Mercurial > libavcodec.hg
view aac.c @ 11186:90de754f70d6 libavcodec
2 cpu cycles faster context calculation for decode_cabac_intra_mb_type()
| author | michael |
|---|---|
| date | Tue, 16 Feb 2010 02:51:37 +0000 |
| parents | 69020e1846fe |
| children | f5d50932acc0 |
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/* * 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> #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 uint32_t cbrt_tab[1<<13]; static ChannelElement *get_che(AACContext *ac, int type, int elem_id) { if (ac->tag_che_map[type][elem_id]) { return ac->tag_che_map[type][elem_id]; } if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) { return NULL; } switch (ac->m4ac.chan_config) { case 7: if (ac->tags_mapped == 3 && type == TYPE_CPE) { ac->tags_mapped++; return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2]; } case 6: /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1] instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have encountered such a stream, transfer the LFE[0] element to SCE[1] */ if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) { ac->tags_mapped++; return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0]; } case 5: if (ac->tags_mapped == 2 && type == TYPE_CPE) { ac->tags_mapped++; return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1]; } case 4: if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) { ac->tags_mapped++; return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; } case 3: case 2: if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) { ac->tags_mapped++; return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0]; } else if (ac->m4ac.chan_config == 2) { return NULL; } case 1: if (!ac->tags_mapped && type == TYPE_SCE) { ac->tags_mapped++; return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0]; } default: return NULL; } } /** * Check for the channel element in the current channel position configuration. * If it exists, make sure the appropriate element is allocated and map the * channel order to match the internal FFmpeg channel layout. * * @param che_pos current channel position configuration * @param type channel element type * @param id channel element id * @param channels count of the number of channels in the configuration * * @return Returns error status. 0 - OK, !0 - error */ static av_cold int che_configure(AACContext *ac, enum ChannelPosition che_pos[4][MAX_ELEM_ID], int type, int id, int *channels) { if (che_pos[type][id]) { if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement)))) return AVERROR(ENOMEM); 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 av_freep(&ac->che[type][id]); return 0; } /** * Configure output channel order based on the current program configuration element. * * @param che_pos current channel position configuration * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. * * @return Returns error status. 0 - OK, !0 - error */ static av_cold int output_configure(AACContext *ac, enum ChannelPosition che_pos[4][MAX_ELEM_ID], enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], int channel_config, enum OCStatus oc_type) { AVCodecContext *avctx = ac->avccontext; int i, type, channels = 0, ret; memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); if (channel_config) { for (i = 0; i < tags_per_config[channel_config]; i++) { if ((ret = che_configure(ac, che_pos, aac_channel_layout_map[channel_config - 1][i][0], aac_channel_layout_map[channel_config - 1][i][1], &channels))) return ret; } memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); ac->tags_mapped = 0; avctx->channel_layout = aac_channel_layout[channel_config - 1]; } else { /* Allocate or free elements depending on if they are in the * current program configuration. * * Set up default 1:1 output mapping. * * For a 5.1 stream the output order will be: * [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ] */ for (i = 0; i < MAX_ELEM_ID; i++) { for (type = 0; type < 4; type++) { if ((ret = che_configure(ac, che_pos, type, i, &channels))) return ret; } } memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); ac->tags_mapped = 4 * MAX_ELEM_ID; avctx->channel_layout = 0; } avctx->channels = channels; ac->output_configured = oc_type; return 0; } /** * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. * * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. * @param sce_map mono (Single Channel Element) map * @param type speaker type/position for these channels */ static void decode_channel_map(enum ChannelPosition *cpe_map, enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext *gb, int n) { while (n--) { enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map map[get_bits(gb, 4)] = type; } } /** * Decode program configuration element; reference: table 4.2. * * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. * * @return Returns error status. 0 - OK, !0 - error */ static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], GetBitContext *gb) { int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index; 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 av_cold int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], int channel_config) { if (channel_config < 1 || channel_config > 7) { av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", channel_config); return -1; } /* default channel configurations: * * 1ch : front center (mono) * 2ch : L + R (stereo) * 3ch : front center + L + R * 4ch : front center + L + R + back center * 5ch : front center + L + R + back stereo * 6ch : front center + L + R + back stereo + LFE * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE */ if (channel_config != 2) new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) if (channel_config > 1) new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) if (channel_config == 4) new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center if (channel_config > 4) new_che_pos[TYPE_CPE][(channel_config == 7) + 1] = AAC_CHANNEL_BACK; // back stereo if (channel_config > 5) new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE if (channel_config == 7) new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right return 0; } /** * Decode GA "General Audio" specific configuration; reference: table 4.1. * * @return Returns error status. 0 - OK, !0 - error */ static int decode_ga_specific_config(AACContext *ac, GetBitContext *gb, int channel_config) { enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; int extension_flag, ret; if (get_bits1(gb)) { // frameLengthFlag av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1); return -1; } if (get_bits1(gb)) // dependsOnCoreCoder skip_bits(gb, 14); // coreCoderDelay extension_flag = get_bits1(gb); if (ac->m4ac.object_type == AOT_AAC_SCALABLE || ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) skip_bits(gb, 3); // layerNr memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); if (channel_config == 0) { skip_bits(gb, 4); // element_instance_tag if ((ret = decode_pce(ac, new_che_pos, gb))) return ret; } else { if ((ret = set_default_channel_config(ac, new_che_pos, channel_config))) return ret; } if ((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR))) return ret; if (extension_flag) { switch (ac->m4ac.object_type) { case AOT_ER_BSAC: skip_bits(gb, 5); // numOfSubFrame skip_bits(gb, 11); // layer_length break; case AOT_ER_AAC_LC: case AOT_ER_AAC_LTP: case AOT_ER_AAC_SCALABLE: case AOT_ER_AAC_LD: skip_bits(gb, 3); /* aacSectionDataResilienceFlag * aacScalefactorDataResilienceFlag * aacSpectralDataResilienceFlag */ break; } skip_bits1(gb); // extensionFlag3 (TBD in version 3) } return 0; } /** * Decode audio specific configuration; reference: table 1.13. * * @param data pointer to AVCodecContext extradata * @param data_size size of AVCCodecContext extradata * * @return Returns error status. 0 - OK, !0 - error */ static int decode_audio_specific_config(AACContext *ac, void *data, int data_size) { GetBitContext gb; int i; init_get_bits(&gb, data, data_size * 8); if ((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) return -1; if (ac->m4ac.sampling_index > 12) { av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); return -1; } skip_bits_long(&gb, i); switch (ac->m4ac.object_type) { case AOT_AAC_MAIN: case AOT_AAC_LC: if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) return -1; break; default: av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); return -1; } return 0; } /** * linear congruential pseudorandom number generator * * @param previous_val pointer to the current state of the generator * * @return Returns a 32-bit pseudorandom integer */ static av_always_inline int lcg_random(int previous_val) { return previous_val * 1664525 + 1013904223; } static 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) { ac->m4ac.sample_rate = avccontext->sample_rate; } avccontext->sample_fmt = SAMPLE_FMT_S16; avccontext->frame_size = 1024; 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); dsputil_init(&ac->dsp, avccontext); ac->random_state = 0x1f2e3d4c; // -1024 - Compensate wrong IMDCT method. // 32768 - Required to scale values to the correct range for the bias method // for float to int16 conversion. if (ac->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) { ac->add_bias = 385.0f; ac->sf_scale = 1. / (-1024. * 32768.); ac->sf_offset = 0; } else { ac->add_bias = 0.0f; ac->sf_scale = 1. / -1024.; ac->sf_offset = 60; } #if !CONFIG_HARDCODED_TABLES for (i = 0; i < 428; i++) ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.); #endif /* CONFIG_HARDCODED_TABLES */ INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code), ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), 352); ff_mdct_init(&ac->mdct, 11, 1, 1.0); ff_mdct_init(&ac->mdct_small, 8, 1, 1.0); // window initialization ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); ff_init_ff_sine_windows(10); ff_init_ff_sine_windows( 7); if (!cbrt_tab[(1<<13) - 1]) { for (i = 0; i < 1<<13; i++) { union float754 f; f.f = cbrtf(i) * i; cbrt_tab[i] = f.i; } } 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_end = k; int sect_len_incr; int sect_band_type = get_bits(gb, 4); if (sect_band_type == 12) { av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n"); return -1; } while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1) sect_end += sect_len_incr; sect_end += sect_len_incr; if (sect_end > ics->max_sfb) { av_log(ac->avccontext, AV_LOG_ERROR, "Number of bands (%d) exceeds limit (%d).\n", sect_end, ics->max_sfb); return -1; } for (; k < sect_end; k++) { band_type [idx] = sect_band_type; band_type_run_end[idx++] = sect_end; } } } return 0; } /** * Decode scalefactors; reference: table 4.47. * * @param global_gain first scalefactor value as scalefactors are differentially coded * @param band_type array of the used band type * @param band_type_run_end array of the last scalefactor band of a band type run * @param sf array of scalefactors or intensity stereo positions * * @return Returns error status. 0 - OK, !0 - error */ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb, unsigned int global_gain, IndividualChannelStream *ics, enum BandType band_type[120], int band_type_run_end[120]) { const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); int g, i, idx = 0; int offset[3] = { global_gain, global_gain - 90, 100 }; int noise_flag = 1; static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; for (g = 0; g < ics->num_window_groups; g++) { for (i = 0; i < ics->max_sfb;) { int run_end = band_type_run_end[idx]; if (band_type[idx] == ZERO_BT) { for (; i < run_end; i++, idx++) sf[idx] = 0.; } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { for (; i < run_end; i++, idx++) { offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; if (offset[2] > 255U) { av_log(ac->avccontext, AV_LOG_ERROR, "%s (%d) out of range.\n", sf_str[2], offset[2]); return -1; } sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; } } else if (band_type[idx] == NOISE_BT) { for (; i < run_end; i++, idx++) { if (noise_flag-- > 0) offset[1] += get_bits(gb, 9) - 256; else offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; if (offset[1] > 255U) { av_log(ac->avccontext, AV_LOG_ERROR, "%s (%d) out of range.\n", sf_str[1], offset[1]); return -1; } sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100]; } } else { for (; i < run_end; i++, idx++) { offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; if (offset[0] > 255U) { av_log(ac->avccontext, AV_LOG_ERROR, "%s (%d) out of range.\n", sf_str[0], offset[0]); return -1; } sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; } } } } return 0; } /** * Decode pulse data; reference: table 4.7. */ static int decode_pulses(Pulse *pulse, GetBitContext *gb, const uint16_t *swb_offset, int num_swb) { int i, pulse_swb; pulse->num_pulse = get_bits(gb, 2) + 1; pulse_swb = get_bits(gb, 6); if (pulse_swb >= num_swb) return -1; pulse->pos[0] = swb_offset[pulse_swb]; pulse->pos[0] += get_bits(gb, 5); if (pulse->pos[0] > 1023) return -1; pulse->amp[0] = get_bits(gb, 4); for (i = 1; i < pulse->num_pulse; i++) { pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1]; if (pulse->pos[i] > 1023) return -1; pulse->amp[i] = get_bits(gb, 4); } return 0; } /** * Decode Temporal Noise Shaping data; reference: table 4.48. * * @return Returns error status. 0 - OK, !0 - error */ static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns, GetBitContext *gb, const IndividualChannelStream *ics) { int w, filt, i, coef_len, coef_res, coef_compress; const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; for (w = 0; w < ics->num_windows; w++) { if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) { coef_res = get_bits1(gb); for (filt = 0; filt < tns->n_filt[w]; filt++) { int tmp2_idx; tns->length[w][filt] = get_bits(gb, 6 - 2 * is8); if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) { av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.", 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->avccontext, 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->avccontext, 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 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(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->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(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; } /** * 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 (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->avccontext->sample_rate) ac->avccontext->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->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; if (ac->output_configured > OC_TRIAL_PCE) av_log(avccontext, 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; 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); 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); if (ac->output_configured) ac->output_configured = OC_LOCKED; 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 = (const enum SampleFormat[]) { SAMPLE_FMT_S16,SAMPLE_FMT_NONE }, .channel_layouts = aac_channel_layout, };