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78bf446852
FFmpeg/libavcodec/aacdec_template.c
wm4 fcfc78cbab aacdec: do not mutate input packet metadata 
Apparently the demuxer outputs the wrong padding for HE-AAC (based on
the raw sample rate, or so). aacdec contains a hack to adjust the muxer
padding accordingly before it's used to trim the decoder output. This
modified the packet side data, which in combination with the old
decoding API would change the packet the user passed to the decoder.
This is clearly not allowed, and it breaks running some gapless fate
tests with "-fflags +keepside" applied (without keepside, the packet
metadata is typically newly allocated, essentially making a copy and not
modifying the user's input packet).

This should probably be fixed in the demuxer (and consequently also the
muxer), but for now only fix the immediate problem.

Regression since 946ed78f5f (2012).
2017-03-09 10:16:12 +01:00

3263 lines
116 KiB
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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.
*/
#include "libavutil/thread.h"
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, type);
}
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]++;
if (id_map[type][id] >= MAX_ELEM_ID) {
avpriv_request_sample(ac->avctx, "Too large remapped id");
return AVERROR_PATCHWELCOME;
}
}
// 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++;
}
}
static inline void relative_align_get_bits(GetBitContext *gb,
int reference_position) {
int n = (reference_position - get_bits_count(gb) & 7);
if (n)
skip_bits(gb, n);
}
/**
* 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 byte_align_ref)
{
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;
relative_align_get_bits(gb, byte_align_ref);
/* 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,
int get_bit_alignment,
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, get_bit_alignment);
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 gb buffer holding an audio specific config
* @param get_bit_alignment relative alignment for byte align operations
* @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_gb(AACContext *ac,
AVCodecContext *avctx,
MPEG4AudioConfig *m4ac,
GetBitContext *gb,
int get_bit_alignment,
int sync_extension)
{
int i, ret;
GetBitContext gbc = *gb;
if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, 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, get_bit_alignment,
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);
}
static int decode_audio_specific_config(AACContext *ac,
AVCodecContext *avctx,
MPEG4AudioConfig *m4ac,
const uint8_t *data, int64_t bit_size,
int sync_extension)
{
int i, ret;
GetBitContext gb;
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;
return decode_audio_specific_config_gb(ac, avctx, m4ac, &gb, 0,
sync_extension);
}
/**
* 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 void aac_static_table_init(void)
{
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)();
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);
// 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(ff_cbrt_tableinit)();
}
static AVOnce aac_table_init = AV_ONCE_INIT;
static av_cold int aac_decode_init(AVCodecContext *avctx)
{
AACContext *ac = avctx->priv_data;
int ret;
ret = ff_thread_once(&aac_table_init, &aac_static_table_init);
if (ret != 0)
return AVERROR_UNKNOWN;
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;
}
#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;
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_mdct15_init(&ac->mdct480, 1, 5, -1.0f);
if (ret < 0)
return ret;
#endif
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++ = ff_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);
if (!ac->frame->data[0] && samples) {
av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
return AVERROR_INVALIDDATA;
}
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, che_prev_type = TYPE_END;
int err, elem_id;
int samples = 0, multiplier, audio_found = 0, pce_found = 0;
int is_dmono, sce_count = 0;
int payload_alignment;
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;
payload_alignment = get_bits_count(gb);
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,
payload_alignment);
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");
pop_output_configuration(ac);
} else {
err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
if (!err)
ac->oc[1].m4ac.chan_config = 0;
pce_found = 1;
}
break;
}
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, che_prev_type);
err = 0; /* FIXME */
break;
default:
err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
break;
}
if (elem_type < TYPE_DSE) {
che_prev = che;
che_prev_type = 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)
avctx->internal->skip_samples_multiplier = 2;
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_mdct15_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,
};
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