);
for (k = 0; k < 32; k++) {
W[1][i][k][0] = -z[63-k];
W[1][i][k][1] = z[k];
}
x += 32;
}
@ -1179,7 +1173,7 @@ static void sbr_qmf_analysis(DSPContext *dsp, RDFTContext *rdft, const float *in
* (14496-3 sp04 p206)
*/
static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
float *out, float X[2][32][64],
float *out, float X[2][38][64],
float mdct_buf[2][64],
float *v0, int *v_off, const unsigned int div,
float bias, float scale)
@ -1197,21 +1191,22 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
*v_off -= 128 >> div;
}
v = v0 + *v_off;
for (n = 1; n < 64 >> div; n+=2) {
X[1][i][n] = -X[1][i][n];
}
if (div) {
memset(X[0][i]+32, 0, 32*sizeof(float));
memset(X[1][i]+32, 0, 32*sizeof(float));
}
ff_imdct_half(mdct, mdct_buf[0], X[0][i]);
ff_imdct_half(mdct, mdct_buf[1], X[1][i]);
if (div) {
for (n = 0; n < 32; n++) {
v[ n] = -mdct_buf[0][63 - 2*n] + mdct_buf[1][2*n ];
v[ 63 - n] = mdct_buf[0][62 - 2*n] + mdct_buf[1][2*n + 1];
X[0][i][ n] = -X[0][i][n];
X[0][i][32+n] = X[1][i][31-n];
}
ff_imdct_half(mdct, mdct_buf[0], X[0][i]);
for (n = 0; n < 32; n++) {
v[ n] = mdct_buf[0][63 - 2*n];
v[63 - n] = -mdct_buf[0][62 - 2*n];
}
} else {
for (n = 1; n < 64; n+=2) {
X[1][i][n] = -X[1][i][n];
}
ff_imdct_half(mdct, mdct_buf[0], X[0][i]);
ff_imdct_half(mdct, mdct_buf[1], X[1][i]);
for (n = 0; n < 64; n++) {
v[ n] = -mdct_buf[0][63 - n] + mdct_buf[1][ n ];
v[127 - n] = mdct_buf[0][63 - n] + mdct_buf[1][ n ];
@ -1380,7 +1375,7 @@ static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
g--;
if (g < 0) {
av_log(ac->avccontext, AV_LOG_ERROR,
av_log(ac->avctx, AV_LOG_ERROR,
"ERROR : no subband found for frequency %d\n", k);
return -1;
}
@ -1414,7 +1409,7 @@ static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
}
/// Generate the subband filtered lowband
static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][32][64],
static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
const float X_low[32][40][2], const float Y[2][38][64][2],
int ch)
{
@ -1436,7 +1431,7 @@ static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][32][64],
}
for (k = 0; k < sbr->kx[1]; k++) {
for (i = i_Temp; i < i_f; i++) {
for (i = i_Temp; i < 38; i++) {
X[0][i][k] = X_low[k][i + ENVELOPE_ADJUSTMENT_OFFSET][0];
X[1][i][k] = X_low[k][i + ENVELOPE_ADJUSTMENT_OFFSET][1];
}
@ -1730,7 +1725,7 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
}
for (ch = 0; ch < nch; ch++) {
/* decode channel */
sbr_qmf_analysis(&ac->dsp, &sbr->rdft, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
sbr_qmf_analysis(&ac->dsp, &sbr->mdct_ana, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
(float*)sbr->qmf_filter_scratch,
sbr->data[ch].W, 1/(-1024 * ac->sf_scale));
sbr_lf_gen(ac, sbr, sbr->X_low, sbr->data[ch].W);
@ -1752,6 +1747,16 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
/* synthesis */
sbr_x_gen(sbr, sbr->X[ch], sbr->X_low, sbr->data[ch].Y, ch);
}
if (ac->m4ac.ps == 1) {
if (sbr->ps.start) {
ff_ps_apply(ac->avctx, &sbr->ps, sbr->X[0], sbr->X[1], sbr->kx[1] + sbr->m[1]);
} else {
memcpy(sbr->X[1], sbr->X[0], sizeof(sbr->X[0]));
}
nch = 2;
}
sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, L, sbr->X[0], sbr->qmf_filter_scratch,
sbr->data[0].synthesis_filterbank_samples,
&sbr->data[0].synthesis_filterbank_samples_offset,

127
src/add-ons/media/plugins/ffmpeg/libavcodec/aactab.c
View File

@ -29,6 +29,7 @@
#include "libavutil/mem.h"
#include "aac.h"
#include "aac_tablegen.h"
#include <stdint.h>
@ -1204,129 +1205,3 @@ const uint8_t ff_tns_max_bands_128[] = {
9, 9, 10, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14
};
// @}
#if CONFIG_HARDCODED_TABLES
/**
* Table of pow(2, (i - 200)/4.) used for different purposes depending on the
* range of indices to the table:
* [ 0, 255] scale factor decoding when using C dsp.float_to_int16
* [60, 315] scale factor decoding when using SIMD dsp.float_to_int16
* [45, 300] intensity stereo position decoding mapped in reverse order i.e. 0->300, 1->299, ..., 254->46, 255->45
*/
const float ff_aac_pow2sf_tab[428] = {
8.88178420e-16, 1.05622810e-15, 1.25607397e-15, 1.49373210e-15,
1.77635684e-15, 2.11245619e-15, 2.51214793e-15, 2.98746420e-15,
3.55271368e-15, 4.22491238e-15, 5.02429587e-15, 5.97492839e-15,
7.10542736e-15, 8.44982477e-15, 1.00485917e-14, 1.19498568e-14,
1.42108547e-14, 1.68996495e-14, 2.00971835e-14, 2.38997136e-14,
2.84217094e-14, 3.37992991e-14, 4.01943669e-14, 4.77994272e-14,
5.68434189e-14, 6.75985982e-14, 8.03887339e-14, 9.55988543e-14,
1.13686838e-13, 1.35197196e-13, 1.60777468e-13, 1.91197709e-13,
2.27373675e-13, 2.70394393e-13, 3.21554936e-13, 3.82395417e-13,
4.54747351e-13, 5.40788785e-13, 6.43109871e-13, 7.64790834e-13,
9.09494702e-13, 1.08157757e-12, 1.28621974e-12, 1.52958167e-12,
1.81898940e-12, 2.16315514e-12, 2.57243948e-12, 3.05916334e-12,
3.63797881e-12, 4.32631028e-12, 5.14487897e-12, 6.11832668e-12,
7.27595761e-12, 8.65262056e-12, 1.02897579e-11, 1.22366534e-11,
1.45519152e-11, 1.73052411e-11, 2.05795159e-11, 2.44733067e-11,
2.91038305e-11, 3.46104823e-11, 4.11590317e-11, 4.89466134e-11,
5.82076609e-11, 6.92209645e-11, 8.23180635e-11, 9.78932268e-11,
1.16415322e-10, 1.38441929e-10, 1.64636127e-10, 1.95786454e-10,
2.32830644e-10, 2.76883858e-10, 3.29272254e-10, 3.91572907e-10,
4.65661287e-10, 5.53767716e-10, 6.58544508e-10, 7.83145814e-10,
9.31322575e-10, 1.10753543e-09, 1.31708902e-09, 1.56629163e-09,
1.86264515e-09, 2.21507086e-09, 2.63417803e-09, 3.13258326e-09,
3.72529030e-09, 4.43014173e-09, 5.26835606e-09, 6.26516652e-09,
7.45058060e-09, 8.86028346e-09, 1.05367121e-08, 1.25303330e-08,
1.49011612e-08, 1.77205669e-08, 2.10734243e-08, 2.50606661e-08,
2.98023224e-08, 3.54411338e-08, 4.21468485e-08, 5.01213321e-08,
5.96046448e-08, 7.08822677e-08, 8.42936970e-08, 1.00242664e-07,
1.19209290e-07, 1.41764535e-07, 1.68587394e-07, 2.00485328e-07,
2.38418579e-07, 2.83529071e-07, 3.37174788e-07, 4.00970657e-07,
4.76837158e-07, 5.67058141e-07, 6.74349576e-07, 8.01941314e-07,
9.53674316e-07, 1.13411628e-06, 1.34869915e-06, 1.60388263e-06,
1.90734863e-06, 2.26823256e-06, 2.69739830e-06, 3.20776526e-06,
3.81469727e-06, 4.53646513e-06, 5.39479661e-06, 6.41553051e-06,
7.62939453e-06, 9.07293026e-06, 1.07895932e-05, 1.28310610e-05,
1.52587891e-05, 1.81458605e-05, 2.15791864e-05, 2.56621220e-05,
3.05175781e-05, 3.62917210e-05, 4.31583729e-05, 5.13242441e-05,
6.10351562e-05, 7.25834421e-05, 8.63167458e-05, 1.02648488e-04,
1.22070312e-04, 1.45166884e-04, 1.72633492e-04, 2.05296976e-04,
2.44140625e-04, 2.90333768e-04, 3.45266983e-04, 4.10593953e-04,
4.88281250e-04, 5.80667537e-04, 6.90533966e-04, 8.21187906e-04,
9.76562500e-04, 1.16133507e-03, 1.38106793e-03, 1.64237581e-03,
1.95312500e-03, 2.32267015e-03, 2.76213586e-03, 3.28475162e-03,
3.90625000e-03, 4.64534029e-03, 5.52427173e-03, 6.56950324e-03,
7.81250000e-03, 9.29068059e-03, 1.10485435e-02, 1.31390065e-02,
1.56250000e-02, 1.85813612e-02, 2.20970869e-02, 2.62780130e-02,
3.12500000e-02, 3.71627223e-02, 4.41941738e-02, 5.25560260e-02,
6.25000000e-02, 7.43254447e-02, 8.83883476e-02, 1.05112052e-01,
1.25000000e-01, 1.48650889e-01, 1.76776695e-01, 2.10224104e-01,
2.50000000e-01, 2.97301779e-01, 3.53553391e-01, 4.20448208e-01,
5.00000000e-01, 5.94603558e-01, 7.07106781e-01, 8.40896415e-01,
1.00000000e+00, 1.18920712e+00, 1.41421356e+00, 1.68179283e+00,
2.00000000e+00, 2.37841423e+00, 2.82842712e+00, 3.36358566e+00,
4.00000000e+00, 4.75682846e+00, 5.65685425e+00, 6.72717132e+00,
8.00000000e+00, 9.51365692e+00, 1.13137085e+01, 1.34543426e+01,
1.60000000e+01, 1.90273138e+01, 2.26274170e+01, 2.69086853e+01,
3.20000000e+01, 3.80546277e+01, 4.52548340e+01, 5.38173706e+01,
6.40000000e+01, 7.61092554e+01, 9.05096680e+01, 1.07634741e+02,
1.28000000e+02, 1.52218511e+02, 1.81019336e+02, 2.15269482e+02,
2.56000000e+02, 3.04437021e+02, 3.62038672e+02, 4.30538965e+02,
5.12000000e+02, 6.08874043e+02, 7.24077344e+02, 8.61077929e+02,
1.02400000e+03, 1.21774809e+03, 1.44815469e+03, 1.72215586e+03,
2.04800000e+03, 2.43549617e+03, 2.89630938e+03, 3.44431172e+03,
4.09600000e+03, 4.87099234e+03, 5.79261875e+03, 6.88862343e+03,
8.19200000e+03, 9.74198469e+03, 1.15852375e+04, 1.37772469e+04,
1.63840000e+04, 1.94839694e+04, 2.31704750e+04, 2.75544937e+04,
3.27680000e+04, 3.89679387e+04, 4.63409500e+04, 5.51089875e+04,
6.55360000e+04, 7.79358775e+04, 9.26819000e+04, 1.10217975e+05,
1.31072000e+05, 1.55871755e+05, 1.85363800e+05, 2.20435950e+05,
2.62144000e+05, 3.11743510e+05, 3.70727600e+05, 4.40871900e+05,
5.24288000e+05, 6.23487020e+05, 7.41455200e+05, 8.81743800e+05,
1.04857600e+06, 1.24697404e+06, 1.48291040e+06, 1.76348760e+06,
2.09715200e+06, 2.49394808e+06, 2.96582080e+06, 3.52697520e+06,
4.19430400e+06, 4.98789616e+06, 5.93164160e+06, 7.05395040e+06,
8.38860800e+06, 9.97579232e+06, 1.18632832e+07, 1.41079008e+07,
1.67772160e+07, 1.99515846e+07, 2.37265664e+07, 2.82158016e+07,
3.35544320e+07, 3.99031693e+07, 4.74531328e+07, 5.64316032e+07,
6.71088640e+07, 7.98063385e+07, 9.49062656e+07, 1.12863206e+08,
1.34217728e+08, 1.59612677e+08, 1.89812531e+08, 2.25726413e+08,
2.68435456e+08, 3.19225354e+08, 3.79625062e+08, 4.51452825e+08,
5.36870912e+08, 6.38450708e+08, 7.59250125e+08, 9.02905651e+08,
1.07374182e+09, 1.27690142e+09, 1.51850025e+09, 1.80581130e+09,
2.14748365e+09, 2.55380283e+09, 3.03700050e+09, 3.61162260e+09,
4.29496730e+09, 5.10760567e+09, 6.07400100e+09, 7.22324521e+09,
8.58993459e+09, 1.02152113e+10, 1.21480020e+10, 1.44464904e+10,
1.71798692e+10, 2.04304227e+10, 2.42960040e+10, 2.88929808e+10,
3.43597384e+10, 4.08608453e+10, 4.85920080e+10, 5.77859616e+10,
6.87194767e+10, 8.17216907e+10, 9.71840160e+10, 1.15571923e+11,
1.37438953e+11, 1.63443381e+11, 1.94368032e+11, 2.31143847e+11,
2.74877907e+11, 3.26886763e+11, 3.88736064e+11, 4.62287693e+11,
5.49755814e+11, 6.53773525e+11, 7.77472128e+11, 9.24575386e+11,
1.09951163e+12, 1.30754705e+12, 1.55494426e+12, 1.84915077e+12,
2.19902326e+12, 2.61509410e+12, 3.10988851e+12, 3.69830155e+12,
4.39804651e+12, 5.23018820e+12, 6.21977702e+12, 7.39660309e+12,
8.79609302e+12, 1.04603764e+13, 1.24395540e+13, 1.47932062e+13,
1.75921860e+13, 2.09207528e+13, 2.48791081e+13, 2.95864124e+13,
3.51843721e+13, 4.18415056e+13, 4.97582162e+13, 5.91728247e+13,
7.03687442e+13, 8.36830112e+13, 9.95164324e+13, 1.18345649e+14,
1.40737488e+14, 1.67366022e+14, 1.99032865e+14, 2.36691299e+14,
2.81474977e+14, 3.34732045e+14, 3.98065730e+14, 4.73382598e+14,
5.62949953e+14, 6.69464090e+14, 7.96131459e+14, 9.46765196e+14,
1.12589991e+15, 1.33892818e+15, 1.59226292e+15, 1.89353039e+15,
2.25179981e+15, 2.67785636e+15, 3.18452584e+15, 3.78706078e+15,
4.50359963e+15, 5.35571272e+15, 6.36905167e+15, 7.57412156e+15,
9.00719925e+15, 1.07114254e+16, 1.27381033e+16, 1.51482431e+16,
1.80143985e+16, 2.14228509e+16, 2.54762067e+16, 3.02964863e+16,
3.60287970e+16, 4.28457018e+16, 5.09524134e+16, 6.05929725e+16,
7.20575940e+16, 8.56914035e+16, 1.01904827e+17, 1.21185945e+17,
};
#else
float ff_aac_pow2sf_tab[428];
#endif /* CONFIG_HARDCODED_TABLES */

7
src/add-ons/media/plugins/ffmpeg/libavcodec/aactab.h
View File

@ -32,6 +32,7 @@
#include "libavutil/mem.h"
#include "aac.h"
#include "aac_tablegen_decl.h"
#include <stdint.h>
@ -73,10 +74,4 @@ extern const uint16_t * const ff_swb_offset_128 [13];
extern const uint8_t ff_tns_max_bands_1024[13];
extern const uint8_t ff_tns_max_bands_128 [13];
#if CONFIG_HARDCODED_TABLES
extern const float ff_aac_pow2sf_tab[428];
#else
extern float ff_aac_pow2sf_tab[428];
#endif /* CONFIG_HARDCODED_TABLES */
#endif /* AVCODEC_AACTAB_H */

2
src/add-ons/media/plugins/ffmpeg/libavcodec/ac3_parser.c
View File

@ -181,7 +181,7 @@ static int ac3_sync(uint64_t state, AACAC3ParseContext *hdr_info,
hdr_info->samples = hdr.num_blocks * 256;
if(hdr.bitstream_id>10)
hdr_info->codec_id = CODEC_ID_EAC3;
else
else if (hdr_info->codec_id == CODEC_ID_NONE)
hdr_info->codec_id = CODEC_ID_AC3;
*need_next_header = (hdr.frame_type != EAC3_FRAME_TYPE_AC3_CONVERT);

21
src/add-ons/media/plugins/ffmpeg/libavcodec/flicvideo.c
View File

@ -159,7 +159,7 @@ static int flic_decode_frame_8BPP(AVCodecContext *avctx,
int pixel_skip;
int pixel_countdown;
unsigned char *pixels;
int pixel_limit;
unsigned int pixel_limit;
s->frame.reference = 1;
s->frame.buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE;
@ -253,10 +253,13 @@ static int flic_decode_frame_8BPP(AVCodecContext *avctx,
av_log(avctx, AV_LOG_ERROR, "Undefined opcode (%x) in DELTA_FLI\n", line_packets);
} else if ((line_packets & 0xC000) == 0x8000) {
// "last byte" opcode
pixels[y_ptr + s->frame.linesize[0] - 1] = line_packets & 0xff;
pixel_ptr= y_ptr + s->frame.linesize[0] - 1;
CHECK_PIXEL_PTR(0);
pixels[pixel_ptr] = line_packets & 0xff;
} else {
compressed_lines--;
pixel_ptr = y_ptr;
CHECK_PIXEL_PTR(0);
pixel_countdown = s->avctx->width;
for (i = 0; i < line_packets; i++) {
/* account for the skip bytes */
@ -268,7 +271,7 @@ static int flic_decode_frame_8BPP(AVCodecContext *avctx,
byte_run = -byte_run;
palette_idx1 = buf[stream_ptr++];
palette_idx2 = buf[stream_ptr++];
CHECK_PIXEL_PTR(byte_run);
CHECK_PIXEL_PTR(byte_run * 2);
for (j = 0; j < byte_run; j++, pixel_countdown -= 2) {
pixels[pixel_ptr++] = palette_idx1;
pixels[pixel_ptr++] = palette_idx2;
@ -298,6 +301,7 @@ static int flic_decode_frame_8BPP(AVCodecContext *avctx,
stream_ptr += 2;
while (compressed_lines > 0) {
pixel_ptr = y_ptr;
CHECK_PIXEL_PTR(0);
pixel_countdown = s->avctx->width;
line_packets = buf[stream_ptr++];
if (line_packets > 0) {
@ -453,7 +457,7 @@ static int flic_decode_frame_15_16BPP(AVCodecContext *avctx,
int pixel_countdown;
unsigned char *pixels;
int pixel;
int pixel_limit;
unsigned int pixel_limit;
s->frame.reference = 1;
s->frame.buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE;
@ -503,6 +507,7 @@ static int flic_decode_frame_15_16BPP(AVCodecContext *avctx,
} else {
compressed_lines--;
pixel_ptr = y_ptr;
CHECK_PIXEL_PTR(0);
pixel_countdown = s->avctx->width;
for (i = 0; i < line_packets; i++) {
/* account for the skip bytes */
@ -514,13 +519,13 @@ static int flic_decode_frame_15_16BPP(AVCodecContext *avctx,
byte_run = -byte_run;
pixel = AV_RL16(&buf[stream_ptr]);
stream_ptr += 2;
CHECK_PIXEL_PTR(byte_run);
CHECK_PIXEL_PTR(2 * byte_run);
for (j = 0; j < byte_run; j++, pixel_countdown -= 2) {
*((signed short*)(&pixels[pixel_ptr])) = pixel;
pixel_ptr += 2;
}
} else {
CHECK_PIXEL_PTR(byte_run);
CHECK_PIXEL_PTR(2 * byte_run);
for (j = 0; j < byte_run; j++, pixel_countdown--) {
*((signed short*)(&pixels[pixel_ptr])) = AV_RL16(&buf[stream_ptr]);
stream_ptr += 2;
@ -611,7 +616,7 @@ static int flic_decode_frame_15_16BPP(AVCodecContext *avctx,
if (byte_run > 0) {
pixel = AV_RL16(&buf[stream_ptr]);
stream_ptr += 2;
CHECK_PIXEL_PTR(byte_run);
CHECK_PIXEL_PTR(2 * byte_run);
for (j = 0; j < byte_run; j++) {
*((signed short*)(&pixels[pixel_ptr])) = pixel;
pixel_ptr += 2;
@ -622,7 +627,7 @@ static int flic_decode_frame_15_16BPP(AVCodecContext *avctx,
}
} else { /* copy pixels if byte_run < 0 */
byte_run = -byte_run;
CHECK_PIXEL_PTR(byte_run);
CHECK_PIXEL_PTR(2 * byte_run);
for (j = 0; j < byte_run; j++) {
*((signed short*)(&pixels[pixel_ptr])) = AV_RL16(&buf[stream_ptr]);
stream_ptr += 2;

3
src/add-ons/media/plugins/ffmpeg/libavcodec/options.c
View File

@ -488,9 +488,6 @@ int avcodec_copy_context(AVCodecContext *dest, const AVCodecContext *src)
dest->slice_offset = NULL;
dest->internal_buffer = NULL;
dest->hwaccel = NULL;
dest->execute = NULL;
dest->execute2 = NULL;
dest->reget_buffer = NULL;
dest->thread_opaque = NULL;
/* reallocate values that should be allocated separately */

6
src/add-ons/media/plugins/ffmpeg/libavcodec/sbr.h
View File

@ -31,6 +31,7 @@
#include <stdint.h>
#include "fft.h"
#include "aacps.h"
/**
* Spectral Band Replication header - spectrum parameters that invoke a reset if they differ from the previous header.
@ -133,6 +134,7 @@ typedef struct {
///The number of frequency bands in f_master
unsigned n_master;
SBRData data[2];
PSContext ps;
///N_Low and N_High respectively, the number of frequency bands for low and high resolution
unsigned n[2];
///Number of noise floor bands
@ -157,7 +159,7 @@ typedef struct {
///QMF output of the HF generator
float X_high[64][40][2];
///QMF values of the reconstructed signal
DECLARE_ALIGNED(16, float, X)[2][2][32][64];
DECLARE_ALIGNED(16, float, X)[2][2][38][64];
///Zeroth coefficient used to filter the subband signals
float alpha0[64][2];
///First coefficient used to filter the subband signals
@ -176,7 +178,7 @@ typedef struct {
float s_m[7][48];
float gain[7][48];
DECLARE_ALIGNED(16, float, qmf_filter_scratch)[5][64];
RDFTContext rdft;
FFTContext mdct_ana;
FFTContext mdct;
} SpectralBandReplication;

16
src/add-ons/media/plugins/ffmpeg/libavformat/avformat.h
View File

@ -869,8 +869,22 @@ void av_pkt_dump_log(void *avcl, int level, AVPacket *pkt, int dump_payload);
*/
void av_register_all(void);
/** codec tag <-> codec id */
/**
* Gets the CodecID for the given codec tag tag.
* If no codec id is found returns CODEC_ID_NONE.
*
* @param tags list of supported codec_id-codec_tag pairs, as stored
* in AVInputFormat.codec_tag and AVOutputFormat.codec_tag
*/
enum CodecID av_codec_get_id(const struct AVCodecTag * const *tags, unsigned int tag);
/**
* Gets the codec tag for the given codec id id.
* If no codec tag is found returns 0.
*
* @param tags list of supported codec_id-codec_tag pairs, as stored
* in AVInputFormat.codec_tag and AVOutputFormat.codec_tag
*/
unsigned int av_codec_get_tag(const struct AVCodecTag * const *tags, enum CodecID id);
/* media file input */

11
src/add-ons/media/plugins/ffmpeg/libavformat/aviobuf.c
View File

@ -27,6 +27,13 @@
#define IO_BUFFER_SIZE 32768
/**
* Do seeks within this distance ahead of the current buffer by skipping
* data instead of calling the protocol seek function, for seekable
* protocols.
*/
#define SHORT_SEEK_THRESHOLD 4096
static void fill_buffer(ByteIOContext *s);
#if LIBAVFORMAT_VERSION_MAJOR >= 53
static int url_resetbuf(ByteIOContext *s, int flags);
@ -152,7 +159,9 @@ int64_t url_fseek(ByteIOContext *s, int64_t offset, int whence)
offset1 >= 0 && offset1 <= (s->buf_end - s->buffer)) {
/* can do the seek inside the buffer */
s->buf_ptr = s->buffer + offset1;
} else if(s->is_streamed && !s->write_flag && offset1 >= 0 &&
} else if ((s->is_streamed ||
offset1 <= s->buf_end + SHORT_SEEK_THRESHOLD - s->buffer) &&
!s->write_flag && offset1 >= 0 &&
(whence != SEEK_END || force)) {
while(s->pos < offset && !s->eof_reached)
fill_buffer(s);

1
src/add-ons/media/plugins/ffmpeg/libavformat/riff.c
View File

@ -183,6 +183,7 @@ const AVCodecTag ff_codec_bmp_tags[] = {
{ CODEC_ID_VP6, MKTAG('V', 'P', '6', '2') },
{ CODEC_ID_VP6F, MKTAG('V', 'P', '6', 'F') },
{ CODEC_ID_VP6F, MKTAG('F', 'L', 'V', '4') },
{ CODEC_ID_VP8, MKTAG('V', 'P', '8', '0') },
{ CODEC_ID_ASV1, MKTAG('A', 'S', 'V', '1') },
{ CODEC_ID_ASV2, MKTAG('A', 'S', 'V', '2') },
{ CODEC_ID_VCR1, MKTAG('V', 'C', 'R', '1') },