/* * Common parts of the AAC decoders * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) * Copyright (c) 2008-2013 Alex Converse * * AAC LATM decoder * Copyright (c) 2008-2010 Paul Kendall * Copyright (c) 2010 Janne Grunau * * AAC decoder fixed-point implementation * Copyright (c) 2013 * MIPS Technologies, Inc., California. * * 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 */ /* We use several quantization functions here (Q31, Q30), * for which we need this to be defined for them to work as expected. */ #define USE_FIXED 1 #include "config_components.h" #include #include #include "aacdec.h" #include "aacdec_tab.h" #include "aacdec_usac.h" #include "libavcodec/aac.h" #include "libavcodec/aac_defines.h" #include "libavcodec/aacsbr.h" #include "libavcodec/aactab.h" #include "libavcodec/adts_header.h" #include "libavcodec/avcodec.h" #include "libavcodec/internal.h" #include "libavcodec/codec_internal.h" #include "libavcodec/decode.h" #include "libavcodec/profiles.h" #include "libavutil/attributes.h" #include "libavutil/error.h" #include "libavutil/log.h" #include "libavutil/macros.h" #include "libavutil/mem.h" #include "libavutil/opt.h" #include "libavutil/tx.h" #include "libavutil/version.h" /* * 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 * Y Long Term Prediction * Y intensity stereo * Y channel coupling * Y frequency domain prediction * Y Perceptual Noise Substitution * Y Mid/Side stereo * N Scalable Inverse AAC Quantization * N Frequency Selective Switch * N upsampling filter * Y quantization & coding - AAC * N quantization & coding - TwinVQ * N quantization & coding - BSAC * N AAC Error Resilience tools * N Error Resilience payload syntax * N Error Protection tool * N CELP * N Silence Compression * N HVXC * N HVXC 4kbits/s VR * N Structured Audio tools * N Structured Audio Sample Bank Format * N MIDI * N Harmonic and Individual Lines plus Noise * N Text-To-Speech Interface * Y Spectral Band Replication * Y (not in this code) Layer-1 * Y (not in this code) Layer-2 * Y (not in this code) Layer-3 * N SinuSoidal Coding (Transient, Sinusoid, Noise) * Y Parametric Stereo * N Direct Stream Transfer * Y (not in fixed point code) Enhanced AAC Low Delay (ER AAC ELD) * * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication. * - HE AAC v2 comprises LC AAC with Spectral Band Replication and Parametric Stereo. */ #define overread_err "Input buffer exhausted before END element found\n" static int count_channels(uint8_t (*layout)[3], int tags) { int i, sum = 0; for (i = 0; i < tags; i++) { int syn_ele = layout[i][0]; int pos = layout[i][2]; sum += (1 + (syn_ele == TYPE_CPE)) * (pos != AAC_CHANNEL_OFF && pos != AAC_CHANNEL_CC); } return sum; } /** * 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(AACDecContext *ac, enum ChannelPosition che_pos, int type, int id, int *channels) { if (*channels >= MAX_CHANNELS) return AVERROR_INVALIDDATA; if (che_pos) { if (!ac->che[type][id]) { int ret = ac->proc.sbr_ctx_alloc_init(ac, &ac->che[type][id], type); if (ret < 0) return ret; } if (type != TYPE_CCE) { if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) { av_log(ac->avctx, AV_LOG_ERROR, "Too many channels\n"); return AVERROR_INVALIDDATA; } ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0]; if (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) { ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1]; } } } else { if (ac->che[type][id]) { ac->proc.sbr_ctx_close(ac->che[type][id]); } av_freep(&ac->che[type][id]); } return 0; } static int frame_configure_elements(AVCodecContext *avctx) { AACDecContext *ac = avctx->priv_data; int type, id, ch, ret; /* set channel pointers to internal buffers by default */ for (type = 0; type < 4; type++) { for (id = 0; id < MAX_ELEM_ID; id++) { ChannelElement *che = ac->che[type][id]; if (che) { che->ch[0].output = che->ch[0].ret_buf; che->ch[1].output = che->ch[1].ret_buf; } } } /* get output buffer */ av_frame_unref(ac->frame); if (!avctx->ch_layout.nb_channels) return 1; ac->frame->nb_samples = 2048; if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0) return ret; /* map output channel pointers to AVFrame data */ for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) { if (ac->output_element[ch]) ac->output_element[ch]->output = (void *)ac->frame->extended_data[ch]; } return 0; } struct elem_to_channel { uint64_t av_position; uint8_t syn_ele; uint8_t elem_id; uint8_t aac_position; }; static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID], uint8_t (*layout_map)[3], int offset, uint64_t left, uint64_t right, int pos, uint64_t *layout) { if (layout_map[offset][0] == TYPE_CPE) { e2c_vec[offset] = (struct elem_to_channel) { .av_position = left | right, .syn_ele = TYPE_CPE, .elem_id = layout_map[offset][1], .aac_position = pos }; if (e2c_vec[offset].av_position != UINT64_MAX) *layout |= e2c_vec[offset].av_position; return 1; } else { e2c_vec[offset] = (struct elem_to_channel) { .av_position = left, .syn_ele = TYPE_SCE, .elem_id = layout_map[offset][1], .aac_position = pos }; e2c_vec[offset + 1] = (struct elem_to_channel) { .av_position = right, .syn_ele = TYPE_SCE, .elem_id = layout_map[offset + 1][1], .aac_position = pos }; if (left != UINT64_MAX) *layout |= left; if (right != UINT64_MAX) *layout |= right; return 2; } } static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos, int current) { int num_pos_channels = 0; int first_cpe = 0; int sce_parity = 0; int i; for (i = current; i < tags; i++) { if (layout_map[i][2] != pos) break; if (layout_map[i][0] == TYPE_CPE) { if (sce_parity) { if (pos == AAC_CHANNEL_FRONT && !first_cpe) { sce_parity = 0; } else { return -1; } } num_pos_channels += 2; first_cpe = 1; } else { num_pos_channels++; sce_parity ^= (pos != AAC_CHANNEL_LFE); } } if (sce_parity && (pos == AAC_CHANNEL_FRONT && first_cpe)) return -1; return num_pos_channels; } static int assign_channels(struct elem_to_channel e2c_vec[MAX_ELEM_ID], uint8_t (*layout_map)[3], uint64_t *layout, int tags, int layer, int pos, int *current) { int i = *current, j = 0; int nb_channels = count_paired_channels(layout_map, tags, pos, i); if (nb_channels < 0 || nb_channels > 5) return 0; if (pos == AAC_CHANNEL_LFE) { while (nb_channels) { if (ff_aac_channel_map[layer][pos - 1][j] == AV_CHAN_NONE) return -1; e2c_vec[i] = (struct elem_to_channel) { .av_position = 1ULL << ff_aac_channel_map[layer][pos - 1][j], .syn_ele = layout_map[i][0], .elem_id = layout_map[i][1], .aac_position = pos }; *layout |= e2c_vec[i].av_position; i++; j++; nb_channels--; } *current = i; return 0; } while (nb_channels & 1) { if (ff_aac_channel_map[layer][pos - 1][0] == AV_CHAN_NONE) return -1; if (ff_aac_channel_map[layer][pos - 1][0] == AV_CHAN_UNUSED) break; e2c_vec[i] = (struct elem_to_channel) { .av_position = 1ULL << ff_aac_channel_map[layer][pos - 1][0], .syn_ele = layout_map[i][0], .elem_id = layout_map[i][1], .aac_position = pos }; *layout |= e2c_vec[i].av_position; i++; nb_channels--; } j = (pos != AAC_CHANNEL_SIDE) && nb_channels <= 3 ? 3 : 1; while (nb_channels >= 2) { if (ff_aac_channel_map[layer][pos - 1][j] == AV_CHAN_NONE || ff_aac_channel_map[layer][pos - 1][j+1] == AV_CHAN_NONE) return -1; i += assign_pair(e2c_vec, layout_map, i, 1ULL << ff_aac_channel_map[layer][pos - 1][j], 1ULL << ff_aac_channel_map[layer][pos - 1][j+1], pos, layout); j += 2; nb_channels -= 2; } while (nb_channels & 1) { if (ff_aac_channel_map[layer][pos - 1][5] == AV_CHAN_NONE) return -1; e2c_vec[i] = (struct elem_to_channel) { .av_position = 1ULL << ff_aac_channel_map[layer][pos - 1][5], .syn_ele = layout_map[i][0], .elem_id = layout_map[i][1], .aac_position = pos }; *layout |= e2c_vec[i].av_position; i++; nb_channels--; } if (nb_channels) return -1; *current = i; return 0; } static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags) { int i, n, total_non_cc_elements; struct elem_to_channel e2c_vec[4 * MAX_ELEM_ID] = { { 0 } }; uint64_t layout = 0; if (FF_ARRAY_ELEMS(e2c_vec) < tags) return 0; for (n = 0, i = 0; n < 3 && i < tags; n++) { int ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_FRONT, &i); if (ret < 0) return 0; ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_SIDE, &i); if (ret < 0) return 0; ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_BACK, &i); if (ret < 0) return 0; ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_LFE, &i); if (ret < 0) return 0; } total_non_cc_elements = n = i; if (layout == AV_CH_LAYOUT_22POINT2) { // For 22.2 reorder the result as needed FFSWAP(struct elem_to_channel, e2c_vec[2], e2c_vec[0]); // FL & FR first (final), FC third FFSWAP(struct elem_to_channel, e2c_vec[2], e2c_vec[1]); // FC second (final), FLc & FRc third FFSWAP(struct elem_to_channel, e2c_vec[6], e2c_vec[2]); // LFE1 third (final), FLc & FRc seventh FFSWAP(struct elem_to_channel, e2c_vec[4], e2c_vec[3]); // BL & BR fourth (final), SiL & SiR fifth FFSWAP(struct elem_to_channel, e2c_vec[6], e2c_vec[4]); // FLc & FRc fifth (final), SiL & SiR seventh FFSWAP(struct elem_to_channel, e2c_vec[7], e2c_vec[6]); // LFE2 seventh (final), SiL & SiR eight (final) FFSWAP(struct elem_to_channel, e2c_vec[9], e2c_vec[8]); // TpFL & TpFR ninth (final), TFC tenth (final) FFSWAP(struct elem_to_channel, e2c_vec[11], e2c_vec[10]); // TC eleventh (final), TpSiL & TpSiR twelth FFSWAP(struct elem_to_channel, e2c_vec[12], e2c_vec[11]); // TpBL & TpBR twelth (final), TpSiL & TpSiR thirteenth (final) } else { // For everything else, utilize the AV channel position define as a // stable sort. do { int next_n = 0; for (i = 1; i < n; i++) if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) { FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]); next_n = i; } n = next_n; } while (n > 0); } for (i = 0; i < total_non_cc_elements; i++) { layout_map[i][0] = e2c_vec[i].syn_ele; layout_map[i][1] = e2c_vec[i].elem_id; layout_map[i][2] = e2c_vec[i].aac_position; } return layout; } /** * Save current output configuration if and only if it has been locked. */ static int push_output_configuration(AACDecContext *ac) { int pushed = 0; if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) { ac->oc[0] = ac->oc[1]; pushed = 1; } ac->oc[1].status = OC_NONE; return pushed; } /** * Restore the previous output configuration if and only if the current * configuration is unlocked. */ static void pop_output_configuration(AACDecContext *ac) { if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) { ac->oc[1] = ac->oc[0]; ac->avctx->ch_layout = ac->oc[1].ch_layout; ff_aac_output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags, ac->oc[1].status, 0); } } /** * Configure output channel order based on the current program * configuration element. * * @return Returns error status. 0 - OK, !0 - error */ int ff_aac_output_configure(AACDecContext *ac, uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags, enum OCStatus oc_type, int get_new_frame) { AVCodecContext *avctx = ac->avctx; int i, channels = 0, ret; uint64_t layout = 0; uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }}; uint8_t type_counts[TYPE_END] = { 0 }; if (ac->oc[1].layout_map != layout_map) { memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0])); ac->oc[1].layout_map_tags = tags; } for (i = 0; i < tags; i++) { int type = layout_map[i][0]; int id = layout_map[i][1]; id_map[type][id] = type_counts[type]++; if (id_map[type][id] >= MAX_ELEM_ID) { avpriv_request_sample(ac->avctx, "Too large remapped id"); return AVERROR_PATCHWELCOME; } } // Try to sniff a reasonable channel order, otherwise output the // channels in the order the PCE declared them. if (ac->output_channel_order == CHANNEL_ORDER_DEFAULT) layout = sniff_channel_order(layout_map, tags); for (i = 0; i < tags; i++) { int type = layout_map[i][0]; int id = layout_map[i][1]; int iid = id_map[type][id]; int position = layout_map[i][2]; // Allocate or free elements depending on if they are in the // current program configuration. ret = che_configure(ac, position, type, iid, &channels); if (ret < 0) return ret; ac->tag_che_map[type][id] = ac->che[type][iid]; } if (ac->oc[1].m4ac.ps == 1 && channels == 2) { if (layout == AV_CH_FRONT_CENTER) { layout = AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT; } else { layout = 0; } } av_channel_layout_uninit(&ac->oc[1].ch_layout); if (layout) av_channel_layout_from_mask(&ac->oc[1].ch_layout, layout); else { ac->oc[1].ch_layout.order = AV_CHANNEL_ORDER_UNSPEC; ac->oc[1].ch_layout.nb_channels = channels; } av_channel_layout_copy(&avctx->ch_layout, &ac->oc[1].ch_layout); ac->oc[1].status = oc_type; if (get_new_frame) { if ((ret = frame_configure_elements(ac->avctx)) < 0) return ret; } return 0; } static av_cold void flush(AVCodecContext *avctx) { AACDecContext *ac= avctx->priv_data; int type, i, j; for (type = 3; type >= 0; type--) { for (i = 0; i < MAX_ELEM_ID; i++) { ChannelElement *che = ac->che[type][i]; if (che) { for (j = 0; j <= 1; j++) { memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved)); } } } } ff_aac_usac_reset_state(ac, &ac->oc[1]); } /** * Set up channel positions based on a default channel configuration * as specified in table 1.17. * * @return Returns error status. 0 - OK, !0 - error */ int ff_aac_set_default_channel_config(AACDecContext *ac, AVCodecContext *avctx, uint8_t (*layout_map)[3], int *tags, int channel_config) { if (channel_config < 1 || (channel_config > 7 && channel_config < 11) || channel_config > 14) { av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", channel_config); return AVERROR_INVALIDDATA; } *tags = ff_tags_per_config[channel_config]; memcpy(layout_map, ff_aac_channel_layout_map[channel_config - 1], *tags * sizeof(*layout_map)); /* * AAC specification has 7.1(wide) as a default layout for 8-channel streams. * However, at least Nero AAC encoder encodes 7.1 streams using the default * channel config 7, mapping the side channels of the original audio stream * to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD * decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding * the incorrect streams as if they were correct (and as the encoder intended). * * As actual intended 7.1(wide) streams are very rare, default to assuming a * 7.1 layout was intended. */ if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) { layout_map[2][2] = AAC_CHANNEL_BACK; if (!ac || !ac->warned_71_wide++) { av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout" " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode" " according to the specification instead.\n", FF_COMPLIANCE_STRICT); } } return 0; } ChannelElement *ff_aac_get_che(AACDecContext *ac, int type, int elem_id) { /* For PCE based channel configurations map the channels solely based * on tags. */ if (!ac->oc[1].m4ac.chan_config) { return ac->tag_che_map[type][elem_id]; } // Allow single CPE stereo files to be signalled with mono configuration. if (!ac->tags_mapped && type == TYPE_CPE && ac->oc[1].m4ac.chan_config == 1) { uint8_t layout_map[MAX_ELEM_ID*4][3]; int layout_map_tags; push_output_configuration(ac); av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n"); if (ff_aac_set_default_channel_config(ac, ac->avctx, layout_map, &layout_map_tags, 2) < 0) return NULL; if (ff_aac_output_configure(ac, layout_map, layout_map_tags, OC_TRIAL_FRAME, 1) < 0) return NULL; ac->oc[1].m4ac.chan_config = 2; ac->oc[1].m4ac.ps = 0; } // And vice-versa if (!ac->tags_mapped && type == TYPE_SCE && ac->oc[1].m4ac.chan_config == 2) { uint8_t layout_map[MAX_ELEM_ID * 4][3]; int layout_map_tags; push_output_configuration(ac); av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n"); layout_map_tags = 2; layout_map[0][0] = layout_map[1][0] = TYPE_SCE; layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT; layout_map[0][1] = 0; layout_map[1][1] = 1; if (ff_aac_output_configure(ac, layout_map, layout_map_tags, OC_TRIAL_FRAME, 1) < 0) return NULL; if (ac->oc[1].m4ac.sbr) ac->oc[1].m4ac.ps = -1; } /* For indexed channel configurations map the channels solely based * on position. */ switch (ac->oc[1].m4ac.chan_config) { case 14: if (ac->tags_mapped > 2 && ((type == TYPE_CPE && elem_id < 3) || (type == TYPE_LFE && elem_id < 1))) { ac->tags_mapped++; return ac->tag_che_map[type][elem_id] = ac->che[type][elem_id]; } case 13: if (ac->tags_mapped > 3 && ((type == TYPE_CPE && elem_id < 8) || (type == TYPE_SCE && elem_id < 6) || (type == TYPE_LFE && elem_id < 2))) { ac->tags_mapped++; return ac->tag_che_map[type][elem_id] = ac->che[type][elem_id]; } case 12: 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 11: if (ac->tags_mapped == 3 && type == TYPE_SCE) { ac->tags_mapped++; return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; } case 6: /* Some streams incorrectly code 5.1 audio as * SCE[0] CPE[0] CPE[1] SCE[1] * instead of * SCE[0] CPE[0] CPE[1] LFE[0]. * If we seem to have encountered such a stream, transfer * the LFE[0] element to the SCE[1]'s mapping */ if (ac->tags_mapped == ff_tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) { if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) { av_log(ac->avctx, AV_LOG_WARNING, "This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n", type == TYPE_SCE ? "SCE" : "LFE", elem_id); ac->warned_remapping_once++; } 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: /* Some streams incorrectly code 4.0 audio as * SCE[0] CPE[0] LFE[0] * instead of * SCE[0] CPE[0] SCE[1]. * If we seem to have encountered such a stream, transfer * the SCE[1] element to the LFE[0]'s mapping */ if (ac->tags_mapped == ff_tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) { if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) { av_log(ac->avctx, AV_LOG_WARNING, "This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n", type == TYPE_SCE ? "SCE" : "LFE", elem_id); ac->warned_remapping_once++; } ac->tags_mapped++; return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1]; } if (ac->tags_mapped == 2 && ac->oc[1].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->oc[1].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->tags_mapped == 1 && ac->oc[1].m4ac.chan_config == 2 && type == TYPE_SCE) { ac->tags_mapped++; return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; } 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; } } /** * Decode an array of 4 bit element IDs, optionally interleaved with a * stereo/mono switching bit. * * @param type speaker type/position for these channels */ static void decode_channel_map(uint8_t layout_map[][3], enum ChannelPosition type, GetBitContext *gb, int n) { while (n--) { enum RawDataBlockType syn_ele; switch (type) { case AAC_CHANNEL_FRONT: case AAC_CHANNEL_BACK: case AAC_CHANNEL_SIDE: syn_ele = get_bits1(gb); break; case AAC_CHANNEL_CC: skip_bits1(gb); syn_ele = TYPE_CCE; break; case AAC_CHANNEL_LFE: syn_ele = TYPE_LFE; break; default: // AAC_CHANNEL_OFF has no channel map av_assert0(0); } layout_map[0][0] = syn_ele; layout_map[0][1] = get_bits(gb, 4); layout_map[0][2] = type; layout_map++; } } static inline void relative_align_get_bits(GetBitContext *gb, int reference_position) { int n = (reference_position - get_bits_count(gb) & 7); if (n) skip_bits(gb, n); } /** * Decode program configuration element; reference: table 4.2. * * @return Returns error status. 0 - OK, !0 - error */ static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac, uint8_t (*layout_map)[3], GetBitContext *gb, int byte_align_ref) { int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc; int sampling_index; int comment_len; int tags; skip_bits(gb, 2); // object_type sampling_index = get_bits(gb, 4); if (m4ac->sampling_index != sampling_index) av_log(avctx, AV_LOG_WARNING, "Sample rate index in program config element does not " "match the sample rate index configured by the container.\n"); 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 if (get_bits_left(gb) < 5 * (num_front + num_side + num_back + num_cc) + 4 *(num_lfe + num_assoc_data + num_cc)) { av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err); return -1; } decode_channel_map(layout_map , AAC_CHANNEL_FRONT, gb, num_front); tags = num_front; decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE, gb, num_side); tags += num_side; decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK, gb, num_back); tags += num_back; decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE, gb, num_lfe); tags += num_lfe; skip_bits_long(gb, 4 * num_assoc_data); decode_channel_map(layout_map + tags, AAC_CHANNEL_CC, gb, num_cc); tags += num_cc; relative_align_get_bits(gb, byte_align_ref); /* comment field, first byte is length */ comment_len = get_bits(gb, 8) * 8; if (get_bits_left(gb) < comment_len) { av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err); return AVERROR_INVALIDDATA; } skip_bits_long(gb, comment_len); return tags; } /** * Decode GA "General Audio" specific configuration; reference: table 4.1. * * @param ac pointer to AACDecContext, may be null * @param avctx pointer to AVCCodecContext, used for logging * * @return Returns error status. 0 - OK, !0 - error */ static int decode_ga_specific_config(AACDecContext *ac, AVCodecContext *avctx, GetBitContext *gb, int get_bit_alignment, MPEG4AudioConfig *m4ac, int channel_config) { int extension_flag, ret, ep_config, res_flags; uint8_t layout_map[MAX_ELEM_ID*4][3]; int tags = 0; m4ac->frame_length_short = get_bits1(gb); if (m4ac->frame_length_short && m4ac->sbr == 1) { avpriv_report_missing_feature(avctx, "SBR with 960 frame length"); if (ac) ac->warned_960_sbr = 1; m4ac->sbr = 0; m4ac->ps = 0; } if (get_bits1(gb)) // dependsOnCoreCoder skip_bits(gb, 14); // coreCoderDelay extension_flag = get_bits1(gb); if (m4ac->object_type == AOT_AAC_SCALABLE || m4ac->object_type == AOT_ER_AAC_SCALABLE) skip_bits(gb, 3); // layerNr if (channel_config == 0) { skip_bits(gb, 4); // element_instance_tag tags = decode_pce(avctx, m4ac, layout_map, gb, get_bit_alignment); if (tags < 0) return tags; } else { if ((ret = ff_aac_set_default_channel_config(ac, avctx, layout_map, &tags, channel_config))) return ret; } if (count_channels(layout_map, tags) > 1) { m4ac->ps = 0; } else if (m4ac->sbr == 1 && m4ac->ps == -1) m4ac->ps = 1; if (ac && (ret = ff_aac_output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0))) return ret; if (extension_flag) { switch (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: res_flags = get_bits(gb, 3); if (res_flags) { avpriv_report_missing_feature(avctx, "AAC data resilience (flags %x)", res_flags); return AVERROR_PATCHWELCOME; } break; } skip_bits1(gb); // extensionFlag3 (TBD in version 3) } switch (m4ac->object_type) { case AOT_ER_AAC_LC: case AOT_ER_AAC_LTP: case AOT_ER_AAC_SCALABLE: case AOT_ER_AAC_LD: ep_config = get_bits(gb, 2); if (ep_config) { avpriv_report_missing_feature(avctx, "epConfig %d", ep_config); return AVERROR_PATCHWELCOME; } } return 0; } static int decode_eld_specific_config(AACDecContext *ac, AVCodecContext *avctx, GetBitContext *gb, MPEG4AudioConfig *m4ac, int channel_config) { int ret, ep_config, res_flags; uint8_t layout_map[MAX_ELEM_ID*4][3]; int tags = 0; const int ELDEXT_TERM = 0; m4ac->ps = 0; m4ac->sbr = 0; m4ac->frame_length_short = get_bits1(gb); res_flags = get_bits(gb, 3); if (res_flags) { avpriv_report_missing_feature(avctx, "AAC data resilience (flags %x)", res_flags); return AVERROR_PATCHWELCOME; } if (get_bits1(gb)) { // ldSbrPresentFlag avpriv_report_missing_feature(avctx, "Low Delay SBR"); return AVERROR_PATCHWELCOME; } while (get_bits(gb, 4) != ELDEXT_TERM) { int len = get_bits(gb, 4); if (len == 15) len += get_bits(gb, 8); if (len == 15 + 255) len += get_bits(gb, 16); if (get_bits_left(gb) < len * 8 + 4) { av_log(avctx, AV_LOG_ERROR, overread_err); return AVERROR_INVALIDDATA; } skip_bits_long(gb, 8 * len); } if ((ret = ff_aac_set_default_channel_config(ac, avctx, layout_map, &tags, channel_config))) return ret; if (ac && (ret = ff_aac_output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0))) return ret; ep_config = get_bits(gb, 2); if (ep_config) { avpriv_report_missing_feature(avctx, "epConfig %d", ep_config); return AVERROR_PATCHWELCOME; } return 0; } /** * Decode audio specific configuration; reference: table 1.13. * * @param ac pointer to AACDecContext, may be null * @param avctx pointer to AVCCodecContext, used for logging * @param m4ac pointer to MPEG4AudioConfig, used for parsing * @param gb buffer holding an audio specific config * @param get_bit_alignment relative alignment for byte align operations * @param sync_extension look for an appended sync extension * * @return Returns error status or number of consumed bits. <0 - error */ static int decode_audio_specific_config_gb(AACDecContext *ac, AVCodecContext *avctx, OutputConfiguration *oc, GetBitContext *gb, int get_bit_alignment, int sync_extension) { int i, ret; GetBitContext gbc = *gb; MPEG4AudioConfig *m4ac = &oc->m4ac; MPEG4AudioConfig m4ac_bak = *m4ac; if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, sync_extension, avctx)) < 0) { *m4ac = m4ac_bak; return AVERROR_INVALIDDATA; } if (m4ac->sampling_index > 12) { av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index); *m4ac = m4ac_bak; return AVERROR_INVALIDDATA; } if (m4ac->object_type == AOT_ER_AAC_LD && (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) { av_log(avctx, AV_LOG_ERROR, "invalid low delay sampling rate index %d\n", m4ac->sampling_index); *m4ac = m4ac_bak; return AVERROR_INVALIDDATA; } skip_bits_long(gb, i); switch (m4ac->object_type) { case AOT_AAC_MAIN: case AOT_AAC_LC: case AOT_AAC_SSR: case AOT_AAC_LTP: case AOT_ER_AAC_LC: case AOT_ER_AAC_LD: if ((ret = decode_ga_specific_config(ac, avctx, gb, get_bit_alignment, &oc->m4ac, m4ac->chan_config)) < 0) return ret; break; case AOT_ER_AAC_ELD: if ((ret = decode_eld_specific_config(ac, avctx, gb, &oc->m4ac, m4ac->chan_config)) < 0) return ret; break; #if CONFIG_AAC_DECODER case AOT_USAC: if ((ret = ff_aac_usac_config_decode(ac, avctx, gb, oc, m4ac->chan_config)) < 0) return ret; break; #endif default: avpriv_report_missing_feature(avctx, "Audio object type %s%d", m4ac->sbr == 1 ? "SBR+" : "", m4ac->object_type); return AVERROR(ENOSYS); } ff_dlog(avctx, "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n", m4ac->object_type, m4ac->chan_config, m4ac->sampling_index, m4ac->sample_rate, m4ac->sbr, m4ac->ps); return get_bits_count(gb); } static int decode_audio_specific_config(AACDecContext *ac, AVCodecContext *avctx, OutputConfiguration *oc, const uint8_t *data, int64_t bit_size, int sync_extension) { int i, ret; GetBitContext gb; if (bit_size < 0 || bit_size > INT_MAX) { av_log(avctx, AV_LOG_ERROR, "Audio specific config size is invalid\n"); return AVERROR_INVALIDDATA; } ff_dlog(avctx, "audio specific config size %d\n", (int)bit_size >> 3); for (i = 0; i < bit_size >> 3; i++) ff_dlog(avctx, "%02x ", data[i]); ff_dlog(avctx, "\n"); if ((ret = init_get_bits(&gb, data, bit_size)) < 0) return ret; return decode_audio_specific_config_gb(ac, avctx, oc, &gb, 0, sync_extension); } static int sample_rate_idx (int rate) { if (92017 <= rate) return 0; else if (75132 <= rate) return 1; else if (55426 <= rate) return 2; else if (46009 <= rate) return 3; else if (37566 <= rate) return 4; else if (27713 <= rate) return 5; else if (23004 <= rate) return 6; else if (18783 <= rate) return 7; else if (13856 <= rate) return 8; else if (11502 <= rate) return 9; else if (9391 <= rate) return 10; else return 11; } static av_cold int decode_close(AVCodecContext *avctx) { AACDecContext *ac = avctx->priv_data; for (int i = 0; i < 2; i++) { OutputConfiguration *oc = &ac->oc[i]; AACUSACConfig *usac = &oc->usac; for (int j = 0; j < usac->nb_elems; j++) { AACUsacElemConfig *ec = &usac->elems[i]; av_freep(&ec->ext.pl_data); } } for (int type = 0; type < FF_ARRAY_ELEMS(ac->che); type++) { for (int i = 0; i < MAX_ELEM_ID; i++) { if (ac->che[type][i]) { ac->proc.sbr_ctx_close(ac->che[type][i]); av_freep(&ac->che[type][i]); } } } av_tx_uninit(&ac->mdct96); av_tx_uninit(&ac->mdct120); av_tx_uninit(&ac->mdct128); av_tx_uninit(&ac->mdct480); av_tx_uninit(&ac->mdct512); av_tx_uninit(&ac->mdct768); av_tx_uninit(&ac->mdct960); av_tx_uninit(&ac->mdct1024); av_tx_uninit(&ac->mdct_ltp); // Compiler will optimize this branch away. if (ac->is_fixed) av_freep(&ac->RENAME_FIXED(fdsp)); else av_freep(&ac->fdsp); return 0; } static av_cold int init_dsp(AVCodecContext *avctx) { AACDecContext *ac = avctx->priv_data; int is_fixed = ac->is_fixed, ret; float scale_fixed, scale_float; const float *const scalep = is_fixed ? &scale_fixed : &scale_float; enum AVTXType tx_type = is_fixed ? AV_TX_INT32_MDCT : AV_TX_FLOAT_MDCT; #define MDCT_INIT(s, fn, len, sval) \ scale_fixed = (sval) * 128.0f; \ scale_float = (sval) / 32768.0f; \ ret = av_tx_init(&s, &fn, tx_type, 1, len, scalep, 0); \ if (ret < 0) \ return ret MDCT_INIT(ac->mdct96, ac->mdct96_fn, 96, 1.0/96); MDCT_INIT(ac->mdct120, ac->mdct120_fn, 120, 1.0/120); MDCT_INIT(ac->mdct128, ac->mdct128_fn, 128, 1.0/128); MDCT_INIT(ac->mdct480, ac->mdct480_fn, 480, 1.0/480); MDCT_INIT(ac->mdct512, ac->mdct512_fn, 512, 1.0/512); MDCT_INIT(ac->mdct768, ac->mdct768_fn, 768, 1.0/768); MDCT_INIT(ac->mdct960, ac->mdct960_fn, 960, 1.0/960); MDCT_INIT(ac->mdct1024, ac->mdct1024_fn, 1024, 1.0/1024); #undef MDCT_INIT /* LTP forward MDCT */ scale_fixed = -1.0; scale_float = -32786.0*2 + 36; ret = av_tx_init(&ac->mdct_ltp, &ac->mdct_ltp_fn, tx_type, 0, 1024, scalep, 0); if (ret < 0) return ret; return 0; } av_cold int ff_aac_decode_init(AVCodecContext *avctx) { AACDecContext *ac = avctx->priv_data; int ret; if (avctx->sample_rate > 96000) return AVERROR_INVALIDDATA; ff_aacdec_common_init_once(); ac->avctx = avctx; ac->oc[1].m4ac.sample_rate = avctx->sample_rate; if (avctx->extradata_size > 0) { if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1], avctx->extradata, avctx->extradata_size * 8LL, 1)) < 0) return ret; } else { int sr, i; uint8_t layout_map[MAX_ELEM_ID*4][3]; int layout_map_tags; sr = sample_rate_idx(avctx->sample_rate); ac->oc[1].m4ac.sampling_index = sr; ac->oc[1].m4ac.channels = avctx->ch_layout.nb_channels; ac->oc[1].m4ac.sbr = -1; ac->oc[1].m4ac.ps = -1; for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++) if (ff_mpeg4audio_channels[i] == avctx->ch_layout.nb_channels) break; if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) { i = 0; } ac->oc[1].m4ac.chan_config = i; if (ac->oc[1].m4ac.chan_config) { int ret = ff_aac_set_default_channel_config(ac, avctx, layout_map, &layout_map_tags, ac->oc[1].m4ac.chan_config); if (!ret) ff_aac_output_configure(ac, layout_map, layout_map_tags, OC_GLOBAL_HDR, 0); else if (avctx->err_recognition & AV_EF_EXPLODE) return AVERROR_INVALIDDATA; } } if (avctx->ch_layout.nb_channels > MAX_CHANNELS) { av_log(avctx, AV_LOG_ERROR, "Too many channels\n"); return AVERROR_INVALIDDATA; } ac->random_state = 0x1f2e3d4c; return init_dsp(avctx); } /** * Skip data_stream_element; reference: table 4.10. */ static int skip_data_stream_element(AACDecContext *ac, GetBitContext *gb) { int byte_align = get_bits1(gb); int count = get_bits(gb, 8); if (count == 255) count += get_bits(gb, 8); if (byte_align) align_get_bits(gb); if (get_bits_left(gb) < 8 * count) { av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err); return AVERROR_INVALIDDATA; } skip_bits_long(gb, 8 * count); return 0; } static int decode_prediction(AACDecContext *ac, IndividualChannelStream *ics, GetBitContext *gb) { int sfb; if (get_bits1(gb)) { ics->predictor_reset_group = get_bits(gb, 5); if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) { av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n"); return AVERROR_INVALIDDATA; } } for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) { ics->prediction_used[sfb] = get_bits1(gb); } return 0; } /** * Decode Long Term Prediction data; reference: table 4.xx. */ static void decode_ltp(AACDecContext *ac, LongTermPrediction *ltp, GetBitContext *gb, uint8_t max_sfb) { int sfb; ltp->lag = get_bits(gb, 11); if (CONFIG_AAC_FIXED_DECODER && ac->is_fixed) ltp->coef_fixed = Q30(ff_ltp_coef[get_bits(gb, 3)]); else if (CONFIG_AAC_DECODER) ltp->coef = ff_ltp_coef[get_bits(gb, 3)]; for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++) ltp->used[sfb] = get_bits1(gb); } /** * Decode Individual Channel Stream info; reference: table 4.6. */ static int decode_ics_info(AACDecContext *ac, IndividualChannelStream *ics, GetBitContext *gb) { const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac; const int aot = m4ac->object_type; const int sampling_index = m4ac->sampling_index; int ret_fail = AVERROR_INVALIDDATA; if (aot != AOT_ER_AAC_ELD) { if (get_bits1(gb)) { av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n"); if (ac->avctx->err_recognition & AV_EF_BITSTREAM) return AVERROR_INVALIDDATA; } ics->window_sequence[1] = ics->window_sequence[0]; ics->window_sequence[0] = get_bits(gb, 2); if (aot == AOT_ER_AAC_LD && ics->window_sequence[0] != ONLY_LONG_SEQUENCE) { av_log(ac->avctx, AV_LOG_ERROR, "AAC LD is only defined for ONLY_LONG_SEQUENCE but " "window sequence %d found.\n", ics->window_sequence[0]); ics->window_sequence[0] = ONLY_LONG_SEQUENCE; return AVERROR_INVALIDDATA; } ics->use_kb_window[1] = ics->use_kb_window[0]; ics->use_kb_window[0] = get_bits1(gb); } ics->prev_num_window_groups = FFMAX(ics->num_window_groups, 1); 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; if (m4ac->frame_length_short) { ics->swb_offset = ff_swb_offset_120[sampling_index]; ics->num_swb = ff_aac_num_swb_120[sampling_index]; } else { ics->swb_offset = ff_swb_offset_128[sampling_index]; ics->num_swb = ff_aac_num_swb_128[sampling_index]; } ics->tns_max_bands = ff_tns_max_bands_128[sampling_index]; ics->predictor_present = 0; } else { ics->max_sfb = get_bits(gb, 6); ics->num_windows = 1; if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) { if (m4ac->frame_length_short) { ics->swb_offset = ff_swb_offset_480[sampling_index]; ics->num_swb = ff_aac_num_swb_480[sampling_index]; ics->tns_max_bands = ff_tns_max_bands_480[sampling_index]; } else { ics->swb_offset = ff_swb_offset_512[sampling_index]; ics->num_swb = ff_aac_num_swb_512[sampling_index]; ics->tns_max_bands = ff_tns_max_bands_512[sampling_index]; } if (!ics->num_swb || !ics->swb_offset) { ret_fail = AVERROR_BUG; goto fail; } } else { if (m4ac->frame_length_short) { ics->num_swb = ff_aac_num_swb_960[sampling_index]; ics->swb_offset = ff_swb_offset_960[sampling_index]; } else { ics->num_swb = ff_aac_num_swb_1024[sampling_index]; ics->swb_offset = ff_swb_offset_1024[sampling_index]; } ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index]; } if (aot != AOT_ER_AAC_ELD) { ics->predictor_present = get_bits1(gb); ics->predictor_reset_group = 0; } if (ics->predictor_present) { if (aot == AOT_AAC_MAIN) { if (decode_prediction(ac, ics, gb)) { goto fail; } } else if (aot == AOT_AAC_LC || aot == AOT_ER_AAC_LC) { av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n"); goto fail; } else { if (aot == AOT_ER_AAC_LD) { av_log(ac->avctx, AV_LOG_ERROR, "LTP in ER AAC LD not yet implemented.\n"); ret_fail = AVERROR_PATCHWELCOME; goto fail; } if ((ics->ltp.present = get_bits(gb, 1))) decode_ltp(ac, &ics->ltp, gb, ics->max_sfb); } } } if (ics->max_sfb > ics->num_swb) { av_log(ac->avctx, AV_LOG_ERROR, "Number of scalefactor bands in group (%d) " "exceeds limit (%d).\n", ics->max_sfb, ics->num_swb); goto fail; } return 0; fail: ics->max_sfb = 0; return ret_fail; } /** * 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(AACDecContext *ac, SingleChannelElement *sce, GetBitContext *gb) { IndividualChannelStream *ics = &sce->ics; const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; for (int g = 0; g < ics->num_window_groups; g++) { int k = 0; while (k < ics->max_sfb) { uint8_t sect_end = k; int sect_len_incr; int sect_band_type = get_bits(gb, 4); if (sect_band_type == 12) { av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n"); return AVERROR_INVALIDDATA; } do { sect_len_incr = get_bits(gb, bits); sect_end += sect_len_incr; if (get_bits_left(gb) < 0) { av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err); return AVERROR_INVALIDDATA; } if (sect_end > ics->max_sfb) { av_log(ac->avctx, AV_LOG_ERROR, "Number of bands (%d) exceeds limit (%d).\n", sect_end, ics->max_sfb); return AVERROR_INVALIDDATA; } } while (sect_len_incr == (1 << bits) - 1); for (; k < sect_end; k++) sce->band_type[g*ics->max_sfb + k] = sect_band_type; } } 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(AACDecContext *ac, SingleChannelElement *sce, GetBitContext *gb, unsigned int global_gain) { IndividualChannelStream *ics = &sce->ics; int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 }; int clipped_offset; int noise_flag = 1; for (int g = 0; g < ics->num_window_groups; g++) { for (int sfb = 0; sfb < ics->max_sfb; sfb++) { switch (sce->band_type[g*ics->max_sfb + sfb]) { case ZERO_BT: sce->sfo[g*ics->max_sfb + sfb] = 0; break; case INTENSITY_BT: /* fallthrough */ case INTENSITY_BT2: offset[2] += get_vlc2(gb, ff_vlc_scalefactors, 7, 3) - SCALE_DIFF_ZERO; clipped_offset = av_clip(offset[2], -155, 100); if (offset[2] != clipped_offset) { avpriv_request_sample(ac->avctx, "If you heard an audible artifact, there may be a bug in the decoder. " "Clipped intensity stereo position (%d -> %d)", offset[2], clipped_offset); } sce->sfo[g*ics->max_sfb + sfb] = clipped_offset - 100; break; case NOISE_BT: if (noise_flag-- > 0) offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE; else offset[1] += get_vlc2(gb, ff_vlc_scalefactors, 7, 3) - SCALE_DIFF_ZERO; clipped_offset = av_clip(offset[1], -100, 155); if (offset[1] != clipped_offset) { avpriv_request_sample(ac->avctx, "If you heard an audible artifact, there may be a bug in the decoder. " "Clipped noise gain (%d -> %d)", offset[1], clipped_offset); } sce->sfo[g*ics->max_sfb + sfb] = clipped_offset; break; default: offset[0] += get_vlc2(gb, ff_vlc_scalefactors, 7, 3) - SCALE_DIFF_ZERO; if (offset[0] > 255U) { av_log(ac->avctx, AV_LOG_ERROR, "Scalefactor (%d) out of range.\n", offset[0]); return AVERROR_INVALIDDATA; } sce->sfo[g*ics->max_sfb + sfb] = offset[0] - 100; break; } } } 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] >= swb_offset[num_swb]) 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] >= swb_offset[num_swb]) 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 */ int ff_aac_decode_tns(AACDecContext *ac, TemporalNoiseShaping *tns, GetBitContext *gb, const IndividualChannelStream *ics) { int tns_max_order = INT32_MAX; const int is_usac = ac->oc[1].m4ac.object_type == AOT_USAC; int w, filt, i, coef_len, coef_res, coef_compress; const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; /* USAC doesn't seem to have a limit */ if (!is_usac) tns_max_order = is8 ? 7 : ac->oc[1].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 (is_usac) tns->order[w][filt] = get_bits(gb, 4 - is8); else tns->order[w][filt] = get_bits(gb, 5 - (2 * is8)); if (tns->order[w][filt] > tns_max_order) { av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n", tns->order[w][filt], tns_max_order); tns->order[w][filt] = 0; return AVERROR_INVALIDDATA; } 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++) { if (CONFIG_AAC_FIXED_DECODER && ac->is_fixed) tns->coef_fixed[w][filt][i] = Q31(ff_tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]); else if (CONFIG_AAC_DECODER) tns->coef[w][filt][i] = ff_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; int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; cpe->max_sfb_ste = cpe->ch[0].ics.max_sfb; if (ms_present == 1) { for (idx = 0; idx < max_idx; idx++) cpe->ms_mask[idx] = get_bits1(gb); } else if (ms_present == 2) { memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0])); } } static void decode_gain_control(SingleChannelElement * sce, GetBitContext * gb) { // wd_num, wd_test, aloc_size static const uint8_t gain_mode[4][3] = { {1, 0, 5}, // ONLY_LONG_SEQUENCE = 0, {2, 1, 2}, // LONG_START_SEQUENCE, {8, 0, 2}, // EIGHT_SHORT_SEQUENCE, {2, 1, 5}, // LONG_STOP_SEQUENCE }; const int mode = sce->ics.window_sequence[0]; uint8_t bd, wd, ad; // FIXME: Store the gain control data on |sce| and do something with it. uint8_t max_band = get_bits(gb, 2); for (bd = 0; bd < max_band; bd++) { for (wd = 0; wd < gain_mode[mode][0]; wd++) { uint8_t adjust_num = get_bits(gb, 3); for (ad = 0; ad < adjust_num; ad++) { skip_bits(gb, 4 + ((wd == 0 && gain_mode[mode][1]) ? 4 : gain_mode[mode][2])); } } } } /** * 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 */ int ff_aac_decode_ics(AACDecContext *ac, SingleChannelElement *sce, GetBitContext *gb, int common_window, int scale_flag) { Pulse pulse; TemporalNoiseShaping *tns = &sce->tns; IndividualChannelStream *ics = &sce->ics; int global_gain, eld_syntax, er_syntax, pulse_present = 0; int ret; eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD; er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC || ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP || ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD || ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD; /* 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) { ret = decode_ics_info(ac, ics, gb); if (ret < 0) goto fail; } if ((ret = decode_band_types(ac, sce, gb)) < 0) goto fail; if ((ret = decode_scalefactors(ac, sce, gb, global_gain)) < 0) goto fail; ac->dsp.dequant_scalefactors(sce); pulse_present = 0; if (!scale_flag) { if (!eld_syntax && (pulse_present = get_bits1(gb))) { if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); ret = AVERROR_INVALIDDATA; goto fail; } if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) { av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n"); ret = AVERROR_INVALIDDATA; goto fail; } } tns->present = get_bits1(gb); if (tns->present && !er_syntax) { ret = ff_aac_decode_tns(ac, tns, gb, ics); if (ret < 0) goto fail; } if (!eld_syntax && get_bits1(gb)) { decode_gain_control(sce, gb); if (!ac->warned_gain_control) { avpriv_report_missing_feature(ac->avctx, "Gain control"); ac->warned_gain_control = 1; } } // I see no textual basis in the spec for this occurring after SSR gain // control, but this is what both reference and real implmentations do if (tns->present && er_syntax) { ret = ff_aac_decode_tns(ac, tns, gb, ics); if (ret < 0) goto fail; } } ret = ac->proc.decode_spectrum_and_dequant(ac, gb, pulse_present ? &pulse : NULL, sce); if (ret < 0) goto fail; if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window) ac->dsp.apply_prediction(ac, sce); return 0; fail: tns->present = 0; return ret; } /** * Decode a channel_pair_element; reference: table 4.4. * * @return Returns error status. 0 - OK, !0 - error */ static int decode_cpe(AACDecContext *ac, GetBitContext *gb, ChannelElement *cpe) { int i, ret, common_window, ms_present = 0; int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD; common_window = eld_syntax || get_bits1(gb); if (common_window) { if (decode_ics_info(ac, &cpe->ch[0].ics, gb)) return AVERROR_INVALIDDATA; 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; if (cpe->ch[1].ics.predictor_present && (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN)) if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1))) decode_ltp(ac, &cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb); ms_present = get_bits(gb, 2); if (ms_present == 3) { av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); return AVERROR_INVALIDDATA; } else if (ms_present) decode_mid_side_stereo(cpe, gb, ms_present); } if ((ret = ff_aac_decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) return ret; if ((ret = ff_aac_decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) return ret; if (common_window) { if (ms_present) ac->dsp.apply_mid_side_stereo(ac, cpe); if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) { ac->dsp.apply_prediction(ac, &cpe->ch[0]); ac->dsp.apply_prediction(ac, &cpe->ch[1]); } } ac->dsp.apply_intensity_stereo(ac, cpe, ms_present); return 0; } /** * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53. * * @return Returns number of bytes consumed. */ static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext *gb) { int i; int num_excl_chan = 0; do { for (i = 0; i < 7; i++) che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); return num_excl_chan / 7; } /** * Decode dynamic range information; reference: table 4.52. * * @return Returns number of bytes consumed. */ static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext *gb) { 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; } static int decode_fill(AACDecContext *ac, GetBitContext *gb, int len) { uint8_t buf[256]; int i, major, minor; if (len < 13+7*8) goto unknown; get_bits(gb, 13); len -= 13; for(i=0; i+1=8; i++, len-=8) buf[i] = get_bits(gb, 8); buf[i] = 0; if (ac->avctx->debug & FF_DEBUG_PICT_INFO) av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf); if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){ ac->avctx->internal->skip_samples = 1024; } unknown: skip_bits_long(gb, len); return 0; } /** * 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(AACDecContext *ac, GetBitContext *gb, int cnt, ChannelElement *che, enum RawDataBlockType elem_type) { int crc_flag = 0; int res = cnt; int type = get_bits(gb, 4); if (ac->avctx->debug & FF_DEBUG_STARTCODE) av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt); switch (type) { // extension type case EXT_SBR_DATA_CRC: crc_flag++; case EXT_SBR_DATA: if (!che) { av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n"); return res; } else if (ac->oc[1].m4ac.frame_length_short) { if (!ac->warned_960_sbr) avpriv_report_missing_feature(ac->avctx, "SBR with 960 frame length"); ac->warned_960_sbr = 1; skip_bits_long(gb, 8 * cnt - 4); return res; } else if (!ac->oc[1].m4ac.sbr) { av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n"); skip_bits_long(gb, 8 * cnt - 4); return res; } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) { av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n"); skip_bits_long(gb, 8 * cnt - 4); return res; } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->ch_layout.nb_channels == 1) { ac->oc[1].m4ac.sbr = 1; ac->oc[1].m4ac.ps = 1; ac->avctx->profile = AV_PROFILE_AAC_HE_V2; ff_aac_output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags, ac->oc[1].status, 1); } else { ac->oc[1].m4ac.sbr = 1; ac->avctx->profile = AV_PROFILE_AAC_HE; } ac->proc.sbr_decode_extension(ac, che, gb, crc_flag, cnt, elem_type); if (ac->oc[1].m4ac.ps == 1 && !ac->warned_he_aac_mono) { av_log(ac->avctx, AV_LOG_VERBOSE, "Treating HE-AAC mono as stereo.\n"); ac->warned_he_aac_mono = 1; } break; case EXT_DYNAMIC_RANGE: res = decode_dynamic_range(&ac->che_drc, gb); break; case EXT_FILL: decode_fill(ac, gb, 8 * cnt - 4); break; case EXT_FILL_DATA: case EXT_DATA_ELEMENT: default: skip_bits_long(gb, 8 * cnt - 4); break; }; return res; } /** * channel coupling transformation interface * * @param apply_coupling_method pointer to (in)dependent coupling function */ static void apply_channel_coupling(AACDecContext *ac, ChannelElement *cc, enum RawDataBlockType type, int elem_id, enum CouplingPoint coupling_point, void (*apply_coupling_method)(AACDecContext *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 samples, applying all supported tools as appropriate. */ static void spectral_to_sample(AACDecContext *ac, int samples) { int i, type; void (*imdct_and_window)(AACDecContext *ac, SingleChannelElement *sce); switch (ac->oc[1].m4ac.object_type) { case AOT_ER_AAC_LD: imdct_and_window = ac->dsp.imdct_and_windowing_ld; break; case AOT_ER_AAC_ELD: imdct_and_window = ac->dsp.imdct_and_windowing_eld; break; default: if (ac->oc[1].m4ac.frame_length_short) imdct_and_window = ac->dsp.imdct_and_windowing_960; else imdct_and_window = ac->dsp.imdct_and_windowing; } for (type = 3; type >= 0; type--) { for (i = 0; i < MAX_ELEM_ID; i++) { ChannelElement *che = ac->che[type][i]; if (che && che->present) { if (type <= TYPE_CPE) apply_channel_coupling(ac, che, type, i, BEFORE_TNS, ac->dsp.apply_dependent_coupling); if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) { if (che->ch[0].ics.predictor_present) { if (che->ch[0].ics.ltp.present) ac->dsp.apply_ltp(ac, &che->ch[0]); if (che->ch[1].ics.ltp.present && type == TYPE_CPE) ac->dsp.apply_ltp(ac, &che->ch[1]); } } if (che->ch[0].tns.present) ac->dsp.apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); if (che->ch[1].tns.present) ac->dsp.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, ac->dsp.apply_dependent_coupling); if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) { imdct_and_window(ac, &che->ch[0]); if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) ac->dsp.update_ltp(ac, &che->ch[0]); if (type == TYPE_CPE) { imdct_and_window(ac, &che->ch[1]); if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) ac->dsp.update_ltp(ac, &che->ch[1]); } if (ac->oc[1].m4ac.sbr > 0) { ac->proc.sbr_apply(ac, che, type, che->ch[0].output, che->ch[1].output); } } if (type <= TYPE_CCE) apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, ac->dsp.apply_independent_coupling); ac->dsp.clip_output(ac, che, type, samples); che->present = 0; } else if (che) { av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i); } } } } static int parse_adts_frame_header(AACDecContext *ac, GetBitContext *gb) { int size; AACADTSHeaderInfo hdr_info; uint8_t layout_map[MAX_ELEM_ID*4][3]; int layout_map_tags, ret; size = ff_adts_header_parse(gb, &hdr_info); if (size > 0) { if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) { // This is 2 for "VLB " audio in NSV files. // See samples/nsv/vlb_audio. avpriv_report_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame"); ac->warned_num_aac_frames = 1; } push_output_configuration(ac); if (hdr_info.chan_config) { ac->oc[1].m4ac.chan_config = hdr_info.chan_config; if ((ret = ff_aac_set_default_channel_config(ac, ac->avctx, layout_map, &layout_map_tags, hdr_info.chan_config)) < 0) return ret; if ((ret = ff_aac_output_configure(ac, layout_map, layout_map_tags, FFMAX(ac->oc[1].status, OC_TRIAL_FRAME), 0)) < 0) return ret; } else { ac->oc[1].m4ac.chan_config = 0; /** * dual mono frames in Japanese DTV can have chan_config 0 * WITHOUT specifying PCE. * thus, set dual mono as default. */ if (ac->dmono_mode && ac->oc[0].status == OC_NONE) { layout_map_tags = 2; layout_map[0][0] = layout_map[1][0] = TYPE_SCE; layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT; layout_map[0][1] = 0; layout_map[1][1] = 1; if (ff_aac_output_configure(ac, layout_map, layout_map_tags, OC_TRIAL_FRAME, 0)) return -7; } } ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate; ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index; ac->oc[1].m4ac.object_type = hdr_info.object_type; ac->oc[1].m4ac.frame_length_short = 0; if (ac->oc[0].status != OC_LOCKED || ac->oc[0].m4ac.chan_config != hdr_info.chan_config || ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) { ac->oc[1].m4ac.sbr = -1; ac->oc[1].m4ac.ps = -1; } if (!hdr_info.crc_absent) skip_bits(gb, 16); } return size; } static int aac_decode_er_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, GetBitContext *gb) { AACDecContext *ac = avctx->priv_data; const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac; ChannelElement *che; int err, i; int samples = m4ac->frame_length_short ? 960 : 1024; int chan_config = m4ac->chan_config; int aot = m4ac->object_type; if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) samples >>= 1; ac->frame = frame; if ((err = frame_configure_elements(avctx)) < 0) return err; // The AV_PROFILE_AAC_* defines are all object_type - 1 // This may lead to an undefined profile being signaled ac->avctx->profile = aot - 1; ac->tags_mapped = 0; if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) { avpriv_request_sample(avctx, "Unknown ER channel configuration %d", chan_config); return AVERROR_INVALIDDATA; } for (i = 0; i < ff_tags_per_config[chan_config]; i++) { const int elem_type = ff_aac_channel_layout_map[chan_config-1][i][0]; const int elem_id = ff_aac_channel_layout_map[chan_config-1][i][1]; if (!(che=ff_aac_get_che(ac, elem_type, elem_id))) { av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id); return AVERROR_INVALIDDATA; } che->present = 1; if (aot != AOT_ER_AAC_ELD) skip_bits(gb, 4); switch (elem_type) { case TYPE_SCE: err = ff_aac_decode_ics(ac, &che->ch[0], gb, 0, 0); break; case TYPE_CPE: err = decode_cpe(ac, gb, che); break; case TYPE_LFE: err = ff_aac_decode_ics(ac, &che->ch[0], gb, 0, 0); break; } if (err < 0) return err; } spectral_to_sample(ac, samples); if (!ac->frame->data[0] && samples) { av_log(avctx, AV_LOG_ERROR, "no frame data found\n"); return AVERROR_INVALIDDATA; } ac->frame->nb_samples = samples; ac->frame->sample_rate = avctx->sample_rate; *got_frame_ptr = 1; skip_bits_long(gb, get_bits_left(gb)); return 0; } static int decode_frame_ga(AVCodecContext *avctx, AACDecContext *ac, GetBitContext *gb, int *got_frame_ptr) { int err; int is_dmono; int elem_id; enum RawDataBlockType elem_type, che_prev_type = TYPE_END; uint8_t che_presence[4][MAX_ELEM_ID] = {{0}}; ChannelElement *che = NULL, *che_prev = NULL; int samples = 0, multiplier, audio_found = 0, pce_found = 0, sce_count = 0; AVFrame *frame = ac->frame; int payload_alignment = get_bits_count(gb); // parse while ((elem_type = get_bits(gb, 3)) != TYPE_END) { elem_id = get_bits(gb, 4); if (avctx->debug & FF_DEBUG_STARTCODE) av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id); if (!avctx->ch_layout.nb_channels && elem_type != TYPE_PCE) return AVERROR_INVALIDDATA; if (elem_type < TYPE_DSE) { if (che_presence[elem_type][elem_id]) { int error = che_presence[elem_type][elem_id] > 1; av_log(ac->avctx, error ? AV_LOG_ERROR : AV_LOG_DEBUG, "channel element %d.%d duplicate\n", elem_type, elem_id); if (error) return AVERROR_INVALIDDATA; } che_presence[elem_type][elem_id]++; if (!(che=ff_aac_get_che(ac, elem_type, elem_id))) { av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id); return AVERROR_INVALIDDATA; } samples = ac->oc[1].m4ac.frame_length_short ? 960 : 1024; che->present = 1; } switch (elem_type) { case TYPE_SCE: err = ff_aac_decode_ics(ac, &che->ch[0], gb, 0, 0); audio_found = 1; sce_count++; break; case TYPE_CPE: err = decode_cpe(ac, gb, che); audio_found = 1; break; case TYPE_CCE: err = ac->proc.decode_cce(ac, gb, che); break; case TYPE_LFE: err = ff_aac_decode_ics(ac, &che->ch[0], gb, 0, 0); audio_found = 1; break; case TYPE_DSE: err = skip_data_stream_element(ac, gb); break; case TYPE_PCE: { uint8_t layout_map[MAX_ELEM_ID*4][3] = {{0}}; int tags; int pushed = push_output_configuration(ac); if (pce_found && !pushed) return AVERROR_INVALIDDATA; tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb, payload_alignment); if (tags < 0) { err = tags; break; } if (pce_found) { av_log(avctx, AV_LOG_ERROR, "Not evaluating a further program_config_element as this construct is dubious at best.\n"); pop_output_configuration(ac); } else { err = ff_aac_output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1); if (!err) ac->oc[1].m4ac.chan_config = 0; pce_found = 1; } break; } case TYPE_FIL: if (elem_id == 15) elem_id += get_bits(gb, 8) - 1; if (get_bits_left(gb) < 8 * elem_id) { av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err); return AVERROR_INVALIDDATA; } err = 0; while (elem_id > 0) { int ret = decode_extension_payload(ac, gb, elem_id, che_prev, che_prev_type); if (ret < 0) { err = ret; break; } elem_id -= ret; } break; default: err = AVERROR_BUG; /* should not happen, but keeps compiler happy */ break; } if (elem_type < TYPE_DSE) { che_prev = che; che_prev_type = elem_type; } if (err) return err; if (get_bits_left(gb) < 3) { av_log(avctx, AV_LOG_ERROR, overread_err); return AVERROR_INVALIDDATA; } } if (!avctx->ch_layout.nb_channels) return 0; multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0; samples <<= multiplier; spectral_to_sample(ac, samples); if (ac->oc[1].status && audio_found) { avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier; avctx->frame_size = samples; ac->oc[1].status = OC_LOCKED; } if (!ac->frame->data[0] && samples) { av_log(avctx, AV_LOG_ERROR, "no frame data found\n"); return AVERROR_INVALIDDATA; } if (samples) { ac->frame->nb_samples = samples; ac->frame->sample_rate = avctx->sample_rate; *got_frame_ptr = 1; } else { av_frame_unref(ac->frame); *got_frame_ptr = 0; } /* for dual-mono audio (SCE + SCE) */ is_dmono = ac->dmono_mode && sce_count == 2 && !av_channel_layout_compare(&ac->oc[1].ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO); if (is_dmono) { if (ac->dmono_mode == 1) frame->data[1] = frame->data[0]; else if (ac->dmono_mode == 2) frame->data[0] = frame->data[1]; } return 0; } static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, GetBitContext *gb, const AVPacket *avpkt) { int err; AACDecContext *ac = avctx->priv_data; ac->frame = frame; *got_frame_ptr = 0; if (show_bits(gb, 12) == 0xfff) { if ((err = parse_adts_frame_header(ac, gb)) < 0) { av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n"); goto fail; } if (ac->oc[1].m4ac.sampling_index > 12) { av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index); err = AVERROR_INVALIDDATA; goto fail; } } if ((err = frame_configure_elements(avctx)) < 0) goto fail; // The AV_PROFILE_AAC_* defines are all object_type - 1 // This may lead to an undefined profile being signaled ac->avctx->profile = ac->oc[1].m4ac.object_type - 1; ac->tags_mapped = 0; if (ac->oc[1].m4ac.object_type == AOT_USAC) { if (ac->is_fixed) { avpriv_report_missing_feature(ac->avctx, "AAC USAC fixed-point decoding"); return AVERROR_PATCHWELCOME; } #if CONFIG_AAC_DECODER err = ff_aac_usac_decode_frame(avctx, ac, gb, got_frame_ptr); if (err < 0) goto fail; #endif } else { err = decode_frame_ga(avctx, ac, gb, got_frame_ptr); if (err < 0) goto fail; } return err; fail: pop_output_configuration(ac); return err; } static int aac_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt) { AACDecContext *ac = avctx->priv_data; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; GetBitContext gb; int buf_consumed; int buf_offset; int err; size_t new_extradata_size; const uint8_t *new_extradata = av_packet_get_side_data(avpkt, AV_PKT_DATA_NEW_EXTRADATA, &new_extradata_size); size_t jp_dualmono_size; const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt, AV_PKT_DATA_JP_DUALMONO, &jp_dualmono_size); if (new_extradata) { /* discard previous configuration */ ac->oc[1].status = OC_NONE; err = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1], new_extradata, new_extradata_size * 8LL, 1); if (err < 0) { return err; } } ac->dmono_mode = 0; if (jp_dualmono && jp_dualmono_size > 0) ac->dmono_mode = 1 + *jp_dualmono; if (ac->force_dmono_mode >= 0) ac->dmono_mode = ac->force_dmono_mode; if (INT_MAX / 8 <= buf_size) return AVERROR_INVALIDDATA; if ((err = init_get_bits8(&gb, buf, buf_size)) < 0) return err; switch (ac->oc[1].m4ac.object_type) { case AOT_ER_AAC_LC: case AOT_ER_AAC_LTP: case AOT_ER_AAC_LD: case AOT_ER_AAC_ELD: err = aac_decode_er_frame(avctx, frame, got_frame_ptr, &gb); break; default: err = aac_decode_frame_int(avctx, frame, got_frame_ptr, &gb, avpkt); } if (err < 0) return err; buf_consumed = (get_bits_count(&gb) + 7) >> 3; for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++) if (buf[buf_offset]) break; return buf_size > buf_offset ? buf_consumed : buf_size; } #if CONFIG_AAC_LATM_DECODER #include "aacdec_latm.h" #endif #define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM #define OFF(field) offsetof(AACDecContext, field) static const AVOption options[] = { /** * AVOptions for Japanese DTV specific extensions (ADTS only) */ {"dual_mono_mode", "Select the channel to decode for dual mono", OFF(force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2, AACDEC_FLAGS, .unit = "dual_mono_mode"}, {"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, .unit = "dual_mono_mode"}, {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, .unit = "dual_mono_mode"}, {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, .unit = "dual_mono_mode"}, {"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, .unit = "dual_mono_mode"}, { "channel_order", "Order in which the channels are to be exported", OFF(output_channel_order), AV_OPT_TYPE_INT, { .i64 = CHANNEL_ORDER_DEFAULT }, 0, 1, AACDEC_FLAGS, .unit = "channel_order" }, { "default", "normal libavcodec channel order", 0, AV_OPT_TYPE_CONST, { .i64 = CHANNEL_ORDER_DEFAULT }, .flags = AACDEC_FLAGS, .unit = "channel_order" }, { "coded", "order in which the channels are coded in the bitstream", 0, AV_OPT_TYPE_CONST, { .i64 = CHANNEL_ORDER_CODED }, .flags = AACDEC_FLAGS, .unit = "channel_order" }, {NULL}, }; static const AVClass decoder_class = { .class_name = "AAC decoder", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, }; #if CONFIG_AAC_DECODER const FFCodec ff_aac_decoder = { .p.name = "aac", CODEC_LONG_NAME("AAC (Advanced Audio Coding)"), .p.type = AVMEDIA_TYPE_AUDIO, .p.id = AV_CODEC_ID_AAC, .p.priv_class = &decoder_class, .priv_data_size = sizeof(AACDecContext), .init = ff_aac_decode_init_float, .close = decode_close, FF_CODEC_DECODE_CB(aac_decode_frame), .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1, .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, .p.ch_layouts = ff_aac_ch_layout, .flush = flush, .p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles), }; #endif #if CONFIG_AAC_FIXED_DECODER const FFCodec ff_aac_fixed_decoder = { .p.name = "aac_fixed", CODEC_LONG_NAME("AAC (Advanced Audio Coding)"), .p.type = AVMEDIA_TYPE_AUDIO, .p.id = AV_CODEC_ID_AAC, .p.priv_class = &decoder_class, .priv_data_size = sizeof(AACDecContext), .init = ff_aac_decode_init_fixed, .close = decode_close, FF_CODEC_DECODE_CB(aac_decode_frame), .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE }, .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1, .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, .p.ch_layouts = ff_aac_ch_layout, .p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles), .flush = flush, }; #endif