| /* |
| * AAC decoder |
| * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) |
| * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) |
| * |
| * AAC LATM decoder |
| * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz> |
| * Copyright (c) 2010 Janne Grunau <janne-ffmpeg@jannau.net> |
| * |
| * 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 ) |
| */ |
| |
| /* |
| * 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 |
| * |
| * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication. |
| * - HE AAC v2 comprises LC AAC with Spectral Band Replication and |
| Parametric Stereo. |
| */ |
| |
| |
| #include "avcodec.h" |
| #include "internal.h" |
| #include "get_bits.h" |
| #include "dsputil.h" |
| #include "fft.h" |
| #include "fmtconvert.h" |
| #include "lpc.h" |
| #include "kbdwin.h" |
| #include "sinewin.h" |
| |
| #include "aac.h" |
| #include "aactab.h" |
| #include "aacdectab.h" |
| #include "cbrt_tablegen.h" |
| #include "sbr.h" |
| #include "aacsbr.h" |
| #include "mpeg4audio.h" |
| #include "aacadtsdec.h" |
| |
| #include <assert.h> |
| #include <errno.h> |
| #include <math.h> |
| #include <string.h> |
| |
| #if ARCH_ARM |
| # include "arm/aac.h" |
| #endif |
| |
| union float754 { |
| float f; |
| uint32_t i; |
| }; |
| |
| static VLC vlc_scalefactors; |
| static VLC vlc_spectral[11]; |
| |
| static const char overread_err[] = "Input buffer exhausted before END element found\n"; |
| |
| 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->m4ac.chan_config) { |
| return ac->tag_che_map[type][elem_id]; |
| } |
| // For indexed channel configurations map the channels solely based on position. |
| switch (ac->m4ac.chan_config) { |
| case 7: |
| if (ac->tags_mapped == 3 && type == TYPE_CPE) { |
| ac->tags_mapped++; |
| return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2]; |
| } |
| case 6: |
| /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1] |
| instead of SCE[0] CPE[0] CPE[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->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) { |
| ac->tags_mapped++; |
| return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0]; |
| } |
| case 5: |
| if (ac->tags_mapped == 2 && type == TYPE_CPE) { |
| ac->tags_mapped++; |
| return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1]; |
| } |
| case 4: |
| if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) { |
| ac->tags_mapped++; |
| return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; |
| } |
| case 3: |
| case 2: |
| if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) { |
| ac->tags_mapped++; |
| return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0]; |
| } else if (ac->m4ac.chan_config == 2) { |
| return NULL; |
| } |
| case 1: |
| if (!ac->tags_mapped && type == TYPE_SCE) { |
| ac->tags_mapped++; |
| return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0]; |
| } |
| default: |
| return NULL; |
| } |
| } |
| |
| /** |
| * Check for the channel element in the current channel position configuration. |
| * If it exists, make sure the appropriate element is allocated and map the |
| * channel order to match the internal FFmpeg channel layout. |
| * |
| * @param che_pos current channel position configuration |
| * @param type channel element type |
| * @param id channel element id |
| * @param channels count of the number of channels in the configuration |
| * |
| * @return Returns error status. 0 - OK, !0 - error |
| */ |
| static av_cold int che_configure(AACContext *ac, |
| enum ChannelPosition che_pos[4][MAX_ELEM_ID], |
| int type, int id, int *channels) |
| { |
| if (che_pos[type][id]) { |
| if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement)))) |
| return AVERROR(ENOMEM); |
| ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr); |
| if (type != TYPE_CCE) { |
| ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret; |
| if (type == TYPE_CPE || |
| (type == TYPE_SCE && ac->m4ac.ps == 1)) { |
| ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret; |
| } |
| } |
| } else { |
| if (ac->che[type][id]) |
| ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr); |
| av_freep(&ac->che[type][id]); |
| } |
| return 0; |
| } |
| |
| /** |
| * Configure output channel order based on the current program configuration element. |
| * |
| * @param che_pos current channel position configuration |
| * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. |
| * |
| * @return Returns error status. 0 - OK, !0 - error |
| */ |
| static av_cold int output_configure(AACContext *ac, |
| enum ChannelPosition che_pos[4][MAX_ELEM_ID], |
| enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], |
| int channel_config, enum OCStatus oc_type) |
| { |
| AVCodecContext *avctx = ac->avctx; |
| int i, type, channels = 0, ret; |
| |
| if (new_che_pos != che_pos) |
| memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
| |
| if (channel_config) { |
| for (i = 0; i < tags_per_config[channel_config]; i++) { |
| if ((ret = che_configure(ac, che_pos, |
| aac_channel_layout_map[channel_config - 1][i][0], |
| aac_channel_layout_map[channel_config - 1][i][1], |
| &channels))) |
| return ret; |
| } |
| |
| memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); |
| |
| avctx->channel_layout = aac_channel_layout[channel_config - 1]; |
| } else { |
| /* Allocate or free elements depending on if they are in the |
| * current program configuration. |
| * |
| * Set up default 1:1 output mapping. |
| * |
| * For a 5.1 stream the output order will be: |
| * [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ] |
| */ |
| |
| for (i = 0; i < MAX_ELEM_ID; i++) { |
| for (type = 0; type < 4; type++) { |
| if ((ret = che_configure(ac, che_pos, type, i, &channels))) |
| return ret; |
| } |
| } |
| |
| memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); |
| } |
| |
| avctx->channels = channels; |
| |
| ac->output_configured = oc_type; |
| |
| return 0; |
| } |
| |
| /** |
| * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. |
| * |
| * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. |
| * @param sce_map mono (Single Channel Element) map |
| * @param type speaker type/position for these channels |
| */ |
| static void decode_channel_map(enum ChannelPosition *cpe_map, |
| enum ChannelPosition *sce_map, |
| enum ChannelPosition type, |
| GetBitContext *gb, int n) |
| { |
| while (n--) { |
| enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map |
| map[get_bits(gb, 4)] = type; |
| } |
| } |
| |
| /** |
| * Decode program configuration element; reference: table 4.2. |
| * |
| * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. |
| * |
| * @return Returns error status. 0 - OK, !0 - error |
| */ |
| static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac, |
| enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], |
| GetBitContext *gb) |
| { |
| int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index; |
| int comment_len; |
| |
| 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, overread_err); |
| return -1; |
| } |
| decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); |
| decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); |
| decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); |
| decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); |
| |
| skip_bits_long(gb, 4 * num_assoc_data); |
| |
| decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); |
| |
| align_get_bits(gb); |
| |
| /* comment field, first byte is length */ |
| comment_len = get_bits(gb, 8) * 8; |
| if (get_bits_left(gb) < comment_len) { |
| av_log(avctx, AV_LOG_ERROR, overread_err); |
| return -1; |
| } |
| skip_bits_long(gb, comment_len); |
| return 0; |
| } |
| |
| /** |
| * Set up channel positions based on a default channel configuration |
| * as specified in table 1.17. |
| * |
| * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. |
| * |
| * @return Returns error status. 0 - OK, !0 - error |
| */ |
| static av_cold int set_default_channel_config(AVCodecContext *avctx, |
| enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], |
| int channel_config) |
| { |
| if (channel_config < 1 || channel_config > 7) { |
| av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", |
| channel_config); |
| return -1; |
| } |
| |
| /* default channel configurations: |
| * |
| * 1ch : front center (mono) |
| * 2ch : L + R (stereo) |
| * 3ch : front center + L + R |
| * 4ch : front center + L + R + back center |
| * 5ch : front center + L + R + back stereo |
| * 6ch : front center + L + R + back stereo + LFE |
| * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE |
| */ |
| |
| if (channel_config != 2) |
| new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) |
| if (channel_config > 1) |
| new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) |
| if (channel_config == 4) |
| new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center |
| if (channel_config > 4) |
| new_che_pos[TYPE_CPE][(channel_config == 7) + 1] |
| = AAC_CHANNEL_BACK; // back stereo |
| if (channel_config > 5) |
| new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE |
| if (channel_config == 7) |
| new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right |
| |
| return 0; |
| } |
| |
| /** |
| * Decode GA "General Audio" specific configuration; reference: table 4.1. |
| * |
| * @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) |
| { |
| enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; |
| int extension_flag, ret; |
| |
| if (get_bits1(gb)) { // frameLengthFlag |
| av_log_missing_feature(avctx, "960/120 MDCT window is", 1); |
| return -1; |
| } |
| |
| 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 |
| |
| memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
| if (channel_config == 0) { |
| skip_bits(gb, 4); // element_instance_tag |
| if ((ret = decode_pce(avctx, m4ac, new_che_pos, gb))) |
| return ret; |
| } else { |
| if ((ret = set_default_channel_config(avctx, new_che_pos, channel_config))) |
| return ret; |
| } |
| if (ac && (ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR))) |
| 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: |
| skip_bits(gb, 3); /* aacSectionDataResilienceFlag |
| * aacScalefactorDataResilienceFlag |
| * aacSpectralDataResilienceFlag |
| */ |
| break; |
| } |
| skip_bits1(gb); // extensionFlag3 (TBD in version 3) |
| } |
| return 0; |
| } |
| |
| /** |
| * Decode audio specific configuration; reference: table 1.13. |
| * |
| * @param 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 AVCodecContext extradata |
| * @param data_size size of AVCCodecContext extradata |
| * |
| * @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, int data_size) |
| { |
| GetBitContext gb; |
| int i; |
| |
| av_dlog(avctx, "extradata size %d\n", avctx->extradata_size); |
| for (i = 0; i < avctx->extradata_size; i++) |
| av_dlog(avctx, "%02x ", avctx->extradata[i]); |
| av_dlog(avctx, "\n"); |
| |
| init_get_bits(&gb, data, data_size * 8); |
| |
| if ((i = ff_mpeg4audio_get_config(m4ac, data, data_size)) < 0) |
| return -1; |
| if (m4ac->sampling_index > 12) { |
| av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index); |
| return -1; |
| } |
| if (m4ac->sbr == 1 && m4ac->ps == -1) |
| m4ac->ps = 1; |
| |
| skip_bits_long(&gb, i); |
| |
| switch (m4ac->object_type) { |
| case AOT_AAC_MAIN: |
| case AOT_AAC_LC: |
| case AOT_AAC_LTP: |
| if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config)) |
| return -1; |
| break; |
| default: |
| av_log(avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", |
| m4ac->sbr == 1? "SBR+" : "", m4ac->object_type); |
| return -1; |
| } |
| |
| av_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(int previous_val) |
| { |
| return previous_val * 1664525 + 1013904223; |
| } |
| |
| static av_always_inline void reset_predict_state(PredictorState *ps) |
| { |
| ps->r0 = 0.0f; |
| ps->r1 = 0.0f; |
| ps->cor0 = 0.0f; |
| ps->cor1 = 0.0f; |
| ps->var0 = 1.0f; |
| ps->var1 = 1.0f; |
| } |
| |
| static void reset_all_predictors(PredictorState *ps) |
| { |
| int i; |
| for (i = 0; i < MAX_PREDICTORS; i++) |
| reset_predict_state(&ps[i]); |
| } |
| |
| static void reset_predictor_group(PredictorState *ps, int group_num) |
| { |
| int i; |
| for (i = group_num - 1; i < MAX_PREDICTORS; i += 30) |
| reset_predict_state(&ps[i]); |
| } |
| |
| #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 av_cold int aac_decode_init(AVCodecContext *avctx) |
| { |
| AACContext *ac = avctx->priv_data; |
| float output_scale_factor; |
| |
| ac->avctx = avctx; |
| ac->m4ac.sample_rate = avctx->sample_rate; |
| |
| if (avctx->extradata_size > 0) { |
| if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac, |
| avctx->extradata, |
| avctx->extradata_size) < 0) |
| return -1; |
| } |
| |
| if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) { |
| avctx->sample_fmt = AV_SAMPLE_FMT_FLT; |
| output_scale_factor = 1.0 / 32768.0; |
| } else { |
| avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
| output_scale_factor = 1.0; |
| } |
| |
| 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); |
| |
| ff_aac_sbr_init(); |
| |
| dsputil_init(&ac->dsp, avctx); |
| ff_fmt_convert_init(&ac->fmt_conv, avctx); |
| |
| ac->random_state = 0x1f2e3d4c; |
| |
| 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); |
| |
| ff_mdct_init(&ac->mdct, 11, 1, output_scale_factor/1024.0); |
| ff_mdct_init(&ac->mdct_small, 8, 1, output_scale_factor/128.0); |
| ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0/output_scale_factor); |
| // window initialization |
| ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); |
| ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); |
| ff_init_ff_sine_windows(10); |
| ff_init_ff_sine_windows( 7); |
| |
| cbrt_tableinit(); |
| |
| 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, overread_err); |
| return -1; |
| } |
| 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 -1; |
| } |
| } |
| for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) { |
| ics->prediction_used[sfb] = get_bits1(gb); |
| } |
| return 0; |
| } |
| |
| /** |
| * Decode Long Term Prediction data; reference: table 4.xx. |
| */ |
| static void decode_ltp(AACContext *ac, 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. |
| * |
| * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. |
| */ |
| static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics, |
| GetBitContext *gb, int common_window) |
| { |
| if (get_bits1(gb)) { |
| av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n"); |
| memset(ics, 0, sizeof(IndividualChannelStream)); |
| return -1; |
| } |
| ics->window_sequence[1] = ics->window_sequence[0]; |
| ics->window_sequence[0] = get_bits(gb, 2); |
| ics->use_kb_window[1] = ics->use_kb_window[0]; |
| ics->use_kb_window[0] = get_bits1(gb); |
| ics->num_window_groups = 1; |
| ics->group_len[0] = 1; |
| if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
| int i; |
| ics->max_sfb = get_bits(gb, 4); |
| for (i = 0; i < 7; i++) { |
| if (get_bits1(gb)) { |
| ics->group_len[ics->num_window_groups - 1]++; |
| } else { |
| ics->num_window_groups++; |
| ics->group_len[ics->num_window_groups - 1] = 1; |
| } |
| } |
| ics->num_windows = 8; |
| ics->swb_offset = ff_swb_offset_128[ac->m4ac.sampling_index]; |
| ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; |
| ics->tns_max_bands = ff_tns_max_bands_128[ac->m4ac.sampling_index]; |
| ics->predictor_present = 0; |
| } else { |
| ics->max_sfb = get_bits(gb, 6); |
| ics->num_windows = 1; |
| ics->swb_offset = ff_swb_offset_1024[ac->m4ac.sampling_index]; |
| ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; |
| ics->tns_max_bands = ff_tns_max_bands_1024[ac->m4ac.sampling_index]; |
| ics->predictor_present = get_bits1(gb); |
| ics->predictor_reset_group = 0; |
| if (ics->predictor_present) { |
| if (ac->m4ac.object_type == AOT_AAC_MAIN) { |
| if (decode_prediction(ac, ics, gb)) { |
| memset(ics, 0, sizeof(IndividualChannelStream)); |
| return -1; |
| } |
| } else if (ac->m4ac.object_type == AOT_AAC_LC) { |
| av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n"); |
| memset(ics, 0, sizeof(IndividualChannelStream)); |
| return -1; |
| } else { |
| if ((ics->ltp.present = get_bits(gb, 1))) |
| decode_ltp(ac, &ics->ltp, gb, ics->max_sfb); |
| } |
| } |
| } |
| |
| if (ics->max_sfb > ics->num_swb) { |
| av_log(ac->avctx, AV_LOG_ERROR, |
| "Number of scalefactor bands in group (%d) exceeds limit (%d).\n", |
| ics->max_sfb, ics->num_swb); |
| memset(ics, 0, sizeof(IndividualChannelStream)); |
| return -1; |
| } |
| |
| return 0; |
| } |
| |
| /** |
| * Decode band types (section_data payload); reference: table 4.46. |
| * |
| * @param band_type array of the used band type |
| * @param band_type_run_end array of the last scalefactor band of a band type run |
| * |
| * @return Returns error status. 0 - OK, !0 - error |
| */ |
| static int decode_band_types(AACContext *ac, enum BandType band_type[120], |
| int band_type_run_end[120], GetBitContext *gb, |
| IndividualChannelStream *ics) |
| { |
| int g, idx = 0; |
| const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; |
| for (g = 0; g < ics->num_window_groups; g++) { |
| int k = 0; |
| while (k < ics->max_sfb) { |
| uint8_t sect_end = k; |
| int sect_len_incr; |
| int sect_band_type = get_bits(gb, 4); |
| if (sect_band_type == 12) { |
| av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n"); |
| return -1; |
| } |
| while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1) |
| sect_end += sect_len_incr; |
| sect_end += sect_len_incr; |
| if (get_bits_left(gb) < 0) { |
| av_log(ac->avctx, AV_LOG_ERROR, overread_err); |
| return -1; |
| } |
| 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 -1; |
| } |
| for (; k < sect_end; k++) { |
| band_type [idx] = sect_band_type; |
| band_type_run_end[idx++] = sect_end; |
| } |
| } |
| } |
| return 0; |
| } |
| |
| /** |
| * Decode scalefactors; reference: table 4.47. |
| * |
| * @param global_gain first scalefactor value as scalefactors are differentially coded |
| * @param band_type array of the used band type |
| * @param band_type_run_end array of the last scalefactor band of a band type run |
| * @param sf array of scalefactors or intensity stereo positions |
| * |
| * @return Returns error status. 0 - OK, !0 - error |
| */ |
| static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb, |
| unsigned int global_gain, |
| IndividualChannelStream *ics, |
| enum BandType band_type[120], |
| int band_type_run_end[120]) |
| { |
| int g, i, idx = 0; |
| int offset[3] = { global_gain, global_gain - 90, 0 }; |
| int clipped_offset; |
| int noise_flag = 1; |
| static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; |
| for (g = 0; g < ics->num_window_groups; g++) { |
| for (i = 0; i < ics->max_sfb;) { |
| int run_end = band_type_run_end[idx]; |
| if (band_type[idx] == ZERO_BT) { |
| for (; i < run_end; i++, idx++) |
| sf[idx] = 0.; |
| } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { |
| for (; i < run_end; i++, idx++) { |
| offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
| clipped_offset = av_clip(offset[2], -155, 100); |
| if (offset[2] != clipped_offset) { |
| av_log_ask_for_sample(ac->avctx, "Intensity stereo " |
| "position clipped (%d -> %d).\nIf you heard an " |
| "audible artifact, there may be a bug in the " |
| "decoder. ", offset[2], clipped_offset); |
| } |
| sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO]; |
| } |
| } else if (band_type[idx] == NOISE_BT) { |
| for (; i < run_end; i++, idx++) { |
| if (noise_flag-- > 0) |
| offset[1] += get_bits(gb, 9) - 256; |
| else |
| offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
| clipped_offset = av_clip(offset[1], -100, 155); |
| if (offset[1] != clipped_offset) { |
| av_log_ask_for_sample(ac->avctx, "Noise gain clipped " |
| "(%d -> %d).\nIf you heard an audible " |
| "artifact, there may be a bug in the decoder. ", |
| offset[1], clipped_offset); |
| } |
| sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO]; |
| } |
| } else { |
| for (; i < run_end; i++, idx++) { |
| offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
| if (offset[0] > 255U) { |
| av_log(ac->avctx, AV_LOG_ERROR, |
| "%s (%d) out of range.\n", sf_str[0], offset[0]); |
| return -1; |
| } |
| sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO]; |
| } |
| } |
| } |
| } |
| return 0; |
| } |
| |
| /** |
| * Decode pulse data; reference: table 4.7. |
| */ |
| static int decode_pulses(Pulse *pulse, GetBitContext *gb, |
| const uint16_t *swb_offset, int num_swb) |
| { |
| int i, pulse_swb; |
| pulse->num_pulse = get_bits(gb, 2) + 1; |
| pulse_swb = get_bits(gb, 6); |
| if (pulse_swb >= num_swb) |
| return -1; |
| pulse->pos[0] = swb_offset[pulse_swb]; |
| pulse->pos[0] += get_bits(gb, 5); |
| if (pulse->pos[0] > 1023) |
| return -1; |
| pulse->amp[0] = get_bits(gb, 4); |
| for (i = 1; i < pulse->num_pulse; i++) { |
| pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1]; |
| if (pulse->pos[i] > 1023) |
| return -1; |
| pulse->amp[i] = get_bits(gb, 4); |
| } |
| return 0; |
| } |
| |
| /** |
| * Decode Temporal Noise Shaping data; reference: table 4.48. |
| * |
| * @return Returns error status. 0 - OK, !0 - error |
| */ |
| static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns, |
| GetBitContext *gb, const IndividualChannelStream *ics) |
| { |
| int w, filt, i, coef_len, coef_res, coef_compress; |
| const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; |
| const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; |
| for (w = 0; w < ics->num_windows; w++) { |
| if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) { |
| coef_res = get_bits1(gb); |
| |
| for (filt = 0; filt < tns->n_filt[w]; filt++) { |
| int tmp2_idx; |
| tns->length[w][filt] = get_bits(gb, 6 - 2 * is8); |
| |
| if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) { |
| av_log(ac->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 -1; |
| } |
| if (tns->order[w][filt]) { |
| tns->direction[w][filt] = get_bits1(gb); |
| coef_compress = get_bits1(gb); |
| coef_len = coef_res + 3 - coef_compress; |
| tmp2_idx = 2 * coef_compress + coef_res; |
| |
| for (i = 0; i < tns->order[w][filt]; i++) |
| tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; |
| } |
| } |
| } |
| } |
| return 0; |
| } |
| |
| /** |
| * Decode Mid/Side data; reference: table 4.54. |
| * |
| * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; |
| * [1] mask is decoded from bitstream; [2] mask is all 1s; |
| * [3] reserved for scalable AAC |
| */ |
| static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb, |
| int ms_present) |
| { |
| int idx; |
| if (ms_present == 1) { |
| for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++) |
| cpe->ms_mask[idx] = get_bits1(gb); |
| } else if (ms_present == 2) { |
| memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0])); |
| } |
| } |
| |
| #ifndef VMUL2 |
| static inline float *VMUL2(float *dst, const float *v, unsigned idx, |
| const float *scale) |
| { |
| float s = *scale; |
| *dst++ = v[idx & 15] * s; |
| *dst++ = v[idx>>4 & 15] * s; |
| return dst; |
| } |
| #endif |
| |
| #ifndef VMUL4 |
| static inline float *VMUL4(float *dst, const float *v, unsigned idx, |
| const float *scale) |
| { |
| float s = *scale; |
| *dst++ = v[idx & 3] * s; |
| *dst++ = v[idx>>2 & 3] * s; |
| *dst++ = v[idx>>4 & 3] * s; |
| *dst++ = v[idx>>6 & 3] * s; |
| return dst; |
| } |
| #endif |
| |
| #ifndef VMUL2S |
| static inline float *VMUL2S(float *dst, const float *v, unsigned idx, |
| unsigned sign, const float *scale) |
| { |
| union float754 s0, s1; |
| |
| s0.f = s1.f = *scale; |
| s0.i ^= sign >> 1 << 31; |
| s1.i ^= sign << 31; |
| |
| *dst++ = v[idx & 15] * s0.f; |
| *dst++ = v[idx>>4 & 15] * s1.f; |
| |
| return dst; |
| } |
| #endif |
| |
| #ifndef VMUL4S |
| static inline float *VMUL4S(float *dst, const float *v, unsigned idx, |
| unsigned sign, const float *scale) |
| { |
| unsigned nz = idx >> 12; |
| union float754 s = { .f = *scale }; |
| union float754 t; |
| |
| t.i = s.i ^ (sign & 1U<<31); |
| *dst++ = v[idx & 3] * t.f; |
| |
| sign <<= nz & 1; nz >>= 1; |
| t.i = s.i ^ (sign & 1U<<31); |
| *dst++ = v[idx>>2 & 3] * t.f; |
| |
| sign <<= nz & 1; nz >>= 1; |
| t.i = s.i ^ (sign & 1U<<31); |
| *dst++ = v[idx>>4 & 3] * t.f; |
| |
| sign <<= nz & 1; nz >>= 1; |
| t.i = s.i ^ (sign & 1U<<31); |
| *dst++ = v[idx>>6 & 3] * t.f; |
| |
| return dst; |
| } |
| #endif |
| |
| /** |
| * Decode spectral data; reference: table 4.50. |
| * Dequantize and scale spectral data; reference: 4.6.3.3. |
| * |
| * @param coef array of dequantized, scaled spectral data |
| * @param sf array of scalefactors or intensity stereo positions |
| * @param pulse_present set if pulses are present |
| * @param pulse pointer to pulse data struct |
| * @param band_type array of the used band type |
| * |
| * @return Returns error status. 0 - OK, !0 - error |
| */ |
| static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024], |
| GetBitContext *gb, const float sf[120], |
| int pulse_present, const Pulse *pulse, |
| const IndividualChannelStream *ics, |
| enum BandType band_type[120]) |
| { |
| int i, k, g, idx = 0; |
| const int c = 1024 / ics->num_windows; |
| const uint16_t *offsets = ics->swb_offset; |
| float *coef_base = coef; |
| |
| for (g = 0; g < ics->num_windows; g++) |
| memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb])); |
| |
| for (g = 0; g < ics->num_window_groups; g++) { |
| unsigned g_len = ics->group_len[g]; |
| |
| for (i = 0; i < ics->max_sfb; i++, idx++) { |
| const unsigned cbt_m1 = band_type[idx] - 1; |
| float *cfo = coef + offsets[i]; |
| int off_len = offsets[i + 1] - offsets[i]; |
| int group; |
| |
| if (cbt_m1 >= INTENSITY_BT2 - 1) { |
| for (group = 0; group < g_len; group++, cfo+=128) { |
| memset(cfo, 0, off_len * sizeof(float)); |
| } |
| } else if (cbt_m1 == NOISE_BT - 1) { |
| for (group = 0; group < g_len; group++, cfo+=128) { |
| float scale; |
| float band_energy; |
| |
| for (k = 0; k < off_len; k++) { |
| ac->random_state = lcg_random(ac->random_state); |
| cfo[k] = ac->random_state; |
| } |
| |
| band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len); |
| scale = sf[idx] / sqrtf(band_energy); |
| ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len); |
| } |
| } else { |
| const float *vq = ff_aac_codebook_vector_vals[cbt_m1]; |
| const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1]; |
| VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table; |
| OPEN_READER(re, gb); |
| |
| switch (cbt_m1 >> 1) { |
| case 0: |
| for (group = 0; group < g_len; group++, cfo+=128) { |
| float *cf = cfo; |
| int len = off_len; |
| |
| do { |
| int code; |
| unsigned cb_idx; |
| |
| UPDATE_CACHE(re, gb); |
| GET_VLC(code, re, gb, vlc_tab, 8, 2); |
| cb_idx = cb_vector_idx[code]; |
| cf = VMUL4(cf, vq, cb_idx, sf + idx); |
| } while (len -= 4); |
| } |
| break; |
| |
| case 1: |
| for (group = 0; group < g_len; group++, cfo+=128) { |
| float *cf = cfo; |
| int len = off_len; |
| |
| do { |
| int code; |
| unsigned nnz; |
| unsigned cb_idx; |
| uint32_t bits; |
| |
| UPDATE_CACHE(re, gb); |
| GET_VLC(code, re, gb, vlc_tab, 8, 2); |
| cb_idx = cb_vector_idx[code]; |
| nnz = cb_idx >> 8 & 15; |
| bits = SHOW_UBITS(re, gb, nnz) << (32-nnz); |
| LAST_SKIP_BITS(re, gb, nnz); |
| cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx); |
| } while (len -= 4); |
| } |
| break; |
| |
| case 2: |
| for (group = 0; group < g_len; group++, cfo+=128) { |
| float *cf = cfo; |
| int len = off_len; |
| |
| do { |
| int code; |
| unsigned cb_idx; |
| |
| UPDATE_CACHE(re, gb); |
| GET_VLC(code, re, gb, vlc_tab, 8, 2); |
| cb_idx = cb_vector_idx[code]; |
| cf = VMUL2(cf, vq, cb_idx, sf + idx); |
| } while (len -= 2); |
| } |
| break; |
| |
| case 3: |
| case 4: |
| for (group = 0; group < g_len; group++, cfo+=128) { |
| float *cf = cfo; |
| int len = off_len; |
| |
| do { |
| int code; |
| unsigned nnz; |
| unsigned cb_idx; |
| unsigned sign; |
| |
| UPDATE_CACHE(re, gb); |
| GET_VLC(code, re, gb, vlc_tab, 8, 2); |
| cb_idx = cb_vector_idx[code]; |
| nnz = cb_idx >> 8 & 15; |
| sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12); |
| LAST_SKIP_BITS(re, gb, nnz); |
| cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx); |
| } while (len -= 2); |
| } |
| break; |
| |
| default: |
| for (group = 0; group < g_len; group++, cfo+=128) { |
| float *cf = cfo; |
| uint32_t *icf = (uint32_t *) cf; |
| int len = off_len; |
| |
| do { |
| int code; |
| unsigned nzt, nnz; |
| unsigned cb_idx; |
| uint32_t bits; |
| int j; |
| |
| UPDATE_CACHE(re, gb); |
| GET_VLC(code, re, gb, vlc_tab, 8, 2); |
| |
| if (!code) { |
| *icf++ = 0; |
| *icf++ = 0; |
| continue; |
| } |
| |
| cb_idx = cb_vector_idx[code]; |
| nnz = cb_idx >> 12; |
| nzt = cb_idx >> 8; |
| bits = SHOW_UBITS(re, gb, nnz) << (32-nnz); |
| LAST_SKIP_BITS(re, gb, nnz); |
| |
| for (j = 0; j < 2; j++) { |
| if (nzt & 1<<j) { |
| uint32_t b; |
| int n; |
| /* The total length of escape_sequence must be < 22 bits according |
| to the specification (i.e. max is 111111110xxxxxxxxxxxx). */ |
| UPDATE_CACHE(re, gb); |
| b = GET_CACHE(re, gb); |
| b = 31 - av_log2(~b); |
| |
| if (b > 8) { |
| av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); |
| return -1; |
| } |
| |
| SKIP_BITS(re, gb, b + 1); |
| b += 4; |
| n = (1 << b) + SHOW_UBITS(re, gb, b); |
| LAST_SKIP_BITS(re, gb, b); |
| *icf++ = cbrt_tab[n] | (bits & 1U<<31); |
| bits <<= 1; |
| } else { |
| unsigned v = ((const uint32_t*)vq)[cb_idx & 15]; |
| *icf++ = (bits & 1U<<31) | v; |
| bits <<= !!v; |
| } |
| cb_idx >>= 4; |
| } |
| } while (len -= 2); |
| |
| ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len); |
| } |
| } |
| |
| CLOSE_READER(re, gb); |
| } |
| } |
| coef += g_len << 7; |
| } |
| |
| if (pulse_present) { |
| idx = 0; |
| for (i = 0; i < pulse->num_pulse; i++) { |
| float co = coef_base[ pulse->pos[i] ]; |
| while (offsets[idx + 1] <= pulse->pos[i]) |
| idx++; |
| if (band_type[idx] != NOISE_BT && sf[idx]) { |
| float ico = -pulse->amp[i]; |
| if (co) { |
| co /= sf[idx]; |
| ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico); |
| } |
| coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx]; |
| } |
| } |
| } |
| return 0; |
| } |
| |
| static av_always_inline float flt16_round(float pf) |
| { |
| union float754 tmp; |
| tmp.f = pf; |
| tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U; |
| return tmp.f; |
| } |
| |
| static av_always_inline float flt16_even(float pf) |
| { |
| union float754 tmp; |
| tmp.f = pf; |
| tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U; |
| return tmp.f; |
| } |
| |
| static av_always_inline float flt16_trunc(float pf) |
| { |
| union float754 pun; |
| pun.f = pf; |
| pun.i &= 0xFFFF0000U; |
| return pun.f; |
| } |
| |
| static av_always_inline void predict(PredictorState *ps, float *coef, |
| int output_enable) |
| { |
| const float a = 0.953125; // 61.0 / 64 |
| const float alpha = 0.90625; // 29.0 / 32 |
| float e0, e1; |
| float pv; |
| float k1, k2; |
| float r0 = ps->r0, r1 = ps->r1; |
| float cor0 = ps->cor0, cor1 = ps->cor1; |
| float var0 = ps->var0, var1 = ps->var1; |
| |
| k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0; |
| k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0; |
| |
| pv = flt16_round(k1 * r0 + k2 * r1); |
| if (output_enable) |
| *coef += pv; |
| |
| e0 = *coef; |
| e1 = e0 - k1 * r0; |
| |
| ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1); |
| ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1)); |
| ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0); |
| ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0)); |
| |
| ps->r1 = flt16_trunc(a * (r0 - k1 * e0)); |
| ps->r0 = flt16_trunc(a * e0); |
| } |
| |
| /** |
| * Apply AAC-Main style frequency domain prediction. |
| */ |
| static void apply_prediction(AACContext *ac, SingleChannelElement *sce) |
| { |
| int sfb, k; |
| |
| if (!sce->ics.predictor_initialized) { |
| reset_all_predictors(sce->predictor_state); |
| sce->ics.predictor_initialized = 1; |
| } |
| |
| if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { |
| for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) { |
| for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) { |
| predict(&sce->predictor_state[k], &sce->coeffs[k], |
| sce->ics.predictor_present && sce->ics.prediction_used[sfb]); |
| } |
| } |
| if (sce->ics.predictor_reset_group) |
| reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group); |
| } else |
| reset_all_predictors(sce->predictor_state); |
| } |
| |
| /** |
| * Decode an individual_channel_stream payload; reference: table 4.44. |
| * |
| * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. |
| * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.) |
| * |
| * @return Returns error status. 0 - OK, !0 - error |
| */ |
| static int decode_ics(AACContext *ac, SingleChannelElement *sce, |
| GetBitContext *gb, int common_window, int scale_flag) |
| { |
| Pulse pulse; |
| TemporalNoiseShaping *tns = &sce->tns; |
| IndividualChannelStream *ics = &sce->ics; |
| float *out = sce->coeffs; |
| int global_gain, pulse_present = 0; |
| |
| /* This assignment is to silence a GCC warning about the variable being used |
| * uninitialized when in fact it always is. |
| */ |
| pulse.num_pulse = 0; |
| |
| global_gain = get_bits(gb, 8); |
| |
| if (!common_window && !scale_flag) { |
| if (decode_ics_info(ac, ics, gb, 0) < 0) |
| return -1; |
| } |
| |
| if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) |
| return -1; |
| if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) |
| return -1; |
| |
| pulse_present = 0; |
| if (!scale_flag) { |
| if ((pulse_present = get_bits1(gb))) { |
| if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
| av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); |
| return -1; |
| } |
| if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) { |
| av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n"); |
| return -1; |
| } |
| } |
| if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics)) |
| return -1; |
| if (get_bits1(gb)) { |
| av_log_missing_feature(ac->avctx, "SSR", 1); |
| return -1; |
| } |
| } |
| |
| if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0) |
| return -1; |
| |
| if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window) |
| apply_prediction(ac, sce); |
| |
| return 0; |
| } |
| |
| /** |
| * Mid/Side stereo decoding; reference: 4.6.8.1.3. |
| */ |
| static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe) |
| { |
| const IndividualChannelStream *ics = &cpe->ch[0].ics; |
| float *ch0 = cpe->ch[0].coeffs; |
| float *ch1 = cpe->ch[1].coeffs; |
| int g, i, group, idx = 0; |
| const uint16_t *offsets = ics->swb_offset; |
| for (g = 0; g < ics->num_window_groups; g++) { |
| for (i = 0; i < ics->max_sfb; i++, idx++) { |
| if (cpe->ms_mask[idx] && |
| cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { |
| for (group = 0; group < ics->group_len[g]; group++) { |
| ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i], |
| ch1 + group * 128 + offsets[i], |
| offsets[i+1] - offsets[i]); |
| } |
| } |
| } |
| ch0 += ics->group_len[g] * 128; |
| ch1 += ics->group_len[g] * 128; |
| } |
| } |
| |
| /** |
| * intensity stereo decoding; reference: 4.6.8.2.3 |
| * |
| * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; |
| * [1] mask is decoded from bitstream; [2] mask is all 1s; |
| * [3] reserved for scalable AAC |
| */ |
| static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present) |
| { |
| const IndividualChannelStream *ics = &cpe->ch[1].ics; |
| SingleChannelElement *sce1 = &cpe->ch[1]; |
| float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs; |
| const uint16_t *offsets = ics->swb_offset; |
| int g, group, i, idx = 0; |
| int c; |
| float scale; |
| for (g = 0; g < ics->num_window_groups; g++) { |
| for (i = 0; i < ics->max_sfb;) { |
| if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) { |
| const int bt_run_end = sce1->band_type_run_end[idx]; |
| for (; i < bt_run_end; i++, idx++) { |
| c = -1 + 2 * (sce1->band_type[idx] - 14); |
| if (ms_present) |
| c *= 1 - 2 * cpe->ms_mask[idx]; |
| scale = c * sce1->sf[idx]; |
| for (group = 0; group < ics->group_len[g]; group++) |
| ac->dsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i], |
| coef0 + group * 128 + offsets[i], |
| scale, |
| offsets[i + 1] - offsets[i]); |
| } |
| } 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; |
| |
| common_window = get_bits1(gb); |
| if (common_window) { |
| if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) |
| return -1; |
| i = cpe->ch[1].ics.use_kb_window[0]; |
| cpe->ch[1].ics = cpe->ch[0].ics; |
| cpe->ch[1].ics.use_kb_window[1] = i; |
| if (cpe->ch[1].ics.predictor_present && (ac->m4ac.object_type != AOT_AAC_MAIN)) |
| if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1))) |
| decode_ltp(ac, &cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb); |
| ms_present = get_bits(gb, 2); |
| if (ms_present == 3) { |
| av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); |
| return -1; |
| } else if (ms_present) |
| decode_mid_side_stereo(cpe, gb, ms_present); |
| } |
| if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) |
| return ret; |
| if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) |
| return ret; |
| |
| if (common_window) { |
| if (ms_present) |
| apply_mid_side_stereo(ac, cpe); |
| if (ac->m4ac.object_type == AOT_AAC_MAIN) { |
| apply_prediction(ac, &cpe->ch[0]); |
| apply_prediction(ac, &cpe->ch[1]); |
| } |
| } |
| |
| apply_intensity_stereo(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; |
| float scale; |
| SingleChannelElement *sce = &che->ch[0]; |
| ChannelCoupling *coup = &che->coup; |
| |
| coup->coupling_point = 2 * get_bits1(gb); |
| coup->num_coupled = get_bits(gb, 3); |
| for (c = 0; c <= coup->num_coupled; c++) { |
| num_gain++; |
| coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE; |
| coup->id_select[c] = get_bits(gb, 4); |
| if (coup->type[c] == TYPE_CPE) { |
| coup->ch_select[c] = get_bits(gb, 2); |
| if (coup->ch_select[c] == 3) |
| num_gain++; |
| } else |
| coup->ch_select[c] = 2; |
| } |
| coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1); |
| |
| sign = get_bits(gb, 1); |
| scale = 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; |
| float gain_cache = 1.; |
| if (c) { |
| cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb); |
| gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0; |
| gain_cache = powf(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 = powf(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. |
| * |
| * @param cnt length of TYPE_FIL syntactic element in bytes |
| * |
| * @return Returns number of bytes consumed. |
| */ |
| static int decode_dynamic_range(DynamicRangeControl *che_drc, |
| GetBitContext *gb, int cnt) |
| { |
| int n = 1; |
| int drc_num_bands = 1; |
| int i; |
| |
| /* pce_tag_present? */ |
| if (get_bits1(gb)) { |
| che_drc->pce_instance_tag = get_bits(gb, 4); |
| skip_bits(gb, 4); // tag_reserved_bits |
| n++; |
| } |
| |
| /* excluded_chns_present? */ |
| if (get_bits1(gb)) { |
| n += decode_drc_channel_exclusions(che_drc, gb); |
| } |
| |
| /* drc_bands_present? */ |
| if (get_bits1(gb)) { |
| che_drc->band_incr = get_bits(gb, 4); |
| che_drc->interpolation_scheme = get_bits(gb, 4); |
| n++; |
| drc_num_bands += che_drc->band_incr; |
| for (i = 0; i < drc_num_bands; i++) { |
| che_drc->band_top[i] = get_bits(gb, 8); |
| n++; |
| } |
| } |
| |
| /* prog_ref_level_present? */ |
| if (get_bits1(gb)) { |
| che_drc->prog_ref_level = get_bits(gb, 7); |
| skip_bits1(gb); // prog_ref_level_reserved_bits |
| n++; |
| } |
| |
| for (i = 0; i < drc_num_bands; i++) { |
| che_drc->dyn_rng_sgn[i] = get_bits1(gb); |
| che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); |
| n++; |
| } |
| |
| return n; |
| } |
| |
| /** |
| * Decode extension data (incomplete); reference: table 4.51. |
| * |
| * @param cnt length of TYPE_FIL syntactic element in bytes |
| * |
| * @return Returns number of bytes consumed |
| */ |
| static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt, |
| ChannelElement *che, enum RawDataBlockType elem_type) |
| { |
| int crc_flag = 0; |
| int res = cnt; |
| switch (get_bits(gb, 4)) { // extension type |
| case EXT_SBR_DATA_CRC: |
| crc_flag++; |
| case EXT_SBR_DATA: |
| if (!che) { |
| av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n"); |
| return res; |
| } else if (!ac->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->m4ac.sbr == -1 && ac->output_configured == 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->m4ac.ps == -1 && ac->output_configured < OC_LOCKED && ac->avctx->channels == 1) { |
| ac->m4ac.sbr = 1; |
| ac->m4ac.ps = 1; |
| output_configure(ac, ac->che_pos, ac->che_pos, ac->m4ac.chan_config, ac->output_configured); |
| } else { |
| ac->m4ac.sbr = 1; |
| } |
| res = 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, cnt); |
| break; |
| case EXT_FILL: |
| case EXT_FILL_DATA: |
| case EXT_DATA_ELEMENT: |
| default: |
| skip_bits_long(gb, 8 * cnt - 4); |
| break; |
| }; |
| return res; |
| } |
| |
| /** |
| * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3. |
| * |
| * @param decode 1 if tool is used normally, 0 if tool is used in LTP. |
| * @param coef spectral coefficients |
| */ |
| static void apply_tns(float coef[1024], TemporalNoiseShaping *tns, |
| IndividualChannelStream *ics, int decode) |
| { |
| const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); |
| int w, filt, m, i; |
| int bottom, top, order, start, end, size, inc; |
| float lpc[TNS_MAX_ORDER]; |
| float tmp[TNS_MAX_ORDER]; |
| |
| for (w = 0; w < ics->num_windows; w++) { |
| bottom = ics->num_swb; |
| for (filt = 0; filt < tns->n_filt[w]; filt++) { |
| top = bottom; |
| bottom = FFMAX(0, top - tns->length[w][filt]); |
| order = tns->order[w][filt]; |
| if (order == 0) |
| continue; |
| |
| // tns_decode_coef |
| compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0); |
| |
| start = ics->swb_offset[FFMIN(bottom, mmm)]; |
| end = ics->swb_offset[FFMIN( top, mmm)]; |
| if ((size = end - start) <= 0) |
| continue; |
| if (tns->direction[w][filt]) { |
| inc = -1; |
| start = end - 1; |
| } else { |
| inc = 1; |
| } |
| start += w * 128; |
| |
| if (decode) { |
| // ar filter |
| for (m = 0; m < size; m++, start += inc) |
| for (i = 1; i <= FFMIN(m, order); i++) |
| coef[start] -= coef[start - i * inc] * lpc[i - 1]; |
| } 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] += 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, float *out, |
| float *in, IndividualChannelStream *ics) |
| { |
| const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024; |
| const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; |
| const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; |
| const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; |
| |
| if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) { |
| ac->dsp.vector_fmul(in, in, lwindow_prev, 1024); |
| } else { |
| memset(in, 0, 448 * sizeof(float)); |
| ac->dsp.vector_fmul(in + 448, in + 448, swindow_prev, 128); |
| } |
| if (ics->window_sequence[0] != LONG_START_SEQUENCE) { |
| ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024); |
| } else { |
| ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128); |
| memset(in + 1024 + 576, 0, 448 * sizeof(float)); |
| } |
| 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) { |
| float *predTime = sce->ret; |
| float *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] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef; |
| memset(&predTime[i], 0, (2048 - i) * sizeof(float)); |
| |
| windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics); |
| |
| if (sce->tns.present) |
| 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; |
| float *saved = sce->saved; |
| float *saved_ltp = sce->coeffs; |
| const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024; |
| const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; |
| int i; |
| |
| if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
| memcpy(saved_ltp, saved, 512 * sizeof(float)); |
| memset(saved_ltp + 576, 0, 448 * sizeof(float)); |
| ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64); |
| for (i = 0; i < 64; i++) |
| saved_ltp[i + 512] = 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(float)); |
| memset(saved_ltp + 576, 0, 448 * sizeof(float)); |
| ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64); |
| for (i = 0; i < 64; i++) |
| saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i]; |
| } else { // LONG_STOP or ONLY_LONG |
| ac->dsp.vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512); |
| for (i = 0; i < 512; i++) |
| saved_ltp[i + 512] = 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; |
| float *in = sce->coeffs; |
| float *out = sce->ret; |
| float *saved = sce->saved; |
| const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; |
| const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; |
| const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; |
| float *buf = ac->buf_mdct; |
| float *temp = ac->temp; |
| int i; |
| |
| // imdct |
| if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
| 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); |
| |
| /* window overlapping |
| * NOTE: To simplify the overlapping code, all 'meaningless' short to long |
| * and long to short transitions are considered to be short to short |
| * transitions. This leaves just two cases (long to long and short to short) |
| * with a little special sauce for EIGHT_SHORT_SEQUENCE. |
| */ |
| if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) && |
| (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) { |
| ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, 512); |
| } else { |
| memcpy( out, saved, 448 * sizeof(float)); |
| |
| if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
| ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64); |
| ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64); |
| ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64); |
| ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64); |
| ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64); |
| memcpy( out + 448 + 4*128, temp, 64 * sizeof(float)); |
| } else { |
| ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64); |
| memcpy( out + 576, buf + 64, 448 * sizeof(float)); |
| } |
| } |
| |
| // buffer update |
| if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
| memcpy( saved, temp + 64, 64 * sizeof(float)); |
| ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64); |
| ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64); |
| ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64); |
| memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
| } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) { |
| memcpy( saved, buf + 512, 448 * sizeof(float)); |
| memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
| } else { // LONG_STOP or ONLY_LONG |
| memcpy( saved, buf + 512, 512 * sizeof(float)); |
| } |
| } |
| |
| /** |
| * Apply dependent channel coupling (applied before IMDCT). |
| * |
| * @param index index into coupling gain array |
| */ |
| static void apply_dependent_coupling(AACContext *ac, |
| SingleChannelElement *target, |
| ChannelElement *cce, int index) |
| { |
| IndividualChannelStream *ics = &cce->ch[0].ics; |
| const uint16_t *offsets = ics->swb_offset; |
| float *dest = target->coeffs; |
| const float *src = cce->ch[0].coeffs; |
| int g, i, group, k, idx = 0; |
| if (ac->m4ac.object_type == AOT_AAC_LTP) { |
| av_log(ac->avctx, AV_LOG_ERROR, |
| "Dependent coupling is not supported together with LTP\n"); |
| return; |
| } |
| for (g = 0; g < ics->num_window_groups; g++) { |
| for (i = 0; i < ics->max_sfb; i++, idx++) { |
| if (cce->ch[0].band_type[idx] != ZERO_BT) { |
| const float gain = cce->coup.gain[index][idx]; |
| for (group = 0; group < ics->group_len[g]; group++) { |
| for (k = offsets[i]; k < offsets[i + 1]; k++) { |
| // XXX dsputil-ize |
| dest[group * 128 + k] += gain * src[group * 128 + k]; |
| } |
| } |
| } |
| } |
| dest += ics->group_len[g] * 128; |
| src += ics->group_len[g] * 128; |
| } |
| } |
| |
| /** |
| * Apply independent channel coupling (applied after IMDCT). |
| * |
| * @param index index into coupling gain array |
| */ |
| static void apply_independent_coupling(AACContext *ac, |
| SingleChannelElement *target, |
| ChannelElement *cce, int index) |
| { |
| int i; |
| const float gain = cce->coup.gain[index][0]; |
| const float *src = cce->ch[0].ret; |
| float *dest = target->ret; |
| const int len = 1024 << (ac->m4ac.sbr == 1); |
| |
| for (i = 0; i < len; i++) |
| dest[i] += gain * src[i]; |
| } |
| |
| /** |
| * 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 float samples, applying all supported tools as appropriate. |
| */ |
| static void spectral_to_sample(AACContext *ac) |
| { |
| int i, type; |
| for (type = 3; type >= 0; type--) { |
| for (i = 0; i < MAX_ELEM_ID; i++) { |
| ChannelElement *che = ac->che[type][i]; |
| if (che) { |
| if (type <= TYPE_CPE) |
| apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling); |
| if (ac->m4ac.object_type == AOT_AAC_LTP) { |
| if (che->ch[0].ics.predictor_present) { |
| if (che->ch[0].ics.ltp.present) |
| apply_ltp(ac, &che->ch[0]); |
| if (che->ch[1].ics.ltp.present && type == TYPE_CPE) |
| apply_ltp(ac, &che->ch[1]); |
| } |
| } |
| if (che->ch[0].tns.present) |
| apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); |
| if (che->ch[1].tns.present) |
| apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1); |
| if (type <= TYPE_CPE) |
| apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling); |
| if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) { |
| imdct_and_windowing(ac, &che->ch[0]); |
| if (ac->m4ac.object_type == AOT_AAC_LTP) |
| update_ltp(ac, &che->ch[0]); |
| if (type == TYPE_CPE) { |
| imdct_and_windowing(ac, &che->ch[1]); |
| if (ac->m4ac.object_type == AOT_AAC_LTP) |
| update_ltp(ac, &che->ch[1]); |
| } |
| if (ac->m4ac.sbr > 0) { |
| 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, apply_independent_coupling); |
| } |
| } |
| } |
| } |
| |
| static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb) |
| { |
| int size; |
| AACADTSHeaderInfo hdr_info; |
| |
| size = ff_aac_parse_header(gb, &hdr_info); |
| if (size > 0) { |
| if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) { |
| enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; |
| memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
| ac->m4ac.chan_config = hdr_info.chan_config; |
| if (set_default_channel_config(ac->avctx, new_che_pos, hdr_info.chan_config)) |
| return -7; |
| if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME)) |
| return -7; |
| } else if (ac->output_configured != OC_LOCKED) { |
| ac->output_configured = OC_NONE; |
| } |
| if (ac->output_configured != OC_LOCKED) { |
| ac->m4ac.sbr = -1; |
| ac->m4ac.ps = -1; |
| } |
| ac->m4ac.sample_rate = hdr_info.sample_rate; |
| ac->m4ac.sampling_index = hdr_info.sampling_index; |
| ac->m4ac.object_type = hdr_info.object_type; |
| if (!ac->avctx->sample_rate) |
| ac->avctx->sample_rate = hdr_info.sample_rate; |
| if (hdr_info.num_aac_frames == 1) { |
| if (!hdr_info.crc_absent) |
| skip_bits(gb, 16); |
| } else { |
| av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0); |
| return -1; |
| } |
| } |
| return size; |
| } |
| |
| static int aac_decode_frame_int(AVCodecContext *avctx, void *data, |
| int *data_size, GetBitContext *gb) |
| { |
| AACContext *ac = avctx->priv_data; |
| ChannelElement *che = NULL, *che_prev = NULL; |
| enum RawDataBlockType elem_type, elem_type_prev = TYPE_END; |
| int err, elem_id, data_size_tmp; |
| int samples = 0, multiplier, audio_found = 0; |
| |
| if (show_bits(gb, 12) == 0xfff) { |
| if (parse_adts_frame_header(ac, gb) < 0) { |
| av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n"); |
| return -1; |
| } |
| if (ac->m4ac.sampling_index > 12) { |
| av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); |
| return -1; |
| } |
| } |
| |
| ac->tags_mapped = 0; |
| // parse |
| while ((elem_type = get_bits(gb, 3)) != TYPE_END) { |
| elem_id = get_bits(gb, 4); |
| |
| 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); |
| return -1; |
| } |
| samples = 1024; |
| } |
| |
| switch (elem_type) { |
| |
| case TYPE_SCE: |
| err = decode_ics(ac, &che->ch[0], gb, 0, 0); |
| audio_found = 1; |
| 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: { |
| enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; |
| memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
| if ((err = decode_pce(avctx, &ac->m4ac, new_che_pos, gb))) |
| break; |
| if (ac->output_configured > OC_TRIAL_PCE) |
| 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, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE); |
| break; |
| } |
| |
| case TYPE_FIL: |
| if (elem_id == 15) |
| elem_id += get_bits(gb, 8) - 1; |
| if (get_bits_left(gb) < 8 * elem_id) { |
| av_log(avctx, AV_LOG_ERROR, overread_err); |
| return -1; |
| } |
| while (elem_id > 0) |
| elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev); |
| err = 0; /* FIXME */ |
| break; |
| |
| default: |
| err = -1; /* should not happen, but keeps compiler happy */ |
| break; |
| } |
| |
| che_prev = che; |
| elem_type_prev = elem_type; |
| |
| if (err) |
| return err; |
| |
| if (get_bits_left(gb) < 3) { |
| av_log(avctx, AV_LOG_ERROR, overread_err); |
| return -1; |
| } |
| } |
| |
| spectral_to_sample(ac); |
| |
| multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0; |
| samples <<= multiplier; |
| if (ac->output_configured < OC_LOCKED) { |
| avctx->sample_rate = ac->m4ac.sample_rate << multiplier; |
| avctx->frame_size = samples; |
| } |
| |
| data_size_tmp = samples * avctx->channels * |
| av_get_bytes_per_sample(avctx->sample_fmt); |
| if (*data_size < data_size_tmp) { |
| av_log(avctx, AV_LOG_ERROR, |
| "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", |
| *data_size, data_size_tmp); |
| return -1; |
| } |
| *data_size = data_size_tmp; |
| |
| if (samples) { |
| if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) |
| ac->fmt_conv.float_interleave(data, (const float **)ac->output_data, |
| samples, avctx->channels); |
| else |
| ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, |
| samples, avctx->channels); |
| } |
| |
| if (ac->output_configured && audio_found) |
| ac->output_configured = OC_LOCKED; |
| |
| return 0; |
| } |
| |
| static int aac_decode_frame(AVCodecContext *avctx, void *data, |
| int *data_size, AVPacket *avpkt) |
| { |
| const uint8_t *buf = avpkt->data; |
| int buf_size = avpkt->size; |
| GetBitContext gb; |
| int buf_consumed; |
| int buf_offset; |
| int err; |
| |
| init_get_bits(&gb, buf, buf_size * 8); |
| |
| if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 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]) |
| 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_ltp); |
| return 0; |
| } |
| |
| |
| #define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word |
| |
| struct LATMContext { |
| AACContext aac_ctx; ///< containing AACContext |
| int initialized; ///< initilized after a valid extradata was seen |
| |
| // parser data |
| int audio_mux_version_A; ///< LATM syntax version |
| int frame_length_type; ///< 0/1 variable/fixed frame length |
| int frame_length; ///< frame length for fixed frame length |
| }; |
| |
| static inline uint32_t latm_get_value(GetBitContext *b) |
| { |
| int length = get_bits(b, 2); |
| |
| return get_bits_long(b, (length+1)*8); |
| } |
| |
| static int latm_decode_audio_specific_config(struct LATMContext *latmctx, |
| GetBitContext *gb) |
| { |
| AVCodecContext *avctx = latmctx->aac_ctx.avctx; |
| MPEG4AudioConfig m4ac; |
| int config_start_bit = get_bits_count(gb); |
| int bits_consumed, esize; |
| |
| if (config_start_bit % 8) { |
| av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific " |
| "config not byte aligned.\n", 1); |
| return AVERROR_INVALIDDATA; |
| } else { |
| bits_consumed = |
| decode_audio_specific_config(NULL, avctx, &m4ac, |
| gb->buffer + (config_start_bit / 8), |
| get_bits_left(gb) / 8); |
| |
| if (bits_consumed < 0) |
| return AVERROR_INVALIDDATA; |
| |
| esize = (bits_consumed+7) / 8; |
| |
| if (avctx->extradata_size <= esize) { |
| av_free(avctx->extradata); |
| avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE); |
| if (!avctx->extradata) |
| return AVERROR(ENOMEM); |
| } |
| |
| avctx->extradata_size = esize; |
| memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize); |
| memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE); |
| |
| skip_bits_long(gb, bits_consumed); |
| } |
| |
| return bits_consumed; |
| } |
| |
| static int read_stream_mux_config(struct LATMContext *latmctx, |
| GetBitContext *gb) |
| { |
| int ret, audio_mux_version = get_bits(gb, 1); |
| |
| latmctx->audio_mux_version_A = 0; |
| if (audio_mux_version) |
| latmctx->audio_mux_version_A = get_bits(gb, 1); |
| |
| if (!latmctx->audio_mux_version_A) { |
| |
| if (audio_mux_version) |
| latm_get_value(gb); // taraFullness |
| |
| skip_bits(gb, 1); // allStreamSameTimeFraming |
| skip_bits(gb, 6); // numSubFrames |
| // numPrograms |
| if (get_bits(gb, 4)) { // numPrograms |
| av_log_missing_feature(latmctx->aac_ctx.avctx, |
| "multiple programs are not supported\n", 1); |
| return AVERROR_PATCHWELCOME; |
| } |
| |
| // for each program (which there is only on in DVB) |
| |
| // for each layer (which there is only on in DVB) |
| if (get_bits(gb, 3)) { // numLayer |
| av_log_missing_feature(latmctx->aac_ctx.avctx, |
| "multiple layers are not supported\n", 1); |
| return AVERROR_PATCHWELCOME; |
| } |
| |
| // for all but first stream: use_same_config = get_bits(gb, 1); |
| if (!audio_mux_version) { |
| if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0) |
| return ret; |
| } else { |
| int ascLen = latm_get_value(gb); |
| if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0) |
| return ret; |
| ascLen -= ret; |
| skip_bits_long(gb, ascLen); |
| } |
| |
| latmctx->frame_length_type = get_bits(gb, 3); |
| switch (latmctx->frame_length_type) { |
| case 0: |
| skip_bits(gb, 8); // latmBufferFullness |
| break; |
| case 1: |
| latmctx->frame_length = get_bits(gb, 9); |
| break; |
| case 3: |
| case 4: |
| case 5: |
| skip_bits(gb, 6); // CELP frame length table index |
| break; |
| case 6: |
| case 7: |
| skip_bits(gb, 1); // HVXC frame length table index |
| break; |
| } |
| |
| if (get_bits(gb, 1)) { // other data |
| if (audio_mux_version) { |
| latm_get_value(gb); // other_data_bits |
| } else { |
| int esc; |
| do { |
| esc = get_bits(gb, 1); |
| skip_bits(gb, 8); |
| } while (esc); |
| } |
| } |
| |
| if (get_bits(gb, 1)) // crc present |
| skip_bits(gb, 8); // config_crc |
| } |
| |
| return 0; |
| } |
| |
| static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb) |
| { |
| uint8_t tmp; |
| |
| if (ctx->frame_length_type == 0) { |
| int mux_slot_length = 0; |
| do { |
| tmp = get_bits(gb, 8); |
| mux_slot_length += tmp; |
| } while (tmp == 255); |
| return mux_slot_length; |
| } else if (ctx->frame_length_type == 1) { |
| return ctx->frame_length; |
| } else if (ctx->frame_length_type == 3 || |
| ctx->frame_length_type == 5 || |
| ctx->frame_length_type == 7) { |
| skip_bits(gb, 2); // mux_slot_length_coded |
| } |
| return 0; |
| } |
| |
| static int read_audio_mux_element(struct LATMContext *latmctx, |
| GetBitContext *gb) |
| { |
| int err; |
| uint8_t use_same_mux = get_bits(gb, 1); |
| if (!use_same_mux) { |
| if ((err = read_stream_mux_config(latmctx, gb)) < 0) |
| return err; |
| } else if (!latmctx->aac_ctx.avctx->extradata) { |
| av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG, |
| "no decoder config found\n"); |
| return AVERROR(EAGAIN); |
| } |
| if (latmctx->audio_mux_version_A == 0) { |
| int mux_slot_length_bytes = read_payload_length_info(latmctx, gb); |
| if (mux_slot_length_bytes * 8 > get_bits_left(gb)) { |
| av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n"); |
| return AVERROR_INVALIDDATA; |
| } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) { |
| av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, |
| "frame length mismatch %d << %d\n", |
| mux_slot_length_bytes * 8, get_bits_left(gb)); |
| return AVERROR_INVALIDDATA; |
| } |
| } |
| return 0; |
| } |
| |
| |
| static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size, |
| AVPacket *avpkt) |
| { |
| struct LATMContext *latmctx = avctx->priv_data; |
| int muxlength, err; |
| GetBitContext gb; |
| |
| if (avpkt->size == 0) |
| return 0; |
| |
| init_get_bits(&gb, avpkt->data, avpkt->size * 8); |
| |
| // check for LOAS sync word |
| if (get_bits(&gb, 11) != LOAS_SYNC_WORD) |
| return AVERROR_INVALIDDATA; |
| |
| muxlength = get_bits(&gb, 13) + 3; |
| // not enough data, the parser should have sorted this |
| if (muxlength > avpkt->size) |
| return AVERROR_INVALIDDATA; |
| |
| if ((err = read_audio_mux_element(latmctx, &gb)) < 0) |
| return err; |
| |
| if (!latmctx->initialized) { |
| if (!avctx->extradata) { |
| *out_size = 0; |
| return avpkt->size; |
| } else { |
| aac_decode_close(avctx); |
| if ((err = aac_decode_init(avctx)) < 0) |
| return err; |
| latmctx->initialized = 1; |
| } |
| } |
| |
| if (show_bits(&gb, 12) == 0xfff) { |
| av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, |
| "ADTS header detected, probably as result of configuration " |
| "misparsing\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0) |
| return err; |
| |
| return muxlength; |
| } |
| |
| av_cold static int latm_decode_init(AVCodecContext *avctx) |
| { |
| struct LATMContext *latmctx = avctx->priv_data; |
| int ret; |
| |
| ret = aac_decode_init(avctx); |
| |
| if (avctx->extradata_size > 0) { |
| latmctx->initialized = !ret; |
| } else { |
| latmctx->initialized = 0; |
| } |
| |
| return ret; |
| } |
| |
| |
| AVCodec ff_aac_decoder = { |
| "aac", |
| AVMEDIA_TYPE_AUDIO, |
| CODEC_ID_AAC, |
| sizeof(AACContext), |
| aac_decode_init, |
| NULL, |
| aac_decode_close, |
| aac_decode_frame, |
| .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), |
| .sample_fmts = (const enum AVSampleFormat[]) { |
| AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE |
| }, |
| .channel_layouts = aac_channel_layout, |
| }; |
| |
| /* |
| Note: This decoder filter is intended to decode LATM streams transferred |
| in MPEG transport streams which only contain one program. |
| To do a more complex LATM demuxing a separate LATM demuxer should be used. |
| */ |
| AVCodec ff_aac_latm_decoder = { |
| .name = "aac_latm", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = CODEC_ID_AAC_LATM, |
| .priv_data_size = sizeof(struct LATMContext), |
| .init = latm_decode_init, |
| .close = aac_decode_close, |
| .decode = latm_decode_frame, |
| .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"), |
| .sample_fmts = (const enum AVSampleFormat[]) { |
| AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE |
| }, |
| .channel_layouts = aac_channel_layout, |
| }; |