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-rw-r--r--libavcodec/aacdec.c403
1 files changed, 324 insertions, 79 deletions
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index 9a931824e8..38796006e8 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -8,20 +8,20 @@
* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
* Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
@@ -74,6 +74,7 @@
* N SinuSoidal Coding (Transient, Sinusoid, Noise)
* Y Parametric Stereo
* N Direct Stream Transfer
+ * Y Enhanced AAC Low Delay (ER AAC ELD)
*
* Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
* - HE AAC v2 comprises LC AAC with Spectral Band Replication and
@@ -81,6 +82,7 @@
*/
#include "libavutil/float_dsp.h"
+#include "libavutil/opt.h"
#include "avcodec.h"
#include "internal.h"
#include "get_bits.h"
@@ -101,7 +103,6 @@
#include "aacadtsdec.h"
#include "libavutil/intfloat.h"
-#include <assert.h>
#include <errno.h>
#include <math.h>
#include <stdint.h>
@@ -109,12 +110,18 @@
#if ARCH_ARM
# include "arm/aac.h"
+#elif ARCH_MIPS
+# include "mips/aacdec_mips.h"
#endif
static VLC vlc_scalefactors;
static VLC vlc_spectral[11];
-static const char overread_err[] = "Input buffer exhausted before END element found\n";
+static int output_configure(AACContext *ac,
+ uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
+ enum OCStatus oc_type, int get_new_frame);
+
+#define overread_err "Input buffer exhausted before END element found\n"
static int count_channels(uint8_t (*layout)[3], int tags)
{
@@ -131,7 +138,7 @@ static int count_channels(uint8_t (*layout)[3], int tags)
/**
* 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 Libav channel layout.
+ * channel order to match the internal FFmpeg channel layout.
*
* @param che_pos current channel position configuration
* @param type channel element type
@@ -144,6 +151,8 @@ static av_cold int che_configure(AACContext *ac,
enum ChannelPosition che_pos,
int type, int id, int *channels)
{
+ if (*channels >= MAX_CHANNELS)
+ return AVERROR_INVALIDDATA;
if (che_pos) {
if (!ac->che[type][id]) {
if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
@@ -151,8 +160,10 @@ static av_cold int che_configure(AACContext *ac,
ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
}
if (type != TYPE_CCE) {
- if (*channels >= MAX_CHANNELS - 2)
+ if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) {
+ av_log(ac->avctx, AV_LOG_ERROR, "Too many channels\n");
return AVERROR_INVALIDDATA;
+ }
ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];
if (type == TYPE_CPE ||
(type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
@@ -185,11 +196,12 @@ static int frame_configure_elements(AVCodecContext *avctx)
/* get output buffer */
av_frame_unref(ac->frame);
+ if (!avctx->channels)
+ return 1;
+
ac->frame->nb_samples = 2048;
- if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0) {
- av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
return ret;
- }
/* map output channel pointers to AVFrame data */
for (ch = 0; ch < avctx->channels; ch++) {
@@ -428,6 +440,8 @@ static void pop_output_configuration(AACContext *ac) {
ac->oc[1] = ac->oc[0];
ac->avctx->channels = ac->oc[1].channels;
ac->avctx->channel_layout = ac->oc[1].channel_layout;
+ output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
+ ac->oc[1].status, 0);
}
}
@@ -473,7 +487,8 @@ static int output_configure(AACContext *ac,
}
memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
- avctx->channel_layout = ac->oc[1].channel_layout = layout;
+ if (layout) avctx->channel_layout = layout;
+ ac->oc[1].channel_layout = layout;
avctx->channels = ac->oc[1].channels = channels;
ac->oc[1].status = oc_type;
@@ -485,6 +500,23 @@ static int output_configure(AACContext *ac,
return 0;
}
+static void flush(AVCodecContext *avctx)
+{
+ AACContext *ac= avctx->priv_data;
+ int type, i, j;
+
+ for (type = 3; type >= 0; type--) {
+ for (i = 0; i < MAX_ELEM_ID; i++) {
+ ChannelElement *che = ac->che[type][i];
+ if (che) {
+ for (j = 0; j <= 1; j++) {
+ memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved));
+ }
+ }
+ }
+ }
+}
+
/**
* Set up channel positions based on a default channel configuration
* as specified in table 1.17.
@@ -505,6 +537,25 @@ static int set_default_channel_config(AVCodecContext *avctx,
*tags = tags_per_config[channel_config];
memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
*tags * sizeof(*layout_map));
+
+ /*
+ * AAC specification has 7.1(wide) as a default layout for 8-channel streams.
+ * However, at least Nero AAC encoder encodes 7.1 streams using the default
+ * channel config 7, mapping the side channels of the original audio stream
+ * to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD
+ * decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
+ * the incorrect streams as if they were correct (and as the encoder intended).
+ *
+ * As actual intended 7.1(wide) streams are very rare, default to assuming a
+ * 7.1 layout was intended.
+ */
+ if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
+ av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
+ " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
+ " according to the specification instead.\n", FF_COMPLIANCE_STRICT);
+ layout_map[2][2] = AAC_CHANNEL_SIDE;
+ }
+
return 0;
}
@@ -522,6 +573,8 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
int layout_map_tags;
push_output_configuration(ac);
+ av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");
+
if (set_default_channel_config(ac->avctx, layout_map,
&layout_map_tags, 2) < 0)
return NULL;
@@ -539,6 +592,8 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
int layout_map_tags;
push_output_configuration(ac);
+ av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");
+
if (set_default_channel_config(ac->avctx, layout_map,
&layout_map_tags, 1) < 0)
return NULL;
@@ -566,6 +621,12 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
* 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->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
+ if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) {
+ av_log(ac->avctx, AV_LOG_WARNING,
+ "This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n",
+ type == TYPE_SCE ? "SCE" : "LFE", elem_id);
+ ac->warned_remapping_once++;
+ }
ac->tags_mapped++;
return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
}
@@ -575,6 +636,22 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
}
case 4:
+ /* Some streams incorrectly code 4.0 audio as
+ * SCE[0] CPE[0] LFE[0]
+ * instead of
+ * SCE[0] CPE[0] SCE[1].
+ * If we seem to have encountered such a stream, transfer
+ * the SCE[1] element to the LFE[0]'s mapping */
+ if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
+ if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) {
+ av_log(ac->avctx, AV_LOG_WARNING,
+ "This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
+ type == TYPE_SCE ? "SCE" : "LFE", elem_id);
+ ac->warned_remapping_once++;
+ }
+ ac->tags_mapped++;
+ return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1];
+ }
if (ac->tags_mapped == 2 &&
ac->oc[1].m4ac.chan_config == 4 &&
type == TYPE_SCE) {
@@ -627,7 +704,7 @@ static void decode_channel_map(uint8_t layout_map[][3],
break;
default:
// AAC_CHANNEL_OFF has no channel map
- return;
+ av_assert0(0);
}
layout_map[0][0] = syn_ele;
layout_map[0][1] = get_bits(gb, 4);
@@ -673,6 +750,10 @@ static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
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, "decode_pce: " overread_err);
+ return -1;
+ }
decode_channel_map(layout_map , AAC_CHANNEL_FRONT, gb, num_front);
tags = num_front;
decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE, gb, num_side);
@@ -692,7 +773,7 @@ static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
/* 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);
+ av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
return AVERROR_INVALIDDATA;
}
skip_bits_long(gb, comment_len);
@@ -863,9 +944,9 @@ static int decode_audio_specific_config(AACContext *ac,
GetBitContext gb;
int i, ret;
- 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, "audio specific config size %d\n", bit_size >> 3);
+ for (i = 0; i < bit_size >> 3; i++)
+ av_dlog(avctx, "%02x ", data[i]);
av_dlog(avctx, "\n");
if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
@@ -929,7 +1010,7 @@ static int decode_audio_specific_config(AACContext *ac,
*
* @return Returns a 32-bit pseudorandom integer
*/
-static av_always_inline int lcg_random(int previous_val)
+static av_always_inline int lcg_random(unsigned previous_val)
{
union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
return v.s;
@@ -983,6 +1064,8 @@ static void reset_predictor_group(PredictorState *ps, int group_num)
sizeof(ff_aac_spectral_codes[num][0]), \
size);
+static void aacdec_init(AACContext *ac);
+
static av_cold int aac_decode_init(AVCodecContext *avctx)
{
AACContext *ac = avctx->priv_data;
@@ -991,6 +1074,8 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ac->avctx = avctx;
ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
+ aacdec_init(ac);
+
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (avctx->extradata_size > 0) {
@@ -1029,6 +1114,11 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
}
}
+ if (avctx->channels > MAX_CHANNELS) {
+ av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
+ return AVERROR_INVALIDDATA;
+ }
+
AAC_INIT_VLC_STATIC( 0, 304);
AAC_INIT_VLC_STATIC( 1, 270);
AAC_INIT_VLC_STATIC( 2, 550);
@@ -1044,7 +1134,10 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ff_aac_sbr_init();
ff_fmt_convert_init(&ac->fmt_conv, avctx);
- avpriv_float_dsp_init(&ac->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+ ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
+ if (!ac->fdsp) {
+ return AVERROR(ENOMEM);
+ }
ac->random_state = 0x1f2e3d4c;
@@ -1093,7 +1186,7 @@ static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
align_get_bits(gb);
if (get_bits_left(gb) < 8 * count) {
- av_log(ac->avctx, AV_LOG_ERROR, overread_err);
+ av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
return AVERROR_INVALIDDATA;
}
skip_bits_long(gb, 8 * count);
@@ -1205,13 +1298,13 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
if (ics->predictor_present) {
if (aot == AOT_AAC_MAIN) {
if (decode_prediction(ac, ics, gb)) {
- return AVERROR_INVALIDDATA;
+ goto fail;
}
} else if (aot == AOT_AAC_LC ||
aot == AOT_ER_AAC_LC) {
av_log(ac->avctx, AV_LOG_ERROR,
"Prediction is not allowed in AAC-LC.\n");
- return AVERROR_INVALIDDATA;
+ goto fail;
} else {
if (aot == AOT_ER_AAC_LD) {
av_log(ac->avctx, AV_LOG_ERROR,
@@ -1229,10 +1322,13 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
"Number of scalefactor bands in group (%d) "
"exceeds limit (%d).\n",
ics->max_sfb, ics->num_swb);
- return AVERROR_INVALIDDATA;
+ goto fail;
}
return 0;
+fail:
+ ics->max_sfb = 0;
+ return AVERROR_INVALIDDATA;
}
/**
@@ -1263,7 +1359,7 @@ static int decode_band_types(AACContext *ac, enum BandType band_type[120],
sect_len_incr = get_bits(gb, bits);
sect_end += sect_len_incr;
if (get_bits_left(gb) < 0) {
- av_log(ac->avctx, AV_LOG_ERROR, overread_err);
+ av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
return AVERROR_INVALIDDATA;
}
if (sect_end > ics->max_sfb) {
@@ -1365,12 +1461,12 @@ static int decode_pulses(Pulse *pulse, GetBitContext *gb,
return -1;
pulse->pos[0] = swb_offset[pulse_swb];
pulse->pos[0] += get_bits(gb, 5);
- if (pulse->pos[0] > 1023)
+ if (pulse->pos[0] >= swb_offset[num_swb])
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)
+ if (pulse->pos[i] >= swb_offset[num_swb])
return -1;
pulse->amp[i] = get_bits(gb, 4);
}
@@ -1556,9 +1652,9 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
cfo[k] = ac->random_state;
}
- band_energy = ac->fdsp.scalarproduct_float(cfo, cfo, off_len);
+ band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
scale = sf[idx] / sqrtf(band_energy);
- ac->fdsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
+ ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
}
} else {
const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
@@ -1704,7 +1800,7 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
}
} while (len -= 2);
- ac->fdsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
+ ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
}
}
@@ -1884,7 +1980,7 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
avpriv_request_sample(ac->avctx, "SSR");
return AVERROR_PATCHWELCOME;
}
- // I see no textual basis in the spec for this occuring after SSR gain
+ // I see no textual basis in the spec for this occurring after SSR gain
// control, but this is what both reference and real implmentations do
if (tns->present && er_syntax)
if (decode_tns(ac, tns, gb, ics) < 0)
@@ -1917,7 +2013,7 @@ static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
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->fdsp.butterflies_float(ch0 + group * 128 + offsets[i],
+ ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
ch1 + group * 128 + offsets[i],
offsets[i+1] - offsets[i]);
}
@@ -1956,7 +2052,7 @@ static void apply_intensity_stereo(AACContext *ac,
c *= 1 - 2 * cpe->ms_mask[idx];
scale = c * sce1->sf[idx];
for (group = 0; group < ics->group_len[g]; group++)
- ac->fdsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
+ ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
coef0 + group * 128 + offsets[i],
scale,
offsets[i + 1] - offsets[i]);
@@ -2168,6 +2264,32 @@ static int decode_dynamic_range(DynamicRangeControl *che_drc,
return n;
}
+static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
+ uint8_t buf[256];
+ int i, major, minor;
+
+ if (len < 13+7*8)
+ goto unknown;
+
+ get_bits(gb, 13); len -= 13;
+
+ for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
+ buf[i] = get_bits(gb, 8);
+
+ buf[i] = 0;
+ if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
+ av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
+
+ if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
+ ac->avctx->internal->skip_samples = 1024;
+ }
+
+unknown:
+ skip_bits_long(gb, len);
+
+ return 0;
+}
+
/**
* Decode extension data (incomplete); reference: table 4.51.
*
@@ -2180,7 +2302,12 @@ 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
+ int type = get_bits(gb, 4);
+
+ if (ac->avctx->debug & FF_DEBUG_STARTCODE)
+ av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
+
+ switch (type) { // extension type
case EXT_SBR_DATA_CRC:
crc_flag++;
case EXT_SBR_DATA:
@@ -2211,6 +2338,8 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
res = decode_dynamic_range(&ac->che_drc, gb);
break;
case EXT_FILL:
+ decode_fill(ac, gb, 8 * cnt - 4);
+ break;
case EXT_FILL_DATA:
case EXT_DATA_ELEMENT:
default:
@@ -2233,7 +2362,7 @@ static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
int w, filt, m, i;
int bottom, top, order, start, end, size, inc;
float lpc[TNS_MAX_ORDER];
- float tmp[TNS_MAX_ORDER + 1];
+ float tmp[TNS_MAX_ORDER+1];
for (w = 0; w < ics->num_windows; w++) {
bottom = ics->num_swb;
@@ -2291,15 +2420,15 @@ static void windowing_and_mdct_ltp(AACContext *ac, float *out,
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->fdsp.vector_fmul(in, in, lwindow_prev, 1024);
+ ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
} else {
memset(in, 0, 448 * sizeof(float));
- ac->fdsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
+ ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128);
}
if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
- ac->fdsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
+ ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
} else {
- ac->fdsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
+ ac->fdsp->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);
@@ -2325,10 +2454,10 @@ static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
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);
+ ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
if (sce->tns.present)
- apply_tns(predFreq, &sce->tns, &sce->ics, 0);
+ ac->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])
@@ -2352,17 +2481,17 @@ static void update_ltp(AACContext *ac, SingleChannelElement *sce)
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
memcpy(saved_ltp, saved, 512 * sizeof(float));
memset(saved_ltp + 576, 0, 448 * sizeof(float));
- ac->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
+ ac->fdsp->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->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
+ ac->fdsp->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->fdsp.vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
+ ac->fdsp->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];
}
@@ -2403,19 +2532,19 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
*/
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->fdsp.vector_fmul_window( out, saved, buf, lwindow_prev, 512);
+ ac->fdsp->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->fdsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
- ac->fdsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
- ac->fdsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
- ac->fdsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
- ac->fdsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
+ ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
+ ac->fdsp->vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
+ ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
+ ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
+ ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
memcpy( out + 448 + 4*128, temp, 64 * sizeof(float));
} else {
- ac->fdsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
+ ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
memcpy( out + 576, buf + 64, 448 * sizeof(float));
}
}
@@ -2423,9 +2552,9 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
// buffer update
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
memcpy( saved, temp + 64, 64 * sizeof(float));
- ac->fdsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
- ac->fdsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
- ac->fdsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
+ ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
+ ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
+ ac->fdsp->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));
@@ -2450,10 +2579,10 @@ static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
if (ics->use_kb_window[1]) {
// AAC LD uses a low overlap sine window instead of a KBD window
memcpy(out, saved, 192 * sizeof(float));
- ac->fdsp.vector_fmul_window(out + 192, saved + 192, buf, ff_sine_128, 64);
+ ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, ff_sine_128, 64);
memcpy( out + 320, buf + 64, 192 * sizeof(float));
} else {
- ac->fdsp.vector_fmul_window(out, saved, buf, ff_sine_512, 256);
+ ac->fdsp->vector_fmul_window(out, saved, buf, ff_sine_512, 256);
}
// buffer update
@@ -2476,7 +2605,7 @@ static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
// Inverse transform, mapped to the conventional IMDCT by
// Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
// "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
- // Audio, Language and Image Processing, 2008. ICALIP 2008. International Conference on
+ // International Conference on Audio, Language and Image Processing, ICALIP 2008.
// URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
for (i = 0; i < n2; i+=2) {
float temp;
@@ -2625,36 +2754,36 @@ static void spectral_to_sample(AACContext *ac)
imdct_and_window = imdct_and_windowing_eld;
break;
default:
- imdct_and_window = imdct_and_windowing;
+ imdct_and_window = ac->imdct_and_windowing;
}
for (type = 3; type >= 0; type--) {
for (i = 0; i < MAX_ELEM_ID; i++) {
ChannelElement *che = ac->che[type][i];
- if (che) {
+ if (che && che->present) {
if (type <= TYPE_CPE)
apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
if (ac->oc[1].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]);
+ ac->apply_ltp(ac, &che->ch[0]);
if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
- apply_ltp(ac, &che->ch[1]);
+ ac->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);
+ ac->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);
+ ac->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_window(ac, &che->ch[0]);
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
- update_ltp(ac, &che->ch[0]);
+ ac->update_ltp(ac, &che->ch[0]);
if (type == TYPE_CPE) {
imdct_and_window(ac, &che->ch[1]);
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
- update_ltp(ac, &che->ch[1]);
+ ac->update_ltp(ac, &che->ch[1]);
}
if (ac->oc[1].m4ac.sbr > 0) {
ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
@@ -2662,6 +2791,9 @@ static void spectral_to_sample(AACContext *ac)
}
if (type <= TYPE_CCE)
apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
+ che->present = 0;
+ } else if (che) {
+ av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
}
}
}
@@ -2676,10 +2808,12 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
size = avpriv_aac_parse_header(gb, &hdr_info);
if (size > 0) {
- if (hdr_info.num_aac_frames != 1) {
+ if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
+ // This is 2 for "VLB " audio in NSV files.
+ // See samples/nsv/vlb_audio.
avpriv_report_missing_feature(ac->avctx,
"More than one AAC RDB per ADTS frame");
- return AVERROR_PATCHWELCOME;
+ ac->warned_num_aac_frames = 1;
}
push_output_configuration(ac);
if (hdr_info.chan_config) {
@@ -2695,6 +2829,21 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
return ret;
} else {
ac->oc[1].m4ac.chan_config = 0;
+ /**
+ * dual mono frames in Japanese DTV can have chan_config 0
+ * WITHOUT specifying PCE.
+ * thus, set dual mono as default.
+ */
+ if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {
+ layout_map_tags = 2;
+ layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
+ layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
+ layout_map[0][1] = 0;
+ layout_map[1][1] = 1;
+ if (output_configure(ac, layout_map, layout_map_tags,
+ OC_TRIAL_FRAME, 0))
+ return -7;
+ }
}
ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
@@ -2751,6 +2900,7 @@ static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
elem_type, elem_id);
return AVERROR_INVALIDDATA;
}
+ che->present = 1;
if (aot != AOT_ER_AAC_ELD)
skip_bits(gb, 4);
switch (elem_type) {
@@ -2779,13 +2929,14 @@ static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
}
static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
- int *got_frame_ptr, GetBitContext *gb)
+ int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
{
AACContext *ac = avctx->priv_data;
ChannelElement *che = NULL, *che_prev = NULL;
enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
int err, elem_id;
int samples = 0, multiplier, audio_found = 0, pce_found = 0;
+ int is_dmono, sce_count = 0;
ac->frame = data;
@@ -2813,6 +2964,9 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
elem_id = get_bits(gb, 4);
+ if (avctx->debug & FF_DEBUG_STARTCODE)
+ av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
+
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",
@@ -2821,6 +2975,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
goto fail;
}
samples = 1024;
+ che->present = 1;
}
switch (elem_type) {
@@ -2828,6 +2983,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
case TYPE_SCE:
err = decode_ics(ac, &che->ch[0], gb, 0, 0);
audio_found = 1;
+ sce_count++;
break;
case TYPE_CPE:
@@ -2860,9 +3016,10 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
if (pce_found) {
av_log(avctx, AV_LOG_ERROR,
"Not evaluating a further program_config_element as this construct is dubious at best.\n");
- pop_output_configuration(ac);
} else {
err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
+ if (!err)
+ ac->oc[1].m4ac.chan_config = 0;
pce_found = 1;
}
break;
@@ -2872,7 +3029,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
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);
+ av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
err = AVERROR_INVALIDDATA;
goto fail;
}
@@ -2910,12 +3067,31 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
ac->oc[1].status = OC_LOCKED;
}
+ if (multiplier) {
+ int side_size;
+ const uint8_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);
+ if (side && side_size>=4)
+ AV_WL32(side, 2*AV_RL32(side));
+ }
+
+ *got_frame_ptr = !!samples;
if (samples) {
ac->frame->nb_samples = samples;
ac->frame->sample_rate = avctx->sample_rate;
- }
+ } else
+ av_frame_unref(ac->frame);
*got_frame_ptr = !!samples;
+ /* for dual-mono audio (SCE + SCE) */
+ is_dmono = ac->dmono_mode && sce_count == 2 &&
+ ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
+ if (is_dmono) {
+ if (ac->dmono_mode == 1)
+ ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
+ else if (ac->dmono_mode == 2)
+ ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
+ }
+
return 0;
fail:
pop_output_configuration(ac);
@@ -2936,8 +3112,12 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
AV_PKT_DATA_NEW_EXTRADATA,
&new_extradata_size);
+ int jp_dualmono_size;
+ const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
+ AV_PKT_DATA_JP_DUALMONO,
+ &jp_dualmono_size);
- if (new_extradata) {
+ if (new_extradata && 0) {
av_free(avctx->extradata);
avctx->extradata = av_mallocz(new_extradata_size +
FF_INPUT_BUFFER_PADDING_SIZE);
@@ -2954,6 +3134,15 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
}
}
+ ac->dmono_mode = 0;
+ if (jp_dualmono && jp_dualmono_size > 0)
+ ac->dmono_mode = 1 + *jp_dualmono;
+ if (ac->force_dmono_mode >= 0)
+ ac->dmono_mode = ac->force_dmono_mode;
+
+ if (INT_MAX / 8 <= buf_size)
+ return AVERROR_INVALIDDATA;
+
if ((err = init_get_bits(&gb, buf, buf_size * 8)) < 0)
return err;
@@ -2965,7 +3154,7 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
break;
default:
- err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb);
+ err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
}
if (err < 0)
return err;
@@ -2996,6 +3185,7 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
ff_mdct_end(&ac->mdct_ld);
ff_mdct_end(&ac->mdct_ltp);
ff_imdct15_uninit(&ac->mdct480);
+ av_freep(&ac->fdsp);
return 0;
}
@@ -3004,7 +3194,7 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
struct LATMContext {
AACContext aac_ctx; ///< containing AACContext
- int initialized; ///< initilized after a valid extradata was seen
+ int initialized; ///< initialized after a valid extradata was seen
// parser data
int audio_mux_version_A; ///< LATM syntax version
@@ -3053,7 +3243,11 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
- av_log(avctx, AV_LOG_INFO, "audio config changed\n");
+ if(latmctx->initialized) {
+ av_log(avctx, AV_LOG_INFO, "audio config changed\n");
+ } else {
+ av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
+ }
latmctx->initialized = 0;
esize = (bits_consumed+7) / 8;
@@ -3096,9 +3290,9 @@ static int read_stream_mux_config(struct LATMContext *latmctx,
return AVERROR_PATCHWELCOME;
}
- // for each program (which there is only on in DVB)
+ // for each program (which there is only one in DVB)
- // for each layer (which there is only on in DVB)
+ // for each layer (which there is only one in DVB)
if (get_bits(gb, 3)) { // numLayer
avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
return AVERROR_PATCHWELCOME;
@@ -3211,7 +3405,7 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out,
int muxlength, err;
GetBitContext gb;
- if ((err = init_get_bits(&gb, avpkt->data, avpkt->size * 8)) < 0)
+ if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
return err;
// check for LOAS sync word
@@ -3219,7 +3413,7 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out,
return AVERROR_INVALIDDATA;
muxlength = get_bits(&gb, 13) + 3;
- // not enough data, the parser should have sorted this
+ // not enough data, the parser should have sorted this out
if (muxlength > avpkt->size)
return AVERROR_INVALIDDATA;
@@ -3257,7 +3451,7 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out,
err = aac_decode_er_frame(avctx, out, got_frame_ptr, &gb);
break;
default:
- err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb);
+ err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt);
}
if (err < 0)
return err;
@@ -3276,6 +3470,52 @@ static av_cold int latm_decode_init(AVCodecContext *avctx)
return ret;
}
+static void aacdec_init(AACContext *c)
+{
+ c->imdct_and_windowing = imdct_and_windowing;
+ c->apply_ltp = apply_ltp;
+ c->apply_tns = apply_tns;
+ c->windowing_and_mdct_ltp = windowing_and_mdct_ltp;
+ c->update_ltp = update_ltp;
+
+ if(ARCH_MIPS)
+ ff_aacdec_init_mips(c);
+}
+/**
+ * AVOptions for Japanese DTV specific extensions (ADTS only)
+ */
+#define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
+static const AVOption options[] = {
+ {"dual_mono_mode", "Select the channel to decode for dual mono",
+ offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
+ AACDEC_FLAGS, "dual_mono_mode"},
+
+ {"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+ {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+ {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+ {"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+
+ {NULL},
+};
+
+static const AVClass aac_decoder_class = {
+ .class_name = "AAC decoder",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+static const AVProfile profiles[] = {
+ { FF_PROFILE_AAC_MAIN, "Main" },
+ { FF_PROFILE_AAC_LOW, "LC" },
+ { FF_PROFILE_AAC_SSR, "SSR" },
+ { FF_PROFILE_AAC_LTP, "LTP" },
+ { FF_PROFILE_AAC_HE, "HE-AAC" },
+ { FF_PROFILE_AAC_HE_V2, "HE-AACv2" },
+ { FF_PROFILE_AAC_LD, "LD" },
+ { FF_PROFILE_AAC_ELD, "ELD" },
+ { FF_PROFILE_UNKNOWN },
+};
AVCodec ff_aac_decoder = {
.name = "aac",
@@ -3291,6 +3531,9 @@ AVCodec ff_aac_decoder = {
},
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
+ .flush = flush,
+ .priv_class = &aac_decoder_class,
+ .profiles = profiles,
};
/*
@@ -3312,4 +3555,6 @@ AVCodec ff_aac_latm_decoder = {
},
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
+ .flush = flush,
+ .profiles = profiles,
};
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-09 21:52:26 +0000