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FFmpeg/libavcodec/aac.c

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/*
* AAC decoder
* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov 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
*/
/**
* @file aac.c
* AAC decoder
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*/
/*
* supported tools
*
* Support? Name
* N (code in SoC repo) gain control
* Y block switching
* Y window shapes - standard
* N window shapes - Low Delay
* Y filterbank - standard
* N (code in SoC repo) filterbank - Scalable Sample Rate
* Y Temporal Noise Shaping
* N (code in SoC repo) Long Term Prediction
* Y intensity stereo
* Y channel coupling
* N frequency domain prediction
* Y Perceptual Noise Substitution
* Y Mid/Side stereo
* N Scalable Inverse AAC Quantization
* N Frequency Selective Switch
* N upsampling filter
* Y quantization & coding - AAC
* N quantization & coding - TwinVQ
* N quantization & coding - BSAC
* N AAC Error Resilience tools
* N Error Resilience payload syntax
* N Error Protection tool
* N CELP
* N Silence Compression
* N HVXC
* N HVXC 4kbits/s VR
* N Structured Audio tools
* N Structured Audio Sample Bank Format
* N MIDI
* N Harmonic and Individual Lines plus Noise
* N Text-To-Speech Interface
* N (in progress) Spectral Band Replication
* Y (not in this code) Layer-1
* Y (not in this code) Layer-2
* Y (not in this code) Layer-3
* N SinuSoidal Coding (Transient, Sinusoid, Noise)
* N (planned) Parametric Stereo
* N Direct Stream Transfer
*
* Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
* - HE AAC v2 comprises LC AAC with Spectral Band Replication and
Parametric Stereo.
*/
#include "avcodec.h"
#include "bitstream.h"
#include "dsputil.h"
#include "aac.h"
#include "aactab.h"
#include "aacdectab.h"
#include "mpeg4audio.h"
#include <assert.h>
#include <errno.h>
#include <math.h>
#include <string.h>
static VLC vlc_scalefactors;
static VLC vlc_spectral[11];
/**
* Configure output channel order based on the current program configuration element.
*
* @param che_pos current channel position configuration
* @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int output_configure(AACContext *ac, enum ChannelPosition che_pos[4][MAX_ELEM_ID],
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]) {
AVCodecContext *avctx = ac->avccontext;
int i, type, channels = 0;
if(!memcmp(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])))
return 0; /* no change */
memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
/* Allocate or free elements depending on if they are in the
* current program configuration.
*
* Set up default 1:1 output mapping.
*
* For a 5.1 stream the output order will be:
* [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
*/
for(i = 0; i < MAX_ELEM_ID; i++) {
for(type = 0; type < 4; type++) {
if(che_pos[type][i]) {
if(!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement))))
return AVERROR(ENOMEM);
if(type != TYPE_CCE) {
ac->output_data[channels++] = ac->che[type][i]->ch[0].ret;
if(type == TYPE_CPE) {
ac->output_data[channels++] = ac->che[type][i]->ch[1].ret;
}
}
} else
av_freep(&ac->che[type][i]);
}
}
avctx->channels = channels;
return 0;
}
/**
* Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
*
* @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
* @param sce_map mono (Single Channel Element) map
* @param type speaker type/position for these channels
*/
static void decode_channel_map(enum ChannelPosition *cpe_map,
enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) {
while(n--) {
enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
map[get_bits(gb, 4)] = type;
}
}
/**
* Decode program configuration element; reference: table 4.2.
*
* @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
GetBitContext * gb) {
int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
skip_bits(gb, 2); // object_type
ac->m4ac.sampling_index = get_bits(gb, 4);
if(ac->m4ac.sampling_index > 11) {
av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
return -1;
}
ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index];
num_front = get_bits(gb, 4);
num_side = get_bits(gb, 4);
num_back = get_bits(gb, 4);
num_lfe = get_bits(gb, 2);
num_assoc_data = get_bits(gb, 3);
num_cc = get_bits(gb, 4);
if (get_bits1(gb))
skip_bits(gb, 4); // mono_mixdown_tag
if (get_bits1(gb))
skip_bits(gb, 4); // stereo_mixdown_tag
if (get_bits1(gb))
skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side );
decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back );
decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe );
skip_bits_long(gb, 4 * num_assoc_data);
decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc );
align_get_bits(gb);
/* comment field, first byte is length */
skip_bits_long(gb, 8 * get_bits(gb, 8));
return 0;
}
/**
* Set up channel positions based on a default channel configuration
* as specified in table 1.17.
*
* @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
int channel_config)
{
if(channel_config < 1 || channel_config > 7) {
av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
channel_config);
return -1;
}
/* default channel configurations:
*
* 1ch : front center (mono)
* 2ch : L + R (stereo)
* 3ch : front center + L + R
* 4ch : front center + L + R + back center
* 5ch : front center + L + R + back stereo
* 6ch : front center + L + R + back stereo + LFE
* 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
*/
if(channel_config != 2)
new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
if(channel_config > 1)
new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
if(channel_config == 4)
new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center
if(channel_config > 4)
new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
= AAC_CHANNEL_BACK; // back stereo
if(channel_config > 5)
new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE
if(channel_config == 7)
new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
return 0;
}
/**
* Decode GA "General Audio" specific configuration; reference: table 4.1.
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_ga_specific_config(AACContext * ac, GetBitContext * gb, int channel_config) {
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
int extension_flag, ret;
if(get_bits1(gb)) { // frameLengthFlag
av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1);
return -1;
}
if (get_bits1(gb)) // dependsOnCoreCoder
skip_bits(gb, 14); // coreCoderDelay
extension_flag = get_bits1(gb);
if(ac->m4ac.object_type == AOT_AAC_SCALABLE ||
ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
skip_bits(gb, 3); // layerNr
memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
if (channel_config == 0) {
skip_bits(gb, 4); // element_instance_tag
if((ret = decode_pce(ac, new_che_pos, gb)))
return ret;
} else {
if((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
return ret;
}
if((ret = output_configure(ac, ac->che_pos, new_che_pos)))
return ret;
if (extension_flag) {
switch (ac->m4ac.object_type) {
case AOT_ER_BSAC:
skip_bits(gb, 5); // numOfSubFrame
skip_bits(gb, 11); // layer_length
break;
case AOT_ER_AAC_LC:
case AOT_ER_AAC_LTP:
case AOT_ER_AAC_SCALABLE:
case AOT_ER_AAC_LD:
skip_bits(gb, 3); /* aacSectionDataResilienceFlag
* aacScalefactorDataResilienceFlag
* aacSpectralDataResilienceFlag
*/
break;
}
skip_bits1(gb); // extensionFlag3 (TBD in version 3)
}
return 0;
}
/**
* Decode audio specific configuration; reference: table 1.13.
*
* @param data pointer to AVCodecContext extradata
* @param data_size size of AVCCodecContext extradata
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) {
GetBitContext gb;
int i;
init_get_bits(&gb, data, data_size * 8);
if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
return -1;
if(ac->m4ac.sampling_index > 11) {
av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
return -1;
}
skip_bits_long(&gb, i);
switch (ac->m4ac.object_type) {
case AOT_AAC_LC:
if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
return -1;
break;
default:
av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
return -1;
}
return 0;
}
/**
* linear congruential pseudorandom number generator
*
* @param previous_val pointer to the current state of the generator
*
* @return Returns a 32-bit pseudorandom integer
*/
static av_always_inline int lcg_random(int previous_val) {
return previous_val * 1664525 + 1013904223;
}
static av_cold int aac_decode_init(AVCodecContext * avccontext) {
AACContext * ac = avccontext->priv_data;
int i;
ac->avccontext = avccontext;
if (avccontext->extradata_size <= 0 ||
decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
return -1;
avccontext->sample_fmt = SAMPLE_FMT_S16;
avccontext->sample_rate = ac->m4ac.sample_rate;
avccontext->frame_size = 1024;
AAC_INIT_VLC_STATIC( 0, 144);
AAC_INIT_VLC_STATIC( 1, 114);
AAC_INIT_VLC_STATIC( 2, 188);
AAC_INIT_VLC_STATIC( 3, 180);
AAC_INIT_VLC_STATIC( 4, 172);
AAC_INIT_VLC_STATIC( 5, 140);
AAC_INIT_VLC_STATIC( 6, 168);
AAC_INIT_VLC_STATIC( 7, 114);
AAC_INIT_VLC_STATIC( 8, 262);
AAC_INIT_VLC_STATIC( 9, 248);
AAC_INIT_VLC_STATIC(10, 384);
dsputil_init(&ac->dsp, avccontext);
ac->random_state = 0x1f2e3d4c;
// -1024 - Compensate wrong IMDCT method.
// 32768 - Required to scale values to the correct range for the bias method
// for float to int16 conversion.
if(ac->dsp.float_to_int16 == ff_float_to_int16_c) {
ac->add_bias = 385.0f;
ac->sf_scale = 1. / (-1024. * 32768.);
ac->sf_offset = 0;
} else {
ac->add_bias = 0.0f;
ac->sf_scale = 1. / -1024.;
ac->sf_offset = 60;
}
#ifndef CONFIG_HARDCODED_TABLES
for (i = 0; i < 316; i++)
ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
#endif /* CONFIG_HARDCODED_TABLES */
INIT_VLC_STATIC(&vlc_scalefactors, 7, sizeof(ff_aac_scalefactor_code)/sizeof(ff_aac_scalefactor_code[0]),
ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
352);
ff_mdct_init(&ac->mdct, 11, 1);
ff_mdct_init(&ac->mdct_small, 8, 1);
// window initialization
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
ff_sine_window_init(ff_sine_1024, 1024);
ff_sine_window_init(ff_sine_128, 128);
return 0;
}
/**
* Skip data_stream_element; reference: table 4.10.
*/
static void skip_data_stream_element(GetBitContext * gb) {
int byte_align = get_bits1(gb);
int count = get_bits(gb, 8);
if (count == 255)
count += get_bits(gb, 8);
if (byte_align)
align_get_bits(gb);
skip_bits_long(gb, 8 * count);
}
/**
* Decode Individual Channel Stream info; reference: table 4.6.
*
* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
*/
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");
memset(ics, 0, sizeof(IndividualChannelStream));
return -1;
}
ics->window_sequence[1] = ics->window_sequence[0];
ics->window_sequence[0] = get_bits(gb, 2);
ics->use_kb_window[1] = ics->use_kb_window[0];
ics->use_kb_window[0] = get_bits1(gb);
ics->num_window_groups = 1;
ics->group_len[0] = 1;
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
int i;
ics->max_sfb = get_bits(gb, 4);
for (i = 0; i < 7; i++) {
if (get_bits1(gb)) {
ics->group_len[ics->num_window_groups-1]++;
} else {
ics->num_window_groups++;
ics->group_len[ics->num_window_groups-1] = 1;
}
}
ics->num_windows = 8;
ics->swb_offset = swb_offset_128[ac->m4ac.sampling_index];
ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index];
ics->tns_max_bands = tns_max_bands_128[ac->m4ac.sampling_index];
} else {
ics->max_sfb = get_bits(gb, 6);
ics->num_windows = 1;
ics->swb_offset = swb_offset_1024[ac->m4ac.sampling_index];
ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index];
ics->tns_max_bands = tns_max_bands_1024[ac->m4ac.sampling_index];
if (get_bits1(gb)) {
av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1);
memset(ics, 0, sizeof(IndividualChannelStream));
return -1;
}
}
if(ics->max_sfb > ics->num_swb) {
av_log(ac->avccontext, 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));
return -1;
}
return 0;
}
/**
* Decode band types (section_data payload); reference: table 4.46.
*
* @param band_type array of the used band type
* @param band_type_run_end array of the last scalefactor band of a band type run
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_band_types(AACContext * ac, enum BandType band_type[120],
int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) {
int g, idx = 0;
const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
for (g = 0; g < ics->num_window_groups; g++) {
int k = 0;
while (k < ics->max_sfb) {
uint8_t sect_len = k;
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");
return -1;
}
while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1)
sect_len += sect_len_incr;
sect_len += sect_len_incr;
if (sect_len > ics->max_sfb) {
av_log(ac->avccontext, AV_LOG_ERROR,
"Number of bands (%d) exceeds limit (%d).\n",
sect_len, ics->max_sfb);
return -1;
}
for (; k < sect_len; k++) {
band_type [idx] = sect_band_type;
band_type_run_end[idx++] = sect_len;
}
}
}
return 0;
}
/**
* Decode scalefactors; reference: table 4.47.
*
* @param global_gain first scalefactor value as scalefactors are differentially coded
* @param band_type array of the used band type
* @param band_type_run_end array of the last scalefactor band of a band type run
* @param sf array of scalefactors or intensity stereo positions
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb,
unsigned int global_gain, IndividualChannelStream * ics,
enum BandType band_type[120], int band_type_run_end[120]) {
const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
int g, i, idx = 0;
int offset[3] = { global_gain, global_gain - 90, 100 };
int noise_flag = 1;
static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
for (g = 0; g < ics->num_window_groups; g++) {
for (i = 0; i < ics->max_sfb;) {
int run_end = band_type_run_end[idx];
if (band_type[idx] == ZERO_BT) {
for(; i < run_end; i++, idx++)
sf[idx] = 0.;
}else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
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,
"%s (%d) out of range.\n", sf_str[2], offset[2]);
return -1;
}
sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
}
}else if(band_type[idx] == NOISE_BT) {
for(; i < run_end; i++, idx++) {
if(noise_flag-- > 0)
offset[1] += get_bits(gb, 9) - 256;
else
offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
if(offset[1] > 255U) {
av_log(ac->avccontext, AV_LOG_ERROR,
"%s (%d) out of range.\n", sf_str[1], offset[1]);
return -1;
}
sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset];
}
}else {
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,
"%s (%d) out of range.\n", sf_str[0], offset[0]);
return -1;
}
sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
}
}
}
}
return 0;
}
/**
* Decode pulse data; reference: table 4.7.
*/
static void decode_pulses(Pulse * pulse, GetBitContext * gb, const uint16_t * swb_offset) {
int i;
pulse->num_pulse = get_bits(gb, 2) + 1;
pulse->pos[0] = get_bits(gb, 5) + swb_offset[get_bits(gb, 6)];
pulse->amp[0] = get_bits(gb, 4);
for (i = 1; i < pulse->num_pulse; i++) {
pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i-1];
pulse->amp[i] = get_bits(gb, 4);
}
}
/**
* Decode Temporal Noise Shaping data; reference: table 4.48.
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_tns(AACContext * ac, TemporalNoiseShaping * tns,
GetBitContext * gb, const IndividualChannelStream * ics) {
int w, filt, i, coef_len, coef_res, coef_compress;
const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
for (w = 0; w < ics->num_windows; w++) {
if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
coef_res = get_bits1(gb);
for (filt = 0; filt < tns->n_filt[w]; filt++) {
int tmp2_idx;
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.",
tns->order[w][filt], tns_max_order);
tns->order[w][filt] = 0;
return -1;
}
tns->direction[w][filt] = get_bits1(gb);
coef_compress = get_bits1(gb);
coef_len = coef_res + 3 - coef_compress;
tmp2_idx = 2*coef_compress + coef_res;
for (i = 0; i < tns->order[w][filt]; i++)
tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
}
}
}
return 0;
}
/**
* Decode Mid/Side data; reference: table 4.54.
*
* @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
* [1] mask is decoded from bitstream; [2] mask is all 1s;
* [3] reserved for scalable AAC
*/
static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb,
int ms_present) {
int idx;
if (ms_present == 1) {
for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
cpe->ms_mask[idx] = get_bits1(gb);
} else if (ms_present == 2) {
memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
}
}
/**
* Decode spectral data; reference: table 4.50.
* Dequantize and scale spectral data; reference: 4.6.3.3.
*
* @param coef array of dequantized, scaled spectral data
* @param sf array of scalefactors or intensity stereo positions
* @param pulse_present set if pulses are present
* @param pulse pointer to pulse data struct
* @param band_type array of the used band type
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_spectrum_and_dequant(AACContext * ac, float coef[1024], GetBitContext * gb, float sf[120],
int pulse_present, const Pulse * pulse, const IndividualChannelStream * ics, enum BandType band_type[120]) {
int i, k, g, idx = 0;
const int c = 1024/ics->num_windows;
const uint16_t * offsets = ics->swb_offset;
float *coef_base = coef;
for (g = 0; g < ics->num_windows; g++)
memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float)*(c - offsets[ics->max_sfb]));
for (g = 0; g < ics->num_window_groups; g++) {
for (i = 0; i < ics->max_sfb; i++, idx++) {
const int cur_band_type = band_type[idx];
const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4;
const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type);
int group;
if (cur_band_type == ZERO_BT) {
for (group = 0; group < ics->group_len[g]; group++) {
memset(coef + group * 128 + offsets[i], 0, (offsets[i+1] - offsets[i])*sizeof(float));
}
}else if (cur_band_type == NOISE_BT) {
const float scale = sf[idx] / ((offsets[i+1] - offsets[i]) * PNS_MEAN_ENERGY);
for (group = 0; group < ics->group_len[g]; group++) {
for (k = offsets[i]; k < offsets[i+1]; k++) {
ac->random_state = lcg_random(ac->random_state);
coef[group*128+k] = ac->random_state * scale;
}
}
}else if (cur_band_type != INTENSITY_BT2 && cur_band_type != INTENSITY_BT) {
for (group = 0; group < ics->group_len[g]; group++) {
for (k = offsets[i]; k < offsets[i+1]; k += dim) {
const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3);
const int coef_tmp_idx = (group << 7) + k;
const float *vq_ptr;
int j;
if(index >= ff_aac_spectral_sizes[cur_band_type - 1]) {
av_log(ac->avccontext, AV_LOG_ERROR,
"Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n",
cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]);
return -1;
}
vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim];
if (is_cb_unsigned) {
for (j = 0; j < dim; j++)
if (vq_ptr[j])
coef[coef_tmp_idx + j] = 1 - 2*(int)get_bits1(gb);
}else {
for (j = 0; j < dim; j++)
coef[coef_tmp_idx + j] = 1.0f;
}
if (cur_band_type == ESC_BT) {
for (j = 0; j < 2; j++) {
if (vq_ptr[j] == 64.0f) {
int n = 4;
/* The total length of escape_sequence must be < 22 bits according
to the specification (i.e. max is 11111111110xxxxxxxxxx). */
while (get_bits1(gb) && n < 15) n++;
if(n == 15) {
av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
return -1;
}
n = (1<<n) + get_bits(gb, n);
coef[coef_tmp_idx + j] *= cbrtf(fabsf(n)) * n;
}else
coef[coef_tmp_idx + j] *= vq_ptr[j];
}
}else
for (j = 0; j < dim; j++)
coef[coef_tmp_idx + j] *= vq_ptr[j];
for (j = 0; j < dim; j++)
coef[coef_tmp_idx + j] *= sf[idx];
}
}
}
}
coef += ics->group_len[g]<<7;
}
if (pulse_present) {
for(i = 0; i < pulse->num_pulse; i++){
float co = coef_base[ pulse->pos[i] ];
float ico = co / sqrtf(sqrtf(fabsf(co))) + pulse->amp[i];
coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico;
}
}
return 0;
}
/**
* Decode an individual_channel_stream payload; reference: table 4.44.
*
* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
* @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) {
Pulse pulse;
TemporalNoiseShaping * tns = &sce->tns;
IndividualChannelStream * ics = &sce->ics;
float * out = sce->coeffs;
int global_gain, pulse_present = 0;
/* This assignment is to silence a GCC warning about the variable being used
* uninitialized when in fact it always is.
*/
pulse.num_pulse = 0;
global_gain = get_bits(gb, 8);
if (!common_window && !scale_flag) {
if (decode_ics_info(ac, ics, gb, 0) < 0)
return -1;
}
if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
return -1;
if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
return -1;
pulse_present = 0;
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");
return -1;
}
decode_pulses(&pulse, gb, ics->swb_offset);
}
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);
return -1;
}
}
if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
return -1;
return 0;
}
/**
* Mid/Side stereo decoding; reference: 4.6.8.1.3.
*/
static void apply_mid_side_stereo(ChannelElement * cpe) {
const IndividualChannelStream * ics = &cpe->ch[0].ics;
float *ch0 = cpe->ch[0].coeffs;
float *ch1 = cpe->ch[1].coeffs;
int g, i, k, group, idx = 0;
const uint16_t * offsets = ics->swb_offset;
for (g = 0; g < ics->num_window_groups; g++) {
for (i = 0; i < ics->max_sfb; i++, idx++) {
if (cpe->ms_mask[idx] &&
cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
for (group = 0; group < ics->group_len[g]; group++) {
for (k = offsets[i]; k < offsets[i+1]; k++) {
float tmp = ch0[group*128 + k] - ch1[group*128 + k];
ch0[group*128 + k] += ch1[group*128 + k];
ch1[group*128 + k] = tmp;
}
}
}
}
ch0 += ics->group_len[g]*128;
ch1 += ics->group_len[g]*128;
}
}
/**
* intensity stereo decoding; reference: 4.6.8.2.3
*
* @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
* [1] mask is decoded from bitstream; [2] mask is all 1s;
* [3] reserved for scalable AAC
*/
static void apply_intensity_stereo(ChannelElement * cpe, int ms_present) {
const IndividualChannelStream * ics = &cpe->ch[1].ics;
SingleChannelElement * sce1 = &cpe->ch[1];
float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
const uint16_t * offsets = ics->swb_offset;
int g, group, i, k, idx = 0;
int c;
float scale;
for (g = 0; g < ics->num_window_groups; g++) {
for (i = 0; i < ics->max_sfb;) {
if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
const int bt_run_end = sce1->band_type_run_end[idx];
for (; i < bt_run_end; i++, idx++) {
c = -1 + 2 * (sce1->band_type[idx] - 14);
if (ms_present)
c *= 1 - 2 * cpe->ms_mask[idx];
scale = c * sce1->sf[idx];
for (group = 0; group < ics->group_len[g]; group++)
for (k = offsets[i]; k < offsets[i+1]; k++)
coef1[group*128 + k] = scale * coef0[group*128 + k];
}
} else {
int bt_run_end = sce1->band_type_run_end[idx];
idx += bt_run_end - i;
i = bt_run_end;
}
}
coef0 += ics->group_len[g]*128;
coef1 += ics->group_len[g]*128;
}
}
/**
* Decode a channel_pair_element; reference: table 4.4.
*
* @param elem_id Identifies the instance of a syntax element.
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) {
int i, ret, common_window, ms_present = 0;
ChannelElement * cpe;
cpe = ac->che[TYPE_CPE][elem_id];
common_window = get_bits1(gb);
if (common_window) {
if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
return -1;
i = cpe->ch[1].ics.use_kb_window[0];
cpe->ch[1].ics = cpe->ch[0].ics;
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");
return -1;
} else if(ms_present)
decode_mid_side_stereo(cpe, gb, ms_present);
}
if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
return ret;
if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
return ret;
if (common_window && ms_present)
apply_mid_side_stereo(cpe);
apply_intensity_stereo(cpe, ms_present);
return 0;
}
/**
* Decode coupling_channel_element; reference: table 4.8.
*
* @param elem_id Identifies the instance of a syntax element.
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_cce(AACContext * ac, GetBitContext * gb, ChannelElement * che) {
int num_gain = 0;
int c, g, sfb, ret, idx = 0;
int sign;
float scale;
SingleChannelElement * sce = &che->ch[0];
ChannelCoupling * coup = &che->coup;
coup->coupling_point = 2*get_bits1(gb);
coup->num_coupled = get_bits(gb, 3);
for (c = 0; c <= coup->num_coupled; c++) {
num_gain++;
coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
coup->id_select[c] = get_bits(gb, 4);
if (coup->type[c] == TYPE_CPE) {
coup->ch_select[c] = get_bits(gb, 2);
if (coup->ch_select[c] == 3)
num_gain++;
} else
coup->ch_select[c] = 1;
}
coup->coupling_point += get_bits1(gb);
if (coup->coupling_point == 2) {
av_log(ac->avccontext, AV_LOG_ERROR,
"Independently switched CCE with 'invalid' domain signalled.\n");
memset(coup, 0, sizeof(ChannelCoupling));
return -1;
}
sign = get_bits(gb, 1);
scale = pow(2., pow(2., get_bits(gb, 2) - 3));
if ((ret = decode_ics(ac, sce, gb, 0, 0)))
return ret;
for (c = 0; c < num_gain; c++) {
int cge = 1;
int gain = 0;
float gain_cache = 1.;
if (c) {
cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
gain_cache = pow(scale, gain);
}
for (g = 0; g < sce->ics.num_window_groups; g++)
for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++)
if (sce->band_type[idx] != ZERO_BT) {
if (!cge) {
int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
if (t) {
int s = 1;
if (sign) {
s -= 2 * (t & 0x1);
t >>= 1;
}
gain += t;
gain_cache = pow(scale, gain) * s;
}
}
coup->gain[c][idx] = gain_cache;
}
}
return 0;
}
/**
* Decode Spectral Band Replication extension data; reference: table 4.55.
*
* @param crc flag indicating the presence of CRC checksum
* @param cnt length of TYPE_FIL syntactic element in bytes
*
* @return Returns number of bytes consumed from the TYPE_FIL element.
*/
static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) {
// TODO : sbr_extension implementation
av_log_missing_feature(ac->avccontext, "SBR", 0);
skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type
return cnt;
}
/**
* Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
*
* @return Returns number of bytes consumed.
*/
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext * gb) {
int i;
int num_excl_chan = 0;
do {
for (i = 0; i < 7; i++)
che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
} while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
return num_excl_chan / 7;
}
/**
* Decode dynamic range information; reference: table 4.52.
*
* @param cnt length of TYPE_FIL syntactic element in bytes
*
* @return Returns number of bytes consumed.
*/
static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) {
int n = 1;
int drc_num_bands = 1;
int i;
/* pce_tag_present? */
if(get_bits1(gb)) {
che_drc->pce_instance_tag = get_bits(gb, 4);
skip_bits(gb, 4); // tag_reserved_bits
n++;
}
/* excluded_chns_present? */
if(get_bits1(gb)) {
n += decode_drc_channel_exclusions(che_drc, gb);
}
/* drc_bands_present? */
if (get_bits1(gb)) {
che_drc->band_incr = get_bits(gb, 4);
che_drc->interpolation_scheme = get_bits(gb, 4);
n++;
drc_num_bands += che_drc->band_incr;
for (i = 0; i < drc_num_bands; i++) {
che_drc->band_top[i] = get_bits(gb, 8);
n++;
}
}
/* prog_ref_level_present? */
if (get_bits1(gb)) {
che_drc->prog_ref_level = get_bits(gb, 7);
skip_bits1(gb); // prog_ref_level_reserved_bits
n++;
}
for (i = 0; i < drc_num_bands; i++) {
che_drc->dyn_rng_sgn[i] = get_bits1(gb);
che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
n++;
}
return n;
}
/**
* Decode extension data (incomplete); reference: table 4.51.
*
* @param cnt length of TYPE_FIL syntactic element in bytes
*
* @return Returns number of bytes consumed
*/
static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) {
int crc_flag = 0;
int res = cnt;
switch (get_bits(gb, 4)) { // extension type
case EXT_SBR_DATA_CRC:
crc_flag++;
case EXT_SBR_DATA:
res = decode_sbr_extension(ac, gb, crc_flag, cnt);
break;
case EXT_DYNAMIC_RANGE:
res = decode_dynamic_range(&ac->che_drc, gb, cnt);
break;
case EXT_FILL:
case EXT_FILL_DATA:
case EXT_DATA_ELEMENT:
default:
skip_bits_long(gb, 8*cnt - 4);
break;
};
return res;
}
/**
* Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
*
* @param decode 1 if tool is used normally, 0 if tool is used in LTP.
* @param coef spectral coefficients
*/
static void apply_tns(float coef[1024], TemporalNoiseShaping * tns, IndividualChannelStream * ics, int decode) {
const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
int w, filt, m, i;
int bottom, top, order, start, end, size, inc;
float lpc[TNS_MAX_ORDER];
for (w = 0; w < ics->num_windows; w++) {
bottom = ics->num_swb;
for (filt = 0; filt < tns->n_filt[w]; filt++) {
top = bottom;
bottom = FFMAX(0, top - tns->length[w][filt]);
order = tns->order[w][filt];
if (order == 0)
continue;
/* tns_decode_coef
* FIXME: This duplicates the functionality of some double code in lpc.c.
*/
for (m = 0; m < order; m++) {
float tmp;
lpc[m] = tns->coef[w][filt][m];
for (i = 0; i < m/2; i++) {
tmp = lpc[i];
lpc[i] += lpc[m] * lpc[m-1-i];
lpc[m-1-i] += lpc[m] * tmp;
}
if(m & 1)
lpc[i] += lpc[m] * lpc[i];
}
start = ics->swb_offset[FFMIN(bottom, mmm)];
end = ics->swb_offset[FFMIN( top, mmm)];
if ((size = end - start) <= 0)
continue;
if (tns->direction[w][filt]) {
inc = -1; start = end - 1;
} else {
inc = 1;
}
start += w * 128;
// ar filter
for (m = 0; m < size; m++, start += inc)
for (i = 1; i <= FFMIN(m, order); i++)
coef[start] -= coef[start - i*inc] * lpc[i-1];
}
}
}
/**
* Conduct IMDCT and windowing.
*/
static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) {
IndividualChannelStream * ics = &sce->ics;
float * in = sce->coeffs;
float * out = sce->ret;
float * saved = sce->saved;
const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
float * buf = ac->buf_mdct;
DECLARE_ALIGNED(16, float, temp[128]);
int i;
// 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,
"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)
ff_imdct_half(&ac->mdct_small, buf + i, in + i);
} else
ff_imdct_half(&ac->mdct, buf, in);
/* window overlapping
* NOTE: To simplify the overlapping code, all 'meaningless' short to long
* and long to short transitions are considered to be short to short
* transitions. This leaves just two cases (long to long and short to short)
* with a little special sauce for EIGHT_SHORT_SEQUENCE.
*/
if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
(ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, ac->add_bias, 512);
} else {
for (i = 0; i < 448; i++)
out[i] = saved[i] + ac->add_bias;
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, ac->add_bias, 64);
ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, ac->add_bias, 64);
ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, ac->add_bias, 64);
ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, ac->add_bias, 64);
ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, ac->add_bias, 64);
memcpy( out + 448 + 4*128, temp, 64 * sizeof(float));
} else {
ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, ac->add_bias, 64);
for (i = 576; i < 1024; i++)
out[i] = buf[i-512] + ac->add_bias;
}
}
// buffer update
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
for (i = 0; i < 64; i++)
saved[i] = temp[64 + i] - ac->add_bias;
ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64);
ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64);
ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64);
memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
memcpy( saved, buf + 512, 448 * sizeof(float));
memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
} else { // LONG_STOP or ONLY_LONG
memcpy( saved, buf + 512, 512 * sizeof(float));
}
}
/**
* Apply dependent channel coupling (applied before IMDCT).
*
* @param index index into coupling gain array
*/
static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) {
IndividualChannelStream * ics = &cc->ch[0].ics;
const uint16_t * offsets = ics->swb_offset;
float * dest = sce->coeffs;
const float * src = cc->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,
"Dependent coupling is not supported together with LTP\n");
return;
}
for (g = 0; g < ics->num_window_groups; g++) {
for (i = 0; i < ics->max_sfb; i++, idx++) {
if (cc->ch[0].band_type[idx] != ZERO_BT) {
for (group = 0; group < ics->group_len[g]; group++) {
for (k = offsets[i]; k < offsets[i+1]; k++) {
// XXX dsputil-ize
dest[group*128+k] += cc->coup.gain[index][idx] * src[group*128+k];
}
}
}
}
dest += ics->group_len[g]*128;
src += ics->group_len[g]*128;
}
}
/**
* Apply independent channel coupling (applied after IMDCT).
*
* @param index index into coupling gain array
*/
static void apply_independent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) {
int i;
for (i = 0; i < 1024; i++)
sce->ret[i] += cc->coup.gain[index][0] * (cc->ch[0].ret[i] - ac->add_bias);
}
/**
* channel coupling transformation interface
*
* @param index index into coupling gain array
* @param apply_coupling_method pointer to (in)dependent coupling function
*/
static void apply_channel_coupling(AACContext * ac, ChannelElement * cc,
void (*apply_coupling_method)(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index))
{
int c;
int index = 0;
ChannelCoupling * coup = &cc->coup;
for (c = 0; c <= coup->num_coupled; c++) {
if (ac->che[coup->type[c]][coup->id_select[c]]) {
if (coup->ch_select[c] != 2) {
apply_coupling_method(ac, &ac->che[coup->type[c]][coup->id_select[c]]->ch[0], cc, index);
if (coup->ch_select[c] != 0)
index++;
}
if (coup->ch_select[c] != 1)
apply_coupling_method(ac, &ac->che[coup->type[c]][coup->id_select[c]]->ch[1], cc, index++);
} else {
av_log(ac->avccontext, AV_LOG_ERROR,
"coupling target %sE[%d] not available\n",
coup->type[c] == TYPE_CPE ? "CP" : "SC", coup->id_select[c]);
break;
}
}
}
/**
* Convert spectral data to float samples, applying all supported tools as appropriate.
*/
static void spectral_to_sample(AACContext * ac) {
int i, type;
for (i = 0; i < MAX_ELEM_ID; i++) {
for(type = 0; type < 4; type++) {
ChannelElement *che = ac->che[type][i];
if(che) {
if(che->coup.coupling_point == BEFORE_TNS)
apply_channel_coupling(ac, che, apply_dependent_coupling);
if(che->ch[0].tns.present)
apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
if(che->ch[1].tns.present)
apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
if(che->coup.coupling_point == BETWEEN_TNS_AND_IMDCT)
apply_channel_coupling(ac, che, apply_dependent_coupling);
imdct_and_windowing(ac, &che->ch[0]);
if(type == TYPE_CPE)
imdct_and_windowing(ac, &che->ch[1]);
if(che->coup.coupling_point == AFTER_IMDCT)
apply_channel_coupling(ac, che, apply_independent_coupling);
}
}
}
}
static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, const uint8_t * buf, int buf_size) {
AACContext * ac = avccontext->priv_data;
GetBitContext gb;
enum RawDataBlockType elem_type;
int err, elem_id, data_size_tmp;
init_get_bits(&gb, buf, buf_size*8);
// parse
while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
elem_id = get_bits(&gb, 4);
err = -1;
if(elem_type == TYPE_SCE && elem_id == 1 &&
!ac->che[TYPE_SCE][elem_id] && ac->che[TYPE_LFE][0]) {
/* 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] */
ac->che[TYPE_SCE][elem_id] = ac->che[TYPE_LFE][0];
ac->che[TYPE_LFE][0] = NULL;
}
if(elem_type < TYPE_DSE) {
if(!ac->che[elem_type][elem_id])
return -1;
if(elem_type != TYPE_CCE)
ac->che[elem_type][elem_id]->coup.coupling_point = 4;
}
switch (elem_type) {
case TYPE_SCE:
err = decode_ics(ac, &ac->che[TYPE_SCE][elem_id]->ch[0], &gb, 0, 0);
break;
case TYPE_CPE:
err = decode_cpe(ac, &gb, elem_id);
break;
case TYPE_CCE:
err = decode_cce(ac, &gb, ac->che[TYPE_CCE][elem_id]);
break;
case TYPE_LFE:
err = decode_ics(ac, &ac->che[TYPE_LFE][elem_id]->ch[0], &gb, 0, 0);
break;
case TYPE_DSE:
skip_data_stream_element(&gb);
err = 0;
break;
case TYPE_PCE:
{
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
if((err = decode_pce(ac, new_che_pos, &gb)))
break;
err = output_configure(ac, ac->che_pos, new_che_pos);
break;
}
case TYPE_FIL:
if (elem_id == 15)
elem_id += get_bits(&gb, 8) - 1;
while (elem_id > 0)
elem_id -= decode_extension_payload(ac, &gb, elem_id);
err = 0; /* FIXME */
break;
default:
err = -1; /* should not happen, but keeps compiler happy */
break;
}
if(err)
return err;
}
spectral_to_sample(ac);
if (!ac->is_saved) {
ac->is_saved = 1;
*data_size = 0;
return buf_size;
}
data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t);
if(*data_size < data_size_tmp) {
av_log(avccontext, 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, 1024, avccontext->channels);
return buf_size;
}
static av_cold int aac_decode_close(AVCodecContext * avccontext) {
AACContext * ac = avccontext->priv_data;
int i, type;
for (i = 0; i < MAX_ELEM_ID; i++) {
for(type = 0; type < 4; type++)
av_freep(&ac->che[type][i]);
}
ff_mdct_end(&ac->mdct);
ff_mdct_end(&ac->mdct_small);
return 0 ;
}
AVCodec aac_decoder = {
"aac",
CODEC_TYPE_AUDIO,
CODEC_ID_AAC,
sizeof(AACContext),
aac_decode_init,
NULL,
aac_decode_close,
aac_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
.sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
};