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authorLynne <dev@lynne.ee>2024-03-21 08:20:43 +0100
committerLynne <dev@lynne.ee>2024-04-23 08:31:40 +0200
commitae7c6cc17d57e4ff73f88dc4a4284c1676a7e19a (patch)
tree4b3588fa4ccbdde30263aff70d9a4f2a344a6498 /libavcodec/aac
parent551ce16b59b109093516e2f4000ae809fcd0b9f3 (diff)
downloadffmpeg-ae7c6cc17d57e4ff73f88dc4a4284c1676a7e19a.tar.gz
aac: move aacdec.c to aac/aacdec.c
Diffstat (limited to 'libavcodec/aac')
-rw-r--r--libavcodec/aac/aacdec.c2445
-rw-r--r--libavcodec/aac/aacdec.h7
-rw-r--r--libavcodec/aac/aacdec_latm.h2
3 files changed, 2442 insertions, 12 deletions
diff --git a/libavcodec/aac/aacdec.c b/libavcodec/aac/aacdec.c
index dfbc309583..01a10468fa 100644
--- a/libavcodec/aac/aacdec.c
+++ b/libavcodec/aac/aacdec.c
@@ -32,10 +32,20 @@
#include <limits.h>
#include <stddef.h>
+#include "aacdec.h"
+#include "aacdec_tab.h"
+
#include "libavcodec/aac.h"
#include "libavcodec/aacsbr.h"
-#include "aacdec.h"
+#include "libavcodec/aactab.h"
+#include "libavcodec/adts_header.h"
+
#include "libavcodec/avcodec.h"
+#include "libavcodec/internal.h"
+#include "libavcodec/codec_internal.h"
+#include "libavcodec/decode.h"
+#include "libavcodec/profiles.h"
+
#include "libavutil/attributes.h"
#include "libavutil/error.h"
#include "libavutil/log.h"
@@ -44,14 +54,1067 @@
#include "libavutil/opt.h"
#include "libavutil/tx.h"
#include "libavutil/version.h"
+#include "libavutil/thread.h"
+
+/*
+ * 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 int output_configure(AACDecContext *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(AACDecContext *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]) {
+ int ret;
+ if (ac->is_fixed)
+ ret = ff_aac_sbr_ctx_alloc_init_fixed(ac, &ac->che[type][id], type);
+ else
+ ret = ff_aac_sbr_ctx_alloc_init(ac, &ac->che[type][id], type);
+ if (ret < 0)
+ return ret;
+ }
+ 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]) {
+ if (ac->is_fixed)
+ ff_aac_sbr_ctx_close_fixed(ac->che[type][id]);
+ else
+ ff_aac_sbr_ctx_close(ac->che[type][id]);
+ }
+ av_freep(&ac->che[type][id]);
+ }
+ return 0;
+}
+
+static int frame_configure_elements(AVCodecContext *avctx)
+{
+ AACDecContext *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].AAC_RENAME(output) = che->ch[0].AAC_RENAME(ret_buf);
+ che->ch[1].AAC_RENAME(output) = che->ch[1].AAC_RENAME(ret_buf);
+ }
+ }
+ }
+
+ /* get output buffer */
+ av_frame_unref(ac->frame);
+ if (!avctx->ch_layout.nb_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->ch_layout.nb_channels; ch++) {
+ if (ac->output_element[ch])
+ ac->output_element[ch]->AAC_RENAME(output) = (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, uint64_t *layout)
+{
+ 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
+ };
+ if (e2c_vec[offset].av_position != UINT64_MAX)
+ *layout |= e2c_vec[offset].av_position;
+
+ 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
+ };
+ if (left != UINT64_MAX)
+ *layout |= left;
+
+ if (right != UINT64_MAX)
+ *layout |= right;
+
+ 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 ^= (pos != AAC_CHANNEL_LFE);
+ }
+ }
+ if (sce_parity &&
+ (pos == AAC_CHANNEL_FRONT && first_cpe))
+ return -1;
+
+ return num_pos_channels;
+}
+
+static int assign_channels(struct elem_to_channel e2c_vec[MAX_ELEM_ID], uint8_t (*layout_map)[3],
+ uint64_t *layout, int tags, int layer, int pos, int *current)
+{
+ int i = *current, j = 0;
+ int nb_channels = count_paired_channels(layout_map, tags, pos, i);
+
+ if (nb_channels < 0 || nb_channels > 5)
+ return 0;
+
+ if (pos == AAC_CHANNEL_LFE) {
+ while (nb_channels) {
+ if (ff_aac_channel_map[layer][pos - 1][j] == AV_CHAN_NONE)
+ return -1;
+ e2c_vec[i] = (struct elem_to_channel) {
+ .av_position = 1ULL << ff_aac_channel_map[layer][pos - 1][j],
+ .syn_ele = layout_map[i][0],
+ .elem_id = layout_map[i][1],
+ .aac_position = pos
+ };
+ *layout |= e2c_vec[i].av_position;
+ i++;
+ j++;
+ nb_channels--;
+ }
+ *current = i;
+
+ return 0;
+ }
+
+ while (nb_channels & 1) {
+ if (ff_aac_channel_map[layer][pos - 1][0] == AV_CHAN_NONE)
+ return -1;
+ if (ff_aac_channel_map[layer][pos - 1][0] == AV_CHAN_UNUSED)
+ break;
+ e2c_vec[i] = (struct elem_to_channel) {
+ .av_position = 1ULL << ff_aac_channel_map[layer][pos - 1][0],
+ .syn_ele = layout_map[i][0],
+ .elem_id = layout_map[i][1],
+ .aac_position = pos
+ };
+ *layout |= e2c_vec[i].av_position;
+ i++;
+ nb_channels--;
+ }
+
+ j = (pos != AAC_CHANNEL_SIDE) && nb_channels <= 3 ? 3 : 1;
+ while (nb_channels >= 2) {
+ if (ff_aac_channel_map[layer][pos - 1][j] == AV_CHAN_NONE ||
+ ff_aac_channel_map[layer][pos - 1][j+1] == AV_CHAN_NONE)
+ return -1;
+ i += assign_pair(e2c_vec, layout_map, i,
+ 1ULL << ff_aac_channel_map[layer][pos - 1][j],
+ 1ULL << ff_aac_channel_map[layer][pos - 1][j+1],
+ pos, layout);
+ j += 2;
+ nb_channels -= 2;
+ }
+ while (nb_channels & 1) {
+ if (ff_aac_channel_map[layer][pos - 1][5] == AV_CHAN_NONE)
+ return -1;
+ e2c_vec[i] = (struct elem_to_channel) {
+ .av_position = 1ULL << ff_aac_channel_map[layer][pos - 1][5],
+ .syn_ele = layout_map[i][0],
+ .elem_id = layout_map[i][1],
+ .aac_position = pos
+ };
+ *layout |= e2c_vec[i].av_position;
+ i++;
+ nb_channels--;
+ }
+ if (nb_channels)
+ return -1;
+
+ *current = i;
+
+ return 0;
+}
+
+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 } };
+ uint64_t layout = 0;
+
+ if (FF_ARRAY_ELEMS(e2c_vec) < tags)
+ return 0;
+
+ for (n = 0, i = 0; n < 3 && i < tags; n++) {
+ int ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_FRONT, &i);
+ if (ret < 0)
+ return 0;
+ ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_SIDE, &i);
+ if (ret < 0)
+ return 0;
+ ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_BACK, &i);
+ if (ret < 0)
+ return 0;
+ ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_LFE, &i);
+ if (ret < 0)
+ return 0;
+ }
+
+ total_non_cc_elements = n = i;
+
+ if (layout == AV_CH_LAYOUT_22POINT2) {
+ // For 22.2 reorder the result as needed
+ FFSWAP(struct elem_to_channel, e2c_vec[2], e2c_vec[0]); // FL & FR first (final), FC third
+ FFSWAP(struct elem_to_channel, e2c_vec[2], e2c_vec[1]); // FC second (final), FLc & FRc third
+ FFSWAP(struct elem_to_channel, e2c_vec[6], e2c_vec[2]); // LFE1 third (final), FLc & FRc seventh
+ FFSWAP(struct elem_to_channel, e2c_vec[4], e2c_vec[3]); // BL & BR fourth (final), SiL & SiR fifth
+ FFSWAP(struct elem_to_channel, e2c_vec[6], e2c_vec[4]); // FLc & FRc fifth (final), SiL & SiR seventh
+ FFSWAP(struct elem_to_channel, e2c_vec[7], e2c_vec[6]); // LFE2 seventh (final), SiL & SiR eight (final)
+ FFSWAP(struct elem_to_channel, e2c_vec[9], e2c_vec[8]); // TpFL & TpFR ninth (final), TFC tenth (final)
+ FFSWAP(struct elem_to_channel, e2c_vec[11], e2c_vec[10]); // TC eleventh (final), TpSiL & TpSiR twelth
+ FFSWAP(struct elem_to_channel, e2c_vec[12], e2c_vec[11]); // TpBL & TpBR twelth (final), TpSiL & TpSiR thirteenth (final)
+ } else {
+ // For everything else, utilize the AV channel position define as a
+ // stable sort.
+ 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);
+
+ }
+
+ 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;
+ }
+
+ return layout;
+}
+
+/**
+ * Save current output configuration if and only if it has been locked.
+ */
+static int push_output_configuration(AACDecContext *ac)
+{
+ int pushed = 0;
+
+ if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) {
+ ac->oc[0] = ac->oc[1];
+ pushed = 1;
+ }
+ ac->oc[1].status = OC_NONE;
+ return pushed;
+}
+
+/**
+ * Restore the previous output configuration if and only if the current
+ * configuration is unlocked.
+ */
+static void pop_output_configuration(AACDecContext *ac)
+{
+ if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
+ ac->oc[1] = ac->oc[0];
+ ac->avctx->ch_layout = ac->oc[1].ch_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(AACDecContext *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 (ac->output_channel_order == CHANNEL_ORDER_DEFAULT)
+ 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;
+ }
+ }
+
+ av_channel_layout_uninit(&ac->oc[1].ch_layout);
+ if (layout)
+ av_channel_layout_from_mask(&ac->oc[1].ch_layout, layout);
+ else {
+ ac->oc[1].ch_layout.order = AV_CHANNEL_ORDER_UNSPEC;
+ ac->oc[1].ch_layout.nb_channels = channels;
+ }
+
+ av_channel_layout_copy(&avctx->ch_layout, &ac->oc[1].ch_layout);
+ 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)
+{
+ AACDecContext *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(AACDecContext *ac, AVCodecContext *avctx,
+ uint8_t (*layout_map)[3],
+ int *tags,
+ int channel_config)
+{
+ if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
+ channel_config > 14) {
+ av_log(avctx, AV_LOG_ERROR,
+ "invalid default channel configuration (%d)\n",
+ channel_config);
+ return AVERROR_INVALIDDATA;
+ }
+ *tags = ff_tags_per_config[channel_config];
+ memcpy(layout_map, ff_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) {
+ layout_map[2][2] = AAC_CHANNEL_BACK;
+
+ if (!ac || !ac->warned_71_wide++) {
+ 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);
+ }
+ }
+
+ return 0;
+}
+
+static ChannelElement *get_che(AACDecContext *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, 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");
+
+ 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, 1) < 0)
+ return NULL;
+
+ 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 14:
+ if (ac->tags_mapped > 2 && ((type == TYPE_CPE && elem_id < 3) ||
+ (type == TYPE_LFE && elem_id < 1))) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[type][elem_id] = ac->che[type][elem_id];
+ }
+ case 13:
+ if (ac->tags_mapped > 3 && ((type == TYPE_CPE && elem_id < 8) ||
+ (type == TYPE_SCE && elem_id < 6) ||
+ (type == TYPE_LFE && elem_id < 2))) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[type][elem_id] = ac->che[type][elem_id];
+ }
+ 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 == 3 && 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 == ff_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 == ff_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->tags_mapped == 1 && ac->oc[1].m4ac.chan_config == 2 &&
+ type == TYPE_SCE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
+ }
+ 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) < 5 * (num_front + num_side + num_back + num_cc) + 4 *(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 AACDecContext, 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(AACDecContext *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;
+
+ m4ac->frame_length_short = get_bits1(gb);
+ if (m4ac->frame_length_short && m4ac->sbr == 1) {
+ avpriv_report_missing_feature(avctx, "SBR with 960 frame length");
+ if (ac) ac->warned_960_sbr = 1;
+ m4ac->sbr = 0;
+ m4ac->ps = 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(ac, 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(AACDecContext *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;
+ m4ac->frame_length_short = get_bits1(gb);
+
+ 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(ac, 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 AACDecContext, 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(AACDecContext *ac,
+ AVCodecContext *avctx,
+ MPEG4AudioConfig *m4ac,
+ GetBitContext *gb,
+ int get_bit_alignment,
+ int sync_extension)
+{
+ int i, ret;
+ GetBitContext gbc = *gb;
+ MPEG4AudioConfig m4ac_bak = *m4ac;
+
+ if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, sync_extension, avctx)) < 0) {
+ *m4ac = m4ac_bak;
+ return AVERROR_INVALIDDATA;
+ }
+
+ if (m4ac->sampling_index > 12) {
+ av_log(avctx, AV_LOG_ERROR,
+ "invalid sampling rate index %d\n",
+ m4ac->sampling_index);
+ *m4ac = m4ac_bak;
+ 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);
+ *m4ac = m4ac_bak;
+ return AVERROR_INVALIDDATA;
+ }
+
+ skip_bits_long(gb, i);
-extern const AACDecDSP aac_dsp;
-extern const AACDecDSP aac_dsp_fixed;
+ switch (m4ac->object_type) {
+ case AOT_AAC_MAIN:
+ case AOT_AAC_LC:
+ case AOT_AAC_SSR:
+ 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(AACDecContext *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);
+}
+
+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 av_cold void aac_static_table_init(void)
+{
+ ff_aac_sbr_init();
+ ff_aac_sbr_init_fixed();
-extern const AACDecProc aac_proc;
-extern const AACDecProc aac_proc_fixed;
+ ff_aacdec_common_init_once();
+}
+static AVOnce aac_table_init = AV_ONCE_INIT;
-av_cold int ff_aac_decode_close(AVCodecContext *avctx)
+static av_cold int decode_close(AVCodecContext *avctx)
{
AACDecContext *ac = avctx->priv_data;
int is_fixed = ac->is_fixed;
@@ -84,7 +1147,7 @@ av_cold int ff_aac_decode_close(AVCodecContext *avctx)
return 0;
}
-av_cold int ff_aac_decode_init_common(AVCodecContext *avctx)
+static av_cold int init_dsp(AVCodecContext *avctx)
{
AACDecContext *ac = avctx->priv_data;
int is_fixed = ac->is_fixed, ret;
@@ -127,6 +1190,1332 @@ av_cold int ff_aac_decode_init_common(AVCodecContext *avctx)
return ac->dsp.init(ac);
}
+static av_cold int aac_decode_init_internal(AVCodecContext *avctx)
+{
+ AACDecContext *ac = avctx->priv_data;
+ int ret;
+
+ if (avctx->sample_rate > 96000)
+ return AVERROR_INVALIDDATA;
+
+ 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;
+
+ if (ac->is_fixed)
+ avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
+ else
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+
+ 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->ch_layout.nb_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->ch_layout.nb_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(ac, 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;
+ }
+ }
+
+ return init_dsp(avctx);
+}
+
+static av_cold int aac_decode_init(AVCodecContext *avctx)
+{
+ AACDecContext *ac = avctx->priv_data;
+ ac->is_fixed = 0;
+ return aac_decode_init_internal(avctx);
+}
+
+static av_cold int aac_decode_init_fixed(AVCodecContext *avctx)
+{
+ AACDecContext *ac = avctx->priv_data;
+ ac->is_fixed = 1;
+ return aac_decode_init_internal(avctx);
+}
+
+/**
+ * Skip data_stream_element; reference: table 4.10.
+ */
+static int skip_data_stream_element(AACDecContext *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(AACDecContext *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(AACDecContext *ac, LongTermPrediction *ltp,
+ GetBitContext *gb, uint8_t max_sfb)
+{
+ int sfb;
+
+ ltp->lag = get_bits(gb, 11);
+ if (ac->is_fixed)
+ ltp->coef_fixed = Q30(ff_ltp_coef[get_bits(gb, 3)]);
+ else
+ ltp->coef = ff_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(AACDecContext *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;
+ int ret_fail = AVERROR_INVALIDDATA;
+
+ 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;
+ if (m4ac->frame_length_short) {
+ ics->swb_offset = ff_swb_offset_120[sampling_index];
+ ics->num_swb = ff_aac_num_swb_120[sampling_index];
+ } else {
+ 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) {
+ ret_fail = AVERROR_BUG;
+ goto fail;
+ }
+ } else {
+ if (m4ac->frame_length_short) {
+ ics->num_swb = ff_aac_num_swb_960[sampling_index];
+ ics->swb_offset = ff_swb_offset_960[sampling_index];
+ } else {
+ ics->num_swb = ff_aac_num_swb_1024[sampling_index];
+ ics->swb_offset = ff_swb_offset_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");
+ ret_fail = AVERROR_PATCHWELCOME;
+ goto fail;
+ }
+ if ((ics->ltp.present = get_bits(gb, 1)))
+ decode_ltp(ac, &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 ret_fail;
+}
+
+/**
+ * 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(AACDecContext *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(AACDecContext *ac, int sfo[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];
+ switch (band_type[idx]) {
+ case ZERO_BT:
+ for (; i < run_end; i++, idx++)
+ sfo[idx] = 0;
+ break;
+ case INTENSITY_BT: /* fallthrough */
+ case INTENSITY_BT2:
+ for (; i < run_end; i++, idx++) {
+ offset[2] += get_vlc2(gb, ff_vlc_scalefactors, 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);
+ }
+ sfo[idx] = clipped_offset;
+ }
+ break;
+ case 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, ff_vlc_scalefactors, 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);
+ }
+ sfo[idx] = clipped_offset;
+ }
+ break;
+ default:
+ for (; i < run_end; i++, idx++) {
+ offset[0] += get_vlc2(gb, ff_vlc_scalefactors, 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;
+ }
+ sfo[idx] = offset[0];
+ }
+ break;
+ }
+ }
+ }
+ 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(AACDecContext *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++) {
+ if (ac->is_fixed)
+ tns->coef_fixed[w][filt][i] = Q31(ff_tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]);
+ else
+ tns->coef[w][filt][i] = ff_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]));
+ }
+}
+
+static void decode_gain_control(SingleChannelElement * sce, GetBitContext * gb)
+{
+ // wd_num, wd_test, aloc_size
+ static const uint8_t gain_mode[4][3] = {
+ {1, 0, 5}, // ONLY_LONG_SEQUENCE = 0,
+ {2, 1, 2}, // LONG_START_SEQUENCE,
+ {8, 0, 2}, // EIGHT_SHORT_SEQUENCE,
+ {2, 1, 5}, // LONG_STOP_SEQUENCE
+ };
+
+ const int mode = sce->ics.window_sequence[0];
+ uint8_t bd, wd, ad;
+
+ // FIXME: Store the gain control data on |sce| and do something with it.
+ uint8_t max_band = get_bits(gb, 2);
+ for (bd = 0; bd < max_band; bd++) {
+ for (wd = 0; wd < gain_mode[mode][0]; wd++) {
+ uint8_t adjust_num = get_bits(gb, 3);
+ for (ad = 0; ad < adjust_num; ad++) {
+ skip_bits(gb, 4 + ((wd == 0 && gain_mode[mode][1])
+ ? 4
+ : gain_mode[mode][2]));
+ }
+ }
+ }
+}
+
+/**
+ * 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
+ */
+int ff_aac_decode_ics(AACDecContext *ac, SingleChannelElement *sce,
+ GetBitContext *gb, int common_window, int scale_flag)
+{
+ Pulse pulse;
+ TemporalNoiseShaping *tns = &sce->tns;
+ IndividualChannelStream *ics = &sce->ics;
+ 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) {
+ ret = decode_ics_info(ac, ics, gb);
+ if (ret < 0)
+ goto fail;
+ }
+
+ if ((ret = decode_band_types(ac, sce->band_type,
+ sce->band_type_run_end, gb, ics)) < 0)
+ goto fail;
+ if ((ret = decode_scalefactors(ac, sce->sfo, gb, global_gain, ics,
+ sce->band_type, sce->band_type_run_end)) < 0)
+ goto fail;
+
+ ac->dsp.dequant_scalefactors(sce);
+
+ 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");
+ ret = AVERROR_INVALIDDATA;
+ goto fail;
+ }
+ if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "Pulse data corrupt or invalid.\n");
+ ret = AVERROR_INVALIDDATA;
+ goto fail;
+ }
+ }
+ tns->present = get_bits1(gb);
+ if (tns->present && !er_syntax) {
+ ret = decode_tns(ac, tns, gb, ics);
+ if (ret < 0)
+ goto fail;
+ }
+ if (!eld_syntax && get_bits1(gb)) {
+ decode_gain_control(sce, gb);
+ if (!ac->warned_gain_control) {
+ avpriv_report_missing_feature(ac->avctx, "Gain control");
+ ac->warned_gain_control = 1;
+ }
+ }
+ // 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) {
+ ret = decode_tns(ac, tns, gb, ics);
+ if (ret < 0)
+ goto fail;
+ }
+ }
+
+ ret = ac->proc.decode_spectrum_and_dequant(ac, gb,
+ pulse_present ? &pulse : NULL,
+ sce);
+ if (ret < 0)
+ goto fail;
+
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
+ ac->dsp.apply_prediction(ac, sce);
+
+ return 0;
+fail:
+ tns->present = 0;
+ return ret;
+}
+
+/**
+ * Decode a channel_pair_element; reference: table 4.4.
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_cpe(AACDecContext *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(ac, &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 = ff_aac_decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
+ return ret;
+ if ((ret = ff_aac_decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
+ return ret;
+
+ if (common_window) {
+ if (ms_present)
+ ac->dsp.apply_mid_side_stereo(ac, cpe);
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
+ ac->dsp.apply_prediction(ac, &cpe->ch[0]);
+ ac->dsp.apply_prediction(ac, &cpe->ch[1]);
+ }
+ }
+
+ ac->dsp.apply_intensity_stereo(ac, cpe, ms_present);
+ 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(AACDecContext *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(AACDecContext *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.frame_length_short) {
+ if (!ac->warned_960_sbr)
+ avpriv_report_missing_feature(ac->avctx,
+ "SBR with 960 frame length");
+ ac->warned_960_sbr = 1;
+ skip_bits_long(gb, 8 * cnt - 4);
+ 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->ch_layout.nb_channels == 1) {
+ ac->oc[1].m4ac.sbr = 1;
+ ac->oc[1].m4ac.ps = 1;
+ ac->avctx->profile = AV_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 = AV_PROFILE_AAC_HE;
+ }
+
+ if (ac->is_fixed)
+ res = ff_aac_sbr_decode_extension_fixed(ac, che, gb, crc_flag, cnt, elem_type);
+ else
+ res = ff_aac_sbr_decode_extension(ac, che, gb, crc_flag, cnt, elem_type);
+
+
+ if (ac->oc[1].m4ac.ps == 1 && !ac->warned_he_aac_mono) {
+ av_log(ac->avctx, AV_LOG_VERBOSE, "Treating HE-AAC mono as stereo.\n");
+ ac->warned_he_aac_mono = 1;
+ }
+ 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;
+}
+
+/**
+ * channel coupling transformation interface
+ *
+ * @param apply_coupling_method pointer to (in)dependent coupling function
+ */
+static void apply_channel_coupling(AACDecContext *ac, ChannelElement *cc,
+ enum RawDataBlockType type, int elem_id,
+ enum CouplingPoint coupling_point,
+ void (*apply_coupling_method)(AACDecContext *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(AACDecContext *ac, int samples)
+{
+ int i, type;
+ void (*imdct_and_window)(AACDecContext *ac, SingleChannelElement *sce);
+ switch (ac->oc[1].m4ac.object_type) {
+ case AOT_ER_AAC_LD:
+ imdct_and_window = ac->dsp.imdct_and_windowing_ld;
+ break;
+ case AOT_ER_AAC_ELD:
+ imdct_and_window = ac->dsp.imdct_and_windowing_eld;
+ break;
+ default:
+ if (ac->oc[1].m4ac.frame_length_short)
+ imdct_and_window = ac->dsp.imdct_and_windowing_960;
+ else
+ imdct_and_window = ac->dsp.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, ac->dsp.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->dsp.apply_ltp(ac, &che->ch[0]);
+ if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
+ ac->dsp.apply_ltp(ac, &che->ch[1]);
+ }
+ }
+ if (che->ch[0].tns.present)
+ ac->dsp.apply_tns(che->ch[0].AAC_RENAME(coeffs),
+ &che->ch[0].tns, &che->ch[0].ics, 1);
+ if (che->ch[1].tns.present)
+ ac->dsp.apply_tns(che->ch[1].AAC_RENAME(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, ac->dsp.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->dsp.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->dsp.update_ltp(ac, &che->ch[1]);
+ }
+ if (ac->oc[1].m4ac.sbr > 0) {
+ if (ac->is_fixed)
+ ff_aac_sbr_apply_fixed(ac, che, type,
+ che->ch[0].AAC_RENAME(output),
+ che->ch[1].AAC_RENAME(output));
+ else
+ ff_aac_sbr_apply(ac, che, type,
+ che->ch[0].AAC_RENAME(output),
+ che->ch[1].AAC_RENAME(output));
+ }
+ }
+ if (type <= TYPE_CCE)
+ apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, ac->dsp.apply_independent_coupling);
+ ac->dsp.clip_output(ac, che, type, samples);
+ 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(AACDecContext *ac, GetBitContext *gb)
+{
+ int size;
+ AACADTSHeaderInfo hdr_info;
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int layout_map_tags, ret;
+
+ size = ff_adts_header_parse(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, 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, AVFrame *frame,
+ int *got_frame_ptr, GetBitContext *gb)
+{
+ AACDecContext *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 = frame;
+
+ if ((err = frame_configure_elements(avctx)) < 0)
+ return err;
+
+ // The AV_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 < ff_tags_per_config[chan_config]; i++) {
+ const int elem_type = ff_aac_channel_layout_map[chan_config-1][i][0];
+ const int elem_id = ff_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 = ff_aac_decode_ics(ac, &che->ch[0], gb, 0, 0);
+ break;
+ case TYPE_CPE:
+ err = decode_cpe(ac, gb, che);
+ break;
+ case TYPE_LFE:
+ err = ff_aac_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, AVFrame *frame,
+ int *got_frame_ptr, GetBitContext *gb,
+ const AVPacket *avpkt)
+{
+ AACDecContext *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;
+ uint8_t che_presence[4][MAX_ELEM_ID] = {{0}};
+
+ ac->frame = frame;
+
+ 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 AV_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->ch_layout.nb_channels && elem_type != TYPE_PCE) {
+ err = AVERROR_INVALIDDATA;
+ goto fail;
+ }
+
+ if (elem_type < TYPE_DSE) {
+ if (che_presence[elem_type][elem_id]) {
+ int error = che_presence[elem_type][elem_id] > 1;
+ av_log(ac->avctx, error ? AV_LOG_ERROR : AV_LOG_DEBUG, "channel element %d.%d duplicate\n",
+ elem_type, elem_id);
+ if (error) {
+ err = AVERROR_INVALIDDATA;
+ goto fail;
+ }
+ }
+ che_presence[elem_type][elem_id]++;
+
+ 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 = ac->oc[1].m4ac.frame_length_short ? 960 : 1024;
+ che->present = 1;
+ }
+
+ switch (elem_type) {
+
+ case TYPE_SCE:
+ err = ff_aac_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 = ac->proc.decode_cce(ac, gb, che);
+ break;
+
+ case TYPE_LFE:
+ err = ff_aac_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] = {{0}};
+ int tags;
+
+ int pushed = push_output_configuration(ac);
+ if (pce_found && !pushed) {
+ err = AVERROR_INVALIDDATA;
+ goto fail;
+ }
+
+ 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;
+ }
+ err = 0;
+ while (elem_id > 0) {
+ int ret = decode_extension_payload(ac, gb, elem_id, che_prev, che_prev_type);
+ if (ret < 0) {
+ err = ret;
+ break;
+ }
+ elem_id -= ret;
+ }
+ 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->ch_layout.nb_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 (!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 &&
+ !av_channel_layout_compare(&ac->oc[1].ch_layout,
+ &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
+ if (is_dmono) {
+ if (ac->dmono_mode == 1)
+ frame->data[1] = frame->data[0];
+ else if (ac->dmono_mode == 2)
+ frame->data[0] = frame->data[1];
+ }
+
+ return 0;
+fail:
+ pop_output_configuration(ac);
+ return err;
+}
+
+static int aac_decode_frame(AVCodecContext *avctx, AVFrame *frame,
+ int *got_frame_ptr, AVPacket *avpkt)
+{
+ AACDecContext *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;
+ size_t new_extradata_size;
+ const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
+ AV_PKT_DATA_NEW_EXTRADATA,
+ &new_extradata_size);
+ size_t 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) {
+ /* discard previous configuration */
+ ac->oc[1].status = OC_NONE;
+ err = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
+ new_extradata,
+ new_extradata_size * 8LL, 1);
+ if (err < 0) {
+ return err;
+ }
+ }
+
+ 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, frame, got_frame_ptr, &gb);
+ break;
+ default:
+ err = aac_decode_frame_int(avctx, frame, 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;
+}
+
+#include "aacdec_latm.h"
+
#define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
#define OFF(field) offsetof(AACDecContext, field)
static const AVOption options[] = {
@@ -153,9 +2542,49 @@ static const AVOption options[] = {
{NULL},
};
-const AVClass ff_aac_decoder_class = {
+static const AVClass decoder_class = {
.class_name = "AAC decoder",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
+
+const FFCodec ff_aac_decoder = {
+ .p.name = "aac",
+ CODEC_LONG_NAME("AAC (Advanced Audio Coding)"),
+ .p.type = AVMEDIA_TYPE_AUDIO,
+ .p.id = AV_CODEC_ID_AAC,
+ .p.priv_class = &decoder_class,
+ .priv_data_size = sizeof(AACDecContext),
+ .init = aac_decode_init,
+ .close = decode_close,
+ FF_CODEC_DECODE_CB(aac_decode_frame),
+ .p.sample_fmts = (const enum AVSampleFormat[]) {
+ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
+ },
+ .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
+ .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
+ .p.ch_layouts = ff_aac_ch_layout,
+ .flush = flush,
+ .p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
+};
+
+const FFCodec ff_aac_fixed_decoder = {
+ .p.name = "aac_fixed",
+ CODEC_LONG_NAME("AAC (Advanced Audio Coding)"),
+ .p.type = AVMEDIA_TYPE_AUDIO,
+ .p.id = AV_CODEC_ID_AAC,
+ .p.priv_class = &decoder_class,
+ .priv_data_size = sizeof(AACDecContext),
+ .init = aac_decode_init_fixed,
+ .close = decode_close,
+ FF_CODEC_DECODE_CB(aac_decode_frame),
+ .p.sample_fmts = (const enum AVSampleFormat[]) {
+ AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
+ },
+ .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
+ .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
+ .p.ch_layouts = ff_aac_ch_layout,
+ .p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
+ .flush = flush,
+};
diff --git a/libavcodec/aac/aacdec.h b/libavcodec/aac/aacdec.h
index c8107f6bce..d994fe7981 100644
--- a/libavcodec/aac/aacdec.h
+++ b/libavcodec/aac/aacdec.h
@@ -339,10 +339,11 @@ struct AACDecContext {
#define fdsp RENAME_FIXED(fdsp)
#endif
-extern const struct AVClass ff_aac_decoder_class;
+extern const AACDecDSP aac_dsp;
+extern const AACDecDSP aac_dsp_fixed;
-int ff_aac_decode_init_common(struct AVCodecContext *avctx);
-int ff_aac_decode_close(struct AVCodecContext *avctx);
+extern const AACDecProc aac_proc;
+extern const AACDecProc aac_proc_fixed;
void ff_aacdec_init_mips(AACDecContext *c);
diff --git a/libavcodec/aac/aacdec_latm.h b/libavcodec/aac/aacdec_latm.h
index 0226aebba4..22153dec83 100644
--- a/libavcodec/aac/aacdec_latm.h
+++ b/libavcodec/aac/aacdec_latm.h
@@ -335,7 +335,7 @@ const FFCodec ff_aac_latm_decoder = {
.p.id = AV_CODEC_ID_AAC_LATM,
.priv_data_size = sizeof(struct LATMContext),
.init = latm_decode_init,
- .close = ff_aac_decode_close,
+ .close = decode_close,
FF_CODEC_DECODE_CB(latm_decode_frame),
.p.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
generated by cgit v1.2.3 (git 2.25.1) at 2024-12-03 01:50:09 +0000