Mercurial > libavcodec.hg
changeset 11834:a469cc7f9edb libavcodec
Rename aac.c to aacdec.c.
| author | alexc |
|---|---|
| date | Sat, 05 Jun 2010 15:27:53 +0000 |
| parents | 9103a9b3573a |
| children | 758c052eb8a9 |
| files | Makefile aac.c aacdec.c |
| diffstat | 3 files changed, 2126 insertions(+), 2126 deletions(-) [+] |
line wrap: on
line diff
--- a/Makefile Sat Jun 05 15:22:19 2010 +0000 +++ b/Makefile Sat Jun 05 15:27:53 2010 +0000 @@ -42,7 +42,7 @@ OBJS-$(CONFIG_VDPAU) += vdpau.o # decoders/encoders/hardware accelerators -OBJS-$(CONFIG_AAC_DECODER) += aac.o aactab.o aacsbr.o +OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o \ aacpsy.o aactab.o \ psymodel.o iirfilter.o \
--- a/aac.c Sat Jun 05 15:22:19 2010 +0000 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,2125 +0,0 @@ -/* - * AAC decoder - * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) - * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) - * - * This file is part of FFmpeg. - * - * FFmpeg is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public - * License as published by the Free Software Foundation; either - * version 2.1 of the License, or (at your option) any later version. - * - * FFmpeg is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -/** - * @file - * AAC decoder - * @author Oded Shimon ( ods15 ods15 dyndns org ) - * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) - */ - -/* - * supported tools - * - * Support? Name - * N (code in SoC repo) gain control - * Y block switching - * Y window shapes - standard - * N window shapes - Low Delay - * Y filterbank - standard - * N (code in SoC repo) filterbank - Scalable Sample Rate - * Y Temporal Noise Shaping - * N (code in SoC repo) Long Term Prediction - * Y intensity stereo - * Y channel coupling - * Y frequency domain prediction - * Y Perceptual Noise Substitution - * Y Mid/Side stereo - * N Scalable Inverse AAC Quantization - * N Frequency Selective Switch - * N upsampling filter - * Y quantization & coding - AAC - * N quantization & coding - TwinVQ - * N quantization & coding - BSAC - * N AAC Error Resilience tools - * N Error Resilience payload syntax - * N Error Protection tool - * N CELP - * N Silence Compression - * N HVXC - * N HVXC 4kbits/s VR - * N Structured Audio tools - * N Structured Audio Sample Bank Format - * N MIDI - * N Harmonic and Individual Lines plus Noise - * N Text-To-Speech Interface - * Y Spectral Band Replication - * Y (not in this code) Layer-1 - * Y (not in this code) Layer-2 - * Y (not in this code) Layer-3 - * N SinuSoidal Coding (Transient, Sinusoid, Noise) - * N (planned) Parametric Stereo - * N Direct Stream Transfer - * - * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication. - * - HE AAC v2 comprises LC AAC with Spectral Band Replication and - Parametric Stereo. - */ - - -#include "avcodec.h" -#include "internal.h" -#include "get_bits.h" -#include "dsputil.h" -#include "fft.h" -#include "lpc.h" - -#include "aac.h" -#include "aactab.h" -#include "aacdectab.h" -#include "cbrt_tablegen.h" -#include "sbr.h" -#include "aacsbr.h" -#include "mpeg4audio.h" -#include "aac_parser.h" - -#include <assert.h> -#include <errno.h> -#include <math.h> -#include <string.h> - -#if ARCH_ARM -# include "arm/aac.h" -#endif - -union float754 { - float f; - uint32_t i; -}; - -static VLC vlc_scalefactors; -static VLC vlc_spectral[11]; - -static const char overread_err[] = "Input buffer exhausted before END element found\n"; - -static ChannelElement *get_che(AACContext *ac, int type, int elem_id) -{ - /* Some buggy encoders appear to set all elem_ids to zero and rely on - channels always occurring in the same order. This is expressly forbidden - by the spec but we will try to work around it. - */ - int err_printed = 0; - while (ac->tags_seen_this_frame[type][elem_id] && elem_id < MAX_ELEM_ID) { - if (ac->output_configured < OC_LOCKED && !err_printed) { - av_log(ac->avctx, AV_LOG_WARNING, "Duplicate channel tag found, attempting to remap.\n"); - err_printed = 1; - } - elem_id++; - } - if (elem_id == MAX_ELEM_ID) - return NULL; - ac->tags_seen_this_frame[type][elem_id] = 1; - - if (ac->tag_che_map[type][elem_id]) { - return ac->tag_che_map[type][elem_id]; - } - if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) { - return NULL; - } - switch (ac->m4ac.chan_config) { - case 7: - if (ac->tags_mapped == 3 && type == TYPE_CPE) { - ac->tags_mapped++; - return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2]; - } - case 6: - /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1] - instead of SCE[0] CPE[0] CPE[1] LFE[0]. If we seem to have - encountered such a stream, transfer the LFE[0] element to the SCE[1]'s mapping */ - if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) { - ac->tags_mapped++; - return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0]; - } - case 5: - if (ac->tags_mapped == 2 && type == TYPE_CPE) { - ac->tags_mapped++; - return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1]; - } - case 4: - if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) { - ac->tags_mapped++; - return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; - } - case 3: - case 2: - if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) { - ac->tags_mapped++; - return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0]; - } else if (ac->m4ac.chan_config == 2) { - return NULL; - } - case 1: - if (!ac->tags_mapped && type == TYPE_SCE) { - ac->tags_mapped++; - return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0]; - } - default: - return NULL; - } -} - -/** - * Check for the channel element in the current channel position configuration. - * If it exists, make sure the appropriate element is allocated and map the - * channel order to match the internal FFmpeg channel layout. - * - * @param che_pos current channel position configuration - * @param type channel element type - * @param id channel element id - * @param channels count of the number of channels in the configuration - * - * @return Returns error status. 0 - OK, !0 - error - */ -static av_cold int che_configure(AACContext *ac, - enum ChannelPosition che_pos[4][MAX_ELEM_ID], - int type, int id, - int *channels) -{ - if (che_pos[type][id]) { - if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement)))) - return AVERROR(ENOMEM); - ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr); - if (type != TYPE_CCE) { - ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret; - if (type == TYPE_CPE) { - ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret; - } - } - } else { - if (ac->che[type][id]) - ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr); - av_freep(&ac->che[type][id]); - } - return 0; -} - -/** - * Configure output channel order based on the current program configuration element. - * - * @param che_pos current channel position configuration - * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. - * - * @return Returns error status. 0 - OK, !0 - error - */ -static av_cold int output_configure(AACContext *ac, - enum ChannelPosition che_pos[4][MAX_ELEM_ID], - enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], - int channel_config, enum OCStatus oc_type) -{ - AVCodecContext *avctx = ac->avctx; - int i, type, channels = 0, ret; - - memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); - - if (channel_config) { - for (i = 0; i < tags_per_config[channel_config]; i++) { - if ((ret = che_configure(ac, che_pos, - aac_channel_layout_map[channel_config - 1][i][0], - aac_channel_layout_map[channel_config - 1][i][1], - &channels))) - return ret; - } - - memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); - ac->tags_mapped = 0; - - avctx->channel_layout = aac_channel_layout[channel_config - 1]; - } else { - /* Allocate or free elements depending on if they are in the - * current program configuration. - * - * Set up default 1:1 output mapping. - * - * For a 5.1 stream the output order will be: - * [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ] - */ - - for (i = 0; i < MAX_ELEM_ID; i++) { - for (type = 0; type < 4; type++) { - if ((ret = che_configure(ac, che_pos, type, i, &channels))) - return ret; - } - } - - memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); - ac->tags_mapped = 4 * MAX_ELEM_ID; - - avctx->channel_layout = 0; - } - - avctx->channels = channels; - - ac->output_configured = oc_type; - - return 0; -} - -/** - * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. - * - * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. - * @param sce_map mono (Single Channel Element) map - * @param type speaker type/position for these channels - */ -static void decode_channel_map(enum ChannelPosition *cpe_map, - enum ChannelPosition *sce_map, - enum ChannelPosition type, - GetBitContext *gb, int n) -{ - while (n--) { - enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map - map[get_bits(gb, 4)] = type; - } -} - -/** - * Decode program configuration element; reference: table 4.2. - * - * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], - GetBitContext *gb) -{ - int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index; - int comment_len; - - skip_bits(gb, 2); // object_type - - sampling_index = get_bits(gb, 4); - if (ac->m4ac.sampling_index != sampling_index) - av_log(ac->avctx, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n"); - - num_front = get_bits(gb, 4); - num_side = get_bits(gb, 4); - num_back = get_bits(gb, 4); - num_lfe = get_bits(gb, 2); - num_assoc_data = get_bits(gb, 3); - num_cc = get_bits(gb, 4); - - if (get_bits1(gb)) - skip_bits(gb, 4); // mono_mixdown_tag - if (get_bits1(gb)) - skip_bits(gb, 4); // stereo_mixdown_tag - - if (get_bits1(gb)) - skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround - - decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); - decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); - decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); - decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); - - skip_bits_long(gb, 4 * num_assoc_data); - - decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); - - align_get_bits(gb); - - /* comment field, first byte is length */ - comment_len = get_bits(gb, 8) * 8; - if (get_bits_left(gb) < comment_len) { - av_log(ac->avctx, AV_LOG_ERROR, overread_err); - return -1; - } - skip_bits_long(gb, comment_len); - return 0; -} - -/** - * Set up channel positions based on a default channel configuration - * as specified in table 1.17. - * - * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. - * - * @return Returns error status. 0 - OK, !0 - error - */ -static av_cold int set_default_channel_config(AACContext *ac, - enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], - int channel_config) -{ - if (channel_config < 1 || channel_config > 7) { - av_log(ac->avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", - channel_config); - return -1; - } - - /* default channel configurations: - * - * 1ch : front center (mono) - * 2ch : L + R (stereo) - * 3ch : front center + L + R - * 4ch : front center + L + R + back center - * 5ch : front center + L + R + back stereo - * 6ch : front center + L + R + back stereo + LFE - * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE - */ - - if (channel_config != 2) - new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) - if (channel_config > 1) - new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) - if (channel_config == 4) - new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center - if (channel_config > 4) - new_che_pos[TYPE_CPE][(channel_config == 7) + 1] - = AAC_CHANNEL_BACK; // back stereo - if (channel_config > 5) - new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE - if (channel_config == 7) - new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right - - return 0; -} - -/** - * Decode GA "General Audio" specific configuration; reference: table 4.1. - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_ga_specific_config(AACContext *ac, GetBitContext *gb, - int channel_config) -{ - enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; - int extension_flag, ret; - - if (get_bits1(gb)) { // frameLengthFlag - av_log_missing_feature(ac->avctx, "960/120 MDCT window is", 1); - return -1; - } - - if (get_bits1(gb)) // dependsOnCoreCoder - skip_bits(gb, 14); // coreCoderDelay - extension_flag = get_bits1(gb); - - if (ac->m4ac.object_type == AOT_AAC_SCALABLE || - ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) - skip_bits(gb, 3); // layerNr - - memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); - if (channel_config == 0) { - skip_bits(gb, 4); // element_instance_tag - if ((ret = decode_pce(ac, new_che_pos, gb))) - return ret; - } else { - if ((ret = set_default_channel_config(ac, new_che_pos, channel_config))) - return ret; - } - if ((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR))) - return ret; - - if (extension_flag) { - switch (ac->m4ac.object_type) { - case AOT_ER_BSAC: - skip_bits(gb, 5); // numOfSubFrame - skip_bits(gb, 11); // layer_length - break; - case AOT_ER_AAC_LC: - case AOT_ER_AAC_LTP: - case AOT_ER_AAC_SCALABLE: - case AOT_ER_AAC_LD: - skip_bits(gb, 3); /* aacSectionDataResilienceFlag - * aacScalefactorDataResilienceFlag - * aacSpectralDataResilienceFlag - */ - break; - } - skip_bits1(gb); // extensionFlag3 (TBD in version 3) - } - return 0; -} - -/** - * Decode audio specific configuration; reference: table 1.13. - * - * @param data pointer to AVCodecContext extradata - * @param data_size size of AVCCodecContext extradata - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_audio_specific_config(AACContext *ac, void *data, - int data_size) -{ - GetBitContext gb; - int i; - - init_get_bits(&gb, data, data_size * 8); - - if ((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) - return -1; - if (ac->m4ac.sampling_index > 12) { - av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); - return -1; - } - - skip_bits_long(&gb, i); - - switch (ac->m4ac.object_type) { - case AOT_AAC_MAIN: - case AOT_AAC_LC: - if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) - return -1; - break; - default: - av_log(ac->avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", - ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); - return -1; - } - return 0; -} - -/** - * linear congruential pseudorandom number generator - * - * @param previous_val pointer to the current state of the generator - * - * @return Returns a 32-bit pseudorandom integer - */ -static av_always_inline int lcg_random(int previous_val) -{ - return previous_val * 1664525 + 1013904223; -} - -static av_always_inline void reset_predict_state(PredictorState *ps) -{ - ps->r0 = 0.0f; - ps->r1 = 0.0f; - ps->cor0 = 0.0f; - ps->cor1 = 0.0f; - ps->var0 = 1.0f; - ps->var1 = 1.0f; -} - -static void reset_all_predictors(PredictorState *ps) -{ - int i; - for (i = 0; i < MAX_PREDICTORS; i++) - reset_predict_state(&ps[i]); -} - -static void reset_predictor_group(PredictorState *ps, int group_num) -{ - int i; - for (i = group_num - 1; i < MAX_PREDICTORS; i += 30) - reset_predict_state(&ps[i]); -} - -static av_cold int aac_decode_init(AVCodecContext *avctx) -{ - AACContext *ac = avctx->priv_data; - int i; - - ac->avctx = avctx; - ac->m4ac.sample_rate = avctx->sample_rate; - - if (avctx->extradata_size > 0) { - if (decode_audio_specific_config(ac, avctx->extradata, avctx->extradata_size)) - return -1; - } - - avctx->sample_fmt = SAMPLE_FMT_S16; - - AAC_INIT_VLC_STATIC( 0, 304); - AAC_INIT_VLC_STATIC( 1, 270); - AAC_INIT_VLC_STATIC( 2, 550); - AAC_INIT_VLC_STATIC( 3, 300); - AAC_INIT_VLC_STATIC( 4, 328); - AAC_INIT_VLC_STATIC( 5, 294); - AAC_INIT_VLC_STATIC( 6, 306); - AAC_INIT_VLC_STATIC( 7, 268); - AAC_INIT_VLC_STATIC( 8, 510); - AAC_INIT_VLC_STATIC( 9, 366); - AAC_INIT_VLC_STATIC(10, 462); - - ff_aac_sbr_init(); - - dsputil_init(&ac->dsp, avctx); - - ac->random_state = 0x1f2e3d4c; - - // -1024 - Compensate wrong IMDCT method. - // 32768 - Required to scale values to the correct range for the bias method - // for float to int16 conversion. - - if (ac->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) { - ac->add_bias = 385.0f; - ac->sf_scale = 1. / (-1024. * 32768.); - ac->sf_offset = 0; - } else { - ac->add_bias = 0.0f; - ac->sf_scale = 1. / -1024.; - ac->sf_offset = 60; - } - -#if !CONFIG_HARDCODED_TABLES - for (i = 0; i < 428; i++) - ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.); -#endif /* CONFIG_HARDCODED_TABLES */ - - INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code), - ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), - ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), - 352); - - ff_mdct_init(&ac->mdct, 11, 1, 1.0); - ff_mdct_init(&ac->mdct_small, 8, 1, 1.0); - // window initialization - ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); - ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); - ff_init_ff_sine_windows(10); - ff_init_ff_sine_windows( 7); - - cbrt_tableinit(); - - return 0; -} - -/** - * Skip data_stream_element; reference: table 4.10. - */ -static int skip_data_stream_element(AACContext *ac, GetBitContext *gb) -{ - int byte_align = get_bits1(gb); - int count = get_bits(gb, 8); - if (count == 255) - count += get_bits(gb, 8); - if (byte_align) - align_get_bits(gb); - - if (get_bits_left(gb) < 8 * count) { - av_log(ac->avctx, AV_LOG_ERROR, overread_err); - return -1; - } - skip_bits_long(gb, 8 * count); - return 0; -} - -static int decode_prediction(AACContext *ac, IndividualChannelStream *ics, - GetBitContext *gb) -{ - int sfb; - if (get_bits1(gb)) { - ics->predictor_reset_group = get_bits(gb, 5); - if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) { - av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n"); - return -1; - } - } - for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) { - ics->prediction_used[sfb] = get_bits1(gb); - } - return 0; -} - -/** - * Decode Individual Channel Stream info; reference: table 4.6. - * - * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. - */ -static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics, - GetBitContext *gb, int common_window) -{ - if (get_bits1(gb)) { - av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n"); - memset(ics, 0, sizeof(IndividualChannelStream)); - return -1; - } - ics->window_sequence[1] = ics->window_sequence[0]; - ics->window_sequence[0] = get_bits(gb, 2); - ics->use_kb_window[1] = ics->use_kb_window[0]; - ics->use_kb_window[0] = get_bits1(gb); - ics->num_window_groups = 1; - ics->group_len[0] = 1; - if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { - int i; - ics->max_sfb = get_bits(gb, 4); - for (i = 0; i < 7; i++) { - if (get_bits1(gb)) { - ics->group_len[ics->num_window_groups - 1]++; - } else { - ics->num_window_groups++; - ics->group_len[ics->num_window_groups - 1] = 1; - } - } - ics->num_windows = 8; - ics->swb_offset = ff_swb_offset_128[ac->m4ac.sampling_index]; - ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; - ics->tns_max_bands = ff_tns_max_bands_128[ac->m4ac.sampling_index]; - ics->predictor_present = 0; - } else { - ics->max_sfb = get_bits(gb, 6); - ics->num_windows = 1; - ics->swb_offset = ff_swb_offset_1024[ac->m4ac.sampling_index]; - ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; - ics->tns_max_bands = ff_tns_max_bands_1024[ac->m4ac.sampling_index]; - ics->predictor_present = get_bits1(gb); - ics->predictor_reset_group = 0; - if (ics->predictor_present) { - if (ac->m4ac.object_type == AOT_AAC_MAIN) { - if (decode_prediction(ac, ics, gb)) { - memset(ics, 0, sizeof(IndividualChannelStream)); - return -1; - } - } else if (ac->m4ac.object_type == AOT_AAC_LC) { - av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n"); - memset(ics, 0, sizeof(IndividualChannelStream)); - return -1; - } else { - av_log_missing_feature(ac->avctx, "Predictor bit set but LTP is", 1); - memset(ics, 0, sizeof(IndividualChannelStream)); - return -1; - } - } - } - - if (ics->max_sfb > ics->num_swb) { - av_log(ac->avctx, AV_LOG_ERROR, - "Number of scalefactor bands in group (%d) exceeds limit (%d).\n", - ics->max_sfb, ics->num_swb); - memset(ics, 0, sizeof(IndividualChannelStream)); - return -1; - } - - return 0; -} - -/** - * Decode band types (section_data payload); reference: table 4.46. - * - * @param band_type array of the used band type - * @param band_type_run_end array of the last scalefactor band of a band type run - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_band_types(AACContext *ac, enum BandType band_type[120], - int band_type_run_end[120], GetBitContext *gb, - IndividualChannelStream *ics) -{ - int g, idx = 0; - const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; - for (g = 0; g < ics->num_window_groups; g++) { - int k = 0; - while (k < ics->max_sfb) { - uint8_t sect_end = k; - int sect_len_incr; - int sect_band_type = get_bits(gb, 4); - if (sect_band_type == 12) { - av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n"); - return -1; - } - while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1) - sect_end += sect_len_incr; - sect_end += sect_len_incr; - if (get_bits_left(gb) < 0) { - av_log(ac->avctx, AV_LOG_ERROR, overread_err); - return -1; - } - if (sect_end > ics->max_sfb) { - av_log(ac->avctx, AV_LOG_ERROR, - "Number of bands (%d) exceeds limit (%d).\n", - sect_end, ics->max_sfb); - return -1; - } - for (; k < sect_end; k++) { - band_type [idx] = sect_band_type; - band_type_run_end[idx++] = sect_end; - } - } - } - return 0; -} - -/** - * Decode scalefactors; reference: table 4.47. - * - * @param global_gain first scalefactor value as scalefactors are differentially coded - * @param band_type array of the used band type - * @param band_type_run_end array of the last scalefactor band of a band type run - * @param sf array of scalefactors or intensity stereo positions - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb, - unsigned int global_gain, - IndividualChannelStream *ics, - enum BandType band_type[120], - int band_type_run_end[120]) -{ - const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); - int g, i, idx = 0; - int offset[3] = { global_gain, global_gain - 90, 100 }; - int noise_flag = 1; - static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; - for (g = 0; g < ics->num_window_groups; g++) { - for (i = 0; i < ics->max_sfb;) { - int run_end = band_type_run_end[idx]; - if (band_type[idx] == ZERO_BT) { - for (; i < run_end; i++, idx++) - sf[idx] = 0.; - } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { - for (; i < run_end; i++, idx++) { - offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; - if (offset[2] > 255U) { - av_log(ac->avctx, AV_LOG_ERROR, - "%s (%d) out of range.\n", sf_str[2], offset[2]); - return -1; - } - sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; - } - } else if (band_type[idx] == NOISE_BT) { - for (; i < run_end; i++, idx++) { - if (noise_flag-- > 0) - offset[1] += get_bits(gb, 9) - 256; - else - offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; - if (offset[1] > 255U) { - av_log(ac->avctx, AV_LOG_ERROR, - "%s (%d) out of range.\n", sf_str[1], offset[1]); - return -1; - } - sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100]; - } - } else { - for (; i < run_end; i++, idx++) { - offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; - if (offset[0] > 255U) { - av_log(ac->avctx, AV_LOG_ERROR, - "%s (%d) out of range.\n", sf_str[0], offset[0]); - return -1; - } - sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; - } - } - } - } - return 0; -} - -/** - * Decode pulse data; reference: table 4.7. - */ -static int decode_pulses(Pulse *pulse, GetBitContext *gb, - const uint16_t *swb_offset, int num_swb) -{ - int i, pulse_swb; - pulse->num_pulse = get_bits(gb, 2) + 1; - pulse_swb = get_bits(gb, 6); - if (pulse_swb >= num_swb) - return -1; - pulse->pos[0] = swb_offset[pulse_swb]; - pulse->pos[0] += get_bits(gb, 5); - if (pulse->pos[0] > 1023) - return -1; - pulse->amp[0] = get_bits(gb, 4); - for (i = 1; i < pulse->num_pulse; i++) { - pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1]; - if (pulse->pos[i] > 1023) - return -1; - pulse->amp[i] = get_bits(gb, 4); - } - return 0; -} - -/** - * Decode Temporal Noise Shaping data; reference: table 4.48. - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns, - GetBitContext *gb, const IndividualChannelStream *ics) -{ - int w, filt, i, coef_len, coef_res, coef_compress; - const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; - const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; - for (w = 0; w < ics->num_windows; w++) { - if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) { - coef_res = get_bits1(gb); - - for (filt = 0; filt < tns->n_filt[w]; filt++) { - int tmp2_idx; - tns->length[w][filt] = get_bits(gb, 6 - 2 * is8); - - if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) { - av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n", - tns->order[w][filt], tns_max_order); - tns->order[w][filt] = 0; - return -1; - } - if (tns->order[w][filt]) { - tns->direction[w][filt] = get_bits1(gb); - coef_compress = get_bits1(gb); - coef_len = coef_res + 3 - coef_compress; - tmp2_idx = 2 * coef_compress + coef_res; - - for (i = 0; i < tns->order[w][filt]; i++) - tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; - } - } - } - } - return 0; -} - -/** - * Decode Mid/Side data; reference: table 4.54. - * - * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; - * [1] mask is decoded from bitstream; [2] mask is all 1s; - * [3] reserved for scalable AAC - */ -static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb, - int ms_present) -{ - int idx; - if (ms_present == 1) { - for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++) - cpe->ms_mask[idx] = get_bits1(gb); - } else if (ms_present == 2) { - memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0])); - } -} - -#ifndef VMUL2 -static inline float *VMUL2(float *dst, const float *v, unsigned idx, - const float *scale) -{ - float s = *scale; - *dst++ = v[idx & 15] * s; - *dst++ = v[idx>>4 & 15] * s; - return dst; -} -#endif - -#ifndef VMUL4 -static inline float *VMUL4(float *dst, const float *v, unsigned idx, - const float *scale) -{ - float s = *scale; - *dst++ = v[idx & 3] * s; - *dst++ = v[idx>>2 & 3] * s; - *dst++ = v[idx>>4 & 3] * s; - *dst++ = v[idx>>6 & 3] * s; - return dst; -} -#endif - -#ifndef VMUL2S -static inline float *VMUL2S(float *dst, const float *v, unsigned idx, - unsigned sign, const float *scale) -{ - union float754 s0, s1; - - s0.f = s1.f = *scale; - s0.i ^= sign >> 1 << 31; - s1.i ^= sign << 31; - - *dst++ = v[idx & 15] * s0.f; - *dst++ = v[idx>>4 & 15] * s1.f; - - return dst; -} -#endif - -#ifndef VMUL4S -static inline float *VMUL4S(float *dst, const float *v, unsigned idx, - unsigned sign, const float *scale) -{ - unsigned nz = idx >> 12; - union float754 s = { .f = *scale }; - union float754 t; - - t.i = s.i ^ (sign & 1<<31); - *dst++ = v[idx & 3] * t.f; - - sign <<= nz & 1; nz >>= 1; - t.i = s.i ^ (sign & 1<<31); - *dst++ = v[idx>>2 & 3] * t.f; - - sign <<= nz & 1; nz >>= 1; - t.i = s.i ^ (sign & 1<<31); - *dst++ = v[idx>>4 & 3] * t.f; - - sign <<= nz & 1; nz >>= 1; - t.i = s.i ^ (sign & 1<<31); - *dst++ = v[idx>>6 & 3] * t.f; - - return dst; -} -#endif - -/** - * Decode spectral data; reference: table 4.50. - * Dequantize and scale spectral data; reference: 4.6.3.3. - * - * @param coef array of dequantized, scaled spectral data - * @param sf array of scalefactors or intensity stereo positions - * @param pulse_present set if pulses are present - * @param pulse pointer to pulse data struct - * @param band_type array of the used band type - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024], - GetBitContext *gb, const float sf[120], - int pulse_present, const Pulse *pulse, - const IndividualChannelStream *ics, - enum BandType band_type[120]) -{ - int i, k, g, idx = 0; - const int c = 1024 / ics->num_windows; - const uint16_t *offsets = ics->swb_offset; - float *coef_base = coef; - int err_idx; - - for (g = 0; g < ics->num_windows; g++) - memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb])); - - for (g = 0; g < ics->num_window_groups; g++) { - unsigned g_len = ics->group_len[g]; - - for (i = 0; i < ics->max_sfb; i++, idx++) { - const unsigned cbt_m1 = band_type[idx] - 1; - float *cfo = coef + offsets[i]; - int off_len = offsets[i + 1] - offsets[i]; - int group; - - if (cbt_m1 >= INTENSITY_BT2 - 1) { - for (group = 0; group < g_len; group++, cfo+=128) { - memset(cfo, 0, off_len * sizeof(float)); - } - } else if (cbt_m1 == NOISE_BT - 1) { - for (group = 0; group < g_len; group++, cfo+=128) { - float scale; - float band_energy; - - for (k = 0; k < off_len; k++) { - ac->random_state = lcg_random(ac->random_state); - cfo[k] = ac->random_state; - } - - band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len); - scale = sf[idx] / sqrtf(band_energy); - ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len); - } - } else { - const float *vq = ff_aac_codebook_vector_vals[cbt_m1]; - const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1]; - VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table; - const int cb_size = ff_aac_spectral_sizes[cbt_m1]; - OPEN_READER(re, gb); - - switch (cbt_m1 >> 1) { - case 0: - for (group = 0; group < g_len; group++, cfo+=128) { - float *cf = cfo; - int len = off_len; - - do { - int code; - unsigned cb_idx; - - UPDATE_CACHE(re, gb); - GET_VLC(code, re, gb, vlc_tab, 8, 2); - - if (code >= cb_size) { - err_idx = code; - goto err_cb_overflow; - } - - cb_idx = cb_vector_idx[code]; - cf = VMUL4(cf, vq, cb_idx, sf + idx); - } while (len -= 4); - } - break; - - case 1: - for (group = 0; group < g_len; group++, cfo+=128) { - float *cf = cfo; - int len = off_len; - - do { - int code; - unsigned nnz; - unsigned cb_idx; - uint32_t bits; - - UPDATE_CACHE(re, gb); - GET_VLC(code, re, gb, vlc_tab, 8, 2); - - if (code >= cb_size) { - err_idx = code; - goto err_cb_overflow; - } - -#if MIN_CACHE_BITS < 20 - UPDATE_CACHE(re, gb); -#endif - cb_idx = cb_vector_idx[code]; - nnz = cb_idx >> 8 & 15; - bits = SHOW_UBITS(re, gb, nnz) << (32-nnz); - LAST_SKIP_BITS(re, gb, nnz); - cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx); - } while (len -= 4); - } - break; - - case 2: - for (group = 0; group < g_len; group++, cfo+=128) { - float *cf = cfo; - int len = off_len; - - do { - int code; - unsigned cb_idx; - - UPDATE_CACHE(re, gb); - GET_VLC(code, re, gb, vlc_tab, 8, 2); - - if (code >= cb_size) { - err_idx = code; - goto err_cb_overflow; - } - - cb_idx = cb_vector_idx[code]; - cf = VMUL2(cf, vq, cb_idx, sf + idx); - } while (len -= 2); - } - break; - - case 3: - case 4: - for (group = 0; group < g_len; group++, cfo+=128) { - float *cf = cfo; - int len = off_len; - - do { - int code; - unsigned nnz; - unsigned cb_idx; - unsigned sign; - - UPDATE_CACHE(re, gb); - GET_VLC(code, re, gb, vlc_tab, 8, 2); - - if (code >= cb_size) { - err_idx = code; - goto err_cb_overflow; - } - - cb_idx = cb_vector_idx[code]; - nnz = cb_idx >> 8 & 15; - sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12); - LAST_SKIP_BITS(re, gb, nnz); - cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx); - } while (len -= 2); - } - break; - - default: - for (group = 0; group < g_len; group++, cfo+=128) { - float *cf = cfo; - uint32_t *icf = (uint32_t *) cf; - int len = off_len; - - do { - int code; - unsigned nzt, nnz; - unsigned cb_idx; - uint32_t bits; - int j; - - UPDATE_CACHE(re, gb); - GET_VLC(code, re, gb, vlc_tab, 8, 2); - - if (!code) { - *icf++ = 0; - *icf++ = 0; - continue; - } - - if (code >= cb_size) { - err_idx = code; - goto err_cb_overflow; - } - - cb_idx = cb_vector_idx[code]; - nnz = cb_idx >> 12; - nzt = cb_idx >> 8; - bits = SHOW_UBITS(re, gb, nnz) << (32-nnz); - LAST_SKIP_BITS(re, gb, nnz); - - for (j = 0; j < 2; j++) { - if (nzt & 1<<j) { - uint32_t b; - int n; - /* The total length of escape_sequence must be < 22 bits according - to the specification (i.e. max is 111111110xxxxxxxxxxxx). */ - UPDATE_CACHE(re, gb); - b = GET_CACHE(re, gb); - b = 31 - av_log2(~b); - - if (b > 8) { - av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); - return -1; - } - -#if MIN_CACHE_BITS < 21 - LAST_SKIP_BITS(re, gb, b + 1); - UPDATE_CACHE(re, gb); -#else - SKIP_BITS(re, gb, b + 1); -#endif - b += 4; - n = (1 << b) + SHOW_UBITS(re, gb, b); - LAST_SKIP_BITS(re, gb, b); - *icf++ = cbrt_tab[n] | (bits & 1<<31); - bits <<= 1; - } else { - unsigned v = ((const uint32_t*)vq)[cb_idx & 15]; - *icf++ = (bits & 1<<31) | v; - bits <<= !!v; - } - cb_idx >>= 4; - } - } while (len -= 2); - - ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len); - } - } - - CLOSE_READER(re, gb); - } - } - coef += g_len << 7; - } - - if (pulse_present) { - idx = 0; - for (i = 0; i < pulse->num_pulse; i++) { - float co = coef_base[ pulse->pos[i] ]; - while (offsets[idx + 1] <= pulse->pos[i]) - idx++; - if (band_type[idx] != NOISE_BT && sf[idx]) { - float ico = -pulse->amp[i]; - if (co) { - co /= sf[idx]; - ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico); - } - coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx]; - } - } - } - return 0; - -err_cb_overflow: - av_log(ac->avctx, AV_LOG_ERROR, - "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n", - band_type[idx], err_idx, ff_aac_spectral_sizes[band_type[idx]]); - return -1; -} - -static av_always_inline float flt16_round(float pf) -{ - union float754 tmp; - tmp.f = pf; - tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U; - return tmp.f; -} - -static av_always_inline float flt16_even(float pf) -{ - union float754 tmp; - tmp.f = pf; - tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U; - return tmp.f; -} - -static av_always_inline float flt16_trunc(float pf) -{ - union float754 pun; - pun.f = pf; - pun.i &= 0xFFFF0000U; - return pun.f; -} - -static av_always_inline void predict(AACContext *ac, PredictorState *ps, float *coef, - int output_enable) -{ - const float a = 0.953125; // 61.0 / 64 - const float alpha = 0.90625; // 29.0 / 32 - float e0, e1; - float pv; - float k1, k2; - - k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0; - k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0; - - pv = flt16_round(k1 * ps->r0 + k2 * ps->r1); - if (output_enable) - *coef += pv * ac->sf_scale; - - e0 = *coef / ac->sf_scale; - e1 = e0 - k1 * ps->r0; - - ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1); - ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1)); - ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0); - ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0)); - - ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0)); - ps->r0 = flt16_trunc(a * e0); -} - -/** - * Apply AAC-Main style frequency domain prediction. - */ -static void apply_prediction(AACContext *ac, SingleChannelElement *sce) -{ - int sfb, k; - - if (!sce->ics.predictor_initialized) { - reset_all_predictors(sce->predictor_state); - sce->ics.predictor_initialized = 1; - } - - if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { - for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) { - for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) { - predict(ac, &sce->predictor_state[k], &sce->coeffs[k], - sce->ics.predictor_present && sce->ics.prediction_used[sfb]); - } - } - if (sce->ics.predictor_reset_group) - reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group); - } else - reset_all_predictors(sce->predictor_state); -} - -/** - * Decode an individual_channel_stream payload; reference: table 4.44. - * - * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. - * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.) - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_ics(AACContext *ac, SingleChannelElement *sce, - GetBitContext *gb, int common_window, int scale_flag) -{ - Pulse pulse; - TemporalNoiseShaping *tns = &sce->tns; - IndividualChannelStream *ics = &sce->ics; - float *out = sce->coeffs; - int global_gain, pulse_present = 0; - - /* This assignment is to silence a GCC warning about the variable being used - * uninitialized when in fact it always is. - */ - pulse.num_pulse = 0; - - global_gain = get_bits(gb, 8); - - if (!common_window && !scale_flag) { - if (decode_ics_info(ac, ics, gb, 0) < 0) - return -1; - } - - if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) - return -1; - if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) - return -1; - - pulse_present = 0; - if (!scale_flag) { - if ((pulse_present = get_bits1(gb))) { - if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { - av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); - return -1; - } - if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) { - av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n"); - return -1; - } - } - if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics)) - return -1; - if (get_bits1(gb)) { - av_log_missing_feature(ac->avctx, "SSR", 1); - return -1; - } - } - - if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0) - return -1; - - if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window) - apply_prediction(ac, sce); - - return 0; -} - -/** - * Mid/Side stereo decoding; reference: 4.6.8.1.3. - */ -static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe) -{ - const IndividualChannelStream *ics = &cpe->ch[0].ics; - float *ch0 = cpe->ch[0].coeffs; - float *ch1 = cpe->ch[1].coeffs; - int g, i, group, idx = 0; - const uint16_t *offsets = ics->swb_offset; - for (g = 0; g < ics->num_window_groups; g++) { - for (i = 0; i < ics->max_sfb; i++, idx++) { - if (cpe->ms_mask[idx] && - cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { - for (group = 0; group < ics->group_len[g]; group++) { - ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i], - ch1 + group * 128 + offsets[i], - offsets[i+1] - offsets[i]); - } - } - } - ch0 += ics->group_len[g] * 128; - ch1 += ics->group_len[g] * 128; - } -} - -/** - * intensity stereo decoding; reference: 4.6.8.2.3 - * - * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; - * [1] mask is decoded from bitstream; [2] mask is all 1s; - * [3] reserved for scalable AAC - */ -static void apply_intensity_stereo(ChannelElement *cpe, int ms_present) -{ - const IndividualChannelStream *ics = &cpe->ch[1].ics; - SingleChannelElement *sce1 = &cpe->ch[1]; - float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs; - const uint16_t *offsets = ics->swb_offset; - int g, group, i, k, idx = 0; - int c; - float scale; - for (g = 0; g < ics->num_window_groups; g++) { - for (i = 0; i < ics->max_sfb;) { - if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) { - const int bt_run_end = sce1->band_type_run_end[idx]; - for (; i < bt_run_end; i++, idx++) { - c = -1 + 2 * (sce1->band_type[idx] - 14); - if (ms_present) - c *= 1 - 2 * cpe->ms_mask[idx]; - scale = c * sce1->sf[idx]; - for (group = 0; group < ics->group_len[g]; group++) - for (k = offsets[i]; k < offsets[i + 1]; k++) - coef1[group * 128 + k] = scale * coef0[group * 128 + k]; - } - } else { - int bt_run_end = sce1->band_type_run_end[idx]; - idx += bt_run_end - i; - i = bt_run_end; - } - } - coef0 += ics->group_len[g] * 128; - coef1 += ics->group_len[g] * 128; - } -} - -/** - * Decode a channel_pair_element; reference: table 4.4. - * - * @param elem_id Identifies the instance of a syntax element. - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe) -{ - int i, ret, common_window, ms_present = 0; - - common_window = get_bits1(gb); - if (common_window) { - if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) - return -1; - i = cpe->ch[1].ics.use_kb_window[0]; - cpe->ch[1].ics = cpe->ch[0].ics; - cpe->ch[1].ics.use_kb_window[1] = i; - ms_present = get_bits(gb, 2); - if (ms_present == 3) { - av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); - return -1; - } else if (ms_present) - decode_mid_side_stereo(cpe, gb, ms_present); - } - if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) - return ret; - if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) - return ret; - - if (common_window) { - if (ms_present) - apply_mid_side_stereo(ac, cpe); - if (ac->m4ac.object_type == AOT_AAC_MAIN) { - apply_prediction(ac, &cpe->ch[0]); - apply_prediction(ac, &cpe->ch[1]); - } - } - - apply_intensity_stereo(cpe, ms_present); - return 0; -} - -/** - * Decode coupling_channel_element; reference: table 4.8. - * - * @param elem_id Identifies the instance of a syntax element. - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che) -{ - int num_gain = 0; - int c, g, sfb, ret; - int sign; - float scale; - SingleChannelElement *sce = &che->ch[0]; - ChannelCoupling *coup = &che->coup; - - coup->coupling_point = 2 * get_bits1(gb); - coup->num_coupled = get_bits(gb, 3); - for (c = 0; c <= coup->num_coupled; c++) { - num_gain++; - coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE; - coup->id_select[c] = get_bits(gb, 4); - if (coup->type[c] == TYPE_CPE) { - coup->ch_select[c] = get_bits(gb, 2); - if (coup->ch_select[c] == 3) - num_gain++; - } else - coup->ch_select[c] = 2; - } - coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1); - - sign = get_bits(gb, 1); - scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3)); - - if ((ret = decode_ics(ac, sce, gb, 0, 0))) - return ret; - - for (c = 0; c < num_gain; c++) { - int idx = 0; - int cge = 1; - int gain = 0; - float gain_cache = 1.; - if (c) { - cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb); - gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0; - gain_cache = pow(scale, -gain); - } - if (coup->coupling_point == AFTER_IMDCT) { - coup->gain[c][0] = gain_cache; - } else { - for (g = 0; g < sce->ics.num_window_groups; g++) { - for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) { - if (sce->band_type[idx] != ZERO_BT) { - if (!cge) { - int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; - if (t) { - int s = 1; - t = gain += t; - if (sign) { - s -= 2 * (t & 0x1); - t >>= 1; - } - gain_cache = pow(scale, -t) * s; - } - } - coup->gain[c][idx] = gain_cache; - } - } - } - } - } - return 0; -} - -/** - * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53. - * - * @return Returns number of bytes consumed. - */ -static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, - GetBitContext *gb) -{ - int i; - int num_excl_chan = 0; - - do { - for (i = 0; i < 7; i++) - che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); - } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); - - return num_excl_chan / 7; -} - -/** - * Decode dynamic range information; reference: table 4.52. - * - * @param cnt length of TYPE_FIL syntactic element in bytes - * - * @return Returns number of bytes consumed. - */ -static int decode_dynamic_range(DynamicRangeControl *che_drc, - GetBitContext *gb, int cnt) -{ - int n = 1; - int drc_num_bands = 1; - int i; - - /* pce_tag_present? */ - if (get_bits1(gb)) { - che_drc->pce_instance_tag = get_bits(gb, 4); - skip_bits(gb, 4); // tag_reserved_bits - n++; - } - - /* excluded_chns_present? */ - if (get_bits1(gb)) { - n += decode_drc_channel_exclusions(che_drc, gb); - } - - /* drc_bands_present? */ - if (get_bits1(gb)) { - che_drc->band_incr = get_bits(gb, 4); - che_drc->interpolation_scheme = get_bits(gb, 4); - n++; - drc_num_bands += che_drc->band_incr; - for (i = 0; i < drc_num_bands; i++) { - che_drc->band_top[i] = get_bits(gb, 8); - n++; - } - } - - /* prog_ref_level_present? */ - if (get_bits1(gb)) { - che_drc->prog_ref_level = get_bits(gb, 7); - skip_bits1(gb); // prog_ref_level_reserved_bits - n++; - } - - for (i = 0; i < drc_num_bands; i++) { - che_drc->dyn_rng_sgn[i] = get_bits1(gb); - che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); - n++; - } - - return n; -} - -/** - * Decode extension data (incomplete); reference: table 4.51. - * - * @param cnt length of TYPE_FIL syntactic element in bytes - * - * @return Returns number of bytes consumed - */ -static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt, - ChannelElement *che, enum RawDataBlockType elem_type) -{ - int crc_flag = 0; - int res = cnt; - switch (get_bits(gb, 4)) { // extension type - case EXT_SBR_DATA_CRC: - crc_flag++; - case EXT_SBR_DATA: - if (!che) { - av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n"); - return res; - } else if (!ac->m4ac.sbr) { - av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n"); - skip_bits_long(gb, 8 * cnt - 4); - return res; - } else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) { - av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n"); - skip_bits_long(gb, 8 * cnt - 4); - return res; - } else { - ac->m4ac.sbr = 1; - } - res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type); - break; - case EXT_DYNAMIC_RANGE: - res = decode_dynamic_range(&ac->che_drc, gb, cnt); - break; - case EXT_FILL: - case EXT_FILL_DATA: - case EXT_DATA_ELEMENT: - default: - skip_bits_long(gb, 8 * cnt - 4); - break; - }; - return res; -} - -/** - * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3. - * - * @param decode 1 if tool is used normally, 0 if tool is used in LTP. - * @param coef spectral coefficients - */ -static void apply_tns(float coef[1024], TemporalNoiseShaping *tns, - IndividualChannelStream *ics, int decode) -{ - const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); - int w, filt, m, i; - int bottom, top, order, start, end, size, inc; - float lpc[TNS_MAX_ORDER]; - - for (w = 0; w < ics->num_windows; w++) { - bottom = ics->num_swb; - for (filt = 0; filt < tns->n_filt[w]; filt++) { - top = bottom; - bottom = FFMAX(0, top - tns->length[w][filt]); - order = tns->order[w][filt]; - if (order == 0) - continue; - - // tns_decode_coef - compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0); - - start = ics->swb_offset[FFMIN(bottom, mmm)]; - end = ics->swb_offset[FFMIN( top, mmm)]; - if ((size = end - start) <= 0) - continue; - if (tns->direction[w][filt]) { - inc = -1; - start = end - 1; - } else { - inc = 1; - } - start += w * 128; - - // ar filter - for (m = 0; m < size; m++, start += inc) - for (i = 1; i <= FFMIN(m, order); i++) - coef[start] -= coef[start - i * inc] * lpc[i - 1]; - } - } -} - -/** - * Conduct IMDCT and windowing. - */ -static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce, float bias) -{ - IndividualChannelStream *ics = &sce->ics; - float *in = sce->coeffs; - float *out = sce->ret; - float *saved = sce->saved; - const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; - const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; - const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; - float *buf = ac->buf_mdct; - float *temp = ac->temp; - int i; - - // imdct - if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { - if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) - av_log(ac->avctx, AV_LOG_WARNING, - "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. " - "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n"); - for (i = 0; i < 1024; i += 128) - ff_imdct_half(&ac->mdct_small, buf + i, in + i); - } else - ff_imdct_half(&ac->mdct, buf, in); - - /* window overlapping - * NOTE: To simplify the overlapping code, all 'meaningless' short to long - * and long to short transitions are considered to be short to short - * transitions. This leaves just two cases (long to long and short to short) - * with a little special sauce for EIGHT_SHORT_SEQUENCE. - */ - if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) && - (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) { - ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, bias, 512); - } else { - for (i = 0; i < 448; i++) - out[i] = saved[i] + bias; - - if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { - ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, bias, 64); - ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, bias, 64); - ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, bias, 64); - ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, bias, 64); - ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, bias, 64); - memcpy( out + 448 + 4*128, temp, 64 * sizeof(float)); - } else { - ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, bias, 64); - for (i = 576; i < 1024; i++) - out[i] = buf[i-512] + bias; - } - } - - // buffer update - if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { - for (i = 0; i < 64; i++) - saved[i] = temp[64 + i] - bias; - ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64); - ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64); - ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64); - memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); - } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) { - memcpy( saved, buf + 512, 448 * sizeof(float)); - memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); - } else { // LONG_STOP or ONLY_LONG - memcpy( saved, buf + 512, 512 * sizeof(float)); - } -} - -/** - * Apply dependent channel coupling (applied before IMDCT). - * - * @param index index into coupling gain array - */ -static void apply_dependent_coupling(AACContext *ac, - SingleChannelElement *target, - ChannelElement *cce, int index) -{ - IndividualChannelStream *ics = &cce->ch[0].ics; - const uint16_t *offsets = ics->swb_offset; - float *dest = target->coeffs; - const float *src = cce->ch[0].coeffs; - int g, i, group, k, idx = 0; - if (ac->m4ac.object_type == AOT_AAC_LTP) { - av_log(ac->avctx, AV_LOG_ERROR, - "Dependent coupling is not supported together with LTP\n"); - return; - } - for (g = 0; g < ics->num_window_groups; g++) { - for (i = 0; i < ics->max_sfb; i++, idx++) { - if (cce->ch[0].band_type[idx] != ZERO_BT) { - const float gain = cce->coup.gain[index][idx]; - for (group = 0; group < ics->group_len[g]; group++) { - for (k = offsets[i]; k < offsets[i + 1]; k++) { - // XXX dsputil-ize - dest[group * 128 + k] += gain * src[group * 128 + k]; - } - } - } - } - dest += ics->group_len[g] * 128; - src += ics->group_len[g] * 128; - } -} - -/** - * Apply independent channel coupling (applied after IMDCT). - * - * @param index index into coupling gain array - */ -static void apply_independent_coupling(AACContext *ac, - SingleChannelElement *target, - ChannelElement *cce, int index) -{ - int i; - const float gain = cce->coup.gain[index][0]; - const float bias = ac->add_bias; - const float *src = cce->ch[0].ret; - float *dest = target->ret; - const int len = 1024 << (ac->m4ac.sbr == 1); - - for (i = 0; i < len; i++) - dest[i] += gain * (src[i] - bias); -} - -/** - * channel coupling transformation interface - * - * @param index index into coupling gain array - * @param apply_coupling_method pointer to (in)dependent coupling function - */ -static void apply_channel_coupling(AACContext *ac, ChannelElement *cc, - enum RawDataBlockType type, int elem_id, - enum CouplingPoint coupling_point, - void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)) -{ - int i, c; - - for (i = 0; i < MAX_ELEM_ID; i++) { - ChannelElement *cce = ac->che[TYPE_CCE][i]; - int index = 0; - - if (cce && cce->coup.coupling_point == coupling_point) { - ChannelCoupling *coup = &cce->coup; - - for (c = 0; c <= coup->num_coupled; c++) { - if (coup->type[c] == type && coup->id_select[c] == elem_id) { - if (coup->ch_select[c] != 1) { - apply_coupling_method(ac, &cc->ch[0], cce, index); - if (coup->ch_select[c] != 0) - index++; - } - if (coup->ch_select[c] != 2) - apply_coupling_method(ac, &cc->ch[1], cce, index++); - } else - index += 1 + (coup->ch_select[c] == 3); - } - } - } -} - -/** - * Convert spectral data to float samples, applying all supported tools as appropriate. - */ -static void spectral_to_sample(AACContext *ac) -{ - int i, type; - float imdct_bias = (ac->m4ac.sbr <= 0) ? ac->add_bias : 0.0f; - for (type = 3; type >= 0; type--) { - for (i = 0; i < MAX_ELEM_ID; i++) { - ChannelElement *che = ac->che[type][i]; - if (che) { - if (type <= TYPE_CPE) - apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling); - if (che->ch[0].tns.present) - apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); - if (che->ch[1].tns.present) - apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1); - if (type <= TYPE_CPE) - apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling); - if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) { - imdct_and_windowing(ac, &che->ch[0], imdct_bias); - if (type == TYPE_CPE) { - imdct_and_windowing(ac, &che->ch[1], imdct_bias); - } - if (ac->m4ac.sbr > 0) { - ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret); - } - } - if (type <= TYPE_CCE) - apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling); - } - } - } -} - -static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb) -{ - int size; - AACADTSHeaderInfo hdr_info; - - size = ff_aac_parse_header(gb, &hdr_info); - if (size > 0) { - if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) { - enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; - memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); - ac->m4ac.chan_config = hdr_info.chan_config; - if (set_default_channel_config(ac, new_che_pos, hdr_info.chan_config)) - return -7; - if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME)) - return -7; - } else if (ac->output_configured != OC_LOCKED) { - ac->output_configured = OC_NONE; - } - if (ac->output_configured != OC_LOCKED) - ac->m4ac.sbr = -1; - ac->m4ac.sample_rate = hdr_info.sample_rate; - ac->m4ac.sampling_index = hdr_info.sampling_index; - ac->m4ac.object_type = hdr_info.object_type; - if (!ac->avctx->sample_rate) - ac->avctx->sample_rate = hdr_info.sample_rate; - if (hdr_info.num_aac_frames == 1) { - if (!hdr_info.crc_absent) - skip_bits(gb, 16); - } else { - av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0); - return -1; - } - } - return size; -} - -static int aac_decode_frame(AVCodecContext *avctx, void *data, - int *data_size, AVPacket *avpkt) -{ - const uint8_t *buf = avpkt->data; - int buf_size = avpkt->size; - AACContext *ac = avctx->priv_data; - ChannelElement *che = NULL, *che_prev = NULL; - GetBitContext gb; - enum RawDataBlockType elem_type, elem_type_prev = TYPE_END; - int err, elem_id, data_size_tmp; - int buf_consumed; - int samples = 1024, multiplier; - int buf_offset; - - init_get_bits(&gb, buf, buf_size * 8); - - if (show_bits(&gb, 12) == 0xfff) { - if (parse_adts_frame_header(ac, &gb) < 0) { - av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n"); - return -1; - } - if (ac->m4ac.sampling_index > 12) { - av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); - return -1; - } - } - - memset(ac->tags_seen_this_frame, 0, sizeof(ac->tags_seen_this_frame)); - // parse - while ((elem_type = get_bits(&gb, 3)) != TYPE_END) { - elem_id = get_bits(&gb, 4); - - if (elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) { - av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id); - return -1; - } - - switch (elem_type) { - - case TYPE_SCE: - err = decode_ics(ac, &che->ch[0], &gb, 0, 0); - break; - - case TYPE_CPE: - err = decode_cpe(ac, &gb, che); - break; - - case TYPE_CCE: - err = decode_cce(ac, &gb, che); - break; - - case TYPE_LFE: - err = decode_ics(ac, &che->ch[0], &gb, 0, 0); - break; - - case TYPE_DSE: - err = skip_data_stream_element(ac, &gb); - break; - - case TYPE_PCE: { - enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; - memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); - if ((err = decode_pce(ac, new_che_pos, &gb))) - break; - if (ac->output_configured > OC_TRIAL_PCE) - av_log(avctx, AV_LOG_ERROR, - "Not evaluating a further program_config_element as this construct is dubious at best.\n"); - else - err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE); - break; - } - - case TYPE_FIL: - if (elem_id == 15) - elem_id += get_bits(&gb, 8) - 1; - if (get_bits_left(&gb) < 8 * elem_id) { - av_log(avctx, AV_LOG_ERROR, overread_err); - return -1; - } - while (elem_id > 0) - elem_id -= decode_extension_payload(ac, &gb, elem_id, che_prev, elem_type_prev); - err = 0; /* FIXME */ - break; - - default: - err = -1; /* should not happen, but keeps compiler happy */ - break; - } - - che_prev = che; - elem_type_prev = elem_type; - - if (err) - return err; - - if (get_bits_left(&gb) < 3) { - av_log(avctx, AV_LOG_ERROR, overread_err); - return -1; - } - } - - spectral_to_sample(ac); - - multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0; - samples <<= multiplier; - if (ac->output_configured < OC_LOCKED) { - avctx->sample_rate = ac->m4ac.sample_rate << multiplier; - avctx->frame_size = samples; - } - - data_size_tmp = samples * avctx->channels * sizeof(int16_t); - if (*data_size < data_size_tmp) { - av_log(avctx, AV_LOG_ERROR, - "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", - *data_size, data_size_tmp); - return -1; - } - *data_size = data_size_tmp; - - ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels); - - if (ac->output_configured) - ac->output_configured = OC_LOCKED; - - buf_consumed = (get_bits_count(&gb) + 7) >> 3; - for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++) - if (buf[buf_offset]) - break; - - return buf_size > buf_offset ? buf_consumed : buf_size; -} - -static av_cold int aac_decode_close(AVCodecContext *avctx) -{ - AACContext *ac = avctx->priv_data; - int i, type; - - for (i = 0; i < MAX_ELEM_ID; i++) { - for (type = 0; type < 4; type++) { - if (ac->che[type][i]) - ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr); - av_freep(&ac->che[type][i]); - } - } - - ff_mdct_end(&ac->mdct); - ff_mdct_end(&ac->mdct_small); - return 0; -} - -AVCodec aac_decoder = { - "aac", - AVMEDIA_TYPE_AUDIO, - CODEC_ID_AAC, - sizeof(AACContext), - aac_decode_init, - NULL, - aac_decode_close, - aac_decode_frame, - .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), - .sample_fmts = (const enum SampleFormat[]) { - SAMPLE_FMT_S16,SAMPLE_FMT_NONE - }, - .channel_layouts = aac_channel_layout, -};
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/aacdec.c Sat Jun 05 15:27:53 2010 +0000 @@ -0,0 +1,2125 @@ +/* + * AAC decoder + * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) + * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * AAC decoder + * @author Oded Shimon ( ods15 ods15 dyndns org ) + * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) + */ + +/* + * supported tools + * + * Support? Name + * N (code in SoC repo) gain control + * Y block switching + * Y window shapes - standard + * N window shapes - Low Delay + * Y filterbank - standard + * N (code in SoC repo) filterbank - Scalable Sample Rate + * Y Temporal Noise Shaping + * N (code in SoC repo) Long Term Prediction + * Y intensity stereo + * Y channel coupling + * Y frequency domain prediction + * Y Perceptual Noise Substitution + * Y Mid/Side stereo + * N Scalable Inverse AAC Quantization + * N Frequency Selective Switch + * N upsampling filter + * Y quantization & coding - AAC + * N quantization & coding - TwinVQ + * N quantization & coding - BSAC + * N AAC Error Resilience tools + * N Error Resilience payload syntax + * N Error Protection tool + * N CELP + * N Silence Compression + * N HVXC + * N HVXC 4kbits/s VR + * N Structured Audio tools + * N Structured Audio Sample Bank Format + * N MIDI + * N Harmonic and Individual Lines plus Noise + * N Text-To-Speech Interface + * Y Spectral Band Replication + * Y (not in this code) Layer-1 + * Y (not in this code) Layer-2 + * Y (not in this code) Layer-3 + * N SinuSoidal Coding (Transient, Sinusoid, Noise) + * N (planned) Parametric Stereo + * N Direct Stream Transfer + * + * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication. + * - HE AAC v2 comprises LC AAC with Spectral Band Replication and + Parametric Stereo. + */ + + +#include "avcodec.h" +#include "internal.h" +#include "get_bits.h" +#include "dsputil.h" +#include "fft.h" +#include "lpc.h" + +#include "aac.h" +#include "aactab.h" +#include "aacdectab.h" +#include "cbrt_tablegen.h" +#include "sbr.h" +#include "aacsbr.h" +#include "mpeg4audio.h" +#include "aac_parser.h" + +#include <assert.h> +#include <errno.h> +#include <math.h> +#include <string.h> + +#if ARCH_ARM +# include "arm/aac.h" +#endif + +union float754 { + float f; + uint32_t i; +}; + +static VLC vlc_scalefactors; +static VLC vlc_spectral[11]; + +static const char overread_err[] = "Input buffer exhausted before END element found\n"; + +static ChannelElement *get_che(AACContext *ac, int type, int elem_id) +{ + /* Some buggy encoders appear to set all elem_ids to zero and rely on + channels always occurring in the same order. This is expressly forbidden + by the spec but we will try to work around it. + */ + int err_printed = 0; + while (ac->tags_seen_this_frame[type][elem_id] && elem_id < MAX_ELEM_ID) { + if (ac->output_configured < OC_LOCKED && !err_printed) { + av_log(ac->avctx, AV_LOG_WARNING, "Duplicate channel tag found, attempting to remap.\n"); + err_printed = 1; + } + elem_id++; + } + if (elem_id == MAX_ELEM_ID) + return NULL; + ac->tags_seen_this_frame[type][elem_id] = 1; + + if (ac->tag_che_map[type][elem_id]) { + return ac->tag_che_map[type][elem_id]; + } + if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) { + return NULL; + } + switch (ac->m4ac.chan_config) { + case 7: + if (ac->tags_mapped == 3 && type == TYPE_CPE) { + ac->tags_mapped++; + return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2]; + } + case 6: + /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1] + instead of SCE[0] CPE[0] CPE[1] LFE[0]. If we seem to have + encountered such a stream, transfer the LFE[0] element to the SCE[1]'s mapping */ + if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) { + ac->tags_mapped++; + return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0]; + } + case 5: + if (ac->tags_mapped == 2 && type == TYPE_CPE) { + ac->tags_mapped++; + return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1]; + } + case 4: + if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) { + ac->tags_mapped++; + return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; + } + case 3: + case 2: + if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) { + ac->tags_mapped++; + return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0]; + } else if (ac->m4ac.chan_config == 2) { + return NULL; + } + case 1: + if (!ac->tags_mapped && type == TYPE_SCE) { + ac->tags_mapped++; + return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0]; + } + default: + return NULL; + } +} + +/** + * Check for the channel element in the current channel position configuration. + * If it exists, make sure the appropriate element is allocated and map the + * channel order to match the internal FFmpeg channel layout. + * + * @param che_pos current channel position configuration + * @param type channel element type + * @param id channel element id + * @param channels count of the number of channels in the configuration + * + * @return Returns error status. 0 - OK, !0 - error + */ +static av_cold int che_configure(AACContext *ac, + enum ChannelPosition che_pos[4][MAX_ELEM_ID], + int type, int id, + int *channels) +{ + if (che_pos[type][id]) { + if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement)))) + return AVERROR(ENOMEM); + ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr); + if (type != TYPE_CCE) { + ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret; + if (type == TYPE_CPE) { + ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret; + } + } + } else { + if (ac->che[type][id]) + ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr); + av_freep(&ac->che[type][id]); + } + return 0; +} + +/** + * Configure output channel order based on the current program configuration element. + * + * @param che_pos current channel position configuration + * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. + * + * @return Returns error status. 0 - OK, !0 - error + */ +static av_cold int output_configure(AACContext *ac, + enum ChannelPosition che_pos[4][MAX_ELEM_ID], + enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], + int channel_config, enum OCStatus oc_type) +{ + AVCodecContext *avctx = ac->avctx; + int i, type, channels = 0, ret; + + memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); + + if (channel_config) { + for (i = 0; i < tags_per_config[channel_config]; i++) { + if ((ret = che_configure(ac, che_pos, + aac_channel_layout_map[channel_config - 1][i][0], + aac_channel_layout_map[channel_config - 1][i][1], + &channels))) + return ret; + } + + memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); + ac->tags_mapped = 0; + + avctx->channel_layout = aac_channel_layout[channel_config - 1]; + } else { + /* Allocate or free elements depending on if they are in the + * current program configuration. + * + * Set up default 1:1 output mapping. + * + * For a 5.1 stream the output order will be: + * [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ] + */ + + for (i = 0; i < MAX_ELEM_ID; i++) { + for (type = 0; type < 4; type++) { + if ((ret = che_configure(ac, che_pos, type, i, &channels))) + return ret; + } + } + + memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); + ac->tags_mapped = 4 * MAX_ELEM_ID; + + avctx->channel_layout = 0; + } + + avctx->channels = channels; + + ac->output_configured = oc_type; + + return 0; +} + +/** + * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. + * + * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. + * @param sce_map mono (Single Channel Element) map + * @param type speaker type/position for these channels + */ +static void decode_channel_map(enum ChannelPosition *cpe_map, + enum ChannelPosition *sce_map, + enum ChannelPosition type, + GetBitContext *gb, int n) +{ + while (n--) { + enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map + map[get_bits(gb, 4)] = type; + } +} + +/** + * Decode program configuration element; reference: table 4.2. + * + * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], + GetBitContext *gb) +{ + int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index; + int comment_len; + + skip_bits(gb, 2); // object_type + + sampling_index = get_bits(gb, 4); + if (ac->m4ac.sampling_index != sampling_index) + av_log(ac->avctx, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n"); + + num_front = get_bits(gb, 4); + num_side = get_bits(gb, 4); + num_back = get_bits(gb, 4); + num_lfe = get_bits(gb, 2); + num_assoc_data = get_bits(gb, 3); + num_cc = get_bits(gb, 4); + + if (get_bits1(gb)) + skip_bits(gb, 4); // mono_mixdown_tag + if (get_bits1(gb)) + skip_bits(gb, 4); // stereo_mixdown_tag + + if (get_bits1(gb)) + skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround + + decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); + decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); + decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); + decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); + + skip_bits_long(gb, 4 * num_assoc_data); + + decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); + + align_get_bits(gb); + + /* comment field, first byte is length */ + comment_len = get_bits(gb, 8) * 8; + if (get_bits_left(gb) < comment_len) { + av_log(ac->avctx, AV_LOG_ERROR, overread_err); + return -1; + } + skip_bits_long(gb, comment_len); + return 0; +} + +/** + * Set up channel positions based on a default channel configuration + * as specified in table 1.17. + * + * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. + * + * @return Returns error status. 0 - OK, !0 - error + */ +static av_cold int set_default_channel_config(AACContext *ac, + enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], + int channel_config) +{ + if (channel_config < 1 || channel_config > 7) { + av_log(ac->avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", + channel_config); + return -1; + } + + /* default channel configurations: + * + * 1ch : front center (mono) + * 2ch : L + R (stereo) + * 3ch : front center + L + R + * 4ch : front center + L + R + back center + * 5ch : front center + L + R + back stereo + * 6ch : front center + L + R + back stereo + LFE + * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE + */ + + if (channel_config != 2) + new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) + if (channel_config > 1) + new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) + if (channel_config == 4) + new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center + if (channel_config > 4) + new_che_pos[TYPE_CPE][(channel_config == 7) + 1] + = AAC_CHANNEL_BACK; // back stereo + if (channel_config > 5) + new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE + if (channel_config == 7) + new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right + + return 0; +} + +/** + * Decode GA "General Audio" specific configuration; reference: table 4.1. + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int decode_ga_specific_config(AACContext *ac, GetBitContext *gb, + int channel_config) +{ + enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; + int extension_flag, ret; + + if (get_bits1(gb)) { // frameLengthFlag + av_log_missing_feature(ac->avctx, "960/120 MDCT window is", 1); + return -1; + } + + if (get_bits1(gb)) // dependsOnCoreCoder + skip_bits(gb, 14); // coreCoderDelay + extension_flag = get_bits1(gb); + + if (ac->m4ac.object_type == AOT_AAC_SCALABLE || + ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) + skip_bits(gb, 3); // layerNr + + memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); + if (channel_config == 0) { + skip_bits(gb, 4); // element_instance_tag + if ((ret = decode_pce(ac, new_che_pos, gb))) + return ret; + } else { + if ((ret = set_default_channel_config(ac, new_che_pos, channel_config))) + return ret; + } + if ((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR))) + return ret; + + if (extension_flag) { + switch (ac->m4ac.object_type) { + case AOT_ER_BSAC: + skip_bits(gb, 5); // numOfSubFrame + skip_bits(gb, 11); // layer_length + break; + case AOT_ER_AAC_LC: + case AOT_ER_AAC_LTP: + case AOT_ER_AAC_SCALABLE: + case AOT_ER_AAC_LD: + skip_bits(gb, 3); /* aacSectionDataResilienceFlag + * aacScalefactorDataResilienceFlag + * aacSpectralDataResilienceFlag + */ + break; + } + skip_bits1(gb); // extensionFlag3 (TBD in version 3) + } + return 0; +} + +/** + * Decode audio specific configuration; reference: table 1.13. + * + * @param data pointer to AVCodecContext extradata + * @param data_size size of AVCCodecContext extradata + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int decode_audio_specific_config(AACContext *ac, void *data, + int data_size) +{ + GetBitContext gb; + int i; + + init_get_bits(&gb, data, data_size * 8); + + if ((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) + return -1; + if (ac->m4ac.sampling_index > 12) { + av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); + return -1; + } + + skip_bits_long(&gb, i); + + switch (ac->m4ac.object_type) { + case AOT_AAC_MAIN: + case AOT_AAC_LC: + if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) + return -1; + break; + default: + av_log(ac->avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", + ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); + return -1; + } + return 0; +} + +/** + * linear congruential pseudorandom number generator + * + * @param previous_val pointer to the current state of the generator + * + * @return Returns a 32-bit pseudorandom integer + */ +static av_always_inline int lcg_random(int previous_val) +{ + return previous_val * 1664525 + 1013904223; +} + +static av_always_inline void reset_predict_state(PredictorState *ps) +{ + ps->r0 = 0.0f; + ps->r1 = 0.0f; + ps->cor0 = 0.0f; + ps->cor1 = 0.0f; + ps->var0 = 1.0f; + ps->var1 = 1.0f; +} + +static void reset_all_predictors(PredictorState *ps) +{ + int i; + for (i = 0; i < MAX_PREDICTORS; i++) + reset_predict_state(&ps[i]); +} + +static void reset_predictor_group(PredictorState *ps, int group_num) +{ + int i; + for (i = group_num - 1; i < MAX_PREDICTORS; i += 30) + reset_predict_state(&ps[i]); +} + +static av_cold int aac_decode_init(AVCodecContext *avctx) +{ + AACContext *ac = avctx->priv_data; + int i; + + ac->avctx = avctx; + ac->m4ac.sample_rate = avctx->sample_rate; + + if (avctx->extradata_size > 0) { + if (decode_audio_specific_config(ac, avctx->extradata, avctx->extradata_size)) + return -1; + } + + avctx->sample_fmt = SAMPLE_FMT_S16; + + AAC_INIT_VLC_STATIC( 0, 304); + AAC_INIT_VLC_STATIC( 1, 270); + AAC_INIT_VLC_STATIC( 2, 550); + AAC_INIT_VLC_STATIC( 3, 300); + AAC_INIT_VLC_STATIC( 4, 328); + AAC_INIT_VLC_STATIC( 5, 294); + AAC_INIT_VLC_STATIC( 6, 306); + AAC_INIT_VLC_STATIC( 7, 268); + AAC_INIT_VLC_STATIC( 8, 510); + AAC_INIT_VLC_STATIC( 9, 366); + AAC_INIT_VLC_STATIC(10, 462); + + ff_aac_sbr_init(); + + dsputil_init(&ac->dsp, avctx); + + ac->random_state = 0x1f2e3d4c; + + // -1024 - Compensate wrong IMDCT method. + // 32768 - Required to scale values to the correct range for the bias method + // for float to int16 conversion. + + if (ac->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) { + ac->add_bias = 385.0f; + ac->sf_scale = 1. / (-1024. * 32768.); + ac->sf_offset = 0; + } else { + ac->add_bias = 0.0f; + ac->sf_scale = 1. / -1024.; + ac->sf_offset = 60; + } + +#if !CONFIG_HARDCODED_TABLES + for (i = 0; i < 428; i++) + ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.); +#endif /* CONFIG_HARDCODED_TABLES */ + + INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code), + ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), + ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), + 352); + + ff_mdct_init(&ac->mdct, 11, 1, 1.0); + ff_mdct_init(&ac->mdct_small, 8, 1, 1.0); + // window initialization + ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); + ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); + ff_init_ff_sine_windows(10); + ff_init_ff_sine_windows( 7); + + cbrt_tableinit(); + + return 0; +} + +/** + * Skip data_stream_element; reference: table 4.10. + */ +static int skip_data_stream_element(AACContext *ac, GetBitContext *gb) +{ + int byte_align = get_bits1(gb); + int count = get_bits(gb, 8); + if (count == 255) + count += get_bits(gb, 8); + if (byte_align) + align_get_bits(gb); + + if (get_bits_left(gb) < 8 * count) { + av_log(ac->avctx, AV_LOG_ERROR, overread_err); + return -1; + } + skip_bits_long(gb, 8 * count); + return 0; +} + +static int decode_prediction(AACContext *ac, IndividualChannelStream *ics, + GetBitContext *gb) +{ + int sfb; + if (get_bits1(gb)) { + ics->predictor_reset_group = get_bits(gb, 5); + if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) { + av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n"); + return -1; + } + } + for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) { + ics->prediction_used[sfb] = get_bits1(gb); + } + return 0; +} + +/** + * Decode Individual Channel Stream info; reference: table 4.6. + * + * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. + */ +static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics, + GetBitContext *gb, int common_window) +{ + if (get_bits1(gb)) { + av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n"); + memset(ics, 0, sizeof(IndividualChannelStream)); + return -1; + } + ics->window_sequence[1] = ics->window_sequence[0]; + ics->window_sequence[0] = get_bits(gb, 2); + ics->use_kb_window[1] = ics->use_kb_window[0]; + ics->use_kb_window[0] = get_bits1(gb); + ics->num_window_groups = 1; + ics->group_len[0] = 1; + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { + int i; + ics->max_sfb = get_bits(gb, 4); + for (i = 0; i < 7; i++) { + if (get_bits1(gb)) { + ics->group_len[ics->num_window_groups - 1]++; + } else { + ics->num_window_groups++; + ics->group_len[ics->num_window_groups - 1] = 1; + } + } + ics->num_windows = 8; + ics->swb_offset = ff_swb_offset_128[ac->m4ac.sampling_index]; + ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; + ics->tns_max_bands = ff_tns_max_bands_128[ac->m4ac.sampling_index]; + ics->predictor_present = 0; + } else { + ics->max_sfb = get_bits(gb, 6); + ics->num_windows = 1; + ics->swb_offset = ff_swb_offset_1024[ac->m4ac.sampling_index]; + ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; + ics->tns_max_bands = ff_tns_max_bands_1024[ac->m4ac.sampling_index]; + ics->predictor_present = get_bits1(gb); + ics->predictor_reset_group = 0; + if (ics->predictor_present) { + if (ac->m4ac.object_type == AOT_AAC_MAIN) { + if (decode_prediction(ac, ics, gb)) { + memset(ics, 0, sizeof(IndividualChannelStream)); + return -1; + } + } else if (ac->m4ac.object_type == AOT_AAC_LC) { + av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n"); + memset(ics, 0, sizeof(IndividualChannelStream)); + return -1; + } else { + av_log_missing_feature(ac->avctx, "Predictor bit set but LTP is", 1); + memset(ics, 0, sizeof(IndividualChannelStream)); + return -1; + } + } + } + + if (ics->max_sfb > ics->num_swb) { + av_log(ac->avctx, AV_LOG_ERROR, + "Number of scalefactor bands in group (%d) exceeds limit (%d).\n", + ics->max_sfb, ics->num_swb); + memset(ics, 0, sizeof(IndividualChannelStream)); + return -1; + } + + return 0; +} + +/** + * Decode band types (section_data payload); reference: table 4.46. + * + * @param band_type array of the used band type + * @param band_type_run_end array of the last scalefactor band of a band type run + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int decode_band_types(AACContext *ac, enum BandType band_type[120], + int band_type_run_end[120], GetBitContext *gb, + IndividualChannelStream *ics) +{ + int g, idx = 0; + const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; + for (g = 0; g < ics->num_window_groups; g++) { + int k = 0; + while (k < ics->max_sfb) { + uint8_t sect_end = k; + int sect_len_incr; + int sect_band_type = get_bits(gb, 4); + if (sect_band_type == 12) { + av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n"); + return -1; + } + while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1) + sect_end += sect_len_incr; + sect_end += sect_len_incr; + if (get_bits_left(gb) < 0) { + av_log(ac->avctx, AV_LOG_ERROR, overread_err); + return -1; + } + if (sect_end > ics->max_sfb) { + av_log(ac->avctx, AV_LOG_ERROR, + "Number of bands (%d) exceeds limit (%d).\n", + sect_end, ics->max_sfb); + return -1; + } + for (; k < sect_end; k++) { + band_type [idx] = sect_band_type; + band_type_run_end[idx++] = sect_end; + } + } + } + return 0; +} + +/** + * Decode scalefactors; reference: table 4.47. + * + * @param global_gain first scalefactor value as scalefactors are differentially coded + * @param band_type array of the used band type + * @param band_type_run_end array of the last scalefactor band of a band type run + * @param sf array of scalefactors or intensity stereo positions + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb, + unsigned int global_gain, + IndividualChannelStream *ics, + enum BandType band_type[120], + int band_type_run_end[120]) +{ + const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); + int g, i, idx = 0; + int offset[3] = { global_gain, global_gain - 90, 100 }; + int noise_flag = 1; + static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; + for (g = 0; g < ics->num_window_groups; g++) { + for (i = 0; i < ics->max_sfb;) { + int run_end = band_type_run_end[idx]; + if (band_type[idx] == ZERO_BT) { + for (; i < run_end; i++, idx++) + sf[idx] = 0.; + } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { + for (; i < run_end; i++, idx++) { + offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; + if (offset[2] > 255U) { + av_log(ac->avctx, AV_LOG_ERROR, + "%s (%d) out of range.\n", sf_str[2], offset[2]); + return -1; + } + sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; + } + } else if (band_type[idx] == NOISE_BT) { + for (; i < run_end; i++, idx++) { + if (noise_flag-- > 0) + offset[1] += get_bits(gb, 9) - 256; + else + offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; + if (offset[1] > 255U) { + av_log(ac->avctx, AV_LOG_ERROR, + "%s (%d) out of range.\n", sf_str[1], offset[1]); + return -1; + } + sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100]; + } + } else { + for (; i < run_end; i++, idx++) { + offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; + if (offset[0] > 255U) { + av_log(ac->avctx, AV_LOG_ERROR, + "%s (%d) out of range.\n", sf_str[0], offset[0]); + return -1; + } + sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; + } + } + } + } + return 0; +} + +/** + * Decode pulse data; reference: table 4.7. + */ +static int decode_pulses(Pulse *pulse, GetBitContext *gb, + const uint16_t *swb_offset, int num_swb) +{ + int i, pulse_swb; + pulse->num_pulse = get_bits(gb, 2) + 1; + pulse_swb = get_bits(gb, 6); + if (pulse_swb >= num_swb) + return -1; + pulse->pos[0] = swb_offset[pulse_swb]; + pulse->pos[0] += get_bits(gb, 5); + if (pulse->pos[0] > 1023) + return -1; + pulse->amp[0] = get_bits(gb, 4); + for (i = 1; i < pulse->num_pulse; i++) { + pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1]; + if (pulse->pos[i] > 1023) + return -1; + pulse->amp[i] = get_bits(gb, 4); + } + return 0; +} + +/** + * Decode Temporal Noise Shaping data; reference: table 4.48. + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns, + GetBitContext *gb, const IndividualChannelStream *ics) +{ + int w, filt, i, coef_len, coef_res, coef_compress; + const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; + const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; + for (w = 0; w < ics->num_windows; w++) { + if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) { + coef_res = get_bits1(gb); + + for (filt = 0; filt < tns->n_filt[w]; filt++) { + int tmp2_idx; + tns->length[w][filt] = get_bits(gb, 6 - 2 * is8); + + if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) { + av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n", + tns->order[w][filt], tns_max_order); + tns->order[w][filt] = 0; + return -1; + } + if (tns->order[w][filt]) { + tns->direction[w][filt] = get_bits1(gb); + coef_compress = get_bits1(gb); + coef_len = coef_res + 3 - coef_compress; + tmp2_idx = 2 * coef_compress + coef_res; + + for (i = 0; i < tns->order[w][filt]; i++) + tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; + } + } + } + } + return 0; +} + +/** + * Decode Mid/Side data; reference: table 4.54. + * + * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; + * [1] mask is decoded from bitstream; [2] mask is all 1s; + * [3] reserved for scalable AAC + */ +static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb, + int ms_present) +{ + int idx; + if (ms_present == 1) { + for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++) + cpe->ms_mask[idx] = get_bits1(gb); + } else if (ms_present == 2) { + memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0])); + } +} + +#ifndef VMUL2 +static inline float *VMUL2(float *dst, const float *v, unsigned idx, + const float *scale) +{ + float s = *scale; + *dst++ = v[idx & 15] * s; + *dst++ = v[idx>>4 & 15] * s; + return dst; +} +#endif + +#ifndef VMUL4 +static inline float *VMUL4(float *dst, const float *v, unsigned idx, + const float *scale) +{ + float s = *scale; + *dst++ = v[idx & 3] * s; + *dst++ = v[idx>>2 & 3] * s; + *dst++ = v[idx>>4 & 3] * s; + *dst++ = v[idx>>6 & 3] * s; + return dst; +} +#endif + +#ifndef VMUL2S +static inline float *VMUL2S(float *dst, const float *v, unsigned idx, + unsigned sign, const float *scale) +{ + union float754 s0, s1; + + s0.f = s1.f = *scale; + s0.i ^= sign >> 1 << 31; + s1.i ^= sign << 31; + + *dst++ = v[idx & 15] * s0.f; + *dst++ = v[idx>>4 & 15] * s1.f; + + return dst; +} +#endif + +#ifndef VMUL4S +static inline float *VMUL4S(float *dst, const float *v, unsigned idx, + unsigned sign, const float *scale) +{ + unsigned nz = idx >> 12; + union float754 s = { .f = *scale }; + union float754 t; + + t.i = s.i ^ (sign & 1<<31); + *dst++ = v[idx & 3] * t.f; + + sign <<= nz & 1; nz >>= 1; + t.i = s.i ^ (sign & 1<<31); + *dst++ = v[idx>>2 & 3] * t.f; + + sign <<= nz & 1; nz >>= 1; + t.i = s.i ^ (sign & 1<<31); + *dst++ = v[idx>>4 & 3] * t.f; + + sign <<= nz & 1; nz >>= 1; + t.i = s.i ^ (sign & 1<<31); + *dst++ = v[idx>>6 & 3] * t.f; + + return dst; +} +#endif + +/** + * Decode spectral data; reference: table 4.50. + * Dequantize and scale spectral data; reference: 4.6.3.3. + * + * @param coef array of dequantized, scaled spectral data + * @param sf array of scalefactors or intensity stereo positions + * @param pulse_present set if pulses are present + * @param pulse pointer to pulse data struct + * @param band_type array of the used band type + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024], + GetBitContext *gb, const float sf[120], + int pulse_present, const Pulse *pulse, + const IndividualChannelStream *ics, + enum BandType band_type[120]) +{ + int i, k, g, idx = 0; + const int c = 1024 / ics->num_windows; + const uint16_t *offsets = ics->swb_offset; + float *coef_base = coef; + int err_idx; + + for (g = 0; g < ics->num_windows; g++) + memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb])); + + for (g = 0; g < ics->num_window_groups; g++) { + unsigned g_len = ics->group_len[g]; + + for (i = 0; i < ics->max_sfb; i++, idx++) { + const unsigned cbt_m1 = band_type[idx] - 1; + float *cfo = coef + offsets[i]; + int off_len = offsets[i + 1] - offsets[i]; + int group; + + if (cbt_m1 >= INTENSITY_BT2 - 1) { + for (group = 0; group < g_len; group++, cfo+=128) { + memset(cfo, 0, off_len * sizeof(float)); + } + } else if (cbt_m1 == NOISE_BT - 1) { + for (group = 0; group < g_len; group++, cfo+=128) { + float scale; + float band_energy; + + for (k = 0; k < off_len; k++) { + ac->random_state = lcg_random(ac->random_state); + cfo[k] = ac->random_state; + } + + band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len); + scale = sf[idx] / sqrtf(band_energy); + ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len); + } + } else { + const float *vq = ff_aac_codebook_vector_vals[cbt_m1]; + const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1]; + VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table; + const int cb_size = ff_aac_spectral_sizes[cbt_m1]; + OPEN_READER(re, gb); + + switch (cbt_m1 >> 1) { + case 0: + for (group = 0; group < g_len; group++, cfo+=128) { + float *cf = cfo; + int len = off_len; + + do { + int code; + unsigned cb_idx; + + UPDATE_CACHE(re, gb); + GET_VLC(code, re, gb, vlc_tab, 8, 2); + + if (code >= cb_size) { + err_idx = code; + goto err_cb_overflow; + } + + cb_idx = cb_vector_idx[code]; + cf = VMUL4(cf, vq, cb_idx, sf + idx); + } while (len -= 4); + } + break; + + case 1: + for (group = 0; group < g_len; group++, cfo+=128) { + float *cf = cfo; + int len = off_len; + + do { + int code; + unsigned nnz; + unsigned cb_idx; + uint32_t bits; + + UPDATE_CACHE(re, gb); + GET_VLC(code, re, gb, vlc_tab, 8, 2); + + if (code >= cb_size) { + err_idx = code; + goto err_cb_overflow; + } + +#if MIN_CACHE_BITS < 20 + UPDATE_CACHE(re, gb); +#endif + cb_idx = cb_vector_idx[code]; + nnz = cb_idx >> 8 & 15; + bits = SHOW_UBITS(re, gb, nnz) << (32-nnz); + LAST_SKIP_BITS(re, gb, nnz); + cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx); + } while (len -= 4); + } + break; + + case 2: + for (group = 0; group < g_len; group++, cfo+=128) { + float *cf = cfo; + int len = off_len; + + do { + int code; + unsigned cb_idx; + + UPDATE_CACHE(re, gb); + GET_VLC(code, re, gb, vlc_tab, 8, 2); + + if (code >= cb_size) { + err_idx = code; + goto err_cb_overflow; + } + + cb_idx = cb_vector_idx[code]; + cf = VMUL2(cf, vq, cb_idx, sf + idx); + } while (len -= 2); + } + break; + + case 3: + case 4: + for (group = 0; group < g_len; group++, cfo+=128) { + float *cf = cfo; + int len = off_len; + + do { + int code; + unsigned nnz; + unsigned cb_idx; + unsigned sign; + + UPDATE_CACHE(re, gb); + GET_VLC(code, re, gb, vlc_tab, 8, 2); + + if (code >= cb_size) { + err_idx = code; + goto err_cb_overflow; + } + + cb_idx = cb_vector_idx[code]; + nnz = cb_idx >> 8 & 15; + sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12); + LAST_SKIP_BITS(re, gb, nnz); + cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx); + } while (len -= 2); + } + break; + + default: + for (group = 0; group < g_len; group++, cfo+=128) { + float *cf = cfo; + uint32_t *icf = (uint32_t *) cf; + int len = off_len; + + do { + int code; + unsigned nzt, nnz; + unsigned cb_idx; + uint32_t bits; + int j; + + UPDATE_CACHE(re, gb); + GET_VLC(code, re, gb, vlc_tab, 8, 2); + + if (!code) { + *icf++ = 0; + *icf++ = 0; + continue; + } + + if (code >= cb_size) { + err_idx = code; + goto err_cb_overflow; + } + + cb_idx = cb_vector_idx[code]; + nnz = cb_idx >> 12; + nzt = cb_idx >> 8; + bits = SHOW_UBITS(re, gb, nnz) << (32-nnz); + LAST_SKIP_BITS(re, gb, nnz); + + for (j = 0; j < 2; j++) { + if (nzt & 1<<j) { + uint32_t b; + int n; + /* The total length of escape_sequence must be < 22 bits according + to the specification (i.e. max is 111111110xxxxxxxxxxxx). */ + UPDATE_CACHE(re, gb); + b = GET_CACHE(re, gb); + b = 31 - av_log2(~b); + + if (b > 8) { + av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); + return -1; + } + +#if MIN_CACHE_BITS < 21 + LAST_SKIP_BITS(re, gb, b + 1); + UPDATE_CACHE(re, gb); +#else + SKIP_BITS(re, gb, b + 1); +#endif + b += 4; + n = (1 << b) + SHOW_UBITS(re, gb, b); + LAST_SKIP_BITS(re, gb, b); + *icf++ = cbrt_tab[n] | (bits & 1<<31); + bits <<= 1; + } else { + unsigned v = ((const uint32_t*)vq)[cb_idx & 15]; + *icf++ = (bits & 1<<31) | v; + bits <<= !!v; + } + cb_idx >>= 4; + } + } while (len -= 2); + + ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len); + } + } + + CLOSE_READER(re, gb); + } + } + coef += g_len << 7; + } + + if (pulse_present) { + idx = 0; + for (i = 0; i < pulse->num_pulse; i++) { + float co = coef_base[ pulse->pos[i] ]; + while (offsets[idx + 1] <= pulse->pos[i]) + idx++; + if (band_type[idx] != NOISE_BT && sf[idx]) { + float ico = -pulse->amp[i]; + if (co) { + co /= sf[idx]; + ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico); + } + coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx]; + } + } + } + return 0; + +err_cb_overflow: + av_log(ac->avctx, AV_LOG_ERROR, + "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n", + band_type[idx], err_idx, ff_aac_spectral_sizes[band_type[idx]]); + return -1; +} + +static av_always_inline float flt16_round(float pf) +{ + union float754 tmp; + tmp.f = pf; + tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U; + return tmp.f; +} + +static av_always_inline float flt16_even(float pf) +{ + union float754 tmp; + tmp.f = pf; + tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U; + return tmp.f; +} + +static av_always_inline float flt16_trunc(float pf) +{ + union float754 pun; + pun.f = pf; + pun.i &= 0xFFFF0000U; + return pun.f; +} + +static av_always_inline void predict(AACContext *ac, PredictorState *ps, float *coef, + int output_enable) +{ + const float a = 0.953125; // 61.0 / 64 + const float alpha = 0.90625; // 29.0 / 32 + float e0, e1; + float pv; + float k1, k2; + + k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0; + k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0; + + pv = flt16_round(k1 * ps->r0 + k2 * ps->r1); + if (output_enable) + *coef += pv * ac->sf_scale; + + e0 = *coef / ac->sf_scale; + e1 = e0 - k1 * ps->r0; + + ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1); + ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1)); + ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0); + ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0)); + + ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0)); + ps->r0 = flt16_trunc(a * e0); +} + +/** + * Apply AAC-Main style frequency domain prediction. + */ +static void apply_prediction(AACContext *ac, SingleChannelElement *sce) +{ + int sfb, k; + + if (!sce->ics.predictor_initialized) { + reset_all_predictors(sce->predictor_state); + sce->ics.predictor_initialized = 1; + } + + if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { + for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) { + for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) { + predict(ac, &sce->predictor_state[k], &sce->coeffs[k], + sce->ics.predictor_present && sce->ics.prediction_used[sfb]); + } + } + if (sce->ics.predictor_reset_group) + reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group); + } else + reset_all_predictors(sce->predictor_state); +} + +/** + * Decode an individual_channel_stream payload; reference: table 4.44. + * + * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. + * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.) + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int decode_ics(AACContext *ac, SingleChannelElement *sce, + GetBitContext *gb, int common_window, int scale_flag) +{ + Pulse pulse; + TemporalNoiseShaping *tns = &sce->tns; + IndividualChannelStream *ics = &sce->ics; + float *out = sce->coeffs; + int global_gain, pulse_present = 0; + + /* This assignment is to silence a GCC warning about the variable being used + * uninitialized when in fact it always is. + */ + pulse.num_pulse = 0; + + global_gain = get_bits(gb, 8); + + if (!common_window && !scale_flag) { + if (decode_ics_info(ac, ics, gb, 0) < 0) + return -1; + } + + if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) + return -1; + if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) + return -1; + + pulse_present = 0; + if (!scale_flag) { + if ((pulse_present = get_bits1(gb))) { + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { + av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); + return -1; + } + if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) { + av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n"); + return -1; + } + } + if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics)) + return -1; + if (get_bits1(gb)) { + av_log_missing_feature(ac->avctx, "SSR", 1); + return -1; + } + } + + if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0) + return -1; + + if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window) + apply_prediction(ac, sce); + + return 0; +} + +/** + * Mid/Side stereo decoding; reference: 4.6.8.1.3. + */ +static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe) +{ + const IndividualChannelStream *ics = &cpe->ch[0].ics; + float *ch0 = cpe->ch[0].coeffs; + float *ch1 = cpe->ch[1].coeffs; + int g, i, group, idx = 0; + const uint16_t *offsets = ics->swb_offset; + for (g = 0; g < ics->num_window_groups; g++) { + for (i = 0; i < ics->max_sfb; i++, idx++) { + if (cpe->ms_mask[idx] && + cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { + for (group = 0; group < ics->group_len[g]; group++) { + ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i], + ch1 + group * 128 + offsets[i], + offsets[i+1] - offsets[i]); + } + } + } + ch0 += ics->group_len[g] * 128; + ch1 += ics->group_len[g] * 128; + } +} + +/** + * intensity stereo decoding; reference: 4.6.8.2.3 + * + * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; + * [1] mask is decoded from bitstream; [2] mask is all 1s; + * [3] reserved for scalable AAC + */ +static void apply_intensity_stereo(ChannelElement *cpe, int ms_present) +{ + const IndividualChannelStream *ics = &cpe->ch[1].ics; + SingleChannelElement *sce1 = &cpe->ch[1]; + float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs; + const uint16_t *offsets = ics->swb_offset; + int g, group, i, k, idx = 0; + int c; + float scale; + for (g = 0; g < ics->num_window_groups; g++) { + for (i = 0; i < ics->max_sfb;) { + if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) { + const int bt_run_end = sce1->band_type_run_end[idx]; + for (; i < bt_run_end; i++, idx++) { + c = -1 + 2 * (sce1->band_type[idx] - 14); + if (ms_present) + c *= 1 - 2 * cpe->ms_mask[idx]; + scale = c * sce1->sf[idx]; + for (group = 0; group < ics->group_len[g]; group++) + for (k = offsets[i]; k < offsets[i + 1]; k++) + coef1[group * 128 + k] = scale * coef0[group * 128 + k]; + } + } else { + int bt_run_end = sce1->band_type_run_end[idx]; + idx += bt_run_end - i; + i = bt_run_end; + } + } + coef0 += ics->group_len[g] * 128; + coef1 += ics->group_len[g] * 128; + } +} + +/** + * Decode a channel_pair_element; reference: table 4.4. + * + * @param elem_id Identifies the instance of a syntax element. + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe) +{ + int i, ret, common_window, ms_present = 0; + + common_window = get_bits1(gb); + if (common_window) { + if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) + return -1; + i = cpe->ch[1].ics.use_kb_window[0]; + cpe->ch[1].ics = cpe->ch[0].ics; + cpe->ch[1].ics.use_kb_window[1] = i; + ms_present = get_bits(gb, 2); + if (ms_present == 3) { + av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); + return -1; + } else if (ms_present) + decode_mid_side_stereo(cpe, gb, ms_present); + } + if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) + return ret; + if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) + return ret; + + if (common_window) { + if (ms_present) + apply_mid_side_stereo(ac, cpe); + if (ac->m4ac.object_type == AOT_AAC_MAIN) { + apply_prediction(ac, &cpe->ch[0]); + apply_prediction(ac, &cpe->ch[1]); + } + } + + apply_intensity_stereo(cpe, ms_present); + return 0; +} + +/** + * Decode coupling_channel_element; reference: table 4.8. + * + * @param elem_id Identifies the instance of a syntax element. + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che) +{ + int num_gain = 0; + int c, g, sfb, ret; + int sign; + float scale; + SingleChannelElement *sce = &che->ch[0]; + ChannelCoupling *coup = &che->coup; + + coup->coupling_point = 2 * get_bits1(gb); + coup->num_coupled = get_bits(gb, 3); + for (c = 0; c <= coup->num_coupled; c++) { + num_gain++; + coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE; + coup->id_select[c] = get_bits(gb, 4); + if (coup->type[c] == TYPE_CPE) { + coup->ch_select[c] = get_bits(gb, 2); + if (coup->ch_select[c] == 3) + num_gain++; + } else + coup->ch_select[c] = 2; + } + coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1); + + sign = get_bits(gb, 1); + scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3)); + + if ((ret = decode_ics(ac, sce, gb, 0, 0))) + return ret; + + for (c = 0; c < num_gain; c++) { + int idx = 0; + int cge = 1; + int gain = 0; + float gain_cache = 1.; + if (c) { + cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb); + gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0; + gain_cache = pow(scale, -gain); + } + if (coup->coupling_point == AFTER_IMDCT) { + coup->gain[c][0] = gain_cache; + } else { + for (g = 0; g < sce->ics.num_window_groups; g++) { + for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) { + if (sce->band_type[idx] != ZERO_BT) { + if (!cge) { + int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; + if (t) { + int s = 1; + t = gain += t; + if (sign) { + s -= 2 * (t & 0x1); + t >>= 1; + } + gain_cache = pow(scale, -t) * s; + } + } + coup->gain[c][idx] = gain_cache; + } + } + } + } + } + return 0; +} + +/** + * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53. + * + * @return Returns number of bytes consumed. + */ +static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, + GetBitContext *gb) +{ + int i; + int num_excl_chan = 0; + + do { + for (i = 0; i < 7; i++) + che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); + } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); + + return num_excl_chan / 7; +} + +/** + * Decode dynamic range information; reference: table 4.52. + * + * @param cnt length of TYPE_FIL syntactic element in bytes + * + * @return Returns number of bytes consumed. + */ +static int decode_dynamic_range(DynamicRangeControl *che_drc, + GetBitContext *gb, int cnt) +{ + int n = 1; + int drc_num_bands = 1; + int i; + + /* pce_tag_present? */ + if (get_bits1(gb)) { + che_drc->pce_instance_tag = get_bits(gb, 4); + skip_bits(gb, 4); // tag_reserved_bits + n++; + } + + /* excluded_chns_present? */ + if (get_bits1(gb)) { + n += decode_drc_channel_exclusions(che_drc, gb); + } + + /* drc_bands_present? */ + if (get_bits1(gb)) { + che_drc->band_incr = get_bits(gb, 4); + che_drc->interpolation_scheme = get_bits(gb, 4); + n++; + drc_num_bands += che_drc->band_incr; + for (i = 0; i < drc_num_bands; i++) { + che_drc->band_top[i] = get_bits(gb, 8); + n++; + } + } + + /* prog_ref_level_present? */ + if (get_bits1(gb)) { + che_drc->prog_ref_level = get_bits(gb, 7); + skip_bits1(gb); // prog_ref_level_reserved_bits + n++; + } + + for (i = 0; i < drc_num_bands; i++) { + che_drc->dyn_rng_sgn[i] = get_bits1(gb); + che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); + n++; + } + + return n; +} + +/** + * Decode extension data (incomplete); reference: table 4.51. + * + * @param cnt length of TYPE_FIL syntactic element in bytes + * + * @return Returns number of bytes consumed + */ +static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt, + ChannelElement *che, enum RawDataBlockType elem_type) +{ + int crc_flag = 0; + int res = cnt; + switch (get_bits(gb, 4)) { // extension type + case EXT_SBR_DATA_CRC: + crc_flag++; + case EXT_SBR_DATA: + if (!che) { + av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n"); + return res; + } else if (!ac->m4ac.sbr) { + av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n"); + skip_bits_long(gb, 8 * cnt - 4); + return res; + } else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) { + av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n"); + skip_bits_long(gb, 8 * cnt - 4); + return res; + } else { + ac->m4ac.sbr = 1; + } + res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type); + break; + case EXT_DYNAMIC_RANGE: + res = decode_dynamic_range(&ac->che_drc, gb, cnt); + break; + case EXT_FILL: + case EXT_FILL_DATA: + case EXT_DATA_ELEMENT: + default: + skip_bits_long(gb, 8 * cnt - 4); + break; + }; + return res; +} + +/** + * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3. + * + * @param decode 1 if tool is used normally, 0 if tool is used in LTP. + * @param coef spectral coefficients + */ +static void apply_tns(float coef[1024], TemporalNoiseShaping *tns, + IndividualChannelStream *ics, int decode) +{ + const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); + int w, filt, m, i; + int bottom, top, order, start, end, size, inc; + float lpc[TNS_MAX_ORDER]; + + for (w = 0; w < ics->num_windows; w++) { + bottom = ics->num_swb; + for (filt = 0; filt < tns->n_filt[w]; filt++) { + top = bottom; + bottom = FFMAX(0, top - tns->length[w][filt]); + order = tns->order[w][filt]; + if (order == 0) + continue; + + // tns_decode_coef + compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0); + + start = ics->swb_offset[FFMIN(bottom, mmm)]; + end = ics->swb_offset[FFMIN( top, mmm)]; + if ((size = end - start) <= 0) + continue; + if (tns->direction[w][filt]) { + inc = -1; + start = end - 1; + } else { + inc = 1; + } + start += w * 128; + + // ar filter + for (m = 0; m < size; m++, start += inc) + for (i = 1; i <= FFMIN(m, order); i++) + coef[start] -= coef[start - i * inc] * lpc[i - 1]; + } + } +} + +/** + * Conduct IMDCT and windowing. + */ +static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce, float bias) +{ + IndividualChannelStream *ics = &sce->ics; + float *in = sce->coeffs; + float *out = sce->ret; + float *saved = sce->saved; + const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; + const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; + const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; + float *buf = ac->buf_mdct; + float *temp = ac->temp; + int i; + + // imdct + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { + if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) + av_log(ac->avctx, AV_LOG_WARNING, + "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. " + "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n"); + for (i = 0; i < 1024; i += 128) + ff_imdct_half(&ac->mdct_small, buf + i, in + i); + } else + ff_imdct_half(&ac->mdct, buf, in); + + /* window overlapping + * NOTE: To simplify the overlapping code, all 'meaningless' short to long + * and long to short transitions are considered to be short to short + * transitions. This leaves just two cases (long to long and short to short) + * with a little special sauce for EIGHT_SHORT_SEQUENCE. + */ + if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) && + (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) { + ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, bias, 512); + } else { + for (i = 0; i < 448; i++) + out[i] = saved[i] + bias; + + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { + ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, bias, 64); + ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, bias, 64); + ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, bias, 64); + ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, bias, 64); + ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, bias, 64); + memcpy( out + 448 + 4*128, temp, 64 * sizeof(float)); + } else { + ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, bias, 64); + for (i = 576; i < 1024; i++) + out[i] = buf[i-512] + bias; + } + } + + // buffer update + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { + for (i = 0; i < 64; i++) + saved[i] = temp[64 + i] - bias; + ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64); + ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64); + ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64); + memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); + } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) { + memcpy( saved, buf + 512, 448 * sizeof(float)); + memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); + } else { // LONG_STOP or ONLY_LONG + memcpy( saved, buf + 512, 512 * sizeof(float)); + } +} + +/** + * Apply dependent channel coupling (applied before IMDCT). + * + * @param index index into coupling gain array + */ +static void apply_dependent_coupling(AACContext *ac, + SingleChannelElement *target, + ChannelElement *cce, int index) +{ + IndividualChannelStream *ics = &cce->ch[0].ics; + const uint16_t *offsets = ics->swb_offset; + float *dest = target->coeffs; + const float *src = cce->ch[0].coeffs; + int g, i, group, k, idx = 0; + if (ac->m4ac.object_type == AOT_AAC_LTP) { + av_log(ac->avctx, AV_LOG_ERROR, + "Dependent coupling is not supported together with LTP\n"); + return; + } + for (g = 0; g < ics->num_window_groups; g++) { + for (i = 0; i < ics->max_sfb; i++, idx++) { + if (cce->ch[0].band_type[idx] != ZERO_BT) { + const float gain = cce->coup.gain[index][idx]; + for (group = 0; group < ics->group_len[g]; group++) { + for (k = offsets[i]; k < offsets[i + 1]; k++) { + // XXX dsputil-ize + dest[group * 128 + k] += gain * src[group * 128 + k]; + } + } + } + } + dest += ics->group_len[g] * 128; + src += ics->group_len[g] * 128; + } +} + +/** + * Apply independent channel coupling (applied after IMDCT). + * + * @param index index into coupling gain array + */ +static void apply_independent_coupling(AACContext *ac, + SingleChannelElement *target, + ChannelElement *cce, int index) +{ + int i; + const float gain = cce->coup.gain[index][0]; + const float bias = ac->add_bias; + const float *src = cce->ch[0].ret; + float *dest = target->ret; + const int len = 1024 << (ac->m4ac.sbr == 1); + + for (i = 0; i < len; i++) + dest[i] += gain * (src[i] - bias); +} + +/** + * channel coupling transformation interface + * + * @param index index into coupling gain array + * @param apply_coupling_method pointer to (in)dependent coupling function + */ +static void apply_channel_coupling(AACContext *ac, ChannelElement *cc, + enum RawDataBlockType type, int elem_id, + enum CouplingPoint coupling_point, + void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)) +{ + int i, c; + + for (i = 0; i < MAX_ELEM_ID; i++) { + ChannelElement *cce = ac->che[TYPE_CCE][i]; + int index = 0; + + if (cce && cce->coup.coupling_point == coupling_point) { + ChannelCoupling *coup = &cce->coup; + + for (c = 0; c <= coup->num_coupled; c++) { + if (coup->type[c] == type && coup->id_select[c] == elem_id) { + if (coup->ch_select[c] != 1) { + apply_coupling_method(ac, &cc->ch[0], cce, index); + if (coup->ch_select[c] != 0) + index++; + } + if (coup->ch_select[c] != 2) + apply_coupling_method(ac, &cc->ch[1], cce, index++); + } else + index += 1 + (coup->ch_select[c] == 3); + } + } + } +} + +/** + * Convert spectral data to float samples, applying all supported tools as appropriate. + */ +static void spectral_to_sample(AACContext *ac) +{ + int i, type; + float imdct_bias = (ac->m4ac.sbr <= 0) ? ac->add_bias : 0.0f; + for (type = 3; type >= 0; type--) { + for (i = 0; i < MAX_ELEM_ID; i++) { + ChannelElement *che = ac->che[type][i]; + if (che) { + if (type <= TYPE_CPE) + apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling); + if (che->ch[0].tns.present) + apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); + if (che->ch[1].tns.present) + apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1); + if (type <= TYPE_CPE) + apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling); + if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) { + imdct_and_windowing(ac, &che->ch[0], imdct_bias); + if (type == TYPE_CPE) { + imdct_and_windowing(ac, &che->ch[1], imdct_bias); + } + if (ac->m4ac.sbr > 0) { + ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret); + } + } + if (type <= TYPE_CCE) + apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling); + } + } + } +} + +static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb) +{ + int size; + AACADTSHeaderInfo hdr_info; + + size = ff_aac_parse_header(gb, &hdr_info); + if (size > 0) { + if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) { + enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; + memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); + ac->m4ac.chan_config = hdr_info.chan_config; + if (set_default_channel_config(ac, new_che_pos, hdr_info.chan_config)) + return -7; + if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME)) + return -7; + } else if (ac->output_configured != OC_LOCKED) { + ac->output_configured = OC_NONE; + } + if (ac->output_configured != OC_LOCKED) + ac->m4ac.sbr = -1; + ac->m4ac.sample_rate = hdr_info.sample_rate; + ac->m4ac.sampling_index = hdr_info.sampling_index; + ac->m4ac.object_type = hdr_info.object_type; + if (!ac->avctx->sample_rate) + ac->avctx->sample_rate = hdr_info.sample_rate; + if (hdr_info.num_aac_frames == 1) { + if (!hdr_info.crc_absent) + skip_bits(gb, 16); + } else { + av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0); + return -1; + } + } + return size; +} + +static int aac_decode_frame(AVCodecContext *avctx, void *data, + int *data_size, AVPacket *avpkt) +{ + const uint8_t *buf = avpkt->data; + int buf_size = avpkt->size; + AACContext *ac = avctx->priv_data; + ChannelElement *che = NULL, *che_prev = NULL; + GetBitContext gb; + enum RawDataBlockType elem_type, elem_type_prev = TYPE_END; + int err, elem_id, data_size_tmp; + int buf_consumed; + int samples = 1024, multiplier; + int buf_offset; + + init_get_bits(&gb, buf, buf_size * 8); + + if (show_bits(&gb, 12) == 0xfff) { + if (parse_adts_frame_header(ac, &gb) < 0) { + av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n"); + return -1; + } + if (ac->m4ac.sampling_index > 12) { + av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); + return -1; + } + } + + memset(ac->tags_seen_this_frame, 0, sizeof(ac->tags_seen_this_frame)); + // parse + while ((elem_type = get_bits(&gb, 3)) != TYPE_END) { + elem_id = get_bits(&gb, 4); + + if (elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) { + av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id); + return -1; + } + + switch (elem_type) { + + case TYPE_SCE: + err = decode_ics(ac, &che->ch[0], &gb, 0, 0); + break; + + case TYPE_CPE: + err = decode_cpe(ac, &gb, che); + break; + + case TYPE_CCE: + err = decode_cce(ac, &gb, che); + break; + + case TYPE_LFE: + err = decode_ics(ac, &che->ch[0], &gb, 0, 0); + break; + + case TYPE_DSE: + err = skip_data_stream_element(ac, &gb); + break; + + case TYPE_PCE: { + enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; + memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); + if ((err = decode_pce(ac, new_che_pos, &gb))) + break; + if (ac->output_configured > OC_TRIAL_PCE) + av_log(avctx, AV_LOG_ERROR, + "Not evaluating a further program_config_element as this construct is dubious at best.\n"); + else + err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE); + break; + } + + case TYPE_FIL: + if (elem_id == 15) + elem_id += get_bits(&gb, 8) - 1; + if (get_bits_left(&gb) < 8 * elem_id) { + av_log(avctx, AV_LOG_ERROR, overread_err); + return -1; + } + while (elem_id > 0) + elem_id -= decode_extension_payload(ac, &gb, elem_id, che_prev, elem_type_prev); + err = 0; /* FIXME */ + break; + + default: + err = -1; /* should not happen, but keeps compiler happy */ + break; + } + + che_prev = che; + elem_type_prev = elem_type; + + if (err) + return err; + + if (get_bits_left(&gb) < 3) { + av_log(avctx, AV_LOG_ERROR, overread_err); + return -1; + } + } + + spectral_to_sample(ac); + + multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0; + samples <<= multiplier; + if (ac->output_configured < OC_LOCKED) { + avctx->sample_rate = ac->m4ac.sample_rate << multiplier; + avctx->frame_size = samples; + } + + data_size_tmp = samples * avctx->channels * sizeof(int16_t); + if (*data_size < data_size_tmp) { + av_log(avctx, AV_LOG_ERROR, + "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", + *data_size, data_size_tmp); + return -1; + } + *data_size = data_size_tmp; + + ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels); + + if (ac->output_configured) + ac->output_configured = OC_LOCKED; + + buf_consumed = (get_bits_count(&gb) + 7) >> 3; + for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++) + if (buf[buf_offset]) + break; + + return buf_size > buf_offset ? buf_consumed : buf_size; +} + +static av_cold int aac_decode_close(AVCodecContext *avctx) +{ + AACContext *ac = avctx->priv_data; + int i, type; + + for (i = 0; i < MAX_ELEM_ID; i++) { + for (type = 0; type < 4; type++) { + if (ac->che[type][i]) + ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr); + av_freep(&ac->che[type][i]); + } + } + + ff_mdct_end(&ac->mdct); + ff_mdct_end(&ac->mdct_small); + return 0; +} + +AVCodec aac_decoder = { + "aac", + AVMEDIA_TYPE_AUDIO, + CODEC_ID_AAC, + sizeof(AACContext), + aac_decode_init, + NULL, + aac_decode_close, + aac_decode_frame, + .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), + .sample_fmts = (const enum SampleFormat[]) { + SAMPLE_FMT_S16,SAMPLE_FMT_NONE + }, + .channel_layouts = aac_channel_layout, +};