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FFmpeg/libavcodec/aacdec_template.c

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/*
* 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>
* Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
*
* AAC decoder fixed-point implementation
* Copyright (c) 2013
* MIPS Technologies, Inc., California.
*
* This file is part of FFmpeg.
*
* 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.
*
* 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAC decoder
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*
* AAC decoder fixed-point implementation
* @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
* @author Nedeljko Babic ( nedeljko.babic imgtec com )
*/
/*
* supported tools
*
* Support? Name
* N (code in SoC repo) gain control
* Y block switching
* Y window shapes - standard
* N window shapes - Low Delay
* Y filterbank - standard
* N (code in SoC repo) filterbank - Scalable Sample Rate
* Y Temporal Noise Shaping
* Y Long Term Prediction
* Y intensity stereo
* Y channel coupling
* Y frequency domain prediction
* Y Perceptual Noise Substitution
* Y Mid/Side stereo
* N Scalable Inverse AAC Quantization
* N Frequency Selective Switch
* N upsampling filter
* Y quantization & coding - AAC
* N quantization & coding - TwinVQ
* N quantization & coding - BSAC
* N AAC Error Resilience tools
* N Error Resilience payload syntax
* N Error Protection tool
* N CELP
* N Silence Compression
* N HVXC
* N HVXC 4kbits/s VR
* N Structured Audio tools
* N Structured Audio Sample Bank Format
* N MIDI
* N Harmonic and Individual Lines plus Noise
* N Text-To-Speech Interface
* Y Spectral Band Replication
* Y (not in this code) Layer-1
* Y (not in this code) Layer-2
* Y (not in this code) Layer-3
* N SinuSoidal Coding (Transient, Sinusoid, Noise)
* Y Parametric Stereo
* N Direct Stream Transfer
* Y (not in fixed point code) 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
Parametric Stereo.
*/
static VLC vlc_scalefactors;
static VLC vlc_spectral[11];
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)
{
int i, sum = 0;
for (i = 0; i < tags; i++) {
int syn_ele = layout[i][0];
int pos = layout[i][2];
sum += (1 + (syn_ele == TYPE_CPE)) *
(pos != AAC_CHANNEL_OFF && pos != AAC_CHANNEL_CC);
}
return sum;
}
/**
* 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 FFmpeg channel layout.
*
* @param che_pos current channel position configuration
* @param type channel element type
* @param id channel element id
* @param channels count of the number of channels in the configuration
*
* @return Returns error status. 0 - OK, !0 - error
*/
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))))
return AVERROR(ENOMEM);
AAC_RENAME(ff_aac_sbr_ctx_init)(ac, &ac->che[type][id]->sbr);
}
if (type != TYPE_CCE) {
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)) {
ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];
}
}
} else {
if (ac->che[type][id])
AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][id]->sbr);
av_freep(&ac->che[type][id]);
}
return 0;
}
static int frame_configure_elements(AVCodecContext *avctx)
{
AACContext *ac = avctx->priv_data;
int type, id, ch, ret;
/* set channel pointers to internal buffers by default */
for (type = 0; type < 4; type++) {
for (id = 0; id < MAX_ELEM_ID; id++) {
ChannelElement *che = ac->che[type][id];
if (che) {
che->ch[0].ret = che->ch[0].ret_buf;
che->ch[1].ret = che->ch[1].ret_buf;
}
}
}
/* 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)
return ret;
/* map output channel pointers to AVFrame data */
for (ch = 0; ch < avctx->channels; ch++) {
if (ac->output_element[ch])
ac->output_element[ch]->ret = (INTFLOAT *)ac->frame->extended_data[ch];
}
return 0;
}
struct elem_to_channel {
uint64_t av_position;
uint8_t syn_ele;
uint8_t elem_id;
uint8_t aac_position;
};
static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID],
uint8_t (*layout_map)[3], int offset, uint64_t left,
uint64_t right, int pos)
{
if (layout_map[offset][0] == TYPE_CPE) {
e2c_vec[offset] = (struct elem_to_channel) {
.av_position = left | right,
.syn_ele = TYPE_CPE,
.elem_id = layout_map[offset][1],
.aac_position = pos
};
return 1;
} else {
e2c_vec[offset] = (struct elem_to_channel) {
.av_position = left,
.syn_ele = TYPE_SCE,
.elem_id = layout_map[offset][1],
.aac_position = pos
};
e2c_vec[offset + 1] = (struct elem_to_channel) {
.av_position = right,
.syn_ele = TYPE_SCE,
.elem_id = layout_map[offset + 1][1],
.aac_position = pos
};
return 2;
}
}
static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos,
int *current)
{
int num_pos_channels = 0;
int first_cpe = 0;
int sce_parity = 0;
int i;
for (i = *current; i < tags; i++) {
if (layout_map[i][2] != pos)
break;
if (layout_map[i][0] == TYPE_CPE) {
if (sce_parity) {
if (pos == AAC_CHANNEL_FRONT && !first_cpe) {
sce_parity = 0;
} else {
return -1;
}
}
num_pos_channels += 2;
first_cpe = 1;
} else {
num_pos_channels++;
sce_parity ^= 1;
}
}
if (sce_parity &&
((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE))
return -1;
*current = i;
return num_pos_channels;
}
static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
{
int i, n, total_non_cc_elements;
struct elem_to_channel e2c_vec[4 * MAX_ELEM_ID] = { { 0 } };
int num_front_channels, num_side_channels, num_back_channels;
uint64_t layout;
if (FF_ARRAY_ELEMS(e2c_vec) < tags)
return 0;
i = 0;
num_front_channels =
count_paired_channels(layout_map, tags, AAC_CHANNEL_FRONT, &i);
if (num_front_channels < 0)
return 0;
num_side_channels =
count_paired_channels(layout_map, tags, AAC_CHANNEL_SIDE, &i);
if (num_side_channels < 0)
return 0;
num_back_channels =
count_paired_channels(layout_map, tags, AAC_CHANNEL_BACK, &i);
if (num_back_channels < 0)
return 0;
if (num_side_channels == 0 && num_back_channels >= 4) {
num_side_channels = 2;
num_back_channels -= 2;
}
i = 0;
if (num_front_channels & 1) {
e2c_vec[i] = (struct elem_to_channel) {
.av_position = AV_CH_FRONT_CENTER,
.syn_ele = TYPE_SCE,
.elem_id = layout_map[i][1],
.aac_position = AAC_CHANNEL_FRONT
};
i++;
num_front_channels--;
}
if (num_front_channels >= 4) {
i += assign_pair(e2c_vec, layout_map, i,
AV_CH_FRONT_LEFT_OF_CENTER,
AV_CH_FRONT_RIGHT_OF_CENTER,
AAC_CHANNEL_FRONT);
num_front_channels -= 2;
}
if (num_front_channels >= 2) {
i += assign_pair(e2c_vec, layout_map, i,
AV_CH_FRONT_LEFT,
AV_CH_FRONT_RIGHT,
AAC_CHANNEL_FRONT);
num_front_channels -= 2;
}
while (num_front_channels >= 2) {
i += assign_pair(e2c_vec, layout_map, i,
UINT64_MAX,
UINT64_MAX,
AAC_CHANNEL_FRONT);
num_front_channels -= 2;
}
if (num_side_channels >= 2) {
i += assign_pair(e2c_vec, layout_map, i,
AV_CH_SIDE_LEFT,
AV_CH_SIDE_RIGHT,
AAC_CHANNEL_FRONT);
num_side_channels -= 2;
}
while (num_side_channels >= 2) {
i += assign_pair(e2c_vec, layout_map, i,
UINT64_MAX,
UINT64_MAX,
AAC_CHANNEL_SIDE);
num_side_channels -= 2;
}
while (num_back_channels >= 4) {
i += assign_pair(e2c_vec, layout_map, i,
UINT64_MAX,
UINT64_MAX,
AAC_CHANNEL_BACK);
num_back_channels -= 2;
}
if (num_back_channels >= 2) {
i += assign_pair(e2c_vec, layout_map, i,
AV_CH_BACK_LEFT,
AV_CH_BACK_RIGHT,
AAC_CHANNEL_BACK);
num_back_channels -= 2;
}
if (num_back_channels) {
e2c_vec[i] = (struct elem_to_channel) {
.av_position = AV_CH_BACK_CENTER,
.syn_ele = TYPE_SCE,
.elem_id = layout_map[i][1],
.aac_position = AAC_CHANNEL_BACK
};
i++;
num_back_channels--;
}
if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
e2c_vec[i] = (struct elem_to_channel) {
.av_position = AV_CH_LOW_FREQUENCY,
.syn_ele = TYPE_LFE,
.elem_id = layout_map[i][1],
.aac_position = AAC_CHANNEL_LFE
};
i++;
}
while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
e2c_vec[i] = (struct elem_to_channel) {
.av_position = UINT64_MAX,
.syn_ele = TYPE_LFE,
.elem_id = layout_map[i][1],
.aac_position = AAC_CHANNEL_LFE
};
i++;
}
// Must choose a stable sort
total_non_cc_elements = n = i;
do {
int next_n = 0;
for (i = 1; i < n; i++)
if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]);
next_n = i;
}
n = next_n;
} while (n > 0);
layout = 0;
for (i = 0; i < total_non_cc_elements; i++) {
layout_map[i][0] = e2c_vec[i].syn_ele;
layout_map[i][1] = e2c_vec[i].elem_id;
layout_map[i][2] = e2c_vec[i].aac_position;
if (e2c_vec[i].av_position != UINT64_MAX) {
layout |= e2c_vec[i].av_position;
}
}
return layout;
}
/**
* Save current output configuration if and only if it has been locked.
*/
static void push_output_configuration(AACContext *ac) {
if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) {
ac->oc[0] = ac->oc[1];
}
ac->oc[1].status = OC_NONE;
}
/**
* Restore the previous output configuration if and only if the current
* configuration is unlocked.
*/
static void pop_output_configuration(AACContext *ac) {
if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
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);
}
}
/**
* Configure output channel order based on the current program
* configuration element.
*
* @return Returns error status. 0 - OK, !0 - error
*/
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)
{
AVCodecContext *avctx = ac->avctx;
int i, channels = 0, ret;
uint64_t layout = 0;
uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }};
uint8_t type_counts[TYPE_END] = { 0 };
if (ac->oc[1].layout_map != layout_map) {
memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
ac->oc[1].layout_map_tags = tags;
}
for (i = 0; i < tags; i++) {
int type = layout_map[i][0];
int id = layout_map[i][1];
id_map[type][id] = type_counts[type]++;
}
// Try to sniff a reasonable channel order, otherwise output the
// channels in the order the PCE declared them.
if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE)
layout = sniff_channel_order(layout_map, tags);
for (i = 0; i < tags; i++) {
int type = layout_map[i][0];
int id = layout_map[i][1];
int iid = id_map[type][id];
int position = layout_map[i][2];
// Allocate or free elements depending on if they are in the
// current program configuration.
ret = che_configure(ac, position, type, iid, &channels);
if (ret < 0)
return ret;
ac->tag_che_map[type][id] = ac->che[type][iid];
}
if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
if (layout == AV_CH_FRONT_CENTER) {
layout = AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT;
} else {
layout = 0;
}
}
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;
if (get_new_frame) {
if ((ret = frame_configure_elements(ac->avctx)) < 0)
return ret;
}
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.
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int set_default_channel_config(AVCodecContext *avctx,
uint8_t (*layout_map)[3],
int *tags,
int channel_config)
{
if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
channel_config > 12) {
av_log(avctx, AV_LOG_ERROR,
"invalid default channel configuration (%d)\n",
channel_config);
return AVERROR_INVALIDDATA;
}
*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;
}
static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
{
/* For PCE based channel configurations map the channels solely based
* on tags. */
if (!ac->oc[1].m4ac.chan_config) {
return ac->tag_che_map[type][elem_id];
}
// Allow single CPE stereo files to be signalled with mono configuration.
if (!ac->tags_mapped && type == TYPE_CPE &&
ac->oc[1].m4ac.chan_config == 1) {
uint8_t layout_map[MAX_ELEM_ID*4][3];
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;
if (output_configure(ac, layout_map, layout_map_tags,
OC_TRIAL_FRAME, 1) < 0)
return NULL;
ac->oc[1].m4ac.chan_config = 2;
ac->oc[1].m4ac.ps = 0;
}
// And vice-versa
if (!ac->tags_mapped && type == TYPE_SCE &&
ac->oc[1].m4ac.chan_config == 2) {
uint8_t layout_map[MAX_ELEM_ID * 4][3];
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;
if (output_configure(ac, layout_map, layout_map_tags,
OC_TRIAL_FRAME, 1) < 0)
return NULL;
ac->oc[1].m4ac.chan_config = 1;
if (ac->oc[1].m4ac.sbr)
ac->oc[1].m4ac.ps = -1;
}
/* For indexed channel configurations map the channels solely based
* on position. */
switch (ac->oc[1].m4ac.chan_config) {
case 12:
case 7:
if (ac->tags_mapped == 3 && type == TYPE_CPE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
}
case 11:
if (ac->tags_mapped == 2 &&
ac->oc[1].m4ac.chan_config == 11 &&
type == TYPE_SCE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
}
case 6:
/* Some streams incorrectly code 5.1 audio as
* SCE[0] CPE[0] CPE[1] SCE[1]
* instead of
* SCE[0] CPE[0] CPE[1] LFE[0].
* 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];
}
case 5:
if (ac->tags_mapped == 2 && type == TYPE_CPE) {
ac->tags_mapped++;
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) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
}
case 3:
case 2:
if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
type == TYPE_CPE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
} else if (ac->oc[1].m4ac.chan_config == 2) {
return NULL;
}
case 1:
if (!ac->tags_mapped && type == TYPE_SCE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
}
default:
return NULL;
}
}
/**
* Decode an array of 4 bit element IDs, optionally interleaved with a
* stereo/mono switching bit.
*
* @param type speaker type/position for these channels
*/
static void decode_channel_map(uint8_t layout_map[][3],
enum ChannelPosition type,
GetBitContext *gb, int n)
{
while (n--) {
enum RawDataBlockType syn_ele;
switch (type) {
case AAC_CHANNEL_FRONT:
case AAC_CHANNEL_BACK:
case AAC_CHANNEL_SIDE:
syn_ele = get_bits1(gb);
break;
case AAC_CHANNEL_CC:
skip_bits1(gb);
syn_ele = TYPE_CCE;
break;
case AAC_CHANNEL_LFE:
syn_ele = TYPE_LFE;
break;
default:
// AAC_CHANNEL_OFF has no channel map
av_assert0(0);
}
layout_map[0][0] = syn_ele;
layout_map[0][1] = get_bits(gb, 4);
layout_map[0][2] = type;
layout_map++;
}
}
/**
* Decode program configuration element; reference: table 4.2.
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
uint8_t (*layout_map)[3],
GetBitContext *gb)
{
int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
int sampling_index;
int comment_len;
int tags;
skip_bits(gb, 2); // object_type
sampling_index = get_bits(gb, 4);
if (m4ac->sampling_index != sampling_index)
av_log(avctx, AV_LOG_WARNING,
"Sample rate index in program config element does not "
"match the sample rate index configured by the container.\n");
num_front = get_bits(gb, 4);
num_side = get_bits(gb, 4);
num_back = get_bits(gb, 4);
num_lfe = get_bits(gb, 2);
num_assoc_data = get_bits(gb, 3);
num_cc = get_bits(gb, 4);
if (get_bits1(gb))
skip_bits(gb, 4); // mono_mixdown_tag
if (get_bits1(gb))
skip_bits(gb, 4); // stereo_mixdown_tag
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);
tags += num_side;
decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK, gb, num_back);
tags += num_back;
decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE, gb, num_lfe);
tags += num_lfe;
skip_bits_long(gb, 4 * num_assoc_data);
decode_channel_map(layout_map + tags, AAC_CHANNEL_CC, gb, num_cc);
tags += num_cc;
align_get_bits(gb);
/* 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, "decode_pce: " overread_err);
return AVERROR_INVALIDDATA;
}
skip_bits_long(gb, comment_len);
return tags;
}
/**
* Decode GA "General Audio" specific configuration; reference: table 4.1.
*
* @param ac pointer to AACContext, may be null
* @param avctx pointer to AVCCodecContext, used for logging
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
GetBitContext *gb,
MPEG4AudioConfig *m4ac,
int channel_config)
{
int extension_flag, ret, ep_config, res_flags;
uint8_t layout_map[MAX_ELEM_ID*4][3];
int tags = 0;
if (get_bits1(gb)) { // frameLengthFlag
avpriv_request_sample(avctx, "960/120 MDCT window");
return AVERROR_PATCHWELCOME;
}
m4ac->frame_length_short = 0;
if (get_bits1(gb)) // dependsOnCoreCoder
skip_bits(gb, 14); // coreCoderDelay
extension_flag = get_bits1(gb);
if (m4ac->object_type == AOT_AAC_SCALABLE ||
m4ac->object_type == AOT_ER_AAC_SCALABLE)
skip_bits(gb, 3); // layerNr
if (channel_config == 0) {
skip_bits(gb, 4); // element_instance_tag
tags = decode_pce(avctx, m4ac, layout_map, gb);
if (tags < 0)
return tags;
} else {
if ((ret = set_default_channel_config(avctx, layout_map,
&tags, channel_config)))
return ret;
}
if (count_channels(layout_map, tags) > 1) {
m4ac->ps = 0;
} else if (m4ac->sbr == 1 && m4ac->ps == -1)
m4ac->ps = 1;
if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
return ret;
if (extension_flag) {
switch (m4ac->object_type) {
case AOT_ER_BSAC:
skip_bits(gb, 5); // numOfSubFrame
skip_bits(gb, 11); // layer_length
break;
case AOT_ER_AAC_LC:
case AOT_ER_AAC_LTP:
case AOT_ER_AAC_SCALABLE:
case AOT_ER_AAC_LD:
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 USE_FIXED
if (get_bits1(gb)) { // frameLengthFlag
avpriv_request_sample(avctx, "960/120 MDCT window");
return AVERROR_PATCHWELCOME;
}
#else
m4ac->frame_length_short = get_bits1(gb);
#endif
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(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.
*
* @param ac pointer to AACContext, may be null
* @param avctx pointer to AVCCodecContext, used for logging
* @param m4ac pointer to MPEG4AudioConfig, used for parsing
* @param data pointer to buffer holding an audio specific config
* @param bit_size size of audio specific config or data in bits
* @param sync_extension look for an appended sync extension
*
* @return Returns error status or number of consumed bits. <0 - error
*/
static int decode_audio_specific_config(AACContext *ac,
AVCodecContext *avctx,
MPEG4AudioConfig *m4ac,
const uint8_t *data, int64_t bit_size,
int sync_extension)
{
GetBitContext gb;
int i, ret;
if (bit_size < 0 || bit_size > INT_MAX) {
av_log(avctx, AV_LOG_ERROR, "Audio specific config size is invalid\n");
return AVERROR_INVALIDDATA;
}
ff_dlog(avctx, "audio specific config size %d\n", (int)bit_size >> 3);
for (i = 0; i < bit_size >> 3; i++)
ff_dlog(avctx, "%02x ", data[i]);
ff_dlog(avctx, "\n");
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 AVERROR_INVALIDDATA;
if (m4ac->sampling_index > 12) {
av_log(avctx, AV_LOG_ERROR,
"invalid sampling rate index %d\n",
m4ac->sampling_index);
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);
switch (m4ac->object_type) {
case AOT_AAC_MAIN:
case AOT_AAC_LC:
case AOT_AAC_LTP:
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:
avpriv_report_missing_feature(avctx,
"Audio object type %s%d",
m4ac->sbr == 1 ? "SBR+" : "",
m4ac->object_type);
return AVERROR(ENOSYS);
}
ff_dlog(avctx,
"AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
m4ac->sample_rate, m4ac->sbr,
m4ac->ps);
return get_bits_count(&gb);
}
/**
* linear congruential pseudorandom number generator
*
* @param previous_val pointer to the current state of the generator
*
* @return Returns a 32-bit pseudorandom integer
*/
static av_always_inline int lcg_random(unsigned previous_val)
{
union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
return v.s;
}
static void reset_all_predictors(PredictorState *ps)
{
int i;
for (i = 0; i < MAX_PREDICTORS; i++)
reset_predict_state(&ps[i]);
}
static int sample_rate_idx (int rate)
{
if (92017 <= rate) return 0;
else if (75132 <= rate) return 1;
else if (55426 <= rate) return 2;
else if (46009 <= rate) return 3;
else if (37566 <= rate) return 4;
else if (27713 <= rate) return 5;
else if (23004 <= rate) return 6;
else if (18783 <= rate) return 7;
else if (13856 <= rate) return 8;
else if (11502 <= rate) return 9;
else if (9391 <= rate) return 10;
else return 11;
}
static void reset_predictor_group(PredictorState *ps, int group_num)
{
int i;
for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
reset_predict_state(&ps[i]);
}
#define AAC_INIT_VLC_STATIC(num, size) \
INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]), \
sizeof(ff_aac_spectral_bits[num][0]), \
ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
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;
int ret;
ac->avctx = avctx;
ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
aacdec_init(ac);
#if USE_FIXED
avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
#else
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
#endif /* USE_FIXED */
if (avctx->extradata_size > 0) {
if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
avctx->extradata,
avctx->extradata_size * 8LL,
1)) < 0)
return ret;
} else {
int sr, i;
uint8_t layout_map[MAX_ELEM_ID*4][3];
int layout_map_tags;
sr = sample_rate_idx(avctx->sample_rate);
ac->oc[1].m4ac.sampling_index = sr;
ac->oc[1].m4ac.channels = avctx->channels;
ac->oc[1].m4ac.sbr = -1;
ac->oc[1].m4ac.ps = -1;
for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
if (ff_mpeg4audio_channels[i] == avctx->channels)
break;
if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
i = 0;
}
ac->oc[1].m4ac.chan_config = i;
if (ac->oc[1].m4ac.chan_config) {
int ret = set_default_channel_config(avctx, layout_map,
&layout_map_tags, ac->oc[1].m4ac.chan_config);
if (!ret)
output_configure(ac, layout_map, layout_map_tags,
OC_GLOBAL_HDR, 0);
else if (avctx->err_recognition & AV_EF_EXPLODE)
return AVERROR_INVALIDDATA;
}
}
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);
AAC_INIT_VLC_STATIC( 3, 300);
AAC_INIT_VLC_STATIC( 4, 328);
AAC_INIT_VLC_STATIC( 5, 294);
AAC_INIT_VLC_STATIC( 6, 306);
AAC_INIT_VLC_STATIC( 7, 268);
AAC_INIT_VLC_STATIC( 8, 510);
AAC_INIT_VLC_STATIC( 9, 366);
AAC_INIT_VLC_STATIC(10, 462);
AAC_RENAME(ff_aac_sbr_init)();
#if USE_FIXED
ac->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & AV_CODEC_FLAG_BITEXACT);
#else
ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
#endif /* USE_FIXED */
if (!ac->fdsp) {
return AVERROR(ENOMEM);
}
ac->random_state = 0x1f2e3d4c;
ff_aac_tableinit();
INIT_VLC_STATIC(&vlc_scalefactors, 7,
FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
ff_aac_scalefactor_bits,
sizeof(ff_aac_scalefactor_bits[0]),
sizeof(ff_aac_scalefactor_bits[0]),
ff_aac_scalefactor_code,
sizeof(ff_aac_scalefactor_code[0]),
sizeof(ff_aac_scalefactor_code[0]),
352);
AAC_RENAME_32(ff_mdct_init)(&ac->mdct, 11, 1, 1.0 / RANGE15(1024.0));
AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ld, 10, 1, 1.0 / RANGE15(512.0));
AAC_RENAME_32(ff_mdct_init)(&ac->mdct_small, 8, 1, 1.0 / RANGE15(128.0));
AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ltp, 11, 0, RANGE15(-2.0));
#if !USE_FIXED
ret = ff_imdct15_init(&ac->mdct480, 5);
if (ret < 0)
return ret;
#endif
// window initialization
AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_long_1024), 4.0, 1024);
AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_short_128), 6.0, 128);
AAC_RENAME(ff_init_ff_sine_windows)(10);
AAC_RENAME(ff_init_ff_sine_windows)( 9);
AAC_RENAME(ff_init_ff_sine_windows)( 7);
AAC_RENAME(cbrt_tableinit)();
return 0;
}
/**
* Skip data_stream_element; reference: table 4.10.
*/
static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
{
int byte_align = get_bits1(gb);
int count = get_bits(gb, 8);
if (count == 255)
count += get_bits(gb, 8);
if (byte_align)
align_get_bits(gb);
if (get_bits_left(gb) < 8 * count) {
av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
return AVERROR_INVALIDDATA;
}
skip_bits_long(gb, 8 * count);
return 0;
}
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
GetBitContext *gb)
{
int sfb;
if (get_bits1(gb)) {
ics->predictor_reset_group = get_bits(gb, 5);
if (ics->predictor_reset_group == 0 ||
ics->predictor_reset_group > 30) {
av_log(ac->avctx, AV_LOG_ERROR,
"Invalid Predictor Reset Group.\n");
return AVERROR_INVALIDDATA;
}
}
for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
ics->prediction_used[sfb] = get_bits1(gb);
}
return 0;
}
/**
* Decode Long Term Prediction data; reference: table 4.xx.
*/
static void decode_ltp(LongTermPrediction *ltp,
GetBitContext *gb, uint8_t max_sfb)
{
int sfb;
ltp->lag = get_bits(gb, 11);
ltp->coef = ltp_coef[get_bits(gb, 3)];
for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
ltp->used[sfb] = get_bits1(gb);
}
/**
* Decode Individual Channel Stream info; reference: table 4.6.
*/
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
GetBitContext *gb)
{
const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
const int aot = m4ac->object_type;
const int sampling_index = m4ac->sampling_index;
if (aot != AOT_ER_AAC_ELD) {
if (get_bits1(gb)) {
av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
if (ac->avctx->err_recognition & AV_EF_BITSTREAM)
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->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 = ff_swb_offset_128[sampling_index];
ics->num_swb = ff_aac_num_swb_128[sampling_index];
ics->tns_max_bands = ff_tns_max_bands_128[sampling_index];
ics->predictor_present = 0;
} else {
ics->max_sfb = get_bits(gb, 6);
ics->num_windows = 1;
if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
if (m4ac->frame_length_short) {
ics->swb_offset = ff_swb_offset_480[sampling_index];
ics->num_swb = ff_aac_num_swb_480[sampling_index];
ics->tns_max_bands = ff_tns_max_bands_480[sampling_index];
} else {
ics->swb_offset = ff_swb_offset_512[sampling_index];
ics->num_swb = ff_aac_num_swb_512[sampling_index];
ics->tns_max_bands = ff_tns_max_bands_512[sampling_index];
}
if (!ics->num_swb || !ics->swb_offset)
return AVERROR_BUG;
} else {
ics->swb_offset = ff_swb_offset_1024[sampling_index];
ics->num_swb = ff_aac_num_swb_1024[sampling_index];
ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index];
}
if (aot != AOT_ER_AAC_ELD) {
ics->predictor_present = get_bits1(gb);
ics->predictor_reset_group = 0;
}
if (ics->predictor_present) {
if (aot == AOT_AAC_MAIN) {
if (decode_prediction(ac, ics, gb)) {
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");
goto fail;
} 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);
}
}
}
if (ics->max_sfb > ics->num_swb) {
av_log(ac->avctx, AV_LOG_ERROR,
"Number of scalefactor bands in group (%d) "
"exceeds limit (%d).\n",
ics->max_sfb, ics->num_swb);
goto fail;
}
return 0;
fail:
ics->max_sfb = 0;
return AVERROR_INVALIDDATA;
}
/**
* Decode band types (section_data payload); reference: table 4.46.
*
* @param band_type array of the used band type
* @param band_type_run_end array of the last scalefactor band of a band type run
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_band_types(AACContext *ac, enum BandType band_type[120],
int band_type_run_end[120], GetBitContext *gb,
IndividualChannelStream *ics)
{
int g, idx = 0;
const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
for (g = 0; g < ics->num_window_groups; g++) {
int k = 0;
while (k < ics->max_sfb) {
uint8_t sect_end = k;
int sect_len_incr;
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 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, "decode_band_types: "overread_err);
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 AVERROR_INVALIDDATA;
}
} while (sect_len_incr == (1 << bits) - 1);
for (; k < sect_end; k++) {
band_type [idx] = sect_band_type;
band_type_run_end[idx++] = sect_end;
}
}
}
return 0;
}
/**
* Decode scalefactors; reference: table 4.47.
*
* @param global_gain first scalefactor value as scalefactors are differentially coded
* @param band_type array of the used band type
* @param band_type_run_end array of the last scalefactor band of a band type run
* @param sf array of scalefactors or intensity stereo positions
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_scalefactors(AACContext *ac, INTFLOAT sf[120], GetBitContext *gb,
unsigned int global_gain,
IndividualChannelStream *ics,
enum BandType band_type[120],
int band_type_run_end[120])
{
int g, i, idx = 0;
int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 };
int clipped_offset;
int noise_flag = 1;
for (g = 0; g < ics->num_window_groups; g++) {
for (i = 0; i < ics->max_sfb;) {
int run_end = band_type_run_end[idx];
if (band_type[idx] == ZERO_BT) {
for (; i < run_end; i++, idx++)
sf[idx] = FIXR(0.);
} else if ((band_type[idx] == INTENSITY_BT) ||
(band_type[idx] == INTENSITY_BT2)) {
for (; i < run_end; i++, idx++) {
offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
clipped_offset = av_clip(offset[2], -155, 100);
if (offset[2] != clipped_offset) {
avpriv_request_sample(ac->avctx,
"If you heard an audible artifact, there may be a bug in the decoder. "
"Clipped intensity stereo position (%d -> %d)",
offset[2], clipped_offset);
}
#if USE_FIXED
sf[idx] = 100 - clipped_offset;
#else
sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
#endif /* USE_FIXED */
}
} else if (band_type[idx] == NOISE_BT) {
for (; i < run_end; i++, idx++) {
if (noise_flag-- > 0)
offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE;
else
offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
clipped_offset = av_clip(offset[1], -100, 155);
if (offset[1] != clipped_offset) {
avpriv_request_sample(ac->avctx,
"If you heard an audible artifact, there may be a bug in the decoder. "
"Clipped noise gain (%d -> %d)",
offset[1], clipped_offset);
}
#if USE_FIXED
sf[idx] = -(100 + clipped_offset);
#else
sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
#endif /* USE_FIXED */
}
} else {
for (; i < run_end; i++, idx++) {
offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
if (offset[0] > 255U) {
av_log(ac->avctx, AV_LOG_ERROR,
"Scalefactor (%d) out of range.\n", offset[0]);
return AVERROR_INVALIDDATA;
}
#if USE_FIXED
sf[idx] = -offset[0];
#else
sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
#endif /* USE_FIXED */
}
}
}
}
return 0;
}
/**
* Decode pulse data; reference: table 4.7.
*/
static int decode_pulses(Pulse *pulse, GetBitContext *gb,
const uint16_t *swb_offset, int num_swb)
{
int i, pulse_swb;
pulse->num_pulse = get_bits(gb, 2) + 1;
pulse_swb = get_bits(gb, 6);
if (pulse_swb >= num_swb)
return -1;
pulse->pos[0] = swb_offset[pulse_swb];
pulse->pos[0] += get_bits(gb, 5);
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] >= swb_offset[num_swb])
return -1;
pulse->amp[i] = get_bits(gb, 4);
}
return 0;
}
/**
* Decode Temporal Noise Shaping data; reference: table 4.48.
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
GetBitContext *gb, const IndividualChannelStream *ics)
{
int w, filt, i, coef_len, coef_res, coef_compress;
const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
for (w = 0; w < ics->num_windows; w++) {
if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
coef_res = get_bits1(gb);
for (filt = 0; filt < tns->n_filt[w]; filt++) {
int tmp2_idx;
tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
av_log(ac->avctx, AV_LOG_ERROR,
"TNS filter order %d is greater than maximum %d.\n",
tns->order[w][filt], tns_max_order);
tns->order[w][filt] = 0;
return AVERROR_INVALIDDATA;
}
if (tns->order[w][filt]) {
tns->direction[w][filt] = get_bits1(gb);
coef_compress = get_bits1(gb);
coef_len = coef_res + 3 - coef_compress;
tmp2_idx = 2 * coef_compress + coef_res;
for (i = 0; i < tns->order[w][filt]; i++)
tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
}
}
}
}
return 0;
}
/**
* Decode Mid/Side data; reference: table 4.54.
*
* @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 decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
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 < max_idx; idx++)
cpe->ms_mask[idx] = get_bits1(gb);
} else if (ms_present == 2) {
memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
}
}
/**
* 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, INTFLOAT coef[1024],
GetBitContext *gb, const INTFLOAT 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;
INTFLOAT *coef_base = coef;
for (g = 0; g < ics->num_windows; g++)
memset(coef + g * 128 + offsets[ics->max_sfb], 0,
sizeof(INTFLOAT) * (c - offsets[ics->max_sfb]));
for (g = 0; g < ics->num_window_groups; g++) {
unsigned g_len = ics->group_len[g];
for (i = 0; i < ics->max_sfb; i++, idx++) {
const unsigned cbt_m1 = band_type[idx] - 1;
INTFLOAT *cfo = coef + offsets[i];
int off_len = offsets[i + 1] - offsets[i];
int group;
if (cbt_m1 >= INTENSITY_BT2 - 1) {
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
memset(cfo, 0, off_len * sizeof(*cfo));
}
} else if (cbt_m1 == NOISE_BT - 1) {
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
#if !USE_FIXED
float scale;
#endif /* !USE_FIXED */
INTFLOAT band_energy;
for (k = 0; k < off_len; k++) {
ac->random_state = lcg_random(ac->random_state);
#if USE_FIXED
cfo[k] = ac->random_state >> 3;
#else
cfo[k] = ac->random_state;
#endif /* USE_FIXED */
}
#if USE_FIXED
band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
band_energy = fixed_sqrt(band_energy, 31);
noise_scale(cfo, sf[idx], band_energy, off_len);
#else
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);
#endif /* USE_FIXED */
}
} else {
#if !USE_FIXED
const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
#endif /* !USE_FIXED */
const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
OPEN_READER(re, gb);
switch (cbt_m1 >> 1) {
case 0:
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
int code;
unsigned cb_idx;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_idx[code];
#if USE_FIXED
cf = DEC_SQUAD(cf, cb_idx);
#else
cf = VMUL4(cf, vq, cb_idx, sf + idx);
#endif /* USE_FIXED */
} while (len -= 4);
}
break;
case 1:
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
int code;
unsigned nnz;
unsigned cb_idx;
uint32_t bits;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_idx[code];
nnz = cb_idx >> 8 & 15;
bits = nnz ? GET_CACHE(re, gb) : 0;
LAST_SKIP_BITS(re, gb, nnz);
#if USE_FIXED
cf = DEC_UQUAD(cf, cb_idx, bits);
#else
cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
#endif /* USE_FIXED */
} while (len -= 4);
}
break;
case 2:
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
int code;
unsigned cb_idx;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_idx[code];
#if USE_FIXED
cf = DEC_SPAIR(cf, cb_idx);
#else
cf = VMUL2(cf, vq, cb_idx, sf + idx);
#endif /* USE_FIXED */
} while (len -= 2);
}
break;
case 3:
case 4:
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
int code;
unsigned nnz;
unsigned cb_idx;
unsigned sign;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_idx[code];
nnz = cb_idx >> 8 & 15;
sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
LAST_SKIP_BITS(re, gb, nnz);
#if USE_FIXED
cf = DEC_UPAIR(cf, cb_idx, sign);
#else
cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
#endif /* USE_FIXED */
} while (len -= 2);
}
break;
default:
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
#if USE_FIXED
int *icf = cfo;
int v;
#else
float *cf = cfo;
uint32_t *icf = (uint32_t *) cf;
#endif /* USE_FIXED */
int len = off_len;
do {
int code;
unsigned nzt, nnz;
unsigned cb_idx;
uint32_t bits;
int j;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
if (!code) {
*icf++ = 0;
*icf++ = 0;
continue;
}
cb_idx = cb_vector_idx[code];
nnz = cb_idx >> 12;
nzt = cb_idx >> 8;
bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
LAST_SKIP_BITS(re, gb, nnz);
for (j = 0; j < 2; j++) {
if (nzt & 1<<j) {
uint32_t b;
int n;
/* The total length of escape_sequence must be < 22 bits according
to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
UPDATE_CACHE(re, gb);
b = GET_CACHE(re, gb);
b = 31 - av_log2(~b);
if (b > 8) {
av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
return AVERROR_INVALIDDATA;
}
SKIP_BITS(re, gb, b + 1);
b += 4;
n = (1 << b) + SHOW_UBITS(re, gb, b);
LAST_SKIP_BITS(re, gb, b);
#if USE_FIXED
v = n;
if (bits & 1U<<31)
v = -v;
*icf++ = v;
#else
*icf++ = cbrt_tab[n] | (bits & 1U<<31);
#endif /* USE_FIXED */
bits <<= 1;
} else {
#if USE_FIXED
v = cb_idx & 15;
if (bits & 1U<<31)
v = -v;
*icf++ = v;
#else
unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
*icf++ = (bits & 1U<<31) | v;
#endif /* USE_FIXED */
bits <<= !!v;
}
cb_idx >>= 4;
}
} while (len -= 2);
#if !USE_FIXED
ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
#endif /* !USE_FIXED */
}
}
CLOSE_READER(re, gb);
}
}
coef += g_len << 7;
}
if (pulse_present) {
idx = 0;
for (i = 0; i < pulse->num_pulse; i++) {
INTFLOAT co = coef_base[ pulse->pos[i] ];
while (offsets[idx + 1] <= pulse->pos[i])
idx++;
if (band_type[idx] != NOISE_BT && sf[idx]) {
INTFLOAT ico = -pulse->amp[i];
#if USE_FIXED
if (co) {
ico = co + (co > 0 ? -ico : ico);
}
coef_base[ pulse->pos[i] ] = ico;
#else
if (co) {
co /= sf[idx];
ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
}
coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
#endif /* USE_FIXED */
}
}
}
#if USE_FIXED
coef = coef_base;
idx = 0;
for (g = 0; g < ics->num_window_groups; g++) {
unsigned g_len = ics->group_len[g];
for (i = 0; i < ics->max_sfb; i++, idx++) {
const unsigned cbt_m1 = band_type[idx] - 1;
int *cfo = coef + offsets[i];
int off_len = offsets[i + 1] - offsets[i];
int group;
if (cbt_m1 < NOISE_BT - 1) {
for (group = 0; group < (int)g_len; group++, cfo+=128) {
ac->vector_pow43(cfo, off_len);
ac->subband_scale(cfo, cfo, sf[idx], 34, off_len);
}
}
}
coef += g_len << 7;
}
#endif /* USE_FIXED */
return 0;
}
/**
* Apply AAC-Main style frequency domain prediction.
*/
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
{
int sfb, k;
if (!sce->ics.predictor_initialized) {
reset_all_predictors(sce->predictor_state);
sce->ics.predictor_initialized = 1;
}
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
for (sfb = 0;
sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
sfb++) {
for (k = sce->ics.swb_offset[sfb];
k < sce->ics.swb_offset[sfb + 1];
k++) {
predict(&sce->predictor_state[k], &sce->coeffs[k],
sce->ics.predictor_present &&
sce->ics.prediction_used[sfb]);
}
}
if (sce->ics.predictor_reset_group)
reset_predictor_group(sce->predictor_state,
sce->ics.predictor_reset_group);
} else
reset_all_predictors(sce->predictor_state);
}
/**
* Decode an individual_channel_stream payload; reference: table 4.44.
*
* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
* @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_ics(AACContext *ac, SingleChannelElement *sce,
GetBitContext *gb, int common_window, int scale_flag)
{
Pulse pulse;
TemporalNoiseShaping *tns = &sce->tns;
IndividualChannelStream *ics = &sce->ics;
INTFLOAT *out = sce->coeffs;
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.
*/
pulse.num_pulse = 0;
global_gain = get_bits(gb, 8);
if (!common_window && !scale_flag) {
if (decode_ics_info(ac, ics, gb) < 0)
return AVERROR_INVALIDDATA;
}
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 (!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 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 AVERROR_INVALIDDATA;
}
}
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 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)
return AVERROR_INVALIDDATA;
}
if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
&pulse, ics, sce->band_type) < 0)
return AVERROR_INVALIDDATA;
if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
apply_prediction(ac, sce);
return 0;
}
/**
* Mid/Side stereo decoding; reference: 4.6.8.1.3.
*/
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
{
const IndividualChannelStream *ics = &cpe->ch[0].ics;
INTFLOAT *ch0 = cpe->ch[0].coeffs;
INTFLOAT *ch1 = cpe->ch[1].coeffs;
int g, i, 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) {
#if USE_FIXED
for (group = 0; group < ics->group_len[g]; group++) {
ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i],
ch1 + group * 128 + offsets[i],
offsets[i+1] - offsets[i]);
#else
for (group = 0; group < ics->group_len[g]; group++) {
ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
ch1 + group * 128 + offsets[i],
offsets[i+1] - offsets[i]);
#endif /* USE_FIXED */
}
}
}
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(AACContext *ac,
ChannelElement *cpe, int ms_present)
{
const IndividualChannelStream *ics = &cpe->ch[1].ics;
SingleChannelElement *sce1 = &cpe->ch[1];
INTFLOAT *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
const uint16_t *offsets = ics->swb_offset;
int g, group, i, idx = 0;
int c;
INTFLOAT 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++)
#if USE_FIXED
ac->subband_scale(coef1 + group * 128 + offsets[i],
coef0 + group * 128 + offsets[i],
scale,
23,
offsets[i + 1] - offsets[i]);
#else
ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
coef0 + group * 128 + offsets[i],
scale,
offsets[i + 1] - offsets[i]);
#endif /* USE_FIXED */
}
} 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.
*
* @return Returns error status. 0 - OK, !0 - error
*/
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 = eld_syntax || get_bits1(gb);
if (common_window) {
if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
return AVERROR_INVALIDDATA;
i = cpe->ch[1].ics.use_kb_window[0];
cpe->ch[1].ics = cpe->ch[0].ics;
cpe->ch[1].ics.use_kb_window[1] = i;
if (cpe->ch[1].ics.predictor_present &&
(ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
ms_present = get_bits(gb, 2);
if (ms_present == 3) {
av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
return AVERROR_INVALIDDATA;
} else if (ms_present)
decode_mid_side_stereo(cpe, gb, ms_present);
}
if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
return ret;
if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
return ret;
if (common_window) {
if (ms_present)
apply_mid_side_stereo(ac, cpe);
if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
apply_prediction(ac, &cpe->ch[0]);
apply_prediction(ac, &cpe->ch[1]);
}
}
apply_intensity_stereo(ac, cpe, ms_present);
return 0;
}
static const float cce_scale[] = {
1.09050773266525765921, //2^(1/8)
1.18920711500272106672, //2^(1/4)
M_SQRT2,
2,
};
/**
* Decode coupling_channel_element; reference: table 4.8.
*
* @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;
int sign;
INTFLOAT 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++) {
num_gain++;
coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
coup->id_select[c] = get_bits(gb, 4);
if (coup->type[c] == TYPE_CPE) {
coup->ch_select[c] = get_bits(gb, 2);
if (coup->ch_select[c] == 3)
num_gain++;
} else
coup->ch_select[c] = 2;
}
coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
sign = get_bits(gb, 1);
scale = AAC_RENAME(cce_scale)[get_bits(gb, 2)];
if ((ret = decode_ics(ac, sce, gb, 0, 0)))
return ret;
for (c = 0; c < num_gain; c++) {
int idx = 0;
int cge = 1;
int gain = 0;
INTFLOAT gain_cache = FIXR10(1.);
if (c) {
cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
gain_cache = GET_GAIN(scale, gain);
}
if (coup->coupling_point == AFTER_IMDCT) {
coup->gain[c][0] = gain_cache;
} else {
for (g = 0; g < sce->ics.num_window_groups; g++) {
for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
if (sce->band_type[idx] != ZERO_BT) {
if (!cge) {
int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
if (t) {
int s = 1;
t = gain += t;
if (sign) {
s -= 2 * (t & 0x1);
t >>= 1;
}
gain_cache = GET_GAIN(scale, t) * s;
}
}
coup->gain[c][idx] = gain_cache;
}
}
}
}
}
return 0;
}
/**
* Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
*
* @return Returns number of bytes consumed.
*/
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
GetBitContext *gb)
{
int i;
int num_excl_chan = 0;
do {
for (i = 0; i < 7; i++)
che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
} while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
return num_excl_chan / 7;
}
/**
* Decode dynamic range information; reference: table 4.52.
*
* @return Returns number of bytes consumed.
*/
static int decode_dynamic_range(DynamicRangeControl *che_drc,
GetBitContext *gb)
{
int n = 1;
int drc_num_bands = 1;
int i;
/* pce_tag_present? */
if (get_bits1(gb)) {
che_drc->pce_instance_tag = get_bits(gb, 4);
skip_bits(gb, 4); // tag_reserved_bits
n++;
}
/* excluded_chns_present? */
if (get_bits1(gb)) {
n += decode_drc_channel_exclusions(che_drc, gb);
}
/* drc_bands_present? */
if (get_bits1(gb)) {
che_drc->band_incr = get_bits(gb, 4);
che_drc->interpolation_scheme = get_bits(gb, 4);
n++;
drc_num_bands += che_drc->band_incr;
for (i = 0; i < drc_num_bands; i++) {
che_drc->band_top[i] = get_bits(gb, 8);
n++;
}
}
/* prog_ref_level_present? */
if (get_bits1(gb)) {
che_drc->prog_ref_level = get_bits(gb, 7);
skip_bits1(gb); // prog_ref_level_reserved_bits
n++;
}
for (i = 0; i < drc_num_bands; i++) {
che_drc->dyn_rng_sgn[i] = get_bits1(gb);
che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
n++;
}
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.
*
* @param cnt length of TYPE_FIL syntactic element in bytes
*
* @return Returns number of bytes consumed
*/
static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
ChannelElement *che, enum RawDataBlockType elem_type)
{
int crc_flag = 0;
int res = cnt;
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:
if (!che) {
av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
return res;
} else if (!ac->oc[1].m4ac.sbr) {
av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
skip_bits_long(gb, 8 * cnt - 4);
return res;
} else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
skip_bits_long(gb, 8 * cnt - 4);
return res;
} 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 = AAC_RENAME(ff_decode_sbr_extension)(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
break;
case EXT_DYNAMIC_RANGE:
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:
skip_bits_long(gb, 8 * cnt - 4);
break;
};
return res;
}
/**
* Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
*
* @param decode 1 if tool is used normally, 0 if tool is used in LTP.
* @param coef spectral coefficients
*/
static void apply_tns(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
IndividualChannelStream *ics, int decode)
{
const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
int w, filt, m, i;
int bottom, top, order, start, end, size, inc;
INTFLOAT lpc[TNS_MAX_ORDER];
INTFLOAT tmp[TNS_MAX_ORDER+1];
for (w = 0; w < ics->num_windows; w++) {
bottom = ics->num_swb;
for (filt = 0; filt < tns->n_filt[w]; filt++) {
top = bottom;
bottom = FFMAX(0, top - tns->length[w][filt]);
order = tns->order[w][filt];
if (order == 0)
continue;
// tns_decode_coef
AAC_RENAME(compute_lpc_coefs)(tns->coef[w][filt], order, lpc, 0, 0, 0);
start = ics->swb_offset[FFMIN(bottom, mmm)];
end = ics->swb_offset[FFMIN( top, mmm)];
if ((size = end - start) <= 0)
continue;
if (tns->direction[w][filt]) {
inc = -1;
start = end - 1;
} else {
inc = 1;
}
start += w * 128;
if (decode) {
// ar filter
for (m = 0; m < size; m++, start += inc)
for (i = 1; i <= FFMIN(m, order); i++)
coef[start] -= AAC_MUL26(coef[start - i * inc], lpc[i - 1]);
} else {
// ma filter
for (m = 0; m < size; m++, start += inc) {
tmp[0] = coef[start];
for (i = 1; i <= FFMIN(m, order); i++)
coef[start] += AAC_MUL26(tmp[i], lpc[i - 1]);
for (i = order; i > 0; i--)
tmp[i] = tmp[i - 1];
}
}
}
}
}
/**
* Apply windowing and MDCT to obtain the spectral
* coefficient from the predicted sample by LTP.
*/
static void windowing_and_mdct_ltp(AACContext *ac, INTFLOAT *out,
INTFLOAT *in, IndividualChannelStream *ics)
{
const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
} else {
memset(in, 0, 448 * sizeof(*in));
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);
} else {
ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
memset(in + 1024 + 576, 0, 448 * sizeof(*in));
}
ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
}
/**
* Apply the long term prediction
*/
static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
{
const LongTermPrediction *ltp = &sce->ics.ltp;
const uint16_t *offsets = sce->ics.swb_offset;
int i, sfb;
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
INTFLOAT *predTime = sce->ret;
INTFLOAT *predFreq = ac->buf_mdct;
int16_t num_samples = 2048;
if (ltp->lag < 1024)
num_samples = ltp->lag + 1024;
for (i = 0; i < num_samples; i++)
predTime[i] = AAC_MUL30(sce->ltp_state[i + 2048 - ltp->lag], ltp->coef);
memset(&predTime[i], 0, (2048 - i) * sizeof(*predTime));
ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
if (sce->tns.present)
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])
for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
sce->coeffs[i] += predFreq[i];
}
}
/**
* Update the LTP buffer for next frame
*/
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
{
IndividualChannelStream *ics = &sce->ics;
INTFLOAT *saved = sce->saved;
INTFLOAT *saved_ltp = sce->coeffs;
const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
int i;
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
memcpy(saved_ltp, saved, 512 * sizeof(*saved_ltp));
memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
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] = AAC_MUL31(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(*saved_ltp));
memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
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] = AAC_MUL31(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);
for (i = 0; i < 512; i++)
saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], lwindow[511 - i]);
}
memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
}
/**
* Conduct IMDCT and windowing.
*/
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
{
IndividualChannelStream *ics = &sce->ics;
INTFLOAT *in = sce->coeffs;
INTFLOAT *out = sce->ret;
INTFLOAT *saved = sce->saved;
const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
INTFLOAT *buf = ac->buf_mdct;
INTFLOAT *temp = ac->temp;
int i;
// imdct
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
for (i = 0; i < 1024; i += 128)
ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
} else {
ac->mdct.imdct_half(&ac->mdct, buf, in);
#if USE_FIXED
for (i=0; i<1024; i++)
buf[i] = (buf[i] + 4) >> 3;
#endif /* USE_FIXED */
}
/* window overlapping
* NOTE: To simplify the overlapping code, all 'meaningless' short to long
* and long to short transitions are considered to be short to short
* transitions. This leaves just two cases (long to long and short to short)
* with a little special sauce for EIGHT_SHORT_SEQUENCE.
*/
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);
} else {
memcpy( out, saved, 448 * sizeof(*out));
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);
memcpy( out + 448 + 4*128, temp, 64 * sizeof(*out));
} else {
ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
memcpy( out + 576, buf + 64, 448 * sizeof(*out));
}
}
// buffer update
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
memcpy( saved, temp + 64, 64 * sizeof(*saved));
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(*saved));
} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
memcpy( saved, buf + 512, 448 * sizeof(*saved));
memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
} else { // LONG_STOP or ONLY_LONG
memcpy( saved, buf + 512, 512 * sizeof(*saved));
}
}
static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
{
IndividualChannelStream *ics = &sce->ics;
INTFLOAT *in = sce->coeffs;
INTFLOAT *out = sce->ret;
INTFLOAT *saved = sce->saved;
INTFLOAT *buf = ac->buf_mdct;
#if USE_FIXED
int i;
#endif /* USE_FIXED */
// imdct
ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
#if USE_FIXED
for (i = 0; i < 1024; i++)
buf[i] = (buf[i] + 2) >> 2;
#endif /* USE_FIXED */
// 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(*out));
ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, AAC_RENAME(ff_sine_128), 64);
memcpy( out + 320, buf + 64, 192 * sizeof(*out));
} else {
ac->fdsp->vector_fmul_window(out, saved, buf, AAC_RENAME(ff_sine_512), 256);
}
// buffer update
memcpy(saved, buf + 256, 256 * sizeof(*saved));
}
static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
{
INTFLOAT *in = sce->coeffs;
INTFLOAT *out = sce->ret;
INTFLOAT *saved = sce->saved;
INTFLOAT *buf = ac->buf_mdct;
int i;
const int n = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
const int n2 = n >> 1;
const int n4 = n >> 2;
const INTFLOAT *const window = n == 480 ? AAC_RENAME(ff_aac_eld_window_480) :
AAC_RENAME(ff_aac_eld_window_512);
// 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,"
// 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) {
INTFLOAT 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;
}
#if !USE_FIXED
if (n == 480)
ac->mdct480->imdct_half(ac->mdct480, buf, in, 1, -1.f/(16*1024*960));
else
#endif
ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
#if USE_FIXED
for (i = 0; i < 1024; i++)
buf[i] = (buf[i] + 1) >> 1;
#endif /* USE_FIXED */
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] = AAC_MUL31( buf[ n2 - 1 - i] , window[i - n4]) +
AAC_MUL31( saved[ i + n2] , window[i + n - n4]) +
AAC_MUL31(-saved[n + n2 - 1 - i] , window[i + 2*n - n4]) +
AAC_MUL31(-saved[ 2*n + n2 + i] , window[i + 3*n - n4]);
}
for (i = 0; i < n2; i ++) {
out[n4 + i] = AAC_MUL31( buf[ i] , window[i + n2 - n4]) +
AAC_MUL31(-saved[ n - 1 - i] , window[i + n2 + n - n4]) +
AAC_MUL31(-saved[ n + i] , window[i + n2 + 2*n - n4]) +
AAC_MUL31( saved[2*n + n - 1 - i] , window[i + n2 + 3*n - n4]);
}
for (i = 0; i < n4; i ++) {
out[n2 + n4 + i] = AAC_MUL31( buf[ i + n2] , window[i + n - n4]) +
AAC_MUL31(-saved[n2 - 1 - i] , window[i + 2*n - n4]) +
AAC_MUL31(-saved[n + n2 + i] , window[i + 3*n - n4]);
}
// buffer update
memmove(saved + n, saved, 2 * n * sizeof(*saved));
memcpy( saved, buf, n * sizeof(*saved));
}
/**
* channel coupling transformation interface
*
* @param apply_coupling_method pointer to (in)dependent coupling function
*/
static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
enum RawDataBlockType type, int elem_id,
enum CouplingPoint coupling_point,
void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
{
int i, c;
for (i = 0; i < MAX_ELEM_ID; i++) {
ChannelElement *cce = ac->che[TYPE_CCE][i];
int index = 0;
if (cce && cce->coup.coupling_point == coupling_point) {
ChannelCoupling *coup = &cce->coup;
for (c = 0; c <= coup->num_coupled; c++) {
if (coup->type[c] == type && coup->id_select[c] == elem_id) {
if (coup->ch_select[c] != 1) {
apply_coupling_method(ac, &cc->ch[0], cce, index);
if (coup->ch_select[c] != 0)
index++;
}
if (coup->ch_select[c] != 2)
apply_coupling_method(ac, &cc->ch[1], cce, index++);
} else
index += 1 + (coup->ch_select[c] == 3);
}
}
}
}
/**
* Convert spectral data to samples, applying all supported tools as appropriate.
*/
static void spectral_to_sample(AACContext *ac, int samples)
{
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 = 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 && che->present) {
if (type <= TYPE_CPE)
apply_channel_coupling(ac, che, type, i, BEFORE_TNS, AAC_RENAME(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)
ac->apply_ltp(ac, &che->ch[0]);
if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
ac->apply_ltp(ac, &che->ch[1]);
}
}
if (che->ch[0].tns.present)
ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
if (che->ch[1].tns.present)
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, AAC_RENAME(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)
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)
ac->update_ltp(ac, &che->ch[1]);
}
if (ac->oc[1].m4ac.sbr > 0) {
AAC_RENAME(ff_sbr_apply)(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
}
}
if (type <= TYPE_CCE)
apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, AAC_RENAME(apply_independent_coupling));
#if USE_FIXED
{
int j;
/* preparation for resampler */
for(j = 0; j<samples; j++){
che->ch[0].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[0].ret[j]<<7)+0x8000;
if(type == TYPE_CPE)
che->ch[1].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[1].ret[j]<<7)+0x8000;
}
}
#endif /* USE_FIXED */
che->present = 0;
} else if (che) {
av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
}
}
}
}
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
{
int size;
AACADTSHeaderInfo hdr_info;
uint8_t layout_map[MAX_ELEM_ID*4][3];
int layout_map_tags, ret;
size = avpriv_aac_parse_header(gb, &hdr_info);
if (size > 0) {
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");
ac->warned_num_aac_frames = 1;
}
push_output_configuration(ac);
if (hdr_info.chan_config) {
ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
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;
/**
* 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;
ac->oc[1].m4ac.object_type = hdr_info.object_type;
ac->oc[1].m4ac.frame_length_short = 0;
if (ac->oc[0].status != OC_LOCKED ||
ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
ac->oc[1].m4ac.sbr = -1;
ac->oc[1].m4ac.ps = -1;
}
if (!hdr_info.crc_absent)
skip_bits(gb, 16);
}
return size;
}
static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, GetBitContext *gb)
{
AACContext *ac = avctx->priv_data;
const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
ChannelElement *che;
int err, i;
int samples = m4ac->frame_length_short ? 960 : 1024;
int chan_config = m4ac->chan_config;
int aot = 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 = aot - 1;
ac->tags_mapped = 0;
if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
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;
}
che->present = 1;
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, samples);
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, 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;
if (show_bits(gb, 12) == 0xfff) {
if ((err = parse_adts_frame_header(ac, gb)) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
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 = AVERROR_INVALIDDATA;
goto fail;
}
}
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) {
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 (!avctx->channels && elem_type != TYPE_PCE) {
err = AVERROR_INVALIDDATA;
goto fail;
}
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",
elem_type, elem_id);
err = AVERROR_INVALIDDATA;
goto fail;
}
samples = 1024;
che->present = 1;
}
switch (elem_type) {
case TYPE_SCE:
err = decode_ics(ac, &che->ch[0], gb, 0, 0);
audio_found = 1;
sce_count++;
break;
case TYPE_CPE:
err = decode_cpe(ac, gb, che);
audio_found = 1;
break;
case TYPE_CCE:
err = decode_cce(ac, gb, che);
break;
case TYPE_LFE:
err = decode_ics(ac, &che->ch[0], gb, 0, 0);
audio_found = 1;
break;
case TYPE_DSE:
err = skip_data_stream_element(ac, gb);
break;
case TYPE_PCE: {
uint8_t layout_map[MAX_ELEM_ID*4][3];
int tags;
push_output_configuration(ac);
tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb);
if (tags < 0) {
err = tags;
break;
}
if (pce_found) {
av_log(avctx, AV_LOG_ERROR,
"Not evaluating a further program_config_element as this construct is dubious at best.\n");
} 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;
}
case TYPE_FIL:
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, "TYPE_FIL: "overread_err);
err = AVERROR_INVALIDDATA;
goto fail;
}
while (elem_id > 0)
elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
err = 0; /* FIXME */
break;
default:
err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
break;
}
che_prev = che;
elem_type_prev = elem_type;
if (err)
goto fail;
if (get_bits_left(gb) < 3) {
av_log(avctx, AV_LOG_ERROR, overread_err);
err = AVERROR_INVALIDDATA;
goto fail;
}
}
if (!avctx->channels) {
*got_frame_ptr = 0;
return 0;
}
multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
samples <<= multiplier;
spectral_to_sample(ac, 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 (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));
}
if (!ac->frame->data[0] && samples) {
av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
err = AVERROR_INVALIDDATA;
goto fail;
}
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);
return err;
}
static int aac_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AACContext *ac = avctx->priv_data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
GetBitContext gb;
int buf_consumed;
int buf_offset;
int err;
int new_extradata_size;
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 && 0) {
av_free(avctx->extradata);
avctx->extradata = av_mallocz(new_extradata_size +
AV_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata)
return AVERROR(ENOMEM);
avctx->extradata_size = new_extradata_size;
memcpy(avctx->extradata, new_extradata, new_extradata_size);
push_output_configuration(ac);
if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
avctx->extradata,
avctx->extradata_size*8LL, 1) < 0) {
pop_output_configuration(ac);
return AVERROR_INVALIDDATA;
}
}
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_bits8(&gb, buf, buf_size)) < 0)
return err;
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, avpkt);
}
if (err < 0)
return err;
buf_consumed = (get_bits_count(&gb) + 7) >> 3;
for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
if (buf[buf_offset])
break;
return buf_size > buf_offset ? buf_consumed : buf_size;
}
static av_cold int aac_decode_close(AVCodecContext *avctx)
{
AACContext *ac = avctx->priv_data;
int i, type;
for (i = 0; i < MAX_ELEM_ID; i++) {
for (type = 0; type < 4; type++) {
if (ac->che[type][i])
AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][i]->sbr);
av_freep(&ac->che[type][i]);
}
}
ff_mdct_end(&ac->mdct);
ff_mdct_end(&ac->mdct_small);
ff_mdct_end(&ac->mdct_ld);
ff_mdct_end(&ac->mdct_ltp);
#if !USE_FIXED
ff_imdct15_uninit(&ac->mdct480);
#endif
av_freep(&ac->fdsp);
return 0;
}
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 USE_FIXED
c->vector_pow43 = vector_pow43;
c->subband_scale = subband_scale;
#endif
#if !USE_FIXED
if(ARCH_MIPS)
ff_aacdec_init_mips(c);
#endif /* !USE_FIXED */
}
/**
* 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 },
};