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ffmpeg/libavcodec/aacdec.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 )
*
* AAC LATM decoder
* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
* Copyright (c) 2010 Janne Grunau <janne-ffmpeg@jannau.net>
*
* 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 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
* Y Long Term Prediction
* Y intensity stereo
* Y channel coupling
* Y 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
* Y 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)
* Y 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 "internal.h"
#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
#include "fmtconvert.h"
#include "lpc.h"
#include "kbdwin.h"
#include "sinewin.h"
#include "aac.h"
#include "aactab.h"
#include "aacdectab.h"
#include "cbrt_tablegen.h"
#include "sbr.h"
#include "aacsbr.h"
#include "mpeg4audio.h"
#include "aacadtsdec.h"
#include <assert.h>
#include <errno.h>
#include <math.h>
#include <string.h>
#if ARCH_ARM
# include "arm/aac.h"
#endif
union float754 {
float f;
uint32_t i;
};
static VLC vlc_scalefactors;
static VLC vlc_spectral[11];
static const char overread_err[] = "Input buffer exhausted before END element found\n";
static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
{
// For PCE based channel configurations map the channels solely based on tags.
if (!ac->m4ac.chan_config) {
return ac->tag_che_map[type][elem_id];
}
// For indexed channel configurations map the channels solely based on position.
switch (ac->m4ac.chan_config) {
case 7:
if (ac->tags_mapped == 3 && type == TYPE_CPE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
}
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[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];
}
case 5:
if (ac->tags_mapped == 2 && type == TYPE_CPE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
}
case 4:
if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
}
case 3:
case 2:
if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
} else if (ac->m4ac.chan_config == 2) {
return NULL;
}
case 1:
if (!ac->tags_mapped && type == TYPE_SCE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
}
default:
return NULL;
}
}
/**
* Check for the channel element in the current channel position configuration.
* If it exists, make sure the appropriate element is allocated and map the
* channel order to match the internal FFmpeg channel layout.
*
* @param che_pos current channel position configuration
* @param type channel element type
* @param id channel element id
* @param channels count of the number of channels in the configuration
*
* @return Returns error status. 0 - OK, !0 - error
*/
static av_cold int che_configure(AACContext *ac,
enum ChannelPosition che_pos[4][MAX_ELEM_ID],
int type, int id, int *channels)
{
if (che_pos[type][id]) {
if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
return AVERROR(ENOMEM);
ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
if (type != TYPE_CCE) {
ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
if (type == TYPE_CPE ||
(type == TYPE_SCE && ac->m4ac.ps == 1)) {
ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
}
}
} else {
if (ac->che[type][id])
ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);
av_freep(&ac->che[type][id]);
}
return 0;
}
/**
* 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 av_cold int output_configure(AACContext *ac,
enum ChannelPosition che_pos[4][MAX_ELEM_ID],
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
int channel_config, enum OCStatus oc_type)
{
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) {
for (i = 0; i < tags_per_config[channel_config]; i++) {
if ((ret = che_configure(ac, che_pos,
aac_channel_layout_map[channel_config - 1][i][0],
aac_channel_layout_map[channel_config - 1][i][1],
&channels)))
return ret;
}
memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
avctx->channel_layout = aac_channel_layout[channel_config - 1];
} else {
/* 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 ((ret = che_configure(ac, che_pos, type, i, &channels)))
return ret;
}
}
memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
}
avctx->channels = channels;
ac->output_configured = oc_type;
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(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
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, sampling_index;
int comment_len;
skip_bits(gb, 2); // object_type
sampling_index = get_bits(gb, 4);
if (m4ac->sampling_index != sampling_index)
av_log(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);
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
if (get_bits_left(gb) < 4 * (num_front + num_side + num_back + num_lfe + num_assoc_data + num_cc)) {
av_log(avctx, AV_LOG_ERROR, overread_err);
return -1;
}
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 */
comment_len = get_bits(gb, 8) * 8;
if (get_bits_left(gb) < comment_len) {
av_log(avctx, AV_LOG_ERROR, overread_err);
return -1;
}
skip_bits_long(gb, comment_len);
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 av_cold int set_default_channel_config(AVCodecContext *avctx,
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
int channel_config)
{
if (channel_config < 1 || channel_config > 7) {
av_log(avctx, 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.
*
* @param ac pointer to AACContext, may be null
* @param avctx pointer to AVCCodecContext, used for logging
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
GetBitContext *gb,
MPEG4AudioConfig *m4ac,
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(avctx, "960/120 MDCT window is", 1);
return -1;
}
if (get_bits1(gb)) // dependsOnCoreCoder
skip_bits(gb, 14); // coreCoderDelay
extension_flag = get_bits1(gb);
if (m4ac->object_type == AOT_AAC_SCALABLE ||
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(avctx, m4ac, new_che_pos, gb)))
return ret;
} else {
if ((ret = set_default_channel_config(avctx, new_che_pos, channel_config)))
return ret;
}
if (ac && (ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
return ret;
if (extension_flag) {
switch (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 ac pointer to AACContext, may be null
* @param avctx pointer to AVCCodecContext, used for logging
* @param m4ac pointer to MPEG4AudioConfig, used for parsing
* @param data pointer to AVCodecContext extradata
* @param data_size size of AVCCodecContext extradata
*
* @return Returns error status or number of consumed bits. <0 - error
*/
static int decode_audio_specific_config(AACContext *ac,
AVCodecContext *avctx,
MPEG4AudioConfig *m4ac,
const uint8_t *data, int data_size)
{
GetBitContext gb;
int i;
av_dlog(avctx, "extradata size %d\n", avctx->extradata_size);
for (i = 0; i < avctx->extradata_size; i++)
av_dlog(avctx, "%02x ", avctx->extradata[i]);
av_dlog(avctx, "\n");
init_get_bits(&gb, data, data_size * 8);
if ((i = ff_mpeg4audio_get_config(m4ac, data, data_size)) < 0)
return -1;
if (m4ac->sampling_index > 12) {
av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
return -1;
}
if (m4ac->sbr == 1 && m4ac->ps == -1)
m4ac->ps = 1;
skip_bits_long(&gb, i);
switch (m4ac->object_type) {
case AOT_AAC_MAIN:
case AOT_AAC_LC:
case AOT_AAC_LTP:
if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config))
return -1;
break;
default:
av_log(avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
m4ac->sbr == 1? "SBR+" : "", m4ac->object_type);
return -1;
}
av_dlog(avctx, "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
m4ac->sample_rate, m4ac->sbr, m4ac->ps);
return get_bits_count(&gb);
}
/**
* 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_always_inline void reset_predict_state(PredictorState *ps)
{
ps->r0 = 0.0f;
ps->r1 = 0.0f;
ps->cor0 = 0.0f;
ps->cor1 = 0.0f;
ps->var0 = 1.0f;
ps->var1 = 1.0f;
}
static void reset_all_predictors(PredictorState *ps)
{
int i;
for (i = 0; i < MAX_PREDICTORS; i++)
reset_predict_state(&ps[i]);
}
static int sample_rate_idx (int rate)
{
if (92017 <= rate) return 0;
else if (75132 <= rate) return 1;
else if (55426 <= rate) return 2;
else if (46009 <= rate) return 3;
else if (37566 <= rate) return 4;
else if (27713 <= rate) return 5;
else if (23004 <= rate) return 6;
else if (18783 <= rate) return 7;
else if (13856 <= rate) return 8;
else if (11502 <= rate) return 9;
else if (9391 <= rate) return 10;
else return 11;
}
static void reset_predictor_group(PredictorState *ps, int group_num)
{
int i;
for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
reset_predict_state(&ps[i]);
}
#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 = avctx->priv_data;
float output_scale_factor;
ac->avctx = avctx;
ac->m4ac.sample_rate = avctx->sample_rate;
if (avctx->extradata_size > 0) {
if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
avctx->extradata,
avctx->extradata_size) < 0)
return -1;
} else {
int sr, i;
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
sr = sample_rate_idx(avctx->sample_rate);
ac->m4ac.sampling_index = sr;
ac->m4ac.channels = avctx->channels;
ac->m4ac.sbr = -1;
ac->m4ac.ps = -1;
for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
if (ff_mpeg4audio_channels[i] == avctx->channels)
break;
if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
i = 0;
}
ac->m4ac.chan_config = i;
if (ac->m4ac.chan_config) {
int ret = set_default_channel_config(avctx, new_che_pos, ac->m4ac.chan_config);
if (!ret)
output_configure(ac, ac->che_pos, new_che_pos, ac->m4ac.chan_config, OC_GLOBAL_HDR);
else if (avctx->error_recognition >= FF_ER_EXPLODE)
return AVERROR_INVALIDDATA;
}
}
if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
output_scale_factor = 1.0 / 32768.0;
} else {
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
output_scale_factor = 1.0;
}
AAC_INIT_VLC_STATIC( 0, 304);
AAC_INIT_VLC_STATIC( 1, 270);
AAC_INIT_VLC_STATIC( 2, 550);
AAC_INIT_VLC_STATIC( 3, 300);
AAC_INIT_VLC_STATIC( 4, 328);
AAC_INIT_VLC_STATIC( 5, 294);
AAC_INIT_VLC_STATIC( 6, 306);
AAC_INIT_VLC_STATIC( 7, 268);
AAC_INIT_VLC_STATIC( 8, 510);
AAC_INIT_VLC_STATIC( 9, 366);
AAC_INIT_VLC_STATIC(10, 462);
ff_aac_sbr_init();
dsputil_init(&ac->dsp, avctx);
ff_fmt_convert_init(&ac->fmt_conv, avctx);
ac->random_state = 0x1f2e3d4c;
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]),
ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
352);
ff_mdct_init(&ac->mdct, 11, 1, output_scale_factor/1024.0);
ff_mdct_init(&ac->mdct_small, 8, 1, output_scale_factor/128.0);
ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0/output_scale_factor);
// 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_init_ff_sine_windows(10);
ff_init_ff_sine_windows( 7);
cbrt_tableinit();
return 0;
}
/**
* Skip data_stream_element; reference: table 4.10.
*/
static int skip_data_stream_element(AACContext *ac, 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);
if (get_bits_left(gb) < 8 * count) {
av_log(ac->avctx, AV_LOG_ERROR, overread_err);
return -1;
}
skip_bits_long(gb, 8 * count);
return 0;
}
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
GetBitContext *gb)
{
int sfb;
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->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
return -1;
}
}
for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
ics->prediction_used[sfb] = get_bits1(gb);
}
return 0;
}
/**
* Decode Long Term Prediction data; reference: table 4.xx.
*/
static void decode_ltp(AACContext *ac, LongTermPrediction *ltp,
GetBitContext *gb, uint8_t max_sfb)
{
int sfb;
ltp->lag = get_bits(gb, 11);
ltp->coef = ltp_coef[get_bits(gb, 3)];
for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
ltp->used[sfb] = get_bits1(gb);
}
/**
* 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->avctx, 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 = ff_swb_offset_128[ac->m4ac.sampling_index];
ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index];
ics->tns_max_bands = ff_tns_max_bands_128[ac->m4ac.sampling_index];
ics->predictor_present = 0;
} else {
ics->max_sfb = get_bits(gb, 6);
ics->num_windows = 1;
ics->swb_offset = ff_swb_offset_1024[ac->m4ac.sampling_index];
ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index];
ics->tns_max_bands = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
ics->predictor_present = get_bits1(gb);
ics->predictor_reset_group = 0;
if (ics->predictor_present) {
if (ac->m4ac.object_type == AOT_AAC_MAIN) {
if (decode_prediction(ac, ics, gb)) {
memset(ics, 0, sizeof(IndividualChannelStream));
return -1;
}
} else if (ac->m4ac.object_type == AOT_AAC_LC) {
av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
memset(ics, 0, sizeof(IndividualChannelStream));
return -1;
} else {
if ((ics->ltp.present = get_bits(gb, 1)))
decode_ltp(ac, &ics->ltp, gb, ics->max_sfb);
}
}
}
if (ics->max_sfb > ics->num_swb) {
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));
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_end = k;
int sect_len_incr;
int sect_band_type = get_bits(gb, 4);
if (sect_band_type == 12) {
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->avctx, AV_LOG_ERROR, overread_err);
return -1;
}
if (sect_end > ics->max_sfb) {
av_log(ac->avctx, AV_LOG_ERROR,
"Number of bands (%d) exceeds limit (%d).\n",
sect_end, ics->max_sfb);
return -1;
}
for (; k < sect_end; k++) {
band_type [idx] = sect_band_type;
band_type_run_end[idx++] = sect_end;
}
}
}
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])
{
int g, i, idx = 0;
int offset[3] = { global_gain, global_gain - 90, 0 };
int clipped_offset;
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;
clipped_offset = av_clip(offset[2], -155, 100);
if (offset[2] != clipped_offset) {
av_log_ask_for_sample(ac->avctx, "Intensity stereo "
"position clipped (%d -> %d).\nIf you heard an "
"audible artifact, there may be a bug in the "
"decoder. ", offset[2], clipped_offset);
}
sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
}
} 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;
clipped_offset = av_clip(offset[1], -100, 155);
if (offset[1] != clipped_offset) {
av_log_ask_for_sample(ac->avctx, "Noise gain clipped "
"(%d -> %d).\nIf you heard an audible "
"artifact, there may be a bug in the decoder. ",
offset[1], clipped_offset);
}
sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
}
} 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->avctx, AV_LOG_ERROR,
"%s (%d) out of range.\n", sf_str[0], offset[0]);
return -1;
}
sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
}
}
}
}
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->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;
}
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]));
}
}
#ifndef VMUL2
static inline float *VMUL2(float *dst, const float *v, unsigned idx,
const float *scale)
{
float s = *scale;
*dst++ = v[idx & 15] * s;
*dst++ = v[idx>>4 & 15] * s;
return dst;
}
#endif
#ifndef VMUL4
static inline float *VMUL4(float *dst, const float *v, unsigned idx,
const float *scale)
{
float s = *scale;
*dst++ = v[idx & 3] * s;
*dst++ = v[idx>>2 & 3] * s;
*dst++ = v[idx>>4 & 3] * s;
*dst++ = v[idx>>6 & 3] * s;
return dst;
}
#endif
#ifndef VMUL2S
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
unsigned sign, const float *scale)
{
union float754 s0, s1;
s0.f = s1.f = *scale;
s0.i ^= sign >> 1 << 31;
s1.i ^= sign << 31;
*dst++ = v[idx & 15] * s0.f;
*dst++ = v[idx>>4 & 15] * s1.f;
return dst;
}
#endif
#ifndef VMUL4S
static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
unsigned sign, const float *scale)
{
unsigned nz = idx >> 12;
union float754 s = { .f = *scale };
union float754 t;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx & 3] * t.f;
sign <<= nz & 1; nz >>= 1;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>2 & 3] * t.f;
sign <<= nz & 1; nz >>= 1;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>4 & 3] * t.f;
sign <<= nz & 1; nz >>= 1;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>6 & 3] * t.f;
return dst;
}
#endif
/**
* 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, const 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++) {
unsigned g_len = ics->group_len[g];
for (i = 0; i < ics->max_sfb; i++, idx++) {
const unsigned cbt_m1 = band_type[idx] - 1;
float *cfo = coef + offsets[i];
int off_len = offsets[i + 1] - offsets[i];
int group;
if (cbt_m1 >= INTENSITY_BT2 - 1) {
for (group = 0; group < g_len; group++, cfo+=128) {
memset(cfo, 0, off_len * sizeof(float));
}
} else if (cbt_m1 == NOISE_BT - 1) {
for (group = 0; group < g_len; group++, cfo+=128) {
float scale;
float band_energy;
for (k = 0; k < off_len; k++) {
ac->random_state = lcg_random(ac->random_state);
cfo[k] = ac->random_state;
}
band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
scale = sf[idx] / sqrtf(band_energy);
ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
}
} else {
const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
OPEN_READER(re, gb);
switch (cbt_m1 >> 1) {
case 0:
for (group = 0; group < g_len; group++, cfo+=128) {
float *cf = cfo;
int len = off_len;
do {
int code;
unsigned cb_idx;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_idx[code];
cf = VMUL4(cf, vq, cb_idx, sf + idx);
} while (len -= 4);
}
break;
case 1:
for (group = 0; group < g_len; group++, cfo+=128) {
float *cf = cfo;
int len = off_len;
do {
int code;
unsigned nnz;
unsigned cb_idx;
uint32_t bits;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_idx[code];
nnz = cb_idx >> 8 & 15;
bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
LAST_SKIP_BITS(re, gb, nnz);
cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
} while (len -= 4);
}
break;
case 2:
for (group = 0; group < g_len; group++, cfo+=128) {
float *cf = cfo;
int len = off_len;
do {
int code;
unsigned cb_idx;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_idx[code];
cf = VMUL2(cf, vq, cb_idx, sf + idx);
} while (len -= 2);
}
break;
case 3:
case 4:
for (group = 0; group < g_len; group++, cfo+=128) {
float *cf = cfo;
int len = off_len;
do {
int code;
unsigned nnz;
unsigned cb_idx;
unsigned sign;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_idx[code];
nnz = cb_idx >> 8 & 15;
sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12);
LAST_SKIP_BITS(re, gb, nnz);
cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
} while (len -= 2);
}
break;
default:
for (group = 0; group < g_len; group++, cfo+=128) {
float *cf = cfo;
uint32_t *icf = (uint32_t *) cf;
int len = off_len;
do {
int code;
unsigned nzt, nnz;
unsigned cb_idx;
uint32_t bits;
int j;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
if (!code) {
*icf++ = 0;
*icf++ = 0;
continue;
}
cb_idx = cb_vector_idx[code];
nnz = cb_idx >> 12;
nzt = cb_idx >> 8;
bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
LAST_SKIP_BITS(re, gb, nnz);
for (j = 0; j < 2; j++) {
if (nzt & 1<<j) {
uint32_t b;
int n;
/* The total length of escape_sequence must be < 22 bits according
to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
UPDATE_CACHE(re, gb);
b = GET_CACHE(re, gb);
b = 31 - av_log2(~b);
if (b > 8) {
av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
return -1;
}
SKIP_BITS(re, gb, b + 1);
b += 4;
n = (1 << b) + SHOW_UBITS(re, gb, b);
LAST_SKIP_BITS(re, gb, b);
*icf++ = cbrt_tab[n] | (bits & 1U<<31);
bits <<= 1;
} else {
unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
*icf++ = (bits & 1U<<31) | v;
bits <<= !!v;
}
cb_idx >>= 4;
}
} while (len -= 2);
ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
}
}
CLOSE_READER(re, gb);
}
}
coef += g_len << 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;
}
static av_always_inline float flt16_round(float pf)
{
union float754 tmp;
tmp.f = pf;
tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
return tmp.f;
}
static av_always_inline float flt16_even(float pf)
{
union float754 tmp;
tmp.f = pf;
tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
return tmp.f;
}
static av_always_inline float flt16_trunc(float pf)
{
union float754 pun;
pun.f = pf;
pun.i &= 0xFFFF0000U;
return pun.f;
}
static av_always_inline void predict(PredictorState *ps, float *coef,
int output_enable)
{
const float a = 0.953125; // 61.0 / 64
const float alpha = 0.90625; // 29.0 / 32
float e0, e1;
float pv;
float k1, k2;
float r0 = ps->r0, r1 = ps->r1;
float cor0 = ps->cor0, cor1 = ps->cor1;
float var0 = ps->var0, var1 = ps->var1;
k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
pv = flt16_round(k1 * r0 + k2 * r1);
if (output_enable)
*coef += pv;
e0 = *coef;
e1 = e0 - k1 * r0;
ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
ps->r0 = flt16_trunc(a * e0);
}
/**
* Apply AAC-Main style frequency domain prediction.
*/
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
{
int sfb, k;
if (!sce->ics.predictor_initialized) {
reset_all_predictors(sce->predictor_state);
sce->ics.predictor_initialized = 1;
}
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
predict(&sce->predictor_state[k], &sce->coeffs[k],
sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
}
}
if (sce->ics.predictor_reset_group)
reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
} else
reset_all_predictors(sce->predictor_state);
}
/**
* 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->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->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->avctx, "SSR", 1);
return -1;
}
}
if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
return -1;
if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
apply_prediction(ac, sce);
return 0;
}
/**
* Mid/Side stereo decoding; reference: 4.6.8.1.3.
*/
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
{
const IndividualChannelStream *ics = &cpe->ch[0].ics;
float *ch0 = cpe->ch[0].coeffs;
float *ch1 = cpe->ch[1].coeffs;
int g, i, 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++) {
ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
ch1 + group * 128 + offsets[i],
offsets[i+1] - offsets[i]);
}
}
}
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(AACContext *ac, 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, 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++)
ac->dsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
coef0 + group * 128 + offsets[i],
scale,
offsets[i + 1] - offsets[i]);
}
} 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.
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
{
int i, ret, common_window, ms_present = 0;
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;
if (cpe->ch[1].ics.predictor_present && (ac->m4ac.object_type != AOT_AAC_MAIN))
if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
decode_ltp(ac, &cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
ms_present = get_bits(gb, 2);
if (ms_present == 3) {
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);
}
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) {
if (ms_present)
apply_mid_side_stereo(ac, cpe);
if (ac->m4ac.object_type == AOT_AAC_MAIN) {
apply_prediction(ac, &cpe->ch[0]);
apply_prediction(ac, &cpe->ch[1]);
}
}
apply_intensity_stereo(ac, cpe, ms_present);
return 0;
}
static const float cce_scale[] = {
1.09050773266525765921, //2^(1/8)
1.18920711500272106672, //2^(1/4)
M_SQRT2,
2,
};
/**
* Decode coupling_channel_element; reference: table 4.8.
*
* @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) || (coup->coupling_point >> 1);
sign = get_bits(gb, 1);
scale = cce_scale[get_bits(gb, 2)];
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 = powf(scale, -gain);
}
if (coup->coupling_point == AFTER_IMDCT) {
coup->gain[c][0] = gain_cache;
} else {
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 = powf(scale, -t) * s;
}
}
coup->gain[c][idx] = gain_cache;
}
}
}
}
}
return 0;
}
/**
* 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,
ChannelElement *che, enum RawDataBlockType elem_type)
{
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:
if (!che) {
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->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->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;
}
res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
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];
float tmp[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;
if (decode) {
// 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];
} else {
// ma filter
for (m = 0; m < size; m++, start += inc) {
tmp[0] = coef[start];
for (i = 1; i <= FFMIN(m, order); i++)
coef[start] += tmp[i] * lpc[i - 1];
for (i = order; i > 0; i--)
tmp[i] = tmp[i - 1];
}
}
}
}
}
/**
* Apply windowing and MDCT to obtain the spectral
* coefficient from the predicted sample by LTP.
*/
static void windowing_and_mdct_ltp(AACContext *ac, float *out,
float *in, IndividualChannelStream *ics)
{
const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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;
if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
ac->dsp.vector_fmul(in, in, lwindow_prev, 1024);
} else {
memset(in, 0, 448 * sizeof(float));
ac->dsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
}
if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
} else {
ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
memset(in + 1024 + 576, 0, 448 * sizeof(float));
}
ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
}
/**
* Apply the long term prediction
*/
static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
{
const LongTermPrediction *ltp = &sce->ics.ltp;
const uint16_t *offsets = sce->ics.swb_offset;
int i, sfb;
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
float *predTime = sce->ret;
float *predFreq = ac->buf_mdct;
int16_t num_samples = 2048;
if (ltp->lag < 1024)
num_samples = ltp->lag + 1024;
for (i = 0; i < num_samples; i++)
predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
memset(&predTime[i], 0, (2048 - i) * sizeof(float));
windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
if (sce->tns.present)
apply_tns(predFreq, &sce->tns, &sce->ics, 0);
for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
if (ltp->used[sfb])
for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
sce->coeffs[i] += predFreq[i];
}
}
/**
* Update the LTP buffer for next frame
*/
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
{
IndividualChannelStream *ics = &sce->ics;
float *saved = sce->saved;
float *saved_ltp = sce->coeffs;
const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
int i;
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
memcpy(saved_ltp, saved, 512 * sizeof(float));
memset(saved_ltp + 576, 0, 448 * sizeof(float));
ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
for (i = 0; i < 64; i++)
saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(float));
memset(saved_ltp + 576, 0, 448 * sizeof(float));
ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
for (i = 0; i < 64; i++)
saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
} else { // LONG_STOP or ONLY_LONG
ac->dsp.vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
for (i = 0; i < 512; i++)
saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
}
memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
}
/**
* 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;
float *temp = ac->temp;
int i;
// imdct
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
for (i = 0; i < 1024; i += 128)
ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
} else
ac->mdct.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, 512);
} else {
memcpy( out, saved, 448 * sizeof(float));
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
memcpy( out + 448 + 4*128, temp, 64 * sizeof(float));
} else {
ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
memcpy( out + 576, buf + 64, 448 * sizeof(float));
}
}
// buffer update
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
memcpy( saved, temp + 64, 64 * sizeof(float));
ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 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->avctx, 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) {
const float gain = cce->coup.gain[index][idx];
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] += gain * 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;
const float gain = cce->coup.gain[index][0];
const float *src = cce->ch[0].ret;
float *dest = target->ret;
const int len = 1024 << (ac->m4ac.sbr == 1);
for (i = 0; i < len; i++)
dest[i] += gain * src[i];
}
/**
* channel coupling transformation interface
*
* @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 (ac->m4ac.object_type == AOT_AAC_LTP) {
if (che->ch[0].ics.predictor_present) {
if (che->ch[0].ics.ltp.present)
apply_ltp(ac, &che->ch[0]);
if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
apply_ltp(ac, &che->ch[1]);
}
}
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 (ac->m4ac.object_type == AOT_AAC_LTP)
update_ltp(ac, &che->ch[0]);
if (type == TYPE_CPE) {
imdct_and_windowing(ac, &che->ch[1]);
if (ac->m4ac.object_type == AOT_AAC_LTP)
update_ltp(ac, &che->ch[1]);
}
if (ac->m4ac.sbr > 0) {
ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
}
}
if (type <= TYPE_CCE)
apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
}
}
}
}
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
{
int size;
AACADTSHeaderInfo hdr_info;
size = ff_aac_parse_header(gb, &hdr_info);
if (size > 0) {
if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
ac->m4ac.chan_config = hdr_info.chan_config;
if (set_default_channel_config(ac->avctx, new_che_pos, hdr_info.chan_config))
return -7;
if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
return -7;
} else if (ac->output_configured != OC_LOCKED) {
ac->m4ac.chan_config = 0;
ac->output_configured = OC_NONE;
}
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->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->avctx, "More than one AAC RDB per ADTS frame is", 0);
return -1;
}
}
return size;
}
static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
int *data_size, GetBitContext *gb)
{
AACContext *ac = avctx->priv_data;
ChannelElement *che = NULL, *che_prev = NULL;
enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
int err, elem_id, data_size_tmp;
int samples = 0, multiplier, audio_found = 0;
if (show_bits(gb, 12) == 0xfff) {
if (parse_adts_frame_header(ac, gb) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
return -1;
}
if (ac->m4ac.sampling_index > 12) {
av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
return -1;
}
}
ac->tags_mapped = 0;
// parse
while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
elem_id = get_bits(gb, 4);
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) {
case TYPE_SCE:
err = decode_ics(ac, &che->ch[0], gb, 0, 0);
audio_found = 1;
break;
case TYPE_CPE:
err = decode_cpe(ac, gb, che);
audio_found = 1;
break;
case TYPE_CCE:
err = decode_cce(ac, gb, che);
break;
case TYPE_LFE:
err = decode_ics(ac, &che->ch[0], gb, 0, 0);
audio_found = 1;
break;
case TYPE_DSE:
err = skip_data_stream_element(ac, gb);
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(avctx, &ac->m4ac, new_che_pos, gb)))
break;
if (ac->output_configured > OC_TRIAL_PCE)
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);
break;
}
case TYPE_FIL:
if (elem_id == 15)
elem_id += get_bits(gb, 8) - 1;
if (get_bits_left(gb) < 8 * elem_id) {
av_log(avctx, AV_LOG_ERROR, overread_err);
return -1;
}
while (elem_id > 0)
elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
err = 0; /* FIXME */
break;
default:
err = -1; /* should not happen, but keeps compiler happy */
break;
}
che_prev = che;
elem_type_prev = elem_type;
if (err)
return err;
if (get_bits_left(gb) < 3) {
av_log(avctx, AV_LOG_ERROR, overread_err);
return -1;
}
}
spectral_to_sample(ac);
multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0;
samples <<= multiplier;
if (ac->output_configured < OC_LOCKED) {
avctx->sample_rate = ac->m4ac.sample_rate << multiplier;
avctx->frame_size = samples;
}
data_size_tmp = samples * avctx->channels *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < data_size_tmp) {
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;
if (samples) {
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
ac->fmt_conv.float_interleave(data, (const float **)ac->output_data,
samples, avctx->channels);
else
ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data,
samples, avctx->channels);
}
if (ac->output_configured && audio_found)
ac->output_configured = OC_LOCKED;
return 0;
}
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;
GetBitContext gb;
int buf_consumed;
int buf_offset;
int err;
init_get_bits(&gb, buf, buf_size * 8);
if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0)
return err;
buf_consumed = (get_bits_count(&gb) + 7) >> 3;
for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
if (buf[buf_offset])
break;
return buf_size > buf_offset ? buf_consumed : buf_size;
}
static av_cold int aac_decode_close(AVCodecContext *avctx)
{
AACContext *ac = avctx->priv_data;
int i, type;
for (i = 0; i < MAX_ELEM_ID; i++) {
for (type = 0; type < 4; type++) {
if (ac->che[type][i])
ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
av_freep(&ac->che[type][i]);
}
}
ff_mdct_end(&ac->mdct);
ff_mdct_end(&ac->mdct_small);
ff_mdct_end(&ac->mdct_ltp);
return 0;
}
#define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word
struct LATMContext {
AACContext aac_ctx; ///< containing AACContext
int initialized; ///< initilized after a valid extradata was seen
// parser data
int audio_mux_version_A; ///< LATM syntax version
int frame_length_type; ///< 0/1 variable/fixed frame length
int frame_length; ///< frame length for fixed frame length
};
static inline uint32_t latm_get_value(GetBitContext *b)
{
int length = get_bits(b, 2);
return get_bits_long(b, (length+1)*8);
}
static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
GetBitContext *gb)
{
AVCodecContext *avctx = latmctx->aac_ctx.avctx;
MPEG4AudioConfig m4ac;
int config_start_bit = get_bits_count(gb);
int bits_consumed, esize;
if (config_start_bit % 8) {
av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
"config not byte aligned.\n", 1);
return AVERROR_INVALIDDATA;
} else {
bits_consumed =
decode_audio_specific_config(NULL, avctx, &m4ac,
gb->buffer + (config_start_bit / 8),
get_bits_left(gb) / 8);
if (bits_consumed < 0)
return AVERROR_INVALIDDATA;
esize = (bits_consumed+7) / 8;
if (avctx->extradata_size <= esize) {
av_free(avctx->extradata);
avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata)
return AVERROR(ENOMEM);
}
avctx->extradata_size = esize;
memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
skip_bits_long(gb, bits_consumed);
}
return bits_consumed;
}
static int read_stream_mux_config(struct LATMContext *latmctx,
GetBitContext *gb)
{
int ret, audio_mux_version = get_bits(gb, 1);
latmctx->audio_mux_version_A = 0;
if (audio_mux_version)
latmctx->audio_mux_version_A = get_bits(gb, 1);
if (!latmctx->audio_mux_version_A) {
if (audio_mux_version)
latm_get_value(gb); // taraFullness
skip_bits(gb, 1); // allStreamSameTimeFraming
skip_bits(gb, 6); // numSubFrames
// numPrograms
if (get_bits(gb, 4)) { // numPrograms
av_log_missing_feature(latmctx->aac_ctx.avctx,
"multiple programs are not supported\n", 1);
return AVERROR_PATCHWELCOME;
}
// for each program (which there is only on in DVB)
// for each layer (which there is only on in DVB)
if (get_bits(gb, 3)) { // numLayer
av_log_missing_feature(latmctx->aac_ctx.avctx,
"multiple layers are not supported\n", 1);
return AVERROR_PATCHWELCOME;
}
// for all but first stream: use_same_config = get_bits(gb, 1);
if (!audio_mux_version) {
if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
return ret;
} else {
int ascLen = latm_get_value(gb);
if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
return ret;
ascLen -= ret;
skip_bits_long(gb, ascLen);
}
latmctx->frame_length_type = get_bits(gb, 3);
switch (latmctx->frame_length_type) {
case 0:
skip_bits(gb, 8); // latmBufferFullness
break;
case 1:
latmctx->frame_length = get_bits(gb, 9);
break;
case 3:
case 4:
case 5:
skip_bits(gb, 6); // CELP frame length table index
break;
case 6:
case 7:
skip_bits(gb, 1); // HVXC frame length table index
break;
}
if (get_bits(gb, 1)) { // other data
if (audio_mux_version) {
latm_get_value(gb); // other_data_bits
} else {
int esc;
do {
esc = get_bits(gb, 1);
skip_bits(gb, 8);
} while (esc);
}
}
if (get_bits(gb, 1)) // crc present
skip_bits(gb, 8); // config_crc
}
return 0;
}
static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
{
uint8_t tmp;
if (ctx->frame_length_type == 0) {
int mux_slot_length = 0;
do {
tmp = get_bits(gb, 8);
mux_slot_length += tmp;
} while (tmp == 255);
return mux_slot_length;
} else if (ctx->frame_length_type == 1) {
return ctx->frame_length;
} else if (ctx->frame_length_type == 3 ||
ctx->frame_length_type == 5 ||
ctx->frame_length_type == 7) {
skip_bits(gb, 2); // mux_slot_length_coded
}
return 0;
}
static int read_audio_mux_element(struct LATMContext *latmctx,
GetBitContext *gb)
{
int err;
uint8_t use_same_mux = get_bits(gb, 1);
if (!use_same_mux) {
if ((err = read_stream_mux_config(latmctx, gb)) < 0)
return err;
} else if (!latmctx->aac_ctx.avctx->extradata) {
av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
"no decoder config found\n");
return AVERROR(EAGAIN);
}
if (latmctx->audio_mux_version_A == 0) {
int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
return AVERROR_INVALIDDATA;
} else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
"frame length mismatch %d << %d\n",
mux_slot_length_bytes * 8, get_bits_left(gb));
return AVERROR_INVALIDDATA;
}
}
return 0;
}
static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
AVPacket *avpkt)
{
struct LATMContext *latmctx = avctx->priv_data;
int muxlength, err;
GetBitContext gb;
if (avpkt->size == 0)
return 0;
init_get_bits(&gb, avpkt->data, avpkt->size * 8);
// check for LOAS sync word
if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
return AVERROR_INVALIDDATA;
muxlength = get_bits(&gb, 13) + 3;
// not enough data, the parser should have sorted this
if (muxlength > avpkt->size)
return AVERROR_INVALIDDATA;
if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
return err;
if (!latmctx->initialized) {
if (!avctx->extradata) {
*out_size = 0;
return avpkt->size;
} else {
aac_decode_close(avctx);
if ((err = aac_decode_init(avctx)) < 0)
return err;
latmctx->initialized = 1;
}
}
if (show_bits(&gb, 12) == 0xfff) {
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
"ADTS header detected, probably as result of configuration "
"misparsing\n");
return AVERROR_INVALIDDATA;
}
if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
return err;
return muxlength;
}
av_cold static int latm_decode_init(AVCodecContext *avctx)
{
struct LATMContext *latmctx = avctx->priv_data;
int ret;
ret = aac_decode_init(avctx);
if (avctx->extradata_size > 0) {
latmctx->initialized = !ret;
} else {
latmctx->initialized = 0;
}
return ret;
}
AVCodec ff_aac_decoder = {
.name = "aac",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_AAC,
.priv_data_size = sizeof(AACContext),
.init = aac_decode_init,
.close = aac_decode_close,
.decode = aac_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
.capabilities = CODEC_CAP_CHANNEL_CONF,
.channel_layouts = aac_channel_layout,
};
/*
Note: This decoder filter is intended to decode LATM streams transferred
in MPEG transport streams which only contain one program.
To do a more complex LATM demuxing a separate LATM demuxer should be used.
*/
AVCodec ff_aac_latm_decoder = {
.name = "aac_latm",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_AAC_LATM,
.priv_data_size = sizeof(struct LATMContext),
.init = latm_decode_init,
.close = aac_decode_close,
.decode = latm_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
.capabilities = CODEC_CAP_CHANNEL_CONF,
.channel_layouts = aac_channel_layout,
};
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