* AAC decoder
* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+ * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
*
* AAC LATM decoder
* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
#include <assert.h>
#include <errno.h>
#include <math.h>
+#include <stdint.h>
#include <string.h>
#if ARCH_ARM
av_log(avctx, AV_LOG_ERROR,
"invalid default channel configuration (%d)\n",
channel_config);
- return -1;
+ return AVERROR_INVALIDDATA;
}
*tags = tags_per_config[channel_config];
memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
comment_len = get_bits(gb, 8) * 8;
if (get_bits_left(gb) < comment_len) {
av_log(avctx, AV_LOG_ERROR, overread_err);
- return -1;
+ return AVERROR_INVALIDDATA;
}
skip_bits_long(gb, comment_len);
return tags;
MPEG4AudioConfig *m4ac,
int channel_config)
{
- int extension_flag, ret;
+ int extension_flag, ret, ep_config, res_flags;
uint8_t layout_map[MAX_ELEM_ID*4][3];
int tags = 0;
case AOT_ER_AAC_LTP:
case AOT_ER_AAC_SCALABLE:
case AOT_ER_AAC_LD:
- skip_bits(gb, 3); /* aacSectionDataResilienceFlag
- * aacScalefactorDataResilienceFlag
- * aacSpectralDataResilienceFlag
- */
+ res_flags = get_bits(gb, 3);
+ if (res_flags) {
+ avpriv_report_missing_feature(avctx,
+ "AAC data resilience (flags %x)",
+ res_flags);
+ return AVERROR_PATCHWELCOME;
+ }
break;
}
skip_bits1(gb); // extensionFlag3 (TBD in version 3)
}
+ switch (m4ac->object_type) {
+ case AOT_ER_AAC_LC:
+ case AOT_ER_AAC_LTP:
+ case AOT_ER_AAC_SCALABLE:
+ case AOT_ER_AAC_LD:
+ ep_config = get_bits(gb, 2);
+ if (ep_config) {
+ avpriv_report_missing_feature(avctx,
+ "epConfig %d", ep_config);
+ return AVERROR_PATCHWELCOME;
+ }
+ }
+ return 0;
+}
+
+static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
+ GetBitContext *gb,
+ MPEG4AudioConfig *m4ac,
+ int channel_config)
+{
+ int ret, ep_config, res_flags;
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int tags = 0;
+ const int ELDEXT_TERM = 0;
+
+ m4ac->ps = 0;
+ m4ac->sbr = 0;
+
+ if (get_bits1(gb)) { // frameLengthFlag
+ avpriv_request_sample(avctx, "960/120 MDCT window");
+ return AVERROR_PATCHWELCOME;
+ }
+
+ res_flags = get_bits(gb, 3);
+ if (res_flags) {
+ avpriv_report_missing_feature(avctx,
+ "AAC data resilience (flags %x)",
+ res_flags);
+ return AVERROR_PATCHWELCOME;
+ }
+
+ if (get_bits1(gb)) { // ldSbrPresentFlag
+ avpriv_report_missing_feature(avctx,
+ "Low Delay SBR");
+ return AVERROR_PATCHWELCOME;
+ }
+
+ while (get_bits(gb, 4) != ELDEXT_TERM) {
+ int len = get_bits(gb, 4);
+ if (len == 15)
+ len += get_bits(gb, 8);
+ if (len == 15 + 255)
+ len += get_bits(gb, 16);
+ if (get_bits_left(gb) < len * 8 + 4) {
+ av_log(ac->avctx, AV_LOG_ERROR, overread_err);
+ return AVERROR_INVALIDDATA;
+ }
+ skip_bits_long(gb, 8 * len);
+ }
+
+ if ((ret = set_default_channel_config(avctx, layout_map,
+ &tags, channel_config)))
+ return ret;
+
+ if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
+ return ret;
+
+ ep_config = get_bits(gb, 2);
+ if (ep_config) {
+ avpriv_report_missing_feature(avctx,
+ "epConfig %d", ep_config);
+ return AVERROR_PATCHWELCOME;
+ }
return 0;
}
int sync_extension)
{
GetBitContext gb;
- int i;
+ 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, "\n");
- init_get_bits(&gb, data, bit_size);
+ if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
+ return ret;
if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size,
sync_extension)) < 0)
- return -1;
+ return AVERROR_INVALIDDATA;
if (m4ac->sampling_index > 12) {
av_log(avctx, AV_LOG_ERROR,
"invalid sampling rate index %d\n",
m4ac->sampling_index);
- return -1;
+ return AVERROR_INVALIDDATA;
+ }
+ if (m4ac->object_type == AOT_ER_AAC_LD &&
+ (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
+ av_log(avctx, AV_LOG_ERROR,
+ "invalid low delay sampling rate index %d\n",
+ m4ac->sampling_index);
+ return AVERROR_INVALIDDATA;
}
skip_bits_long(&gb, i);
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;
+ case AOT_ER_AAC_LC:
+ case AOT_ER_AAC_LD:
+ if ((ret = decode_ga_specific_config(ac, avctx, &gb,
+ m4ac, m4ac->chan_config)) < 0)
+ return ret;
+ break;
+ case AOT_ER_AAC_ELD:
+ if ((ret = decode_eld_specific_config(ac, avctx, &gb,
+ m4ac, m4ac->chan_config)) < 0)
+ return ret;
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;
+ avpriv_report_missing_feature(avctx,
+ "Audio object type %s%d",
+ m4ac->sbr == 1 ? "SBR+" : "",
+ m4ac->object_type);
+ return AVERROR(ENOSYS);
}
av_dlog(avctx,
static av_cold int aac_decode_init(AVCodecContext *avctx)
{
AACContext *ac = avctx->priv_data;
+ int ret;
ac->avctx = avctx;
ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (avctx->extradata_size > 0) {
- if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
- avctx->extradata,
- avctx->extradata_size * 8, 1) < 0)
- return -1;
+ if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
+ avctx->extradata,
+ avctx->extradata_size * 8,
+ 1)) < 0)
+ return ret;
} else {
int sr, i;
uint8_t layout_map[MAX_ELEM_ID*4][3];
352);
ff_mdct_init(&ac->mdct, 11, 1, 1.0 / (32768.0 * 1024.0));
+ ff_mdct_init(&ac->mdct_ld, 10, 1, 1.0 / (32768.0 * 512.0));
ff_mdct_init(&ac->mdct_small, 8, 1, 1.0 / (32768.0 * 128.0));
ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0 * 32768.0);
// 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( 9);
ff_init_ff_sine_windows( 7);
cbrt_tableinit();
if (get_bits_left(gb) < 8 * count) {
av_log(ac->avctx, AV_LOG_ERROR, overread_err);
- return -1;
+ return AVERROR_INVALIDDATA;
}
skip_bits_long(gb, 8 * count);
return 0;
ics->predictor_reset_group > 30) {
av_log(ac->avctx, AV_LOG_ERROR,
"Invalid Predictor Reset Group.\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
}
for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
GetBitContext *gb)
{
- if (get_bits1(gb)) {
- av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
- return AVERROR_INVALIDDATA;
+ int aot = ac->oc[1].m4ac.object_type;
+ if (aot != AOT_ER_AAC_ELD) {
+ if (get_bits1(gb)) {
+ av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
+ return AVERROR_INVALIDDATA;
+ }
+ ics->window_sequence[1] = ics->window_sequence[0];
+ ics->window_sequence[0] = get_bits(gb, 2);
+ if (aot == AOT_ER_AAC_LD &&
+ ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
+ "window sequence %d found.\n", ics->window_sequence[0]);
+ ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
+ return AVERROR_INVALIDDATA;
+ }
+ ics->use_kb_window[1] = ics->use_kb_window[0];
+ ics->use_kb_window[0] = get_bits1(gb);
}
- 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) {
} else {
ics->max_sfb = get_bits(gb, 6);
ics->num_windows = 1;
- ics->swb_offset = ff_swb_offset_1024[ac->oc[1].m4ac.sampling_index];
- ics->num_swb = ff_aac_num_swb_1024[ac->oc[1].m4ac.sampling_index];
- ics->tns_max_bands = ff_tns_max_bands_1024[ac->oc[1].m4ac.sampling_index];
- ics->predictor_present = get_bits1(gb);
- ics->predictor_reset_group = 0;
+ if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
+ ics->swb_offset = ff_swb_offset_512[ac->oc[1].m4ac.sampling_index];
+ ics->num_swb = ff_aac_num_swb_512[ac->oc[1].m4ac.sampling_index];
+ ics->tns_max_bands = ff_tns_max_bands_512[ac->oc[1].m4ac.sampling_index];
+ if (!ics->num_swb || !ics->swb_offset)
+ return AVERROR_BUG;
+ } else {
+ ics->swb_offset = ff_swb_offset_1024[ac->oc[1].m4ac.sampling_index];
+ ics->num_swb = ff_aac_num_swb_1024[ac->oc[1].m4ac.sampling_index];
+ ics->tns_max_bands = ff_tns_max_bands_1024[ac->oc[1].m4ac.sampling_index];
+ }
+ if (aot != AOT_ER_AAC_ELD) {
+ ics->predictor_present = get_bits1(gb);
+ ics->predictor_reset_group = 0;
+ }
if (ics->predictor_present) {
- if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
+ if (aot == AOT_AAC_MAIN) {
if (decode_prediction(ac, ics, gb)) {
return AVERROR_INVALIDDATA;
}
- } else if (ac->oc[1].m4ac.object_type == AOT_AAC_LC) {
+ } 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;
} else {
+ if (aot == AOT_ER_AAC_LD) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "LTP in ER AAC LD not yet implemented.\n");
+ return AVERROR_PATCHWELCOME;
+ }
if ((ics->ltp.present = get_bits(gb, 1)))
decode_ltp(&ics->ltp, gb, ics->max_sfb);
}
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;
+ return AVERROR_INVALIDDATA;
}
do {
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);
- return -1;
+ return AVERROR_INVALIDDATA;
}
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;
+ return AVERROR_INVALIDDATA;
}
} while (sect_len_incr == (1 << bits) - 1);
for (; k < sect_end; k++) {
int run_end = band_type_run_end[idx];
if (band_type[idx] == ZERO_BT) {
for (; i < run_end; i++, idx++)
- sf[idx] = 0.;
+ sf[idx] = 0.0;
} else if ((band_type[idx] == INTENSITY_BT) ||
(band_type[idx] == INTENSITY_BT2)) {
for (; i < run_end; i++, idx++) {
if (offset[0] > 255U) {
av_log(ac->avctx, AV_LOG_ERROR,
"Scalefactor (%d) out of range.\n", offset[0]);
- return -1;
+ return AVERROR_INVALIDDATA;
}
sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
}
"TNS filter order %d is greater than maximum %d.\n",
tns->order[w][filt], tns_max_order);
tns->order[w][filt] = 0;
- return -1;
+ return AVERROR_INVALIDDATA;
}
if (tns->order[w][filt]) {
tns->direction[w][filt] = get_bits1(gb);
if (b > 8) {
av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
SKIP_BITS(re, gb, b + 1);
TemporalNoiseShaping *tns = &sce->tns;
IndividualChannelStream *ics = &sce->ics;
float *out = sce->coeffs;
- int global_gain, pulse_present = 0;
+ int global_gain, eld_syntax, er_syntax, pulse_present = 0;
+ int ret;
+
+ eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
+ er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
+ ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
+ ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
+ ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
/* This assignment is to silence a GCC warning about the variable being used
* uninitialized when in fact it always is.
return AVERROR_INVALIDDATA;
}
- 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;
+ if ((ret = decode_band_types(ac, sce->band_type,
+ sce->band_type_run_end, gb, ics)) < 0)
+ return ret;
+ if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
+ sce->band_type, sce->band_type_run_end)) < 0)
+ return ret;
pulse_present = 0;
if (!scale_flag) {
- if ((pulse_present = get_bits1(gb))) {
+ if (!eld_syntax && (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;
+ return AVERROR_INVALIDDATA;
}
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;
+ return AVERROR_INVALIDDATA;
}
}
- if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
- return -1;
- if (get_bits1(gb)) {
+ tns->present = get_bits1(gb);
+ if (tns->present && !er_syntax)
+ if (decode_tns(ac, tns, gb, ics) < 0)
+ return AVERROR_INVALIDDATA;
+ if (!eld_syntax && get_bits1(gb)) {
avpriv_request_sample(ac->avctx, "SSR");
return AVERROR_PATCHWELCOME;
}
+ // I see no textual basis in the spec for this occuring 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)
+ return AVERROR_INVALIDDATA;
}
if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
&pulse, ics, sce->band_type) < 0)
- return -1;
+ return AVERROR_INVALIDDATA;
if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
apply_prediction(ac, sce);
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
{
int i, ret, common_window, ms_present = 0;
+ int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
- common_window = get_bits1(gb);
+ common_window = eld_syntax || get_bits1(gb);
if (common_window) {
if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
return AVERROR_INVALIDDATA;
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;
+ return AVERROR_INVALIDDATA;
} else if (ms_present)
decode_mid_side_stereo(cpe, gb, ms_present);
}
int idx = 0;
int cge = 1;
int gain = 0;
- float gain_cache = 1.;
+ float gain_cache = 1.0;
if (c) {
cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
} else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
ac->oc[1].m4ac.sbr = 1;
ac->oc[1].m4ac.ps = 1;
+ ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
ac->oc[1].status, 1);
} else {
ac->oc[1].m4ac.sbr = 1;
+ ac->avctx->profile = FF_PROFILE_AAC_HE;
}
res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
break;
}
}
+static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
+{
+ IndividualChannelStream *ics = &sce->ics;
+ float *in = sce->coeffs;
+ float *out = sce->ret;
+ float *saved = sce->saved;
+ float *buf = ac->buf_mdct;
+
+ // imdct
+ ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
+
+ // window overlapping
+ 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);
+ memcpy( out + 320, buf + 64, 192 * sizeof(float));
+ } else {
+ ac->fdsp.vector_fmul_window(out, saved, buf, ff_sine_512, 256);
+ }
+
+ // buffer update
+ memcpy(saved, buf + 256, 256 * sizeof(float));
+}
+
+static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
+{
+ float *in = sce->coeffs;
+ float *out = sce->ret;
+ float *saved = sce->saved;
+ const float *const window = ff_aac_eld_window;
+ float *buf = ac->buf_mdct;
+ int i;
+ const int n = 512;
+ const int n2 = n >> 1;
+ const int n4 = n >> 2;
+
+ // 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
+ // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
+ for (i = 0; i < n2; i+=2) {
+ float temp;
+ temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
+ temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp;
+ }
+ ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
+ for (i = 0; i < n; i+=2) {
+ buf[i] = -buf[i];
+ }
+ // Like with the regular IMDCT at this point we still have the middle half
+ // of a transform but with even symmetry on the left and odd symmetry on
+ // the right
+
+ // window overlapping
+ // The spec says to use samples [0..511] but the reference decoder uses
+ // samples [128..639].
+ for (i = n4; i < n2; i ++) {
+ out[i - n4] = buf[n2 - 1 - i] * window[i - n4] +
+ saved[ i + n2] * window[i + n - n4] +
+ -saved[ n + n2 - 1 - i] * window[i + 2*n - n4] +
+ -saved[2*n + n2 + i] * window[i + 3*n - n4];
+ }
+ for (i = 0; i < n2; i ++) {
+ out[n4 + i] = buf[i] * window[i + n2 - n4] +
+ -saved[ n - 1 - i] * window[i + n2 + n - n4] +
+ -saved[ n + i] * window[i + n2 + 2*n - n4] +
+ saved[2*n + n - 1 - i] * window[i + n2 + 3*n - n4];
+ }
+ for (i = 0; i < n4; i ++) {
+ out[n2 + n4 + i] = buf[ i + n2] * window[i + n - n4] +
+ -saved[ n2 - 1 - i] * window[i + 2*n - n4] +
+ -saved[ n + n2 + i] * window[i + 3*n - n4];
+ }
+
+ // buffer update
+ memmove(saved + n, saved, 2 * n * sizeof(float));
+ memcpy( saved, buf, n * sizeof(float));
+}
+
/**
* Apply dependent channel coupling (applied before IMDCT).
*
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
+ // FIXME: SIMDify
dest[group * 128 + k] += gain * src[group * 128 + k];
}
}
static void spectral_to_sample(AACContext *ac)
{
int i, type;
+ void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
+ switch (ac->oc[1].m4ac.object_type) {
+ case AOT_ER_AAC_LD:
+ imdct_and_window = imdct_and_windowing_ld;
+ break;
+ case AOT_ER_AAC_ELD:
+ imdct_and_window = imdct_and_windowing_eld;
+ break;
+ default:
+ imdct_and_window = imdct_and_windowing;
+ }
for (type = 3; type >= 0; type--) {
for (i = 0; i < MAX_ELEM_ID; i++) {
ChannelElement *che = ac->che[type][i];
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]);
+ imdct_and_window(ac, &che->ch[0]);
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
update_ltp(ac, &che->ch[0]);
if (type == TYPE_CPE) {
- imdct_and_windowing(ac, &che->ch[1]);
+ imdct_and_window(ac, &che->ch[1]);
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
update_ltp(ac, &che->ch[1]);
}
int size;
AACADTSHeaderInfo hdr_info;
uint8_t layout_map[MAX_ELEM_ID*4][3];
- int layout_map_tags;
+ int layout_map_tags, ret;
size = avpriv_aac_parse_header(gb, &hdr_info);
if (size > 0) {
push_output_configuration(ac);
if (hdr_info.chan_config) {
ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
- if (set_default_channel_config(ac->avctx, layout_map,
- &layout_map_tags, hdr_info.chan_config))
- return -7;
- if (output_configure(ac, layout_map, layout_map_tags,
- FFMAX(ac->oc[1].status, OC_TRIAL_FRAME), 0))
- return -7;
+ if ((ret = set_default_channel_config(ac->avctx,
+ layout_map,
+ &layout_map_tags,
+ hdr_info.chan_config)) < 0)
+ return ret;
+ if ((ret = output_configure(ac, layout_map, layout_map_tags,
+ FFMAX(ac->oc[1].status,
+ OC_TRIAL_FRAME), 0)) < 0)
+ return ret;
} else {
ac->oc[1].m4ac.chan_config = 0;
}
return size;
}
+static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, GetBitContext *gb)
+{
+ AACContext *ac = avctx->priv_data;
+ ChannelElement *che;
+ int err, i;
+ int samples = 1024;
+ int chan_config = ac->oc[1].m4ac.chan_config;
+ int aot = ac->oc[1].m4ac.object_type;
+
+ if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
+ samples >>= 1;
+
+ ac->frame = data;
+
+ if ((err = frame_configure_elements(avctx)) < 0)
+ return err;
+
+ // The FF_PROFILE_AAC_* defines are all object_type - 1
+ // This may lead to an undefined profile being signaled
+ ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
+
+ ac->tags_mapped = 0;
+
+ if (chan_config < 0 || chan_config >= 8) {
+ avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
+ ac->oc[1].m4ac.chan_config);
+ return AVERROR_INVALIDDATA;
+ }
+ for (i = 0; i < tags_per_config[chan_config]; i++) {
+ const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
+ const int elem_id = aac_channel_layout_map[chan_config-1][i][1];
+ 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 AVERROR_INVALIDDATA;
+ }
+ if (aot != AOT_ER_AAC_ELD)
+ skip_bits(gb, 4);
+ switch (elem_type) {
+ case TYPE_SCE:
+ err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+ break;
+ case TYPE_CPE:
+ err = decode_cpe(ac, gb, che);
+ break;
+ case TYPE_LFE:
+ err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+ break;
+ }
+ if (err < 0)
+ return err;
+ }
+
+ spectral_to_sample(ac);
+
+ ac->frame->nb_samples = samples;
+ ac->frame->sample_rate = avctx->sample_rate;
+ *got_frame_ptr = 1;
+
+ skip_bits_long(gb, get_bits_left(gb));
+ return 0;
+}
+
static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
int *got_frame_ptr, GetBitContext *gb)
{
ac->frame = data;
if (show_bits(gb, 12) == 0xfff) {
- if (parse_adts_frame_header(ac, gb) < 0) {
+ if ((err = parse_adts_frame_header(ac, gb)) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
- err = -1;
goto fail;
}
if (ac->oc[1].m4ac.sampling_index > 12) {
av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
- err = -1;
+ err = AVERROR_INVALIDDATA;
goto fail;
}
}
- if (frame_configure_elements(avctx) < 0) {
- err = -1;
+ if ((err = frame_configure_elements(avctx)) < 0)
goto fail;
- }
+
+ // The FF_PROFILE_AAC_* defines are all object_type - 1
+ // This may lead to an undefined profile being signaled
+ ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
ac->tags_mapped = 0;
// parse
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);
- err = -1;
+ err = AVERROR_INVALIDDATA;
goto fail;
}
samples = 1024;
elem_id += get_bits(gb, 8) - 1;
if (get_bits_left(gb) < 8 * elem_id) {
av_log(avctx, AV_LOG_ERROR, overread_err);
- err = -1;
+ err = AVERROR_INVALIDDATA;
goto fail;
}
while (elem_id > 0)
break;
default:
- err = -1; /* should not happen, but keeps compiler happy */
+ err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
break;
}
if (get_bits_left(gb) < 3) {
av_log(avctx, AV_LOG_ERROR, overread_err);
- err = -1;
+ err = AVERROR_INVALIDDATA;
goto fail;
}
}
multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
samples <<= multiplier;
- if (samples)
- ac->frame->nb_samples = samples;
- *got_frame_ptr = !!samples;
-
if (ac->oc[1].status && audio_found) {
avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
avctx->frame_size = samples;
ac->oc[1].status = OC_LOCKED;
}
+ if (samples) {
+ ac->frame->nb_samples = samples;
+ ac->frame->sample_rate = avctx->sample_rate;
+ }
+ *got_frame_ptr = !!samples;
+
return 0;
fail:
pop_output_configuration(ac);
}
}
- init_get_bits(&gb, buf, buf_size * 8);
+ if ((err = init_get_bits(&gb, buf, buf_size * 8)) < 0)
+ return err;
- if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb)) < 0)
+ switch (ac->oc[1].m4ac.object_type) {
+ case AOT_ER_AAC_LC:
+ case AOT_ER_AAC_LTP:
+ case AOT_ER_AAC_LD:
+ case AOT_ER_AAC_ELD:
+ err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
+ break;
+ default:
+ err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb);
+ }
+ if (err < 0)
return err;
buf_consumed = (get_bits_count(&gb) + 7) >> 3;
ff_mdct_end(&ac->mdct);
ff_mdct_end(&ac->mdct_small);
+ ff_mdct_end(&ac->mdct_ld);
ff_mdct_end(&ac->mdct_ltp);
return 0;
}
if (bits_consumed < 0)
return AVERROR_INVALIDDATA;
- if (ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
+ if (!latmctx->initialized ||
+ 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");
int muxlength, err;
GetBitContext gb;
- init_get_bits(&gb, avpkt->data, avpkt->size * 8);
+ if ((err = init_get_bits(&gb, avpkt->data, avpkt->size * 8)) < 0)
+ return err;
// check for LOAS sync word
if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
AVCodec ff_aac_decoder = {
.name = "aac",
+ .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_AAC,
.priv_data_size = sizeof(AACContext),
.init = aac_decode_init,
.close = aac_decode_close,
.decode = aac_decode_frame,
- .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
},
*/
AVCodec ff_aac_latm_decoder = {
.name = "aac_latm",
+ .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_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 Coding LATM syntax)"),
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
},
projects / ffmpeg / blobdiff