@@ -43,7 +43,7 @@ OBJS-$(CONFIG_VAAPI)                   += vaapi.o

 OBJS-$(CONFIG_VDPAU)                   += vdpau.o

 # decoders/encoders/hardware accelerators

-OBJS-$(CONFIG_AAC_DECODER)             += aac.o aactab.o aacsbr.o

+OBJS-$(CONFIG_AAC_DECODER)             += aacdec.o aactab.o aacsbr.o aacps.o

 OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o    \

                                           aacpsy.o aactab.o      \

                                           psymodel.o iirfilter.o \

deleted file mode 100644

@@ -1,2108 +0,0 @@

-/*

- * 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 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

- * 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)

- * 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 "internal.h"

-#include "get_bits.h"

-#include "dsputil.h"

-#include "fft.h"

-#include "lpc.h"

-

-#include "aac.h"

-#include "aactab.h"

-#include "aacdectab.h"

-#include "cbrt_tablegen.h"

-#include "sbr.h"

-#include "aacsbr.h"

-#include "mpeg4audio.h"

-#include "aac_parser.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)

-{

-    if (ac->tag_che_map[type][elem_id]) {

-        return ac->tag_che_map[type][elem_id];

-    }

-    if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) {

-        return NULL;

-    }

-    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[0] LFE[0]. If we seem to have

-           encountered such a stream, transfer the LFE[0] element to SCE[1] */

-        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->che[type][id]->sbr);

-        if (type != TYPE_CCE) {

-            ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;

-            if (type == TYPE_CPE) {

-                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->avccontext;

-    int i, type, channels = 0, ret;

-

-    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]));

-        ac->tags_mapped = 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]));

-        ac->tags_mapped = 4 * MAX_ELEM_ID;

-

-        avctx->channel_layout = 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(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, sampling_index;

-    int comment_len;

-

-    skip_bits(gb, 2);  // object_type

-

-    sampling_index = get_bits(gb, 4);

-    if (ac->m4ac.sampling_index != sampling_index)

-        av_log(ac->avccontext, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");

-

-    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 */

-    comment_len = get_bits(gb, 8) * 8;

-    if (get_bits_left(gb) < comment_len) {

-        av_log(ac->avccontext, 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(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, channel_config, OC_GLOBAL_HDR)))

-        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 > 12) {

-        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_MAIN:

-    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_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 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]);

-}

-

-static av_cold int aac_decode_init(AVCodecContext *avccontext)

-{

-    AACContext *ac = avccontext->priv_data;

-    int i;

-

-    ac->avccontext = avccontext;

-    ac->m4ac.sample_rate = avccontext->sample_rate;

-

-    if (avccontext->extradata_size > 0) {

-        if (decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))

-            return -1;

-    }

-

-    avccontext->sample_fmt = SAMPLE_FMT_S16;

-

-    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, 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_interleave == ff_float_to_int16_interleave_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;

-    }

-

-#if !CONFIG_HARDCODED_TABLES

-    for (i = 0; i < 428; i++)

-        ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);

-#endif /* CONFIG_HARDCODED_TABLES */

-

-    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, 1.0);

-    ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);

-    // 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->avccontext, 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->avccontext, 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 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        =    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->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");

-                memset(ics, 0, sizeof(IndividualChannelStream));

-                return -1;

-            } else {

-                av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1);

-                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_end = 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_end += sect_len_incr;

-            sect_end += sect_len_incr;

-            if (get_bits_left(gb) < 0) {

-                av_log(ac->avccontext, AV_LOG_ERROR, overread_err);

-                return -1;

-            }

-            if (sect_end > ics->max_sfb) {

-                av_log(ac->avccontext, 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])

-{

-    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 + 100];

-                }

-            } 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.\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)];

-                }

-            }

-        }

-    }