virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-07 11:13:41 +02:00
Code Issues Releases Activity
def4f63b85
FFmpeg/libavcodec/aac.c
Alex Converse fbdae895f2 Rename variables to clarify the channel coupling element and corresponding 
target channel element.

Patch by Alex Converse (alex converse gmail com)

Originally committed as revision 15575 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-10-06 16:22:11 +00:00

1477 lines
54 KiB
C
Raw Blame History

/*
* 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 "lpc.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 int decode_pulses(Pulse * pulse, GetBitContext * gb, const uint16_t * swb_offset, int num_swb) {
int i, pulse_swb;
pulse->num_pulse = get_bits(gb, 2) + 1;
pulse_swb = get_bits(gb, 6);
if (pulse_swb >= num_swb)
return -1;
pulse->pos[0] = swb_offset[pulse_swb];
pulse->pos[0] += get_bits(gb, 5);
if (pulse->pos[0] > 1023)
return -1;
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];
if (pulse->pos[i] > 1023)
return -1;
pulse->amp[i] = get_bits(gb, 4);
}
return 0;
}
/**
* 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;
}
if (tns->order[w][filt]) {
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) {
idx = 0;
for(i = 0; i < pulse->num_pulse; i++){
float co = coef_base[ pulse->pos[i] ];
while(offsets[idx + 1] <= pulse->pos[i])
idx++;
if (band_type[idx] != NOISE_BT && sf[idx]) {
float ico = -pulse->amp[i];
if (co) {
co /= sf[idx];
ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
}
coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
}
}
}
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;
}
if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
av_log(ac->avccontext, 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);
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;
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] = 2;
}
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., (int)get_bits(gb, 2) - 3));
if ((ret = decode_ics(ac, sce, gb, 0, 0)))
return ret;
for (c = 0; c < num_gain; c++) {
int idx = 0;
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;
t = gain += t;
if (sign) {
s -= 2 * (t & 0x1);
t >>= 1;
}
gain_cache = pow(scale, -t) * 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
compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
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 * target, ChannelElement * cce, int index) {
IndividualChannelStream * ics = &cce->ch[0].ics;
const uint16_t * offsets = ics->swb_offset;
float * dest = target->coeffs;
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,
"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 (cce->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] += cce->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 * target, ChannelElement * cce, int index) {
int i;
for (i = 0; i < 1024; i++)
target->ret[i] += cce->coup.gain[index][0] * (cce->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,
enum RawDataBlockType type, int elem_id, enum CouplingPoint coupling_point,
void (*apply_coupling_method)(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index))
{
int i, c;
for (i = 0; i < MAX_ELEM_ID; i++) {
ChannelElement *cce = ac->che[TYPE_CCE][i];
int index = 0;
if (cce && cce->coup.coupling_point == coupling_point) {
ChannelCoupling * coup = &cce->coup;
for (c = 0; c <= coup->num_coupled; c++) {
if (coup->type[c] == type && coup->id_select[c] == elem_id) {
if (coup->ch_select[c] != 1) {
apply_coupling_method(ac, &cc->ch[0], cce, index);
if (coup->ch_select[c] != 0)
index++;
}
if (coup->ch_select[c] != 2)
apply_coupling_method(ac, &cc->ch[1], cce, index++);
} else
index += 1 + (coup->ch_select[c] == 3);
}
}
}
}
/**
* Convert spectral data to float samples, applying all supported tools as appropriate.
*/
static void spectral_to_sample(AACContext * ac) {
int i, type;
for(type = 3; type >= 0; type--) {
for (i = 0; i < MAX_ELEM_ID; i++) {
ChannelElement *che = ac->che[type][i];
if(che) {
if(type <= TYPE_CPE)
apply_channel_coupling(ac, che, type, i, BEFORE_TNS, 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(type <= TYPE_CPE)
apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
if(type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT)
imdct_and_windowing(ac, &che->ch[0]);
if(type == TYPE_CPE)
imdct_and_windowing(ac, &che->ch[1]);
if(type <= TYPE_CCE)
apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, 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},
};
Reference in New Issue View Git Blame Copy Permalink