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libavcodec/aacdec.c

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00001 /*
00002  * AAC decoder
00003  * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
00004  * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
00005  *
00006  * AAC LATM decoder
00007  * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
00008  * Copyright (c) 2010      Janne Grunau <janne-ffmpeg@jannau.net>
00009  *
00010  * This file is part of FFmpeg.
00011  *
00012  * FFmpeg is free software; you can redistribute it and/or
00013  * modify it under the terms of the GNU Lesser General Public
00014  * License as published by the Free Software Foundation; either
00015  * version 2.1 of the License, or (at your option) any later version.
00016  *
00017  * FFmpeg is distributed in the hope that it will be useful,
00018  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00019  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00020  * Lesser General Public License for more details.
00021  *
00022  * You should have received a copy of the GNU Lesser General Public
00023  * License along with FFmpeg; if not, write to the Free Software
00024  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00025  */
00026 
00034 /*
00035  * supported tools
00036  *
00037  * Support?             Name
00038  * N (code in SoC repo) gain control
00039  * Y                    block switching
00040  * Y                    window shapes - standard
00041  * N                    window shapes - Low Delay
00042  * Y                    filterbank - standard
00043  * N (code in SoC repo) filterbank - Scalable Sample Rate
00044  * Y                    Temporal Noise Shaping
00045  * Y                    Long Term Prediction
00046  * Y                    intensity stereo
00047  * Y                    channel coupling
00048  * Y                    frequency domain prediction
00049  * Y                    Perceptual Noise Substitution
00050  * Y                    Mid/Side stereo
00051  * N                    Scalable Inverse AAC Quantization
00052  * N                    Frequency Selective Switch
00053  * N                    upsampling filter
00054  * Y                    quantization & coding - AAC
00055  * N                    quantization & coding - TwinVQ
00056  * N                    quantization & coding - BSAC
00057  * N                    AAC Error Resilience tools
00058  * N                    Error Resilience payload syntax
00059  * N                    Error Protection tool
00060  * N                    CELP
00061  * N                    Silence Compression
00062  * N                    HVXC
00063  * N                    HVXC 4kbits/s VR
00064  * N                    Structured Audio tools
00065  * N                    Structured Audio Sample Bank Format
00066  * N                    MIDI
00067  * N                    Harmonic and Individual Lines plus Noise
00068  * N                    Text-To-Speech Interface
00069  * Y                    Spectral Band Replication
00070  * Y (not in this code) Layer-1
00071  * Y (not in this code) Layer-2
00072  * Y (not in this code) Layer-3
00073  * N                    SinuSoidal Coding (Transient, Sinusoid, Noise)
00074  * Y                    Parametric Stereo
00075  * N                    Direct Stream Transfer
00076  *
00077  * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
00078  *       - HE AAC v2 comprises LC AAC with Spectral Band Replication and
00079            Parametric Stereo.
00080  */
00081 
00082 
00083 #include "avcodec.h"
00084 #include "internal.h"
00085 #include "get_bits.h"
00086 #include "dsputil.h"
00087 #include "fft.h"
00088 #include "fmtconvert.h"
00089 #include "lpc.h"
00090 #include "kbdwin.h"
00091 #include "sinewin.h"
00092 
00093 #include "aac.h"
00094 #include "aactab.h"
00095 #include "aacdectab.h"
00096 #include "cbrt_tablegen.h"
00097 #include "sbr.h"
00098 #include "aacsbr.h"
00099 #include "mpeg4audio.h"
00100 #include "aacadtsdec.h"
00101 
00102 #include <assert.h>
00103 #include <errno.h>
00104 #include <math.h>
00105 #include <string.h>
00106 
00107 #if ARCH_ARM
00108 #   include "arm/aac.h"
00109 #endif
00110 
00111 union float754 {
00112     float f;
00113     uint32_t i;
00114 };
00115 
00116 static VLC vlc_scalefactors;
00117 static VLC vlc_spectral[11];
00118 
00119 static const char overread_err[] = "Input buffer exhausted before END element found\n";
00120 
00121 static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
00122 {
00123     // For PCE based channel configurations map the channels solely based on tags.
00124     if (!ac->m4ac.chan_config) {
00125         return ac->tag_che_map[type][elem_id];
00126     }
00127     // For indexed channel configurations map the channels solely based on position.
00128     switch (ac->m4ac.chan_config) {
00129     case 7:
00130         if (ac->tags_mapped == 3 && type == TYPE_CPE) {
00131             ac->tags_mapped++;
00132             return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
00133         }
00134     case 6:
00135         /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
00136            instead of SCE[0] CPE[0] CPE[1] LFE[0]. If we seem to have
00137            encountered such a stream, transfer the LFE[0] element to the SCE[1]'s mapping */
00138         if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
00139             ac->tags_mapped++;
00140             return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
00141         }
00142     case 5:
00143         if (ac->tags_mapped == 2 && type == TYPE_CPE) {
00144             ac->tags_mapped++;
00145             return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
00146         }
00147     case 4:
00148         if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) {
00149             ac->tags_mapped++;
00150             return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
00151         }
00152     case 3:
00153     case 2:
00154         if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) {
00155             ac->tags_mapped++;
00156             return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
00157         } else if (ac->m4ac.chan_config == 2) {
00158             return NULL;
00159         }
00160     case 1:
00161         if (!ac->tags_mapped && type == TYPE_SCE) {
00162             ac->tags_mapped++;
00163             return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
00164         }
00165     default:
00166         return NULL;
00167     }
00168 }
00169 
00182 static av_cold int che_configure(AACContext *ac,
00183                                  enum ChannelPosition che_pos[4][MAX_ELEM_ID],
00184                                  int type, int id, int *channels)
00185 {
00186     if (che_pos[type][id]) {
00187         if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
00188             return AVERROR(ENOMEM);
00189         ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
00190         if (type != TYPE_CCE) {
00191             ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
00192             if (type == TYPE_CPE ||
00193                 (type == TYPE_SCE && ac->m4ac.ps == 1)) {
00194                 ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
00195             }
00196         }
00197     } else {
00198         if (ac->che[type][id])
00199             ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);
00200         av_freep(&ac->che[type][id]);
00201     }
00202     return 0;
00203 }
00204 
00213 static av_cold int output_configure(AACContext *ac,
00214                                     enum ChannelPosition che_pos[4][MAX_ELEM_ID],
00215                                     enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
00216                                     int channel_config, enum OCStatus oc_type)
00217 {
00218     AVCodecContext *avctx = ac->avctx;
00219     int i, type, channels = 0, ret;
00220 
00221     if (new_che_pos != che_pos)
00222     memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
00223 
00224     if (channel_config) {
00225         for (i = 0; i < tags_per_config[channel_config]; i++) {
00226             if ((ret = che_configure(ac, che_pos,
00227                                      aac_channel_layout_map[channel_config - 1][i][0],
00228                                      aac_channel_layout_map[channel_config - 1][i][1],
00229                                      &channels)))
00230                 return ret;
00231         }
00232 
00233         memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
00234 
00235         avctx->channel_layout = aac_channel_layout[channel_config - 1];
00236     } else {
00237         /* Allocate or free elements depending on if they are in the
00238          * current program configuration.
00239          *
00240          * Set up default 1:1 output mapping.
00241          *
00242          * For a 5.1 stream the output order will be:
00243          *    [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
00244          */
00245 
00246         for (i = 0; i < MAX_ELEM_ID; i++) {
00247             for (type = 0; type < 4; type++) {
00248                 if ((ret = che_configure(ac, che_pos, type, i, &channels)))
00249                     return ret;
00250             }
00251         }
00252 
00253         memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
00254     }
00255 
00256     avctx->channels = channels;
00257 
00258     ac->output_configured = oc_type;
00259 
00260     return 0;
00261 }
00262 
00270 static void decode_channel_map(enum ChannelPosition *cpe_map,
00271                                enum ChannelPosition *sce_map,
00272                                enum ChannelPosition type,
00273                                GetBitContext *gb, int n)
00274 {
00275     while (n--) {
00276         enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
00277         map[get_bits(gb, 4)] = type;
00278     }
00279 }
00280 
00288 static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
00289                       enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
00290                       GetBitContext *gb)
00291 {
00292     int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
00293     int comment_len;
00294 
00295     skip_bits(gb, 2);  // object_type
00296 
00297     sampling_index = get_bits(gb, 4);
00298     if (m4ac->sampling_index != sampling_index)
00299         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");
00300 
00301     num_front       = get_bits(gb, 4);
00302     num_side        = get_bits(gb, 4);
00303     num_back        = get_bits(gb, 4);
00304     num_lfe         = get_bits(gb, 2);
00305     num_assoc_data  = get_bits(gb, 3);
00306     num_cc          = get_bits(gb, 4);
00307 
00308     if (get_bits1(gb))
00309         skip_bits(gb, 4); // mono_mixdown_tag
00310     if (get_bits1(gb))
00311         skip_bits(gb, 4); // stereo_mixdown_tag
00312 
00313     if (get_bits1(gb))
00314         skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
00315 
00316     if (get_bits_left(gb) < 4 * (num_front + num_side + num_back + num_lfe + num_assoc_data + num_cc)) {
00317         av_log(avctx, AV_LOG_ERROR, overread_err);
00318         return -1;
00319     }
00320     decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
00321     decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
00322     decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK,  gb, num_back );
00323     decode_channel_map(NULL,                  new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
00324 
00325     skip_bits_long(gb, 4 * num_assoc_data);
00326 
00327     decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
00328 
00329     align_get_bits(gb);
00330 
00331     /* comment field, first byte is length */
00332     comment_len = get_bits(gb, 8) * 8;
00333     if (get_bits_left(gb) < comment_len) {
00334         av_log(avctx, AV_LOG_ERROR, overread_err);
00335         return -1;
00336     }
00337     skip_bits_long(gb, comment_len);
00338     return 0;
00339 }
00340 
00349 static av_cold int set_default_channel_config(AVCodecContext *avctx,
00350                                               enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
00351                                               int channel_config)
00352 {
00353     if (channel_config < 1 || channel_config > 7) {
00354         av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
00355                channel_config);
00356         return -1;
00357     }
00358 
00359     /* default channel configurations:
00360      *
00361      * 1ch : front center (mono)
00362      * 2ch : L + R (stereo)
00363      * 3ch : front center + L + R
00364      * 4ch : front center + L + R + back center
00365      * 5ch : front center + L + R + back stereo
00366      * 6ch : front center + L + R + back stereo + LFE
00367      * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
00368      */
00369 
00370     if (channel_config != 2)
00371         new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
00372     if (channel_config > 1)
00373         new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
00374     if (channel_config == 4)
00375         new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK;  // back center
00376     if (channel_config > 4)
00377         new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
00378         = AAC_CHANNEL_BACK;  // back stereo
00379     if (channel_config > 5)
00380         new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE;   // LFE
00381     if (channel_config == 7)
00382         new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
00383 
00384     return 0;
00385 }
00386 
00395 static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
00396                                      GetBitContext *gb,
00397                                      MPEG4AudioConfig *m4ac,
00398                                      int channel_config)
00399 {
00400     enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
00401     int extension_flag, ret;
00402 
00403     if (get_bits1(gb)) { // frameLengthFlag
00404         av_log_missing_feature(avctx, "960/120 MDCT window is", 1);
00405         return -1;
00406     }
00407 
00408     if (get_bits1(gb))       // dependsOnCoreCoder
00409         skip_bits(gb, 14);   // coreCoderDelay
00410     extension_flag = get_bits1(gb);
00411 
00412     if (m4ac->object_type == AOT_AAC_SCALABLE ||
00413         m4ac->object_type == AOT_ER_AAC_SCALABLE)
00414         skip_bits(gb, 3);     // layerNr
00415 
00416     memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
00417     if (channel_config == 0) {
00418         skip_bits(gb, 4);  // element_instance_tag
00419         if ((ret = decode_pce(avctx, m4ac, new_che_pos, gb)))
00420             return ret;
00421     } else {
00422         if ((ret = set_default_channel_config(avctx, new_che_pos, channel_config)))
00423             return ret;
00424     }
00425     if (ac && (ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
00426         return ret;
00427 
00428     if (extension_flag) {
00429         switch (m4ac->object_type) {
00430         case AOT_ER_BSAC:
00431             skip_bits(gb, 5);    // numOfSubFrame
00432             skip_bits(gb, 11);   // layer_length
00433             break;
00434         case AOT_ER_AAC_LC:
00435         case AOT_ER_AAC_LTP:
00436         case AOT_ER_AAC_SCALABLE:
00437         case AOT_ER_AAC_LD:
00438             skip_bits(gb, 3);  /* aacSectionDataResilienceFlag
00439                                     * aacScalefactorDataResilienceFlag
00440                                     * aacSpectralDataResilienceFlag
00441                                     */
00442             break;
00443         }
00444         skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
00445     }
00446     return 0;
00447 }
00448 
00460 static int decode_audio_specific_config(AACContext *ac,
00461                                         AVCodecContext *avctx,
00462                                         MPEG4AudioConfig *m4ac,
00463                                         const uint8_t *data, int data_size)
00464 {
00465     GetBitContext gb;
00466     int i;
00467 
00468     av_dlog(avctx, "extradata size %d\n", avctx->extradata_size);
00469     for (i = 0; i < avctx->extradata_size; i++)
00470          av_dlog(avctx, "%02x ", avctx->extradata[i]);
00471     av_dlog(avctx, "\n");
00472 
00473     init_get_bits(&gb, data, data_size * 8);
00474 
00475     if ((i = ff_mpeg4audio_get_config(m4ac, data, data_size)) < 0)
00476         return -1;
00477     if (m4ac->sampling_index > 12) {
00478         av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
00479         return -1;
00480     }
00481     if (m4ac->sbr == 1 && m4ac->ps == -1)
00482         m4ac->ps = 1;
00483 
00484     skip_bits_long(&gb, i);
00485 
00486     switch (m4ac->object_type) {
00487     case AOT_AAC_MAIN:
00488     case AOT_AAC_LC:
00489     case AOT_AAC_LTP:
00490         if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config))
00491             return -1;
00492         break;
00493     default:
00494         av_log(avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
00495                m4ac->sbr == 1? "SBR+" : "", m4ac->object_type);
00496         return -1;
00497     }
00498 
00499     av_dlog(avctx, "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
00500             m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
00501             m4ac->sample_rate, m4ac->sbr, m4ac->ps);
00502 
00503     return get_bits_count(&gb);
00504 }
00505 
00513 static av_always_inline int lcg_random(int previous_val)
00514 {
00515     return previous_val * 1664525 + 1013904223;
00516 }
00517 
00518 static av_always_inline void reset_predict_state(PredictorState *ps)
00519 {
00520     ps->r0   = 0.0f;
00521     ps->r1   = 0.0f;
00522     ps->cor0 = 0.0f;
00523     ps->cor1 = 0.0f;
00524     ps->var0 = 1.0f;
00525     ps->var1 = 1.0f;
00526 }
00527 
00528 static void reset_all_predictors(PredictorState *ps)
00529 {
00530     int i;
00531     for (i = 0; i < MAX_PREDICTORS; i++)
00532         reset_predict_state(&ps[i]);
00533 }
00534 
00535 static void reset_predictor_group(PredictorState *ps, int group_num)
00536 {
00537     int i;
00538     for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
00539         reset_predict_state(&ps[i]);
00540 }
00541 
00542 #define AAC_INIT_VLC_STATIC(num, size) \
00543     INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
00544          ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
00545         ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
00546         size);
00547 
00548 static av_cold int aac_decode_init(AVCodecContext *avctx)
00549 {
00550     AACContext *ac = avctx->priv_data;
00551     float output_scale_factor;
00552 
00553     ac->avctx = avctx;
00554     ac->m4ac.sample_rate = avctx->sample_rate;
00555 
00556     if (avctx->extradata_size > 0) {
00557         if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
00558                                          avctx->extradata,
00559                                          avctx->extradata_size) < 0)
00560             return -1;
00561     }
00562 
00563     if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
00564         avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
00565         output_scale_factor = 1.0 / 32768.0;
00566     } else {
00567         avctx->sample_fmt = AV_SAMPLE_FMT_S16;
00568         output_scale_factor = 1.0;
00569     }
00570 
00571     if (avctx->channels > MAX_CHANNELS) {
00572         av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
00573         return AVERROR_INVALIDDATA;
00574     }
00575 
00576     AAC_INIT_VLC_STATIC( 0, 304);
00577     AAC_INIT_VLC_STATIC( 1, 270);
00578     AAC_INIT_VLC_STATIC( 2, 550);
00579     AAC_INIT_VLC_STATIC( 3, 300);
00580     AAC_INIT_VLC_STATIC( 4, 328);
00581     AAC_INIT_VLC_STATIC( 5, 294);
00582     AAC_INIT_VLC_STATIC( 6, 306);
00583     AAC_INIT_VLC_STATIC( 7, 268);
00584     AAC_INIT_VLC_STATIC( 8, 510);
00585     AAC_INIT_VLC_STATIC( 9, 366);
00586     AAC_INIT_VLC_STATIC(10, 462);
00587 
00588     ff_aac_sbr_init();
00589 
00590     dsputil_init(&ac->dsp, avctx);
00591     ff_fmt_convert_init(&ac->fmt_conv, avctx);
00592 
00593     ac->random_state = 0x1f2e3d4c;
00594 
00595     ff_aac_tableinit();
00596 
00597     INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
00598                     ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
00599                     ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
00600                     352);
00601 
00602     ff_mdct_init(&ac->mdct,       11, 1, output_scale_factor/1024.0);
00603     ff_mdct_init(&ac->mdct_small,  8, 1, output_scale_factor/128.0);
00604     ff_mdct_init(&ac->mdct_ltp,   11, 0, -2.0/output_scale_factor);
00605     // window initialization
00606     ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
00607     ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
00608     ff_init_ff_sine_windows(10);
00609     ff_init_ff_sine_windows( 7);
00610 
00611     cbrt_tableinit();
00612 
00613     return 0;
00614 }
00615 
00619 static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
00620 {
00621     int byte_align = get_bits1(gb);
00622     int count = get_bits(gb, 8);
00623     if (count == 255)
00624         count += get_bits(gb, 8);
00625     if (byte_align)
00626         align_get_bits(gb);
00627 
00628     if (get_bits_left(gb) < 8 * count) {
00629         av_log(ac->avctx, AV_LOG_ERROR, overread_err);
00630         return -1;
00631     }
00632     skip_bits_long(gb, 8 * count);
00633     return 0;
00634 }
00635 
00636 static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
00637                              GetBitContext *gb)
00638 {
00639     int sfb;
00640     if (get_bits1(gb)) {
00641         ics->predictor_reset_group = get_bits(gb, 5);
00642         if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
00643             av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
00644             return -1;
00645         }
00646     }
00647     for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
00648         ics->prediction_used[sfb] = get_bits1(gb);
00649     }
00650     return 0;
00651 }
00652 
00656 static void decode_ltp(AACContext *ac, LongTermPrediction *ltp,
00657                        GetBitContext *gb, uint8_t max_sfb)
00658 {
00659     int sfb;
00660 
00661     ltp->lag  = get_bits(gb, 11);
00662     ltp->coef = ltp_coef[get_bits(gb, 3)];
00663     for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
00664         ltp->used[sfb] = get_bits1(gb);
00665 }
00666 
00672 static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
00673                            GetBitContext *gb, int common_window)
00674 {
00675     if (get_bits1(gb)) {
00676         av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
00677         memset(ics, 0, sizeof(IndividualChannelStream));
00678         return -1;
00679     }
00680     ics->window_sequence[1] = ics->window_sequence[0];
00681     ics->window_sequence[0] = get_bits(gb, 2);
00682     ics->use_kb_window[1]   = ics->use_kb_window[0];
00683     ics->use_kb_window[0]   = get_bits1(gb);
00684     ics->num_window_groups  = 1;
00685     ics->group_len[0]       = 1;
00686     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
00687         int i;
00688         ics->max_sfb = get_bits(gb, 4);
00689         for (i = 0; i < 7; i++) {
00690             if (get_bits1(gb)) {
00691                 ics->group_len[ics->num_window_groups - 1]++;
00692             } else {
00693                 ics->num_window_groups++;
00694                 ics->group_len[ics->num_window_groups - 1] = 1;
00695             }
00696         }
00697         ics->num_windows       = 8;
00698         ics->swb_offset        =    ff_swb_offset_128[ac->m4ac.sampling_index];
00699         ics->num_swb           =   ff_aac_num_swb_128[ac->m4ac.sampling_index];
00700         ics->tns_max_bands     = ff_tns_max_bands_128[ac->m4ac.sampling_index];
00701         ics->predictor_present = 0;
00702     } else {
00703         ics->max_sfb               = get_bits(gb, 6);
00704         ics->num_windows           = 1;
00705         ics->swb_offset            =    ff_swb_offset_1024[ac->m4ac.sampling_index];
00706         ics->num_swb               =   ff_aac_num_swb_1024[ac->m4ac.sampling_index];
00707         ics->tns_max_bands         = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
00708         ics->predictor_present     = get_bits1(gb);
00709         ics->predictor_reset_group = 0;
00710         if (ics->predictor_present) {
00711             if (ac->m4ac.object_type == AOT_AAC_MAIN) {
00712                 if (decode_prediction(ac, ics, gb)) {
00713                     memset(ics, 0, sizeof(IndividualChannelStream));
00714                     return -1;
00715                 }
00716             } else if (ac->m4ac.object_type == AOT_AAC_LC) {
00717                 av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
00718                 memset(ics, 0, sizeof(IndividualChannelStream));
00719                 return -1;
00720             } else {
00721                 if ((ics->ltp.present = get_bits(gb, 1)))
00722                     decode_ltp(ac, &ics->ltp, gb, ics->max_sfb);
00723             }
00724         }
00725     }
00726 
00727     if (ics->max_sfb > ics->num_swb) {
00728         av_log(ac->avctx, AV_LOG_ERROR,
00729                "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
00730                ics->max_sfb, ics->num_swb);
00731         memset(ics, 0, sizeof(IndividualChannelStream));
00732         return -1;
00733     }
00734 
00735     return 0;
00736 }
00737 
00746 static int decode_band_types(AACContext *ac, enum BandType band_type[120],
00747                              int band_type_run_end[120], GetBitContext *gb,
00748                              IndividualChannelStream *ics)
00749 {
00750     int g, idx = 0;
00751     const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
00752     for (g = 0; g < ics->num_window_groups; g++) {
00753         int k = 0;
00754         while (k < ics->max_sfb) {
00755             uint8_t sect_end = k;
00756             int sect_len_incr;
00757             int sect_band_type = get_bits(gb, 4);
00758             if (sect_band_type == 12) {
00759                 av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
00760                 return -1;
00761             }
00762             do {
00763                 sect_len_incr = get_bits(gb, bits);
00764                 sect_end += sect_len_incr;
00765                 if (get_bits_left(gb) < 0) {
00766                     av_log(ac->avctx, AV_LOG_ERROR, overread_err);
00767                     return -1;
00768                 }
00769                 if (sect_end > ics->max_sfb) {
00770                     av_log(ac->avctx, AV_LOG_ERROR,
00771                            "Number of bands (%d) exceeds limit (%d).\n",
00772                            sect_end, ics->max_sfb);
00773                     return -1;
00774                 }
00775             } while (sect_len_incr == (1 << bits) - 1);
00776             for (; k < sect_end; k++) {
00777                 band_type        [idx]   = sect_band_type;
00778                 band_type_run_end[idx++] = sect_end;
00779             }
00780         }
00781     }
00782     return 0;
00783 }
00784 
00795 static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
00796                                unsigned int global_gain,
00797                                IndividualChannelStream *ics,
00798                                enum BandType band_type[120],
00799                                int band_type_run_end[120])
00800 {
00801     int g, i, idx = 0;
00802     int offset[3] = { global_gain, global_gain - 90, 0 };
00803     int clipped_offset;
00804     int noise_flag = 1;
00805     static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
00806     for (g = 0; g < ics->num_window_groups; g++) {
00807         for (i = 0; i < ics->max_sfb;) {
00808             int run_end = band_type_run_end[idx];
00809             if (band_type[idx] == ZERO_BT) {
00810                 for (; i < run_end; i++, idx++)
00811                     sf[idx] = 0.;
00812             } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
00813                 for (; i < run_end; i++, idx++) {
00814                     offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
00815                     clipped_offset = av_clip(offset[2], -155, 100);
00816                     if (offset[2] != clipped_offset) {
00817                         av_log_ask_for_sample(ac->avctx, "Intensity stereo "
00818                                 "position clipped (%d -> %d).\nIf you heard an "
00819                                 "audible artifact, there may be a bug in the "
00820                                 "decoder. ", offset[2], clipped_offset);
00821                     }
00822                     sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
00823                 }
00824             } else if (band_type[idx] == NOISE_BT) {
00825                 for (; i < run_end; i++, idx++) {
00826                     if (noise_flag-- > 0)
00827                         offset[1] += get_bits(gb, 9) - 256;
00828                     else
00829                         offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
00830                     clipped_offset = av_clip(offset[1], -100, 155);
00831                     if (offset[1] != clipped_offset) {
00832                         av_log_ask_for_sample(ac->avctx, "Noise gain clipped "
00833                                 "(%d -> %d).\nIf you heard an audible "
00834                                 "artifact, there may be a bug in the decoder. ",
00835                                 offset[1], clipped_offset);
00836                     }
00837                     sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
00838                 }
00839             } else {
00840                 for (; i < run_end; i++, idx++) {
00841                     offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
00842                     if (offset[0] > 255U) {
00843                         av_log(ac->avctx, AV_LOG_ERROR,
00844                                "%s (%d) out of range.\n", sf_str[0], offset[0]);
00845                         return -1;
00846                     }
00847                     sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
00848                 }
00849             }
00850         }
00851     }
00852     return 0;
00853 }
00854 
00858 static int decode_pulses(Pulse *pulse, GetBitContext *gb,
00859                          const uint16_t *swb_offset, int num_swb)
00860 {
00861     int i, pulse_swb;
00862     pulse->num_pulse = get_bits(gb, 2) + 1;
00863     pulse_swb        = get_bits(gb, 6);
00864     if (pulse_swb >= num_swb)
00865         return -1;
00866     pulse->pos[0]    = swb_offset[pulse_swb];
00867     pulse->pos[0]   += get_bits(gb, 5);
00868     if (pulse->pos[0] > 1023)
00869         return -1;
00870     pulse->amp[0]    = get_bits(gb, 4);
00871     for (i = 1; i < pulse->num_pulse; i++) {
00872         pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
00873         if (pulse->pos[i] > 1023)
00874             return -1;
00875         pulse->amp[i] = get_bits(gb, 4);
00876     }
00877     return 0;
00878 }
00879 
00885 static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
00886                       GetBitContext *gb, const IndividualChannelStream *ics)
00887 {
00888     int w, filt, i, coef_len, coef_res, coef_compress;
00889     const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
00890     const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
00891     for (w = 0; w < ics->num_windows; w++) {
00892         if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
00893             coef_res = get_bits1(gb);
00894 
00895             for (filt = 0; filt < tns->n_filt[w]; filt++) {
00896                 int tmp2_idx;
00897                 tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
00898 
00899                 if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
00900                     av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
00901                            tns->order[w][filt], tns_max_order);
00902                     tns->order[w][filt] = 0;
00903                     return -1;
00904                 }
00905                 if (tns->order[w][filt]) {
00906                     tns->direction[w][filt] = get_bits1(gb);
00907                     coef_compress = get_bits1(gb);
00908                     coef_len = coef_res + 3 - coef_compress;
00909                     tmp2_idx = 2 * coef_compress + coef_res;
00910 
00911                     for (i = 0; i < tns->order[w][filt]; i++)
00912                         tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
00913                 }
00914             }
00915         }
00916     }
00917     return 0;
00918 }
00919 
00927 static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
00928                                    int ms_present)
00929 {
00930     int idx;
00931     if (ms_present == 1) {
00932         for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
00933             cpe->ms_mask[idx] = get_bits1(gb);
00934     } else if (ms_present == 2) {
00935         memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
00936     }
00937 }
00938 
00939 #ifndef VMUL2
00940 static inline float *VMUL2(float *dst, const float *v, unsigned idx,
00941                            const float *scale)
00942 {
00943     float s = *scale;
00944     *dst++ = v[idx    & 15] * s;
00945     *dst++ = v[idx>>4 & 15] * s;
00946     return dst;
00947 }
00948 #endif
00949 
00950 #ifndef VMUL4
00951 static inline float *VMUL4(float *dst, const float *v, unsigned idx,
00952                            const float *scale)
00953 {
00954     float s = *scale;
00955     *dst++ = v[idx    & 3] * s;
00956     *dst++ = v[idx>>2 & 3] * s;
00957     *dst++ = v[idx>>4 & 3] * s;
00958     *dst++ = v[idx>>6 & 3] * s;
00959     return dst;
00960 }
00961 #endif
00962 
00963 #ifndef VMUL2S
00964 static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
00965                             unsigned sign, const float *scale)
00966 {
00967     union float754 s0, s1;
00968 
00969     s0.f = s1.f = *scale;
00970     s0.i ^= sign >> 1 << 31;
00971     s1.i ^= sign      << 31;
00972 
00973     *dst++ = v[idx    & 15] * s0.f;
00974     *dst++ = v[idx>>4 & 15] * s1.f;
00975 
00976     return dst;
00977 }
00978 #endif
00979 
00980 #ifndef VMUL4S
00981 static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
00982                             unsigned sign, const float *scale)
00983 {
00984     unsigned nz = idx >> 12;
00985     union float754 s = { .f = *scale };
00986     union float754 t;
00987 
00988     t.i = s.i ^ (sign & 1U<<31);
00989     *dst++ = v[idx    & 3] * t.f;
00990 
00991     sign <<= nz & 1; nz >>= 1;
00992     t.i = s.i ^ (sign & 1U<<31);
00993     *dst++ = v[idx>>2 & 3] * t.f;
00994 
00995     sign <<= nz & 1; nz >>= 1;
00996     t.i = s.i ^ (sign & 1U<<31);
00997     *dst++ = v[idx>>4 & 3] * t.f;
00998 
00999     sign <<= nz & 1; nz >>= 1;
01000     t.i = s.i ^ (sign & 1U<<31);
01001     *dst++ = v[idx>>6 & 3] * t.f;
01002 
01003     return dst;
01004 }
01005 #endif
01006 
01019 static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
01020                                        GetBitContext *gb, const float sf[120],
01021                                        int pulse_present, const Pulse *pulse,
01022                                        const IndividualChannelStream *ics,
01023                                        enum BandType band_type[120])
01024 {
01025     int i, k, g, idx = 0;
01026     const int c = 1024 / ics->num_windows;
01027     const uint16_t *offsets = ics->swb_offset;
01028     float *coef_base = coef;
01029 
01030     for (g = 0; g < ics->num_windows; g++)
01031         memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
01032 
01033     for (g = 0; g < ics->num_window_groups; g++) {
01034         unsigned g_len = ics->group_len[g];
01035 
01036         for (i = 0; i < ics->max_sfb; i++, idx++) {
01037             const unsigned cbt_m1 = band_type[idx] - 1;
01038             float *cfo = coef + offsets[i];
01039             int off_len = offsets[i + 1] - offsets[i];
01040             int group;
01041 
01042             if (cbt_m1 >= INTENSITY_BT2 - 1) {
01043                 for (group = 0; group < g_len; group++, cfo+=128) {
01044                     memset(cfo, 0, off_len * sizeof(float));
01045                 }
01046             } else if (cbt_m1 == NOISE_BT - 1) {
01047                 for (group = 0; group < g_len; group++, cfo+=128) {
01048                     float scale;
01049                     float band_energy;
01050 
01051                     for (k = 0; k < off_len; k++) {
01052                         ac->random_state  = lcg_random(ac->random_state);
01053                         cfo[k] = ac->random_state;
01054                     }
01055 
01056                     band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
01057                     scale = sf[idx] / sqrtf(band_energy);
01058                     ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
01059                 }
01060             } else {
01061                 const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
01062                 const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
01063                 VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
01064                 OPEN_READER(re, gb);
01065 
01066                 switch (cbt_m1 >> 1) {
01067                 case 0:
01068                     for (group = 0; group < g_len; group++, cfo+=128) {
01069                         float *cf = cfo;
01070                         int len = off_len;
01071 
01072                         do {
01073                             int code;
01074                             unsigned cb_idx;
01075 
01076                             UPDATE_CACHE(re, gb);
01077                             GET_VLC(code, re, gb, vlc_tab, 8, 2);
01078                             cb_idx = cb_vector_idx[code];
01079                             cf = VMUL4(cf, vq, cb_idx, sf + idx);
01080                         } while (len -= 4);
01081                     }
01082                     break;
01083 
01084                 case 1:
01085                     for (group = 0; group < g_len; group++, cfo+=128) {
01086                         float *cf = cfo;
01087                         int len = off_len;
01088 
01089                         do {
01090                             int code;
01091                             unsigned nnz;
01092                             unsigned cb_idx;
01093                             uint32_t bits;
01094 
01095                             UPDATE_CACHE(re, gb);
01096                             GET_VLC(code, re, gb, vlc_tab, 8, 2);
01097                             cb_idx = cb_vector_idx[code];
01098                             nnz = cb_idx >> 8 & 15;
01099                             bits = nnz ? GET_CACHE(re, gb) : 0;
01100                             LAST_SKIP_BITS(re, gb, nnz);
01101                             cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
01102                         } while (len -= 4);
01103                     }
01104                     break;
01105 
01106                 case 2:
01107                     for (group = 0; group < g_len; group++, cfo+=128) {
01108                         float *cf = cfo;
01109                         int len = off_len;
01110 
01111                         do {
01112                             int code;
01113                             unsigned cb_idx;
01114 
01115                             UPDATE_CACHE(re, gb);
01116                             GET_VLC(code, re, gb, vlc_tab, 8, 2);
01117                             cb_idx = cb_vector_idx[code];
01118                             cf = VMUL2(cf, vq, cb_idx, sf + idx);
01119                         } while (len -= 2);
01120                     }
01121                     break;
01122 
01123                 case 3:
01124                 case 4:
01125                     for (group = 0; group < g_len; group++, cfo+=128) {
01126                         float *cf = cfo;
01127                         int len = off_len;
01128 
01129                         do {
01130                             int code;
01131                             unsigned nnz;
01132                             unsigned cb_idx;
01133                             unsigned sign;
01134 
01135                             UPDATE_CACHE(re, gb);
01136                             GET_VLC(code, re, gb, vlc_tab, 8, 2);
01137                             cb_idx = cb_vector_idx[code];
01138                             nnz = cb_idx >> 8 & 15;
01139                             sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
01140                             LAST_SKIP_BITS(re, gb, nnz);
01141                             cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
01142                         } while (len -= 2);
01143                     }
01144                     break;
01145 
01146                 default:
01147                     for (group = 0; group < g_len; group++, cfo+=128) {
01148                         float *cf = cfo;
01149                         uint32_t *icf = (uint32_t *) cf;
01150                         int len = off_len;
01151 
01152                         do {
01153                             int code;
01154                             unsigned nzt, nnz;
01155                             unsigned cb_idx;
01156                             uint32_t bits;
01157                             int j;
01158 
01159                             UPDATE_CACHE(re, gb);
01160                             GET_VLC(code, re, gb, vlc_tab, 8, 2);
01161 
01162                             if (!code) {
01163                                 *icf++ = 0;
01164                                 *icf++ = 0;
01165                                 continue;
01166                             }
01167 
01168                             cb_idx = cb_vector_idx[code];
01169                             nnz = cb_idx >> 12;
01170                             nzt = cb_idx >> 8;
01171                             bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
01172                             LAST_SKIP_BITS(re, gb, nnz);
01173 
01174                             for (j = 0; j < 2; j++) {
01175                                 if (nzt & 1<<j) {
01176                                     uint32_t b;
01177                                     int n;
01178                                     /* The total length of escape_sequence must be < 22 bits according
01179                                        to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
01180                                     UPDATE_CACHE(re, gb);
01181                                     b = GET_CACHE(re, gb);
01182                                     b = 31 - av_log2(~b);
01183 
01184                                     if (b > 8) {
01185                                         av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
01186                                         return -1;
01187                                     }
01188 
01189                                     SKIP_BITS(re, gb, b + 1);
01190                                     b += 4;
01191                                     n = (1 << b) + SHOW_UBITS(re, gb, b);
01192                                     LAST_SKIP_BITS(re, gb, b);
01193                                     *icf++ = cbrt_tab[n] | (bits & 1U<<31);
01194                                     bits <<= 1;
01195                                 } else {
01196                                     unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
01197                                     *icf++ = (bits & 1U<<31) | v;
01198                                     bits <<= !!v;
01199                                 }
01200                                 cb_idx >>= 4;
01201                             }
01202                         } while (len -= 2);
01203 
01204                         ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
01205                     }
01206                 }
01207 
01208                 CLOSE_READER(re, gb);
01209             }
01210         }
01211         coef += g_len << 7;
01212     }
01213 
01214     if (pulse_present) {
01215         idx = 0;
01216         for (i = 0; i < pulse->num_pulse; i++) {
01217             float co = coef_base[ pulse->pos[i] ];
01218             while (offsets[idx + 1] <= pulse->pos[i])
01219                 idx++;
01220             if (band_type[idx] != NOISE_BT && sf[idx]) {
01221                 float ico = -pulse->amp[i];
01222                 if (co) {
01223                     co /= sf[idx];
01224                     ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
01225                 }
01226                 coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
01227             }
01228         }
01229     }
01230     return 0;
01231 }
01232 
01233 static av_always_inline float flt16_round(float pf)
01234 {
01235     union float754 tmp;
01236     tmp.f = pf;
01237     tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
01238     return tmp.f;
01239 }
01240 
01241 static av_always_inline float flt16_even(float pf)
01242 {
01243     union float754 tmp;
01244     tmp.f = pf;
01245     tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
01246     return tmp.f;
01247 }
01248 
01249 static av_always_inline float flt16_trunc(float pf)
01250 {
01251     union float754 pun;
01252     pun.f = pf;
01253     pun.i &= 0xFFFF0000U;
01254     return pun.f;
01255 }
01256 
01257 static av_always_inline void predict(PredictorState *ps, float *coef,
01258                                      int output_enable)
01259 {
01260     const float a     = 0.953125; // 61.0 / 64
01261     const float alpha = 0.90625;  // 29.0 / 32
01262     float e0, e1;
01263     float pv;
01264     float k1, k2;
01265     float   r0 = ps->r0,     r1 = ps->r1;
01266     float cor0 = ps->cor0, cor1 = ps->cor1;
01267     float var0 = ps->var0, var1 = ps->var1;
01268 
01269     k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
01270     k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
01271 
01272     pv = flt16_round(k1 * r0 + k2 * r1);
01273     if (output_enable)
01274         *coef += pv;
01275 
01276     e0 = *coef;
01277     e1 = e0 - k1 * r0;
01278 
01279     ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
01280     ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
01281     ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
01282     ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
01283 
01284     ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
01285     ps->r0 = flt16_trunc(a * e0);
01286 }
01287 
01291 static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
01292 {
01293     int sfb, k;
01294 
01295     if (!sce->ics.predictor_initialized) {
01296         reset_all_predictors(sce->predictor_state);
01297         sce->ics.predictor_initialized = 1;
01298     }
01299 
01300     if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
01301         for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
01302             for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
01303                 predict(&sce->predictor_state[k], &sce->coeffs[k],
01304                         sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
01305             }
01306         }
01307         if (sce->ics.predictor_reset_group)
01308             reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
01309     } else
01310         reset_all_predictors(sce->predictor_state);
01311 }
01312 
01321 static int decode_ics(AACContext *ac, SingleChannelElement *sce,
01322                       GetBitContext *gb, int common_window, int scale_flag)
01323 {
01324     Pulse pulse;
01325     TemporalNoiseShaping    *tns = &sce->tns;
01326     IndividualChannelStream *ics = &sce->ics;
01327     float *out = sce->coeffs;
01328     int global_gain, pulse_present = 0;
01329 
01330     /* This assignment is to silence a GCC warning about the variable being used
01331      * uninitialized when in fact it always is.
01332      */
01333     pulse.num_pulse = 0;
01334 
01335     global_gain = get_bits(gb, 8);
01336 
01337     if (!common_window && !scale_flag) {
01338         if (decode_ics_info(ac, ics, gb, 0) < 0)
01339             return -1;
01340     }
01341 
01342     if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
01343         return -1;
01344     if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
01345         return -1;
01346 
01347     pulse_present = 0;
01348     if (!scale_flag) {
01349         if ((pulse_present = get_bits1(gb))) {
01350             if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
01351                 av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
01352                 return -1;
01353             }
01354             if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
01355                 av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
01356                 return -1;
01357             }
01358         }
01359         if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
01360             return -1;
01361         if (get_bits1(gb)) {
01362             av_log_missing_feature(ac->avctx, "SSR", 1);
01363             return -1;
01364         }
01365     }
01366 
01367     if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
01368         return -1;
01369 
01370     if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
01371         apply_prediction(ac, sce);
01372 
01373     return 0;
01374 }
01375 
01379 static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
01380 {
01381     const IndividualChannelStream *ics = &cpe->ch[0].ics;
01382     float *ch0 = cpe->ch[0].coeffs;
01383     float *ch1 = cpe->ch[1].coeffs;
01384     int g, i, group, idx = 0;
01385     const uint16_t *offsets = ics->swb_offset;
01386     for (g = 0; g < ics->num_window_groups; g++) {
01387         for (i = 0; i < ics->max_sfb; i++, idx++) {
01388             if (cpe->ms_mask[idx] &&
01389                     cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
01390                 for (group = 0; group < ics->group_len[g]; group++) {
01391                     ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
01392                                               ch1 + group * 128 + offsets[i],
01393                                               offsets[i+1] - offsets[i]);
01394                 }
01395             }
01396         }
01397         ch0 += ics->group_len[g] * 128;
01398         ch1 += ics->group_len[g] * 128;
01399     }
01400 }
01401 
01409 static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present)
01410 {
01411     const IndividualChannelStream *ics = &cpe->ch[1].ics;
01412     SingleChannelElement         *sce1 = &cpe->ch[1];
01413     float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
01414     const uint16_t *offsets = ics->swb_offset;
01415     int g, group, i, idx = 0;
01416     int c;
01417     float scale;
01418     for (g = 0; g < ics->num_window_groups; g++) {
01419         for (i = 0; i < ics->max_sfb;) {
01420             if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
01421                 const int bt_run_end = sce1->band_type_run_end[idx];
01422                 for (; i < bt_run_end; i++, idx++) {
01423                     c = -1 + 2 * (sce1->band_type[idx] - 14);
01424                     if (ms_present)
01425                         c *= 1 - 2 * cpe->ms_mask[idx];
01426                     scale = c * sce1->sf[idx];
01427                     for (group = 0; group < ics->group_len[g]; group++)
01428                         ac->dsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
01429                                                    coef0 + group * 128 + offsets[i],
01430                                                    scale,
01431                                                    offsets[i + 1] - offsets[i]);
01432                 }
01433             } else {
01434                 int bt_run_end = sce1->band_type_run_end[idx];
01435                 idx += bt_run_end - i;
01436                 i    = bt_run_end;
01437             }
01438         }
01439         coef0 += ics->group_len[g] * 128;
01440         coef1 += ics->group_len[g] * 128;
01441     }
01442 }
01443 
01449 static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
01450 {
01451     int i, ret, common_window, ms_present = 0;
01452 
01453     common_window = get_bits1(gb);
01454     if (common_window) {
01455         if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
01456             return -1;
01457         i = cpe->ch[1].ics.use_kb_window[0];
01458         cpe->ch[1].ics = cpe->ch[0].ics;
01459         cpe->ch[1].ics.use_kb_window[1] = i;
01460         if (cpe->ch[1].ics.predictor_present && (ac->m4ac.object_type != AOT_AAC_MAIN))
01461             if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
01462                 decode_ltp(ac, &cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
01463         ms_present = get_bits(gb, 2);
01464         if (ms_present == 3) {
01465             av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
01466             return -1;
01467         } else if (ms_present)
01468             decode_mid_side_stereo(cpe, gb, ms_present);
01469     }
01470     if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
01471         return ret;
01472     if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
01473         return ret;
01474 
01475     if (common_window) {
01476         if (ms_present)
01477             apply_mid_side_stereo(ac, cpe);
01478         if (ac->m4ac.object_type == AOT_AAC_MAIN) {
01479             apply_prediction(ac, &cpe->ch[0]);
01480             apply_prediction(ac, &cpe->ch[1]);
01481         }
01482     }
01483 
01484     apply_intensity_stereo(ac, cpe, ms_present);
01485     return 0;
01486 }
01487 
01488 static const float cce_scale[] = {
01489     1.09050773266525765921, //2^(1/8)
01490     1.18920711500272106672, //2^(1/4)
01491     M_SQRT2,
01492     2,
01493 };
01494 
01500 static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
01501 {
01502     int num_gain = 0;
01503     int c, g, sfb, ret;
01504     int sign;
01505     float scale;
01506     SingleChannelElement *sce = &che->ch[0];
01507     ChannelCoupling     *coup = &che->coup;
01508 
01509     coup->coupling_point = 2 * get_bits1(gb);
01510     coup->num_coupled = get_bits(gb, 3);
01511     for (c = 0; c <= coup->num_coupled; c++) {
01512         num_gain++;
01513         coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
01514         coup->id_select[c] = get_bits(gb, 4);
01515         if (coup->type[c] == TYPE_CPE) {
01516             coup->ch_select[c] = get_bits(gb, 2);
01517             if (coup->ch_select[c] == 3)
01518                 num_gain++;
01519         } else
01520             coup->ch_select[c] = 2;
01521     }
01522     coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
01523 
01524     sign  = get_bits(gb, 1);
01525     scale = cce_scale[get_bits(gb, 2)];
01526 
01527     if ((ret = decode_ics(ac, sce, gb, 0, 0)))
01528         return ret;
01529 
01530     for (c = 0; c < num_gain; c++) {
01531         int idx  = 0;
01532         int cge  = 1;
01533         int gain = 0;
01534         float gain_cache = 1.;
01535         if (c) {
01536             cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
01537             gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
01538             gain_cache = powf(scale, -gain);
01539         }
01540         if (coup->coupling_point == AFTER_IMDCT) {
01541             coup->gain[c][0] = gain_cache;
01542         } else {
01543             for (g = 0; g < sce->ics.num_window_groups; g++) {
01544                 for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
01545                     if (sce->band_type[idx] != ZERO_BT) {
01546                         if (!cge) {
01547                             int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
01548                             if (t) {
01549                                 int s = 1;
01550                                 t = gain += t;
01551                                 if (sign) {
01552                                     s  -= 2 * (t & 0x1);
01553                                     t >>= 1;
01554                                 }
01555                                 gain_cache = powf(scale, -t) * s;
01556                             }
01557                         }
01558                         coup->gain[c][idx] = gain_cache;
01559                     }
01560                 }
01561             }
01562         }
01563     }
01564     return 0;
01565 }
01566 
01572 static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
01573                                          GetBitContext *gb)
01574 {
01575     int i;
01576     int num_excl_chan = 0;
01577 
01578     do {
01579         for (i = 0; i < 7; i++)
01580             che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
01581     } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
01582 
01583     return num_excl_chan / 7;
01584 }
01585 
01593 static int decode_dynamic_range(DynamicRangeControl *che_drc,
01594                                 GetBitContext *gb, int cnt)
01595 {
01596     int n             = 1;
01597     int drc_num_bands = 1;
01598     int i;
01599 
01600     /* pce_tag_present? */
01601     if (get_bits1(gb)) {
01602         che_drc->pce_instance_tag  = get_bits(gb, 4);
01603         skip_bits(gb, 4); // tag_reserved_bits
01604         n++;
01605     }
01606 
01607     /* excluded_chns_present? */
01608     if (get_bits1(gb)) {
01609         n += decode_drc_channel_exclusions(che_drc, gb);
01610     }
01611 
01612     /* drc_bands_present? */
01613     if (get_bits1(gb)) {
01614         che_drc->band_incr            = get_bits(gb, 4);
01615         che_drc->interpolation_scheme = get_bits(gb, 4);
01616         n++;
01617         drc_num_bands += che_drc->band_incr;
01618         for (i = 0; i < drc_num_bands; i++) {
01619             che_drc->band_top[i] = get_bits(gb, 8);
01620             n++;
01621         }
01622     }
01623 
01624     /* prog_ref_level_present? */
01625     if (get_bits1(gb)) {
01626         che_drc->prog_ref_level = get_bits(gb, 7);
01627         skip_bits1(gb); // prog_ref_level_reserved_bits
01628         n++;
01629     }
01630 
01631     for (i = 0; i < drc_num_bands; i++) {
01632         che_drc->dyn_rng_sgn[i] = get_bits1(gb);
01633         che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
01634         n++;
01635     }
01636 
01637     return n;
01638 }
01639 
01647 static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
01648                                     ChannelElement *che, enum RawDataBlockType elem_type)
01649 {
01650     int crc_flag = 0;
01651     int res = cnt;
01652     switch (get_bits(gb, 4)) { // extension type
01653     case EXT_SBR_DATA_CRC:
01654         crc_flag++;
01655     case EXT_SBR_DATA:
01656         if (!che) {
01657             av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
01658             return res;
01659         } else if (!ac->m4ac.sbr) {
01660             av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
01661             skip_bits_long(gb, 8 * cnt - 4);
01662             return res;
01663         } else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) {
01664             av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
01665             skip_bits_long(gb, 8 * cnt - 4);
01666             return res;
01667         } else if (ac->m4ac.ps == -1 && ac->output_configured < OC_LOCKED && ac->avctx->channels == 1) {
01668             ac->m4ac.sbr = 1;
01669             ac->m4ac.ps = 1;
01670             output_configure(ac, ac->che_pos, ac->che_pos, ac->m4ac.chan_config, ac->output_configured);
01671         } else {
01672             ac->m4ac.sbr = 1;
01673         }
01674         res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
01675         break;
01676     case EXT_DYNAMIC_RANGE:
01677         res = decode_dynamic_range(&ac->che_drc, gb, cnt);
01678         break;
01679     case EXT_FILL:
01680     case EXT_FILL_DATA:
01681     case EXT_DATA_ELEMENT:
01682     default:
01683         skip_bits_long(gb, 8 * cnt - 4);
01684         break;
01685     };
01686     return res;
01687 }
01688 
01695 static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
01696                       IndividualChannelStream *ics, int decode)
01697 {
01698     const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
01699     int w, filt, m, i;
01700     int bottom, top, order, start, end, size, inc;
01701     float lpc[TNS_MAX_ORDER];
01702     float tmp[TNS_MAX_ORDER + 1];
01703 
01704     for (w = 0; w < ics->num_windows; w++) {
01705         bottom = ics->num_swb;
01706         for (filt = 0; filt < tns->n_filt[w]; filt++) {
01707             top    = bottom;
01708             bottom = FFMAX(0, top - tns->length[w][filt]);
01709             order  = tns->order[w][filt];
01710             if (order == 0)
01711                 continue;
01712 
01713             // tns_decode_coef
01714             compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
01715 
01716             start = ics->swb_offset[FFMIN(bottom, mmm)];
01717             end   = ics->swb_offset[FFMIN(   top, mmm)];
01718             if ((size = end - start) <= 0)
01719                 continue;
01720             if (tns->direction[w][filt]) {
01721                 inc = -1;
01722                 start = end - 1;
01723             } else {
01724                 inc = 1;
01725             }
01726             start += w * 128;
01727 
01728             if (decode) {
01729                 // ar filter
01730                 for (m = 0; m < size; m++, start += inc)
01731                     for (i = 1; i <= FFMIN(m, order); i++)
01732                         coef[start] -= coef[start - i * inc] * lpc[i - 1];
01733             } else {
01734                 // ma filter
01735                 for (m = 0; m < size; m++, start += inc) {
01736                     tmp[0] = coef[start];
01737                     for (i = 1; i <= FFMIN(m, order); i++)
01738                         coef[start] += tmp[i] * lpc[i - 1];
01739                     for (i = order; i > 0; i--)
01740                         tmp[i] = tmp[i - 1];
01741                 }
01742             }
01743         }
01744     }
01745 }
01746 
01751 static void windowing_and_mdct_ltp(AACContext *ac, float *out,
01752                                    float *in, IndividualChannelStream *ics)
01753 {
01754     const float *lwindow      = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
01755     const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
01756     const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
01757     const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
01758 
01759     if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
01760         ac->dsp.vector_fmul(in, in, lwindow_prev, 1024);
01761     } else {
01762         memset(in, 0, 448 * sizeof(float));
01763         ac->dsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
01764     }
01765     if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
01766         ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
01767     } else {
01768         ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
01769         memset(in + 1024 + 576, 0, 448 * sizeof(float));
01770     }
01771     ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
01772 }
01773 
01777 static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
01778 {
01779     const LongTermPrediction *ltp = &sce->ics.ltp;
01780     const uint16_t *offsets = sce->ics.swb_offset;
01781     int i, sfb;
01782 
01783     if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
01784         float *predTime = sce->ret;
01785         float *predFreq = ac->buf_mdct;
01786         int16_t num_samples = 2048;
01787 
01788         if (ltp->lag < 1024)
01789             num_samples = ltp->lag + 1024;
01790         for (i = 0; i < num_samples; i++)
01791             predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
01792         memset(&predTime[i], 0, (2048 - i) * sizeof(float));
01793 
01794         windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
01795 
01796         if (sce->tns.present)
01797             apply_tns(predFreq, &sce->tns, &sce->ics, 0);
01798 
01799         for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
01800             if (ltp->used[sfb])
01801                 for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
01802                     sce->coeffs[i] += predFreq[i];
01803     }
01804 }
01805 
01809 static void update_ltp(AACContext *ac, SingleChannelElement *sce)
01810 {
01811     IndividualChannelStream *ics = &sce->ics;
01812     float *saved     = sce->saved;
01813     float *saved_ltp = sce->coeffs;
01814     const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
01815     const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
01816     int i;
01817 
01818     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
01819         memcpy(saved_ltp,       saved, 512 * sizeof(float));
01820         memset(saved_ltp + 576, 0,     448 * sizeof(float));
01821         ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
01822         for (i = 0; i < 64; i++)
01823             saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
01824     } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
01825         memcpy(saved_ltp,       ac->buf_mdct + 512, 448 * sizeof(float));
01826         memset(saved_ltp + 576, 0,                  448 * sizeof(float));
01827         ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
01828         for (i = 0; i < 64; i++)
01829             saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
01830     } else { // LONG_STOP or ONLY_LONG
01831         ac->dsp.vector_fmul_reverse(saved_ltp,       ac->buf_mdct + 512,     &lwindow[512],     512);
01832         for (i = 0; i < 512; i++)
01833             saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
01834     }
01835 
01836     memcpy(sce->ltp_state,      sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
01837     memcpy(sce->ltp_state+1024, sce->ret,            1024 * sizeof(*sce->ltp_state));
01838     memcpy(sce->ltp_state+2048, saved_ltp,           1024 * sizeof(*sce->ltp_state));
01839 }
01840 
01844 static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
01845 {
01846     IndividualChannelStream *ics = &sce->ics;
01847     float *in    = sce->coeffs;
01848     float *out   = sce->ret;
01849     float *saved = sce->saved;
01850     const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
01851     const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
01852     const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
01853     float *buf  = ac->buf_mdct;
01854     float *temp = ac->temp;
01855     int i;
01856 
01857     // imdct
01858     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
01859         for (i = 0; i < 1024; i += 128)
01860             ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
01861     } else
01862         ac->mdct.imdct_half(&ac->mdct, buf, in);
01863 
01864     /* window overlapping
01865      * NOTE: To simplify the overlapping code, all 'meaningless' short to long
01866      * and long to short transitions are considered to be short to short
01867      * transitions. This leaves just two cases (long to long and short to short)
01868      * with a little special sauce for EIGHT_SHORT_SEQUENCE.
01869      */
01870     if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
01871             (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
01872         ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);
01873     } else {
01874         memcpy(                        out,               saved,            448 * sizeof(float));
01875 
01876         if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
01877             ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, 64);
01878             ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      64);
01879             ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      64);
01880             ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      64);
01881             ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      64);
01882             memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
01883         } else {
01884             ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);
01885             memcpy(                    out + 576,         buf + 64,         448 * sizeof(float));
01886         }
01887     }
01888 
01889     // buffer update
01890     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
01891         memcpy(                    saved,       temp + 64,         64 * sizeof(float));
01892         ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 64);
01893         ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
01894         ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
01895         memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
01896     } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
01897         memcpy(                    saved,       buf + 512,        448 * sizeof(float));
01898         memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
01899     } else { // LONG_STOP or ONLY_LONG
01900         memcpy(                    saved,       buf + 512,        512 * sizeof(float));
01901     }
01902 }
01903 
01909 static void apply_dependent_coupling(AACContext *ac,
01910                                      SingleChannelElement *target,
01911                                      ChannelElement *cce, int index)
01912 {
01913     IndividualChannelStream *ics = &cce->ch[0].ics;
01914     const uint16_t *offsets = ics->swb_offset;
01915     float *dest = target->coeffs;
01916     const float *src = cce->ch[0].coeffs;
01917     int g, i, group, k, idx = 0;
01918     if (ac->m4ac.object_type == AOT_AAC_LTP) {
01919         av_log(ac->avctx, AV_LOG_ERROR,
01920                "Dependent coupling is not supported together with LTP\n");
01921         return;
01922     }
01923     for (g = 0; g < ics->num_window_groups; g++) {
01924         for (i = 0; i < ics->max_sfb; i++, idx++) {
01925             if (cce->ch[0].band_type[idx] != ZERO_BT) {
01926                 const float gain = cce->coup.gain[index][idx];
01927                 for (group = 0; group < ics->group_len[g]; group++) {
01928                     for (k = offsets[i]; k < offsets[i + 1]; k++) {
01929                         // XXX dsputil-ize
01930                         dest[group * 128 + k] += gain * src[group * 128 + k];
01931                     }
01932                 }
01933             }
01934         }
01935         dest += ics->group_len[g] * 128;
01936         src  += ics->group_len[g] * 128;
01937     }
01938 }
01939 
01945 static void apply_independent_coupling(AACContext *ac,
01946                                        SingleChannelElement *target,
01947                                        ChannelElement *cce, int index)
01948 {
01949     int i;
01950     const float gain = cce->coup.gain[index][0];
01951     const float *src = cce->ch[0].ret;
01952     float *dest = target->ret;
01953     const int len = 1024 << (ac->m4ac.sbr == 1);
01954 
01955     for (i = 0; i < len; i++)
01956         dest[i] += gain * src[i];
01957 }
01958 
01964 static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
01965                                    enum RawDataBlockType type, int elem_id,
01966                                    enum CouplingPoint coupling_point,
01967                                    void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
01968 {
01969     int i, c;
01970 
01971     for (i = 0; i < MAX_ELEM_ID; i++) {
01972         ChannelElement *cce = ac->che[TYPE_CCE][i];
01973         int index = 0;
01974 
01975         if (cce && cce->coup.coupling_point == coupling_point) {
01976             ChannelCoupling *coup = &cce->coup;
01977 
01978             for (c = 0; c <= coup->num_coupled; c++) {
01979                 if (coup->type[c] == type && coup->id_select[c] == elem_id) {
01980                     if (coup->ch_select[c] != 1) {
01981                         apply_coupling_method(ac, &cc->ch[0], cce, index);
01982                         if (coup->ch_select[c] != 0)
01983                             index++;
01984                     }
01985                     if (coup->ch_select[c] != 2)
01986                         apply_coupling_method(ac, &cc->ch[1], cce, index++);
01987                 } else
01988                     index += 1 + (coup->ch_select[c] == 3);
01989             }
01990         }
01991     }
01992 }
01993 
01997 static void spectral_to_sample(AACContext *ac)
01998 {
01999     int i, type;
02000     for (type = 3; type >= 0; type--) {
02001         for (i = 0; i < MAX_ELEM_ID; i++) {
02002             ChannelElement *che = ac->che[type][i];
02003             if (che) {
02004                 if (type <= TYPE_CPE)
02005                     apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
02006                 if (ac->m4ac.object_type == AOT_AAC_LTP) {
02007                     if (che->ch[0].ics.predictor_present) {
02008                         if (che->ch[0].ics.ltp.present)
02009                             apply_ltp(ac, &che->ch[0]);
02010                         if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
02011                             apply_ltp(ac, &che->ch[1]);
02012                     }
02013                 }
02014                 if (che->ch[0].tns.present)
02015                     apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
02016                 if (che->ch[1].tns.present)
02017                     apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
02018                 if (type <= TYPE_CPE)
02019                     apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
02020                 if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
02021                     imdct_and_windowing(ac, &che->ch[0]);
02022                     if (ac->m4ac.object_type == AOT_AAC_LTP)
02023                         update_ltp(ac, &che->ch[0]);
02024                     if (type == TYPE_CPE) {
02025                         imdct_and_windowing(ac, &che->ch[1]);
02026                         if (ac->m4ac.object_type == AOT_AAC_LTP)
02027                             update_ltp(ac, &che->ch[1]);
02028                     }
02029                     if (ac->m4ac.sbr > 0) {
02030                         ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
02031                     }
02032                 }
02033                 if (type <= TYPE_CCE)
02034                     apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
02035             }
02036         }
02037     }
02038 }
02039 
02040 static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
02041 {
02042     int size;
02043     AACADTSHeaderInfo hdr_info;
02044 
02045     size = ff_aac_parse_header(gb, &hdr_info);
02046     if (size > 0) {
02047         if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
02048             enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
02049             memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
02050             ac->m4ac.chan_config = hdr_info.chan_config;
02051             if (set_default_channel_config(ac->avctx, new_che_pos, hdr_info.chan_config))
02052                 return -7;
02053             if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
02054                 return -7;
02055         } else if (ac->output_configured != OC_LOCKED) {
02056             ac->output_configured = OC_NONE;
02057         }
02058         if (ac->output_configured != OC_LOCKED) {
02059             ac->m4ac.sbr = -1;
02060             ac->m4ac.ps  = -1;
02061         }
02062         ac->m4ac.sample_rate     = hdr_info.sample_rate;
02063         ac->m4ac.sampling_index  = hdr_info.sampling_index;
02064         ac->m4ac.object_type     = hdr_info.object_type;
02065         if (!ac->avctx->sample_rate)
02066             ac->avctx->sample_rate = hdr_info.sample_rate;
02067         if (hdr_info.num_aac_frames == 1) {
02068             if (!hdr_info.crc_absent)
02069                 skip_bits(gb, 16);
02070         } else {
02071             av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0);
02072             return -1;
02073         }
02074     }
02075     return size;
02076 }
02077 
02078 static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
02079                                 int *data_size, GetBitContext *gb)
02080 {
02081     AACContext *ac = avctx->priv_data;
02082     ChannelElement *che = NULL, *che_prev = NULL;
02083     enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
02084     int err, elem_id, data_size_tmp;
02085     int samples = 0, multiplier, audio_found = 0;
02086 
02087     if (show_bits(gb, 12) == 0xfff) {
02088         if (parse_adts_frame_header(ac, gb) < 0) {
02089             av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
02090             return -1;
02091         }
02092         if (ac->m4ac.sampling_index > 12) {
02093             av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
02094             return -1;
02095         }
02096     }
02097 
02098     ac->tags_mapped = 0;
02099     // parse
02100     while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
02101         elem_id = get_bits(gb, 4);
02102 
02103         if (elem_type < TYPE_DSE) {
02104             if (!(che=get_che(ac, elem_type, elem_id))) {
02105                 av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
02106                        elem_type, elem_id);
02107                 return -1;
02108             }
02109             samples = 1024;
02110         }
02111 
02112         switch (elem_type) {
02113 
02114         case TYPE_SCE:
02115             err = decode_ics(ac, &che->ch[0], gb, 0, 0);
02116             audio_found = 1;
02117             break;
02118 
02119         case TYPE_CPE:
02120             err = decode_cpe(ac, gb, che);
02121             audio_found = 1;
02122             break;
02123 
02124         case TYPE_CCE:
02125             err = decode_cce(ac, gb, che);
02126             break;
02127 
02128         case TYPE_LFE:
02129             err = decode_ics(ac, &che->ch[0], gb, 0, 0);
02130             audio_found = 1;
02131             break;
02132 
02133         case TYPE_DSE:
02134             err = skip_data_stream_element(ac, gb);
02135             break;
02136 
02137         case TYPE_PCE: {
02138             enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
02139             memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
02140             if ((err = decode_pce(avctx, &ac->m4ac, new_che_pos, gb)))
02141                 break;
02142             if (ac->output_configured > OC_TRIAL_PCE)
02143                 av_log(avctx, AV_LOG_ERROR,
02144                        "Not evaluating a further program_config_element as this construct is dubious at best.\n");
02145             else
02146                 err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
02147             break;
02148         }
02149 
02150         case TYPE_FIL:
02151             if (elem_id == 15)
02152                 elem_id += get_bits(gb, 8) - 1;
02153             if (get_bits_left(gb) < 8 * elem_id) {
02154                     av_log(avctx, AV_LOG_ERROR, overread_err);
02155                     return -1;
02156             }
02157             while (elem_id > 0)
02158                 elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
02159             err = 0; /* FIXME */
02160             break;
02161 
02162         default:
02163             err = -1; /* should not happen, but keeps compiler happy */
02164             break;
02165         }
02166 
02167         che_prev       = che;
02168         elem_type_prev = elem_type;
02169 
02170         if (err)
02171             return err;
02172 
02173         if (get_bits_left(gb) < 3) {
02174             av_log(avctx, AV_LOG_ERROR, overread_err);
02175             return -1;
02176         }
02177     }
02178 
02179     spectral_to_sample(ac);
02180 
02181     multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0;
02182     samples <<= multiplier;
02183     if (ac->output_configured < OC_LOCKED) {
02184         avctx->sample_rate = ac->m4ac.sample_rate << multiplier;
02185         avctx->frame_size = samples;
02186     }
02187 
02188     data_size_tmp = samples * avctx->channels *
02189                     av_get_bytes_per_sample(avctx->sample_fmt);
02190     if (*data_size < data_size_tmp) {
02191         av_log(avctx, AV_LOG_ERROR,
02192                "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
02193                *data_size, data_size_tmp);
02194         return -1;
02195     }
02196     *data_size = data_size_tmp;
02197 
02198     if (samples) {
02199         if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
02200             ac->fmt_conv.float_interleave(data, (const float **)ac->output_data,
02201                                           samples, avctx->channels);
02202         else
02203             ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data,
02204                                                    samples, avctx->channels);
02205     }
02206 
02207     if (ac->output_configured && audio_found)
02208         ac->output_configured = OC_LOCKED;
02209 
02210     return 0;
02211 }
02212 
02213 static int aac_decode_frame(AVCodecContext *avctx, void *data,
02214                             int *data_size, AVPacket *avpkt)
02215 {
02216     const uint8_t *buf = avpkt->data;
02217     int buf_size = avpkt->size;
02218     GetBitContext gb;
02219     int buf_consumed;
02220     int buf_offset;
02221     int err;
02222 
02223     init_get_bits(&gb, buf, buf_size * 8);
02224 
02225     if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0)
02226         return err;
02227 
02228     buf_consumed = (get_bits_count(&gb) + 7) >> 3;
02229     for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
02230         if (buf[buf_offset])
02231             break;
02232 
02233     return buf_size > buf_offset ? buf_consumed : buf_size;
02234 }
02235 
02236 static av_cold int aac_decode_close(AVCodecContext *avctx)
02237 {
02238     AACContext *ac = avctx->priv_data;
02239     int i, type;
02240 
02241     for (i = 0; i < MAX_ELEM_ID; i++) {
02242         for (type = 0; type < 4; type++) {
02243             if (ac->che[type][i])
02244                 ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
02245             av_freep(&ac->che[type][i]);
02246         }
02247     }
02248 
02249     ff_mdct_end(&ac->mdct);
02250     ff_mdct_end(&ac->mdct_small);
02251     ff_mdct_end(&ac->mdct_ltp);
02252     return 0;
02253 }
02254 
02255 
02256 #define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
02257 
02258 struct LATMContext {
02259     AACContext      aac_ctx;             
02260     int             initialized;         
02261 
02262     // parser data
02263     int             audio_mux_version_A; 
02264     int             frame_length_type;   
02265     int             frame_length;        
02266 };
02267 
02268 static inline uint32_t latm_get_value(GetBitContext *b)
02269 {
02270     int length = get_bits(b, 2);
02271 
02272     return get_bits_long(b, (length+1)*8);
02273 }
02274 
02275 static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
02276                                              GetBitContext *gb)
02277 {
02278     AVCodecContext *avctx = latmctx->aac_ctx.avctx;
02279     MPEG4AudioConfig m4ac;
02280     int  config_start_bit = get_bits_count(gb);
02281     int     bits_consumed, esize;
02282 
02283     if (config_start_bit % 8) {
02284         av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
02285                                "config not byte aligned.\n", 1);
02286         return AVERROR_INVALIDDATA;
02287     } else {
02288         bits_consumed =
02289             decode_audio_specific_config(NULL, avctx, &m4ac,
02290                                          gb->buffer + (config_start_bit / 8),
02291                                          get_bits_left(gb) / 8);
02292 
02293         if (bits_consumed < 0)
02294             return AVERROR_INVALIDDATA;
02295 
02296         esize = (bits_consumed+7) / 8;
02297 
02298         if (avctx->extradata_size <= esize) {
02299             av_free(avctx->extradata);
02300             avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
02301             if (!avctx->extradata)
02302                 return AVERROR(ENOMEM);
02303         }
02304 
02305         avctx->extradata_size = esize;
02306         memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
02307         memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
02308 
02309         skip_bits_long(gb, bits_consumed);
02310     }
02311 
02312     return bits_consumed;
02313 }
02314 
02315 static int read_stream_mux_config(struct LATMContext *latmctx,
02316                                   GetBitContext *gb)
02317 {
02318     int ret, audio_mux_version = get_bits(gb, 1);
02319 
02320     latmctx->audio_mux_version_A = 0;
02321     if (audio_mux_version)
02322         latmctx->audio_mux_version_A = get_bits(gb, 1);
02323 
02324     if (!latmctx->audio_mux_version_A) {
02325 
02326         if (audio_mux_version)
02327             latm_get_value(gb);                 // taraFullness
02328 
02329         skip_bits(gb, 1);                       // allStreamSameTimeFraming
02330         skip_bits(gb, 6);                       // numSubFrames
02331         // numPrograms
02332         if (get_bits(gb, 4)) {                  // numPrograms
02333             av_log_missing_feature(latmctx->aac_ctx.avctx,
02334                                    "multiple programs are not supported\n", 1);
02335             return AVERROR_PATCHWELCOME;
02336         }
02337 
02338         // for each program (which there is only on in DVB)
02339 
02340         // for each layer (which there is only on in DVB)
02341         if (get_bits(gb, 3)) {                   // numLayer
02342             av_log_missing_feature(latmctx->aac_ctx.avctx,
02343                                    "multiple layers are not supported\n", 1);
02344             return AVERROR_PATCHWELCOME;
02345         }
02346 
02347         // for all but first stream: use_same_config = get_bits(gb, 1);
02348         if (!audio_mux_version) {
02349             if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
02350                 return ret;
02351         } else {
02352             int ascLen = latm_get_value(gb);
02353             if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
02354                 return ret;
02355             ascLen -= ret;
02356             skip_bits_long(gb, ascLen);
02357         }
02358 
02359         latmctx->frame_length_type = get_bits(gb, 3);
02360         switch (latmctx->frame_length_type) {
02361         case 0:
02362             skip_bits(gb, 8);       // latmBufferFullness
02363             break;
02364         case 1:
02365             latmctx->frame_length = get_bits(gb, 9);
02366             break;
02367         case 3:
02368         case 4:
02369         case 5:
02370             skip_bits(gb, 6);       // CELP frame length table index
02371             break;
02372         case 6:
02373         case 7:
02374             skip_bits(gb, 1);       // HVXC frame length table index
02375             break;
02376         }
02377 
02378         if (get_bits(gb, 1)) {                  // other data
02379             if (audio_mux_version) {
02380                 latm_get_value(gb);             // other_data_bits
02381             } else {
02382                 int esc;
02383                 do {
02384                     esc = get_bits(gb, 1);
02385                     skip_bits(gb, 8);
02386                 } while (esc);
02387             }
02388         }
02389 
02390         if (get_bits(gb, 1))                     // crc present
02391             skip_bits(gb, 8);                    // config_crc
02392     }
02393 
02394     return 0;
02395 }
02396 
02397 static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
02398 {
02399     uint8_t tmp;
02400 
02401     if (ctx->frame_length_type == 0) {
02402         int mux_slot_length = 0;
02403         do {
02404             tmp = get_bits(gb, 8);
02405             mux_slot_length += tmp;
02406         } while (tmp == 255);
02407         return mux_slot_length;
02408     } else if (ctx->frame_length_type == 1) {
02409         return ctx->frame_length;
02410     } else if (ctx->frame_length_type == 3 ||
02411                ctx->frame_length_type == 5 ||
02412                ctx->frame_length_type == 7) {
02413         skip_bits(gb, 2);          // mux_slot_length_coded
02414     }
02415     return 0;
02416 }
02417 
02418 static int read_audio_mux_element(struct LATMContext *latmctx,
02419                                   GetBitContext *gb)
02420 {
02421     int err;
02422     uint8_t use_same_mux = get_bits(gb, 1);
02423     if (!use_same_mux) {
02424         if ((err = read_stream_mux_config(latmctx, gb)) < 0)
02425             return err;
02426     } else if (!latmctx->aac_ctx.avctx->extradata) {
02427         av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
02428                "no decoder config found\n");
02429         return AVERROR(EAGAIN);
02430     }
02431     if (latmctx->audio_mux_version_A == 0) {
02432         int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
02433         if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
02434             av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
02435             return AVERROR_INVALIDDATA;
02436         } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
02437             av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
02438                    "frame length mismatch %d << %d\n",
02439                    mux_slot_length_bytes * 8, get_bits_left(gb));
02440             return AVERROR_INVALIDDATA;
02441         }
02442     }
02443     return 0;
02444 }
02445 
02446 
02447 static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
02448                              AVPacket *avpkt)
02449 {
02450     struct LATMContext *latmctx = avctx->priv_data;
02451     int                 muxlength, err;
02452     GetBitContext       gb;
02453 
02454     if (avpkt->size == 0)
02455         return 0;
02456 
02457     init_get_bits(&gb, avpkt->data, avpkt->size * 8);
02458 
02459     // check for LOAS sync word
02460     if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
02461         return AVERROR_INVALIDDATA;
02462 
02463     muxlength = get_bits(&gb, 13) + 3;
02464     // not enough data, the parser should have sorted this
02465     if (muxlength > avpkt->size)
02466         return AVERROR_INVALIDDATA;
02467 
02468     if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
02469         return err;
02470 
02471     if (!latmctx->initialized) {
02472         if (!avctx->extradata) {
02473             *out_size = 0;
02474             return avpkt->size;
02475         } else {
02476             aac_decode_close(avctx);
02477             if ((err = aac_decode_init(avctx)) < 0)
02478                 return err;
02479             latmctx->initialized = 1;
02480         }
02481     }
02482 
02483     if (show_bits(&gb, 12) == 0xfff) {
02484         av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
02485                "ADTS header detected, probably as result of configuration "
02486                "misparsing\n");
02487         return AVERROR_INVALIDDATA;
02488     }
02489 
02490     if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
02491         return err;
02492 
02493     return muxlength;
02494 }
02495 
02496 av_cold static int latm_decode_init(AVCodecContext *avctx)
02497 {
02498     struct LATMContext *latmctx = avctx->priv_data;
02499     int ret;
02500 
02501     ret = aac_decode_init(avctx);
02502 
02503     if (avctx->extradata_size > 0) {
02504         latmctx->initialized = !ret;
02505     } else {
02506         latmctx->initialized = 0;
02507     }
02508 
02509     return ret;
02510 }
02511 
02512 
02513 AVCodec ff_aac_decoder = {
02514     "aac",
02515     AVMEDIA_TYPE_AUDIO,
02516     CODEC_ID_AAC,
02517     sizeof(AACContext),
02518     aac_decode_init,
02519     NULL,
02520     aac_decode_close,
02521     aac_decode_frame,
02522     .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
02523     .sample_fmts = (const enum AVSampleFormat[]) {
02524         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
02525     },
02526     .channel_layouts = aac_channel_layout,
02527 };
02528 
02529 /*
02530     Note: This decoder filter is intended to decode LATM streams transferred
02531     in MPEG transport streams which only contain one program.
02532     To do a more complex LATM demuxing a separate LATM demuxer should be used.
02533 */
02534 AVCodec ff_aac_latm_decoder = {
02535     .name = "aac_latm",
02536     .type = AVMEDIA_TYPE_AUDIO,
02537     .id   = CODEC_ID_AAC_LATM,
02538     .priv_data_size = sizeof(struct LATMContext),
02539     .init   = latm_decode_init,
02540     .close  = aac_decode_close,
02541     .decode = latm_decode_frame,
02542     .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
02543     .sample_fmts = (const enum AVSampleFormat[]) {
02544         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
02545     },
02546     .channel_layouts = aac_channel_layout,
02547 };

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