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authorRobert Swain <robert.swain@gmail.com>2008-08-15 08:01:31 +0000
committerRobert Swain <robert.swain@gmail.com>2008-08-15 08:01:31 +0000
commit9ffd5c1cee8d46128bf6b3faad8befb886763ae3 (patch)
tree8801ba1d882069c38b0cf0c3ce59986f879b0e23 /libavcodec
parentaa6ed60895e3dcd2425ae1ad9552a04cd6c7983a (diff)
downloadffmpeg-9ffd5c1cee8d46128bf6b3faad8befb886763ae3.tar.gz
More OKed AAC decoder hunks
Originally committed as revision 14774 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec')
-rw-r--r--libavcodec/aac.c269
-rw-r--r--libavcodec/aac.h3
-rw-r--r--libavcodec/aactab.c7
-rw-r--r--libavcodec/aactab.h9
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 */