/* * AAC decoder * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file aac.c * AAC decoder * @author Oded Shimon ( ods15 ods15 dyndns org ) * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) */ /* * supported tools * * Support? Name * N (code in SoC repo) gain control * Y block switching * Y window shapes - standard * N window shapes - Low Delay * Y filterbank - standard * N (code in SoC repo) filterbank - Scalable Sample Rate * Y Temporal Noise Shaping * N (code in SoC repo) Long Term Prediction * Y intensity stereo * Y channel coupling * N frequency domain prediction * Y Perceptual Noise Substitution * Y Mid/Side stereo * N Scalable Inverse AAC Quantization * N Frequency Selective Switch * N upsampling filter * Y quantization & coding - AAC * N quantization & coding - TwinVQ * N quantization & coding - BSAC * N AAC Error Resilience tools * N Error Resilience payload syntax * N Error Protection tool * N CELP * N Silence Compression * N HVXC * N HVXC 4kbits/s VR * N Structured Audio tools * N Structured Audio Sample Bank Format * N MIDI * N Harmonic and Individual Lines plus Noise * N Text-To-Speech Interface * N (in progress) Spectral Band Replication * Y (not in this code) Layer-1 * Y (not in this code) Layer-2 * Y (not in this code) Layer-3 * N SinuSoidal Coding (Transient, Sinusoid, Noise) * N (planned) Parametric Stereo * N Direct Stream Transfer * * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication. * - HE AAC v2 comprises LC AAC with Spectral Band Replication and Parametric Stereo. */ #include "avcodec.h" #include "bitstream.h" #include "dsputil.h" #include "aac.h" #include "aactab.h" #include "aacdectab.h" #include "mpeg4audio.h" #include #include #include #include #ifndef CONFIG_HARDCODED_TABLES static float ff_aac_ivquant_tab[IVQUANT_SIZE]; static float ff_aac_pow2sf_tab[316]; #endif /* CONFIG_HARDCODED_TABLES */ static VLC vlc_scalefactors; static VLC vlc_spectral[11]; /** * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. * * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. * @param sce_map mono (Single Channel Element) map * @param type speaker type/position for these channels */ static void decode_channel_map(enum ChannelPosition *cpe_map, enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) { while(n--) { enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map map[get_bits(gb, 4)] = type; } } /** * Decode program configuration element; reference: table 4.2. * * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. * * @return Returns error status. 0 - OK, !0 - error */ static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], GetBitContext * gb) { int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc; skip_bits(gb, 2); // object_type ac->m4ac.sampling_index = get_bits(gb, 4); if(ac->m4ac.sampling_index > 11) { av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); return -1; } ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index]; num_front = get_bits(gb, 4); num_side = get_bits(gb, 4); num_back = get_bits(gb, 4); num_lfe = get_bits(gb, 2); num_assoc_data = get_bits(gb, 3); num_cc = get_bits(gb, 4); if (get_bits1(gb)) skip_bits(gb, 4); // mono_mixdown_tag if (get_bits1(gb)) skip_bits(gb, 4); // stereo_mixdown_tag if (get_bits1(gb)) skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); skip_bits_long(gb, 4 * num_assoc_data); decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); align_get_bits(gb); /* comment field, first byte is length */ skip_bits_long(gb, 8 * get_bits(gb, 8)); return 0; } /** * Set up channel positions based on a default channel configuration * as specified in table 1.17. * * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. * * @return Returns error status. 0 - OK, !0 - error */ static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], int channel_config) { if(channel_config < 1 || channel_config > 7) { av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", channel_config); return -1; } /* default channel configurations: * * 1ch : front center (mono) * 2ch : L + R (stereo) * 3ch : front center + L + R * 4ch : front center + L + R + back center * 5ch : front center + L + R + back stereo * 6ch : front center + L + R + back stereo + LFE * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE */ if(channel_config != 2) new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) if(channel_config > 1) new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) if(channel_config == 4) new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center if(channel_config > 4) new_che_pos[TYPE_CPE][(channel_config == 7) + 1] = AAC_CHANNEL_BACK; // back stereo if(channel_config > 5) new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE if(channel_config == 7) new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right return 0; } return -1; } if (get_bits1(gb)) // dependsOnCoreCoder skip_bits(gb, 14); // coreCoderDelay extension_flag = get_bits1(gb); if(ac->m4ac.object_type == AOT_AAC_SCALABLE || ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) skip_bits(gb, 3); // layerNr memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); if (channel_config == 0) { skip_bits(gb, 4); // element_instance_tag if((ret = decode_pce(ac, new_che_pos, gb))) return ret; } else { if((ret = set_default_channel_config(ac, new_che_pos, channel_config))) return ret; } if((ret = output_configure(ac, ac->che_pos, new_che_pos))) return ret; if (extension_flag) { switch (ac->m4ac.object_type) { case AOT_ER_BSAC: skip_bits(gb, 5); // numOfSubFrame skip_bits(gb, 11); // layer_length break; case AOT_ER_AAC_LC: case AOT_ER_AAC_LTP: case AOT_ER_AAC_SCALABLE: case AOT_ER_AAC_LD: skip_bits(gb, 3); /* aacSectionDataResilienceFlag * aacScalefactorDataResilienceFlag * aacSpectralDataResilienceFlag */ break; } skip_bits1(gb); // extensionFlag3 (TBD in version 3) } return 0; } /** * Decode audio specific configuration; reference: table 1.13. * * @param data pointer to AVCodecContext extradata * @param data_size size of AVCCodecContext extradata * * @return Returns error status. 0 - OK, !0 - error */ static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) { GetBitContext gb; int i; init_get_bits(&gb, data, data_size * 8); if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) return -1; if(ac->m4ac.sampling_index > 11) { av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); return -1; } skip_bits_long(&gb, i); switch (ac->m4ac.object_type) { case AOT_AAC_LC: if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) return -1; break; default: av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); return -1; } return 0; } static av_cold int aac_decode_init(AVCodecContext * avccontext) { AACContext * ac = avccontext->priv_data; int i; ac->avccontext = avccontext; if (avccontext->extradata_size <= 0 || decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size)) return -1; avccontext->sample_fmt = SAMPLE_FMT_S16; avccontext->sample_rate = ac->m4ac.sample_rate; avccontext->frame_size = 1024; AAC_INIT_VLC_STATIC( 0, 144); AAC_INIT_VLC_STATIC( 1, 114); AAC_INIT_VLC_STATIC( 2, 188); AAC_INIT_VLC_STATIC( 3, 180); AAC_INIT_VLC_STATIC( 4, 172); AAC_INIT_VLC_STATIC( 5, 140); AAC_INIT_VLC_STATIC( 6, 168); AAC_INIT_VLC_STATIC( 7, 114); AAC_INIT_VLC_STATIC( 8, 262); AAC_INIT_VLC_STATIC( 9, 248); AAC_INIT_VLC_STATIC(10, 384); dsputil_init(&ac->dsp, avccontext); ac->random_state = 0x1f2e3d4c; // -1024 - Compensate wrong IMDCT method. // 32768 - Required to scale values to the correct range for the bias method // for float to int16 conversion. if(ac->dsp.float_to_int16 == ff_float_to_int16_c) { ac->add_bias = 385.0f; ac->sf_scale = 1. / (-1024. * 32768.); ac->sf_offset = 0; } else { ac->add_bias = 0.0f; ac->sf_scale = 1. / -1024.; ac->sf_offset = 60; } #ifndef CONFIG_HARDCODED_TABLES for (i = 1 - IVQUANT_SIZE/2; i < IVQUANT_SIZE/2; i++) ff_aac_ivquant_tab[i + IVQUANT_SIZE/2 - 1] = cbrt(fabs(i)) * i; for (i = 0; i < 316; i++) ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.); #endif /* CONFIG_HARDCODED_TABLES */ INIT_VLC_STATIC(&vlc_scalefactors, 7, sizeof(ff_aac_scalefactor_code)/sizeof(ff_aac_scalefactor_code[0]), 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); ff_mdct_init(&ac->mdct_small, 8, 1); return 0; } /** * Skip data_stream_element; reference: table 4.10. */ static void skip_data_stream_element(GetBitContext * gb) { int byte_align = get_bits1(gb); int count = get_bits(gb, 8); if (count == 255) count += get_bits(gb, 8); if (byte_align) align_get_bits(gb); skip_bits_long(gb, 8 * count); } /** * Decode Individual Channel Stream info; reference: table 4.6. * * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. */ static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) { if (get_bits1(gb)) { av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n"); memset(ics, 0, sizeof(IndividualChannelStream)); return -1; } ics->window_sequence[1] = ics->window_sequence[0]; ics->window_sequence[0] = get_bits(gb, 2); ics->use_kb_window[1] = ics->use_kb_window[0]; ics->use_kb_window[0] = get_bits1(gb); ics->num_window_groups = 1; ics->group_len[0] = 1; return 0; } /** * inverse quantization * * @param a quantized value to be dequantized * @return Returns dequantized value. */ static inline float ivquant(int a) { if (a + (unsigned int)IVQUANT_SIZE/2 - 1 < (unsigned int)IVQUANT_SIZE - 1) return ff_aac_ivquant_tab[a + IVQUANT_SIZE/2 - 1]; else return cbrtf(fabsf(a)) * a; } /** * 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_len = k; int sect_len_incr; int sect_band_type = get_bits(gb, 4); if (sect_band_type == 12) { av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n"); return -1; } while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1) sect_len += sect_len_incr; sect_len += sect_len_incr; if (sect_len > ics->max_sfb) { av_log(ac->avccontext, AV_LOG_ERROR, "Number of bands (%d) exceeds limit (%d).\n", sect_len, ics->max_sfb); return -1; } } } return 0; } /** * Decode scalefactors; reference: table 4.47. * * @param global_gain first scalefactor value as scalefactors are differentially coded * @param band_type array of the used band type * @param band_type_run_end array of the last scalefactor band of a band type run * @param sf array of scalefactors or intensity stereo positions * * @return Returns error status. 0 - OK, !0 - error */ static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb, unsigned int global_gain, IndividualChannelStream * ics, enum BandType band_type[120], int band_type_run_end[120]) { const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); int g, i, idx = 0; int offset[3] = { global_gain, global_gain - 90, 100 }; int noise_flag = 1; static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; ics->intensity_present = 0; 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)) { ics->intensity_present = 1; for(; i < run_end; i++, idx++) { offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; if(offset[2] > 255U) { av_log(ac->avccontext, AV_LOG_ERROR, "%s (%d) out of range.\n", sf_str[2], offset[2]); return -1; } sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; } }else if(band_type[idx] == NOISE_BT) { for(; i < run_end; i++, idx++) { if(noise_flag-- > 0) offset[1] += get_bits(gb, 9) - 256; else offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; if(offset[1] > 255U) { av_log(ac->avccontext, AV_LOG_ERROR, "%s (%d) out of range.\n", sf_str[1], offset[1]); return -1; } sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset]; } }else { for(; i < run_end; i++, idx++) { offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; if(offset[0] > 255U) { av_log(ac->avccontext, AV_LOG_ERROR, "%s (%d) out of range.\n", sf_str[0], offset[0]); return -1; } sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; } } } } return 0; } /** * Decode pulse data; reference: table 4.7. */ static void decode_pulses(Pulse * pulse, GetBitContext * gb) { int i; pulse->num_pulse = get_bits(gb, 2) + 1; pulse->start = get_bits(gb, 6); for (i = 0; i < pulse->num_pulse; i++) { pulse->offset[i] = get_bits(gb, 5); pulse->amp [i] = get_bits(gb, 4); } } /** * 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) { /** * Add pulses with particular amplitudes to the quantized spectral data; reference: 4.6.3.3. * * @param pulse pointer to pulse data struct * @param icoef array of quantized spectral data */ static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualChannelStream * ics) { int i, off = ics->swb_offset[pulse->start]; for (i = 0; i < pulse->num_pulse; i++) { int ic; off += pulse->offset[i]; ic = (icoef[off] - 1)>>31; icoef[off] += (pulse->amp[i]^ic) - ic; } } /** * 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) { int icoeffs[1024]; Pulse pulse; TemporalNoiseShaping * tns = &sce->tns; IndividualChannelStream * ics = &sce->ics; float * out = sce->coeffs; int global_gain, pulse_present = 0; /* These two assignments are to silence some GCC warnings about the * variables being used uninitialised when in fact they always are. */ pulse.num_pulse = 0; pulse.start = 0; global_gain = get_bits(gb, 8); if (!common_window && !scale_flag) { if (decode_ics_info(ac, ics, gb, 0) < 0) return -1; } if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) return -1; if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) return -1; pulse_present = 0; if (!scale_flag) { if ((pulse_present = get_bits1(gb))) { if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); return -1; } decode_pulses(&pulse, gb); } if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics)) return -1; if (get_bits1(gb)) { av_log_missing_feature(ac->avccontext, "SSR", 1); return -1; } } if (decode_spectrum(ac, icoeffs, gb, ics, sce->band_type) < 0) return -1; if (pulse_present) add_pulses(icoeffs, &pulse, ics); dequant(ac, out, icoeffs, sce->sf, ics, sce->band_type); return 0; } /** * Decode a channel_pair_element; reference: table 4.4. * * @param elem_id Identifies the instance of a syntax element. * * @return Returns error status. 0 - OK, !0 - error */ static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) { int i, ret, common_window, ms_present = 0; ChannelElement * cpe; cpe = ac->che[TYPE_CPE][elem_id]; common_window = get_bits1(gb); if (common_window) { if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) return -1; i = cpe->ch[1].ics.use_kb_window[0]; cpe->ch[1].ics = cpe->ch[0].ics; cpe->ch[1].ics.use_kb_window[1] = i; ms_present = get_bits(gb, 2); if(ms_present == 3) { av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); return -1; } else if(ms_present) decode_mid_side_stereo(cpe, gb, ms_present); } if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) return ret; if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) return ret; if (common_window && ms_present) apply_mid_side_stereo(cpe); if (cpe->ch[1].ics.intensity_present) apply_intensity_stereo(cpe, ms_present); return 0; } /** * Decode Spectral Band Replication extension data; reference: table 4.55. * * @param crc flag indicating the presence of CRC checksum * @param cnt length of TYPE_FIL syntactic element in bytes * * @return Returns number of bytes consumed from the TYPE_FIL element. */ static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) { // TODO : sbr_extension implementation av_log_missing_feature(ac->avccontext, "SBR", 0); skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type return cnt; } /** * Decode dynamic range information; reference: table 4.52. * * @param cnt length of TYPE_FIL syntactic element in bytes * * @return Returns number of bytes consumed. */ static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) { int n = 1; int drc_num_bands = 1; int i; /* pce_tag_present? */ if(get_bits1(gb)) { che_drc->pce_instance_tag = get_bits(gb, 4); skip_bits(gb, 4); // tag_reserved_bits n++; } /* excluded_chns_present? */ if(get_bits1(gb)) { n += decode_drc_channel_exclusions(che_drc, gb); } /* drc_bands_present? */ if (get_bits1(gb)) { che_drc->band_incr = get_bits(gb, 4); che_drc->interpolation_scheme = get_bits(gb, 4); n++; drc_num_bands += che_drc->band_incr; for (i = 0; i < drc_num_bands; i++) { che_drc->band_top[i] = get_bits(gb, 8); n++; } } /* prog_ref_level_present? */ if (get_bits1(gb)) { che_drc->prog_ref_level = get_bits(gb, 7); skip_bits1(gb); // prog_ref_level_reserved_bits n++; } for (i = 0; i < drc_num_bands; i++) { che_drc->dyn_rng_sgn[i] = get_bits1(gb); che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); n++; } return n; } /** * Decode extension data (incomplete); reference: table 4.51. * * @param cnt length of TYPE_FIL syntactic element in bytes * * @return Returns number of bytes consumed */ static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) { int crc_flag = 0; int res = cnt; switch (get_bits(gb, 4)) { // extension type case EXT_SBR_DATA_CRC: crc_flag++; case EXT_SBR_DATA: res = decode_sbr_extension(ac, gb, crc_flag, cnt); break; case EXT_DYNAMIC_RANGE: res = decode_dynamic_range(&ac->che_drc, gb, cnt); break; case EXT_FILL: case EXT_FILL_DATA: case EXT_DATA_ELEMENT: default: skip_bits_long(gb, 8*cnt - 4); break; }; return res; } /** * 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 * lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024; const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128; const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024; const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128; float * buf = ac->buf_mdct; int i; /** * Apply dependent channel coupling (applied before IMDCT). * * @param index index into coupling gain array */ static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) { IndividualChannelStream * ics = &cc->ch[0].ics; const uint16_t * offsets = ics->swb_offset; float * dest = sce->coeffs; const float * src = cc->ch[0].coeffs; int g, i, group, k, idx = 0; if(ac->m4ac.object_type == AOT_AAC_LTP) { av_log(ac->avccontext, AV_LOG_ERROR, "Dependent coupling is not supported together with LTP\n"); return; } for (g = 0; g < ics->num_window_groups; g++) { for (i = 0; i < ics->max_sfb; i++, idx++) { if (cc->ch[0].band_type[idx] != ZERO_BT) { 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] += cc->coup.gain[index][idx] * 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 * sce, ChannelElement * cc, int index) { int i; for (i = 0; i < 1024; i++) sce->ret[i] += cc->coup.gain[index][0] * (cc->ch[0].ret[i] - ac->add_bias); } if (!ac->is_saved) { ac->is_saved = 1; *data_size = 0; return 0; } data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t); if(*data_size < data_size_tmp) { av_log(avccontext, AV_LOG_ERROR, "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", *data_size, data_size_tmp); return -1; } *data_size = data_size_tmp; ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels); return buf_size; } static av_cold int aac_decode_close(AVCodecContext * avccontext) { AACContext * ac = avccontext->priv_data; int i, type; for (i = 0; i < MAX_ELEM_ID; i++) { for(type = 0; type < 4; type++) av_freep(&ac->che[type][i]); } ff_mdct_end(&ac->mdct); ff_mdct_end(&ac->mdct_small); return 0 ; } AVCodec aac_decoder = { "aac", CODEC_TYPE_AUDIO, CODEC_ID_AAC, sizeof(AACContext), aac_decode_init, NULL, aac_decode_close, aac_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, };