diff --git a/libavcodec/aac/Makefile b/libavcodec/aac/Makefile index c3e525d373..70b1dca274 100644 --- a/libavcodec/aac/Makefile +++ b/libavcodec/aac/Makefile @@ -2,6 +2,7 @@ clean:: $(RM) $(CLEANSUFFIXES:%=libavcodec/aac/%) OBJS-$(CONFIG_AAC_DECODER) += aac/aacdec.o aac/aacdec_tab.o \ - aac/aacdec_float.o + aac/aacdec_float.o aac/aacdec_usac.o \ + aac/aacdec_ac.o aac/aacdec_lpd.o OBJS-$(CONFIG_AAC_FIXED_DECODER) += aac/aacdec.o aac/aacdec_tab.o \ aac/aacdec_fixed.o diff --git a/libavcodec/aac/aacdec.c b/libavcodec/aac/aacdec.c index 6f37ac5361..2b8322fc68 100644 --- a/libavcodec/aac/aacdec.c +++ b/libavcodec/aac/aacdec.c @@ -40,6 +40,7 @@ #include "aacdec.h" #include "aacdec_tab.h" +#include "aacdec_usac.h" #include "libavcodec/aac.h" #include "libavcodec/aac_defines.h" @@ -535,6 +536,8 @@ static av_cold void flush(AVCodecContext *avctx) } } } + + ff_aac_usac_reset_state(ac, &ac->oc[1]); } /** @@ -993,13 +996,14 @@ static int decode_eld_specific_config(AACDecContext *ac, AVCodecContext *avctx, */ static int decode_audio_specific_config_gb(AACDecContext *ac, AVCodecContext *avctx, - MPEG4AudioConfig *m4ac, + 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) { @@ -1033,14 +1037,22 @@ static int decode_audio_specific_config_gb(AACDecContext *ac, case AOT_ER_AAC_LC: case AOT_ER_AAC_LD: if ((ret = decode_ga_specific_config(ac, avctx, gb, get_bit_alignment, - m4ac, m4ac->chan_config)) < 0) + &oc->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) + &oc->m4ac, m4ac->chan_config)) < 0) return ret; break; +#if CONFIG_AAC_DECODER + case AOT_USAC_NOSBR: /* fallthrough */ + 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", @@ -1060,7 +1072,7 @@ static int decode_audio_specific_config_gb(AACDecContext *ac, static int decode_audio_specific_config(AACDecContext *ac, AVCodecContext *avctx, - MPEG4AudioConfig *m4ac, + OutputConfiguration *oc, const uint8_t *data, int64_t bit_size, int sync_extension) { @@ -1080,7 +1092,7 @@ static int decode_audio_specific_config(AACDecContext *ac, if ((ret = init_get_bits(&gb, data, bit_size)) < 0) return ret; - return decode_audio_specific_config_gb(ac, avctx, m4ac, &gb, 0, + return decode_audio_specific_config_gb(ac, avctx, oc, &gb, 0, sync_extension); } @@ -1104,6 +1116,15 @@ 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]) { @@ -1181,7 +1202,7 @@ av_cold int ff_aac_decode_init(AVCodecContext *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].m4ac, + if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1], avctx->extradata, avctx->extradata_size * 8LL, 1)) < 0) @@ -1549,9 +1570,16 @@ static int decode_pulses(Pulse *pulse, GetBitContext *gb, 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 || + ac->oc[1].m4ac.object_type == AOT_USAC_NOSBR; 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; + + /* 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); @@ -1560,7 +1588,12 @@ int ff_aac_decode_tns(AACDecContext *ac, TemporalNoiseShaping *tns, 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) { + 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); @@ -1598,6 +1631,7 @@ static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb, { 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); @@ -2182,42 +2216,19 @@ static int aac_decode_er_frame(AVCodecContext *avctx, AVFrame *frame, return 0; } -static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, - int *got_frame_ptr, GetBitContext *gb, - const AVPacket *avpkt) +static int decode_frame_ga(AVCodecContext *avctx, AACDecContext *ac, + GetBitContext *gb, int *got_frame_ptr) { - AACDecContext *ac = avctx->priv_data; - ChannelElement *che = NULL, *che_prev = NULL; + int err; + int is_dmono; + int elem_id; enum RawDataBlockType elem_type, che_prev_type = TYPE_END; - int err, elem_id; - int samples = 0, multiplier, audio_found = 0, pce_found = 0; - int is_dmono, sce_count = 0; - int payload_alignment; 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; - ac->frame = frame; - - 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; - - payload_alignment = get_bits_count(gb); - ac->tags_mapped = 0; + int payload_alignment = get_bits_count(gb); // parse while ((elem_type = get_bits(gb, 3)) != TYPE_END) { elem_id = get_bits(gb, 4); @@ -2225,28 +2236,23 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, 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) { - err = AVERROR_INVALIDDATA; - goto fail; - } + 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) { - err = AVERROR_INVALIDDATA; - goto fail; - } + 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); - err = AVERROR_INVALIDDATA; - goto fail; + return AVERROR_INVALIDDATA; } samples = ac->oc[1].m4ac.frame_length_short ? 960 : 1024; che->present = 1; @@ -2283,10 +2289,8 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, int tags; int pushed = push_output_configuration(ac); - if (pce_found && !pushed) { - err = AVERROR_INVALIDDATA; - goto fail; - } + if (pce_found && !pushed) + return AVERROR_INVALIDDATA; tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb, payload_alignment); @@ -2312,8 +2316,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, 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; + return AVERROR_INVALIDDATA; } err = 0; while (elem_id > 0) { @@ -2337,19 +2340,16 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, } if (err) - goto fail; + return err; if (get_bits_left(gb) < 3) { av_log(avctx, AV_LOG_ERROR, overread_err); - err = AVERROR_INVALIDDATA; - goto fail; + return AVERROR_INVALIDDATA; } } - if (!avctx->ch_layout.nb_channels) { - *got_frame_ptr = 0; + 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; @@ -2364,16 +2364,17 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, if (!ac->frame->data[0] && samples) { av_log(avctx, AV_LOG_ERROR, "no frame data found\n"); - err = AVERROR_INVALIDDATA; - goto fail; + return AVERROR_INVALIDDATA; } if (samples) { ac->frame->nb_samples = samples; ac->frame->sample_rate = avctx->sample_rate; - } else + *got_frame_ptr = 1; + } else { av_frame_unref(ac->frame); - *got_frame_ptr = !!samples; + *got_frame_ptr = 0; + } /* for dual-mono audio (SCE + SCE) */ is_dmono = ac->dmono_mode && sce_count == 2 && @@ -2387,6 +2388,59 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, } 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) || + (ac->oc[1].m4ac.object_type == AOT_USAC_NOSBR)) { + 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; @@ -2414,7 +2468,7 @@ static int aac_decode_frame(AVCodecContext *avctx, AVFrame *frame, if (new_extradata) { /* discard previous configuration */ ac->oc[1].status = OC_NONE; - err = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac, + err = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1], new_extradata, new_extradata_size * 8LL, 1); if (err < 0) { diff --git a/libavcodec/aac/aacdec.h b/libavcodec/aac/aacdec.h index 8d1eb74066..ee21a94007 100644 --- a/libavcodec/aac/aacdec.h +++ b/libavcodec/aac/aacdec.h @@ -42,6 +42,8 @@ #include "libavcodec/avcodec.h" #include "libavcodec/mpeg4audio.h" +#include "aacdec_ac.h" + typedef struct AACDecContext AACDecContext; /** @@ -69,6 +71,32 @@ enum CouplingPoint { AFTER_IMDCT = 3, }; +enum AACUsacElem { + ID_USAC_SCE = 0, + ID_USAC_CPE = 1, + ID_USAC_LFE = 2, + ID_USAC_EXT = 3, +}; + +enum ExtensionHeaderType { + ID_CONFIG_EXT_FILL = 0, + ID_CONFIG_EXT_LOUDNESS_INFO = 2, + ID_CONFIG_EXT_STREAM_ID = 7, +}; + +enum AACUsacExtension { + ID_EXT_ELE_FILL, + ID_EXT_ELE_MPEGS, + ID_EXT_ELE_SAOC, + ID_EXT_ELE_AUDIOPREROLL, + ID_EXT_ELE_UNI_DRC, +}; + +enum AACUSACLoudnessExt { + UNIDRCLOUDEXT_TERM = 0x0, + UNIDRCLOUDEXT_EQ = 0x1, +}; + // Supposed to be equal to AAC_RENAME() in case of USE_FIXED. #define RENAME_FIXED(name) name ## _fixed @@ -93,6 +121,40 @@ typedef struct LongTermPrediction { int8_t used[MAX_LTP_LONG_SFB]; } LongTermPrediction; +/* Per channel core mode */ +typedef struct AACUsacElemData { + uint8_t core_mode; + uint8_t scale_factor_grouping; + + /* Timewarping ratio */ +#define NUM_TW_NODES 16 + uint8_t tw_ratio[NUM_TW_NODES]; + + struct { + uint8_t acelp_core_mode : 3; + uint8_t lpd_mode : 5; + + uint8_t bpf_control_info : 1; + uint8_t core_mode_last : 1; + uint8_t fac_data_present : 1; + + int last_lpd_mode; + } ldp; + + struct { + unsigned int seed; + uint8_t level : 3; + uint8_t offset : 5; + } noise; + + struct { + uint8_t gain; + uint32_t kv[8 /* (1024 / 16) / 8 */][8]; + } fac; + + AACArithState ac; +} AACUsacElemData; + /** * Individual Channel Stream */ @@ -145,11 +207,13 @@ typedef struct ChannelCoupling { */ typedef struct SingleChannelElement { IndividualChannelStream ics; + AACUsacElemData ue; ///< USAC element data TemporalNoiseShaping tns; enum BandType band_type[128]; ///< band types int sfo[128]; ///< scalefactor offsets INTFLOAT_UNION(sf, [128]); ///< scalefactors (8 windows * 16 sfb max) INTFLOAT_ALIGNED_UNION(32, coeffs, 1024); ///< coefficients for IMDCT, maybe processed + INTFLOAT_ALIGNED_UNION(32, prev_coeffs, 1024); ///< unscaled previous contents of coeffs[] for USAC INTFLOAT_ALIGNED_UNION(32, saved, 1536); ///< overlap INTFLOAT_ALIGNED_UNION(32, ret_buf, 2048); ///< PCM output buffer INTFLOAT_ALIGNED_UNION(16, ltp_state, 3072); ///< time signal for LTP @@ -163,25 +227,148 @@ typedef struct SingleChannelElement { }; } SingleChannelElement; +typedef struct AACUsacStereo { + uint8_t common_window; + uint8_t common_tw; + + uint8_t ms_mask_mode; + uint8_t config_idx; + + /* Complex prediction */ + uint8_t use_prev_frame; + uint8_t pred_dir; + uint8_t complex_coef; + + uint8_t pred_used[128]; + + INTFLOAT_ALIGNED_UNION(32, alpha_q_re, 1024); + INTFLOAT_ALIGNED_UNION(32, alpha_q_im, 1024); + INTFLOAT_ALIGNED_UNION(32, prev_alpha_q_re, 1024); + INTFLOAT_ALIGNED_UNION(32, prev_alpha_q_im, 1024); + + INTFLOAT_ALIGNED_UNION(32, dmix_re, 1024); + INTFLOAT_ALIGNED_UNION(32, prev_dmix_re, 1024); /* Recalculated on every frame */ + INTFLOAT_ALIGNED_UNION(32, dmix_im, 1024); /* Final prediction data */ +} AACUsacStereo; + /** * channel element - generic struct for SCE/CPE/CCE/LFE */ typedef struct ChannelElement { int present; // CPE specific + uint8_t max_sfb_ste; ///< (USAC) Maximum of both max_sfb values uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band // shared SingleChannelElement ch[2]; // CCE specific ChannelCoupling coup; + // USAC stereo coupling data + AACUsacStereo us; } ChannelElement; +typedef struct AACUSACLoudnessInfo { + uint8_t drc_set_id : 6; + uint8_t downmix_id : 7; + struct { + uint16_t lvl : 12; + uint8_t present : 1; + } sample_peak; + + struct { + uint16_t lvl : 12; + uint8_t measurement : 4; + uint8_t reliability : 2; + uint8_t present : 1; + } true_peak; + + uint8_t nb_measurements : 4; + struct { + uint8_t method_def : 4; + uint8_t method_val; + uint8_t measurement : 4; + uint8_t reliability : 2; + } measurements[16]; +} AACUSACLoudnessInfo; + +typedef struct AACUsacElemConfig { + enum AACUsacElem type; + + uint8_t tw_mdct : 1; + uint8_t noise_fill : 1; + + uint8_t stereo_config_index; + + struct { + int ratio; + + uint8_t harmonic_sbr : 1; /* harmonicSBR */ + uint8_t bs_intertes : 1; /* bs_interTes */ + uint8_t bs_pvc : 1; /* bs_pvc */ + + struct { + uint8_t start_freq; /* dflt_start_freq */ + uint8_t stop_freq; /* dflt_stop_freq */ + + uint8_t freq_scale; /* dflt_freq_scale */ + uint8_t alter_scale : 1; /* dflt_alter_scale */ + uint8_t noise_scale; /* dflt_noise_scale */ + + uint8_t limiter_bands; /* dflt_limiter_bands */ + uint8_t limiter_gains; /* dflt_limiter_gains */ + uint8_t interpol_freq : 1; /* dflt_interpol_freq */ + uint8_t smoothing_mode : 1; /* dflt_smoothing_mode */ + } dflt; + } sbr; + + struct { + uint8_t freq_res; /* bsFreqRes */ + uint8_t fixed_gain; /* bsFixedGainDMX */ + uint8_t temp_shape_config; /* bsTempShapeConfig */ + uint8_t decorr_config; /* bsDecorrConfig */ + uint8_t high_rate_mode : 1; /* bsHighRateMode */ + uint8_t phase_coding : 1; /* bsPhaseCoding */ + + uint8_t otts_bands_phase; /* bsOttBandsPhase */ + uint8_t residual_coding; /* bsResidualCoding */ + uint8_t residual_bands; /* bsResidualBands */ + uint8_t pseudo_lr : 1; /* bsPseudoLr */ + uint8_t env_quant_mode : 1; /* bsEnvQuantMode */ + } mps; + + struct { + enum AACUsacExtension type; + uint8_t payload_frag; + uint32_t default_len; + uint32_t pl_data_offset; + uint8_t *pl_data; + } ext; +} AACUsacElemConfig; + +typedef struct AACUSACConfig { + uint8_t core_sbr_frame_len_idx; /* coreSbrFrameLengthIndex */ + uint8_t rate_idx; + uint16_t core_frame_len; + uint16_t stream_identifier; + + AACUsacElemConfig elems[64]; + int nb_elems; + + struct { + uint8_t nb_album; + AACUSACLoudnessInfo album_info[64]; + uint8_t nb_info; + AACUSACLoudnessInfo info[64]; + } loudness; +} AACUSACConfig; + typedef struct OutputConfiguration { MPEG4AudioConfig m4ac; uint8_t layout_map[MAX_ELEM_ID*4][3]; int layout_map_tags; AVChannelLayout ch_layout; enum OCStatus status; + AACUSACConfig usac; } OutputConfiguration; /** diff --git a/libavcodec/aac/aacdec_ac.c b/libavcodec/aac/aacdec_ac.c new file mode 100644 index 0000000000..7e5077cd19 --- /dev/null +++ b/libavcodec/aac/aacdec_ac.c @@ -0,0 +1,208 @@ +/* + * AAC definitions and structures + * Copyright (c) 2024 Lynne + * + * 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 + */ + +#include "libavcodec/aactab.h" +#include "aacdec_ac.h" + +uint32_t ff_aac_ac_map_process(AACArithState *state, int reset, int N) +{ + float ratio; + if (reset) { + memset(state->last, 0, sizeof(state->last)); + state->last_len = N; + } else if (state->last_len != N) { + int i; + uint8_t last[512 /* 2048 / 4 */]; + memcpy(last, state->last, sizeof(last)); + + ratio = state->last_len / (float)N; + for (i = 0; i < N/2; i++) { + int k = (int)(i * ratio); + state->last[i] = last[k]; + } + + for (; i < FF_ARRAY_ELEMS(state->last); i++) + state->last[i] = 0; + + state->last_len = N; + } + + state->cur[3] = 0; + state->cur[2] = 0; + state->cur[1] = 0; + state->cur[0] = 1; + + state->state_pre = state->last[0] << 12; + return state->last[0] << 12; +} + +uint32_t ff_aac_ac_get_context(AACArithState *state, uint32_t c, int i, int N) +{ + c = state->state_pre >> 8; + c = c + (state->last[i + 1] << 8); + c = (c << 4); + c += state->cur[1]; + + state->state_pre = c; + + if (i > 3 && + ((state->cur[3] + state->cur[2] + state->cur[1]) < 5)) + return c + 0x10000; + + return c; +} + +uint32_t ff_aac_ac_get_pk(uint32_t c) +{ + int i_min = -1; + int i, j; + int i_max = FF_ARRAY_ELEMS(ff_aac_ac_lookup_m) - 1; + while ((i_max - i_min) > 1) { + i = i_min + ((i_max - i_min) / 2); + j = ff_aac_ac_hash_m[i]; + if (c < (j >> 8)) + i_max = i; + else if (c > (j >> 8)) + i_min = i; + else + return (j & 0xFF); + } + return ff_aac_ac_lookup_m[i_max]; +} + +void ff_aac_ac_update_context(AACArithState *state, int idx, + uint16_t a, uint16_t b) +{ + state->cur[0] = a + b + 1; + if (state->cur[0] > 0xF) + state->cur[0] = 0xF; + + state->cur[3] = state->cur[2]; + state->cur[2] = state->cur[1]; + state->cur[1] = state->cur[0]; + + state->last[idx] = state->cur[0]; +} + +/* Initialize AC */ +void ff_aac_ac_init(AACArith *ac, GetBitContext *gb) +{ + ac->low = 0; + ac->high = UINT16_MAX; + ac->val = get_bits(gb, 16); +} + +uint16_t ff_aac_ac_decode(AACArith *ac, GetBitContext *gb, + const uint16_t *cdf, uint16_t cdf_len) +{ + int val = ac->val; + int low = ac->low; + int high = ac->high; + + int sym; + int rng = high - low + 1; + int c = ((((int)(val - low + 1)) << 14) - ((int)1)); + + const uint16_t *p = cdf - 1; + + /* One for each possible CDF length in the spec */ + switch (cdf_len) { + case 2: + if ((p[1] * rng) > c) + p += 1; + break; + case 4: + if ((p[2] * rng) > c) + p += 2; + if ((p[1] * rng) > c) + p += 1; + break; + case 17: + /* First check if the current probability is even met at all */ + if ((p[1] * rng) <= c) + break; + p += 1; + for (int i = 8; i >= 1; i >>= 1) + if ((p[i] * rng) > c) + p += i; + break; + case 27: + if ((p[16] * rng) > c) + p += 16; + if ((p[8] * rng) > c) + p += 8; + if (p != (cdf - 1 + 24)) + if ((p[4] * rng) > c) + p += 4; + if ((p[2] * rng) > c) + p += 2; + + if (p != (cdf - 1 + 24 + 2)) + if ((p[1] * rng) > c) + p += 1; + break; + default: + /* This should never happen */ + av_assert2(0); + } + + sym = (int)((ptrdiff_t)(p - cdf)) + 1; + if (sym) + high = low + ((rng * cdf[sym - 1]) >> 14) - 1; + low += (rng * cdf[sym]) >> 14; + + /* This loop could be done faster */ + while (1) { + if (high < 32768) { + ; + } else if (low >= 32768) { + val -= 32768; + low -= 32768; + high -= 32768; + } else if (low >= 16384 && high < 49152) { + val -= 16384; + low -= 16384; + high -= 16384; + } else { + break; + } + low += low; + high += high + 1; + val = (val << 1) | get_bits1(gb); + }; + + ac->low = low; + ac->high = high; + ac->val = val; + + return sym; +} + +void ff_aac_ac_finish(AACArithState *state, int offset, int N) +{ + int i; + + for (i = offset; i < N/2; i++) + state->last[i] = 1; + + for (; i < FF_ARRAY_ELEMS(state->last); i++) + state->last[i] = 0; +} diff --git a/libavcodec/aac/aacdec_ac.h b/libavcodec/aac/aacdec_ac.h new file mode 100644 index 0000000000..0b98c0f0d9 --- /dev/null +++ b/libavcodec/aac/aacdec_ac.h @@ -0,0 +1,54 @@ +/* + * AAC definitions and structures + * Copyright (c) 2024 Lynne + * + * 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 + */ + +#ifndef AVCODEC_AAC_AACDEC_AC_H +#define AVCODEC_AAC_AACDEC_AC_H + +#include "libavcodec/get_bits.h" + +typedef struct AACArithState { + uint8_t last[512 /* 2048 / 4 */]; + int last_len; + uint8_t cur[4]; + uint16_t state_pre; +} AACArithState; + +typedef struct AACArith { + uint16_t low; + uint16_t high; + uint16_t val; +} AACArith; + +#define FF_AAC_AC_ESCAPE 16 + +uint32_t ff_aac_ac_map_process(AACArithState *state, int reset, int len); +uint32_t ff_aac_ac_get_context(AACArithState *state, uint32_t old_c, int idx, int len); +uint32_t ff_aac_ac_get_pk(uint32_t c); + +void ff_aac_ac_update_context(AACArithState *state, int idx, uint16_t a, uint16_t b); +void ff_aac_ac_init(AACArith *ac, GetBitContext *gb); + +uint16_t ff_aac_ac_decode(AACArith *ac, GetBitContext *gb, + const uint16_t *cdf, uint16_t cdf_len); + +void ff_aac_ac_finish(AACArithState *state, int offset, int nb); + +#endif /* AVCODEC_AACDEC_AC_H */ diff --git a/libavcodec/aac/aacdec_dsp_template.c b/libavcodec/aac/aacdec_dsp_template.c index 59a69d88f3..8d31af22f8 100644 --- a/libavcodec/aac/aacdec_dsp_template.c +++ b/libavcodec/aac/aacdec_dsp_template.c @@ -88,8 +88,8 @@ static void AAC_RENAME(apply_mid_side_stereo)(AACDecContext *ac, ChannelElement INTFLOAT *ch1 = cpe->ch[1].AAC_RENAME(coeffs); const uint16_t *offsets = ics->swb_offset; for (int g = 0; g < ics->num_window_groups; g++) { - for (int sfb = 0; sfb < ics->max_sfb; sfb++) { - const int idx = g*ics->max_sfb + sfb; + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) { + const int idx = g*cpe->max_sfb_ste + sfb; if (cpe->ms_mask[idx] && cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { diff --git a/libavcodec/aac/aacdec_latm.h b/libavcodec/aac/aacdec_latm.h index e40a2fe1a7..047c11e0fb 100644 --- a/libavcodec/aac/aacdec_latm.h +++ b/libavcodec/aac/aacdec_latm.h @@ -56,7 +56,8 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, { AACDecContext *ac = &latmctx->aac_ctx; AVCodecContext *avctx = ac->avctx; - MPEG4AudioConfig m4ac = { 0 }; + OutputConfiguration oc = { 0 }; + MPEG4AudioConfig *m4ac = &oc.m4ac; GetBitContext gbc; int config_start_bit = get_bits_count(gb); int sync_extension = 0; @@ -76,7 +77,7 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, if (get_bits_left(gb) <= 0) return AVERROR_INVALIDDATA; - bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac, + bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &oc, &gbc, config_start_bit, sync_extension); @@ -88,11 +89,12 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, asclen = bits_consumed; if (!latmctx->initialized || - ac->oc[1].m4ac.sample_rate != m4ac.sample_rate || - ac->oc[1].m4ac.chan_config != m4ac.chan_config) { + ac->oc[1].m4ac.sample_rate != m4ac->sample_rate || + ac->oc[1].m4ac.chan_config != m4ac->chan_config) { if (latmctx->initialized) { - av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config); + av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", + m4ac->sample_rate, m4ac->chan_config); } else { av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n"); } @@ -280,7 +282,7 @@ static int latm_decode_frame(AVCodecContext *avctx, AVFrame *out, } else { push_output_configuration(&latmctx->aac_ctx); if ((err = decode_audio_specific_config( - &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac, + &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1], avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) { pop_output_configuration(&latmctx->aac_ctx); return err; diff --git a/libavcodec/aac/aacdec_lpd.c b/libavcodec/aac/aacdec_lpd.c new file mode 100644 index 0000000000..796edd2ab5 --- /dev/null +++ b/libavcodec/aac/aacdec_lpd.c @@ -0,0 +1,198 @@ +/* + * Copyright (c) 2024 Lynne + * + * 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 + */ + +#include "aacdec_lpd.h" +#include "aacdec_usac.h" +#include "libavcodec/unary.h" + +const uint8_t ff_aac_lpd_mode_tab[32][4] = { + { 0, 0, 0, 0 }, + { 1, 0, 0, 0 }, + { 0, 1, 0, 0 }, + { 1, 1, 0, 0 }, + { 0, 0, 1, 0 }, + { 1, 0, 1, 0 }, + { 0, 1, 1, 0 }, + { 1, 1, 1, 0 }, + { 0, 0, 0, 1 }, + { 1, 0, 0, 1 }, + { 0, 1, 0, 1 }, + { 1, 1, 0, 1 }, + { 0, 0, 1, 1 }, + { 1, 0, 1, 1 }, + { 0, 1, 1, 1 }, + { 1, 1, 1, 1 }, + { 2, 2, 0, 0 }, + { 2, 2, 1, 0 }, + { 2, 2, 0, 1 }, + { 2, 2, 1, 1 }, + { 0, 0, 2, 2 }, + { 1, 0, 2, 2 }, + { 0, 1, 2, 2 }, + { 1, 1, 2, 2 }, + { 2, 2, 2, 2 }, + { 3, 3, 3, 3 }, + /* Larger values are reserved, but permit them for resilience */ + { 0, 0, 0, 0 }, + { 0, 0, 0, 0 }, + { 0, 0, 0, 0 }, + { 0, 0, 0, 0 }, + { 0, 0, 0, 0 }, + { 0, 0, 0, 0 }, +}; + +static void parse_qn(GetBitContext *gb, int *qn, int nk_mode, int no_qn) +{ + if (nk_mode == 1) { + for (int k = 0; k < no_qn; k++) { + qn[k] = get_unary(gb, 0, INT32_MAX); // TODO: find proper ranges + if (qn[k]) + qn[k]++; + } + return; + } + + for (int k = 0; k < no_qn; k++) + qn[k] = get_bits(gb, 2) + 2; + + if (nk_mode == 2) { + for (int k = 0; k < no_qn; k++) { + if (qn[k] > 4) { + qn[k] = get_unary(gb, 0, INT32_MAX);; + if (qn[k]) + qn[k] += 4; + } + } + return; + } + + for (int k = 0; k < no_qn; k++) { + if (qn[k] > 4) { + int qn_ext = get_unary(gb, 0, INT32_MAX);; + switch (qn_ext) { + case 0: qn[k] = 5; break; + case 1: qn[k] = 6; break; + case 2: qn[k] = 0; break; + default: qn[k] = qn_ext + 4; break; + } + } + } +} + +static int parse_codebook_idx(GetBitContext *gb, uint32_t *kv, + int nk_mode, int no_qn) +{ + int idx, n, nk; + + int qn[2]; + parse_qn(gb, qn, nk_mode, no_qn); + + for (int k = 0; k < no_qn; k++) { + if (qn[k] > 4) { + nk = (qn[k] - 3) / 2; + n = qn[k] - nk*2; + } else { + nk = 0; + n = qn[k]; + } + } + + idx = get_bits(gb, 4*n); + + if (nk > 0) + for (int i = 0; i < 8; i++) + kv[i] = get_bits(gb, nk); + + return 0; +} + +int ff_aac_parse_fac_data(AACUsacElemData *ce, GetBitContext *gb, + int use_gain, int len) +{ + int ret; + if (use_gain) + ce->fac.gain = get_bits(gb, 7); + + for (int i = 0; i < len/8; i++) { + ret = parse_codebook_idx(gb, ce->fac.kv[i], 1, 1); + if (ret < 0) + return ret; + } + + return 0; +} + +int ff_aac_ldp_parse_channel_stream(AACDecContext *ac, AACUSACConfig *usac, + AACUsacElemData *ce, GetBitContext *gb) +{ + int k; + const uint8_t *mod; + int first_ldp_flag; + int first_tcx_flag; + + ce->ldp.acelp_core_mode = get_bits(gb, 3); + ce->ldp.lpd_mode = get_bits(gb, 5); + + ce->ldp.bpf_control_info = get_bits1(gb); + ce->ldp.core_mode_last = get_bits1(gb); + ce->ldp.fac_data_present = get_bits1(gb); + + mod = ff_aac_lpd_mode_tab[ce->ldp.lpd_mode]; + + first_ldp_flag = !ce->ldp.core_mode_last; + first_tcx_flag = 1; + if (first_ldp_flag) + ce->ldp.last_lpd_mode = -1; /* last_ldp_mode is a **STATEFUL** value */ + + k = 0; + while (k < 0) { + if (!k) { + if (ce->ldp.core_mode_last && ce->ldp.fac_data_present) + ff_aac_parse_fac_data(ce, gb, 0, usac->core_frame_len/8); + } else { + if (!ce->ldp.last_lpd_mode && mod[k] > 0 || + ce->ldp.last_lpd_mode && !mod[k]) + ff_aac_parse_fac_data(ce, gb, 0, usac->core_frame_len/8); + } + if (!mod[k]) { +// parse_acelp_coding(); + ce->ldp.last_lpd_mode = 0; + k++; + } else { +// parse_tcx_coding(); + ce->ldp.last_lpd_mode = mod[k]; + k += (1 << (mod[k] - 1)); + first_tcx_flag = 0; + } + } + +// parse_lpc_data(first_lpd_flag); + + if (!ce->ldp.core_mode_last && ce->ldp.fac_data_present) { + uint16_t len_8 = usac->core_frame_len / 8; + uint16_t len_16 = usac->core_frame_len / 16; + uint16_t fac_len = get_bits1(gb) /* short_fac_flag */ ? len_8 : len_16; + int ret = ff_aac_parse_fac_data(ce, gb, 1, fac_len); + if (ret < 0) + return ret; + } + + return 0; +} diff --git a/libavcodec/aac/aacdec_lpd.h b/libavcodec/aac/aacdec_lpd.h new file mode 100644 index 0000000000..924ff75e52 --- /dev/null +++ b/libavcodec/aac/aacdec_lpd.h @@ -0,0 +1,33 @@ +/* + * Copyright (c) 2024 Lynne + * + * 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 + */ + +#ifndef AVCODEC_AAC_AACDEC_LPD_H +#define AVCODEC_AAC_AACDEC_LPD_H + +#include "aacdec.h" +#include "libavcodec/get_bits.h" + +int ff_aac_parse_fac_data(AACUsacElemData *ce, GetBitContext *gb, + int use_gain, int len); + +int ff_aac_ldp_parse_channel_stream(AACDecContext *ac, AACUSACConfig *usac, + AACUsacElemData *ce, GetBitContext *gb); + +#endif /* AVCODEC_AAC_AACDEC_LPD_H */ diff --git a/libavcodec/aac/aacdec_usac.c b/libavcodec/aac/aacdec_usac.c new file mode 100644 index 0000000000..c3c9137a2e --- /dev/null +++ b/libavcodec/aac/aacdec_usac.c @@ -0,0 +1,1608 @@ +/* + * Copyright (c) 2024 Lynne + * + * 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 + */ + +#include "aacdec_usac.h" +#include "aacdec_tab.h" +#include "aacdec_lpd.h" +#include "aacdec_ac.h" + +#include "libavcodec/aactab.h" +#include "libavutil/mem.h" +#include "libavcodec/mpeg4audio.h" +#include "libavcodec/unary.h" + +/* Number of scalefactor bands per complex prediction band, equal to 2. */ +#define SFB_PER_PRED_BAND 2 + +static inline uint32_t get_escaped_value(GetBitContext *gb, int nb1, int nb2, int nb3) +{ + uint32_t val = get_bits(gb, nb1), val2; + if (val < ((1 << nb1) - 1)) + return val; + + val += val2 = get_bits(gb, nb2); + if (val2 == ((1 << nb2) - 1)) + val += get_bits(gb, nb3); + + return val; +} + +/* ISO/IEC 23003-3, Table 74 — bsOutputChannelPos */ +static const enum AVChannel usac_ch_pos_to_av[64] = { + [0] = AV_CHAN_FRONT_LEFT, + [1] = AV_CHAN_FRONT_RIGHT, + [2] = AV_CHAN_FRONT_CENTER, + [3] = AV_CHAN_LOW_FREQUENCY, + [4] = AV_CHAN_SIDE_LEFT, // +110 degrees, Ls|LS|kAudioChannelLabel_LeftSurround + [5] = AV_CHAN_SIDE_RIGHT, // -110 degrees, Rs|RS|kAudioChannelLabel_RightSurround + [6] = AV_CHAN_FRONT_LEFT_OF_CENTER, + [7] = AV_CHAN_FRONT_RIGHT_OF_CENTER, + [8] = AV_CHAN_BACK_LEFT, // +135 degrees, Lsr|BL|kAudioChannelLabel_RearSurroundLeft + [9] = AV_CHAN_BACK_RIGHT, // -135 degrees, Rsr|BR|kAudioChannelLabel_RearSurroundRight + [10] = AV_CHAN_BACK_CENTER, + [11] = AV_CHAN_SURROUND_DIRECT_LEFT, + [12] = AV_CHAN_SURROUND_DIRECT_RIGHT, + [13] = AV_CHAN_SIDE_SURROUND_LEFT, // +90 degrees, Lss|SL|kAudioChannelLabel_LeftSideSurround + [14] = AV_CHAN_SIDE_SURROUND_RIGHT, // -90 degrees, Rss|SR|kAudioChannelLabel_RightSideSurround + [15] = AV_CHAN_WIDE_LEFT, // +60 degrees, Lw|FLw|kAudioChannelLabel_LeftWide + [16] = AV_CHAN_WIDE_RIGHT, // -60 degrees, Rw|FRw|kAudioChannelLabel_RightWide + [17] = AV_CHAN_TOP_FRONT_LEFT, + [18] = AV_CHAN_TOP_FRONT_RIGHT, + [19] = AV_CHAN_TOP_FRONT_CENTER, + [20] = AV_CHAN_TOP_BACK_LEFT, + [21] = AV_CHAN_TOP_BACK_RIGHT, + [22] = AV_CHAN_TOP_BACK_CENTER, + [23] = AV_CHAN_TOP_SIDE_LEFT, + [24] = AV_CHAN_TOP_SIDE_RIGHT, + [25] = AV_CHAN_TOP_CENTER, + [26] = AV_CHAN_LOW_FREQUENCY_2, + [27] = AV_CHAN_BOTTOM_FRONT_LEFT, + [28] = AV_CHAN_BOTTOM_FRONT_RIGHT, + [29] = AV_CHAN_BOTTOM_FRONT_CENTER, + [30] = AV_CHAN_TOP_SURROUND_LEFT, ///< +110 degrees, Lvs, TpLS + [31] = AV_CHAN_TOP_SURROUND_RIGHT, ///< -110 degrees, Rvs, TpRS +}; + +static int decode_loudness_info(AACDecContext *ac, AACUSACLoudnessInfo *info, + GetBitContext *gb) +{ + info->drc_set_id = get_bits(gb, 6); + info->downmix_id = get_bits(gb, 7); + + if ((info->sample_peak.present = get_bits1(gb))) /* samplePeakLevelPresent */ + info->sample_peak.lvl = get_bits(gb, 12); + + if ((info->true_peak.present = get_bits1(gb))) { /* truePeakLevelPresent */ + info->true_peak.lvl = get_bits(gb, 12); + info->true_peak.measurement = get_bits(gb, 4); + info->true_peak.reliability = get_bits(gb, 2); + } + + info->nb_measurements = get_bits(gb, 4); + for (int i = 0; i < info->nb_measurements; i++) { + info->measurements[i].method_def = get_bits(gb, 4); + info->measurements[i].method_val = get_unary(gb, 0, 8); + info->measurements[i].measurement = get_bits(gb, 4); + info->measurements[i].reliability = get_bits(gb, 2); + } + + return 0; +} + +static int decode_loudness_set(AACDecContext *ac, AACUSACConfig *usac, + GetBitContext *gb) +{ + int ret; + + usac->loudness.nb_album = get_bits(gb, 6); /* loudnessInfoAlbumCount */ + usac->loudness.nb_info = get_bits(gb, 6); /* loudnessInfoCount */ + + for (int i = 0; i < usac->loudness.nb_album; i++) { + ret = decode_loudness_info(ac, &usac->loudness.album_info[i], gb); + if (ret < 0) + return ret; + } + + for (int i = 0; i < usac->loudness.nb_info; i++) { + ret = decode_loudness_info(ac, &usac->loudness.info[i], gb); + if (ret < 0) + return ret; + } + + if (get_bits1(gb)) { /* loudnessInfoSetExtPresent */ + enum AACUSACLoudnessExt type; + while ((type = get_bits(gb, 4)) != UNIDRCLOUDEXT_TERM) { + uint8_t size_bits = get_bits(gb, 4) + 4; + uint8_t bit_size = get_bits(gb, size_bits) + 1; + switch (type) { + case UNIDRCLOUDEXT_EQ: + avpriv_report_missing_feature(ac->avctx, "loudnessInfoV1"); + return AVERROR_PATCHWELCOME; + default: + for (int i = 0; i < bit_size; i++) + skip_bits1(gb); + } + } + } + + return 0; +} + +static void decode_usac_sbr_data(AACUsacElemConfig *e, GetBitContext *gb) +{ + uint8_t header_extra1; + uint8_t header_extra2; + + e->sbr.harmonic_sbr = get_bits1(gb); /* harmonicSBR */ + e->sbr.bs_intertes = get_bits1(gb); /* bs_interTes */ + e->sbr.bs_pvc = get_bits1(gb); /* bs_pvc */ + + e->sbr.dflt.start_freq = get_bits(gb, 4); /* dflt_start_freq */ + e->sbr.dflt.stop_freq = get_bits(gb, 4); /* dflt_stop_freq */ + + header_extra1 = get_bits1(gb); /* dflt_header_extra1 */ + header_extra2 = get_bits1(gb); /* dflt_header_extra2 */ + + e->sbr.dflt.freq_scale = 2; + e->sbr.dflt.alter_scale = 1; + e->sbr.dflt.noise_scale = 2; + if (header_extra1) { + e->sbr.dflt.freq_scale = get_bits(gb, 2); /* dflt_freq_scale */ + e->sbr.dflt.alter_scale = get_bits1(gb); /* dflt_alter_scale */ + e->sbr.dflt.noise_scale = get_bits(gb, 2); /* dflt_noise_scale */ + } + + e->sbr.dflt.limiter_bands = 2; + e->sbr.dflt.limiter_gains = 2; + e->sbr.dflt.interpol_freq = 1; + e->sbr.dflt.smoothing_mode = 1; + if (header_extra2) { + e->sbr.dflt.limiter_bands = get_bits(gb, 2); /* dflt_limiter_bands */ + e->sbr.dflt.limiter_gains = get_bits(gb, 2); /* dflt_limiter_gains */ + e->sbr.dflt.interpol_freq = get_bits1(gb); /* dflt_interpol_freq */ + e->sbr.dflt.smoothing_mode = get_bits1(gb); /* dflt_smoothing_mode */ + } +} + +static void decode_usac_element_core(AACUsacElemConfig *e, + GetBitContext *gb, + int sbr_ratio) +{ + e->tw_mdct = get_bits1(gb); /* tw_mdct */ + e->noise_fill = get_bits1(gb); + e->sbr.ratio = sbr_ratio; +} + +static void decode_usac_element_pair(AACUsacElemConfig *e, GetBitContext *gb) +{ + e->stereo_config_index = 0; + if (e->sbr.ratio) { + decode_usac_sbr_data(e, gb); + e->stereo_config_index = get_bits(gb, 2); + } + if (e->stereo_config_index) { + e->mps.freq_res = get_bits(gb, 3); /* bsFreqRes */ + e->mps.fixed_gain = get_bits(gb, 3); /* bsFixedGainDMX */ + e->mps.temp_shape_config = get_bits(gb, 2); /* bsTempShapeConfig */ + e->mps.decorr_config = get_bits(gb, 2); /* bsDecorrConfig */ + e->mps.high_rate_mode = get_bits1(gb); /* bsHighRateMode */ + e->mps.phase_coding = get_bits1(gb); /* bsPhaseCoding */ + + if (get_bits1(gb)) /* bsOttBandsPhasePresent */ + e->mps.otts_bands_phase = get_bits(gb, 5); /* bsOttBandsPhase */ + + e->mps.residual_coding = e->stereo_config_index >= 2; /* bsResidualCoding */ + if (e->mps.residual_coding) { + e->mps.residual_bands = get_bits(gb, 5); /* bsResidualBands */ + e->mps.pseudo_lr = get_bits1(gb); /* bsPseudoLr */ + } + if (e->mps.temp_shape_config == 2) + e->mps.env_quant_mode = get_bits1(gb); /* bsEnvQuantMode */ + } +} + +static int decode_usac_extension(AACDecContext *ac, AACUsacElemConfig *e, + GetBitContext *gb) +{ + int len = 0, ext_config_len; + + e->ext.type = get_escaped_value(gb, 4, 8, 16); /* usacExtElementType */ + ext_config_len = get_escaped_value(gb, 4, 8, 16); /* usacExtElementConfigLength */ + + if (get_bits1(gb)) /* usacExtElementDefaultLengthPresent */ + len = get_escaped_value(gb, 8, 16, 0) + 1; + + e->ext.default_len = len; + e->ext.payload_frag = get_bits1(gb); /* usacExtElementPayloadFrag */ + + av_log(ac->avctx, AV_LOG_DEBUG, "Extension present: type %i, len %i\n", + e->ext.type, ext_config_len); + + switch (e->ext.type) { +#if 0 /* Skip unsupported values */ + case ID_EXT_ELE_MPEGS: + break; + case ID_EXT_ELE_SAOC: + break; + case ID_EXT_ELE_UNI_DRC: + break; +#endif + case ID_EXT_ELE_FILL: + break; /* This is what the spec does */ + case ID_EXT_ELE_AUDIOPREROLL: + /* No configuration needed - fallthrough (len should be 0) */ + default: + skip_bits(gb, 8*ext_config_len); + break; + }; + + return 0; +} + +int ff_aac_usac_reset_state(AACDecContext *ac, OutputConfiguration *oc) +{ + AACUSACConfig *usac = &oc->usac; + int elem_id[3 /* SCE, CPE, LFE */] = { 0, 0, 0 }; + + ChannelElement *che; + enum RawDataBlockType type; + int id, ch; + + /* Initialize state */ + for (int i = 0; i < usac->nb_elems; i++) { + AACUsacElemConfig *e = &usac->elems[i]; + if (e->type != ID_USAC_SCE && e->type != ID_USAC_CPE) + continue; + + if (e->type == ID_USAC_SCE) { + ch = 1; + type = TYPE_SCE; + id = elem_id[0]++; + } else { + ch = 2; + type = TYPE_CPE; + id = elem_id[1]++; + } + + che = ff_aac_get_che(ac, type, id); + if (che) { + AACUsacStereo *us = &che->us; + memset(us, 0, sizeof(*us)); + + for (int j = 0; j < ch; j++) { + SingleChannelElement *sce = &che->ch[ch]; + AACUsacElemData *ue = &sce->ue; + + memset(ue, 0, sizeof(*ue)); + + if (!ch) + ue->noise.seed = 0x3039; + else + che->ch[1].ue.noise.seed = 0x10932; + } + } + } + + return 0; +} + +/* UsacConfig */ +int ff_aac_usac_config_decode(AACDecContext *ac, AVCodecContext *avctx, + GetBitContext *gb, OutputConfiguration *oc, + int channel_config) +{ + int ret, idx; + uint8_t freq_idx; + uint8_t channel_config_idx; + int nb_elements; + int samplerate; + int sbr_ratio; + MPEG4AudioConfig *m4ac = &oc->m4ac; + AACUSACConfig *usac = &oc->usac; + int elem_id[3 /* SCE, CPE, LFE */]; + + uint8_t layout_map[MAX_ELEM_ID*4][3]; + + memset(usac, 0, sizeof(*usac)); + + freq_idx = get_bits(gb, 5); /* usacSamplingFrequencyIndex */ + if (freq_idx == 0x1f) { + samplerate = get_bits(gb, 24); /* usacSamplingFrequency */ + + /* Try to match up an index for the custom sample rate. + * TODO: not sure if correct */ + for (idx = 0; idx < /* FF_ARRAY_ELEMS(ff_aac_usac_samplerate) */ 32; idx++) { + if (ff_aac_usac_samplerate[idx] >= samplerate) + break; + } + idx = FFMIN(idx, /* FF_ARRAY_ELEMS(ff_aac_usac_samplerate) */ 32 - 1); + usac->rate_idx = idx; + } else { + samplerate = ff_aac_usac_samplerate[freq_idx]; + if (samplerate < 0) + return AVERROR(EINVAL); + usac->rate_idx = freq_idx; + } + + m4ac->sample_rate = avctx->sample_rate = samplerate; + + usac->core_sbr_frame_len_idx = get_bits(gb, 3); /* coreSbrFrameLengthIndex */ + m4ac->frame_length_short = usac->core_sbr_frame_len_idx == 0 || + usac->core_sbr_frame_len_idx == 2; + + usac->core_frame_len = (usac->core_sbr_frame_len_idx == 0 || + usac->core_sbr_frame_len_idx == 2) ? 768 : 1024; + + sbr_ratio = usac->core_sbr_frame_len_idx == 2 ? 2 : + usac->core_sbr_frame_len_idx == 3 ? 3 : + usac->core_sbr_frame_len_idx == 4 ? 1 : + 0; + + channel_config_idx = get_bits(gb, 5); /* channelConfigurationIndex */ + if (!channel_config_idx) { + /* UsacChannelConfig() */ + uint8_t nb_channels = get_escaped_value(gb, 5, 8, 16); /* numOutChannels */ + if (nb_channels >= 64) + return AVERROR(EINVAL); + + av_channel_layout_uninit(&ac->oc[1].ch_layout); + + ret = av_channel_layout_custom_init(&ac->oc[1].ch_layout, nb_channels); + if (ret < 0) + return ret; + + for (int i = 0; i < nb_channels; i++) { + AVChannelCustom *cm = &ac->oc[1].ch_layout.u.map[i]; + cm->id = usac_ch_pos_to_av[get_bits(gb, 5)]; /* bsOutputChannelPos */ + if (cm->id == AV_CHAN_NONE) + cm->id = AV_CHAN_UNKNOWN; + } + + ret = av_channel_layout_retype(&ac->oc[1].ch_layout, + AV_CHANNEL_ORDER_NATIVE, + AV_CHANNEL_LAYOUT_RETYPE_FLAG_CANONICAL); + if (ret < 0) + return ret; + + ret = av_channel_layout_copy(&avctx->ch_layout, &ac->oc[1].ch_layout); + if (ret < 0) + return ret; + } else { + if ((ret = ff_aac_set_default_channel_config(ac, avctx, layout_map, + &nb_elements, channel_config_idx))) + return ret; + } + + /* UsacDecoderConfig */ + elem_id[0] = elem_id[1] = elem_id[2] = 0; + usac->nb_elems = get_escaped_value(gb, 4, 8, 16) + 1; + + for (int i = 0; i < usac->nb_elems; i++) { + AACUsacElemConfig *e = &usac->elems[i]; + memset(e, 0, sizeof(*e)); + + e->type = get_bits(gb, 2); /* usacElementType */ + av_log(ac->avctx, AV_LOG_DEBUG, "Element present: idx %i, type %i\n", + i, e->type); + + switch (e->type) { + case ID_USAC_SCE: /* SCE */ + /* UsacCoreConfig */ + decode_usac_element_core(e, gb, sbr_ratio); + if (e->sbr.ratio > 0) + decode_usac_sbr_data(e, gb); + layout_map[i][0] = TYPE_SCE; + layout_map[i][1] = i; + layout_map[i][2] = AAC_CHANNEL_FRONT; + elem_id[0]++; + + break; + case ID_USAC_CPE: /* UsacChannelPairElementConf */ + /* UsacCoreConfig */ + decode_usac_element_core(e, gb, sbr_ratio); + decode_usac_element_pair(e, gb); + layout_map[i][0] = TYPE_CPE; + layout_map[i][1] = i; + layout_map[i][2] = AAC_CHANNEL_FRONT; + elem_id[1]++; + + break; + case ID_USAC_LFE: /* LFE */ + /* LFE has no need for any configuration */ + e->tw_mdct = 0; + e->noise_fill = 0; + elem_id[2]++; + break; + case ID_USAC_EXT: /* EXT */ + ret = decode_usac_extension(ac, e, gb); + if (ret < 0) + return ret; + break; + }; + } + + ret = ff_aac_output_configure(ac, layout_map, elem_id[0] + elem_id[1] + elem_id[2], OC_GLOBAL_HDR, 0); + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "Unable to parse channel config!\n"); + return ret; + } + + if (get_bits1(gb)) { /* usacConfigExtensionPresent */ + int invalid; + int nb_extensions = get_escaped_value(gb, 2, 4, 8) + 1; /* numConfigExtensions */ + for (int i = 0; i < nb_extensions; i++) { + int type = get_escaped_value(gb, 4, 8, 16); + int len = get_escaped_value(gb, 4, 8, 16); + switch (type) { + case ID_CONFIG_EXT_LOUDNESS_INFO: + ret = decode_loudness_set(ac, usac, gb); + if (ret < 0) + return ret; + break; + case ID_CONFIG_EXT_STREAM_ID: + usac->stream_identifier = get_bits(gb, 16); + break; + case ID_CONFIG_EXT_FILL: /* fallthrough */ + invalid = 0; + while (len--) { + if (get_bits(gb, 8) != 0xA5) + invalid++; + } + if (invalid) + av_log(avctx, AV_LOG_WARNING, "Invalid fill bytes: %i\n", + invalid); + break; + default: + while (len--) + skip_bits(gb, 8); + break; + } + } + } + + ret = ff_aac_usac_reset_state(ac, oc); + if (ret < 0) + return ret; + + return 0; +} + +static int decode_usac_scale_factors(AACDecContext *ac, + SingleChannelElement *sce, + GetBitContext *gb, uint8_t global_gain) +{ + IndividualChannelStream *ics = &sce->ics; + + /* Decode all scalefactors. */ + int offset_sf = global_gain; + for (int g = 0; g < ics->num_window_groups; g++) { + for (int sfb = 0; sfb < ics->max_sfb; sfb++) { + /* First coefficient is just the global gain */ + if (!g && !sfb) { + /* The cannonical representation of quantized scalefactors + * in the spec is with 100 subtracted. */ + sce->sfo[0] = offset_sf - 100; + continue; + } + + offset_sf += get_vlc2(gb, ff_vlc_scalefactors, 7, 3) - SCALE_DIFF_ZERO; + if (offset_sf > 255U) { + av_log(ac->avctx, AV_LOG_ERROR, + "Scalefactor (%d) out of range.\n", offset_sf); + return AVERROR_INVALIDDATA; + } + + sce->sfo[g*ics->max_sfb + sfb] = offset_sf - 100; + } + } + + return 0; +} + +/** + * Decode and dequantize arithmetically coded, uniformly quantized value + * + * @param coef array of dequantized, scaled spectral data + * @param sf array of scalefactors or intensity stereo positions + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int decode_spectrum_and_dequant_ac(AACDecContext *s, float coef[1024], + GetBitContext *gb, const float sf[120], + AACArithState *state, int reset, + uint16_t len, uint16_t N) +{ + AACArith ac; + int i, a, b; + uint32_t c; + + int gb_count; + GetBitContext gb2; + + ff_aac_ac_init(&ac, gb); + c = ff_aac_ac_map_process(state, reset, N); + + /* Backup reader for rolling back by 14 bits at the end */ + gb2 = (GetBitContext)*gb; + gb_count = get_bits_count(&gb2); + + for (i = 0; i < len/2; i++) { + /* MSB */ + int lvl, esc_nb, m; + c = ff_aac_ac_get_context(state, c, i, N); + for (lvl=esc_nb=0;;) { + uint32_t pki = ff_aac_ac_get_pk(c + (esc_nb << 17)); + m = ff_aac_ac_decode(&ac, &gb2, ff_aac_ac_msb_cdfs[pki], + FF_ARRAY_ELEMS(ff_aac_ac_msb_cdfs[pki])); + if (m < FF_AAC_AC_ESCAPE) + break; + lvl++; + + /* Cargo-culted value. */ + if (lvl > 23) + return AVERROR(EINVAL); + + if ((esc_nb = lvl) > 7) + esc_nb = 7; + } + + b = m >> 2; + a = m - (b << 2); + + /* ARITH_STOP detection */ + if (!m) { + if (esc_nb) + break; + a = b = 0; + } + + /* LSB */ + for (int l = lvl; l > 0; l--) { + int lsbidx = !a ? 1 : (!b ? 0 : 2); + uint8_t r = ff_aac_ac_decode(&ac, &gb2, ff_aac_ac_lsb_cdfs[lsbidx], + FF_ARRAY_ELEMS(ff_aac_ac_lsb_cdfs[lsbidx])); + a = (a << 1) | (r & 1); + b = (b << 1) | ((r >> 1) & 1); + } + + /* Dequantize coeffs here */ + coef[2*i + 0] = a * cbrt(a); + coef[2*i + 1] = b * cbrt(b); + ff_aac_ac_update_context(state, i, a, b); + } + + if (len > 1) { + /* "Rewind" bitstream back by 14 bits */ + int gb_count2 = get_bits_count(&gb2); + skip_bits(gb, gb_count2 - gb_count - 14); + } else { + *gb = gb2; + } + + ff_aac_ac_finish(state, i, N); + + for (; i < N/2; i++) { + coef[2*i + 0] = 0; + coef[2*i + 1] = 0; + } + + /* Signs */ + for (i = 0; i < len; i++) { + if (coef[i]) { + if (!get_bits1(gb)) /* s */ + coef[i] *= -1; + } + } + + return 0; +} + +static int decode_usac_stereo_cplx(AACDecContext *ac, AACUsacStereo *us, + ChannelElement *cpe, GetBitContext *gb, + int num_window_groups, int indep_flag) +{ + int delta_code_time; + IndividualChannelStream *ics = &cpe->ch[0].ics; + + if (!get_bits1(gb)) { /* cplx_pred_all */ + for (int g = 0; g < num_window_groups; g++) { + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb += SFB_PER_PRED_BAND) { + const uint8_t val = get_bits1(gb); + us->pred_used[g*cpe->max_sfb_ste + sfb] = val; + if ((sfb + 1) < cpe->max_sfb_ste) + us->pred_used[g*cpe->max_sfb_ste + sfb + 1] = val; + } + } + } else { + for (int g = 0; g < num_window_groups; g++) + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) + us->pred_used[g*cpe->max_sfb_ste + sfb] = 1; + } + + us->pred_dir = get_bits1(gb); + us->complex_coef = get_bits1(gb); + + us->use_prev_frame = 0; + if (us->complex_coef && !indep_flag) + us->use_prev_frame = get_bits1(gb); + + delta_code_time = 0; + if (!indep_flag) + delta_code_time = get_bits1(gb); + + /* TODO: shouldn't be needed */ + for (int g = 0; g < num_window_groups; g++) { + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb += SFB_PER_PRED_BAND) { + float last_alpha_q_re = 0; + float last_alpha_q_im = 0; + if (delta_code_time) { + if (g) { + last_alpha_q_re = us->prev_alpha_q_re[(g - 1)*cpe->max_sfb_ste + sfb]; + last_alpha_q_im = us->prev_alpha_q_im[(g - 1)*cpe->max_sfb_ste + sfb]; + } else if ((ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) && + ics->window_sequence[1] == EIGHT_SHORT_SEQUENCE || + ics->window_sequence[1] == EIGHT_SHORT_SEQUENCE) { + /* The spec doesn't explicitly mention this, but it doesn't make + * any other sense otherwise! */ + last_alpha_q_re = us->prev_alpha_q_re[7*cpe->max_sfb_ste + sfb]; + last_alpha_q_im = us->prev_alpha_q_im[7*cpe->max_sfb_ste + sfb]; + } else { + last_alpha_q_re = us->prev_alpha_q_re[g*cpe->max_sfb_ste + sfb]; + last_alpha_q_im = us->prev_alpha_q_im[g*cpe->max_sfb_ste + sfb]; + } + } else { + if (sfb) { + last_alpha_q_re = us->alpha_q_re[g*cpe->max_sfb_ste + sfb - 1]; + last_alpha_q_im = us->alpha_q_im[g*cpe->max_sfb_ste + sfb - 1]; + } + } + + if (us->pred_used[g*cpe->max_sfb_ste + sfb]) { + int val = -get_vlc2(gb, ff_vlc_scalefactors, 7, 3) + 60; + last_alpha_q_re += val * 0.1f; + if (us->complex_coef) { + val = -get_vlc2(gb, ff_vlc_scalefactors, 7, 3) + 60; + last_alpha_q_im += val * 0.1f; + } + us->alpha_q_re[g*cpe->max_sfb_ste + sfb] = last_alpha_q_re; + us->alpha_q_im[g*cpe->max_sfb_ste + sfb] = last_alpha_q_im; + } else { + us->alpha_q_re[g*cpe->max_sfb_ste + sfb] = 0; + us->alpha_q_im[g*cpe->max_sfb_ste + sfb] = 0; + } + + if ((sfb + 1) < cpe->max_sfb_ste) { + us->alpha_q_re[g*cpe->max_sfb_ste + sfb + 1] = + us->alpha_q_re[g*cpe->max_sfb_ste + sfb]; + us->alpha_q_im[g*cpe->max_sfb_ste + sfb + 1] = + us->alpha_q_im[g*cpe->max_sfb_ste + sfb]; + } + } + } + + return 0; +} + +static int setup_sce(AACDecContext *ac, SingleChannelElement *sce, + AACUSACConfig *usac) +{ + AACUsacElemData *ue = &sce->ue; + IndividualChannelStream *ics = &sce->ics; + + /* Setup window parameters */ + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { + if (usac->core_frame_len == 768) { + ics->swb_offset = ff_swb_offset_96[usac->rate_idx]; + ics->num_swb = ff_aac_num_swb_96[usac->rate_idx]; + } else { + ics->swb_offset = ff_swb_offset_128[usac->rate_idx]; + ics->num_swb = ff_aac_num_swb_128[usac->rate_idx]; + } + ics->tns_max_bands = ff_tns_max_bands_128[usac->rate_idx]; + + /* Setup scalefactor grouping. 7 bit mask. */ + ics->num_window_groups = 0; + for (int j = 0; j < 7; j++) { + ics->group_len[j] = 1; + if (ue->scale_factor_grouping & (1 << (6 - j))) + ics->group_len[ics->num_window_groups] += 1; + else + ics->num_window_groups++; + } + + ics->group_len[7] = 1; + ics->num_window_groups++; + ics->num_windows = 8; + } else { + if (usac->core_frame_len == 768) { + ics->swb_offset = ff_swb_offset_768[usac->rate_idx]; + ics->num_swb = ff_aac_num_swb_768[usac->rate_idx]; + } else { + ics->swb_offset = ff_swb_offset_1024[usac->rate_idx]; + ics->num_swb = ff_aac_num_swb_1024[usac->rate_idx]; + } + ics->tns_max_bands = ff_tns_max_bands_1024[usac->rate_idx]; + + ics->group_len[0] = 1; + ics->num_window_groups = 1; + ics->num_windows = 1; + } + + if (ics->max_sfb > ics->num_swb) { + av_log(ac->avctx, AV_LOG_ERROR, + "Number of scalefactor bands in group (%d) " + "exceeds limit (%d).\n", + ics->max_sfb, ics->num_swb); + return AVERROR(EINVAL); + } + + /* Just some defaults for the band types */ + for (int i = 0; i < FF_ARRAY_ELEMS(sce->band_type); i++) + sce->band_type[i] = ESC_BT; + + return 0; +} + +static int decode_usac_stereo_info(AACDecContext *ac, AACUSACConfig *usac, + AACUsacElemConfig *ec, ChannelElement *cpe, + GetBitContext *gb, int indep_flag) +{ + int ret, tns_active; + + AACUsacStereo *us = &cpe->us; + SingleChannelElement *sce1 = &cpe->ch[0]; + SingleChannelElement *sce2 = &cpe->ch[1]; + IndividualChannelStream *ics1 = &sce1->ics; + IndividualChannelStream *ics2 = &sce2->ics; + AACUsacElemData *ue1 = &sce1->ue; + AACUsacElemData *ue2 = &sce2->ue; + + us->common_window = 0; + us->common_tw = 0; + + if (!(!ue1->core_mode && !ue2->core_mode)) + return 0; + + tns_active = get_bits1(gb); + us->common_window = get_bits1(gb); + + if (us->common_window) { + /* ics_info() */ + ics1->window_sequence[1] = ics1->window_sequence[0]; + ics2->window_sequence[1] = ics2->window_sequence[0]; + ics1->window_sequence[0] = ics2->window_sequence[0] = get_bits(gb, 2); + + ics1->use_kb_window[1] = ics1->use_kb_window[0]; + ics2->use_kb_window[1] = ics2->use_kb_window[0]; + ics1->use_kb_window[0] = ics2->use_kb_window[0] = get_bits1(gb); + + if (ics1->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { + ics1->max_sfb = ics2->max_sfb = get_bits(gb, 4); + ue1->scale_factor_grouping = ue2->scale_factor_grouping = get_bits(gb, 7); + } else { + ics1->max_sfb = ics2->max_sfb = get_bits(gb, 6); + } + + if (!get_bits1(gb)) { /* common_max_sfb */ + if (ics2->window_sequence[0] == EIGHT_SHORT_SEQUENCE) + ics2->max_sfb = get_bits(gb, 4); + else + ics2->max_sfb = get_bits(gb, 6); + } + + ret = setup_sce(ac, sce1, usac); + if (ret < 0) + return ret; + + ret = setup_sce(ac, sce2, usac); + if (ret < 0) + return ret; + + cpe->max_sfb_ste = FFMAX(ics1->max_sfb, ics2->max_sfb); + + us->ms_mask_mode = get_bits(gb, 2); /* ms_mask_present */ + memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask)); + if (us->ms_mask_mode == 1) { + for (int g = 0; g < ics1->num_window_groups; g++) + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) + cpe->ms_mask[g*cpe->max_sfb_ste + sfb] = get_bits1(gb); + } else if (us->ms_mask_mode == 2) { + memset(cpe->ms_mask, 0xFF, sizeof(cpe->ms_mask)); + } else if ((us->ms_mask_mode == 3) && !ec->stereo_config_index) { + ret = decode_usac_stereo_cplx(ac, us, cpe, gb, + ics1->num_window_groups, indep_flag); + if (ret < 0) + return ret; + } + } + + if (ec->tw_mdct) { + us->common_tw = get_bits1(gb); + avpriv_report_missing_feature(ac->avctx, + "AAC USAC timewarping"); + return AVERROR_PATCHWELCOME; + } + + sce1->tns.present = sce2->tns.present = 0; + if (tns_active) { + av_unused int tns_on_lr; + int common_tns = 0; + if (us->common_window) + common_tns = get_bits1(gb); + + tns_on_lr = get_bits1(gb); + if (common_tns) { + ret = ff_aac_decode_tns(ac, &sce1->tns, gb, ics1); + if (ret < 0) + return ret; + memcpy(&sce2->tns, &sce1->tns, sizeof(sce1->tns)); + sce2->tns.present = 0; + sce1->tns.present = 0; + } else { + if (get_bits1(gb)) { + sce2->tns.present = 1; + sce1->tns.present = 1; + } else { + sce2->tns.present = get_bits1(gb); + sce1->tns.present = !sce2->tns.present; + } + } + } + + return 0; +} + +/* 7.2.4 Generation of random signs for spectral noise filling + * This function is exactly defined, though we've helped the definition + * along with being slightly faster. */ +static inline float noise_random_sign(unsigned int *seed) +{ + unsigned int new_seed = *seed = ((*seed) * 69069) + 5; + if (((new_seed) & 0x10000) > 0) + return -1.f; + return +1.f; +} + +static void apply_noise_fill(AACDecContext *ac, SingleChannelElement *sce, + AACUsacElemData *ue) +{ + float *coef; + IndividualChannelStream *ics = &sce->ics; + + float noise_val = pow(2, (ue->noise.level - 14)/3); + int noise_offset = ue->noise.offset - 16; + int band_off; + + band_off = ff_usac_noise_fill_start_offset[ac->oc[1].m4ac.frame_length_short] + [ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE]; + + coef = sce->coeffs; + for (int g = 0; g < ics->num_window_groups; g++) { + unsigned g_len = ics->group_len[g]; + + for (int sfb = 0; sfb < ics->max_sfb; sfb++) { + float *cb = coef + ics->swb_offset[sfb]; + int cb_len = ics->swb_offset[sfb + 1] - ics->swb_offset[sfb]; + int band_quantized_to_zero = 1; + + if (ics->swb_offset[sfb] < band_off) + continue; + + for (int group = 0; group < (unsigned)g_len; group++, cb += 128) { + for (int z = 0; z < cb_len; z++) { + if (cb[z] == 0) + cb[z] = noise_random_sign(&sce->ue.noise.seed) * noise_val; + else + band_quantized_to_zero = 0; + } + } + + if (band_quantized_to_zero) + sce->sf[g*ics->max_sfb + sfb] += noise_offset; + } + coef += g_len << 7; + } +} + +static void spectrum_scale(AACDecContext *ac, SingleChannelElement *sce, + AACUsacElemData *ue) +{ + IndividualChannelStream *ics = &sce->ics; + float *coef; + + /* Synthesise noise */ + if (ue->noise.level) + apply_noise_fill(ac, sce, ue); + + /* Apply scalefactors */ + coef = sce->coeffs; + for (int g = 0; g < ics->num_window_groups; g++) { + unsigned g_len = ics->group_len[g]; + + for (int sfb = 0; sfb < ics->max_sfb; sfb++) { + float *cb = coef + ics->swb_offset[sfb]; + int cb_len = ics->swb_offset[sfb + 1] - ics->swb_offset[sfb]; + float sf = sce->sf[g*ics->max_sfb + sfb]; + + for (int group = 0; group < (unsigned)g_len; group++, cb += 128) + ac->fdsp->vector_fmul_scalar(cb, cb, sf, cb_len); + } + coef += g_len << 7; + } +} + +static void complex_stereo_downmix_prev(AACDecContext *ac, ChannelElement *cpe, + float *dmix_re) +{ + IndividualChannelStream *ics = &cpe->ch[0].ics; + int sign = !cpe->us.pred_dir ? +1 : -1; + float *coef1 = cpe->ch[0].coeffs; + float *coef2 = cpe->ch[1].coeffs; + + for (int g = 0; g < ics->num_window_groups; g++) { + unsigned g_len = ics->group_len[g]; + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) { + int off = ics->swb_offset[sfb]; + int cb_len = ics->swb_offset[sfb + 1] - off; + + float *c1 = coef1 + off; + float *c2 = coef2 + off; + float *dm = dmix_re + off; + + for (int group = 0; group < (unsigned)g_len; + group++, c1 += 128, c2 += 128, dm += 128) { + for (int z = 0; z < cb_len; z++) + dm[z] = 0.5*(c1[z] + sign*c2[z]); + } + } + + coef1 += g_len << 7; + coef2 += g_len << 7; + dmix_re += g_len << 7; + } +} + +static void complex_stereo_downmix_cur(AACDecContext *ac, ChannelElement *cpe, + float *dmix_re) +{ + AACUsacStereo *us = &cpe->us; + IndividualChannelStream *ics = &cpe->ch[0].ics; + int sign = !cpe->us.pred_dir ? +1 : -1; + float *coef1 = cpe->ch[0].coeffs; + float *coef2 = cpe->ch[1].coeffs; + + for (int g = 0; g < ics->num_window_groups; g++) { + unsigned g_len = ics->group_len[g]; + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) { + int off = ics->swb_offset[sfb]; + int cb_len = ics->swb_offset[sfb + 1] - off; + + float *c1 = coef1 + off; + float *c2 = coef2 + off; + float *dm = dmix_re + off; + + if (us->pred_used[g*cpe->max_sfb_ste + sfb]) { + for (int group = 0; group < (unsigned)g_len; + group++, c1 += 128, c2 += 128, dm += 128) { + for (int z = 0; z < cb_len; z++) + dm[z] = 0.5*(c1[z] + sign*c2[z]); + } + } else { + for (int group = 0; group < (unsigned)g_len; + group++, c1 += 128, c2 += 128, dm += 128) { + for (int z = 0; z < cb_len; z++) + dm[z] = c1[z]; + } + } + } + + coef1 += g_len << 7; + coef2 += g_len << 7; + dmix_re += g_len << 7; + } +} + +static void complex_stereo_interpolate_imag(float *im, float *re, const float f[6], + int len, int factor_even, int factor_odd) +{ + int i = 0; + float s; + + s = f[6]*re[2] + f[5]*re[1] + f[4]*re[0] + + f[3]*re[0] + + f[2]*re[1] + f[1]*re[2] + f[0]*re[3]; + im[i] += s*factor_even; + + i = 1; + s = f[6]*re[1] + f[5]*re[0] + f[4]*re[0] + + f[3]*re[1] + + f[2]*re[2] + f[1]*re[3] + f[0]*re[4]; + im[i] += s*factor_odd; + + i = 2; + s = f[6]*re[0] + f[5]*re[0] + f[4]*re[1] + + f[3]*re[2] + + f[2]*re[3] + f[1]*re[4] + f[0]*re[5]; + + im[i] += s*factor_even; + for (i = 3; i < len - 4; i += 2) { + s = f[6]*re[i-3] + f[5]*re[i-2] + f[4]*re[i-1] + + f[3]*re[i] + + f[2]*re[i+1] + f[1]*re[i+2] + f[0]*re[i+3]; + im[i+0] += s*factor_odd; + + s = f[6]*re[i-2] + f[5]*re[i-1] + f[4]*re[i] + + f[3]*re[i+1] + + f[2]*re[i+2] + f[1]*re[i+3] + f[0]*re[i+4]; + im[i+1] += s*factor_even; + } + + i = len - 3; + s = f[6]*re[i-3] + f[5]*re[i-2] + f[4]*re[i-1] + + f[3]*re[i] + + f[2]*re[i+1] + f[1]*re[i+2] + f[0]*re[i+2]; + im[i] += s*factor_odd; + + i = len - 2; + s = f[6]*re[i-3] + f[5]*re[i-2] + f[4]*re[i-1] + + f[3]*re[i] + + f[2]*re[i+1] + f[1]*re[i+1] + f[0]*re[i]; + im[i] += s*factor_even; + + i = len - 1; + s = f[6]*re[i-3] + f[5]*re[i-2] + f[4]*re[i-1] + + f[3]*re[i] + + f[2]*re[i] + f[1]*re[i-1] + f[0]*re[i-2]; + im[i] += s*factor_odd; +} + +static void apply_complex_stereo(AACDecContext *ac, ChannelElement *cpe) +{ + AACUsacStereo *us = &cpe->us; + IndividualChannelStream *ics = &cpe->ch[0].ics; + float *coef1 = cpe->ch[0].coeffs; + float *coef2 = cpe->ch[1].coeffs; + float *dmix_im = us->dmix_im; + + for (int g = 0; g < ics->num_window_groups; g++) { + unsigned g_len = ics->group_len[g]; + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) { + int off = ics->swb_offset[sfb]; + int cb_len = ics->swb_offset[sfb + 1] - off; + + float *c1 = coef1 + off; + float *c2 = coef2 + off; + float *dm_im = dmix_im + off; + float alpha_re = us->alpha_q_re[g*cpe->max_sfb_ste + sfb]; + float alpha_im = us->alpha_q_im[g*cpe->max_sfb_ste + sfb]; + + if (!us->pred_used[g*cpe->max_sfb_ste + sfb]) + continue; + + if (!cpe->us.pred_dir) { + for (int group = 0; group < (unsigned)g_len; + group++, c1 += 128, c2 += 128, dm_im += 128) { + for (int z = 0; z < cb_len; z++) { + float side; + side = c2[z] - alpha_re*c1[z] - alpha_im*dm_im[z]; + c2[z] = c1[z] - side; + c1[z] = c1[z] + side; + } + } + } else { + for (int group = 0; group < (unsigned)g_len; + group++, c1 += 128, c2 += 128, dm_im += 128) { + for (int z = 0; z < cb_len; z++) { + float mid; + mid = c2[z] - alpha_re*c1[z] - alpha_im*dm_im[z]; + c2[z] = mid - c1[z]; + c1[z] = mid + c1[z]; + } + } + } + } + + coef1 += g_len << 7; + coef2 += g_len << 7; + dmix_im += g_len << 7; + } +} + +static const float *complex_stereo_get_filter(ChannelElement *cpe, int is_prev) +{ + int win, shape; + if (!is_prev) { + switch (cpe->ch[0].ics.window_sequence[0]) { + default: + case ONLY_LONG_SEQUENCE: + case EIGHT_SHORT_SEQUENCE: + win = 0; + break; + case LONG_START_SEQUENCE: + win = 1; + break; + case LONG_STOP_SEQUENCE: + win = 2; + break; + } + + if (cpe->ch[0].ics.use_kb_window[0] == 0 && + cpe->ch[0].ics.use_kb_window[1] == 0) + shape = 0; + else if (cpe->ch[0].ics.use_kb_window[0] == 1 && + cpe->ch[0].ics.use_kb_window[1] == 1) + shape = 1; + else if (cpe->ch[0].ics.use_kb_window[0] == 0 && + cpe->ch[0].ics.use_kb_window[1] == 1) + shape = 2; + else if (cpe->ch[0].ics.use_kb_window[0] == 1 && + cpe->ch[0].ics.use_kb_window[1] == 0) + shape = 3; + else + shape = 3; + } else { + win = cpe->ch[0].ics.window_sequence[0] == LONG_STOP_SEQUENCE; + shape = cpe->ch[0].ics.use_kb_window[1]; + } + + return ff_aac_usac_mdst_filt_cur[win][shape]; +} + +static void spectrum_decode(AACDecContext *ac, AACUSACConfig *usac, + ChannelElement *cpe, int nb_channels) +{ + AACUsacStereo *us = &cpe->us; + + for (int ch = 0; ch < nb_channels; ch++) { + SingleChannelElement *sce = &cpe->ch[ch]; + AACUsacElemData *ue = &sce->ue; + + spectrum_scale(ac, sce, ue); + } + + if (nb_channels > 1 && us->common_window) { + if (us->ms_mask_mode == 3) { + const float *filt; + complex_stereo_downmix_cur(ac, cpe, us->dmix_re); + complex_stereo_downmix_prev(ac, cpe, us->prev_dmix_re); + + filt = complex_stereo_get_filter(cpe, 0); + complex_stereo_interpolate_imag(us->dmix_im, us->dmix_re, filt, + usac->core_frame_len, 1, 1); + if (us->use_prev_frame) { + filt = complex_stereo_get_filter(cpe, 1); + complex_stereo_interpolate_imag(us->dmix_im, us->prev_dmix_re, filt, + usac->core_frame_len, -1, 1); + } + + apply_complex_stereo(ac, cpe); + } else if (us->ms_mask_mode > 0) { + ac->dsp.apply_mid_side_stereo(ac, cpe); + } + } + + /* Save coefficients and alpha values for prediction reasons */ + if (nb_channels > 1) { + AACUsacStereo *us = &cpe->us; + for (int ch = 0; ch < nb_channels; ch++) { + SingleChannelElement *sce = &cpe->ch[ch]; + memcpy(sce->prev_coeffs, sce->coeffs, sizeof(sce->coeffs)); + } + memcpy(us->prev_alpha_q_re, us->alpha_q_re, sizeof(us->alpha_q_re)); + memcpy(us->prev_alpha_q_im, us->alpha_q_im, sizeof(us->alpha_q_im)); + } + + for (int ch = 0; ch < nb_channels; ch++) { + SingleChannelElement *sce = &cpe->ch[ch]; + + /* Apply TNS */ + if (sce->tns.present) + ac->dsp.apply_tns(sce->coeffs, &sce->tns, &sce->ics, 1); + + ac->oc[1].m4ac.frame_length_short ? ac->dsp.imdct_and_windowing_768(ac, sce) : + ac->dsp.imdct_and_windowing(ac, sce); + } +} + +static int decode_usac_core_coder(AACDecContext *ac, AACUSACConfig *usac, + AACUsacElemConfig *ec, ChannelElement *che, + GetBitContext *gb, int indep_flag, int nb_channels) +{ + int ret; + int arith_reset_flag; + AACUsacStereo *us = &che->us; + + /* Local symbols */ + uint8_t global_gain; + + us->common_window = 0; + che->ch[0].tns.present = che->ch[1].tns.present = 0; + + for (int ch = 0; ch < nb_channels; ch++) { + SingleChannelElement *sce = &che->ch[ch]; + AACUsacElemData *ue = &sce->ue; + + ue->core_mode = get_bits1(gb); + } + + if (nb_channels == 2) { + ret = decode_usac_stereo_info(ac, usac, ec, che, gb, indep_flag); + if (ret) + return ret; + } + + for (int ch = 0; ch < nb_channels; ch++) { + SingleChannelElement *sce = &che->ch[ch]; + IndividualChannelStream *ics = &sce->ics; + AACUsacElemData *ue = &sce->ue; + + if (ue->core_mode) { /* lpd_channel_stream */ + ret = ff_aac_ldp_parse_channel_stream(ac, usac, ue, gb); + if (ret < 0) + return ret; + } + + if ((nb_channels == 1) || + (che->ch[0].ue.core_mode != che->ch[1].ue.core_mode)) + sce->tns.present = get_bits1(gb); + + /* fd_channel_stream */ + global_gain = get_bits(gb, 8); + + ue->noise.level = 0; + if (ec->noise_fill) { + ue->noise.level = get_bits(gb, 3); + ue->noise.offset = get_bits(gb, 5); + } + + if (!us->common_window) { + /* ics_info() */ + ics->window_sequence[1] = ics->window_sequence[0]; + ics->window_sequence[0] = get_bits(gb, 2); + ics->use_kb_window[1] = ics->use_kb_window[0]; + ics->use_kb_window[0] = get_bits1(gb); + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { + ics->max_sfb = get_bits(gb, 4); + ue->scale_factor_grouping = get_bits(gb, 7); + } else { + ics->max_sfb = get_bits(gb, 6); + } + + ret = setup_sce(ac, sce, usac); + if (ret < 0) + return ret; + } + + if (ec->tw_mdct && !us->common_tw) { + /* tw_data() */ + if (get_bits1(gb)) { /* tw_data_present */ + /* Time warping is not supported in baseline profile streams. */ + avpriv_report_missing_feature(ac->avctx, + "AAC USAC timewarping"); + return AVERROR_PATCHWELCOME; + } + } + + ret = decode_usac_scale_factors(ac, sce, gb, global_gain); + if (ret < 0) + return ret; + + ac->dsp.dequant_scalefactors(sce); + + if (sce->tns.present) { + ret = ff_aac_decode_tns(ac, &sce->tns, gb, ics); + if (ret < 0) + return ret; + } + + /* ac_spectral_data */ + arith_reset_flag = indep_flag; + if (!arith_reset_flag) + arith_reset_flag = get_bits1(gb); + + /* Decode coeffs */ + memset(&sce->coeffs[0], 0, 1024*sizeof(float)); + for (int win = 0; win < ics->num_windows; win++) { + int lg = ics->swb_offset[ics->max_sfb]; + int N; + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) + N = usac->core_frame_len / 8; + else + N = usac->core_frame_len; + + ret = decode_spectrum_and_dequant_ac(ac, sce->coeffs + win*128, gb, + sce->sf, &ue->ac, + arith_reset_flag && (win == 0), + lg, N); + if (ret < 0) + return ret; + } + + if (get_bits1(gb)) { /* fac_data_present */ + const uint16_t len_8 = usac->core_frame_len / 8; + const uint16_t len_16 = usac->core_frame_len / 16; + const uint16_t fac_len = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? len_8 : len_16; + ret = ff_aac_parse_fac_data(ue, gb, 1, fac_len); + if (ret < 0) + return ret; + } + } + + spectrum_decode(ac, usac, che, nb_channels); + + return 0; +} + +static int parse_audio_preroll(AACDecContext *ac, GetBitContext *gb) +{ + int ret = 0; + GetBitContext gbc; + OutputConfiguration *oc = &ac->oc[1]; + MPEG4AudioConfig *m4ac = &oc->m4ac; + MPEG4AudioConfig m4ac_bak = oc->m4ac; + uint8_t temp_data[512]; + uint8_t *tmp_buf = temp_data; + size_t tmp_buf_size = sizeof(temp_data); + + av_unused int crossfade; + int num_preroll_frames; + + int config_len = get_escaped_value(gb, 4, 4, 8); + + /* Implementations are free to pad the config to any length, so use a + * different reader for this. */ + gbc = *gb; + ret = ff_aac_usac_config_decode(ac, ac->avctx, &gbc, oc, m4ac->chan_config); + if (ret < 0) { + *m4ac = m4ac_bak; + return ret; + } else { + ac->oc[1].m4ac.chan_config = 0; + } + + /* 7.18.3.3 Bitrate adaption + * If configuration didn't change after applying preroll, continue + * without decoding it. */ + if (!memcmp(m4ac, &m4ac_bak, sizeof(m4ac_bak))) + return 0; + + skip_bits_long(gb, config_len*8); + + crossfade = get_bits1(gb); /* applyCrossfade */ + skip_bits1(gb); /* reserved */ + num_preroll_frames = get_escaped_value(gb, 2, 4, 0); /* numPreRollFrames */ + + for (int i = 0; i < num_preroll_frames; i++) { + int got_frame_ptr = 0; + int au_len = get_escaped_value(gb, 16, 16, 0); + + if (au_len*8 > tmp_buf_size) { + uint8_t *tmp2; + tmp_buf = tmp_buf == temp_data ? NULL : tmp_buf; + tmp2 = realloc(tmp_buf, au_len*8); + if (!tmp2) { + if (tmp_buf != temp_data) + av_free(tmp_buf); + return AVERROR(ENOMEM); + } + tmp_buf = tmp2; + } + + /* Byte alignment is not guaranteed. */ + for (int i = 0; i < au_len; i++) + tmp_buf[i] = get_bits(gb, 8); + + ret = init_get_bits8(&gbc, tmp_buf, au_len); + if (ret < 0) + break; + + ret = ff_aac_usac_decode_frame(ac->avctx, ac, &gbc, &got_frame_ptr); + if (ret < 0) + break; + } + + if (tmp_buf != temp_data) + av_free(tmp_buf); + + return 0; +} + +static int parse_ext_ele(AACDecContext *ac, AACUsacElemConfig *e, + GetBitContext *gb) +{ + uint8_t *tmp; + uint8_t pl_frag_start = 1; + uint8_t pl_frag_end = 1; + uint32_t len; + + if (!get_bits1(gb)) /* usacExtElementPresent */ + return 0; + + if (get_bits1(gb)) { /* usacExtElementUseDefaultLength */ + len = e->ext.default_len; + } else { + len = get_bits(gb, 8); /* usacExtElementPayloadLength */ + if (len == 255) + len += get_bits(gb, 16) - 2; + } + + if (!len) + return 0; + + if (e->ext.payload_frag) { + pl_frag_start = get_bits1(gb); /* usacExtElementStart */ + pl_frag_end = get_bits1(gb); /* usacExtElementStop */ + } + + if (pl_frag_start) + e->ext.pl_data_offset = 0; + + /* If an extension starts and ends this packet, we can directly use it */ + if (!(pl_frag_start && pl_frag_end)) { + tmp = av_realloc(e->ext.pl_data, e->ext.pl_data_offset + len); + if (!tmp) { + av_free(e->ext.pl_data); + return AVERROR(ENOMEM); + } + e->ext.pl_data = tmp; + + /* Readout data to a buffer */ + for (int i = 0; i < len; i++) + e->ext.pl_data[e->ext.pl_data_offset + i] = get_bits(gb, 8); + } + + e->ext.pl_data_offset += len; + + if (pl_frag_end) { + int ret = 0; + int start_bits = get_bits_count(gb); + const int pl_len = e->ext.pl_data_offset; + GetBitContext *gb2 = gb; + GetBitContext gbc; + if (!(pl_frag_start && pl_frag_end)) { + ret = init_get_bits8(&gbc, e->ext.pl_data, pl_len); + if (ret < 0) + return ret; + + gb2 = &gbc; + } + + switch (e->ext.type) { + case ID_EXT_ELE_FILL: + /* Filler elements have no usable payload */ + break; + case ID_EXT_ELE_AUDIOPREROLL: + ret = parse_audio_preroll(ac, gb2); + break; + default: + /* This should never happen */ + av_assert0(0); + } + av_freep(&e->ext.pl_data); + if (ret < 0) + return ret; + + skip_bits_long(gb, pl_len*8 - (get_bits_count(gb) - start_bits)); + } + + return 0; +} + +int ff_aac_usac_decode_frame(AVCodecContext *avctx, AACDecContext *ac, + GetBitContext *gb, int *got_frame_ptr) +{ + int ret, nb_ch_el, is_dmono = 0; + int indep_flag, samples = 0; + int audio_found = 0, sce_count = 0; + AVFrame *frame = ac->frame; + + ff_aac_output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags, + ac->oc[1].status, 0); + + indep_flag = get_bits1(gb); + + nb_ch_el = 0; + for (int i = 0; i < ac->oc[1].usac.nb_elems; i++) { + AACUsacElemConfig *e = &ac->oc[1].usac.elems[i]; + ChannelElement *che; + + switch (e->type) { + case ID_USAC_LFE: + /* Fallthrough */ + case ID_USAC_SCE: + che = ff_aac_get_che(ac, TYPE_SCE, nb_ch_el++); + if (!che) { + av_log(ac->avctx, AV_LOG_ERROR, + "channel element %d.%d is not allocated\n", + TYPE_SCE, nb_ch_el - 1); + return AVERROR_INVALIDDATA; + } + + ret = decode_usac_core_coder(ac, &ac->oc[1].usac, e, che, gb, + indep_flag, 1); + if (ret < 0) + return ret; + + sce_count++; + audio_found = 1; + che->present = 1; + samples = ac->oc[1].m4ac.frame_length_short ? 768 : 1024; + break; + case ID_USAC_CPE: + che = ff_aac_get_che(ac, TYPE_CPE, nb_ch_el++); + if (!che) { + av_log(ac->avctx, AV_LOG_ERROR, + "channel element %d.%d is not allocated\n", + TYPE_SCE, nb_ch_el - 1); + return AVERROR_INVALIDDATA; + } + + ret = decode_usac_core_coder(ac, &ac->oc[1].usac, e, che, gb, + indep_flag, 2); + if (ret < 0) + return ret; + + audio_found = 1; + che->present = 1; + samples = ac->oc[1].m4ac.frame_length_short ? 768 : 1024; + break; + case ID_USAC_EXT: + ret = parse_ext_ele(ac, e, gb); + if (ret < 0) + return ret; + break; + } + } + + if (ac->oc[1].status && audio_found) { + avctx->sample_rate = ac->oc[1].m4ac.sample_rate; + avctx->frame_size = samples; + ac->oc[1].status = OC_LOCKED; + } + + if (!frame->data[0] && samples) { + av_log(avctx, AV_LOG_ERROR, "no frame data found\n"); + return AVERROR_INVALIDDATA; + } + + if (samples) { + frame->nb_samples = samples; + frame->sample_rate = avctx->sample_rate; + frame->flags = indep_flag ? AV_FRAME_FLAG_KEY : 0x0; + *got_frame_ptr = 1; + } else { + av_frame_unref(ac->frame); + frame->flags = indep_flag ? AV_FRAME_FLAG_KEY : 0x0; + *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; +} diff --git a/libavcodec/aac/aacdec_usac.h b/libavcodec/aac/aacdec_usac.h new file mode 100644 index 0000000000..4116a2073a --- /dev/null +++ b/libavcodec/aac/aacdec_usac.h @@ -0,0 +1,37 @@ +/* + * Copyright (c) 2024 Lynne + * + * 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 + */ + +#ifndef AVCODEC_AAC_AACDEC_USAC_H +#define AVCODEC_AAC_AACDEC_USAC_H + +#include "aacdec.h" + +#include "libavcodec/get_bits.h" + +int ff_aac_usac_config_decode(AACDecContext *ac, AVCodecContext *avctx, + GetBitContext *gb, OutputConfiguration *oc, + int channel_config); + +int ff_aac_usac_reset_state(AACDecContext *ac, OutputConfiguration *oc); + +int ff_aac_usac_decode_frame(AVCodecContext *avctx, AACDecContext *ac, + GetBitContext *gb, int *got_frame_ptr); + +#endif /* AVCODEC_AAC_AACDEC_USAC_H */ diff --git a/libavcodec/aactab.c b/libavcodec/aactab.c index 18afa69bad..7b040531aa 100644 --- a/libavcodec/aactab.c +++ b/libavcodec/aactab.c @@ -1998,6 +1998,11 @@ const uint8_t ff_tns_max_bands_128[] = { }; // @} +const uint8_t ff_usac_noise_fill_start_offset[2][2] = { + { 160, 20 }, + { 120, 15 }, +}; + const DECLARE_ALIGNED(32, float, ff_aac_eld_window_512)[1920] = { 0.00338834, 0.00567745, 0.00847677, 0.01172641, 0.01532555, 0.01917664, 0.02318809, 0.02729259, @@ -3895,3 +3900,40 @@ DECLARE_ALIGNED(16, const float, ff_aac_deemph_weights)[16] = { 0, USAC_EMPH_COEFF, }; + +const int ff_aac_usac_samplerate[32] = { + 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, + 16000, 12000, 11025, 8000, 7350, -1, -1, 57600, + 51200, 40000, 38400, 34150, 28800, 25600, 20000, 19200, + 17075, 14400, 12800, 9600, -1, -1, -1, -1, +}; + +/* Window type (only long+eight, start/stop/stopstart), sine+sine, kbd+kbd, sine+kbd, kbd+sine */ +const float ff_aac_usac_mdst_filt_cur[4 /* Window */][4 /* Shape */][7] = +{ + { { 0.000000, 0.000000, 0.500000, 0.000000, -0.500000, 0.000000, 0.000000 }, + { 0.091497, 0.000000, 0.581427, 0.000000, -0.581427, 0.000000, -0.091497 }, + { 0.045748, 0.057238, 0.540714, 0.000000, -0.540714, -0.057238, -0.045748 }, + { 0.045748, -0.057238, 0.540714, 0.000000, -0.540714, 0.057238, -0.045748 } }, + { { 0.102658, 0.103791, 0.567149, 0.000000, -0.567149, -0.103791, -0.102658 }, + { 0.150512, 0.047969, 0.608574, 0.000000, -0.608574, -0.047969, -0.150512 }, + { 0.104763, 0.105207, 0.567861, 0.000000, -0.567861, -0.105207, -0.104763 }, + { 0.148406, 0.046553, 0.607863, 0.000000, -0.607863, -0.046553, -0.148406 } }, + { { 0.102658, -0.103791, 0.567149, 0.000000, -0.567149, 0.103791, -0.102658 }, + { 0.150512, -0.047969, 0.608574, 0.000000, -0.608574, 0.047969, -0.150512 }, + { 0.148406, -0.046553, 0.607863, 0.000000, -0.607863, 0.046553, -0.148406 }, + { 0.104763, -0.105207, 0.567861, 0.000000, -0.567861, 0.105207, -0.104763 } }, + { { 0.205316, 0.000000, 0.634298, 0.000000, -0.634298, 0.000000, -0.205316 }, + { 0.209526, 0.000000, 0.635722, 0.000000, -0.635722, 0.000000, -0.209526 }, + { 0.207421, 0.001416, 0.635010, 0.000000, -0.635010, -0.001416, -0.207421 }, + { 0.207421, -0.001416, 0.635010, 0.000000, -0.635010, 0.001416, -0.207421 } } +}; + +/* Window type (everything/longstop+stopstart), sine or kbd */ +const float ff_aac_usac_mdst_filt_prev[2 /* Window */][2 /* sine/kbd */][7] = +{ + { { 0.000000, 0.106103, 0.250000, 0.318310, 0.250000, 0.106103, 0.000000 }, + { 0.059509, 0.123714, 0.186579, 0.213077, 0.186579, 0.123714, 0.059509 } }, + { { 0.038498, 0.039212, 0.039645, 0.039790, 0.039645, 0.039212, 0.038498 }, + { 0.026142, 0.026413, 0.026577, 0.026631, 0.026577, 0.026413, 0.026142 } } +}; diff --git a/libavcodec/aactab.h b/libavcodec/aactab.h index 481fc57d93..8dbb2098c5 100644 --- a/libavcodec/aactab.h +++ b/libavcodec/aactab.h @@ -115,4 +115,14 @@ extern const uint8_t ff_tns_max_bands_512 [13]; extern const uint8_t ff_tns_max_bands_480 [13]; extern const uint8_t ff_tns_max_bands_128 [13]; +/* [x][y], x == 1 -> frame len is 768 frames, y == 1 -> is eight_short */ +extern const uint8_t ff_usac_noise_fill_start_offset[2][2]; + +extern const int ff_aac_usac_samplerate[32]; + +/* Window type (only long+eight, start/stop/stopstart), sine+sine, kbd+kbd, sine+kbd, kbd+sine */ +extern const float ff_aac_usac_mdst_filt_cur[4 /* Window */][4 /* Shape */][7]; +/* Window type (everything/longstop+stopstart), sine or kbd */ +extern const float ff_aac_usac_mdst_filt_prev[2 /* Window */][2 /* sine/kbd */][7]; + #endif /* AVCODEC_AACTAB_H */