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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     AAC_INIT_VLC_STATIC( 0, 304);
00572     AAC_INIT_VLC_STATIC( 1, 270);
00573     AAC_INIT_VLC_STATIC( 2, 550);
00574     AAC_INIT_VLC_STATIC( 3, 300);
00575     AAC_INIT_VLC_STATIC( 4, 328);
00576     AAC_INIT_VLC_STATIC( 5, 294);
00577     AAC_INIT_VLC_STATIC( 6, 306);
00578     AAC_INIT_VLC_STATIC( 7, 268);
00579     AAC_INIT_VLC_STATIC( 8, 510);
00580     AAC_INIT_VLC_STATIC( 9, 366);
00581     AAC_INIT_VLC_STATIC(10, 462);
00582 
00583     ff_aac_sbr_init();
00584 
00585     dsputil_init(&ac->dsp, avctx);
00586     ff_fmt_convert_init(&ac->fmt_conv, avctx);
00587 
00588     ac->random_state = 0x1f2e3d4c;
00589 
00590     ff_aac_tableinit();
00591 
00592     INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
00593                     ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
00594                     ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
00595                     352);
00596 
00597     ff_mdct_init(&ac->mdct,       11, 1, output_scale_factor/1024.0);
00598     ff_mdct_init(&ac->mdct_small,  8, 1, output_scale_factor/128.0);
00599     ff_mdct_init(&ac->mdct_ltp,   11, 0, -2.0/output_scale_factor);
00600     // window initialization
00601     ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
00602     ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
00603     ff_init_ff_sine_windows(10);
00604     ff_init_ff_sine_windows( 7);
00605 
00606     cbrt_tableinit();
00607 
00608     return 0;
00609 }
00610 
00614 static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
00615 {
00616     int byte_align = get_bits1(gb);
00617     int count = get_bits(gb, 8);
00618     if (count == 255)
00619         count += get_bits(gb, 8);
00620     if (byte_align)
00621         align_get_bits(gb);
00622 
00623     if (get_bits_left(gb) < 8 * count) {
00624         av_log(ac->avctx, AV_LOG_ERROR, overread_err);
00625         return -1;
00626     }
00627     skip_bits_long(gb, 8 * count);
00628     return 0;
00629 }
00630 
00631 static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
00632                              GetBitContext *gb)
00633 {
00634     int sfb;
00635     if (get_bits1(gb)) {
00636         ics->predictor_reset_group = get_bits(gb, 5);
00637         if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
00638             av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
00639             return -1;
00640         }
00641     }
00642     for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
00643         ics->prediction_used[sfb] = get_bits1(gb);
00644     }
00645     return 0;
00646 }
00647 
00651 static void decode_ltp(AACContext *ac, LongTermPrediction *ltp,
00652                        GetBitContext *gb, uint8_t max_sfb)
00653 {
00654     int sfb;
00655 
00656     ltp->lag  = get_bits(gb, 11);
00657     ltp->coef = ltp_coef[get_bits(gb, 3)];
00658     for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
00659         ltp->used[sfb] = get_bits1(gb);
00660 }
00661 
00667 static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
00668                            GetBitContext *gb, int common_window)
00669 {
00670     if (get_bits1(gb)) {
00671         av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
00672         memset(ics, 0, sizeof(IndividualChannelStream));
00673         return -1;
00674     }
00675     ics->window_sequence[1] = ics->window_sequence[0];
00676     ics->window_sequence[0] = get_bits(gb, 2);
00677     ics->use_kb_window[1]   = ics->use_kb_window[0];
00678     ics->use_kb_window[0]   = get_bits1(gb);
00679     ics->num_window_groups  = 1;
00680     ics->group_len[0]       = 1;
00681     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
00682         int i;
00683         ics->max_sfb = get_bits(gb, 4);
00684         for (i = 0; i < 7; i++) {
00685             if (get_bits1(gb)) {
00686                 ics->group_len[ics->num_window_groups - 1]++;
00687             } else {
00688                 ics->num_window_groups++;
00689                 ics->group_len[ics->num_window_groups - 1] = 1;
00690             }
00691         }
00692         ics->num_windows       = 8;
00693         ics->swb_offset        =    ff_swb_offset_128[ac->m4ac.sampling_index];
00694         ics->num_swb           =   ff_aac_num_swb_128[ac->m4ac.sampling_index];
00695         ics->tns_max_bands     = ff_tns_max_bands_128[ac->m4ac.sampling_index];
00696         ics->predictor_present = 0;
00697     } else {
00698         ics->max_sfb               = get_bits(gb, 6);
00699         ics->num_windows           = 1;
00700         ics->swb_offset            =    ff_swb_offset_1024[ac->m4ac.sampling_index];
00701         ics->num_swb               =   ff_aac_num_swb_1024[ac->m4ac.sampling_index];
00702         ics->tns_max_bands         = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
00703         ics->predictor_present     = get_bits1(gb);
00704         ics->predictor_reset_group = 0;
00705         if (ics->predictor_present) {
00706             if (ac->m4ac.object_type == AOT_AAC_MAIN) {
00707                 if (decode_prediction(ac, ics, gb)) {
00708                     memset(ics, 0, sizeof(IndividualChannelStream));
00709                     return -1;
00710                 }
00711             } else if (ac->m4ac.object_type == AOT_AAC_LC) {
00712                 av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
00713                 memset(ics, 0, sizeof(IndividualChannelStream));
00714                 return -1;
00715             } else {
00716                 if ((ics->ltp.present = get_bits(gb, 1)))
00717                     decode_ltp(ac, &ics->ltp, gb, ics->max_sfb);
00718             }
00719         }
00720     }
00721 
00722     if (ics->max_sfb > ics->num_swb) {
00723         av_log(ac->avctx, AV_LOG_ERROR,
00724                "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
00725                ics->max_sfb, ics->num_swb);
00726         memset(ics, 0, sizeof(IndividualChannelStream));
00727         return -1;
00728     }
00729 
00730     return 0;
00731 }
00732 
00741 static int decode_band_types(AACContext *ac, enum BandType band_type[120],
00742                              int band_type_run_end[120], GetBitContext *gb,
00743                              IndividualChannelStream *ics)
00744 {
00745     int g, idx = 0;
00746     const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
00747     for (g = 0; g < ics->num_window_groups; g++) {
00748         int k = 0;
00749         while (k < ics->max_sfb) {
00750             uint8_t sect_end = k;
00751             int sect_len_incr;
00752             int sect_band_type = get_bits(gb, 4);
00753             if (sect_band_type == 12) {
00754                 av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
00755                 return -1;
00756             }
00757             do {
00758                 sect_len_incr = get_bits(gb, bits);
00759                 sect_end += sect_len_incr;
00760                 if (get_bits_left(gb) < 0) {
00761                     av_log(ac->avctx, AV_LOG_ERROR, overread_err);
00762                     return -1;
00763                 }
00764                 if (sect_end > ics->max_sfb) {
00765                     av_log(ac->avctx, AV_LOG_ERROR,
00766                            "Number of bands (%d) exceeds limit (%d).\n",
00767                            sect_end, ics->max_sfb);
00768                     return -1;
00769                 }
00770             } while (sect_len_incr == (1 << bits) - 1);
00771             for (; k < sect_end; k++) {
00772                 band_type        [idx]   = sect_band_type;
00773                 band_type_run_end[idx++] = sect_end;
00774             }
00775         }
00776     }
00777     return 0;
00778 }
00779 
00790 static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
00791                                unsigned int global_gain,
00792                                IndividualChannelStream *ics,
00793                                enum BandType band_type[120],
00794                                int band_type_run_end[120])
00795 {
00796     int g, i, idx = 0;
00797     int offset[3] = { global_gain, global_gain - 90, 0 };
00798     int clipped_offset;
00799     int noise_flag = 1;
00800     static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
00801     for (g = 0; g < ics->num_window_groups; g++) {
00802         for (i = 0; i < ics->max_sfb;) {
00803             int run_end = band_type_run_end[idx];
00804             if (band_type[idx] == ZERO_BT) {
00805                 for (; i < run_end; i++, idx++)
00806                     sf[idx] = 0.;
00807             } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
00808                 for (; i < run_end; i++, idx++) {
00809                     offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
00810                     clipped_offset = av_clip(offset[2], -155, 100);
00811                     if (offset[2] != clipped_offset) {
00812                         av_log_ask_for_sample(ac->avctx, "Intensity stereo "
00813                                 "position clipped (%d -> %d).\nIf you heard an "
00814                                 "audible artifact, there may be a bug in the "
00815                                 "decoder. ", offset[2], clipped_offset);
00816                     }
00817                     sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
00818                 }
00819             } else if (band_type[idx] == NOISE_BT) {
00820                 for (; i < run_end; i++, idx++) {
00821                     if (noise_flag-- > 0)
00822                         offset[1] += get_bits(gb, 9) - 256;
00823                     else
00824                         offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
00825                     clipped_offset = av_clip(offset[1], -100, 155);
00826                     if (offset[1] != clipped_offset) {
00827                         av_log_ask_for_sample(ac->avctx, "Noise gain clipped "
00828                                 "(%d -> %d).\nIf you heard an audible "
00829                                 "artifact, there may be a bug in the decoder. ",
00830                                 offset[1], clipped_offset);
00831                     }
00832                     sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
00833                 }
00834             } else {
00835                 for (; i < run_end; i++, idx++) {
00836                     offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
00837                     if (offset[0] > 255U) {
00838                         av_log(ac->avctx, AV_LOG_ERROR,
00839                                "%s (%d) out of range.\n", sf_str[0], offset[0]);
00840                         return -1;
00841                     }
00842                     sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
00843                 }
00844             }
00845         }
00846     }
00847     return 0;
00848 }
00849 
00853 static int decode_pulses(Pulse *pulse, GetBitContext *gb,
00854                          const uint16_t *swb_offset, int num_swb)
00855 {
00856     int i, pulse_swb;
00857     pulse->num_pulse = get_bits(gb, 2) + 1;
00858     pulse_swb        = get_bits(gb, 6);
00859     if (pulse_swb >= num_swb)
00860         return -1;
00861     pulse->pos[0]    = swb_offset[pulse_swb];
00862     pulse->pos[0]   += get_bits(gb, 5);
00863     if (pulse->pos[0] > 1023)
00864         return -1;
00865     pulse->amp[0]    = get_bits(gb, 4);
00866     for (i = 1; i < pulse->num_pulse; i++) {
00867         pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
00868         if (pulse->pos[i] > 1023)
00869             return -1;
00870         pulse->amp[i] = get_bits(gb, 4);
00871     }
00872     return 0;
00873 }
00874 
00880 static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
00881                       GetBitContext *gb, const IndividualChannelStream *ics)
00882 {
00883     int w, filt, i, coef_len, coef_res, coef_compress;
00884     const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
00885     const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
00886     for (w = 0; w < ics->num_windows; w++) {
00887         if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
00888             coef_res = get_bits1(gb);
00889 
00890             for (filt = 0; filt < tns->n_filt[w]; filt++) {
00891                 int tmp2_idx;
00892                 tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
00893 
00894                 if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
00895                     av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
00896                            tns->order[w][filt], tns_max_order);
00897                     tns->order[w][filt] = 0;
00898                     return -1;
00899                 }
00900                 if (tns->order[w][filt]) {
00901                     tns->direction[w][filt] = get_bits1(gb);
00902                     coef_compress = get_bits1(gb);
00903                     coef_len = coef_res + 3 - coef_compress;
00904                     tmp2_idx = 2 * coef_compress + coef_res;
00905 
00906                     for (i = 0; i < tns->order[w][filt]; i++)
00907                         tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
00908                 }
00909             }
00910         }
00911     }
00912     return 0;
00913 }
00914 
00922 static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
00923                                    int ms_present)
00924 {
00925     int idx;
00926     if (ms_present == 1) {
00927         for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
00928             cpe->ms_mask[idx] = get_bits1(gb);
00929     } else if (ms_present == 2) {
00930         memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
00931     }
00932 }
00933 
00934 #ifndef VMUL2
00935 static inline float *VMUL2(float *dst, const float *v, unsigned idx,
00936                            const float *scale)
00937 {
00938     float s = *scale;
00939     *dst++ = v[idx    & 15] * s;
00940     *dst++ = v[idx>>4 & 15] * s;
00941     return dst;
00942 }
00943 #endif
00944 
00945 #ifndef VMUL4
00946 static inline float *VMUL4(float *dst, const float *v, unsigned idx,
00947                            const float *scale)
00948 {
00949     float s = *scale;
00950     *dst++ = v[idx    & 3] * s;
00951     *dst++ = v[idx>>2 & 3] * s;
00952     *dst++ = v[idx>>4 & 3] * s;
00953     *dst++ = v[idx>>6 & 3] * s;
00954     return dst;
00955 }
00956 #endif
00957 
00958 #ifndef VMUL2S
00959 static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
00960                             unsigned sign, const float *scale)
00961 {
00962     union float754 s0, s1;
00963 
00964     s0.f = s1.f = *scale;
00965     s0.i ^= sign >> 1 << 31;
00966     s1.i ^= sign      << 31;
00967 
00968     *dst++ = v[idx    & 15] * s0.f;
00969     *dst++ = v[idx>>4 & 15] * s1.f;
00970 
00971     return dst;
00972 }
00973 #endif
00974 
00975 #ifndef VMUL4S
00976 static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
00977                             unsigned sign, const float *scale)
00978 {
00979     unsigned nz = idx >> 12;
00980     union float754 s = { .f = *scale };
00981     union float754 t;
00982 
00983     t.i = s.i ^ (sign & 1U<<31);
00984     *dst++ = v[idx    & 3] * t.f;
00985 
00986     sign <<= nz & 1; nz >>= 1;
00987     t.i = s.i ^ (sign & 1U<<31);
00988     *dst++ = v[idx>>2 & 3] * t.f;
00989 
00990     sign <<= nz & 1; nz >>= 1;
00991     t.i = s.i ^ (sign & 1U<<31);
00992     *dst++ = v[idx>>4 & 3] * t.f;
00993 
00994     sign <<= nz & 1; nz >>= 1;
00995     t.i = s.i ^ (sign & 1U<<31);
00996     *dst++ = v[idx>>6 & 3] * t.f;
00997 
00998     return dst;
00999 }
01000 #endif
01001 
01014 static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
01015                                        GetBitContext *gb, const float sf[120],
01016                                        int pulse_present, const Pulse *pulse,
01017                                        const IndividualChannelStream *ics,
01018                                        enum BandType band_type[120])
01019 {
01020     int i, k, g, idx = 0;
01021     const int c = 1024 / ics->num_windows;
01022     const uint16_t *offsets = ics->swb_offset;
01023     float *coef_base = coef;
01024 
01025     for (g = 0; g < ics->num_windows; g++)
01026         memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
01027 
01028     for (g = 0; g < ics->num_window_groups; g++) {
01029         unsigned g_len = ics->group_len[g];
01030 
01031         for (i = 0; i < ics->max_sfb; i++, idx++) {
01032             const unsigned cbt_m1 = band_type[idx] - 1;
01033             float *cfo = coef + offsets[i];
01034             int off_len = offsets[i + 1] - offsets[i];
01035             int group;
01036 
01037             if (cbt_m1 >= INTENSITY_BT2 - 1) {
01038                 for (group = 0; group < g_len; group++, cfo+=128) {
01039                     memset(cfo, 0, off_len * sizeof(float));
01040                 }
01041             } else if (cbt_m1 == NOISE_BT - 1) {
01042                 for (group = 0; group < g_len; group++, cfo+=128) {
01043                     float scale;
01044                     float band_energy;
01045 
01046                     for (k = 0; k < off_len; k++) {
01047                         ac->random_state  = lcg_random(ac->random_state);
01048                         cfo[k] = ac->random_state;
01049                     }
01050 
01051                     band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
01052                     scale = sf[idx] / sqrtf(band_energy);
01053                     ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
01054                 }
01055             } else {
01056                 const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
01057                 const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
01058                 VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
01059                 OPEN_READER(re, gb);
01060 
01061                 switch (cbt_m1 >> 1) {
01062                 case 0:
01063                     for (group = 0; group < g_len; group++, cfo+=128) {
01064                         float *cf = cfo;
01065                         int len = off_len;
01066 
01067                         do {
01068                             int code;
01069                             unsigned cb_idx;
01070 
01071                             UPDATE_CACHE(re, gb);
01072                             GET_VLC(code, re, gb, vlc_tab, 8, 2);
01073                             cb_idx = cb_vector_idx[code];
01074                             cf = VMUL4(cf, vq, cb_idx, sf + idx);
01075                         } while (len -= 4);
01076                     }
01077                     break;
01078 
01079                 case 1:
01080                     for (group = 0; group < g_len; group++, cfo+=128) {
01081                         float *cf = cfo;
01082                         int len = off_len;
01083 
01084                         do {
01085                             int code;
01086                             unsigned nnz;
01087                             unsigned cb_idx;
01088                             uint32_t bits;
01089 
01090                             UPDATE_CACHE(re, gb);
01091                             GET_VLC(code, re, gb, vlc_tab, 8, 2);
01092                             cb_idx = cb_vector_idx[code];
01093                             nnz = cb_idx >> 8 & 15;
01094                             bits = nnz ? GET_CACHE(re, gb) : 0;
01095                             LAST_SKIP_BITS(re, gb, nnz);
01096                             cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
01097                         } while (len -= 4);
01098                     }
01099                     break;
01100 
01101                 case 2:
01102                     for (group = 0; group < g_len; group++, cfo+=128) {
01103                         float *cf = cfo;
01104                         int len = off_len;
01105 
01106                         do {
01107                             int code;
01108                             unsigned cb_idx;
01109 
01110                             UPDATE_CACHE(re, gb);
01111                             GET_VLC(code, re, gb, vlc_tab, 8, 2);
01112                             cb_idx = cb_vector_idx[code];
01113                             cf = VMUL2(cf, vq, cb_idx, sf + idx);
01114                         } while (len -= 2);
01115                     }
01116                     break;
01117 
01118                 case 3:
01119                 case 4:
01120                     for (group = 0; group < g_len; group++, cfo+=128) {
01121                         float *cf = cfo;
01122                         int len = off_len;
01123 
01124                         do {
01125                             int code;
01126                             unsigned nnz;
01127                             unsigned cb_idx;
01128                             unsigned sign;
01129 
01130                             UPDATE_CACHE(re, gb);
01131                             GET_VLC(code, re, gb, vlc_tab, 8, 2);
01132                             cb_idx = cb_vector_idx[code];
01133                             nnz = cb_idx >> 8 & 15;
01134                             sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
01135                             LAST_SKIP_BITS(re, gb, nnz);
01136                             cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
01137                         } while (len -= 2);
01138                     }
01139                     break;
01140 
01141                 default:
01142                     for (group = 0; group < g_len; group++, cfo+=128) {
01143                         float *cf = cfo;
01144                         uint32_t *icf = (uint32_t *) cf;
01145                         int len = off_len;
01146 
01147                         do {
01148                             int code;
01149                             unsigned nzt, nnz;
01150                             unsigned cb_idx;
01151                             uint32_t bits;
01152                             int j;
01153 
01154                             UPDATE_CACHE(re, gb);
01155                             GET_VLC(code, re, gb, vlc_tab, 8, 2);
01156 
01157                             if (!code) {
01158                                 *icf++ = 0;
01159                                 *icf++ = 0;
01160                                 continue;
01161                             }
01162 
01163                             cb_idx = cb_vector_idx[code];
01164                             nnz = cb_idx >> 12;
01165                             nzt = cb_idx >> 8;
01166                             bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
01167                             LAST_SKIP_BITS(re, gb, nnz);
01168 
01169                             for (j = 0; j < 2; j++) {
01170                                 if (nzt & 1<<j) {
01171                                     uint32_t b;
01172                                     int n;
01173                                     /* The total length of escape_sequence must be < 22 bits according
01174                                        to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
01175                                     UPDATE_CACHE(re, gb);
01176                                     b = GET_CACHE(re, gb);
01177                                     b = 31 - av_log2(~b);
01178 
01179                                     if (b > 8) {
01180                                         av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
01181                                         return -1;
01182                                     }
01183 
01184                                     SKIP_BITS(re, gb, b + 1);
01185                                     b += 4;
01186                                     n = (1 << b) + SHOW_UBITS(re, gb, b);
01187                                     LAST_SKIP_BITS(re, gb, b);
01188                                     *icf++ = cbrt_tab[n] | (bits & 1U<<31);
01189                                     bits <<= 1;
01190                                 } else {
01191                                     unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
01192                                     *icf++ = (bits & 1U<<31) | v;
01193                                     bits <<= !!v;
01194                                 }
01195                                 cb_idx >>= 4;
01196                             }
01197                         } while (len -= 2);
01198 
01199                         ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
01200                     }
01201                 }
01202 
01203                 CLOSE_READER(re, gb);
01204             }
01205         }
01206         coef += g_len << 7;
01207     }
01208 
01209     if (pulse_present) {
01210         idx = 0;
01211         for (i = 0; i < pulse->num_pulse; i++) {
01212             float co = coef_base[ pulse->pos[i] ];
01213             while (offsets[idx + 1] <= pulse->pos[i])
01214                 idx++;
01215             if (band_type[idx] != NOISE_BT && sf[idx]) {
01216                 float ico = -pulse->amp[i];
01217                 if (co) {
01218                     co /= sf[idx];
01219                     ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
01220                 }
01221                 coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
01222             }
01223         }
01224     }
01225     return 0;
01226 }
01227 
01228 static av_always_inline float flt16_round(float pf)
01229 {
01230     union float754 tmp;
01231     tmp.f = pf;
01232     tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
01233     return tmp.f;
01234 }
01235 
01236 static av_always_inline float flt16_even(float pf)
01237 {
01238     union float754 tmp;
01239     tmp.f = pf;
01240     tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
01241     return tmp.f;
01242 }
01243 
01244 static av_always_inline float flt16_trunc(float pf)
01245 {
01246     union float754 pun;
01247     pun.f = pf;
01248     pun.i &= 0xFFFF0000U;
01249     return pun.f;
01250 }
01251 
01252 static av_always_inline void predict(PredictorState *ps, float *coef,
01253                                      int output_enable)
01254 {
01255     const float a     = 0.953125; // 61.0 / 64
01256     const float alpha = 0.90625;  // 29.0 / 32
01257     float e0, e1;
01258     float pv;
01259     float k1, k2;
01260     float   r0 = ps->r0,     r1 = ps->r1;
01261     float cor0 = ps->cor0, cor1 = ps->cor1;
01262     float var0 = ps->var0, var1 = ps->var1;
01263 
01264     k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
01265     k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
01266 
01267     pv = flt16_round(k1 * r0 + k2 * r1);
01268     if (output_enable)
01269         *coef += pv;
01270 
01271     e0 = *coef;
01272     e1 = e0 - k1 * r0;
01273 
01274     ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
01275     ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
01276     ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
01277     ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
01278 
01279     ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
01280     ps->r0 = flt16_trunc(a * e0);
01281 }
01282 
01286 static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
01287 {
01288     int sfb, k;
01289 
01290     if (!sce->ics.predictor_initialized) {
01291         reset_all_predictors(sce->predictor_state);
01292         sce->ics.predictor_initialized = 1;
01293     }
01294 
01295     if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
01296         for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
01297             for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
01298                 predict(&sce->predictor_state[k], &sce->coeffs[k],
01299                         sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
01300             }
01301         }
01302         if (sce->ics.predictor_reset_group)
01303             reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
01304     } else
01305         reset_all_predictors(sce->predictor_state);
01306 }
01307 
01316 static int decode_ics(AACContext *ac, SingleChannelElement *sce,
01317                       GetBitContext *gb, int common_window, int scale_flag)
01318 {
01319     Pulse pulse;
01320     TemporalNoiseShaping    *tns = &sce->tns;
01321     IndividualChannelStream *ics = &sce->ics;
01322     float *out = sce->coeffs;
01323     int global_gain, pulse_present = 0;
01324 
01325     /* This assignment is to silence a GCC warning about the variable being used
01326      * uninitialized when in fact it always is.
01327      */
01328     pulse.num_pulse = 0;
01329 
01330     global_gain = get_bits(gb, 8);
01331 
01332     if (!common_window && !scale_flag) {
01333         if (decode_ics_info(ac, ics, gb, 0) < 0)
01334             return -1;
01335     }
01336 
01337     if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
01338         return -1;
01339     if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
01340         return -1;
01341 
01342     pulse_present = 0;
01343     if (!scale_flag) {
01344         if ((pulse_present = get_bits1(gb))) {
01345             if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
01346                 av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
01347                 return -1;
01348             }
01349             if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
01350                 av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
01351                 return -1;
01352             }
01353         }
01354         if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
01355             return -1;
01356         if (get_bits1(gb)) {
01357             av_log_missing_feature(ac->avctx, "SSR", 1);
01358             return -1;
01359         }
01360     }
01361 
01362     if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
01363         return -1;
01364 
01365     if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
01366         apply_prediction(ac, sce);
01367 
01368     return 0;
01369 }
01370 
01374 static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
01375 {
01376     const IndividualChannelStream *ics = &cpe->ch[0].ics;
01377     float *ch0 = cpe->ch[0].coeffs;
01378     float *ch1 = cpe->ch[1].coeffs;
01379     int g, i, group, idx = 0;
01380     const uint16_t *offsets = ics->swb_offset;
01381     for (g = 0; g < ics->num_window_groups; g++) {
01382         for (i = 0; i < ics->max_sfb; i++, idx++) {
01383             if (cpe->ms_mask[idx] &&
01384                     cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
01385                 for (group = 0; group < ics->group_len[g]; group++) {
01386                     ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
01387                                               ch1 + group * 128 + offsets[i],
01388                                               offsets[i+1] - offsets[i]);
01389                 }
01390             }
01391         }
01392         ch0 += ics->group_len[g] * 128;
01393         ch1 += ics->group_len[g] * 128;
01394     }
01395 }
01396 
01404 static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present)
01405 {
01406     const IndividualChannelStream *ics = &cpe->ch[1].ics;
01407     SingleChannelElement         *sce1 = &cpe->ch[1];
01408     float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
01409     const uint16_t *offsets = ics->swb_offset;
01410     int g, group, i, idx = 0;
01411     int c;
01412     float scale;
01413     for (g = 0; g < ics->num_window_groups; g++) {
01414         for (i = 0; i < ics->max_sfb;) {
01415             if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
01416                 const int bt_run_end = sce1->band_type_run_end[idx];
01417                 for (; i < bt_run_end; i++, idx++) {
01418                     c = -1 + 2 * (sce1->band_type[idx] - 14);
01419                     if (ms_present)
01420                         c *= 1 - 2 * cpe->ms_mask[idx];
01421                     scale = c * sce1->sf[idx];
01422                     for (group = 0; group < ics->group_len[g]; group++)
01423                         ac->dsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
01424                                                    coef0 + group * 128 + offsets[i],
01425                                                    scale,
01426                                                    offsets[i + 1] - offsets[i]);
01427                 }
01428             } else {
01429                 int bt_run_end = sce1->band_type_run_end[idx];
01430                 idx += bt_run_end - i;
01431                 i    = bt_run_end;
01432             }
01433         }
01434         coef0 += ics->group_len[g] * 128;
01435         coef1 += ics->group_len[g] * 128;
01436     }
01437 }
01438 
01444 static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
01445 {
01446     int i, ret, common_window, ms_present = 0;
01447 
01448     common_window = get_bits1(gb);
01449     if (common_window) {
01450         if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
01451             return -1;
01452         i = cpe->ch[1].ics.use_kb_window[0];
01453         cpe->ch[1].ics = cpe->ch[0].ics;
01454         cpe->ch[1].ics.use_kb_window[1] = i;
01455         if (cpe->ch[1].ics.predictor_present && (ac->m4ac.object_type != AOT_AAC_MAIN))
01456             if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
01457                 decode_ltp(ac, &cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
01458         ms_present = get_bits(gb, 2);
01459         if (ms_present == 3) {
01460             av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
01461             return -1;
01462         } else if (ms_present)
01463             decode_mid_side_stereo(cpe, gb, ms_present);
01464     }
01465     if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
01466         return ret;
01467     if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
01468         return ret;
01469 
01470     if (common_window) {
01471         if (ms_present)
01472             apply_mid_side_stereo(ac, cpe);
01473         if (ac->m4ac.object_type == AOT_AAC_MAIN) {
01474             apply_prediction(ac, &cpe->ch[0]);
01475             apply_prediction(ac, &cpe->ch[1]);
01476         }
01477     }
01478 
01479     apply_intensity_stereo(ac, cpe, ms_present);
01480     return 0;
01481 }
01482 
01483 static const float cce_scale[] = {
01484     1.09050773266525765921, //2^(1/8)
01485     1.18920711500272106672, //2^(1/4)
01486     M_SQRT2,
01487     2,
01488 };
01489 
01495 static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
01496 {
01497     int num_gain = 0;
01498     int c, g, sfb, ret;
01499     int sign;
01500     float scale;
01501     SingleChannelElement *sce = &che->ch[0];
01502     ChannelCoupling     *coup = &che->coup;
01503 
01504     coup->coupling_point = 2 * get_bits1(gb);
01505     coup->num_coupled = get_bits(gb, 3);
01506     for (c = 0; c <= coup->num_coupled; c++) {
01507         num_gain++;
01508         coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
01509         coup->id_select[c] = get_bits(gb, 4);
01510         if (coup->type[c] == TYPE_CPE) {
01511             coup->ch_select[c] = get_bits(gb, 2);
01512             if (coup->ch_select[c] == 3)
01513                 num_gain++;
01514         } else
01515             coup->ch_select[c] = 2;
01516     }
01517     coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
01518 
01519     sign  = get_bits(gb, 1);
01520     scale = cce_scale[get_bits(gb, 2)];
01521 
01522     if ((ret = decode_ics(ac, sce, gb, 0, 0)))
01523         return ret;
01524 
01525     for (c = 0; c < num_gain; c++) {
01526         int idx  = 0;
01527         int cge  = 1;
01528         int gain = 0;
01529         float gain_cache = 1.;
01530         if (c) {
01531             cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
01532             gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
01533             gain_cache = powf(scale, -gain);
01534         }
01535         if (coup->coupling_point == AFTER_IMDCT) {
01536             coup->gain[c][0] = gain_cache;
01537         } else {
01538             for (g = 0; g < sce->ics.num_window_groups; g++) {
01539                 for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
01540                     if (sce->band_type[idx] != ZERO_BT) {
01541                         if (!cge) {
01542                             int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
01543                             if (t) {
01544                                 int s = 1;
01545                                 t = gain += t;
01546                                 if (sign) {
01547                                     s  -= 2 * (t & 0x1);
01548                                     t >>= 1;
01549                                 }
01550                                 gain_cache = powf(scale, -t) * s;
01551                             }
01552                         }
01553                         coup->gain[c][idx] = gain_cache;
01554                     }
01555                 }
01556             }
01557         }
01558     }
01559     return 0;
01560 }
01561 
01567 static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
01568                                          GetBitContext *gb)
01569 {
01570     int i;
01571     int num_excl_chan = 0;
01572 
01573     do {
01574         for (i = 0; i < 7; i++)
01575             che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
01576     } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
01577 
01578     return num_excl_chan / 7;
01579 }
01580 
01588 static int decode_dynamic_range(DynamicRangeControl *che_drc,
01589                                 GetBitContext *gb, int cnt)
01590 {
01591     int n             = 1;
01592     int drc_num_bands = 1;
01593     int i;
01594 
01595     /* pce_tag_present? */
01596     if (get_bits1(gb)) {
01597         che_drc->pce_instance_tag  = get_bits(gb, 4);
01598         skip_bits(gb, 4); // tag_reserved_bits
01599         n++;
01600     }
01601 
01602     /* excluded_chns_present? */
01603     if (get_bits1(gb)) {
01604         n += decode_drc_channel_exclusions(che_drc, gb);
01605     }
01606 
01607     /* drc_bands_present? */
01608     if (get_bits1(gb)) {
01609         che_drc->band_incr            = get_bits(gb, 4);
01610         che_drc->interpolation_scheme = get_bits(gb, 4);
01611         n++;
01612         drc_num_bands += che_drc->band_incr;
01613         for (i = 0; i < drc_num_bands; i++) {
01614             che_drc->band_top[i] = get_bits(gb, 8);
01615             n++;
01616         }
01617     }
01618 
01619     /* prog_ref_level_present? */
01620     if (get_bits1(gb)) {
01621         che_drc->prog_ref_level = get_bits(gb, 7);
01622         skip_bits1(gb); // prog_ref_level_reserved_bits
01623         n++;
01624     }
01625 
01626     for (i = 0; i < drc_num_bands; i++) {
01627         che_drc->dyn_rng_sgn[i] = get_bits1(gb);
01628         che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
01629         n++;
01630     }
01631 
01632     return n;
01633 }
01634 
01642 static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
01643                                     ChannelElement *che, enum RawDataBlockType elem_type)
01644 {
01645     int crc_flag = 0;
01646     int res = cnt;
01647     switch (get_bits(gb, 4)) { // extension type
01648     case EXT_SBR_DATA_CRC:
01649         crc_flag++;
01650     case EXT_SBR_DATA:
01651         if (!che) {
01652             av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
01653             return res;
01654         } else if (!ac->m4ac.sbr) {
01655             av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
01656             skip_bits_long(gb, 8 * cnt - 4);
01657             return res;
01658         } else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) {
01659             av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
01660             skip_bits_long(gb, 8 * cnt - 4);
01661             return res;
01662         } else if (ac->m4ac.ps == -1 && ac->output_configured < OC_LOCKED && ac->avctx->channels == 1) {
01663             ac->m4ac.sbr = 1;
01664             ac->m4ac.ps = 1;
01665             output_configure(ac, ac->che_pos, ac->che_pos, ac->m4ac.chan_config, ac->output_configured);
01666         } else {
01667             ac->m4ac.sbr = 1;
01668         }
01669         res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
01670         break;
01671     case EXT_DYNAMIC_RANGE:
01672         res = decode_dynamic_range(&ac->che_drc, gb, cnt);
01673         break;
01674     case EXT_FILL:
01675     case EXT_FILL_DATA:
01676     case EXT_DATA_ELEMENT:
01677     default:
01678         skip_bits_long(gb, 8 * cnt - 4);
01679         break;
01680     };
01681     return res;
01682 }
01683 
01690 static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
01691                       IndividualChannelStream *ics, int decode)
01692 {
01693     const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
01694     int w, filt, m, i;
01695     int bottom, top, order, start, end, size, inc;
01696     float lpc[TNS_MAX_ORDER];
01697     float tmp[TNS_MAX_ORDER];
01698 
01699     for (w = 0; w < ics->num_windows; w++) {
01700         bottom = ics->num_swb;
01701         for (filt = 0; filt < tns->n_filt[w]; filt++) {
01702             top    = bottom;
01703             bottom = FFMAX(0, top - tns->length[w][filt]);
01704             order  = tns->order[w][filt];
01705             if (order == 0)
01706                 continue;
01707 
01708             // tns_decode_coef
01709             compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
01710 
01711             start = ics->swb_offset[FFMIN(bottom, mmm)];
01712             end   = ics->swb_offset[FFMIN(   top, mmm)];
01713             if ((size = end - start) <= 0)
01714                 continue;
01715             if (tns->direction[w][filt]) {
01716                 inc = -1;
01717                 start = end - 1;
01718             } else {
01719                 inc = 1;
01720             }
01721             start += w * 128;
01722 
01723             if (decode) {
01724                 // ar filter
01725                 for (m = 0; m < size; m++, start += inc)
01726                     for (i = 1; i <= FFMIN(m, order); i++)
01727                         coef[start] -= coef[start - i * inc] * lpc[i - 1];
01728             } else {
01729                 // ma filter
01730                 for (m = 0; m < size; m++, start += inc) {
01731                     tmp[0] = coef[start];
01732                     for (i = 1; i <= FFMIN(m, order); i++)
01733                         coef[start] += tmp[i] * lpc[i - 1];
01734                     for (i = order; i > 0; i--)
01735                         tmp[i] = tmp[i - 1];
01736                 }
01737             }
01738         }
01739     }
01740 }
01741 
01746 static void windowing_and_mdct_ltp(AACContext *ac, float *out,
01747                                    float *in, IndividualChannelStream *ics)
01748 {
01749     const float *lwindow      = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
01750     const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
01751     const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
01752     const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
01753 
01754     if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
01755         ac->dsp.vector_fmul(in, in, lwindow_prev, 1024);
01756     } else {
01757         memset(in, 0, 448 * sizeof(float));
01758         ac->dsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
01759     }
01760     if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
01761         ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
01762     } else {
01763         ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
01764         memset(in + 1024 + 576, 0, 448 * sizeof(float));
01765     }
01766     ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
01767 }
01768 
01772 static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
01773 {
01774     const LongTermPrediction *ltp = &sce->ics.ltp;
01775     const uint16_t *offsets = sce->ics.swb_offset;
01776     int i, sfb;
01777 
01778     if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
01779         float *predTime = sce->ret;
01780         float *predFreq = ac->buf_mdct;
01781         int16_t num_samples = 2048;
01782 
01783         if (ltp->lag < 1024)
01784             num_samples = ltp->lag + 1024;
01785         for (i = 0; i < num_samples; i++)
01786             predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
01787         memset(&predTime[i], 0, (2048 - i) * sizeof(float));
01788 
01789         windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
01790 
01791         if (sce->tns.present)
01792             apply_tns(predFreq, &sce->tns, &sce->ics, 0);
01793 
01794         for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
01795             if (ltp->used[sfb])
01796                 for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
01797                     sce->coeffs[i] += predFreq[i];
01798     }
01799 }
01800 
01804 static void update_ltp(AACContext *ac, SingleChannelElement *sce)
01805 {
01806     IndividualChannelStream *ics = &sce->ics;
01807     float *saved     = sce->saved;
01808     float *saved_ltp = sce->coeffs;
01809     const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
01810     const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
01811     int i;
01812 
01813     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
01814         memcpy(saved_ltp,       saved, 512 * sizeof(float));
01815         memset(saved_ltp + 576, 0,     448 * sizeof(float));
01816         ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
01817         for (i = 0; i < 64; i++)
01818             saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
01819     } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
01820         memcpy(saved_ltp,       ac->buf_mdct + 512, 448 * sizeof(float));
01821         memset(saved_ltp + 576, 0,                  448 * sizeof(float));
01822         ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
01823         for (i = 0; i < 64; i++)
01824             saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
01825     } else { // LONG_STOP or ONLY_LONG
01826         ac->dsp.vector_fmul_reverse(saved_ltp,       ac->buf_mdct + 512,     &lwindow[512],     512);
01827         for (i = 0; i < 512; i++)
01828             saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
01829     }
01830 
01831     memcpy(sce->ltp_state,      sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
01832     memcpy(sce->ltp_state+1024, sce->ret,            1024 * sizeof(*sce->ltp_state));
01833     memcpy(sce->ltp_state+2048, saved_ltp,           1024 * sizeof(*sce->ltp_state));
01834 }
01835 
01839 static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
01840 {
01841     IndividualChannelStream *ics = &sce->ics;
01842     float *in    = sce->coeffs;
01843     float *out   = sce->ret;
01844     float *saved = sce->saved;
01845     const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
01846     const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
01847     const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
01848     float *buf  = ac->buf_mdct;
01849     float *temp = ac->temp;
01850     int i;
01851 
01852     // imdct
01853     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
01854         for (i = 0; i < 1024; i += 128)
01855             ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
01856     } else
01857         ac->mdct.imdct_half(&ac->mdct, buf, in);
01858 
01859     /* window overlapping
01860      * NOTE: To simplify the overlapping code, all 'meaningless' short to long
01861      * and long to short transitions are considered to be short to short
01862      * transitions. This leaves just two cases (long to long and short to short)
01863      * with a little special sauce for EIGHT_SHORT_SEQUENCE.
01864      */
01865     if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
01866             (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
01867         ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);
01868     } else {
01869         memcpy(                        out,               saved,            448 * sizeof(float));
01870 
01871         if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
01872             ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, 64);
01873             ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      64);
01874             ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      64);
01875             ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      64);
01876             ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      64);
01877             memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
01878         } else {
01879             ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);
01880             memcpy(                    out + 576,         buf + 64,         448 * sizeof(float));
01881         }
01882     }
01883 
01884     // buffer update
01885     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
01886         memcpy(                    saved,       temp + 64,         64 * sizeof(float));
01887         ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 64);
01888         ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
01889         ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
01890         memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
01891     } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
01892         memcpy(                    saved,       buf + 512,        448 * sizeof(float));
01893         memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
01894     } else { // LONG_STOP or ONLY_LONG
01895         memcpy(                    saved,       buf + 512,        512 * sizeof(float));
01896     }
01897 }
01898 
01904 static void apply_dependent_coupling(AACContext *ac,
01905                                      SingleChannelElement *target,
01906                                      ChannelElement *cce, int index)
01907 {
01908     IndividualChannelStream *ics = &cce->ch[0].ics;
01909     const uint16_t *offsets = ics->swb_offset;
01910     float *dest = target->coeffs;
01911     const float *src = cce->ch[0].coeffs;
01912     int g, i, group, k, idx = 0;
01913     if (ac->m4ac.object_type == AOT_AAC_LTP) {
01914         av_log(ac->avctx, AV_LOG_ERROR,
01915                "Dependent coupling is not supported together with LTP\n");
01916         return;
01917     }
01918     for (g = 0; g < ics->num_window_groups; g++) {
01919         for (i = 0; i < ics->max_sfb; i++, idx++) {
01920             if (cce->ch[0].band_type[idx] != ZERO_BT) {
01921                 const float gain = cce->coup.gain[index][idx];
01922                 for (group = 0; group < ics->group_len[g]; group++) {
01923                     for (k = offsets[i]; k < offsets[i + 1]; k++) {
01924                         // XXX dsputil-ize
01925                         dest[group * 128 + k] += gain * src[group * 128 + k];
01926                     }
01927                 }
01928             }
01929         }
01930         dest += ics->group_len[g] * 128;
01931         src  += ics->group_len[g] * 128;
01932     }
01933 }
01934 
01940 static void apply_independent_coupling(AACContext *ac,
01941                                        SingleChannelElement *target,
01942                                        ChannelElement *cce, int index)
01943 {
01944     int i;
01945     const float gain = cce->coup.gain[index][0];
01946     const float *src = cce->ch[0].ret;
01947     float *dest = target->ret;
01948     const int len = 1024 << (ac->m4ac.sbr == 1);
01949 
01950     for (i = 0; i < len; i++)
01951         dest[i] += gain * src[i];
01952 }
01953 
01959 static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
01960                                    enum RawDataBlockType type, int elem_id,
01961                                    enum CouplingPoint coupling_point,
01962                                    void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
01963 {
01964     int i, c;
01965 
01966     for (i = 0; i < MAX_ELEM_ID; i++) {
01967         ChannelElement *cce = ac->che[TYPE_CCE][i];
01968         int index = 0;
01969 
01970         if (cce && cce->coup.coupling_point == coupling_point) {
01971             ChannelCoupling *coup = &cce->coup;
01972 
01973             for (c = 0; c <= coup->num_coupled; c++) {
01974                 if (coup->type[c] == type && coup->id_select[c] == elem_id) {
01975                     if (coup->ch_select[c] != 1) {
01976                         apply_coupling_method(ac, &cc->ch[0], cce, index);
01977                         if (coup->ch_select[c] != 0)
01978                             index++;
01979                     }
01980                     if (coup->ch_select[c] != 2)
01981                         apply_coupling_method(ac, &cc->ch[1], cce, index++);
01982                 } else
01983                     index += 1 + (coup->ch_select[c] == 3);
01984             }
01985         }
01986     }
01987 }
01988 
01992 static void spectral_to_sample(AACContext *ac)
01993 {
01994     int i, type;
01995     for (type = 3; type >= 0; type--) {
01996         for (i = 0; i < MAX_ELEM_ID; i++) {
01997             ChannelElement *che = ac->che[type][i];
01998             if (che) {
01999                 if (type <= TYPE_CPE)
02000                     apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
02001                 if (ac->m4ac.object_type == AOT_AAC_LTP) {
02002                     if (che->ch[0].ics.predictor_present) {
02003                         if (che->ch[0].ics.ltp.present)
02004                             apply_ltp(ac, &che->ch[0]);
02005                         if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
02006                             apply_ltp(ac, &che->ch[1]);
02007                     }
02008                 }
02009                 if (che->ch[0].tns.present)
02010                     apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
02011                 if (che->ch[1].tns.present)
02012                     apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
02013                 if (type <= TYPE_CPE)
02014                     apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
02015                 if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
02016                     imdct_and_windowing(ac, &che->ch[0]);
02017                     if (ac->m4ac.object_type == AOT_AAC_LTP)
02018                         update_ltp(ac, &che->ch[0]);
02019                     if (type == TYPE_CPE) {
02020                         imdct_and_windowing(ac, &che->ch[1]);
02021                         if (ac->m4ac.object_type == AOT_AAC_LTP)
02022                             update_ltp(ac, &che->ch[1]);
02023                     }
02024                     if (ac->m4ac.sbr > 0) {
02025                         ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
02026                     }
02027                 }
02028                 if (type <= TYPE_CCE)
02029                     apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
02030             }
02031         }
02032     }
02033 }
02034 
02035 static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
02036 {
02037     int size;
02038     AACADTSHeaderInfo hdr_info;
02039 
02040     size = ff_aac_parse_header(gb, &hdr_info);
02041     if (size > 0) {
02042         if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
02043             enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
02044             memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
02045             ac->m4ac.chan_config = hdr_info.chan_config;
02046             if (set_default_channel_config(ac->avctx, new_che_pos, hdr_info.chan_config))
02047                 return -7;
02048             if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
02049                 return -7;
02050         } else if (ac->output_configured != OC_LOCKED) {
02051             ac->output_configured = OC_NONE;
02052         }
02053         if (ac->output_configured != OC_LOCKED) {
02054             ac->m4ac.sbr = -1;
02055             ac->m4ac.ps  = -1;
02056         }
02057         ac->m4ac.sample_rate     = hdr_info.sample_rate;
02058         ac->m4ac.sampling_index  = hdr_info.sampling_index;
02059         ac->m4ac.object_type     = hdr_info.object_type;
02060         if (!ac->avctx->sample_rate)
02061             ac->avctx->sample_rate = hdr_info.sample_rate;
02062         if (hdr_info.num_aac_frames == 1) {
02063             if (!hdr_info.crc_absent)
02064                 skip_bits(gb, 16);
02065         } else {
02066             av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0);
02067             return -1;
02068         }
02069     }
02070     return size;
02071 }
02072 
02073 static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
02074                                 int *data_size, GetBitContext *gb)
02075 {
02076     AACContext *ac = avctx->priv_data;
02077     ChannelElement *che = NULL, *che_prev = NULL;
02078     enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
02079     int err, elem_id, data_size_tmp;
02080     int samples = 0, multiplier, audio_found = 0;
02081 
02082     if (show_bits(gb, 12) == 0xfff) {
02083         if (parse_adts_frame_header(ac, gb) < 0) {
02084             av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
02085             return -1;
02086         }
02087         if (ac->m4ac.sampling_index > 12) {
02088             av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
02089             return -1;
02090         }
02091     }
02092 
02093     ac->tags_mapped = 0;
02094     // parse
02095     while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
02096         elem_id = get_bits(gb, 4);
02097 
02098         if (elem_type < TYPE_DSE) {
02099             if (!(che=get_che(ac, elem_type, elem_id))) {
02100                 av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
02101                        elem_type, elem_id);
02102                 return -1;
02103             }
02104             samples = 1024;
02105         }
02106 
02107         switch (elem_type) {
02108 
02109         case TYPE_SCE:
02110             err = decode_ics(ac, &che->ch[0], gb, 0, 0);
02111             audio_found = 1;
02112             break;
02113 
02114         case TYPE_CPE:
02115             err = decode_cpe(ac, gb, che);
02116             audio_found = 1;
02117             break;
02118 
02119         case TYPE_CCE:
02120             err = decode_cce(ac, gb, che);
02121             break;
02122 
02123         case TYPE_LFE:
02124             err = decode_ics(ac, &che->ch[0], gb, 0, 0);
02125             audio_found = 1;
02126             break;
02127 
02128         case TYPE_DSE:
02129             err = skip_data_stream_element(ac, gb);
02130             break;
02131 
02132         case TYPE_PCE: {
02133             enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
02134             memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
02135             if ((err = decode_pce(avctx, &ac->m4ac, new_che_pos, gb)))
02136                 break;
02137             if (ac->output_configured > OC_TRIAL_PCE)
02138                 av_log(avctx, AV_LOG_ERROR,
02139                        "Not evaluating a further program_config_element as this construct is dubious at best.\n");
02140             else
02141                 err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
02142             break;
02143         }
02144 
02145         case TYPE_FIL:
02146             if (elem_id == 15)
02147                 elem_id += get_bits(gb, 8) - 1;
02148             if (get_bits_left(gb) < 8 * elem_id) {
02149                     av_log(avctx, AV_LOG_ERROR, overread_err);
02150                     return -1;
02151             }
02152             while (elem_id > 0)
02153                 elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
02154             err = 0; /* FIXME */
02155             break;
02156 
02157         default:
02158             err = -1; /* should not happen, but keeps compiler happy */
02159             break;
02160         }
02161 
02162         che_prev       = che;
02163         elem_type_prev = elem_type;
02164 
02165         if (err)
02166             return err;
02167 
02168         if (get_bits_left(gb) < 3) {
02169             av_log(avctx, AV_LOG_ERROR, overread_err);
02170             return -1;
02171         }
02172     }
02173 
02174     spectral_to_sample(ac);
02175 
02176     multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0;
02177     samples <<= multiplier;
02178     if (ac->output_configured < OC_LOCKED) {
02179         avctx->sample_rate = ac->m4ac.sample_rate << multiplier;
02180         avctx->frame_size = samples;
02181     }
02182 
02183     data_size_tmp = samples * avctx->channels *
02184                     av_get_bytes_per_sample(avctx->sample_fmt);
02185     if (*data_size < data_size_tmp) {
02186         av_log(avctx, AV_LOG_ERROR,
02187                "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
02188                *data_size, data_size_tmp);
02189         return -1;
02190     }
02191     *data_size = data_size_tmp;
02192 
02193     if (samples) {
02194         if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
02195             ac->fmt_conv.float_interleave(data, (const float **)ac->output_data,
02196                                           samples, avctx->channels);
02197         else
02198             ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data,
02199                                                    samples, avctx->channels);
02200     }
02201 
02202     if (ac->output_configured && audio_found)
02203         ac->output_configured = OC_LOCKED;
02204 
02205     return 0;
02206 }
02207 
02208 static int aac_decode_frame(AVCodecContext *avctx, void *data,
02209                             int *data_size, AVPacket *avpkt)
02210 {
02211     const uint8_t *buf = avpkt->data;
02212     int buf_size = avpkt->size;
02213     GetBitContext gb;
02214     int buf_consumed;
02215     int buf_offset;
02216     int err;
02217 
02218     init_get_bits(&gb, buf, buf_size * 8);
02219 
02220     if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0)
02221         return err;
02222 
02223     buf_consumed = (get_bits_count(&gb) + 7) >> 3;
02224     for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
02225         if (buf[buf_offset])
02226             break;
02227 
02228     return buf_size > buf_offset ? buf_consumed : buf_size;
02229 }
02230 
02231 static av_cold int aac_decode_close(AVCodecContext *avctx)
02232 {
02233     AACContext *ac = avctx->priv_data;
02234     int i, type;
02235 
02236     for (i = 0; i < MAX_ELEM_ID; i++) {
02237         for (type = 0; type < 4; type++) {
02238             if (ac->che[type][i])
02239                 ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
02240             av_freep(&ac->che[type][i]);
02241         }
02242     }
02243 
02244     ff_mdct_end(&ac->mdct);
02245     ff_mdct_end(&ac->mdct_small);
02246     ff_mdct_end(&ac->mdct_ltp);
02247     return 0;
02248 }
02249 
02250 
02251 #define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
02252 
02253 struct LATMContext {
02254     AACContext      aac_ctx;             
02255     int             initialized;         
02256 
02257     // parser data
02258     int             audio_mux_version_A; 
02259     int             frame_length_type;   
02260     int             frame_length;        
02261 };
02262 
02263 static inline uint32_t latm_get_value(GetBitContext *b)
02264 {
02265     int length = get_bits(b, 2);
02266 
02267     return get_bits_long(b, (length+1)*8);
02268 }
02269 
02270 static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
02271                                              GetBitContext *gb)
02272 {
02273     AVCodecContext *avctx = latmctx->aac_ctx.avctx;
02274     MPEG4AudioConfig m4ac;
02275     int  config_start_bit = get_bits_count(gb);
02276     int     bits_consumed, esize;
02277 
02278     if (config_start_bit % 8) {
02279         av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
02280                                "config not byte aligned.\n", 1);
02281         return AVERROR_INVALIDDATA;
02282     } else {
02283         bits_consumed =
02284             decode_audio_specific_config(NULL, avctx, &m4ac,
02285                                          gb->buffer + (config_start_bit / 8),
02286                                          get_bits_left(gb) / 8);
02287 
02288         if (bits_consumed < 0)
02289             return AVERROR_INVALIDDATA;
02290 
02291         esize = (bits_consumed+7) / 8;
02292 
02293         if (avctx->extradata_size <= esize) {
02294             av_free(avctx->extradata);
02295             avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
02296             if (!avctx->extradata)
02297                 return AVERROR(ENOMEM);
02298         }
02299 
02300         avctx->extradata_size = esize;
02301         memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
02302         memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
02303 
02304         skip_bits_long(gb, bits_consumed);
02305     }
02306 
02307     return bits_consumed;
02308 }
02309 
02310 static int read_stream_mux_config(struct LATMContext *latmctx,
02311                                   GetBitContext *gb)
02312 {
02313     int ret, audio_mux_version = get_bits(gb, 1);
02314 
02315     latmctx->audio_mux_version_A = 0;
02316     if (audio_mux_version)
02317         latmctx->audio_mux_version_A = get_bits(gb, 1);
02318 
02319     if (!latmctx->audio_mux_version_A) {
02320 
02321         if (audio_mux_version)
02322             latm_get_value(gb);                 // taraFullness
02323 
02324         skip_bits(gb, 1);                       // allStreamSameTimeFraming
02325         skip_bits(gb, 6);                       // numSubFrames
02326         // numPrograms
02327         if (get_bits(gb, 4)) {                  // numPrograms
02328             av_log_missing_feature(latmctx->aac_ctx.avctx,
02329                                    "multiple programs are not supported\n", 1);
02330             return AVERROR_PATCHWELCOME;
02331         }
02332 
02333         // for each program (which there is only on in DVB)
02334 
02335         // for each layer (which there is only on in DVB)
02336         if (get_bits(gb, 3)) {                   // numLayer
02337             av_log_missing_feature(latmctx->aac_ctx.avctx,
02338                                    "multiple layers are not supported\n", 1);
02339             return AVERROR_PATCHWELCOME;
02340         }
02341 
02342         // for all but first stream: use_same_config = get_bits(gb, 1);
02343         if (!audio_mux_version) {
02344             if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
02345                 return ret;
02346         } else {
02347             int ascLen = latm_get_value(gb);
02348             if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
02349                 return ret;
02350             ascLen -= ret;
02351             skip_bits_long(gb, ascLen);
02352         }
02353 
02354         latmctx->frame_length_type = get_bits(gb, 3);
02355         switch (latmctx->frame_length_type) {
02356         case 0:
02357             skip_bits(gb, 8);       // latmBufferFullness
02358             break;
02359         case 1:
02360             latmctx->frame_length = get_bits(gb, 9);
02361             break;
02362         case 3:
02363         case 4:
02364         case 5:
02365             skip_bits(gb, 6);       // CELP frame length table index
02366             break;
02367         case 6:
02368         case 7:
02369             skip_bits(gb, 1);       // HVXC frame length table index
02370             break;
02371         }
02372 
02373         if (get_bits(gb, 1)) {                  // other data
02374             if (audio_mux_version) {
02375                 latm_get_value(gb);             // other_data_bits
02376             } else {
02377                 int esc;
02378                 do {
02379                     esc = get_bits(gb, 1);
02380                     skip_bits(gb, 8);
02381                 } while (esc);
02382             }
02383         }
02384 
02385         if (get_bits(gb, 1))                     // crc present
02386             skip_bits(gb, 8);                    // config_crc
02387     }
02388 
02389     return 0;
02390 }
02391 
02392 static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
02393 {
02394     uint8_t tmp;
02395 
02396     if (ctx->frame_length_type == 0) {
02397         int mux_slot_length = 0;
02398         do {
02399             tmp = get_bits(gb, 8);
02400             mux_slot_length += tmp;
02401         } while (tmp == 255);
02402         return mux_slot_length;
02403     } else if (ctx->frame_length_type == 1) {
02404         return ctx->frame_length;
02405     } else if (ctx->frame_length_type == 3 ||
02406                ctx->frame_length_type == 5 ||
02407                ctx->frame_length_type == 7) {
02408         skip_bits(gb, 2);          // mux_slot_length_coded
02409     }
02410     return 0;
02411 }
02412 
02413 static int read_audio_mux_element(struct LATMContext *latmctx,
02414                                   GetBitContext *gb)
02415 {
02416     int err;
02417     uint8_t use_same_mux = get_bits(gb, 1);
02418     if (!use_same_mux) {
02419         if ((err = read_stream_mux_config(latmctx, gb)) < 0)
02420             return err;
02421     } else if (!latmctx->aac_ctx.avctx->extradata) {
02422         av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
02423                "no decoder config found\n");
02424         return AVERROR(EAGAIN);
02425     }
02426     if (latmctx->audio_mux_version_A == 0) {
02427         int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
02428         if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
02429             av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
02430             return AVERROR_INVALIDDATA;
02431         } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
02432             av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
02433                    "frame length mismatch %d << %d\n",
02434                    mux_slot_length_bytes * 8, get_bits_left(gb));
02435             return AVERROR_INVALIDDATA;
02436         }
02437     }
02438     return 0;
02439 }
02440 
02441 
02442 static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
02443                              AVPacket *avpkt)
02444 {
02445     struct LATMContext *latmctx = avctx->priv_data;
02446     int                 muxlength, err;
02447     GetBitContext       gb;
02448 
02449     if (avpkt->size == 0)
02450         return 0;
02451 
02452     init_get_bits(&gb, avpkt->data, avpkt->size * 8);
02453 
02454     // check for LOAS sync word
02455     if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
02456         return AVERROR_INVALIDDATA;
02457 
02458     muxlength = get_bits(&gb, 13) + 3;
02459     // not enough data, the parser should have sorted this
02460     if (muxlength > avpkt->size)
02461         return AVERROR_INVALIDDATA;
02462 
02463     if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
02464         return err;
02465 
02466     if (!latmctx->initialized) {
02467         if (!avctx->extradata) {
02468             *out_size = 0;
02469             return avpkt->size;
02470         } else {
02471             aac_decode_close(avctx);
02472             if ((err = aac_decode_init(avctx)) < 0)
02473                 return err;
02474             latmctx->initialized = 1;
02475         }
02476     }
02477 
02478     if (show_bits(&gb, 12) == 0xfff) {
02479         av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
02480                "ADTS header detected, probably as result of configuration "
02481                "misparsing\n");
02482         return AVERROR_INVALIDDATA;
02483     }
02484 
02485     if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
02486         return err;
02487 
02488     return muxlength;
02489 }
02490 
02491 av_cold static int latm_decode_init(AVCodecContext *avctx)
02492 {
02493     struct LATMContext *latmctx = avctx->priv_data;
02494     int ret;
02495 
02496     ret = aac_decode_init(avctx);
02497 
02498     if (avctx->extradata_size > 0) {
02499         latmctx->initialized = !ret;
02500     } else {
02501         latmctx->initialized = 0;
02502     }
02503 
02504     return ret;
02505 }
02506 
02507 
02508 AVCodec ff_aac_decoder = {
02509     "aac",
02510     AVMEDIA_TYPE_AUDIO,
02511     CODEC_ID_AAC,
02512     sizeof(AACContext),
02513     aac_decode_init,
02514     NULL,
02515     aac_decode_close,
02516     aac_decode_frame,
02517     .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
02518     .sample_fmts = (const enum AVSampleFormat[]) {
02519         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
02520     },
02521     .channel_layouts = aac_channel_layout,
02522 };
02523 
02524 /*
02525     Note: This decoder filter is intended to decode LATM streams transferred
02526     in MPEG transport streams which only contain one program.
02527     To do a more complex LATM demuxing a separate LATM demuxer should be used.
02528 */
02529 AVCodec ff_aac_latm_decoder = {
02530     .name = "aac_latm",
02531     .type = AVMEDIA_TYPE_AUDIO,
02532     .id   = CODEC_ID_AAC_LATM,
02533     .priv_data_size = sizeof(struct LATMContext),
02534     .init   = latm_decode_init,
02535     .close  = aac_decode_close,
02536     .decode = latm_decode_frame,
02537     .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
02538     .sample_fmts = (const enum AVSampleFormat[]) {
02539         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
02540     },
02541     .channel_layouts = aac_channel_layout,
02542 };

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