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author | Robert Swain <robert.swain@gmail.com> | 2008-08-15 08:01:31 +0000 |
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committer | Robert Swain <robert.swain@gmail.com> | 2008-08-15 08:01:31 +0000 |
commit | 9ffd5c1cee8d46128bf6b3faad8befb886763ae3 (patch) | |
tree | 8801ba1d882069c38b0cf0c3ce59986f879b0e23 | |
parent | aa6ed60895e3dcd2425ae1ad9552a04cd6c7983a (diff) | |
download | ffmpeg-9ffd5c1cee8d46128bf6b3faad8befb886763ae3.tar.gz |
More OKed AAC decoder hunks
Originally committed as revision 14774 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r-- | libavcodec/aac.c | 269 | ||||
-rw-r--r-- | libavcodec/aac.h | 3 | ||||
-rw-r--r-- | libavcodec/aactab.c | 7 | ||||
-rw-r--r-- | libavcodec/aactab.h | 9 |
4 files changed, 283 insertions, 5 deletions
diff --git a/libavcodec/aac.c b/libavcodec/aac.c index 45d6a55b70..7dd0c5ea8b 100644 --- a/libavcodec/aac.c +++ b/libavcodec/aac.c @@ -90,10 +90,6 @@ #include <math.h> #include <string.h> -#ifndef CONFIG_HARDCODED_TABLES - static float ff_aac_pow2sf_tab[316]; -#endif /* CONFIG_HARDCODED_TABLES */ - static VLC vlc_scalefactors; static VLC vlc_spectral[11]; @@ -413,6 +409,12 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) { ff_mdct_init(&ac->mdct, 11, 1); ff_mdct_init(&ac->mdct_small, 8, 1); + // 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_sine_window_init(ff_sine_1024, 1024); + ff_sine_window_init(ff_sine_128, 128); + return 0; } @@ -446,7 +448,27 @@ static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBi 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 = swb_offset_128[ac->m4ac.sampling_index]; + ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; + ics->tns_max_bands = tns_max_bands_128[ac->m4ac.sampling_index]; + } else { + ics->max_sfb = get_bits(gb, 6); + ics->num_windows = 1; + ics->swb_offset = swb_offset_1024[ac->m4ac.sampling_index]; + ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; + ics->tns_max_bands = tns_max_bands_1024[ac->m4ac.sampling_index]; if (get_bits1(gb)) { av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1); memset(ics, 0, sizeof(IndividualChannelStream)); @@ -496,6 +518,10 @@ static int decode_band_types(AACContext * ac, enum BandType band_type[120], sect_len, ics->max_sfb); return -1; } + for (; k < sect_len; k++) { + band_type [idx] = sect_band_type; + band_type_run_end[idx++] = sect_len; + } } } return 0; @@ -597,6 +623,106 @@ static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb, } /** + * 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, 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++) { + for (i = 0; i < ics->max_sfb; i++, idx++) { + const int cur_band_type = band_type[idx]; + const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4; + const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type); + int group; + if (cur_band_type == ZERO_BT) { + for (group = 0; group < ics->group_len[g]; group++) { + memset(coef + group * 128 + offsets[i], 0, (offsets[i+1] - offsets[i])*sizeof(float)); + } + }else if (cur_band_type == NOISE_BT) { + const float scale = sf[idx] / ((offsets[i+1] - offsets[i]) * PNS_MEAN_ENERGY); + for (group = 0; group < ics->group_len[g]; group++) { + for (k = offsets[i]; k < offsets[i+1]; k++) { + ac->random_state = lcg_random(ac->random_state); + coef[group*128+k] = ac->random_state * scale; + } + } + }else if (cur_band_type != INTENSITY_BT2 && cur_band_type != INTENSITY_BT) { + for (group = 0; group < ics->group_len[g]; group++) { + for (k = offsets[i]; k < offsets[i+1]; k += dim) { + const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3); + const int coef_tmp_idx = (group << 7) + k; + const float *vq_ptr; + int j; + if(index >= ff_aac_spectral_sizes[cur_band_type - 1]) { + av_log(ac->avccontext, AV_LOG_ERROR, + "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n", + cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]); + return -1; + } + vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim]; + if (is_cb_unsigned) { + for (j = 0; j < dim; j++) + if (vq_ptr[j]) + coef[coef_tmp_idx + j] = 1 - 2*(int)get_bits1(gb); + }else { + for (j = 0; j < dim; j++) + coef[coef_tmp_idx + j] = 1.0f; + } + if (cur_band_type == ESC_BT) { + for (j = 0; j < 2; j++) { + if (vq_ptr[j] == 64.0f) { + int n = 4; + /* The total length of escape_sequence must be < 22 bits according + to the specification (i.e. max is 11111111110xxxxxxxxxx). */ + while (get_bits1(gb) && n < 15) n++; + if(n == 15) { + av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); + return -1; + } + n = (1<<n) + get_bits(gb, n); + coef[coef_tmp_idx + j] *= cbrtf(fabsf(n)) * n; + }else + coef[coef_tmp_idx + j] *= vq_ptr[j]; + } + }else + for (j = 0; j < dim; j++) + coef[coef_tmp_idx + j] *= vq_ptr[j]; + for (j = 0; j < dim; j++) + coef[coef_tmp_idx + j] *= sf[idx]; + } + } + } + } + coef += ics->group_len[g]<<7; + } + + if (pulse_present) { + for(i = 0; i < pulse->num_pulse; i++){ + float co = coef_base[ pulse->pos[i] ]; + float ico = co / sqrtf(sqrtf(fabsf(co))) + pulse->amp[i]; + coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico; + } + } + return 0; +} + +/** * Decode an individual_channel_stream payload; reference: table 4.44. * * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. @@ -651,6 +777,72 @@ static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext } /** + * Mid/Side stereo decoding; reference: 4.6.8.1.3. + */ +static void apply_mid_side_stereo(ChannelElement * cpe) { + const IndividualChannelStream * ics = &cpe->ch[0].ics; + float *ch0 = cpe->ch[0].coeffs; + float *ch1 = cpe->ch[1].coeffs; + int g, i, k, 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++) { + for (k = offsets[i]; k < offsets[i+1]; k++) { + float tmp = ch0[group*128 + k] - ch1[group*128 + k]; + ch0[group*128 + k] += ch1[group*128 + k]; + ch1[group*128 + k] = tmp; + } + } + } + } + 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(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, k, 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++) + for (k = offsets[i]; k < offsets[i+1]; k++) + coef1[group*128 + k] = scale * coef0[group*128 + k]; + } + } 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. * * @param elem_id Identifies the instance of a syntax element. @@ -688,6 +880,21 @@ static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) { return 0; } +/** + * Decode coupling_channel_element; reference: table 4.8. + * + * @param elem_id Identifies the instance of a syntax element. + * + * @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, idx = 0; + 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++) { @@ -966,6 +1173,58 @@ static void apply_independent_coupling(AACContext * ac, SingleChannelElement * s sce->ret[i] += cc->coup.gain[index][0] * (cc->ch[0].ret[i] - ac->add_bias); } +/** + * channel coupling transformation interface + * + * @param index index into coupling gain array + * @param apply_coupling_method pointer to (in)dependent coupling function + */ +static void apply_channel_coupling(AACContext * ac, ChannelElement * cc, + void (*apply_coupling_method)(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index)) +{ + int c; + int index = 0; + ChannelCoupling * coup = &cc->coup; + for (c = 0; c <= coup->num_coupled; c++) { + if (ac->che[coup->type[c]][coup->id_select[c]]) { + if (coup->ch_select[c] != 2) { + apply_coupling_method(ac, &ac->che[coup->type[c]][coup->id_select[c]]->ch[0], cc, index); + if (coup->ch_select[c] != 0) + index++; + } + if (coup->ch_select[c] != 1) + apply_coupling_method(ac, &ac->che[coup->type[c]][coup->id_select[c]]->ch[1], cc, index++); + } else { + av_log(ac->avccontext, AV_LOG_ERROR, + "coupling target %sE[%d] not available\n", + coup->type[c] == TYPE_CPE ? "CP" : "SC", coup->id_select[c]); + break; + } + } +} + +/** + * Convert spectral data to float samples, applying all supported tools as appropriate. + */ +static void spectral_to_sample(AACContext * ac) { + int i, type; + for (i = 0; i < MAX_ELEM_ID; i++) { + for(type = 0; type < 4; type++) { + ChannelElement *che = ac->che[type][i]; + if(che) { + if(che->coup.coupling_point == BEFORE_TNS) + apply_channel_coupling(ac, che, apply_dependent_coupling); + 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(che->coup.coupling_point == BETWEEN_TNS_AND_IMDCT) + apply_channel_coupling(ac, che, apply_dependent_coupling); + imdct_and_windowing(ac, &che->ch[0]); + if(type == TYPE_CPE) + imdct_and_windowing(ac, &che->ch[1]); + if(che->coup.coupling_point == AFTER_IMDCT) + apply_channel_coupling(ac, che, apply_independent_coupling); } } } diff --git a/libavcodec/aac.h b/libavcodec/aac.h index 84ae8f0d77..91b18a13f6 100644 --- a/libavcodec/aac.h +++ b/libavcodec/aac.h @@ -45,6 +45,9 @@ #define MAX_CHANNELS 64 #define MAX_ELEM_ID 16 +#define TNS_MAX_ORDER 20 +#define PNS_MEAN_ENERGY 3719550720.0f // sqrt(3.0) * 1<<31 + enum AudioObjectType { AOT_NULL, // Support? Name diff --git a/libavcodec/aactab.c b/libavcodec/aactab.c index 2c30eeb51a..25f8d111b6 100644 --- a/libavcodec/aactab.c +++ b/libavcodec/aactab.c @@ -32,6 +32,9 @@ #include <stdint.h> +DECLARE_ALIGNED(16, float, ff_aac_kbd_long_1024[1024]); +DECLARE_ALIGNED(16, float, ff_aac_kbd_short_128[128]); + const uint8_t ff_aac_num_swb_1024[] = { 41, 41, 47, 49, 49, 51, 47, 47, 43, 43, 43, 40 }; @@ -983,4 +986,8 @@ const float ff_aac_pow2sf_tab[316] = { 2.68435456e+08, 3.19225354e+08, 3.79625062e+08, 4.51452825e+08, }; +#else + +float ff_aac_pow2sf_tab[316]; + #endif /* CONFIG_HARDCODED_TABLES */ diff --git a/libavcodec/aactab.h b/libavcodec/aactab.h index 8dcf94db3e..1a3a8dfe2f 100644 --- a/libavcodec/aactab.h +++ b/libavcodec/aactab.h @@ -40,6 +40,13 @@ * encoder. */ +/* @name window coefficients + * @{ + */ +DECLARE_ALIGNED(16, extern float, ff_aac_kbd_long_1024[1024]); +DECLARE_ALIGNED(16, extern float, ff_aac_kbd_short_128[128]); +// @} + /* @name number of scalefactor window bands for long and short transform windows respectively * @{ */ @@ -58,6 +65,8 @@ extern const float *ff_aac_codebook_vectors[]; #ifdef CONFIG_HARDCODED_TABLES extern const float ff_aac_pow2sf_tab[316]; +#else +extern float ff_aac_pow2sf_tab[316]; #endif /* CONFIG_HARDCODED_TABLES */ #endif /* FFMPEG_AACTAB_H */ |