/* * AAC decoder * 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 */ /** * @file * AAC decoder * @author Oded Shimon ( ods15 ods15 dyndns org ) * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) * * AAC decoder fixed-point implementation * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com ) * @author Nedeljko Babic ( nedeljko.babic imgtec 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 * 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. */ #include "libavutil/thread.h" static VLC vlc_scalefactors; static VLC vlc_spectral[11]; static int output_configure(AACContext *ac, uint8_t layout_map[MAX_ELEM_ID*4][3], int tags, enum OCStatus oc_type, int get_new_frame); #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(AACContext *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]) { if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement)))) return AVERROR(ENOMEM); AAC_RENAME(ff_aac_sbr_ctx_init)(ac, &ac->che[type][id]->sbr); } 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]) AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][id]->sbr); av_freep(&ac->che[type][id]); } return 0; } static int frame_configure_elements(AVCodecContext *avctx) { AACContext *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].ret = che->ch[0].ret_buf; che->ch[1].ret = che->ch[1].ret_buf; } } } /* get output buffer */ av_frame_unref(ac->frame); if (!avctx->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->channels; ch++) { if (ac->output_element[ch]) ac->output_element[ch]->ret = (INTFLOAT *)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) { 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 }; 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 }; 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 ^= 1; } } if (sce_parity && ((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE)) return -1; *current = i; return num_pos_channels; } 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 } }; int num_front_channels, num_side_channels, num_back_channels; uint64_t layout; if (FF_ARRAY_ELEMS(e2c_vec) < tags) return 0; i = 0; num_front_channels = count_paired_channels(layout_map, tags, AAC_CHANNEL_FRONT, &i); if (num_front_channels < 0) return 0; num_side_channels = count_paired_channels(layout_map, tags, AAC_CHANNEL_SIDE, &i); if (num_side_channels < 0) return 0; num_back_channels = count_paired_channels(layout_map, tags, AAC_CHANNEL_BACK, &i); if (num_back_channels < 0) return 0; if (num_side_channels == 0 && num_back_channels >= 4) { num_side_channels = 2; num_back_channels -= 2; } i = 0; if (num_front_channels & 1) { e2c_vec[i] = (struct elem_to_channel) { .av_position = AV_CH_FRONT_CENTER, .syn_ele = TYPE_SCE, .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_FRONT }; i++; num_front_channels--; } if (num_front_channels >= 4) { i += assign_pair(e2c_vec, layout_map, i, AV_CH_FRONT_LEFT_OF_CENTER, AV_CH_FRONT_RIGHT_OF_CENTER, AAC_CHANNEL_FRONT); num_front_channels -= 2; } if (num_front_channels >= 2) { i += assign_pair(e2c_vec, layout_map, i, AV_CH_FRONT_LEFT, AV_CH_FRONT_RIGHT, AAC_CHANNEL_FRONT); num_front_channels -= 2; } while (num_front_channels >= 2) { i += assign_pair(e2c_vec, layout_map, i, UINT64_MAX, UINT64_MAX, AAC_CHANNEL_FRONT); num_front_channels -= 2; } if (num_side_channels >= 2) { i += assign_pair(e2c_vec, layout_map, i, AV_CH_SIDE_LEFT, AV_CH_SIDE_RIGHT, AAC_CHANNEL_FRONT); num_side_channels -= 2; } while (num_side_channels >= 2) { i += assign_pair(e2c_vec, layout_map, i, UINT64_MAX, UINT64_MAX, AAC_CHANNEL_SIDE); num_side_channels -= 2; } while (num_back_channels >= 4) { i += assign_pair(e2c_vec, layout_map, i, UINT64_MAX, UINT64_MAX, AAC_CHANNEL_BACK); num_back_channels -= 2; } if (num_back_channels >= 2) { i += assign_pair(e2c_vec, layout_map, i, AV_CH_BACK_LEFT, AV_CH_BACK_RIGHT, AAC_CHANNEL_BACK); num_back_channels -= 2; } if (num_back_channels) { e2c_vec[i] = (struct elem_to_channel) { .av_position = AV_CH_BACK_CENTER, .syn_ele = TYPE_SCE, .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_BACK }; i++; num_back_channels--; } if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) { e2c_vec[i] = (struct elem_to_channel) { .av_position = AV_CH_LOW_FREQUENCY, .syn_ele = TYPE_LFE, .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_LFE }; i++; } while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) { e2c_vec[i] = (struct elem_to_channel) { .av_position = UINT64_MAX, .syn_ele = TYPE_LFE, .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_LFE }; i++; } // Must choose a stable sort total_non_cc_elements = n = i; 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); layout = 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; if (e2c_vec[i].av_position != UINT64_MAX) { layout |= e2c_vec[i].av_position; } } return layout; } /** * Save current output configuration if and only if it has been locked. */ static void push_output_configuration(AACContext *ac) { if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) { ac->oc[0] = ac->oc[1]; } ac->oc[1].status = OC_NONE; } /** * Restore the previous output configuration if and only if the current * configuration is unlocked. */ static void pop_output_configuration(AACContext *ac) { if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) { ac->oc[1] = ac->oc[0]; ac->avctx->channels = ac->oc[1].channels; ac->avctx->channel_layout = ac->oc[1].channel_layout; 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 */ static int output_configure(AACContext *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]++; } // Try to sniff a reasonable channel order, otherwise output the // channels in the order the PCE declared them. if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE) 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; } } if (layout) avctx->channel_layout = layout; ac->oc[1].channel_layout = layout; avctx->channels = ac->oc[1].channels = channels; ac->oc[1].status = oc_type; if (get_new_frame) { if ((ret = frame_configure_elements(ac->avctx)) < 0) return ret; } return 0; } static void flush(AVCodecContext *avctx) { AACContext *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)); } } } } } /** * Set up channel positions based on a default channel configuration * as specified in table 1.17. * * @return Returns error status. 0 - OK, !0 - error */ static int set_default_channel_config(AVCodecContext *avctx, uint8_t (*layout_map)[3], int *tags, int channel_config) { if (channel_config < 1 || (channel_config > 7 && channel_config < 11) || channel_config > 12) { av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", channel_config); return AVERROR_INVALIDDATA; } *tags = tags_per_config[channel_config]; memcpy(layout_map, 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) { 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); layout_map[2][2] = AAC_CHANNEL_SIDE; } return 0; } static ChannelElement *get_che(AACContext *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 (set_default_channel_config(ac->avctx, layout_map, &layout_map_tags, 2) < 0) return NULL; if (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"); if (set_default_channel_config(ac->avctx, layout_map, &layout_map_tags, 1) < 0) return NULL; if (output_configure(ac, layout_map, layout_map_tags, OC_TRIAL_FRAME, 1) < 0) return NULL; ac->oc[1].m4ac.chan_config = 1; 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 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 == 2 && ac->oc[1].m4ac.chan_config == 11 && 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 == 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 == 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->oc[1].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; } } /** * 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++; } } /** * 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 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) < 4 * (num_front + num_side + num_back + 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; align_get_bits(gb); /* comment field, first byte is length */ comment_len = get_bits(gb, 8) * 8; if (get_bits_left(gb) < comment_len) { av_log(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 AACContext, 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(AACContext *ac, AVCodecContext *avctx, GetBitContext *gb, 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; if (get_bits1(gb)) { // frameLengthFlag avpriv_request_sample(avctx, "960/120 MDCT window"); return AVERROR_PATCHWELCOME; } m4ac->frame_length_short = 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); if (tags < 0) return tags; } else { if ((ret = set_default_channel_config(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 = 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(AACContext *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; #if USE_FIXED if (get_bits1(gb)) { // frameLengthFlag avpriv_request_sample(avctx, "960/120 MDCT window"); return AVERROR_PATCHWELCOME; } #else m4ac->frame_length_short = get_bits1(gb); #endif 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 = set_default_channel_config(avctx, layout_map, &tags, channel_config))) return ret; if (ac && (ret = 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 AACContext, may be null * @param avctx pointer to AVCCodecContext, used for logging * @param m4ac pointer to MPEG4AudioConfig, used for parsing * @param data pointer to buffer holding an audio specific config * @param bit_size size of audio specific config or data in bits * @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(AACContext *ac, AVCodecContext *avctx, MPEG4AudioConfig *m4ac, const uint8_t *data, int64_t bit_size, int sync_extension) { GetBitContext gb; int i, ret; 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; if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size, sync_extension)) < 0) return AVERROR_INVALIDDATA; if (m4ac->sampling_index > 12) { av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index); 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); return AVERROR_INVALIDDATA; } skip_bits_long(&gb, i); switch (m4ac->object_type) { case AOT_AAC_MAIN: case AOT_AAC_LC: case AOT_AAC_LTP: case AOT_ER_AAC_LC: case AOT_ER_AAC_LD: if ((ret = decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config)) < 0) return ret; break; case AOT_ER_AAC_ELD: if ((ret = decode_eld_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config)) < 0) return ret; break; 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); } /** * 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(unsigned previous_val) { union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 }; return v.s; } static void reset_all_predictors(PredictorState *ps) { int i; for (i = 0; i < MAX_PREDICTORS; i++) reset_predict_state(&ps[i]); } 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 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]); } #define AAC_INIT_VLC_STATIC(num, size) \ INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \ ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]), \ sizeof(ff_aac_spectral_bits[num][0]), \ ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \ sizeof(ff_aac_spectral_codes[num][0]), \ size); static void aacdec_init(AACContext *ac); static av_cold void aac_static_table_init(void) { 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); AAC_RENAME(ff_aac_sbr_init)(); ff_aac_tableinit(); 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); // window initialization AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_long_1024), 4.0, 1024); AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_short_128), 6.0, 128); AAC_RENAME(ff_init_ff_sine_windows)(10); AAC_RENAME(ff_init_ff_sine_windows)( 9); AAC_RENAME(ff_init_ff_sine_windows)( 7); AAC_RENAME(cbrt_tableinit)(); } static AVOnce aac_init = AV_ONCE_INIT; static av_cold int aac_decode_init(AVCodecContext *avctx) { AACContext *ac = avctx->priv_data; int ret; ret = ff_thread_once(&aac_init, &aac_static_table_init); if (ret != 0) return AVERROR_UNKNOWN; ac->avctx = avctx; ac->oc[1].m4ac.sample_rate = avctx->sample_rate; aacdec_init(ac); #if USE_FIXED avctx->sample_fmt = AV_SAMPLE_FMT_S32P; #else avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; #endif /* USE_FIXED */ if (avctx->extradata_size > 0) { if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac, 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->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->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 = set_default_channel_config(avctx, layout_map, &layout_map_tags, ac->oc[1].m4ac.chan_config); if (!ret) 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->channels > MAX_CHANNELS) { av_log(avctx, AV_LOG_ERROR, "Too many channels\n"); return AVERROR_INVALIDDATA; } #if USE_FIXED ac->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & AV_CODEC_FLAG_BITEXACT); #else ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT); #endif /* USE_FIXED */ if (!ac->fdsp) { return AVERROR(ENOMEM); } ac->random_state = 0x1f2e3d4c; AAC_RENAME_32(ff_mdct_init)(&ac->mdct, 11, 1, 1.0 / RANGE15(1024.0)); AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ld, 10, 1, 1.0 / RANGE15(512.0)); AAC_RENAME_32(ff_mdct_init)(&ac->mdct_small, 8, 1, 1.0 / RANGE15(128.0)); AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ltp, 11, 0, RANGE15(-2.0)); #if !USE_FIXED ret = ff_imdct15_init(&ac->mdct480, 5); if (ret < 0) return ret; #endif return 0; } /** * Skip data_stream_element; reference: table 4.10. */ static int skip_data_stream_element(AACContext *ac, GetBitContext *gb) { int byte_align = get_bits1(gb); int count = get_bits(gb, 8); if (count == 255) count += get_bits(gb, 8); if (byte_align) align_get_bits(gb); if (get_bits_left(gb) < 8 * count) { av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err); return AVERROR_INVALIDDATA; } skip_bits_long(gb, 8 * count); return 0; } static int decode_prediction(AACContext *ac, IndividualChannelStream *ics, GetBitContext *gb) { int sfb; if (get_bits1(gb)) { ics->predictor_reset_group = get_bits(gb, 5); if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) { av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n"); return 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(LongTermPrediction *ltp, GetBitContext *gb, uint8_t max_sfb) { int sfb; ltp->lag = get_bits(gb, 11); ltp->coef = 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(AACContext *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; 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->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[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) return AVERROR_BUG; } else { ics->swb_offset = ff_swb_offset_1024[sampling_index]; ics->num_swb = ff_aac_num_swb_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"); return AVERROR_PATCHWELCOME; } if ((ics->ltp.present = get_bits(gb, 1))) decode_ltp(&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 AVERROR_INVALIDDATA; } /** * Decode band types (section_data payload); reference: table 4.46. * * @param band_type array of the used band type * @param band_type_run_end array of the last scalefactor band of a band type run * * @return Returns error status. 0 - OK, !0 - error */ static int decode_band_types(AACContext *ac, enum BandType band_type[120], int band_type_run_end[120], GetBitContext *gb, IndividualChannelStream *ics) { int g, idx = 0; const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; for (g = 0; g < ics->num_window_groups; g++) { int k = 0; while (k < ics->max_sfb) { uint8_t sect_end = k; int sect_len_incr; int sect_band_type = get_bits(gb, 4); if (sect_band_type == 12) { av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n"); return 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++) { 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, INTFLOAT sf[120], GetBitContext *gb, unsigned int global_gain, IndividualChannelStream *ics, enum BandType band_type[120], int band_type_run_end[120]) { int g, i, idx = 0; int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 }; int clipped_offset; int noise_flag = 1; 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] = FIXR(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) - 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); } #if USE_FIXED sf[idx] = 100 - clipped_offset; #else sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO]; #endif /* USE_FIXED */ } } else if (band_type[idx] == NOISE_BT) { for (; i < run_end; i++, idx++) { if (noise_flag-- > 0) offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE; else offset[1] += get_vlc2(gb, vlc_scalefactors.table, 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); } #if USE_FIXED sf[idx] = -(100 + clipped_offset); #else sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO]; #endif /* USE_FIXED */ } } else { for (; i < run_end; i++, idx++) { offset[0] += get_vlc2(gb, vlc_scalefactors.table, 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; } #if USE_FIXED sf[idx] = -offset[0]; #else sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO]; #endif /* USE_FIXED */ } } } } 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 */ 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->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 ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) { av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n", tns->order[w][filt], tns_max_order); tns->order[w][filt] = 0; return 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++) 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; int max_idx = cpe->ch[0].ics.num_window_groups * 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])); } } /** * 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, INTFLOAT coef[1024], GetBitContext *gb, const INTFLOAT 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; INTFLOAT *coef_base = coef; for (g = 0; g < ics->num_windows; g++) memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(INTFLOAT) * (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; INTFLOAT *cfo = coef + offsets[i]; int off_len = offsets[i + 1] - offsets[i]; int group; if (cbt_m1 >= INTENSITY_BT2 - 1) { for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { memset(cfo, 0, off_len * sizeof(*cfo)); } } else if (cbt_m1 == NOISE_BT - 1) { for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { #if !USE_FIXED float scale; #endif /* !USE_FIXED */ INTFLOAT band_energy; for (k = 0; k < off_len; k++) { ac->random_state = lcg_random(ac->random_state); #if USE_FIXED cfo[k] = ac->random_state >> 3; #else cfo[k] = ac->random_state; #endif /* USE_FIXED */ } #if USE_FIXED band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len); band_energy = fixed_sqrt(band_energy, 31); noise_scale(cfo, sf[idx], band_energy, off_len); #else band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len); scale = sf[idx] / sqrtf(band_energy); ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len); #endif /* USE_FIXED */ } } else { #if !USE_FIXED const float *vq = ff_aac_codebook_vector_vals[cbt_m1]; #endif /* !USE_FIXED */ const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1]; VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table; OPEN_READER(re, gb); switch (cbt_m1 >> 1) { case 0: for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { INTFLOAT *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); cb_idx = cb_vector_idx[code]; #if USE_FIXED cf = DEC_SQUAD(cf, cb_idx); #else cf = VMUL4(cf, vq, cb_idx, sf + idx); #endif /* USE_FIXED */ } while (len -= 4); } break; case 1: for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { INTFLOAT *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); cb_idx = cb_vector_idx[code]; nnz = cb_idx >> 8 & 15; bits = nnz ? GET_CACHE(re, gb) : 0; LAST_SKIP_BITS(re, gb, nnz); #if USE_FIXED cf = DEC_UQUAD(cf, cb_idx, bits); #else cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx); #endif /* USE_FIXED */ } while (len -= 4); } break; case 2: for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { INTFLOAT *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); cb_idx = cb_vector_idx[code]; #if USE_FIXED cf = DEC_SPAIR(cf, cb_idx); #else cf = VMUL2(cf, vq, cb_idx, sf + idx); #endif /* USE_FIXED */ } while (len -= 2); } break; case 3: case 4: for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { INTFLOAT *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); cb_idx = cb_vector_idx[code]; nnz = cb_idx >> 8 & 15; sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0; LAST_SKIP_BITS(re, gb, nnz); #if USE_FIXED cf = DEC_UPAIR(cf, cb_idx, sign); #else cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx); #endif /* USE_FIXED */ } while (len -= 2); } break; default: for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { #if USE_FIXED int *icf = cfo; int v; #else float *cf = cfo; uint32_t *icf = (uint32_t *) cf; #endif /* USE_FIXED */ 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; } 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< 8) { av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); return AVERROR_INVALIDDATA; } SKIP_BITS(re, gb, b + 1); b += 4; n = (1 << b) + SHOW_UBITS(re, gb, b); LAST_SKIP_BITS(re, gb, b); #if USE_FIXED v = n; if (bits & 1U<<31) v = -v; *icf++ = v; #else *icf++ = cbrt_tab[n] | (bits & 1U<<31); #endif /* USE_FIXED */ bits <<= 1; } else { #if USE_FIXED v = cb_idx & 15; if (bits & 1U<<31) v = -v; *icf++ = v; #else unsigned v = ((const uint32_t*)vq)[cb_idx & 15]; *icf++ = (bits & 1U<<31) | v; #endif /* USE_FIXED */ bits <<= !!v; } cb_idx >>= 4; } } while (len -= 2); #if !USE_FIXED ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len); #endif /* !USE_FIXED */ } } CLOSE_READER(re, gb); } } coef += g_len << 7; } if (pulse_present) { idx = 0; for (i = 0; i < pulse->num_pulse; i++) { INTFLOAT co = coef_base[ pulse->pos[i] ]; while (offsets[idx + 1] <= pulse->pos[i]) idx++; if (band_type[idx] != NOISE_BT && sf[idx]) { INTFLOAT ico = -pulse->amp[i]; #if USE_FIXED if (co) { ico = co + (co > 0 ? -ico : ico); } coef_base[ pulse->pos[i] ] = ico; #else 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]; #endif /* USE_FIXED */ } } } #if USE_FIXED coef = coef_base; idx = 0; 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; int *cfo = coef + offsets[i]; int off_len = offsets[i + 1] - offsets[i]; int group; if (cbt_m1 < NOISE_BT - 1) { for (group = 0; group < (int)g_len; group++, cfo+=128) { ac->vector_pow43(cfo, off_len); ac->subband_scale(cfo, cfo, sf[idx], 34, off_len); } } } coef += g_len << 7; } #endif /* USE_FIXED */ return 0; } /** * 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->oc[1].m4ac.sampling_index]; sfb++) { for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) { predict(&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; INTFLOAT *out = sce->coeffs; 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) { if (decode_ics_info(ac, ics, gb) < 0) return AVERROR_INVALIDDATA; } if ((ret = decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics)) < 0) return ret; if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end)) < 0) return ret; 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"); return AVERROR_INVALIDDATA; } if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) { av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n"); return AVERROR_INVALIDDATA; } } tns->present = get_bits1(gb); if (tns->present && !er_syntax) if (decode_tns(ac, tns, gb, ics) < 0) return AVERROR_INVALIDDATA; if (!eld_syntax && get_bits1(gb)) { avpriv_request_sample(ac->avctx, "SSR"); return AVERROR_PATCHWELCOME; } // 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) if (decode_tns(ac, tns, gb, ics) < 0) return AVERROR_INVALIDDATA; } if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0) return AVERROR_INVALIDDATA; if (ac->oc[1].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; INTFLOAT *ch0 = cpe->ch[0].coeffs; INTFLOAT *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) { #if USE_FIXED for (group = 0; group < ics->group_len[g]; group++) { ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i], ch1 + group * 128 + offsets[i], offsets[i+1] - offsets[i]); #else for (group = 0; group < ics->group_len[g]; group++) { ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i], ch1 + group * 128 + offsets[i], offsets[i+1] - offsets[i]); #endif /* USE_FIXED */ } } } 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(AACContext *ac, ChannelElement *cpe, int ms_present) { const IndividualChannelStream *ics = &cpe->ch[1].ics; SingleChannelElement *sce1 = &cpe->ch[1]; INTFLOAT *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs; const uint16_t *offsets = ics->swb_offset; int g, group, i, idx = 0; int c; INTFLOAT 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++) #if USE_FIXED ac->subband_scale(coef1 + group * 128 + offsets[i], coef0 + group * 128 + offsets[i], scale, 23, offsets[i + 1] - offsets[i]); #else ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i], coef0 + group * 128 + offsets[i], scale, offsets[i + 1] - offsets[i]); #endif /* USE_FIXED */ } } 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. * * @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; 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(&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 = 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->oc[1].m4ac.object_type == AOT_AAC_MAIN) { apply_prediction(ac, &cpe->ch[0]); apply_prediction(ac, &cpe->ch[1]); } } apply_intensity_stereo(ac, cpe, ms_present); return 0; } static const float cce_scale[] = { 1.09050773266525765921, //2^(1/8) 1.18920711500272106672, //2^(1/4) M_SQRT2, 2, }; /** * Decode coupling_channel_element; reference: table 4.8. * * @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; INTFLOAT 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 = AAC_RENAME(cce_scale)[get_bits(gb, 2)]; 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; INTFLOAT gain_cache = FIXR10(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 = GET_GAIN(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 = GET_GAIN(scale, t) * s; } } coup->gain[c][idx] = gain_cache; } } } } } return 0; } /** * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53. * * @return Returns number of bytes consumed. */ static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext *gb) { int i; int num_excl_chan = 0; do { for (i = 0; i < 7; i++) che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); return num_excl_chan / 7; } /** * Decode dynamic range information; reference: table 4.52. * * @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(AACContext *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(AACContext *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.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->channels == 1) { ac->oc[1].m4ac.sbr = 1; ac->oc[1].m4ac.ps = 1; ac->avctx->profile = FF_PROFILE_AAC_HE_V2; 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 = FF_PROFILE_AAC_HE; } res = AAC_RENAME(ff_decode_sbr_extension)(ac, &che->sbr, gb, crc_flag, cnt, elem_type); break; case EXT_DYNAMIC_RANGE: res = decode_dynamic_range(&ac->che_drc, gb); 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; } /** * 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(INTFLOAT 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; INTFLOAT lpc[TNS_MAX_ORDER]; INTFLOAT tmp[TNS_MAX_ORDER+1]; 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 AAC_RENAME(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; if (decode) { // ar filter for (m = 0; m < size; m++, start += inc) for (i = 1; i <= FFMIN(m, order); i++) coef[start] -= AAC_MUL26(coef[start - i * inc], lpc[i - 1]); } else { // ma filter for (m = 0; m < size; m++, start += inc) { tmp[0] = coef[start]; for (i = 1; i <= FFMIN(m, order); i++) coef[start] += AAC_MUL26(tmp[i], lpc[i - 1]); for (i = order; i > 0; i--) tmp[i] = tmp[i - 1]; } } } } } /** * Apply windowing and MDCT to obtain the spectral * coefficient from the predicted sample by LTP. */ static void windowing_and_mdct_ltp(AACContext *ac, INTFLOAT *out, INTFLOAT *in, IndividualChannelStream *ics) { const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024); const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128); const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024); const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128); if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) { ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024); } else { memset(in, 0, 448 * sizeof(*in)); ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128); } if (ics->window_sequence[0] != LONG_START_SEQUENCE) { ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024); } else { ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128); memset(in + 1024 + 576, 0, 448 * sizeof(*in)); } ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in); } /** * Apply the long term prediction */ static void apply_ltp(AACContext *ac, SingleChannelElement *sce) { const LongTermPrediction *ltp = &sce->ics.ltp; const uint16_t *offsets = sce->ics.swb_offset; int i, sfb; if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { INTFLOAT *predTime = sce->ret; INTFLOAT *predFreq = ac->buf_mdct; int16_t num_samples = 2048; if (ltp->lag < 1024) num_samples = ltp->lag + 1024; for (i = 0; i < num_samples; i++) predTime[i] = AAC_MUL30(sce->ltp_state[i + 2048 - ltp->lag], ltp->coef); memset(&predTime[i], 0, (2048 - i) * sizeof(*predTime)); ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics); if (sce->tns.present) ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0); for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++) if (ltp->used[sfb]) for (i = offsets[sfb]; i < offsets[sfb + 1]; i++) sce->coeffs[i] += predFreq[i]; } } /** * Update the LTP buffer for next frame */ static void update_ltp(AACContext *ac, SingleChannelElement *sce) { IndividualChannelStream *ics = &sce->ics; INTFLOAT *saved = sce->saved; INTFLOAT *saved_ltp = sce->coeffs; const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024); const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128); int i; if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { memcpy(saved_ltp, saved, 512 * sizeof(*saved_ltp)); memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp)); ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64); for (i = 0; i < 64; i++) saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]); } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) { memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(*saved_ltp)); memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp)); ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64); for (i = 0; i < 64; i++) saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]); } else { // LONG_STOP or ONLY_LONG ac->fdsp->vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512); for (i = 0; i < 512; i++) saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], lwindow[511 - i]); } memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state)); memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state)); memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state)); } /** * Conduct IMDCT and windowing. */ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce) { IndividualChannelStream *ics = &sce->ics; INTFLOAT *in = sce->coeffs; INTFLOAT *out = sce->ret; INTFLOAT *saved = sce->saved; const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128); const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024); const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128); INTFLOAT *buf = ac->buf_mdct; INTFLOAT *temp = ac->temp; int i; // imdct if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { for (i = 0; i < 1024; i += 128) ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i); } else { ac->mdct.imdct_half(&ac->mdct, buf, in); #if USE_FIXED for (i=0; i<1024; i++) buf[i] = (buf[i] + 4) >> 3; #endif /* USE_FIXED */ } /* 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->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 512); } else { memcpy( out, saved, 448 * sizeof(*out)); if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64); ac->fdsp->vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64); ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64); ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64); ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64); memcpy( out + 448 + 4*128, temp, 64 * sizeof(*out)); } else { ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64); memcpy( out + 576, buf + 64, 448 * sizeof(*out)); } } // buffer update if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { memcpy( saved, temp + 64, 64 * sizeof(*saved)); ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64); ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64); ac->fdsp->vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64); memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved)); } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) { memcpy( saved, buf + 512, 448 * sizeof(*saved)); memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved)); } else { // LONG_STOP or ONLY_LONG memcpy( saved, buf + 512, 512 * sizeof(*saved)); } } static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce) { IndividualChannelStream *ics = &sce->ics; INTFLOAT *in = sce->coeffs; INTFLOAT *out = sce->ret; INTFLOAT *saved = sce->saved; INTFLOAT *buf = ac->buf_mdct; #if USE_FIXED int i; #endif /* USE_FIXED */ // imdct ac->mdct.imdct_half(&ac->mdct_ld, buf, in); #if USE_FIXED for (i = 0; i < 1024; i++) buf[i] = (buf[i] + 2) >> 2; #endif /* USE_FIXED */ // window overlapping if (ics->use_kb_window[1]) { // AAC LD uses a low overlap sine window instead of a KBD window memcpy(out, saved, 192 * sizeof(*out)); ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, AAC_RENAME(ff_sine_128), 64); memcpy( out + 320, buf + 64, 192 * sizeof(*out)); } else { ac->fdsp->vector_fmul_window(out, saved, buf, AAC_RENAME(ff_sine_512), 256); } // buffer update memcpy(saved, buf + 256, 256 * sizeof(*saved)); } static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce) { INTFLOAT *in = sce->coeffs; INTFLOAT *out = sce->ret; INTFLOAT *saved = sce->saved; INTFLOAT *buf = ac->buf_mdct; int i; const int n = ac->oc[1].m4ac.frame_length_short ? 480 : 512; const int n2 = n >> 1; const int n4 = n >> 2; const INTFLOAT *const window = n == 480 ? AAC_RENAME(ff_aac_eld_window_480) : AAC_RENAME(ff_aac_eld_window_512); // Inverse transform, mapped to the conventional IMDCT by // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V., // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks," // International Conference on Audio, Language and Image Processing, ICALIP 2008. // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950 for (i = 0; i < n2; i+=2) { INTFLOAT temp; temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp; temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp; } #if !USE_FIXED if (n == 480) ac->mdct480->imdct_half(ac->mdct480, buf, in, 1, -1.f/(16*1024*960)); else #endif ac->mdct.imdct_half(&ac->mdct_ld, buf, in); #if USE_FIXED for (i = 0; i < 1024; i++) buf[i] = (buf[i] + 1) >> 1; #endif /* USE_FIXED */ for (i = 0; i < n; i+=2) { buf[i] = -buf[i]; } // Like with the regular IMDCT at this point we still have the middle half // of a transform but with even symmetry on the left and odd symmetry on // the right // window overlapping // The spec says to use samples [0..511] but the reference decoder uses // samples [128..639]. for (i = n4; i < n2; i ++) { out[i - n4] = AAC_MUL31( buf[ n2 - 1 - i] , window[i - n4]) + AAC_MUL31( saved[ i + n2] , window[i + n - n4]) + AAC_MUL31(-saved[n + n2 - 1 - i] , window[i + 2*n - n4]) + AAC_MUL31(-saved[ 2*n + n2 + i] , window[i + 3*n - n4]); } for (i = 0; i < n2; i ++) { out[n4 + i] = AAC_MUL31( buf[ i] , window[i + n2 - n4]) + AAC_MUL31(-saved[ n - 1 - i] , window[i + n2 + n - n4]) + AAC_MUL31(-saved[ n + i] , window[i + n2 + 2*n - n4]) + AAC_MUL31( saved[2*n + n - 1 - i] , window[i + n2 + 3*n - n4]); } for (i = 0; i < n4; i ++) { out[n2 + n4 + i] = AAC_MUL31( buf[ i + n2] , window[i + n - n4]) + AAC_MUL31(-saved[n2 - 1 - i] , window[i + 2*n - n4]) + AAC_MUL31(-saved[n + n2 + i] , window[i + 3*n - n4]); } // buffer update memmove(saved + n, saved, 2 * n * sizeof(*saved)); memcpy( saved, buf, n * sizeof(*saved)); } /** * channel coupling transformation interface * * @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 samples, applying all supported tools as appropriate. */ static void spectral_to_sample(AACContext *ac, int samples) { int i, type; void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce); switch (ac->oc[1].m4ac.object_type) { case AOT_ER_AAC_LD: imdct_and_window = imdct_and_windowing_ld; break; case AOT_ER_AAC_ELD: imdct_and_window = imdct_and_windowing_eld; break; default: imdct_and_window = ac->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, AAC_RENAME(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->apply_ltp(ac, &che->ch[0]); if (che->ch[1].ics.ltp.present && type == TYPE_CPE) ac->apply_ltp(ac, &che->ch[1]); } } if (che->ch[0].tns.present) ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); if (che->ch[1].tns.present) ac->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, AAC_RENAME(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->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->update_ltp(ac, &che->ch[1]); } if (ac->oc[1].m4ac.sbr > 0) { AAC_RENAME(ff_sbr_apply)(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret); } } if (type <= TYPE_CCE) apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, AAC_RENAME(apply_independent_coupling)); #if USE_FIXED { int j; /* preparation for resampler */ for(j = 0; jch[0].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[0].ret[j]<<7)+0x8000; if(type == TYPE_CPE) che->ch[1].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[1].ret[j]<<7)+0x8000; } } #endif /* USE_FIXED */ 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(AACContext *ac, GetBitContext *gb) { int size; AACADTSHeaderInfo hdr_info; uint8_t layout_map[MAX_ELEM_ID*4][3]; int layout_map_tags, ret; size = avpriv_aac_parse_header(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 = set_default_channel_config(ac->avctx, layout_map, &layout_map_tags, hdr_info.chan_config)) < 0) return ret; if ((ret = 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 (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, void *data, int *got_frame_ptr, GetBitContext *gb) { AACContext *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 = data; if ((err = frame_configure_elements(avctx)) < 0) return err; // The FF_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 < tags_per_config[chan_config]; i++) { const int elem_type = aac_channel_layout_map[chan_config-1][i][0]; const int elem_id = aac_channel_layout_map[chan_config-1][i][1]; if (!(che=get_che(ac, elem_type, elem_id))) { av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id); return AVERROR_INVALIDDATA; } che->present = 1; if (aot != AOT_ER_AAC_ELD) skip_bits(gb, 4); 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_LFE: err = decode_ics(ac, &che->ch[0], gb, 0, 0); break; } if (err < 0) return err; } spectral_to_sample(ac, samples); 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 aac_decode_frame_int(AVCodecContext *avctx, void *data, int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt) { AACContext *ac = avctx->priv_data; ChannelElement *che = NULL, *che_prev = NULL; enum RawDataBlockType elem_type, elem_type_prev = TYPE_END; int err, elem_id; int samples = 0, multiplier, audio_found = 0, pce_found = 0; int is_dmono, sce_count = 0; ac->frame = data; 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 FF_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; // 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->channels && elem_type != TYPE_PCE) { err = AVERROR_INVALIDDATA; goto fail; } if (elem_type < TYPE_DSE) { if (!(che=get_che(ac, elem_type, elem_id))) { av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id); err = AVERROR_INVALIDDATA; goto fail; } samples = 1024; che->present = 1; } switch (elem_type) { case TYPE_SCE: err = 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 = decode_cce(ac, gb, che); break; case TYPE_LFE: err = 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]; int tags; push_output_configuration(ac); tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb); 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"); } else { err = 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); err = AVERROR_INVALIDDATA; goto fail; } while (elem_id > 0) elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev); err = 0; /* FIXME */ break; default: err = AVERROR_BUG; /* should not happen, but keeps compiler happy */ break; } che_prev = che; elem_type_prev = elem_type; if (err) goto fail; if (get_bits_left(gb) < 3) { av_log(avctx, AV_LOG_ERROR, overread_err); err = AVERROR_INVALIDDATA; goto fail; } } if (!avctx->channels) { *got_frame_ptr = 0; 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 (multiplier) { int side_size; const uint8_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size); if (side && side_size>=4) AV_WL32(side, 2*AV_RL32(side)); } if (!ac->frame->data[0] && samples) { av_log(avctx, AV_LOG_ERROR, "no frame data found\n"); err = AVERROR_INVALIDDATA; goto fail; } if (samples) { ac->frame->nb_samples = samples; ac->frame->sample_rate = avctx->sample_rate; } else av_frame_unref(ac->frame); *got_frame_ptr = !!samples; /* for dual-mono audio (SCE + SCE) */ is_dmono = ac->dmono_mode && sce_count == 2 && ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT); if (is_dmono) { if (ac->dmono_mode == 1) ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0]; else if (ac->dmono_mode == 2) ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1]; } return 0; fail: pop_output_configuration(ac); return err; } static int aac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { AACContext *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; int new_extradata_size; const uint8_t *new_extradata = av_packet_get_side_data(avpkt, AV_PKT_DATA_NEW_EXTRADATA, &new_extradata_size); int 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 && 0) { av_free(avctx->extradata); avctx->extradata = av_mallocz(new_extradata_size + AV_INPUT_BUFFER_PADDING_SIZE); if (!avctx->extradata) return AVERROR(ENOMEM); avctx->extradata_size = new_extradata_size; memcpy(avctx->extradata, new_extradata, new_extradata_size); push_output_configuration(ac); if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac, avctx->extradata, avctx->extradata_size*8LL, 1) < 0) { pop_output_configuration(ac); return AVERROR_INVALIDDATA; } } 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, data, got_frame_ptr, &gb); break; default: err = aac_decode_frame_int(avctx, data, 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; } static av_cold int aac_decode_close(AVCodecContext *avctx) { AACContext *ac = avctx->priv_data; int i, type; for (i = 0; i < MAX_ELEM_ID; i++) { for (type = 0; type < 4; type++) { if (ac->che[type][i]) AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][i]->sbr); av_freep(&ac->che[type][i]); } } ff_mdct_end(&ac->mdct); ff_mdct_end(&ac->mdct_small); ff_mdct_end(&ac->mdct_ld); ff_mdct_end(&ac->mdct_ltp); #if !USE_FIXED ff_imdct15_uninit(&ac->mdct480); #endif av_freep(&ac->fdsp); return 0; } static void aacdec_init(AACContext *c) { c->imdct_and_windowing = imdct_and_windowing; c->apply_ltp = apply_ltp; c->apply_tns = apply_tns; c->windowing_and_mdct_ltp = windowing_and_mdct_ltp; c->update_ltp = update_ltp; #if USE_FIXED c->vector_pow43 = vector_pow43; c->subband_scale = subband_scale; #endif #if !USE_FIXED if(ARCH_MIPS) ff_aacdec_init_mips(c); #endif /* !USE_FIXED */ } /** * AVOptions for Japanese DTV specific extensions (ADTS only) */ #define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM static const AVOption options[] = { {"dual_mono_mode", "Select the channel to decode for dual mono", offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2, AACDEC_FLAGS, "dual_mono_mode"}, {"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"}, {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"}, {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"}, {"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"}, {NULL}, }; static const AVClass aac_decoder_class = { .class_name = "AAC decoder", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, }; static const AVProfile profiles[] = { { FF_PROFILE_AAC_MAIN, "Main" }, { FF_PROFILE_AAC_LOW, "LC" }, { FF_PROFILE_AAC_SSR, "SSR" }, { FF_PROFILE_AAC_LTP, "LTP" }, { FF_PROFILE_AAC_HE, "HE-AAC" }, { FF_PROFILE_AAC_HE_V2, "HE-AACv2" }, { FF_PROFILE_AAC_LD, "LD" }, { FF_PROFILE_AAC_ELD, "ELD" }, { FF_PROFILE_UNKNOWN }, };