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authorTim Redfern <tim@eclectronics.org>2013-09-05 17:57:22 +0100
committerTim Redfern <tim@eclectronics.org>2013-09-05 17:57:22 +0100
commit8992cb1d0d07edc33d274f6d7924ecdf6f83d994 (patch)
tree3a2c86846b7eec8137c1507e623fc7018f13d453 /ffmpeg/libavcodec/aacdec.c
parent741fb4b9e135cfb161a749db88713229038577bb (diff)
making act segmenter
Diffstat (limited to 'ffmpeg/libavcodec/aacdec.c')
-rw-r--r--ffmpeg/libavcodec/aacdec.c3060
1 files changed, 3060 insertions, 0 deletions
diff --git a/ffmpeg/libavcodec/aacdec.c b/ffmpeg/libavcodec/aacdec.c
new file mode 100644
index 0000000..37c7de5
--- /dev/null
+++ b/ffmpeg/libavcodec/aacdec.c
@@ -0,0 +1,3060 @@
+/*
+ * AAC decoder
+ * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
+ * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+ *
+ * AAC LATM decoder
+ * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
+ * Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
+ *
+ * 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 )
+ */
+
+/*
+ * 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
+ *
+ * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
+ * - HE AAC v2 comprises LC AAC with Spectral Band Replication and
+ Parametric Stereo.
+ */
+
+#include "libavutil/float_dsp.h"
+#include "libavutil/opt.h"
+#include "avcodec.h"
+#include "internal.h"
+#include "get_bits.h"
+#include "fft.h"
+#include "fmtconvert.h"
+#include "lpc.h"
+#include "kbdwin.h"
+#include "sinewin.h"
+
+#include "aac.h"
+#include "aactab.h"
+#include "aacdectab.h"
+#include "cbrt_tablegen.h"
+#include "sbr.h"
+#include "aacsbr.h"
+#include "mpeg4audio.h"
+#include "aacadtsdec.h"
+#include "libavutil/intfloat.h"
+
+#include <assert.h>
+#include <errno.h>
+#include <math.h>
+#include <string.h>
+
+#if ARCH_ARM
+# include "arm/aac.h"
+#elif ARCH_MIPS
+# include "mips/aacdec_mips.h"
+#endif
+
+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 (che_pos) {
+ if (!ac->che[type][id]) {
+ if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
+ return AVERROR(ENOMEM);
+ 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])
+ 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);
+ 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 = (float *)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;
+
+ 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] = 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;
+
+ 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;
+ }
+
+ // 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 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, id, &channels);
+ if (ret < 0)
+ return ret;
+ }
+ 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;
+ }
+ }
+
+ memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][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) {
+ av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
+ channel_config);
+ return -1;
+ }
+ *tags = tags_per_config[channel_config];
+ memcpy(layout_map, aac_channel_layout_map[channel_config-1], *tags * sizeof(*layout_map));
+ 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 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 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)) {
+ 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:
+ 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:
+ 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, 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 -1;
+ }
+ 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;
+ 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;
+ }
+
+ 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:
+ skip_bits(gb, 3); /* aacSectionDataResilienceFlag
+ * aacScalefactorDataResilienceFlag
+ * aacSpectralDataResilienceFlag
+ */
+ break;
+ }
+ skip_bits1(gb); // extensionFlag3 (TBD in version 3)
+ }
+ 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;
+ int ret;
+
+ av_dlog(avctx, "audio specific config size %d\n", bit_size >> 3);
+ for (i = 0; i < bit_size >> 3; i++)
+ av_dlog(avctx, "%02x ", data[i]);
+ av_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 -1;
+ if (m4ac->sampling_index > 12) {
+ av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
+ return -1;
+ }
+
+ skip_bits_long(&gb, i);
+
+ switch (m4ac->object_type) {
+ case AOT_AAC_MAIN:
+ case AOT_AAC_LC:
+ case AOT_AAC_LTP:
+ if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config))
+ return -1;
+ break;
+ default:
+ av_log(avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
+ m4ac->sbr == 1? "SBR+" : "", m4ac->object_type);
+ return -1;
+ }
+
+ av_dlog(avctx, "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
+ m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
+ m4ac->sample_rate, m4ac->sbr, m4ac->ps);
+
+ return get_bits_count(&gb);
+}
+
+/**
+ * linear congruential pseudorandom number generator
+ *
+ * @param previous_val pointer to the current state of the generator
+ *
+ * @return Returns a 32-bit pseudorandom integer
+ */
+static av_always_inline int lcg_random(unsigned previous_val)
+{
+ union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
+ return v.s;
+}
+
+static av_always_inline void reset_predict_state(PredictorState *ps)
+{
+ ps->r0 = 0.0f;
+ ps->r1 = 0.0f;
+ ps->cor0 = 0.0f;
+ ps->cor1 = 0.0f;
+ ps->var0 = 1.0f;
+ ps->var1 = 1.0f;
+}
+
+static void reset_all_predictors(PredictorState *ps)
+{
+ int i;
+ for (i = 0; i < MAX_PREDICTORS; i++)
+ reset_predict_state(&ps[i]);
+}
+
+static int sample_rate_idx (int rate)
+{
+ if (92017 <= rate) return 0;
+ else if (75132 <= rate) return 1;
+ else if (55426 <= rate) return 2;
+ else if (46009 <= rate) return 3;
+ else if (37566 <= rate) return 4;
+ else if (27713 <= rate) return 5;
+ else if (23004 <= rate) return 6;
+ else if (18783 <= rate) return 7;
+ else if (13856 <= rate) return 8;
+ else if (11502 <= rate) return 9;
+ else if (9391 <= rate) return 10;
+ else return 11;
+}
+
+static void reset_predictor_group(PredictorState *ps, int group_num)
+{
+ int i;
+ for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
+ reset_predict_state(&ps[i]);
+}
+
+#define AAC_INIT_VLC_STATIC(num, size) \
+ INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
+ ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
+ ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
+ size);
+
+static void aacdec_init(AACContext *ac);
+
+static av_cold int aac_decode_init(AVCodecContext *avctx)
+{
+ AACContext *ac = avctx->priv_data;
+
+ ac->avctx = avctx;
+ ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
+
+ aacdec_init(ac);
+
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+
+ if (avctx->extradata_size > 0) {
+ if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
+ avctx->extradata,
+ avctx->extradata_size*8, 1) < 0)
+ return -1;
+ } else {
+ int sr, i;
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int layout_map_tags;
+
+ sr = sample_rate_idx(avctx->sample_rate);
+ ac->oc[1].m4ac.sampling_index = sr;
+ ac->oc[1].m4ac.channels = avctx->channels;
+ ac->oc[1].m4ac.sbr = -1;
+ ac->oc[1].m4ac.ps = -1;
+
+ for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
+ if (ff_mpeg4audio_channels[i] == avctx->channels)
+ break;
+ if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
+ i = 0;
+ }
+ ac->oc[1].m4ac.chan_config = i;
+
+ if (ac->oc[1].m4ac.chan_config) {
+ int ret = set_default_channel_config(avctx, layout_map,
+ &layout_map_tags, ac->oc[1].m4ac.chan_config);
+ if (!ret)
+ output_configure(ac, layout_map, layout_map_tags,
+ OC_GLOBAL_HDR, 0);
+ else if (avctx->err_recognition & AV_EF_EXPLODE)
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ if (avctx->channels > MAX_CHANNELS) {
+ av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ AAC_INIT_VLC_STATIC( 0, 304);
+ AAC_INIT_VLC_STATIC( 1, 270);
+ AAC_INIT_VLC_STATIC( 2, 550);
+ AAC_INIT_VLC_STATIC( 3, 300);
+ AAC_INIT_VLC_STATIC( 4, 328);
+ AAC_INIT_VLC_STATIC( 5, 294);
+ AAC_INIT_VLC_STATIC( 6, 306);
+ AAC_INIT_VLC_STATIC( 7, 268);
+ AAC_INIT_VLC_STATIC( 8, 510);
+ AAC_INIT_VLC_STATIC( 9, 366);
+ AAC_INIT_VLC_STATIC(10, 462);
+
+ ff_aac_sbr_init();
+
+ ff_fmt_convert_init(&ac->fmt_conv, avctx);
+ avpriv_float_dsp_init(&ac->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+
+ ac->random_state = 0x1f2e3d4c;
+
+ ff_aac_tableinit();
+
+ INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
+ ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
+ ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
+ 352);
+
+ ff_mdct_init(&ac->mdct, 11, 1, 1.0 / (32768.0 * 1024.0));
+ ff_mdct_init(&ac->mdct_small, 8, 1, 1.0 / (32768.0 * 128.0));
+ ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0 * 32768.0);
+ // window initialization
+ ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
+ ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
+ ff_init_ff_sine_windows(10);
+ ff_init_ff_sine_windows( 7);
+
+ cbrt_tableinit();
+
+ return 0;
+}
+
+/**
+ * Skip data_stream_element; reference: table 4.10.
+ */
+static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
+{
+ int byte_align = get_bits1(gb);
+ int count = get_bits(gb, 8);
+ if (count == 255)
+ count += get_bits(gb, 8);
+ if (byte_align)
+ align_get_bits(gb);
+
+ if (get_bits_left(gb) < 8 * count) {
+ av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
+ return -1;
+ }
+ skip_bits_long(gb, 8 * count);
+ return 0;
+}
+
+static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
+ GetBitContext *gb)
+{
+ int sfb;
+ if (get_bits1(gb)) {
+ ics->predictor_reset_group = get_bits(gb, 5);
+ if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
+ av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
+ return -1;
+ }
+ }
+ for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
+ ics->prediction_used[sfb] = get_bits1(gb);
+ }
+ return 0;
+}
+
+/**
+ * Decode Long Term Prediction data; reference: table 4.xx.
+ */
+static void decode_ltp(LongTermPrediction *ltp,
+ GetBitContext *gb, uint8_t max_sfb)
+{
+ int sfb;
+
+ ltp->lag = get_bits(gb, 11);
+ ltp->coef = ltp_coef[get_bits(gb, 3)];
+ for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
+ ltp->used[sfb] = get_bits1(gb);
+}
+
+/**
+ * Decode Individual Channel Stream info; reference: table 4.6.
+ */
+static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
+ GetBitContext *gb)
+{
+ if (get_bits1(gb)) {
+ av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
+ return AVERROR_INVALIDDATA;
+ }
+ ics->window_sequence[1] = ics->window_sequence[0];
+ ics->window_sequence[0] = get_bits(gb, 2);
+ ics->use_kb_window[1] = ics->use_kb_window[0];
+ ics->use_kb_window[0] = get_bits1(gb);
+ ics->num_window_groups = 1;
+ ics->group_len[0] = 1;
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ int i;
+ ics->max_sfb = get_bits(gb, 4);
+ for (i = 0; i < 7; i++) {
+ if (get_bits1(gb)) {
+ ics->group_len[ics->num_window_groups - 1]++;
+ } else {
+ ics->num_window_groups++;
+ ics->group_len[ics->num_window_groups - 1] = 1;
+ }
+ }
+ ics->num_windows = 8;
+ ics->swb_offset = ff_swb_offset_128[ac->oc[1].m4ac.sampling_index];
+ ics->num_swb = ff_aac_num_swb_128[ac->oc[1].m4ac.sampling_index];
+ ics->tns_max_bands = ff_tns_max_bands_128[ac->oc[1].m4ac.sampling_index];
+ ics->predictor_present = 0;
+ } else {
+ ics->max_sfb = get_bits(gb, 6);
+ ics->num_windows = 1;
+ ics->swb_offset = ff_swb_offset_1024[ac->oc[1].m4ac.sampling_index];
+ ics->num_swb = ff_aac_num_swb_1024[ac->oc[1].m4ac.sampling_index];
+ ics->tns_max_bands = ff_tns_max_bands_1024[ac->oc[1].m4ac.sampling_index];
+ ics->predictor_present = get_bits1(gb);
+ ics->predictor_reset_group = 0;
+ if (ics->predictor_present) {
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
+ if (decode_prediction(ac, ics, gb)) {
+ goto fail;
+ }
+ } else if (ac->oc[1].m4ac.object_type == AOT_AAC_LC) {
+ av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
+ goto fail;
+ } else {
+ if ((ics->ltp.present = get_bits(gb, 1)))
+ decode_ltp(&ics->ltp, gb, ics->max_sfb);
+ }
+ }
+ }
+
+ if (ics->max_sfb > ics->num_swb) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
+ ics->max_sfb, ics->num_swb);
+ goto fail;
+ }
+
+ return 0;
+fail:
+ ics->max_sfb = 0;
+ return AVERROR_INVALIDDATA;
+}
+
+/**
+ * Decode band types (section_data payload); reference: table 4.46.
+ *
+ * @param band_type array of the used band type
+ * @param band_type_run_end array of the last scalefactor band of a band type run
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_band_types(AACContext *ac, enum BandType band_type[120],
+ int band_type_run_end[120], GetBitContext *gb,
+ IndividualChannelStream *ics)
+{
+ int g, idx = 0;
+ const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
+ for (g = 0; g < ics->num_window_groups; g++) {
+ int k = 0;
+ while (k < ics->max_sfb) {
+ uint8_t sect_end = k;
+ int sect_len_incr;
+ int sect_band_type = get_bits(gb, 4);
+ if (sect_band_type == 12) {
+ av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
+ return -1;
+ }
+ 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 -1;
+ }
+ 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 -1;
+ }
+ } while (sect_len_incr == (1 << bits) - 1);
+ for (; k < sect_end; k++) {
+ band_type [idx] = sect_band_type;
+ band_type_run_end[idx++] = sect_end;
+ }
+ }
+ }
+ return 0;
+}
+
+/**
+ * Decode scalefactors; reference: table 4.47.
+ *
+ * @param global_gain first scalefactor value as scalefactors are differentially coded
+ * @param band_type array of the used band type
+ * @param band_type_run_end array of the last scalefactor band of a band type run
+ * @param sf array of scalefactors or intensity stereo positions
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
+ unsigned int global_gain,
+ IndividualChannelStream *ics,
+ enum BandType band_type[120],
+ int band_type_run_end[120])
+{
+ int g, i, idx = 0;
+ int offset[3] = { global_gain, global_gain - 90, 0 };
+ int clipped_offset;
+ int noise_flag = 1;
+ for (g = 0; g < ics->num_window_groups; g++) {
+ for (i = 0; i < ics->max_sfb;) {
+ int run_end = band_type_run_end[idx];
+ if (band_type[idx] == ZERO_BT) {
+ for (; i < run_end; i++, idx++)
+ sf[idx] = 0.;
+ } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
+ for (; i < run_end; i++, idx++) {
+ offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
+ 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);
+ }
+ sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
+ }
+ } else if (band_type[idx] == NOISE_BT) {
+ for (; i < run_end; i++, idx++) {
+ if (noise_flag-- > 0)
+ offset[1] += get_bits(gb, 9) - 256;
+ else
+ offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
+ 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);
+ }
+ sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
+ }
+ } else {
+ for (; i < run_end; i++, idx++) {
+ offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
+ if (offset[0] > 255U) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "Scalefactor (%d) out of range.\n", offset[0]);
+ return -1;
+ }
+ sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
+ }
+ }
+ }
+ }
+ 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] > 1023)
+ 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] > 1023)
+ return -1;
+ pulse->amp[i] = get_bits(gb, 4);
+ }
+ return 0;
+}
+
+/**
+ * Decode Temporal Noise Shaping data; reference: table 4.48.
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
+ GetBitContext *gb, const IndividualChannelStream *ics)
+{
+ int w, filt, i, coef_len, coef_res, coef_compress;
+ const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
+ const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
+ for (w = 0; w < ics->num_windows; w++) {
+ if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
+ coef_res = get_bits1(gb);
+
+ for (filt = 0; filt < tns->n_filt[w]; filt++) {
+ int tmp2_idx;
+ tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
+
+ if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
+ av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
+ tns->order[w][filt], tns_max_order);
+ tns->order[w][filt] = 0;
+ return -1;
+ }
+ if (tns->order[w][filt]) {
+ tns->direction[w][filt] = get_bits1(gb);
+ coef_compress = get_bits1(gb);
+ coef_len = coef_res + 3 - coef_compress;
+ tmp2_idx = 2 * coef_compress + coef_res;
+
+ for (i = 0; i < tns->order[w][filt]; i++)
+ tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
+ }
+ }
+ }
+ }
+ return 0;
+}
+
+/**
+ * Decode Mid/Side data; reference: table 4.54.
+ *
+ * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
+ * [1] mask is decoded from bitstream; [2] mask is all 1s;
+ * [3] reserved for scalable AAC
+ */
+static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
+ int ms_present)
+{
+ int idx;
+ if (ms_present == 1) {
+ for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
+ cpe->ms_mask[idx] = get_bits1(gb);
+ } else if (ms_present == 2) {
+ memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask[0]) * cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb);
+ }
+}
+
+#ifndef VMUL2
+static inline float *VMUL2(float *dst, const float *v, unsigned idx,
+ const float *scale)
+{
+ float s = *scale;
+ *dst++ = v[idx & 15] * s;
+ *dst++ = v[idx>>4 & 15] * s;
+ return dst;
+}
+#endif
+
+#ifndef VMUL4
+static inline float *VMUL4(float *dst, const float *v, unsigned idx,
+ const float *scale)
+{
+ float s = *scale;
+ *dst++ = v[idx & 3] * s;
+ *dst++ = v[idx>>2 & 3] * s;
+ *dst++ = v[idx>>4 & 3] * s;
+ *dst++ = v[idx>>6 & 3] * s;
+ return dst;
+}
+#endif
+
+#ifndef VMUL2S
+static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
+ unsigned sign, const float *scale)
+{
+ union av_intfloat32 s0, s1;
+
+ s0.f = s1.f = *scale;
+ s0.i ^= sign >> 1 << 31;
+ s1.i ^= sign << 31;
+
+ *dst++ = v[idx & 15] * s0.f;
+ *dst++ = v[idx>>4 & 15] * s1.f;
+
+ return dst;
+}
+#endif
+
+#ifndef VMUL4S
+static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
+ unsigned sign, const float *scale)
+{
+ unsigned nz = idx >> 12;
+ union av_intfloat32 s = { .f = *scale };
+ union av_intfloat32 t;
+
+ t.i = s.i ^ (sign & 1U<<31);
+ *dst++ = v[idx & 3] * t.f;
+
+ sign <<= nz & 1; nz >>= 1;
+ t.i = s.i ^ (sign & 1U<<31);
+ *dst++ = v[idx>>2 & 3] * t.f;
+
+ sign <<= nz & 1; nz >>= 1;
+ t.i = s.i ^ (sign & 1U<<31);
+ *dst++ = v[idx>>4 & 3] * t.f;
+
+ sign <<= nz & 1;
+ t.i = s.i ^ (sign & 1U<<31);
+ *dst++ = v[idx>>6 & 3] * t.f;
+
+ return dst;
+}
+#endif
+
+/**
+ * Decode spectral data; reference: table 4.50.
+ * Dequantize and scale spectral data; reference: 4.6.3.3.
+ *
+ * @param coef array of dequantized, scaled spectral data
+ * @param sf array of scalefactors or intensity stereo positions
+ * @param pulse_present set if pulses are present
+ * @param pulse pointer to pulse data struct
+ * @param band_type array of the used band type
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
+ GetBitContext *gb, const float sf[120],
+ int pulse_present, const Pulse *pulse,
+ const IndividualChannelStream *ics,
+ enum BandType band_type[120])
+{
+ int i, k, g, idx = 0;
+ const int c = 1024 / ics->num_windows;
+ const uint16_t *offsets = ics->swb_offset;
+ float *coef_base = coef;
+
+ for (g = 0; g < ics->num_windows; g++)
+ memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
+
+ for (g = 0; g < ics->num_window_groups; g++) {
+ unsigned g_len = ics->group_len[g];
+
+ for (i = 0; i < ics->max_sfb; i++, idx++) {
+ const unsigned cbt_m1 = band_type[idx] - 1;
+ float *cfo = coef + offsets[i];
+ int off_len = offsets[i + 1] - offsets[i];
+ int group;
+
+ if (cbt_m1 >= INTENSITY_BT2 - 1) {
+ for (group = 0; group < g_len; group++, cfo+=128) {
+ memset(cfo, 0, off_len * sizeof(float));
+ }
+ } else if (cbt_m1 == NOISE_BT - 1) {
+ for (group = 0; group < g_len; group++, cfo+=128) {
+ float scale;
+ float band_energy;
+
+ for (k = 0; k < off_len; k++) {
+ ac->random_state = lcg_random(ac->random_state);
+ cfo[k] = ac->random_state;
+ }
+
+ band_energy = ac->fdsp.scalarproduct_float(cfo, cfo, off_len);
+ scale = sf[idx] / sqrtf(band_energy);
+ ac->fdsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
+ }
+ } else {
+ const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
+ const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
+ VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
+ OPEN_READER(re, gb);
+
+ switch (cbt_m1 >> 1) {
+ case 0:
+ for (group = 0; group < g_len; group++, cfo+=128) {
+ float *cf = cfo;
+ int len = off_len;
+
+ do {
+ int code;
+ unsigned cb_idx;
+
+ UPDATE_CACHE(re, gb);
+ GET_VLC(code, re, gb, vlc_tab, 8, 2);
+ cb_idx = cb_vector_idx[code];
+ cf = VMUL4(cf, vq, cb_idx, sf + idx);
+ } while (len -= 4);
+ }
+ break;
+
+ case 1:
+ for (group = 0; group < g_len; group++, cfo+=128) {
+ float *cf = cfo;
+ int len = off_len;
+
+ do {
+ int code;
+ unsigned nnz;
+ unsigned cb_idx;
+ uint32_t bits;
+
+ UPDATE_CACHE(re, gb);
+ GET_VLC(code, re, gb, vlc_tab, 8, 2);
+ cb_idx = cb_vector_idx[code];
+ nnz = cb_idx >> 8 & 15;
+ bits = nnz ? GET_CACHE(re, gb) : 0;
+ LAST_SKIP_BITS(re, gb, nnz);
+ cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
+ } while (len -= 4);
+ }
+ break;
+
+ case 2:
+ for (group = 0; group < g_len; group++, cfo+=128) {
+ float *cf = cfo;
+ int len = off_len;
+
+ do {
+ int code;
+ unsigned cb_idx;
+
+ UPDATE_CACHE(re, gb);
+ GET_VLC(code, re, gb, vlc_tab, 8, 2);
+ cb_idx = cb_vector_idx[code];
+ cf = VMUL2(cf, vq, cb_idx, sf + idx);
+ } while (len -= 2);
+ }
+ break;
+
+ case 3:
+ case 4:
+ for (group = 0; group < g_len; group++, cfo+=128) {
+ float *cf = cfo;
+ int len = off_len;
+
+ do {
+ int code;
+ unsigned nnz;
+ unsigned cb_idx;
+ unsigned sign;
+
+ UPDATE_CACHE(re, gb);
+ GET_VLC(code, re, gb, vlc_tab, 8, 2);
+ cb_idx = cb_vector_idx[code];
+ nnz = cb_idx >> 8 & 15;
+ sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
+ LAST_SKIP_BITS(re, gb, nnz);
+ cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
+ } while (len -= 2);
+ }
+ break;
+
+ default:
+ for (group = 0; group < g_len; group++, cfo+=128) {
+ float *cf = cfo;
+ uint32_t *icf = (uint32_t *) cf;
+ int len = off_len;
+
+ do {
+ int code;
+ unsigned nzt, nnz;
+ unsigned cb_idx;
+ uint32_t bits;
+ int j;
+
+ UPDATE_CACHE(re, gb);
+ GET_VLC(code, re, gb, vlc_tab, 8, 2);
+
+ if (!code) {
+ *icf++ = 0;
+ *icf++ = 0;
+ continue;
+ }
+
+ cb_idx = cb_vector_idx[code];
+ nnz = cb_idx >> 12;
+ nzt = cb_idx >> 8;
+ bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
+ LAST_SKIP_BITS(re, gb, nnz);
+
+ for (j = 0; j < 2; j++) {
+ if (nzt & 1<<j) {
+ uint32_t b;
+ int n;
+ /* The total length of escape_sequence must be < 22 bits according
+ to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
+ UPDATE_CACHE(re, gb);
+ b = GET_CACHE(re, gb);
+ b = 31 - av_log2(~b);
+
+ if (b > 8) {
+ av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
+ return -1;
+ }
+
+ SKIP_BITS(re, gb, b + 1);
+ b += 4;
+ n = (1 << b) + SHOW_UBITS(re, gb, b);
+ LAST_SKIP_BITS(re, gb, b);
+ *icf++ = cbrt_tab[n] | (bits & 1U<<31);
+ bits <<= 1;
+ } else {
+ unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
+ *icf++ = (bits & 1U<<31) | v;
+ bits <<= !!v;
+ }
+ cb_idx >>= 4;
+ }
+ } while (len -= 2);
+
+ ac->fdsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
+ }
+ }
+
+ CLOSE_READER(re, gb);
+ }
+ }
+ coef += g_len << 7;
+ }
+
+ if (pulse_present) {
+ idx = 0;
+ for (i = 0; i < pulse->num_pulse; i++) {
+ float co = coef_base[ pulse->pos[i] ];
+ while (offsets[idx + 1] <= pulse->pos[i])
+ idx++;
+ if (band_type[idx] != NOISE_BT && sf[idx]) {
+ float ico = -pulse->amp[i];
+ if (co) {
+ co /= sf[idx];
+ ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
+ }
+ coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
+ }
+ }
+ }
+ return 0;
+}
+
+static av_always_inline float flt16_round(float pf)
+{
+ union av_intfloat32 tmp;
+ tmp.f = pf;
+ tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
+ return tmp.f;
+}
+
+static av_always_inline float flt16_even(float pf)
+{
+ union av_intfloat32 tmp;
+ tmp.f = pf;
+ tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
+ return tmp.f;
+}
+
+static av_always_inline float flt16_trunc(float pf)
+{
+ union av_intfloat32 pun;
+ pun.f = pf;
+ pun.i &= 0xFFFF0000U;
+ return pun.f;
+}
+
+static av_always_inline void predict(PredictorState *ps, float *coef,
+ int output_enable)
+{
+ const float a = 0.953125; // 61.0 / 64
+ const float alpha = 0.90625; // 29.0 / 32
+ float e0, e1;
+ float pv;
+ float k1, k2;
+ float r0 = ps->r0, r1 = ps->r1;
+ float cor0 = ps->cor0, cor1 = ps->cor1;
+ float var0 = ps->var0, var1 = ps->var1;
+
+ k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
+ k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
+
+ pv = flt16_round(k1 * r0 + k2 * r1);
+ if (output_enable)
+ *coef += pv;
+
+ e0 = *coef;
+ e1 = e0 - k1 * r0;
+
+ ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
+ ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
+ ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
+ ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
+
+ ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
+ ps->r0 = flt16_trunc(a * e0);
+}
+
+/**
+ * Apply AAC-Main style frequency domain prediction.
+ */
+static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
+{
+ int sfb, k;
+
+ if (!sce->ics.predictor_initialized) {
+ reset_all_predictors(sce->predictor_state);
+ sce->ics.predictor_initialized = 1;
+ }
+
+ if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
+ for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]; sfb++) {
+ for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
+ predict(&sce->predictor_state[k], &sce->coeffs[k],
+ sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
+ }
+ }
+ if (sce->ics.predictor_reset_group)
+ reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
+ } else
+ reset_all_predictors(sce->predictor_state);
+}
+
+/**
+ * Decode an individual_channel_stream payload; reference: table 4.44.
+ *
+ * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
+ * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_ics(AACContext *ac, SingleChannelElement *sce,
+ GetBitContext *gb, int common_window, int scale_flag)
+{
+ Pulse pulse;
+ TemporalNoiseShaping *tns = &sce->tns;
+ IndividualChannelStream *ics = &sce->ics;
+ float *out = sce->coeffs;
+ int global_gain, pulse_present = 0;
+
+ /* This assignment is to silence a GCC warning about the variable being used
+ * uninitialized when in fact it always is.
+ */
+ pulse.num_pulse = 0;
+
+ global_gain = get_bits(gb, 8);
+
+ if (!common_window && !scale_flag) {
+ if (decode_ics_info(ac, ics, gb) < 0)
+ return AVERROR_INVALIDDATA;
+ }
+
+ if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
+ return -1;
+ if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
+ return -1;
+
+ pulse_present = 0;
+ if (!scale_flag) {
+ if ((pulse_present = get_bits1(gb))) {
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
+ return -1;
+ }
+ if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
+ av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
+ return -1;
+ }
+ }
+ if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
+ return -1;
+ if (get_bits1(gb)) {
+ avpriv_request_sample(ac->avctx, "SSR");
+ return AVERROR_PATCHWELCOME;
+ }
+ }
+
+ if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
+ return -1;
+
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
+ apply_prediction(ac, sce);
+
+ return 0;
+}
+
+/**
+ * Mid/Side stereo decoding; reference: 4.6.8.1.3.
+ */
+static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
+{
+ const IndividualChannelStream *ics = &cpe->ch[0].ics;
+ float *ch0 = cpe->ch[0].coeffs;
+ float *ch1 = cpe->ch[1].coeffs;
+ int g, i, group, idx = 0;
+ const uint16_t *offsets = ics->swb_offset;
+ for (g = 0; g < ics->num_window_groups; g++) {
+ for (i = 0; i < ics->max_sfb; i++, idx++) {
+ if (cpe->ms_mask[idx] &&
+ cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
+ for (group = 0; group < ics->group_len[g]; group++) {
+ ac->fdsp.butterflies_float(ch0 + group * 128 + offsets[i],
+ ch1 + group * 128 + offsets[i],
+ offsets[i+1] - offsets[i]);
+ }
+ }
+ }
+ ch0 += ics->group_len[g] * 128;
+ ch1 += ics->group_len[g] * 128;
+ }
+}
+
+/**
+ * intensity stereo decoding; reference: 4.6.8.2.3
+ *
+ * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
+ * [1] mask is decoded from bitstream; [2] mask is all 1s;
+ * [3] reserved for scalable AAC
+ */
+static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present)
+{
+ const IndividualChannelStream *ics = &cpe->ch[1].ics;
+ SingleChannelElement *sce1 = &cpe->ch[1];
+ float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
+ const uint16_t *offsets = ics->swb_offset;
+ int g, group, i, idx = 0;
+ int c;
+ float scale;
+ for (g = 0; g < ics->num_window_groups; g++) {
+ for (i = 0; i < ics->max_sfb;) {
+ if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
+ const int bt_run_end = sce1->band_type_run_end[idx];
+ for (; i < bt_run_end; i++, idx++) {
+ c = -1 + 2 * (sce1->band_type[idx] - 14);
+ if (ms_present)
+ c *= 1 - 2 * cpe->ms_mask[idx];
+ scale = c * sce1->sf[idx];
+ for (group = 0; group < ics->group_len[g]; group++)
+ ac->fdsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
+ coef0 + group * 128 + offsets[i],
+ scale,
+ offsets[i + 1] - offsets[i]);
+ }
+ } else {
+ int bt_run_end = sce1->band_type_run_end[idx];
+ idx += bt_run_end - i;
+ i = bt_run_end;
+ }
+ }
+ coef0 += ics->group_len[g] * 128;
+ coef1 += ics->group_len[g] * 128;
+ }
+}
+
+/**
+ * Decode a channel_pair_element; reference: table 4.4.
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
+{
+ int i, ret, common_window, ms_present = 0;
+
+ common_window = get_bits1(gb);
+ if (common_window) {
+ if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
+ return AVERROR_INVALIDDATA;
+ i = cpe->ch[1].ics.use_kb_window[0];
+ cpe->ch[1].ics = cpe->ch[0].ics;
+ cpe->ch[1].ics.use_kb_window[1] = i;
+ if (cpe->ch[1].ics.predictor_present && (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
+ if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
+ decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
+ ms_present = get_bits(gb, 2);
+ if (ms_present == 3) {
+ av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
+ return -1;
+ } else if (ms_present)
+ decode_mid_side_stereo(cpe, gb, ms_present);
+ }
+ if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
+ return ret;
+ if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
+ return ret;
+
+ if (common_window) {
+ if (ms_present)
+ apply_mid_side_stereo(ac, cpe);
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
+ apply_prediction(ac, &cpe->ch[0]);
+ apply_prediction(ac, &cpe->ch[1]);
+ }
+ }
+
+ apply_intensity_stereo(ac, cpe, ms_present);
+ return 0;
+}
+
+static const float cce_scale[] = {
+ 1.09050773266525765921, //2^(1/8)
+ 1.18920711500272106672, //2^(1/4)
+ M_SQRT2,
+ 2,
+};
+
+/**
+ * Decode coupling_channel_element; reference: table 4.8.
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
+{
+ int num_gain = 0;
+ int c, g, sfb, ret;
+ int sign;
+ float scale;
+ SingleChannelElement *sce = &che->ch[0];
+ ChannelCoupling *coup = &che->coup;
+
+ coup->coupling_point = 2 * get_bits1(gb);
+ coup->num_coupled = get_bits(gb, 3);
+ for (c = 0; c <= coup->num_coupled; c++) {
+ num_gain++;
+ coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
+ coup->id_select[c] = get_bits(gb, 4);
+ if (coup->type[c] == TYPE_CPE) {
+ coup->ch_select[c] = get_bits(gb, 2);
+ if (coup->ch_select[c] == 3)
+ num_gain++;
+ } else
+ coup->ch_select[c] = 2;
+ }
+ coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
+
+ sign = get_bits(gb, 1);
+ scale = cce_scale[get_bits(gb, 2)];
+
+ if ((ret = decode_ics(ac, sce, gb, 0, 0)))
+ return ret;
+
+ for (c = 0; c < num_gain; c++) {
+ int idx = 0;
+ int cge = 1;
+ int gain = 0;
+ float gain_cache = 1.;
+ if (c) {
+ cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
+ gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
+ gain_cache = powf(scale, -gain);
+ }
+ if (coup->coupling_point == AFTER_IMDCT) {
+ coup->gain[c][0] = gain_cache;
+ } else {
+ for (g = 0; g < sce->ics.num_window_groups; g++) {
+ for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
+ if (sce->band_type[idx] != ZERO_BT) {
+ if (!cge) {
+ int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
+ if (t) {
+ int s = 1;
+ t = gain += t;
+ if (sign) {
+ s -= 2 * (t & 0x1);
+ t >>= 1;
+ }
+ gain_cache = powf(scale, -t) * s;
+ }
+ }
+ coup->gain[c][idx] = gain_cache;
+ }
+ }
+ }
+ }
+ }
+ return 0;
+}
+
+/**
+ * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
+ *
+ * @return Returns number of bytes consumed.
+ */
+static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
+ GetBitContext *gb)
+{
+ int i;
+ int num_excl_chan = 0;
+
+ do {
+ for (i = 0; i < 7; i++)
+ che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
+ } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
+
+ return num_excl_chan / 7;
+}
+
+/**
+ * Decode dynamic range information; reference: table 4.52.
+ *
+ * @return Returns number of bytes consumed.
+ */
+static int decode_dynamic_range(DynamicRangeControl *che_drc,
+ GetBitContext *gb)
+{
+ int n = 1;
+ int drc_num_bands = 1;
+ int i;
+
+ /* pce_tag_present? */
+ if (get_bits1(gb)) {
+ che_drc->pce_instance_tag = get_bits(gb, 4);
+ skip_bits(gb, 4); // tag_reserved_bits
+ n++;
+ }
+
+ /* excluded_chns_present? */
+ if (get_bits1(gb)) {
+ n += decode_drc_channel_exclusions(che_drc, gb);
+ }
+
+ /* drc_bands_present? */
+ if (get_bits1(gb)) {
+ che_drc->band_incr = get_bits(gb, 4);
+ che_drc->interpolation_scheme = get_bits(gb, 4);
+ n++;
+ drc_num_bands += che_drc->band_incr;
+ for (i = 0; i < drc_num_bands; i++) {
+ che_drc->band_top[i] = get_bits(gb, 8);
+ n++;
+ }
+ }
+
+ /* prog_ref_level_present? */
+ if (get_bits1(gb)) {
+ che_drc->prog_ref_level = get_bits(gb, 7);
+ skip_bits1(gb); // prog_ref_level_reserved_bits
+ n++;
+ }
+
+ for (i = 0; i < drc_num_bands; i++) {
+ che_drc->dyn_rng_sgn[i] = get_bits1(gb);
+ che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
+ n++;
+ }
+
+ return n;
+}
+
+static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
+ uint8_t buf[256];
+ int i, major, minor;
+
+ if (len < 13+7*8)
+ goto unknown;
+
+ get_bits(gb, 13); len -= 13;
+
+ for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
+ buf[i] = get_bits(gb, 8);
+
+ buf[i] = 0;
+ if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
+ av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
+
+ if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
+ ac->avctx->internal->skip_samples = 1024;
+ }
+
+unknown:
+ skip_bits_long(gb, len);
+
+ return 0;
+}
+
+/**
+ * Decode extension data (incomplete); reference: table 4.51.
+ *
+ * @param cnt length of TYPE_FIL syntactic element in bytes
+ *
+ * @return Returns number of bytes consumed
+ */
+static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
+ ChannelElement *che, enum RawDataBlockType elem_type)
+{
+ int crc_flag = 0;
+ int res = cnt;
+ switch (get_bits(gb, 4)) { // extension type
+ case EXT_SBR_DATA_CRC:
+ crc_flag++;
+ case EXT_SBR_DATA:
+ if (!che) {
+ av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
+ return res;
+ } else if (!ac->oc[1].m4ac.sbr) {
+ av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
+ skip_bits_long(gb, 8 * cnt - 4);
+ return res;
+ } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
+ av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
+ skip_bits_long(gb, 8 * cnt - 4);
+ return res;
+ } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
+ ac->oc[1].m4ac.sbr = 1;
+ ac->oc[1].m4ac.ps = 1;
+ 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;
+ }
+ res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
+ break;
+ case EXT_DYNAMIC_RANGE:
+ res = decode_dynamic_range(&ac->che_drc, gb);
+ break;
+ case EXT_FILL:
+ decode_fill(ac, gb, 8 * cnt - 4);
+ break;
+ case EXT_FILL_DATA:
+ case EXT_DATA_ELEMENT:
+ default:
+ skip_bits_long(gb, 8 * cnt - 4);
+ break;
+ };
+ return res;
+}
+
+/**
+ * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
+ *
+ * @param decode 1 if tool is used normally, 0 if tool is used in LTP.
+ * @param coef spectral coefficients
+ */
+static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
+ IndividualChannelStream *ics, int decode)
+{
+ const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
+ int w, filt, m, i;
+ int bottom, top, order, start, end, size, inc;
+ float lpc[TNS_MAX_ORDER];
+ float tmp[TNS_MAX_ORDER+1];
+
+ for (w = 0; w < ics->num_windows; w++) {
+ bottom = ics->num_swb;
+ for (filt = 0; filt < tns->n_filt[w]; filt++) {
+ top = bottom;
+ bottom = FFMAX(0, top - tns->length[w][filt]);
+ order = tns->order[w][filt];
+ if (order == 0)
+ continue;
+
+ // tns_decode_coef
+ compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
+
+ start = ics->swb_offset[FFMIN(bottom, mmm)];
+ end = ics->swb_offset[FFMIN( top, mmm)];
+ if ((size = end - start) <= 0)
+ continue;
+ if (tns->direction[w][filt]) {
+ inc = -1;
+ start = end - 1;
+ } else {
+ inc = 1;
+ }
+ start += w * 128;
+
+ if (decode) {
+ // ar filter
+ for (m = 0; m < size; m++, start += inc)
+ for (i = 1; i <= FFMIN(m, order); i++)
+ coef[start] -= coef[start - i * inc] * lpc[i - 1];
+ } else {
+ // ma filter
+ for (m = 0; m < size; m++, start += inc) {
+ tmp[0] = coef[start];
+ for (i = 1; i <= FFMIN(m, order); i++)
+ coef[start] += tmp[i] * lpc[i - 1];
+ for (i = order; i > 0; i--)
+ tmp[i] = tmp[i - 1];
+ }
+ }
+ }
+ }
+}
+
+/**
+ * Apply windowing and MDCT to obtain the spectral
+ * coefficient from the predicted sample by LTP.
+ */
+static void windowing_and_mdct_ltp(AACContext *ac, float *out,
+ float *in, IndividualChannelStream *ics)
+{
+ const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+ const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
+ const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+ const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
+
+ if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
+ ac->fdsp.vector_fmul(in, in, lwindow_prev, 1024);
+ } else {
+ memset(in, 0, 448 * sizeof(float));
+ ac->fdsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
+ }
+ if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
+ ac->fdsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
+ } else {
+ ac->fdsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
+ memset(in + 1024 + 576, 0, 448 * sizeof(float));
+ }
+ ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
+}
+
+/**
+ * Apply the long term prediction
+ */
+static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
+{
+ const LongTermPrediction *ltp = &sce->ics.ltp;
+ const uint16_t *offsets = sce->ics.swb_offset;
+ int i, sfb;
+
+ if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
+ float *predTime = sce->ret;
+ float *predFreq = ac->buf_mdct;
+ int16_t num_samples = 2048;
+
+ if (ltp->lag < 1024)
+ num_samples = ltp->lag + 1024;
+ for (i = 0; i < num_samples; i++)
+ predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
+ memset(&predTime[i], 0, (2048 - i) * sizeof(float));
+
+ ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
+
+ if (sce->tns.present)
+ ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
+
+ for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
+ if (ltp->used[sfb])
+ for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
+ sce->coeffs[i] += predFreq[i];
+ }
+}
+
+/**
+ * Update the LTP buffer for next frame
+ */
+static void update_ltp(AACContext *ac, SingleChannelElement *sce)
+{
+ IndividualChannelStream *ics = &sce->ics;
+ float *saved = sce->saved;
+ float *saved_ltp = sce->coeffs;
+ const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+ const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
+ int i;
+
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ memcpy(saved_ltp, saved, 512 * sizeof(float));
+ memset(saved_ltp + 576, 0, 448 * sizeof(float));
+ ac->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
+ for (i = 0; i < 64; i++)
+ saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
+ } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
+ memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(float));
+ memset(saved_ltp + 576, 0, 448 * sizeof(float));
+ ac->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
+ for (i = 0; i < 64; i++)
+ saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
+ } else { // LONG_STOP or ONLY_LONG
+ ac->fdsp.vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
+ for (i = 0; i < 512; i++)
+ saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
+ }
+
+ memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
+ memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
+ memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
+}
+
+/**
+ * Conduct IMDCT and windowing.
+ */
+static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
+{
+ IndividualChannelStream *ics = &sce->ics;
+ float *in = sce->coeffs;
+ float *out = sce->ret;
+ float *saved = sce->saved;
+ const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
+ const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+ const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
+ float *buf = ac->buf_mdct;
+ float *temp = ac->temp;
+ int i;
+
+ // imdct
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ for (i = 0; i < 1024; i += 128)
+ ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
+ } else
+ ac->mdct.imdct_half(&ac->mdct, buf, in);
+
+ /* window overlapping
+ * NOTE: To simplify the overlapping code, all 'meaningless' short to long
+ * and long to short transitions are considered to be short to short
+ * transitions. This leaves just two cases (long to long and short to short)
+ * with a little special sauce for EIGHT_SHORT_SEQUENCE.
+ */
+ if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
+ (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
+ ac->fdsp.vector_fmul_window( out, saved, buf, lwindow_prev, 512);
+ } else {
+ memcpy( out, saved, 448 * sizeof(float));
+
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ ac->fdsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
+ ac->fdsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
+ ac->fdsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
+ ac->fdsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
+ ac->fdsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
+ memcpy( out + 448 + 4*128, temp, 64 * sizeof(float));
+ } else {
+ ac->fdsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
+ memcpy( out + 576, buf + 64, 448 * sizeof(float));
+ }
+ }
+
+ // buffer update
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ memcpy( saved, temp + 64, 64 * sizeof(float));
+ ac->fdsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
+ ac->fdsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
+ ac->fdsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
+ memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
+ } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
+ memcpy( saved, buf + 512, 448 * sizeof(float));
+ memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
+ } else { // LONG_STOP or ONLY_LONG
+ memcpy( saved, buf + 512, 512 * sizeof(float));
+ }
+}
+
+/**
+ * Apply dependent channel coupling (applied before IMDCT).
+ *
+ * @param index index into coupling gain array
+ */
+static void apply_dependent_coupling(AACContext *ac,
+ SingleChannelElement *target,
+ ChannelElement *cce, int index)
+{
+ IndividualChannelStream *ics = &cce->ch[0].ics;
+ const uint16_t *offsets = ics->swb_offset;
+ float *dest = target->coeffs;
+ const float *src = cce->ch[0].coeffs;
+ int g, i, group, k, idx = 0;
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "Dependent coupling is not supported together with LTP\n");
+ return;
+ }
+ for (g = 0; g < ics->num_window_groups; g++) {
+ for (i = 0; i < ics->max_sfb; i++, idx++) {
+ if (cce->ch[0].band_type[idx] != ZERO_BT) {
+ const float gain = cce->coup.gain[index][idx];
+ for (group = 0; group < ics->group_len[g]; group++) {
+ for (k = offsets[i]; k < offsets[i + 1]; k++) {
+ // XXX dsputil-ize
+ dest[group * 128 + k] += gain * src[group * 128 + k];
+ }
+ }
+ }
+ }
+ dest += ics->group_len[g] * 128;
+ src += ics->group_len[g] * 128;
+ }
+}
+
+/**
+ * Apply independent channel coupling (applied after IMDCT).
+ *
+ * @param index index into coupling gain array
+ */
+static void apply_independent_coupling(AACContext *ac,
+ SingleChannelElement *target,
+ ChannelElement *cce, int index)
+{
+ int i;
+ const float gain = cce->coup.gain[index][0];
+ const float *src = cce->ch[0].ret;
+ float *dest = target->ret;
+ const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
+
+ for (i = 0; i < len; i++)
+ dest[i] += gain * src[i];
+}
+
+/**
+ * channel coupling transformation interface
+ *
+ * @param apply_coupling_method pointer to (in)dependent coupling function
+ */
+static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
+ enum RawDataBlockType type, int elem_id,
+ enum CouplingPoint coupling_point,
+ void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
+{
+ int i, c;
+
+ for (i = 0; i < MAX_ELEM_ID; i++) {
+ ChannelElement *cce = ac->che[TYPE_CCE][i];
+ int index = 0;
+
+ if (cce && cce->coup.coupling_point == coupling_point) {
+ ChannelCoupling *coup = &cce->coup;
+
+ for (c = 0; c <= coup->num_coupled; c++) {
+ if (coup->type[c] == type && coup->id_select[c] == elem_id) {
+ if (coup->ch_select[c] != 1) {
+ apply_coupling_method(ac, &cc->ch[0], cce, index);
+ if (coup->ch_select[c] != 0)
+ index++;
+ }
+ if (coup->ch_select[c] != 2)
+ apply_coupling_method(ac, &cc->ch[1], cce, index++);
+ } else
+ index += 1 + (coup->ch_select[c] == 3);
+ }
+ }
+ }
+}
+
+/**
+ * Convert spectral data to float samples, applying all supported tools as appropriate.
+ */
+static void spectral_to_sample(AACContext *ac)
+{
+ int i, type;
+ for (type = 3; type >= 0; type--) {
+ for (i = 0; i < MAX_ELEM_ID; i++) {
+ ChannelElement *che = ac->che[type][i];
+ if (che) {
+ if (type <= TYPE_CPE)
+ apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
+ if (che->ch[0].ics.predictor_present) {
+ if (che->ch[0].ics.ltp.present)
+ ac->apply_ltp(ac, &che->ch[0]);
+ if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
+ ac->apply_ltp(ac, &che->ch[1]);
+ }
+ }
+ if (che->ch[0].tns.present)
+ ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
+ if (che->ch[1].tns.present)
+ ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
+ if (type <= TYPE_CPE)
+ apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
+ if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
+ ac->imdct_and_windowing(ac, &che->ch[0]);
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
+ ac->update_ltp(ac, &che->ch[0]);
+ if (type == TYPE_CPE) {
+ ac->imdct_and_windowing(ac, &che->ch[1]);
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
+ ac->update_ltp(ac, &che->ch[1]);
+ }
+ if (ac->oc[1].m4ac.sbr > 0) {
+ ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
+ }
+ }
+ if (type <= TYPE_CCE)
+ apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
+ }
+ }
+ }
+}
+
+static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
+{
+ int size;
+ AACADTSHeaderInfo hdr_info;
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int layout_map_tags;
+
+ size = avpriv_aac_parse_header(gb, &hdr_info);
+ if (size > 0) {
+ if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
+ // This is 2 for "VLB " audio in NSV files.
+ // See samples/nsv/vlb_audio.
+ avpriv_report_missing_feature(ac->avctx,
+ "More than one AAC RDB per ADTS frame");
+ ac->warned_num_aac_frames = 1;
+ }
+ push_output_configuration(ac);
+ if (hdr_info.chan_config) {
+ ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
+ if (set_default_channel_config(ac->avctx, layout_map,
+ &layout_map_tags, hdr_info.chan_config))
+ return -7;
+ if (output_configure(ac, layout_map, layout_map_tags,
+ FFMAX(ac->oc[1].status, OC_TRIAL_FRAME), 0))
+ return -7;
+ } 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;
+ 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_frame_int(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
+{
+ AACContext *ac = avctx->priv_data;
+ ChannelElement *che = NULL, *che_prev = NULL;
+ enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
+ int err, elem_id;
+ int samples = 0, multiplier, audio_found = 0, pce_found = 0;
+ int is_dmono, sce_count = 0;
+
+ ac->frame = data;
+
+ if (show_bits(gb, 12) == 0xfff) {
+ if (parse_adts_frame_header(ac, gb) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
+ err = -1;
+ 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 = -1;
+ goto fail;
+ }
+ }
+
+ if (frame_configure_elements(avctx) < 0) {
+ err = -1;
+ goto fail;
+ }
+
+ ac->tags_mapped = 0;
+ // parse
+ while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
+ elem_id = get_bits(gb, 4);
+
+ if (elem_type < TYPE_DSE) {
+ if (!(che=get_che(ac, elem_type, elem_id))) {
+ av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
+ elem_type, elem_id);
+ err = -1;
+ goto fail;
+ }
+ samples = 1024;
+ }
+
+ switch (elem_type) {
+
+ case TYPE_SCE:
+ err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+ audio_found = 1;
+ sce_count++;
+ break;
+
+ case TYPE_CPE:
+ err = decode_cpe(ac, gb, che);
+ audio_found = 1;
+ break;
+
+ case TYPE_CCE:
+ err = decode_cce(ac, gb, che);
+ break;
+
+ case TYPE_LFE:
+ err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+ audio_found = 1;
+ break;
+
+ case TYPE_DSE:
+ err = skip_data_stream_element(ac, gb);
+ break;
+
+ case TYPE_PCE: {
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int tags;
+ push_output_configuration(ac);
+ tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb);
+ if (tags < 0) {
+ err = tags;
+ break;
+ }
+ if (pce_found) {
+ av_log(avctx, AV_LOG_ERROR,
+ "Not evaluating a further program_config_element as this construct is dubious at best.\n");
+ } else {
+ err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
+ if (!err)
+ ac->oc[1].m4ac.chan_config = 0;
+ pce_found = 1;
+ }
+ break;
+ }
+
+ case TYPE_FIL:
+ if (elem_id == 15)
+ elem_id += get_bits(gb, 8) - 1;
+ if (get_bits_left(gb) < 8 * elem_id) {
+ av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
+ err = -1;
+ goto fail;
+ }
+ while (elem_id > 0)
+ elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
+ err = 0; /* FIXME */
+ break;
+
+ default:
+ err = -1; /* should not happen, but keeps compiler happy */
+ break;
+ }
+
+ che_prev = che;
+ elem_type_prev = elem_type;
+
+ if (err)
+ goto fail;
+
+ if (get_bits_left(gb) < 3) {
+ av_log(avctx, AV_LOG_ERROR, overread_err);
+ err = -1;
+ goto fail;
+ }
+ }
+
+ spectral_to_sample(ac);
+
+ multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
+ samples <<= multiplier;
+ /* for dual-mono audio (SCE + SCE) */
+ is_dmono = ac->dmono_mode && sce_count == 2 &&
+ ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
+
+ if (samples)
+ ac->frame->nb_samples = samples;
+ *got_frame_ptr = !!samples;
+
+ if (is_dmono) {
+ if (ac->dmono_mode == 1)
+ ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
+ else if (ac->dmono_mode == 2)
+ ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
+ }
+
+ if (ac->oc[1].status && audio_found) {
+ avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
+ avctx->frame_size = samples;
+ ac->oc[1].status = OC_LOCKED;
+ }
+
+ if (multiplier) {
+ int side_size;
+ const uint8_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);
+ if (side && side_size>=4)
+ AV_WL32(side, 2*AV_RL32(side));
+ }
+ return 0;
+fail:
+ pop_output_configuration(ac);
+ return err;
+}
+
+static int aac_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
+{
+ AACContext *ac = avctx->priv_data;
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+ GetBitContext gb;
+ int buf_consumed;
+ int buf_offset;
+ int err;
+ int new_extradata_size;
+ const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
+ AV_PKT_DATA_NEW_EXTRADATA,
+ &new_extradata_size);
+ int jp_dualmono_size;
+ const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
+ AV_PKT_DATA_JP_DUALMONO,
+ &jp_dualmono_size);
+
+ if (new_extradata && 0) {
+ av_free(avctx->extradata);
+ avctx->extradata = av_mallocz(new_extradata_size +
+ FF_INPUT_BUFFER_PADDING_SIZE);
+ if (!avctx->extradata)
+ return AVERROR(ENOMEM);
+ avctx->extradata_size = new_extradata_size;
+ memcpy(avctx->extradata, new_extradata, new_extradata_size);
+ push_output_configuration(ac);
+ if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
+ avctx->extradata,
+ avctx->extradata_size*8, 1) < 0) {
+ pop_output_configuration(ac);
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ ac->dmono_mode = 0;
+ if (jp_dualmono && jp_dualmono_size > 0)
+ ac->dmono_mode = 1 + *jp_dualmono;
+ if (ac->force_dmono_mode >= 0)
+ ac->dmono_mode = ac->force_dmono_mode;
+
+ if (INT_MAX / 8 <= buf_size)
+ return AVERROR_INVALIDDATA;
+
+ init_get_bits(&gb, buf, buf_size * 8);
+
+ if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt)) < 0)
+ return err;
+
+ buf_consumed = (get_bits_count(&gb) + 7) >> 3;
+ for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
+ if (buf[buf_offset])
+ break;
+
+ return buf_size > buf_offset ? buf_consumed : buf_size;
+}
+
+static av_cold int aac_decode_close(AVCodecContext *avctx)
+{
+ AACContext *ac = avctx->priv_data;
+ int i, type;
+
+ for (i = 0; i < MAX_ELEM_ID; i++) {
+ for (type = 0; type < 4; type++) {
+ if (ac->che[type][i])
+ ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
+ av_freep(&ac->che[type][i]);
+ }
+ }
+
+ ff_mdct_end(&ac->mdct);
+ ff_mdct_end(&ac->mdct_small);
+ ff_mdct_end(&ac->mdct_ltp);
+ return 0;
+}
+
+
+#define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word
+
+struct LATMContext {
+ AACContext aac_ctx; ///< containing AACContext
+ int initialized; ///< initialized after a valid extradata was seen
+
+ // parser data
+ int audio_mux_version_A; ///< LATM syntax version
+ int frame_length_type; ///< 0/1 variable/fixed frame length
+ int frame_length; ///< frame length for fixed frame length
+};
+
+static inline uint32_t latm_get_value(GetBitContext *b)
+{
+ int length = get_bits(b, 2);
+
+ return get_bits_long(b, (length+1)*8);
+}
+
+static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
+ GetBitContext *gb, int asclen)
+{
+ AACContext *ac = &latmctx->aac_ctx;
+ AVCodecContext *avctx = ac->avctx;
+ MPEG4AudioConfig m4ac = { 0 };
+ int config_start_bit = get_bits_count(gb);
+ int sync_extension = 0;
+ int bits_consumed, esize;
+
+ if (asclen) {
+ sync_extension = 1;
+ asclen = FFMIN(asclen, get_bits_left(gb));
+ } else
+ asclen = get_bits_left(gb);
+
+ if (config_start_bit % 8) {
+ avpriv_request_sample(latmctx->aac_ctx.avctx,
+ "Non-byte-aligned audio-specific config");
+ return AVERROR_PATCHWELCOME;
+ }
+ if (asclen <= 0)
+ return AVERROR_INVALIDDATA;
+ bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac,
+ gb->buffer + (config_start_bit / 8),
+ asclen, sync_extension);
+
+ if (bits_consumed < 0)
+ return AVERROR_INVALIDDATA;
+
+ if (!latmctx->initialized ||
+ ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
+ ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
+
+ if(latmctx->initialized) {
+ av_log(avctx, AV_LOG_INFO, "audio config changed\n");
+ } else {
+ av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
+ }
+ latmctx->initialized = 0;
+
+ esize = (bits_consumed+7) / 8;
+
+ if (avctx->extradata_size < esize) {
+ av_free(avctx->extradata);
+ avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
+ if (!avctx->extradata)
+ return AVERROR(ENOMEM);
+ }
+
+ avctx->extradata_size = esize;
+ memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
+ memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
+ }
+ skip_bits_long(gb, bits_consumed);
+
+ return bits_consumed;
+}
+
+static int read_stream_mux_config(struct LATMContext *latmctx,
+ GetBitContext *gb)
+{
+ int ret, audio_mux_version = get_bits(gb, 1);
+
+ latmctx->audio_mux_version_A = 0;
+ if (audio_mux_version)
+ latmctx->audio_mux_version_A = get_bits(gb, 1);
+
+ if (!latmctx->audio_mux_version_A) {
+
+ if (audio_mux_version)
+ latm_get_value(gb); // taraFullness
+
+ skip_bits(gb, 1); // allStreamSameTimeFraming
+ skip_bits(gb, 6); // numSubFrames
+ // numPrograms
+ if (get_bits(gb, 4)) { // numPrograms
+ avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // for each program (which there is only one in DVB)
+
+ // for each layer (which there is only one in DVB)
+ if (get_bits(gb, 3)) { // numLayer
+ avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // for all but first stream: use_same_config = get_bits(gb, 1);
+ if (!audio_mux_version) {
+ if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
+ return ret;
+ } else {
+ int ascLen = latm_get_value(gb);
+ if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
+ return ret;
+ ascLen -= ret;
+ skip_bits_long(gb, ascLen);
+ }
+
+ latmctx->frame_length_type = get_bits(gb, 3);
+ switch (latmctx->frame_length_type) {
+ case 0:
+ skip_bits(gb, 8); // latmBufferFullness
+ break;
+ case 1:
+ latmctx->frame_length = get_bits(gb, 9);
+ break;
+ case 3:
+ case 4:
+ case 5:
+ skip_bits(gb, 6); // CELP frame length table index
+ break;
+ case 6:
+ case 7:
+ skip_bits(gb, 1); // HVXC frame length table index
+ break;
+ }
+
+ if (get_bits(gb, 1)) { // other data
+ if (audio_mux_version) {
+ latm_get_value(gb); // other_data_bits
+ } else {
+ int esc;
+ do {
+ esc = get_bits(gb, 1);
+ skip_bits(gb, 8);
+ } while (esc);
+ }
+ }
+
+ if (get_bits(gb, 1)) // crc present
+ skip_bits(gb, 8); // config_crc
+ }
+
+ return 0;
+}
+
+static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
+{
+ uint8_t tmp;
+
+ if (ctx->frame_length_type == 0) {
+ int mux_slot_length = 0;
+ do {
+ tmp = get_bits(gb, 8);
+ mux_slot_length += tmp;
+ } while (tmp == 255);
+ return mux_slot_length;
+ } else if (ctx->frame_length_type == 1) {
+ return ctx->frame_length;
+ } else if (ctx->frame_length_type == 3 ||
+ ctx->frame_length_type == 5 ||
+ ctx->frame_length_type == 7) {
+ skip_bits(gb, 2); // mux_slot_length_coded
+ }
+ return 0;
+}
+
+static int read_audio_mux_element(struct LATMContext *latmctx,
+ GetBitContext *gb)
+{
+ int err;
+ uint8_t use_same_mux = get_bits(gb, 1);
+ if (!use_same_mux) {
+ if ((err = read_stream_mux_config(latmctx, gb)) < 0)
+ return err;
+ } else if (!latmctx->aac_ctx.avctx->extradata) {
+ av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
+ "no decoder config found\n");
+ return AVERROR(EAGAIN);
+ }
+ if (latmctx->audio_mux_version_A == 0) {
+ int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
+ if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
+ av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
+ return AVERROR_INVALIDDATA;
+ } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
+ av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
+ "frame length mismatch %d << %d\n",
+ mux_slot_length_bytes * 8, get_bits_left(gb));
+ return AVERROR_INVALIDDATA;
+ }
+ }
+ return 0;
+}
+
+
+static int latm_decode_frame(AVCodecContext *avctx, void *out,
+ int *got_frame_ptr, AVPacket *avpkt)
+{
+ struct LATMContext *latmctx = avctx->priv_data;
+ int muxlength, err;
+ GetBitContext gb;
+
+ if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
+ return err;
+
+ // check for LOAS sync word
+ if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
+ return AVERROR_INVALIDDATA;
+
+ muxlength = get_bits(&gb, 13) + 3;
+ // not enough data, the parser should have sorted this out
+ if (muxlength > avpkt->size)
+ return AVERROR_INVALIDDATA;
+
+ if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
+ return err;
+
+ if (!latmctx->initialized) {
+ if (!avctx->extradata) {
+ *got_frame_ptr = 0;
+ return avpkt->size;
+ } else {
+ push_output_configuration(&latmctx->aac_ctx);
+ if ((err = decode_audio_specific_config(
+ &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
+ avctx->extradata, avctx->extradata_size*8, 1)) < 0) {
+ pop_output_configuration(&latmctx->aac_ctx);
+ return err;
+ }
+ latmctx->initialized = 1;
+ }
+ }
+
+ if (show_bits(&gb, 12) == 0xfff) {
+ av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
+ "ADTS header detected, probably as result of configuration "
+ "misparsing\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt)) < 0)
+ return err;
+
+ return muxlength;
+}
+
+static av_cold int latm_decode_init(AVCodecContext *avctx)
+{
+ struct LATMContext *latmctx = avctx->priv_data;
+ int ret = aac_decode_init(avctx);
+
+ if (avctx->extradata_size > 0)
+ latmctx->initialized = !ret;
+
+ return ret;
+}
+
+static void aacdec_init(AACContext *c)
+{
+ c->imdct_and_windowing = imdct_and_windowing;
+ c->apply_ltp = apply_ltp;
+ c->apply_tns = apply_tns;
+ c->windowing_and_mdct_ltp = windowing_and_mdct_ltp;
+ c->update_ltp = update_ltp;
+
+ if(ARCH_MIPS)
+ ff_aacdec_init_mips(c);
+}
+/**
+ * AVOptions for Japanese DTV specific extensions (ADTS only)
+ */
+#define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
+static const AVOption options[] = {
+ {"dual_mono_mode", "Select the channel to decode for dual mono",
+ offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
+ AACDEC_FLAGS, "dual_mono_mode"},
+
+ {"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+ {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+ {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+ {"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+
+ {NULL},
+};
+
+static const AVClass aac_decoder_class = {
+ .class_name = "AAC decoder",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+AVCodec ff_aac_decoder = {
+ .name = "aac",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_AAC,
+ .priv_data_size = sizeof(AACContext),
+ .init = aac_decode_init,
+ .close = aac_decode_close,
+ .decode = aac_decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
+ .sample_fmts = (const enum AVSampleFormat[]) {
+ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
+ },
+ .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
+ .channel_layouts = aac_channel_layout,
+ .flush = flush,
+ .priv_class = &aac_decoder_class,
+};
+
+/*
+ Note: This decoder filter is intended to decode LATM streams transferred
+ in MPEG transport streams which only contain one program.
+ To do a more complex LATM demuxing a separate LATM demuxer should be used.
+*/
+AVCodec ff_aac_latm_decoder = {
+ .name = "aac_latm",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_AAC_LATM,
+ .priv_data_size = sizeof(struct LATMContext),
+ .init = latm_decode_init,
+ .close = aac_decode_close,
+ .decode = latm_decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
+ .sample_fmts = (const enum AVSampleFormat[]) {
+ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
+ },
+ .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
+ .channel_layouts = aac_channel_layout,
+ .flush = flush,
+};
generated by cgit v1.2.3 (git 2.39.1) at 2025-12-06 16:56:26 +0000