* 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
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))))
ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
}
if (type != TYPE_CCE) {
+ if (*channels >= MAX_CHANNELS - 2)
+ 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)) {
case AAC_CHANNEL_LFE:
syn_ele = TYPE_LFE;
break;
+ default:
+ // AAC_CHANNEL_OFF has no channel map
+ return;
}
layout_map[0][0] = syn_ele;
layout_map[0][1] = get_bits(gb, 4);
case AOT_ER_AAC_LD:
res_flags = get_bits(gb, 3);
if (res_flags) {
- av_log(avctx, AV_LOG_ERROR,
- "AAC data resilience not supported (flags %x)\n",
- res_flags);
+ avpriv_report_missing_feature(avctx,
+ "AAC data resilience (flags %x)",
+ res_flags);
return AVERROR_PATCHWELCOME;
}
break;
case AOT_ER_AAC_LD:
ep_config = get_bits(gb, 2);
if (ep_config) {
- av_log(avctx, AV_LOG_ERROR,
- "epConfig %d is not supported.\n",
- 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;
+}
+
/**
* Decode audio specific configuration; reference: table 1.13.
*
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);
+ avpriv_report_missing_feature(avctx,
+ "Audio object type %s%d",
+ m4ac->sbr == 1 ? "SBR+" : "",
+ m4ac->object_type);
return AVERROR(ENOSYS);
}
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_long_512, 4.0, 512);
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
ff_init_ff_sine_windows(10);
ff_init_ff_sine_windows( 9);
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;
- }
- ics->window_sequence[1] = ics->window_sequence[0];
- ics->window_sequence[0] = get_bits(gb, 2);
- if (ac->oc[1].m4ac.object_type == 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;
+ 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->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;
- if (ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD) {
+ 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;
}
- 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 (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 ||
- ac->oc[1].m4ac.object_type == AOT_ER_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 (ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD) {
+ 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;
int ms_present)
{
int idx;
+ int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
if (ms_present == 1) {
- for (idx = 0;
- idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
- idx++)
+ for (idx = 0; idx < max_idx; idx++)
cpe->ms_mask[idx] = get_bits1(gb);
} else if (ms_present == 2) {
- memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
+ memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
}
}
TemporalNoiseShaping *tns = &sce->tns;
IndividualChannelStream *ics = &sce->ics;
float *out = sce->coeffs;
- int global_gain, er_syntax, 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 ret;
pulse_present = 0;
- 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;
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");
if (tns->present && !er_syntax)
if (decode_tns(ac, tns, gb, ics) < 0)
return AVERROR_INVALIDDATA;
- if (get_bits1(gb)) {
+ if (!eld_syntax && get_bits1(gb)) {
avpriv_request_sample(ac->avctx, "SSR");
return AVERROR_PATCHWELCOME;
}
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;
} 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;
float *in = sce->coeffs;
float *out = sce->ret;
float *saved = sce->saved;
- const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_512 : ff_sine_512;
float *buf = ac->buf_mdct;
// imdct
ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
// window overlapping
- ac->fdsp.vector_fmul_window(out, saved, buf, lwindow_prev, 256);
+ 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];
}
}
{
int i, type;
void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
- if (ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD)
+ switch (ac->oc[1].m4ac.object_type) {
+ case AOT_ER_AAC_LD:
imdct_and_window = imdct_and_windowing_ld;
- else
+ 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];
int err, i;
int samples = 1024;
int chan_config = ac->oc[1].m4ac.chan_config;
+ int aot = ac->oc[1].m4ac.object_type;
- if (ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD)
+ 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) {
elem_type, elem_id);
return AVERROR_INVALIDDATA;
}
- skip_bits(gb, 4);
+ 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);
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));
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
while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
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);
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:
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");
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