Mercurial > libavcodec.hg
view aac.c @ 9930:32e856bd5ded libavcodec
Check for CONFIG_LIBFOO_DECODER/CONFIG_LIBFOO_ENCODER instead of just
CONFIG_LIBFOO in the external libraries section.
This is more consistent with the rest of the Makefiles, it makes clearer what
is actually implemented and should be advantageous if we implement an external
library encoder where we previously just had the decoder and vice versa.
| author | diego |
|---|---|
| date | Tue, 07 Jul 2009 09:33:08 +0000 |
| parents | 7ad7d4094d1f |
| children | 98fd723f72e7 |
line wrap: on
line source
/* * 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 libavcodec/aac.c * 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 * N (in progress) 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 "lpc.h" #include "aac.h" #include "aactab.h" #include "aacdectab.h" #include "mpeg4audio.h" #include "aac_parser.h" #include <assert.h> #include <errno.h> #include <math.h> #include <string.h> union float754 { float f; uint32_t i; }; static VLC vlc_scalefactors; static VLC vlc_spectral[11]; static ChannelElement* get_che(AACContext *ac, int type, int elem_id) { static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0 }; if (ac->tag_che_map[type][elem_id]) { return ac->tag_che_map[type][elem_id]; } if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) { return NULL; } switch (ac->m4ac.chan_config) { case 7: if (ac->tags_mapped == 3 && type == TYPE_CPE) { ac->tags_mapped++; return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2]; } case 6: /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1] instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have encountered such a stream, transfer the LFE[0] element to SCE[1] */ if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) { ac->tags_mapped++; return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0]; } case 5: if (ac->tags_mapped == 2 && type == TYPE_CPE) { ac->tags_mapped++; return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1]; } case 4: if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) { ac->tags_mapped++; return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; } case 3: case 2: if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) { ac->tags_mapped++; return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0]; } else if (ac->m4ac.chan_config == 2) { return NULL; } case 1: if (!ac->tags_mapped && type == TYPE_SCE) { ac->tags_mapped++; return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0]; } default: return NULL; } } /** * 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 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) { AVCodecContext *avctx = ac->avccontext; int i, type, channels = 0; memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); /* 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(che_pos[type][i]) { if(!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement)))) return AVERROR(ENOMEM); if(type != TYPE_CCE) { ac->output_data[channels++] = ac->che[type][i]->ch[0].ret; if(type == TYPE_CPE) { ac->output_data[channels++] = ac->che[type][i]->ch[1].ret; } } } else av_freep(&ac->che[type][i]); } } if (channel_config) { memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); ac->tags_mapped = 0; } else { memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); ac->tags_mapped = 4*MAX_ELEM_ID; } avctx->channels = channels; 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; skip_bits(gb, 2); // object_type sampling_index = get_bits(gb, 4); if (ac->m4ac.sampling_index != sampling_index) av_log(ac->avccontext, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n"); num_front = get_bits(gb, 4); num_side = get_bits(gb, 4); num_back = get_bits(gb, 4); num_lfe = get_bits(gb, 2); num_assoc_data = get_bits(gb, 3); num_cc = get_bits(gb, 4); if (get_bits1(gb)) skip_bits(gb, 4); // mono_mixdown_tag if (get_bits1(gb)) skip_bits(gb, 4); // stereo_mixdown_tag if (get_bits1(gb)) skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); skip_bits_long(gb, 4 * num_assoc_data); decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); align_get_bits(gb); /* comment field, first byte is length */ skip_bits_long(gb, 8 * get_bits(gb, 8)); 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 int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], int channel_config) { if(channel_config < 1 || channel_config > 7) { av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", channel_config); return -1; } /* default channel configurations: * * 1ch : front center (mono) * 2ch : L + R (stereo) * 3ch : front center + L + R * 4ch : front center + L + R + back center * 5ch : front center + L + R + back stereo * 6ch : front center + L + R + back stereo + LFE * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE */ if(channel_config != 2) new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) if(channel_config > 1) new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) if(channel_config == 4) new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center if(channel_config > 4) new_che_pos[TYPE_CPE][(channel_config == 7) + 1] = AAC_CHANNEL_BACK; // back stereo if(channel_config > 5) new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE if(channel_config == 7) new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right return 0; } /** * Decode GA "General Audio" specific configuration; reference: table 4.1. * * @return Returns error status. 0 - OK, !0 - error */ static int decode_ga_specific_config(AACContext * ac, GetBitContext * gb, int channel_config) { enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; int extension_flag, ret; if(get_bits1(gb)) { // frameLengthFlag av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1); return -1; } if (get_bits1(gb)) // dependsOnCoreCoder skip_bits(gb, 14); // coreCoderDelay extension_flag = get_bits1(gb); if(ac->m4ac.object_type == AOT_AAC_SCALABLE || ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) skip_bits(gb, 3); // layerNr memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); if (channel_config == 0) { skip_bits(gb, 4); // element_instance_tag if((ret = decode_pce(ac, new_che_pos, gb))) return ret; } else { if((ret = set_default_channel_config(ac, new_che_pos, channel_config))) return ret; } if((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config))) return ret; if (extension_flag) { switch (ac->m4ac.object_type) { case AOT_ER_BSAC: skip_bits(gb, 5); // numOfSubFrame skip_bits(gb, 11); // layer_length break; case AOT_ER_AAC_LC: case AOT_ER_AAC_LTP: case AOT_ER_AAC_SCALABLE: case AOT_ER_AAC_LD: skip_bits(gb, 3); /* aacSectionDataResilienceFlag * aacScalefactorDataResilienceFlag * aacSpectralDataResilienceFlag */ break; } skip_bits1(gb); // extensionFlag3 (TBD in version 3) } return 0; } /** * Decode audio specific configuration; reference: table 1.13. * * @param data pointer to AVCodecContext extradata * @param data_size size of AVCCodecContext extradata * * @return Returns error status. 0 - OK, !0 - error */ static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) { GetBitContext gb; int i; init_get_bits(&gb, data, data_size * 8); if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) return -1; if(ac->m4ac.sampling_index > 12) { av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); return -1; } skip_bits_long(&gb, i); switch (ac->m4ac.object_type) { case AOT_AAC_MAIN: case AOT_AAC_LC: if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) return -1; break; default: av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); return -1; } return 0; } /** * linear congruential pseudorandom number generator * * @param previous_val pointer to the current state of the generator * * @return Returns a 32-bit pseudorandom integer */ static av_always_inline int lcg_random(int previous_val) { return previous_val * 1664525 + 1013904223; } static void reset_predict_state(PredictorState * ps) { ps->r0 = 0.0f; ps->r1 = 0.0f; ps->cor0 = 0.0f; ps->cor1 = 0.0f; ps->var0 = 1.0f; ps->var1 = 1.0f; } static void reset_all_predictors(PredictorState * ps) { int i; for (i = 0; i < MAX_PREDICTORS; i++) reset_predict_state(&ps[i]); } static void reset_predictor_group(PredictorState * ps, int group_num) { int i; for (i = group_num-1; i < MAX_PREDICTORS; i+=30) reset_predict_state(&ps[i]); } static av_cold int aac_decode_init(AVCodecContext * avccontext) { AACContext * ac = avccontext->priv_data; int i; ac->avccontext = avccontext; if (avccontext->extradata_size > 0) { if(decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size)) return -1; avccontext->sample_rate = ac->m4ac.sample_rate; } else if (avccontext->channels > 0) { 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(set_default_channel_config(ac, new_che_pos, avccontext->channels - (avccontext->channels == 8))) return -1; if(output_configure(ac, ac->che_pos, new_che_pos, 1)) return -1; ac->m4ac.sample_rate = avccontext->sample_rate; } avccontext->sample_fmt = SAMPLE_FMT_S16; avccontext->frame_size = 1024; AAC_INIT_VLC_STATIC( 0, 144); AAC_INIT_VLC_STATIC( 1, 114); AAC_INIT_VLC_STATIC( 2, 188); AAC_INIT_VLC_STATIC( 3, 180); AAC_INIT_VLC_STATIC( 4, 172); AAC_INIT_VLC_STATIC( 5, 140); AAC_INIT_VLC_STATIC( 6, 168); AAC_INIT_VLC_STATIC( 7, 114); AAC_INIT_VLC_STATIC( 8, 262); AAC_INIT_VLC_STATIC( 9, 248); AAC_INIT_VLC_STATIC(10, 384); dsputil_init(&ac->dsp, avccontext); ac->random_state = 0x1f2e3d4c; // -1024 - Compensate wrong IMDCT method. // 32768 - Required to scale values to the correct range for the bias method // for float to int16 conversion. if(ac->dsp.float_to_int16 == ff_float_to_int16_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_sine_window_init(ff_sine_1024, 1024); ff_sine_window_init(ff_sine_128, 128); return 0; } /** * Skip data_stream_element; reference: table 4.10. */ static void skip_data_stream_element(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); skip_bits_long(gb, 8 * count); } static int decode_prediction(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb) { int sfb; if (get_bits1(gb)) { ics->predictor_reset_group = get_bits(gb, 5); if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) { av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n"); return -1; } } for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) { ics->prediction_used[sfb] = get_bits1(gb); } return 0; } /** * Decode Individual Channel Stream info; reference: table 4.6. * * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. */ static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) { if (get_bits1(gb)) { av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n"); memset(ics, 0, sizeof(IndividualChannelStream)); return -1; } ics->window_sequence[1] = ics->window_sequence[0]; ics->window_sequence[0] = get_bits(gb, 2); ics->use_kb_window[1] = ics->use_kb_window[0]; ics->use_kb_window[0] = get_bits1(gb); ics->num_window_groups = 1; ics->group_len[0] = 1; if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { int i; ics->max_sfb = get_bits(gb, 4); for (i = 0; i < 7; i++) { if (get_bits1(gb)) { ics->group_len[ics->num_window_groups-1]++; } else { ics->num_window_groups++; ics->group_len[ics->num_window_groups-1] = 1; } } ics->num_windows = 8; ics->swb_offset = ff_swb_offset_128[ac->m4ac.sampling_index]; ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; ics->tns_max_bands = ff_tns_max_bands_128[ac->m4ac.sampling_index]; ics->predictor_present = 0; } else { ics->max_sfb = get_bits(gb, 6); ics->num_windows = 1; ics->swb_offset = ff_swb_offset_1024[ac->m4ac.sampling_index]; ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; ics->tns_max_bands = ff_tns_max_bands_1024[ac->m4ac.sampling_index]; ics->predictor_present = get_bits1(gb); ics->predictor_reset_group = 0; if (ics->predictor_present) { if (ac->m4ac.object_type == AOT_AAC_MAIN) { if (decode_prediction(ac, ics, gb)) { memset(ics, 0, sizeof(IndividualChannelStream)); return -1; } } else if (ac->m4ac.object_type == AOT_AAC_LC) { av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n"); memset(ics, 0, sizeof(IndividualChannelStream)); return -1; } else { av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1); memset(ics, 0, sizeof(IndividualChannelStream)); return -1; } } } if(ics->max_sfb > ics->num_swb) { av_log(ac->avccontext, AV_LOG_ERROR, "Number of scalefactor bands in group (%d) exceeds limit (%d).\n", ics->max_sfb, ics->num_swb); memset(ics, 0, sizeof(IndividualChannelStream)); return -1; } return 0; } /** * Decode band types (section_data payload); reference: table 4.46. * * @param band_type array of the used band type * @param band_type_run_end array of the last scalefactor band of a band type run * * @return Returns error status. 0 - OK, !0 - error */ static int decode_band_types(AACContext * ac, enum BandType band_type[120], int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) { int g, idx = 0; const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; for (g = 0; g < ics->num_window_groups; g++) { int k = 0; while (k < ics->max_sfb) { uint8_t sect_len = k; int sect_len_incr; int sect_band_type = get_bits(gb, 4); if (sect_band_type == 12) { av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n"); return -1; } while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1) sect_len += sect_len_incr; sect_len += sect_len_incr; if (sect_len > ics->max_sfb) { av_log(ac->avccontext, AV_LOG_ERROR, "Number of bands (%d) exceeds limit (%d).\n", sect_len, ics->max_sfb); return -1; } for (; k < sect_len; k++) { band_type [idx] = sect_band_type; band_type_run_end[idx++] = sect_len; } } } return 0; } /** * Decode scalefactors; reference: table 4.47. * * @param global_gain first scalefactor value as scalefactors are differentially coded * @param band_type array of the used band type * @param band_type_run_end array of the last scalefactor band of a band type run * @param sf array of scalefactors or intensity stereo positions * * @return Returns error status. 0 - OK, !0 - error */ static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb, unsigned int global_gain, IndividualChannelStream * ics, enum BandType band_type[120], int band_type_run_end[120]) { const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); int g, i, idx = 0; int offset[3] = { global_gain, global_gain - 90, 100 }; int noise_flag = 1; static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; for (g = 0; g < ics->num_window_groups; g++) { for (i = 0; i < ics->max_sfb;) { int run_end = band_type_run_end[idx]; if (band_type[idx] == ZERO_BT) { for(; i < run_end; i++, idx++) sf[idx] = 0.; }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { for(; i < run_end; i++, idx++) { offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; if(offset[2] > 255U) { av_log(ac->avccontext, AV_LOG_ERROR, "%s (%d) out of range.\n", sf_str[2], offset[2]); return -1; } sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; } }else if(band_type[idx] == NOISE_BT) { for(; i < run_end; i++, idx++) { if(noise_flag-- > 0) offset[1] += get_bits(gb, 9) - 256; else offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; if(offset[1] > 255U) { av_log(ac->avccontext, AV_LOG_ERROR, "%s (%d) out of range.\n", sf_str[1], offset[1]); return -1; } sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset + 100]; } }else { for(; i < run_end; i++, idx++) { offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; if(offset[0] > 255U) { av_log(ac->avccontext, AV_LOG_ERROR, "%s (%d) out of range.\n", sf_str[0], offset[0]); return -1; } sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; } } } } return 0; } /** * Decode pulse data; reference: table 4.7. */ static int decode_pulses(Pulse * pulse, GetBitContext * gb, const uint16_t * swb_offset, int num_swb) { int i, pulse_swb; pulse->num_pulse = get_bits(gb, 2) + 1; pulse_swb = get_bits(gb, 6); if (pulse_swb >= num_swb) return -1; pulse->pos[0] = swb_offset[pulse_swb]; pulse->pos[0] += get_bits(gb, 5); if (pulse->pos[0] > 1023) return -1; pulse->amp[0] = get_bits(gb, 4); for (i = 1; i < pulse->num_pulse; i++) { pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i-1]; if (pulse->pos[i] > 1023) return -1; pulse->amp[i] = get_bits(gb, 4); } return 0; } /** * Decode Temporal Noise Shaping data; reference: table 4.48. * * @return Returns error status. 0 - OK, !0 - error */ static int decode_tns(AACContext * ac, TemporalNoiseShaping * tns, GetBitContext * gb, const IndividualChannelStream * ics) { int w, filt, i, coef_len, coef_res, coef_compress; const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; for (w = 0; w < ics->num_windows; w++) { if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) { coef_res = get_bits1(gb); for (filt = 0; filt < tns->n_filt[w]; filt++) { int tmp2_idx; tns->length[w][filt] = get_bits(gb, 6 - 2*is8); if ((tns->order[w][filt] = get_bits(gb, 5 - 2*is8)) > tns_max_order) { av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.", 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])); } } /** * 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, 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; static const float sign_lookup[] = { 1.0f, -1.0f }; 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++) { for (i = 0; i < ics->max_sfb; i++, idx++) { const int cur_band_type = band_type[idx]; const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4; const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type); int group; if (cur_band_type == ZERO_BT || cur_band_type == INTENSITY_BT2 || cur_band_type == INTENSITY_BT) { for (group = 0; group < ics->group_len[g]; group++) { memset(coef + group * 128 + offsets[i], 0, (offsets[i+1] - offsets[i])*sizeof(float)); } }else if (cur_band_type == NOISE_BT) { for (group = 0; group < ics->group_len[g]; group++) { float scale; float band_energy = 0; for (k = offsets[i]; k < offsets[i+1]; k++) { ac->random_state = lcg_random(ac->random_state); coef[group*128+k] = ac->random_state; band_energy += coef[group*128+k]*coef[group*128+k]; } scale = sf[idx] / sqrtf(band_energy); for (k = offsets[i]; k < offsets[i+1]; k++) { coef[group*128+k] *= scale; } } }else { for (group = 0; group < ics->group_len[g]; group++) { for (k = offsets[i]; k < offsets[i+1]; k += dim) { const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3); const int coef_tmp_idx = (group << 7) + k; const float *vq_ptr; int j; if(index >= ff_aac_spectral_sizes[cur_band_type - 1]) { av_log(ac->avccontext, AV_LOG_ERROR, "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n", cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]); return -1; } vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim]; if (is_cb_unsigned) { if (vq_ptr[0]) coef[coef_tmp_idx ] = sign_lookup[get_bits1(gb)]; if (vq_ptr[1]) coef[coef_tmp_idx + 1] = sign_lookup[get_bits1(gb)]; if (dim == 4) { if (vq_ptr[2]) coef[coef_tmp_idx + 2] = sign_lookup[get_bits1(gb)]; if (vq_ptr[3]) coef[coef_tmp_idx + 3] = sign_lookup[get_bits1(gb)]; } if (cur_band_type == ESC_BT) { for (j = 0; j < 2; j++) { if (vq_ptr[j] == 64.0f) { int n = 4; /* The total length of escape_sequence must be < 22 bits according to the specification (i.e. max is 11111111110xxxxxxxxxx). */ while (get_bits1(gb) && n < 15) n++; if(n == 15) { av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); return -1; } n = (1<<n) + get_bits(gb, n); coef[coef_tmp_idx + j] *= cbrtf(n) * n; }else coef[coef_tmp_idx + j] *= vq_ptr[j]; } }else { coef[coef_tmp_idx ] *= vq_ptr[0]; coef[coef_tmp_idx + 1] *= vq_ptr[1]; if (dim == 4) { coef[coef_tmp_idx + 2] *= vq_ptr[2]; coef[coef_tmp_idx + 3] *= vq_ptr[3]; } } }else { coef[coef_tmp_idx ] = vq_ptr[0]; coef[coef_tmp_idx + 1] = vq_ptr[1]; if (dim == 4) { coef[coef_tmp_idx + 2] = vq_ptr[2]; coef[coef_tmp_idx + 3] = vq_ptr[3]; } } coef[coef_tmp_idx ] *= sf[idx]; coef[coef_tmp_idx + 1] *= sf[idx]; if (dim == 4) { coef[coef_tmp_idx + 2] *= sf[idx]; coef[coef_tmp_idx + 3] *= sf[idx]; } } } } } coef += ics->group_len[g]<<7; } if (pulse_present) { idx = 0; for(i = 0; i < pulse->num_pulse; i++){ float co = coef_base[ pulse->pos[i] ]; while(offsets[idx + 1] <= pulse->pos[i]) idx++; if (band_type[idx] != NOISE_BT && sf[idx]) { float ico = -pulse->amp[i]; if (co) { co /= sf[idx]; ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico); } coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx]; } } } return 0; } static av_always_inline float flt16_round(float pf) { union float754 tmp; tmp.f = pf; tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U; return tmp.f; } static av_always_inline float flt16_even(float pf) { union float754 tmp; tmp.f = pf; tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U>>16)) & 0xFFFF0000U; return tmp.f; } static av_always_inline float flt16_trunc(float pf) { union float754 pun; pun.f = pf; pun.i &= 0xFFFF0000U; return pun.f; } static 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->avccontext, 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->avccontext, 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->avccontext, "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(ChannelElement * cpe) { const IndividualChannelStream * ics = &cpe->ch[0].ics; float *ch0 = cpe->ch[0].coeffs; float *ch1 = cpe->ch[1].coeffs; int g, i, k, 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++) { for (k = offsets[i]; k < offsets[i+1]; k++) { float tmp = ch0[group*128 + k] - ch1[group*128 + k]; ch0[group*128 + k] += ch1[group*128 + k]; ch1[group*128 + k] = tmp; } } } } 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->avccontext, 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(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; } /** * Decode Spectral Band Replication extension data; reference: table 4.55. * * @param crc flag indicating the presence of CRC checksum * @param cnt length of TYPE_FIL syntactic element in bytes * * @return Returns number of bytes consumed from the TYPE_FIL element. */ static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) { // TODO : sbr_extension implementation av_log_missing_feature(ac->avccontext, "SBR", 0); skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type return cnt; } /** * 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) { 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: res = decode_sbr_extension(ac, gb, crc_flag, cnt); 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) { 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->avccontext, 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, ac->add_bias, 512); } else { for (i = 0; i < 448; i++) out[i] = saved[i] + ac->add_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, ac->add_bias, 64); ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, ac->add_bias, 64); ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, ac->add_bias, 64); ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, ac->add_bias, 64); ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, ac->add_bias, 64); memcpy( out + 448 + 4*128, temp, 64 * sizeof(float)); } else { ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, ac->add_bias, 64); for (i = 576; i < 1024; i++) out[i] = buf[i-512] + ac->add_bias; } } // buffer update if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { for (i = 0; i < 64; i++) saved[i] = temp[64 + i] - ac->add_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->avccontext, 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; for (i = 0; i < 1024; 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; 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]); if(type == TYPE_CPE) imdct_and_windowing(ac, &che->ch[1]); 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 (hdr_info.chan_config) ac->m4ac.chan_config = hdr_info.chan_config; 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 (hdr_info.num_aac_frames == 1) { if (!hdr_info.crc_absent) skip_bits(gb, 16); } else { av_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0); return -1; } } return size; } static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; AACContext * ac = avccontext->priv_data; ChannelElement * che = NULL; GetBitContext gb; enum RawDataBlockType elem_type; int err, elem_id, data_size_tmp; init_get_bits(&gb, buf, buf_size*8); if (show_bits(&gb, 12) == 0xfff) { if (parse_adts_frame_header(ac, &gb) < 0) { av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n"); return -1; } if (ac->m4ac.sampling_index > 12) { av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); return -1; } } // 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->avccontext, 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: skip_data_stream_element(&gb); err = 0; 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; err = output_configure(ac, ac->che_pos, new_che_pos, 0); break; } case TYPE_FIL: if (elem_id == 15) elem_id += get_bits(&gb, 8) - 1; while (elem_id > 0) elem_id -= decode_extension_payload(ac, &gb, elem_id); err = 0; /* FIXME */ break; default: err = -1; /* should not happen, but keeps compiler happy */ break; } if(err) return err; } spectral_to_sample(ac); if (!ac->is_saved) { ac->is_saved = 1; *data_size = 0; return buf_size; } data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t); if(*data_size < data_size_tmp) { av_log(avccontext, 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, 1024, avccontext->channels); return buf_size; } static av_cold int aac_decode_close(AVCodecContext * avccontext) { AACContext * ac = avccontext->priv_data; int i, type; for (i = 0; i < MAX_ELEM_ID; i++) { for(type = 0; type < 4; type++) av_freep(&ac->che[type][i]); } ff_mdct_end(&ac->mdct); ff_mdct_end(&ac->mdct_small); return 0 ; } AVCodec aac_decoder = { "aac", CODEC_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 = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, };