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aacdec_template.c
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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 <[email protected]>
*
* AAC LATM decoder
* Copyright (c) 2008-2010 Paul Kendall <[email protected]>
* Copyright (c) 2010 Janne Grunau <[email protected]>
*
* 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, int bit_size,
int sync_extension)
{
GetBitContext gb;
int i, ret;
ff_dlog(avctx, "audio specific config size %d\n", 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",