libavcodec: Implementation of AAC_fixed_decoder (LC-module) [1/4]
authorJovan Zelincevic <jovan.zelincevic@imgtec.com>
Tue, 30 Jun 2015 09:53:03 +0000 (11:53 +0200)
committerMichael Niedermayer <michaelni@gmx.at>
Thu, 9 Jul 2015 12:35:44 +0000 (14:35 +0200)
Move existing code to the new template files

Signed-off-by: Nedeljko Babic <nedeljko.babic@imgtec.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
libavcodec/aacdec.c
libavcodec/aacdec_template.c [new file with mode: 0644]
libavcodec/cbrt_tablegen.c
libavcodec/cbrt_tablegen_template.c [new file with mode: 0644]
libavcodec/sinewin_tablegen.c
libavcodec/sinewin_tablegen_template.c [new file with mode: 0644]

index 622cc5c..1d1abc9 100644 (file)
  * @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
- * Y                    Enhanced AAC Low Delay (ER AAC ELD)
- *
- * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
- *       - HE AAC v2 comprises LC AAC with Spectral Band Replication and
-           Parametric Stereo.
- */
-
 #include "libavutil/float_dsp.h"
 #include "libavutil/opt.h"
 #include "avcodec.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 (*channels >= MAX_CHANNELS)
-        return AVERROR_INVALIDDATA;
-    if (che_pos) {
-        if (!ac->che[type][id]) {
-            if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
-                return AVERROR(ENOMEM);
-            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);
-    if (!avctx->channels)
-        return 1;
-
-    ac->frame->nb_samples = 2048;
-    if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
-        return ret;
-
-    /* map output channel pointers to AVFrame data */
-    for (ch = 0; ch < avctx->channels; ch++) {
-        if (ac->output_element[ch])
-            ac->output_element[ch]->ret = (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;
-
-    if (num_side_channels == 0 && num_back_channels >= 4) {
-        num_side_channels = 2;
-        num_back_channels -= 2;
-    }
-
-    i = 0;
-    if (num_front_channels & 1) {
-        e2c_vec[i] = (struct elem_to_channel) {
-            .av_position  = AV_CH_FRONT_CENTER,
-            .syn_ele      = TYPE_SCE,
-            .elem_id      = layout_map[i][1],
-            .aac_position = AAC_CHANNEL_FRONT
-        };
-        i++;
-        num_front_channels--;
-    }
-    if (num_front_channels >= 4) {
-        i += assign_pair(e2c_vec, layout_map, i,
-                         AV_CH_FRONT_LEFT_OF_CENTER,
-                         AV_CH_FRONT_RIGHT_OF_CENTER,
-                         AAC_CHANNEL_FRONT);
-        num_front_channels -= 2;
-    }
-    if (num_front_channels >= 2) {
-        i += assign_pair(e2c_vec, layout_map, i,
-                         AV_CH_FRONT_LEFT,
-                         AV_CH_FRONT_RIGHT,
-                         AAC_CHANNEL_FRONT);
-        num_front_channels -= 2;
-    }
-    while (num_front_channels >= 2) {
-        i += assign_pair(e2c_vec, layout_map, i,
-                         UINT64_MAX,
-                         UINT64_MAX,
-                         AAC_CHANNEL_FRONT);
-        num_front_channels -= 2;
-    }
-
-    if (num_side_channels >= 2) {
-        i += assign_pair(e2c_vec, layout_map, i,
-                         AV_CH_SIDE_LEFT,
-                         AV_CH_SIDE_RIGHT,
-                         AAC_CHANNEL_FRONT);
-        num_side_channels -= 2;
-    }
-    while (num_side_channels >= 2) {
-        i += assign_pair(e2c_vec, layout_map, i,
-                         UINT64_MAX,
-                         UINT64_MAX,
-                         AAC_CHANNEL_SIDE);
-        num_side_channels -= 2;
-    }
-
-    while (num_back_channels >= 4) {
-        i += assign_pair(e2c_vec, layout_map, i,
-                         UINT64_MAX,
-                         UINT64_MAX,
-                         AAC_CHANNEL_BACK);
-        num_back_channels -= 2;
-    }
-    if (num_back_channels >= 2) {
-        i += assign_pair(e2c_vec, layout_map, i,
-                         AV_CH_BACK_LEFT,
-                         AV_CH_BACK_RIGHT,
-                         AAC_CHANNEL_BACK);
-        num_back_channels -= 2;
-    }
-    if (num_back_channels) {
-        e2c_vec[i] = (struct elem_to_channel) {
-            .av_position  = AV_CH_BACK_CENTER,
-            .syn_ele      = TYPE_SCE,
-            .elem_id      = layout_map[i][1],
-            .aac_position = AAC_CHANNEL_BACK
-        };
-        i++;
-        num_back_channels--;
-    }
-
-    if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
-        e2c_vec[i] = (struct elem_to_channel) {
-            .av_position  = AV_CH_LOW_FREQUENCY,
-            .syn_ele      = TYPE_LFE,
-            .elem_id      = layout_map[i][1],
-            .aac_position = AAC_CHANNEL_LFE
-        };
-        i++;
-    }
-    while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
-        e2c_vec[i] = (struct elem_to_channel) {
-            .av_position  = UINT64_MAX,
-            .syn_ele      = TYPE_LFE,
-            .elem_id      = layout_map[i][1],
-            .aac_position = AAC_CHANNEL_LFE
-        };
-        i++;
-    }
-
-    // Must choose a stable sort
-    total_non_cc_elements = n = i;
-    do {
-        int next_n = 0;
-        for (i = 1; i < n; i++)
-            if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
-                FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]);
-                next_n = i;
-            }
-        n = next_n;
-    } while (n > 0);
-
-    layout = 0;
-    for (i = 0; i < total_non_cc_elements; i++) {
-        layout_map[i][0] = e2c_vec[i].syn_ele;
-        layout_map[i][1] = e2c_vec[i].elem_id;
-        layout_map[i][2] = e2c_vec[i].aac_position;
-        if (e2c_vec[i].av_position != UINT64_MAX) {
-            layout |= e2c_vec[i].av_position;
-        }
-    }
-
-    return layout;
-}
-
-/**
- * Save current output configuration if and only if it has been locked.
- */
-static void push_output_configuration(AACContext *ac) {
-    if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) {
-        ac->oc[0] = ac->oc[1];
-    }
-    ac->oc[1].status = OC_NONE;
-}
-
-/**
- * Restore the previous output configuration if and only if the current
- * configuration is unlocked.
- */
-static void pop_output_configuration(AACContext *ac) {
-    if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
-        ac->oc[1] = ac->oc[0];
-        ac->avctx->channels = ac->oc[1].channels;
-        ac->avctx->channel_layout = ac->oc[1].channel_layout;
-        output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
-                         ac->oc[1].status, 0);
-    }
-}
-
-/**
- * Configure output channel order based on the current program
- * configuration element.
- *
- * @return  Returns error status. 0 - OK, !0 - error
- */
-static int output_configure(AACContext *ac,
-                            uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
-                            enum OCStatus oc_type, int get_new_frame)
-{
-    AVCodecContext *avctx = ac->avctx;
-    int i, channels = 0, ret;
-    uint64_t layout = 0;
-    uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }};
-    uint8_t type_counts[TYPE_END] = { 0 };
-
-    if (ac->oc[1].layout_map != layout_map) {
-        memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
-        ac->oc[1].layout_map_tags = tags;
-    }
-    for (i = 0; i < tags; i++) {
-        int type =         layout_map[i][0];
-        int id =           layout_map[i][1];
-        id_map[type][id] = type_counts[type]++;
-    }
-    // Try to sniff a reasonable channel order, otherwise output the
-    // channels in the order the PCE declared them.
-    if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE)
-        layout = sniff_channel_order(layout_map, tags);
-    for (i = 0; i < tags; i++) {
-        int type =     layout_map[i][0];
-        int id =       layout_map[i][1];
-        int iid =      id_map[type][id];
-        int position = layout_map[i][2];
-        // Allocate or free elements depending on if they are in the
-        // current program configuration.
-        ret = che_configure(ac, position, type, iid, &channels);
-        if (ret < 0)
-            return ret;
-        ac->tag_che_map[type][id] = ac->che[type][iid];
-    }
-    if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
-        if (layout == AV_CH_FRONT_CENTER) {
-            layout = AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT;
-        } else {
-            layout = 0;
-        }
-    }
-
-    if (layout) avctx->channel_layout = layout;
-                            ac->oc[1].channel_layout = layout;
-    avctx->channels       = ac->oc[1].channels       = channels;
-    ac->oc[1].status = oc_type;
-
-    if (get_new_frame) {
-        if ((ret = frame_configure_elements(ac->avctx)) < 0)
-            return ret;
-    }
-
-    return 0;
-}
-
-static void flush(AVCodecContext *avctx)
-{
-    AACContext *ac= avctx->priv_data;
-    int type, i, j;
-
-    for (type = 3; type >= 0; type--) {
-        for (i = 0; i < MAX_ELEM_ID; i++) {
-            ChannelElement *che = ac->che[type][i];
-            if (che) {
-                for (j = 0; j <= 1; j++) {
-                    memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved));
-                }
-            }
-        }
-    }
-}
-
-/**
- * Set up channel positions based on a default channel configuration
- * as specified in table 1.17.
- *
- * @return  Returns error status. 0 - OK, !0 - error
- */
-static int set_default_channel_config(AVCodecContext *avctx,
-                                      uint8_t (*layout_map)[3],
-                                      int *tags,
-                                      int channel_config)
-{
-    if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
-        channel_config > 12) {
-        av_log(avctx, AV_LOG_ERROR,
-               "invalid default channel configuration (%d)\n",
-               channel_config);
-        return AVERROR_INVALIDDATA;
-    }
-    *tags = tags_per_config[channel_config];
-    memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
-           *tags * sizeof(*layout_map));
-
-    /*
-     * AAC specification has 7.1(wide) as a default layout for 8-channel streams.
-     * However, at least Nero AAC encoder encodes 7.1 streams using the default
-     * channel config 7, mapping the side channels of the original audio stream
-     * to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD
-     * decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
-     * the incorrect streams as if they were correct (and as the encoder intended).
-     *
-     * As actual intended 7.1(wide) streams are very rare, default to assuming a
-     * 7.1 layout was intended.
-     */
-    if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
-        av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
-               " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
-               " according to the specification instead.\n", FF_COMPLIANCE_STRICT);
-        layout_map[2][2] = AAC_CHANNEL_SIDE;
-    }
-
-    return 0;
-}
-
-static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
-{
-    /* For PCE based channel configurations map the channels solely based
-     * on tags. */
-    if (!ac->oc[1].m4ac.chan_config) {
-        return ac->tag_che_map[type][elem_id];
-    }
-    // Allow single CPE stereo files to be signalled with mono configuration.
-    if (!ac->tags_mapped && type == TYPE_CPE &&
-        ac->oc[1].m4ac.chan_config == 1) {
-        uint8_t layout_map[MAX_ELEM_ID*4][3];
-        int layout_map_tags;
-        push_output_configuration(ac);
-
-        av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");
-
-        if (set_default_channel_config(ac->avctx, layout_map,
-                                       &layout_map_tags, 2) < 0)
-            return NULL;
-        if (output_configure(ac, layout_map, layout_map_tags,
-                             OC_TRIAL_FRAME, 1) < 0)
-            return NULL;
-
-        ac->oc[1].m4ac.chan_config = 2;
-        ac->oc[1].m4ac.ps = 0;
-    }
-    // And vice-versa
-    if (!ac->tags_mapped && type == TYPE_SCE &&
-        ac->oc[1].m4ac.chan_config == 2) {
-        uint8_t layout_map[MAX_ELEM_ID * 4][3];
-        int layout_map_tags;
-        push_output_configuration(ac);
-
-        av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");
-
-        if (set_default_channel_config(ac->avctx, layout_map,
-                                       &layout_map_tags, 1) < 0)
-            return NULL;
-        if (output_configure(ac, layout_map, layout_map_tags,
-                             OC_TRIAL_FRAME, 1) < 0)
-            return NULL;
-
-        ac->oc[1].m4ac.chan_config = 1;
-        if (ac->oc[1].m4ac.sbr)
-            ac->oc[1].m4ac.ps = -1;
-    }
-    /* For indexed channel configurations map the channels solely based
-     * on position. */
-    switch (ac->oc[1].m4ac.chan_config) {
-    case 12:
-    case 7:
-        if (ac->tags_mapped == 3 && type == TYPE_CPE) {
-            ac->tags_mapped++;
-            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
-        }
-    case 11:
-        if (ac->tags_mapped == 2 &&
-            ac->oc[1].m4ac.chan_config == 11 &&
-            type == TYPE_SCE) {
-            ac->tags_mapped++;
-            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
-        }
-    case 6:
-        /* Some streams incorrectly code 5.1 audio as
-         * SCE[0] CPE[0] CPE[1] SCE[1]
-         * instead of
-         * SCE[0] CPE[0] CPE[1] LFE[0].
-         * If we seem to have encountered such a stream, transfer
-         * the LFE[0] element to the SCE[1]'s mapping */
-        if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
-            if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) {
-                av_log(ac->avctx, AV_LOG_WARNING,
-                   "This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n",
-                   type == TYPE_SCE ? "SCE" : "LFE", elem_id);
-                ac->warned_remapping_once++;
-            }
-            ac->tags_mapped++;
-            return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
-        }
-    case 5:
-        if (ac->tags_mapped == 2 && type == TYPE_CPE) {
-            ac->tags_mapped++;
-            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
-        }
-    case 4:
-        /* Some streams incorrectly code 4.0 audio as
-         * SCE[0] CPE[0] LFE[0]
-         * instead of
-         * SCE[0] CPE[0] SCE[1].
-         * If we seem to have encountered such a stream, transfer
-         * the SCE[1] element to the LFE[0]'s mapping */
-        if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
-            if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) {
-                av_log(ac->avctx, AV_LOG_WARNING,
-                   "This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
-                   type == TYPE_SCE ? "SCE" : "LFE", elem_id);
-                ac->warned_remapping_once++;
-            }
-            ac->tags_mapped++;
-            return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1];
-        }
-        if (ac->tags_mapped == 2 &&
-            ac->oc[1].m4ac.chan_config == 4 &&
-            type == TYPE_SCE) {
-            ac->tags_mapped++;
-            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
-        }
-    case 3:
-    case 2:
-        if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
-            type == TYPE_CPE) {
-            ac->tags_mapped++;
-            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
-        } else if (ac->oc[1].m4ac.chan_config == 2) {
-            return NULL;
-        }
-    case 1:
-        if (!ac->tags_mapped && type == TYPE_SCE) {
-            ac->tags_mapped++;
-            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
-        }
-    default:
-        return NULL;
-    }
-}
-
-/**
- * Decode an array of 4 bit element IDs, optionally interleaved with a
- * stereo/mono switching bit.
- *
- * @param type speaker type/position for these channels
- */
-static void decode_channel_map(uint8_t layout_map[][3],
-                               enum ChannelPosition type,
-                               GetBitContext *gb, int n)
-{
-    while (n--) {
-        enum RawDataBlockType syn_ele;
-        switch (type) {
-        case AAC_CHANNEL_FRONT:
-        case AAC_CHANNEL_BACK:
-        case AAC_CHANNEL_SIDE:
-            syn_ele = get_bits1(gb);
-            break;
-        case AAC_CHANNEL_CC:
-            skip_bits1(gb);
-            syn_ele = TYPE_CCE;
-            break;
-        case AAC_CHANNEL_LFE:
-            syn_ele = TYPE_LFE;
-            break;
-        default:
-            // AAC_CHANNEL_OFF has no channel map
-            av_assert0(0);
-        }
-        layout_map[0][0] = syn_ele;
-        layout_map[0][1] = get_bits(gb, 4);
-        layout_map[0][2] = type;
-        layout_map++;
-    }
-}
-
-/**
- * Decode program configuration element; reference: table 4.2.
- *
- * @return  Returns error status. 0 - OK, !0 - error
- */
-static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
-                      uint8_t (*layout_map)[3],
-                      GetBitContext *gb)
-{
-    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
-    int sampling_index;
-    int comment_len;
-    int tags;
-
-    skip_bits(gb, 2);  // object_type
-
-    sampling_index = get_bits(gb, 4);
-    if (m4ac->sampling_index != sampling_index)
-        av_log(avctx, AV_LOG_WARNING,
-               "Sample rate index in program config element does not "
-               "match the sample rate index configured by the container.\n");
-
-    num_front       = get_bits(gb, 4);
-    num_side        = get_bits(gb, 4);
-    num_back        = get_bits(gb, 4);
-    num_lfe         = get_bits(gb, 2);
-    num_assoc_data  = get_bits(gb, 3);
-    num_cc          = get_bits(gb, 4);
-
-    if (get_bits1(gb))
-        skip_bits(gb, 4); // mono_mixdown_tag
-    if (get_bits1(gb))
-        skip_bits(gb, 4); // stereo_mixdown_tag
-
-    if (get_bits1(gb))
-        skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
-
-    if (get_bits_left(gb) < 4 * (num_front + num_side + num_back + num_lfe + num_assoc_data + num_cc)) {
-        av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
-        return -1;
-    }
-    decode_channel_map(layout_map       , AAC_CHANNEL_FRONT, gb, num_front);
-    tags = num_front;
-    decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE,  gb, num_side);
-    tags += num_side;
-    decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK,  gb, num_back);
-    tags += num_back;
-    decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE,   gb, num_lfe);
-    tags += num_lfe;
-
-    skip_bits_long(gb, 4 * num_assoc_data);
-
-    decode_channel_map(layout_map + tags, AAC_CHANNEL_CC,    gb, num_cc);
-    tags += num_cc;
-
-    align_get_bits(gb);
-
-    /* comment field, first byte is length */
-    comment_len = get_bits(gb, 8) * 8;
-    if (get_bits_left(gb) < comment_len) {
-        av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
-        return AVERROR_INVALIDDATA;
-    }
-    skip_bits_long(gb, comment_len);
-    return tags;
-}
-
-/**
- * Decode GA "General Audio" specific configuration; reference: table 4.1.
- *
- * @param   ac          pointer to AACContext, may be null
- * @param   avctx       pointer to AVCCodecContext, used for logging
- *
- * @return  Returns error status. 0 - OK, !0 - error
- */
-static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
-                                     GetBitContext *gb,
-                                     MPEG4AudioConfig *m4ac,
-                                     int channel_config)
-{
-    int extension_flag, ret, ep_config, res_flags;
-    uint8_t layout_map[MAX_ELEM_ID*4][3];
-    int tags = 0;
-
-    if (get_bits1(gb)) { // frameLengthFlag
-        avpriv_request_sample(avctx, "960/120 MDCT window");
-        return AVERROR_PATCHWELCOME;
-    }
-    m4ac->frame_length_short = 0;
-
-    if (get_bits1(gb))       // dependsOnCoreCoder
-        skip_bits(gb, 14);   // coreCoderDelay
-    extension_flag = get_bits1(gb);
-
-    if (m4ac->object_type == AOT_AAC_SCALABLE ||
-        m4ac->object_type == AOT_ER_AAC_SCALABLE)
-        skip_bits(gb, 3);     // layerNr
-
-    if (channel_config == 0) {
-        skip_bits(gb, 4);  // element_instance_tag
-        tags = decode_pce(avctx, m4ac, layout_map, gb);
-        if (tags < 0)
-            return tags;
-    } else {
-        if ((ret = set_default_channel_config(avctx, layout_map,
-                                              &tags, channel_config)))
-            return ret;
-    }
-
-    if (count_channels(layout_map, tags) > 1) {
-        m4ac->ps = 0;
-    } else if (m4ac->sbr == 1 && m4ac->ps == -1)
-        m4ac->ps = 1;
-
-    if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
-        return ret;
-
-    if (extension_flag) {
-        switch (m4ac->object_type) {
-        case AOT_ER_BSAC:
-            skip_bits(gb, 5);    // numOfSubFrame
-            skip_bits(gb, 11);   // layer_length
-            break;
-        case AOT_ER_AAC_LC:
-        case AOT_ER_AAC_LTP:
-        case AOT_ER_AAC_SCALABLE:
-        case AOT_ER_AAC_LD:
-            res_flags = get_bits(gb, 3);
-            if (res_flags) {
-                avpriv_report_missing_feature(avctx,
-                                              "AAC data resilience (flags %x)",
-                                              res_flags);
-                return AVERROR_PATCHWELCOME;
-            }
-            break;
-        }
-        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
-    }
-    switch (m4ac->object_type) {
-    case AOT_ER_AAC_LC:
-    case AOT_ER_AAC_LTP:
-    case AOT_ER_AAC_SCALABLE:
-    case AOT_ER_AAC_LD:
-        ep_config = get_bits(gb, 2);
-        if (ep_config) {
-            avpriv_report_missing_feature(avctx,
-                                          "epConfig %d", ep_config);
-            return AVERROR_PATCHWELCOME;
-        }
-    }
-    return 0;
-}
-
-static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
-                                     GetBitContext *gb,
-                                     MPEG4AudioConfig *m4ac,
-                                     int channel_config)
-{
-    int ret, ep_config, res_flags;
-    uint8_t layout_map[MAX_ELEM_ID*4][3];
-    int tags = 0;
-    const int ELDEXT_TERM = 0;
-
-    m4ac->ps  = 0;
-    m4ac->sbr = 0;
-
-    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(avctx, layout_map,
-                                          &tags, channel_config)))
-        return ret;
-
-    if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
-        return ret;
-
-    ep_config = get_bits(gb, 2);
-    if (ep_config) {
-        avpriv_report_missing_feature(avctx,
-                                      "epConfig %d", ep_config);
-        return AVERROR_PATCHWELCOME;
-    }
-    return 0;
-}
-
-/**
- * Decode audio specific configuration; reference: table 1.13.
- *
- * @param   ac          pointer to AACContext, may be null
- * @param   avctx       pointer to AVCCodecContext, used for logging
- * @param   m4ac        pointer to MPEG4AudioConfig, used for parsing
- * @param   data        pointer to buffer holding an audio specific config
- * @param   bit_size    size of audio specific config or data in bits
- * @param   sync_extension look for an appended sync extension
- *
- * @return  Returns error status or number of consumed bits. <0 - error
- */
-static int decode_audio_specific_config(AACContext *ac,
-                                        AVCodecContext *avctx,
-                                        MPEG4AudioConfig *m4ac,
-                                        const uint8_t *data, int bit_size,
-                                        int sync_extension)
-{
-    GetBitContext gb;
-    int i, ret;
-
-    ff_dlog(avctx, "audio specific config size %d\n", bit_size >> 3);
-    for (i = 0; i < bit_size >> 3; i++)
-        ff_dlog(avctx, "%02x ", data[i]);
-    ff_dlog(avctx, "\n");
-
-    if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
-        return ret;
-
-    if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size,
-                                          sync_extension)) < 0)
-        return AVERROR_INVALIDDATA;
-    if (m4ac->sampling_index > 12) {
-        av_log(avctx, AV_LOG_ERROR,
-               "invalid sampling rate index %d\n",
-               m4ac->sampling_index);
-        return AVERROR_INVALIDDATA;
-    }
-    if (m4ac->object_type == AOT_ER_AAC_LD &&
-        (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
-        av_log(avctx, AV_LOG_ERROR,
-               "invalid low delay sampling rate index %d\n",
-               m4ac->sampling_index);
-        return AVERROR_INVALIDDATA;
-    }
-
-    skip_bits_long(&gb, i);
-
-    switch (m4ac->object_type) {
-    case AOT_AAC_MAIN:
-    case AOT_AAC_LC:
-    case AOT_AAC_LTP:
-    case AOT_ER_AAC_LC:
-    case AOT_ER_AAC_LD:
-        if ((ret = decode_ga_specific_config(ac, avctx, &gb,
-                                            m4ac, m4ac->chan_config)) < 0)
-            return ret;
-        break;
-    case AOT_ER_AAC_ELD:
-        if ((ret = decode_eld_specific_config(ac, avctx, &gb,
-                                              m4ac, m4ac->chan_config)) < 0)
-            return ret;
-        break;
-    default:
-        avpriv_report_missing_feature(avctx,
-                                      "Audio object type %s%d",
-                                      m4ac->sbr == 1 ? "SBR+" : "",
-                                      m4ac->object_type);
-        return AVERROR(ENOSYS);
-    }
-
-    ff_dlog(avctx,
-            "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
-            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;
-    int ret;
-
-    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 ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
-                                                avctx->extradata,
-                                                avctx->extradata_size * 8,
-                                                1)) < 0)
-            return ret;
-    } else {
-        int sr, i;
-        uint8_t layout_map[MAX_ELEM_ID*4][3];
-        int layout_map_tags;
-
-        sr = sample_rate_idx(avctx->sample_rate);
-        ac->oc[1].m4ac.sampling_index = sr;
-        ac->oc[1].m4ac.channels = avctx->channels;
-        ac->oc[1].m4ac.sbr = -1;
-        ac->oc[1].m4ac.ps = -1;
-
-        for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
-            if (ff_mpeg4audio_channels[i] == avctx->channels)
-                break;
-        if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
-            i = 0;
-        }
-        ac->oc[1].m4ac.chan_config = i;
-
-        if (ac->oc[1].m4ac.chan_config) {
-            int ret = set_default_channel_config(avctx, layout_map,
-                &layout_map_tags, ac->oc[1].m4ac.chan_config);
-            if (!ret)
-                output_configure(ac, layout_map, layout_map_tags,
-                                 OC_GLOBAL_HDR, 0);
-            else if (avctx->err_recognition & AV_EF_EXPLODE)
-                return AVERROR_INVALIDDATA;
-        }
-    }
-
-    if (avctx->channels > MAX_CHANNELS) {
-        av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
-        return AVERROR_INVALIDDATA;
-    }
-
-    AAC_INIT_VLC_STATIC( 0, 304);
-    AAC_INIT_VLC_STATIC( 1, 270);
-    AAC_INIT_VLC_STATIC( 2, 550);
-    AAC_INIT_VLC_STATIC( 3, 300);
-    AAC_INIT_VLC_STATIC( 4, 328);
-    AAC_INIT_VLC_STATIC( 5, 294);
-    AAC_INIT_VLC_STATIC( 6, 306);
-    AAC_INIT_VLC_STATIC( 7, 268);
-    AAC_INIT_VLC_STATIC( 8, 510);
-    AAC_INIT_VLC_STATIC( 9, 366);
-    AAC_INIT_VLC_STATIC(10, 462);
-
-    ff_aac_sbr_init();
-
-    ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
-    if (!ac->fdsp) {
-        return AVERROR(ENOMEM);
-    }
-
-    ac->random_state = 0x1f2e3d4c;
-
-    ff_aac_tableinit();
-
-    INIT_VLC_STATIC(&vlc_scalefactors, 7,
-                    FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
-                    ff_aac_scalefactor_bits,
-                    sizeof(ff_aac_scalefactor_bits[0]),
-                    sizeof(ff_aac_scalefactor_bits[0]),
-                    ff_aac_scalefactor_code,
-                    sizeof(ff_aac_scalefactor_code[0]),
-                    sizeof(ff_aac_scalefactor_code[0]),
-                    352);
-
-    ff_mdct_init(&ac->mdct,       11, 1, 1.0 / (32768.0 * 1024.0));
-    ff_mdct_init(&ac->mdct_ld,    10, 1, 1.0 / (32768.0 * 512.0));
-    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);
-    ret = ff_imdct15_init(&ac->mdct480, 5);
-    if (ret < 0)
-        return ret;
-
-    // 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( 9);
-    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 AVERROR_INVALIDDATA;
-    }
-    skip_bits_long(gb, 8 * count);
-    return 0;
-}
-
-static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
-                             GetBitContext *gb)
-{
-    int sfb;
-    if (get_bits1(gb)) {
-        ics->predictor_reset_group = get_bits(gb, 5);
-        if (ics->predictor_reset_group == 0 ||
-            ics->predictor_reset_group > 30) {
-            av_log(ac->avctx, AV_LOG_ERROR,
-                   "Invalid Predictor Reset Group.\n");
-            return AVERROR_INVALIDDATA;
-        }
-    }
-    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
-        ics->prediction_used[sfb] = get_bits1(gb);
-    }
-    return 0;
-}
-
-/**
- * Decode Long Term Prediction data; reference: table 4.xx.
- */
-static void decode_ltp(LongTermPrediction *ltp,
-                       GetBitContext *gb, uint8_t max_sfb)
-{
-    int sfb;
-
-    ltp->lag  = get_bits(gb, 11);
-    ltp->coef = ltp_coef[get_bits(gb, 3)];
-    for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
-        ltp->used[sfb] = get_bits1(gb);
-}
-
-/**
- * Decode Individual Channel Stream info; reference: table 4.6.
- */
-static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
-                           GetBitContext *gb)
-{
-    const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
-    const int aot = m4ac->object_type;
-    const int sampling_index = m4ac->sampling_index;
-    if (aot != AOT_ER_AAC_ELD) {
-        if (get_bits1(gb)) {
-            av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
-            if (ac->avctx->err_recognition & AV_EF_BITSTREAM)
-                return AVERROR_INVALIDDATA;
-        }
-        ics->window_sequence[1] = ics->window_sequence[0];
-        ics->window_sequence[0] = get_bits(gb, 2);
-        if (aot == AOT_ER_AAC_LD &&
-            ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
-            av_log(ac->avctx, AV_LOG_ERROR,
-                   "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
-                   "window sequence %d found.\n", ics->window_sequence[0]);
-            ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
-            return AVERROR_INVALIDDATA;
-        }
-        ics->use_kb_window[1]   = ics->use_kb_window[0];
-        ics->use_kb_window[0]   = get_bits1(gb);
-    }
-    ics->num_window_groups  = 1;
-    ics->group_len[0]       = 1;
-    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
-        int i;
-        ics->max_sfb = get_bits(gb, 4);
-        for (i = 0; i < 7; i++) {
-            if (get_bits1(gb)) {
-                ics->group_len[ics->num_window_groups - 1]++;
-            } else {
-                ics->num_window_groups++;
-                ics->group_len[ics->num_window_groups - 1] = 1;
-            }
-        }
-        ics->num_windows       = 8;
-        ics->swb_offset        =    ff_swb_offset_128[sampling_index];
-        ics->num_swb           =   ff_aac_num_swb_128[sampling_index];
-        ics->tns_max_bands     = ff_tns_max_bands_128[sampling_index];
-        ics->predictor_present = 0;
-    } else {
-        ics->max_sfb           = get_bits(gb, 6);
-        ics->num_windows       = 1;
-        if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
-            if (m4ac->frame_length_short) {
-                ics->swb_offset    =     ff_swb_offset_480[sampling_index];
-                ics->num_swb       =    ff_aac_num_swb_480[sampling_index];
-                ics->tns_max_bands =  ff_tns_max_bands_480[sampling_index];
-            } else {
-                ics->swb_offset    =     ff_swb_offset_512[sampling_index];
-                ics->num_swb       =    ff_aac_num_swb_512[sampling_index];
-                ics->tns_max_bands =  ff_tns_max_bands_512[sampling_index];
-            }
-            if (!ics->num_swb || !ics->swb_offset)
-                return AVERROR_BUG;
-        } else {
-            ics->swb_offset    =    ff_swb_offset_1024[sampling_index];
-            ics->num_swb       =   ff_aac_num_swb_1024[sampling_index];
-            ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index];
-        }
-        if (aot != AOT_ER_AAC_ELD) {
-            ics->predictor_present     = get_bits1(gb);
-            ics->predictor_reset_group = 0;
-        }
-        if (ics->predictor_present) {
-            if (aot == AOT_AAC_MAIN) {
-                if (decode_prediction(ac, ics, gb)) {
-                    goto fail;
-                }
-            } else if (aot == AOT_AAC_LC ||
-                       aot == AOT_ER_AAC_LC) {
-                av_log(ac->avctx, AV_LOG_ERROR,
-                       "Prediction is not allowed in AAC-LC.\n");
-                goto fail;
-            } else {
-                if (aot == AOT_ER_AAC_LD) {
-                    av_log(ac->avctx, AV_LOG_ERROR,
-                           "LTP in ER AAC LD not yet implemented.\n");
-                    return AVERROR_PATCHWELCOME;
-                }
-                if ((ics->ltp.present = get_bits(gb, 1)))
-                    decode_ltp(&ics->ltp, gb, ics->max_sfb);
-            }
-        }
-    }
-
-    if (ics->max_sfb > ics->num_swb) {
-        av_log(ac->avctx, AV_LOG_ERROR,
-               "Number of scalefactor bands in group (%d) "
-               "exceeds limit (%d).\n",
-               ics->max_sfb, ics->num_swb);
-        goto fail;
-    }
-
-    return 0;
-fail:
-    ics->max_sfb = 0;
-    return AVERROR_INVALIDDATA;
-}
-
-/**
- * Decode band types (section_data payload); reference: table 4.46.
- *
- * @param   band_type           array of the used band type
- * @param   band_type_run_end   array of the last scalefactor band of a band type run
- *
- * @return  Returns error status. 0 - OK, !0 - error
- */
-static int decode_band_types(AACContext *ac, enum BandType band_type[120],
-                             int band_type_run_end[120], GetBitContext *gb,
-                             IndividualChannelStream *ics)
-{
-    int g, idx = 0;
-    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
-    for (g = 0; g < ics->num_window_groups; g++) {
-        int k = 0;
-        while (k < ics->max_sfb) {
-            uint8_t sect_end = k;
-            int sect_len_incr;
-            int sect_band_type = get_bits(gb, 4);
-            if (sect_band_type == 12) {
-                av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
-                return AVERROR_INVALIDDATA;
-            }
-            do {
-                sect_len_incr = get_bits(gb, bits);
-                sect_end += sect_len_incr;
-                if (get_bits_left(gb) < 0) {
-                    av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
-                    return AVERROR_INVALIDDATA;
-                }
-                if (sect_end > ics->max_sfb) {
-                    av_log(ac->avctx, AV_LOG_ERROR,
-                           "Number of bands (%d) exceeds limit (%d).\n",
-                           sect_end, ics->max_sfb);
-                    return AVERROR_INVALIDDATA;
-                }
-            } while (sect_len_incr == (1 << bits) - 1);
-            for (; k < sect_end; k++) {
-                band_type        [idx]   = sect_band_type;
-                band_type_run_end[idx++] = sect_end;
-            }
-        }
-    }
-    return 0;
-}
-
-/**
- * Decode scalefactors; reference: table 4.47.
- *
- * @param   global_gain         first scalefactor value as scalefactors are differentially coded
- * @param   band_type           array of the used band type
- * @param   band_type_run_end   array of the last scalefactor band of a band type run
- * @param   sf                  array of scalefactors or intensity stereo positions
- *
- * @return  Returns error status. 0 - OK, !0 - error
- */
-static int decode_scalefactors(AACContext *ac, 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 - NOISE_OFFSET, 0 };
-    int clipped_offset;
-    int noise_flag = 1;
-    for (g = 0; g < ics->num_window_groups; g++) {
-        for (i = 0; i < ics->max_sfb;) {
-            int run_end = band_type_run_end[idx];
-            if (band_type[idx] == ZERO_BT) {
-                for (; i < run_end; i++, idx++)
-                    sf[idx] = 0.0;
-            } else if ((band_type[idx] == INTENSITY_BT) ||
-                       (band_type[idx] == INTENSITY_BT2)) {
-                for (; i < run_end; i++, idx++) {
-                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
-                    clipped_offset = av_clip(offset[2], -155, 100);
-                    if (offset[2] != clipped_offset) {
-                        avpriv_request_sample(ac->avctx,
-                                              "If you heard an audible artifact, there may be a bug in the decoder. "
-                                              "Clipped intensity stereo position (%d -> %d)",
-                                              offset[2], clipped_offset);
-                    }
-                    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, NOISE_PRE_BITS) - NOISE_PRE;
-                    else
-                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
-                    clipped_offset = av_clip(offset[1], -100, 155);
-                    if (offset[1] != clipped_offset) {
-                        avpriv_request_sample(ac->avctx,
-                                              "If you heard an audible artifact, there may be a bug in the decoder. "
-                                              "Clipped noise gain (%d -> %d)",
-                                              offset[1], clipped_offset);
-                    }
-                    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) - 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;
-                    }
-                    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] >= swb_offset[num_swb])
-        return -1;
-    pulse->amp[0]    = get_bits(gb, 4);
-    for (i = 1; i < pulse->num_pulse; i++) {
-        pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
-        if (pulse->pos[i] >= swb_offset[num_swb])
-            return -1;
-        pulse->amp[i] = get_bits(gb, 4);
-    }
-    return 0;
-}
-
-/**
- * Decode Temporal Noise Shaping data; reference: table 4.48.
- *
- * @return  Returns error status. 0 - OK, !0 - error
- */
-static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
-                      GetBitContext *gb, const IndividualChannelStream *ics)
-{
-    int w, filt, i, coef_len, coef_res, coef_compress;
-    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
-    const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
-    for (w = 0; w < ics->num_windows; w++) {
-        if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
-            coef_res = get_bits1(gb);
-
-            for (filt = 0; filt < tns->n_filt[w]; filt++) {
-                int tmp2_idx;
-                tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
-
-                if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
-                    av_log(ac->avctx, AV_LOG_ERROR,
-                           "TNS filter order %d is greater than maximum %d.\n",
-                           tns->order[w][filt], tns_max_order);
-                    tns->order[w][filt] = 0;
-                    return AVERROR_INVALIDDATA;
-                }
-                if (tns->order[w][filt]) {
-                    tns->direction[w][filt] = get_bits1(gb);
-                    coef_compress = get_bits1(gb);
-                    coef_len = coef_res + 3 - coef_compress;
-                    tmp2_idx = 2 * coef_compress + coef_res;
-
-                    for (i = 0; i < tns->order[w][filt]; i++)
-                        tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
-                }
-            }
-        }
-    }
-    return 0;
-}
+#   include "arm/aac.h"
+#elif ARCH_MIPS
+#   include "mips/aacdec_mips.h"
+#endif
 
-/**
- * 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)
+static av_always_inline void reset_predict_state(PredictorState *ps)
 {
-    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]));
-    }
+    ps->r0   = 0.0f;
+    ps->r1   = 0.0f;
+    ps->cor0 = 0.0f;
+    ps->cor1 = 0.0f;
+    ps->var0 = 1.0f;
+    ps->var1 = 1.0f;
 }
 
 #ifndef VMUL2
@@ -1611,1062 +131,70 @@ static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
     *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 AVERROR_INVALIDDATA;
-                                    }
-
-                                    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, eld_syntax, er_syntax, pulse_present = 0;
-    int ret;
-
-    eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
-    er_syntax  = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
-                 ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
-                 ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
-                 ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
-
-    /* This assignment is to silence a GCC warning about the variable being used
-     * uninitialized when in fact it always is.
-     */
-    pulse.num_pulse = 0;
-
-    global_gain = get_bits(gb, 8);
-
-    if (!common_window && !scale_flag) {
-        if (decode_ics_info(ac, ics, gb) < 0)
-            return AVERROR_INVALIDDATA;
-    }
-
-    if ((ret = decode_band_types(ac, sce->band_type,
-                                 sce->band_type_run_end, gb, ics)) < 0)
-        return ret;
-    if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
-                                  sce->band_type, sce->band_type_run_end)) < 0)
-        return ret;
-
-    pulse_present = 0;
-    if (!scale_flag) {
-        if (!eld_syntax && (pulse_present = get_bits1(gb))) {
-            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
-                av_log(ac->avctx, AV_LOG_ERROR,
-                       "Pulse tool not allowed in eight short sequence.\n");
-                return AVERROR_INVALIDDATA;
-            }
-            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
-                av_log(ac->avctx, AV_LOG_ERROR,
-                       "Pulse data corrupt or invalid.\n");
-                return AVERROR_INVALIDDATA;
-            }
-        }
-        tns->present = get_bits1(gb);
-        if (tns->present && !er_syntax)
-            if (decode_tns(ac, tns, gb, ics) < 0)
-                return AVERROR_INVALIDDATA;
-        if (!eld_syntax && get_bits1(gb)) {
-            avpriv_request_sample(ac->avctx, "SSR");
-            return AVERROR_PATCHWELCOME;
-        }
-        // I see no textual basis in the spec for this occurring after SSR gain
-        // control, but this is what both reference and real implmentations do
-        if (tns->present && er_syntax)
-            if (decode_tns(ac, tns, gb, ics) < 0)
-                return AVERROR_INVALIDDATA;
-    }
-
-    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
-                                    &pulse, ics, sce->band_type) < 0)
-        return AVERROR_INVALIDDATA;
-
-    if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
-        apply_prediction(ac, sce);
-
-    return 0;
-}
-
-/**
- * Mid/Side stereo decoding; reference: 4.6.8.1.3.
- */
-static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
-{
-    const IndividualChannelStream *ics = &cpe->ch[0].ics;
-    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;
-    int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
-
-    common_window = eld_syntax || get_bits1(gb);
-    if (common_window) {
-        if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
-            return AVERROR_INVALIDDATA;
-        i = cpe->ch[1].ics.use_kb_window[0];
-        cpe->ch[1].ics = cpe->ch[0].ics;
-        cpe->ch[1].ics.use_kb_window[1] = i;
-        if (cpe->ch[1].ics.predictor_present &&
-            (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
-            if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
-                decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
-        ms_present = get_bits(gb, 2);
-        if (ms_present == 3) {
-            av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
-            return AVERROR_INVALIDDATA;
-        } else if (ms_present)
-            decode_mid_side_stereo(cpe, gb, ms_present);
-    }
-    if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
-        return ret;
-    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
-        return ret;
-
-    if (common_window) {
-        if (ms_present)
-            apply_mid_side_stereo(ac, cpe);
-        if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
-            apply_prediction(ac, &cpe->ch[0]);
-            apply_prediction(ac, &cpe->ch[1]);
-        }
-    }
-
-    apply_intensity_stereo(ac, cpe, ms_present);
-    return 0;
-}
-
-static const float cce_scale[] = {
-    1.09050773266525765921, //2^(1/8)
-    1.18920711500272106672, //2^(1/4)
-    M_SQRT2,
-    2,
-};
-
-/**
- * Decode coupling_channel_element; reference: table 4.8.
- *
- * @return  Returns error status. 0 - OK, !0 - error
- */
-static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
-{
-    int num_gain = 0;
-    int c, g, sfb, ret;
-    int sign;
-    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 = 1.0;
-        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;
-    int type = get_bits(gb, 4);
-
-    if (ac->avctx->debug & FF_DEBUG_STARTCODE)
-        av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
-
-    switch (type) { // extension type
-    case EXT_SBR_DATA_CRC:
-        crc_flag++;
-    case EXT_SBR_DATA:
-        if (!che) {
-            av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
-            return res;
-        } else if (!ac->oc[1].m4ac.sbr) {
-            av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
-            skip_bits_long(gb, 8 * cnt - 4);
-            return res;
-        } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
-            av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
-            skip_bits_long(gb, 8 * cnt - 4);
-            return res;
-        } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
-            ac->oc[1].m4ac.sbr = 1;
-            ac->oc[1].m4ac.ps = 1;
-            ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
-            output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
-                             ac->oc[1].status, 1);
-        } else {
-            ac->oc[1].m4ac.sbr = 1;
-            ac->avctx->profile = FF_PROFILE_AAC_HE;
-        }
-        res = 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;
-}
+    t.i = s.i ^ (sign & 1U<<31);
+    *dst++ = v[idx>>4 & 3] * t.f;
 
-/**
- * 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;
+    sign <<= nz & 1;
+    t.i = s.i ^ (sign & 1U<<31);
+    *dst++ = v[idx>>6 & 3] * t.f;
 
-            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];
-                }
-            }
-        }
-    }
+    return dst;
 }
+#endif
 
-/**
- *  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)
+static av_always_inline float flt16_round(float pf)
 {
-    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);
+    union av_intfloat32 tmp;
+    tmp.f = pf;
+    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
+    return tmp.f;
 }
 
-/**
- * Apply the long term prediction
- */
-static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
+static av_always_inline float flt16_even(float pf)
 {
-    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];
-    }
+    union av_intfloat32 tmp;
+    tmp.f = pf;
+    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
+    return tmp.f;
 }
 
-/**
- * Update the LTP buffer for next frame
- */
-static void update_ltp(AACContext *ac, SingleChannelElement *sce)
+static av_always_inline float flt16_trunc(float pf)
 {
-    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));
+    union av_intfloat32 pun;
+    pun.f = pf;
+    pun.i &= 0xFFFF0000U;
+    return pun.f;
 }
 
-/**
- * Conduct IMDCT and windowing.
- */
-static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
+static av_always_inline void predict(PredictorState *ps, float *coef,
+                                     int output_enable)
 {
-    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));
-        }
-    }
+    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;
 
-    // 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));
-    }
-}
+    k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
+    k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
 
-static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
-{
-    IndividualChannelStream *ics = &sce->ics;
-    float *in    = sce->coeffs;
-    float *out   = sce->ret;
-    float *saved = sce->saved;
-    float *buf  = ac->buf_mdct;
-
-    // imdct
-    ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
-
-    // window overlapping
-    if (ics->use_kb_window[1]) {
-        // AAC LD uses a low overlap sine window instead of a KBD window
-        memcpy(out, saved, 192 * sizeof(float));
-        ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, ff_sine_128, 64);
-        memcpy(                     out + 320, buf + 64, 192 * sizeof(float));
-    } else {
-        ac->fdsp->vector_fmul_window(out, saved, buf, ff_sine_512, 256);
-    }
+    pv = flt16_round(k1 * r0 + k2 * r1);
+    if (output_enable)
+        *coef += pv;
 
-    // buffer update
-    memcpy(saved, buf + 256, 256 * sizeof(float));
-}
+    e0 = *coef;
+    e1 = e0 - k1 * r0;
 
-static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
-{
-    float *in    = sce->coeffs;
-    float *out   = sce->ret;
-    float *saved = sce->saved;
-    float *buf  = ac->buf_mdct;
-    int i;
-    const int n  = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
-    const int n2 = n >> 1;
-    const int n4 = n >> 2;
-    const float *const window = n == 480 ? ff_aac_eld_window_480 :
-                                           ff_aac_eld_window_512;
-
-    // Inverse transform, mapped to the conventional IMDCT by
-    // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
-    // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
-    // International Conference on Audio, Language and Image Processing, ICALIP 2008.
-    // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
-    for (i = 0; i < n2; i+=2) {
-        float temp;
-        temp =  in[i    ]; in[i    ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
-        temp = -in[i + 1]; in[i + 1] =  in[n - 2 - i]; in[n - 2 - i] = temp;
-    }
-    if (n == 480)
-        ac->mdct480->imdct_half(ac->mdct480, buf, in, 1, -1.f/(16*1024*960));
-    else
-        ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
-    for (i = 0; i < n; i+=2) {
-        buf[i] = -buf[i];
-    }
-    // Like with the regular IMDCT at this point we still have the middle half
-    // of a transform but with even symmetry on the left and odd symmetry on
-    // the right
-
-    // window overlapping
-    // The spec says to use samples [0..511] but the reference decoder uses
-    // samples [128..639].
-    for (i = n4; i < n2; i ++) {
-        out[i - n4] =    buf[n2 - 1 - i]       * window[i       - n4] +
-                       saved[      i + n2]     * window[i +   n - n4] +
-                      -saved[  n + n2 - 1 - i] * window[i + 2*n - n4] +
-                      -saved[2*n + n2 + i]     * window[i + 3*n - n4];
-    }
-    for (i = 0; i < n2; i ++) {
-        out[n4 + i] =    buf[i]               * window[i + n2       - n4] +
-                      -saved[      n - 1 - i] * window[i + n2 +   n - n4] +
-                      -saved[  n + i]         * window[i + n2 + 2*n - n4] +
-                       saved[2*n + n - 1 - i] * window[i + n2 + 3*n - n4];
-    }
-    for (i = 0; i < n4; i ++) {
-        out[n2 + n4 + i] =    buf[      i + n2]     * window[i +   n - n4] +
-                           -saved[      n2 - 1 - i] * window[i + 2*n - n4] +
-                           -saved[  n + n2 + i]     * window[i + 3*n - n4];
-    }
+    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));
 
-    // buffer update
-    memmove(saved + n, saved, 2 * n * sizeof(float));
-    memcpy( saved,       buf,     n * sizeof(float));
+    ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
+    ps->r0 = flt16_trunc(a * e0);
 }
 
 /**
@@ -2724,506 +252,7 @@ static void apply_independent_coupling(AACContext *ac,
         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;
-    void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
-    switch (ac->oc[1].m4ac.object_type) {
-    case AOT_ER_AAC_LD:
-        imdct_and_window = imdct_and_windowing_ld;
-        break;
-    case AOT_ER_AAC_ELD:
-        imdct_and_window = imdct_and_windowing_eld;
-        break;
-    default:
-        imdct_and_window = ac->imdct_and_windowing;
-    }
-    for (type = 3; type >= 0; type--) {
-        for (i = 0; i < MAX_ELEM_ID; i++) {
-            ChannelElement *che = ac->che[type][i];
-            if (che && che->present) {
-                if (type <= TYPE_CPE)
-                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, 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) {
-                    imdct_and_window(ac, &che->ch[0]);
-                    if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
-                        ac->update_ltp(ac, &che->ch[0]);
-                    if (type == TYPE_CPE) {
-                        imdct_and_window(ac, &che->ch[1]);
-                        if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
-                            ac->update_ltp(ac, &che->ch[1]);
-                    }
-                    if (ac->oc[1].m4ac.sbr > 0) {
-                        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);
-                che->present = 0;
-            } else if (che) {
-                av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
-            }
-        }
-    }
-}
-
-static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
-{
-    int size;
-    AACADTSHeaderInfo hdr_info;
-    uint8_t layout_map[MAX_ELEM_ID*4][3];
-    int layout_map_tags, ret;
-
-    size = avpriv_aac_parse_header(gb, &hdr_info);
-    if (size > 0) {
-        if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
-            // This is 2 for "VLB " audio in NSV files.
-            // See samples/nsv/vlb_audio.
-            avpriv_report_missing_feature(ac->avctx,
-                                          "More than one AAC RDB per ADTS frame");
-            ac->warned_num_aac_frames = 1;
-        }
-        push_output_configuration(ac);
-        if (hdr_info.chan_config) {
-            ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
-            if ((ret = set_default_channel_config(ac->avctx,
-                                                  layout_map,
-                                                  &layout_map_tags,
-                                                  hdr_info.chan_config)) < 0)
-                return ret;
-            if ((ret = output_configure(ac, layout_map, layout_map_tags,
-                                        FFMAX(ac->oc[1].status,
-                                              OC_TRIAL_FRAME), 0)) < 0)
-                return ret;
-        } else {
-            ac->oc[1].m4ac.chan_config = 0;
-            /**
-             * dual mono frames in Japanese DTV can have chan_config 0
-             * WITHOUT specifying PCE.
-             *  thus, set dual mono as default.
-             */
-            if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {
-                layout_map_tags = 2;
-                layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
-                layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
-                layout_map[0][1] = 0;
-                layout_map[1][1] = 1;
-                if (output_configure(ac, layout_map, layout_map_tags,
-                                     OC_TRIAL_FRAME, 0))
-                    return -7;
-            }
-        }
-        ac->oc[1].m4ac.sample_rate     = hdr_info.sample_rate;
-        ac->oc[1].m4ac.sampling_index  = hdr_info.sampling_index;
-        ac->oc[1].m4ac.object_type     = hdr_info.object_type;
-        ac->oc[1].m4ac.frame_length_short = 0;
-        if (ac->oc[0].status != OC_LOCKED ||
-            ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
-            ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
-            ac->oc[1].m4ac.sbr = -1;
-            ac->oc[1].m4ac.ps  = -1;
-        }
-        if (!hdr_info.crc_absent)
-            skip_bits(gb, 16);
-    }
-    return size;
-}
-
-static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
-                               int *got_frame_ptr, GetBitContext *gb)
-{
-    AACContext *ac = avctx->priv_data;
-    const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
-    ChannelElement *che;
-    int err, i;
-    int samples = m4ac->frame_length_short ? 960 : 1024;
-    int chan_config = m4ac->chan_config;
-    int aot = m4ac->object_type;
-
-    if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
-        samples >>= 1;
-
-    ac->frame = data;
-
-    if ((err = frame_configure_elements(avctx)) < 0)
-        return err;
-
-    // The FF_PROFILE_AAC_* defines are all object_type - 1
-    // This may lead to an undefined profile being signaled
-    ac->avctx->profile = aot - 1;
-
-    ac->tags_mapped = 0;
-
-    if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
-        avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
-                              chan_config);
-        return AVERROR_INVALIDDATA;
-    }
-    for (i = 0; i < tags_per_config[chan_config]; i++) {
-        const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
-        const int elem_id   = aac_channel_layout_map[chan_config-1][i][1];
-        if (!(che=get_che(ac, elem_type, elem_id))) {
-            av_log(ac->avctx, AV_LOG_ERROR,
-                   "channel element %d.%d is not allocated\n",
-                   elem_type, elem_id);
-            return AVERROR_INVALIDDATA;
-        }
-        che->present = 1;
-        if (aot != AOT_ER_AAC_ELD)
-            skip_bits(gb, 4);
-        switch (elem_type) {
-        case TYPE_SCE:
-            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
-            break;
-        case TYPE_CPE:
-            err = decode_cpe(ac, gb, che);
-            break;
-        case TYPE_LFE:
-            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
-            break;
-        }
-        if (err < 0)
-            return err;
-    }
-
-    spectral_to_sample(ac);
-
-    ac->frame->nb_samples = samples;
-    ac->frame->sample_rate = avctx->sample_rate;
-    *got_frame_ptr = 1;
-
-    skip_bits_long(gb, get_bits_left(gb));
-    return 0;
-}
-
-static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
-                                int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
-{
-    AACContext *ac = avctx->priv_data;
-    ChannelElement *che = NULL, *che_prev = NULL;
-    enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
-    int err, elem_id;
-    int samples = 0, multiplier, audio_found = 0, pce_found = 0;
-    int is_dmono, sce_count = 0;
-
-    ac->frame = data;
-
-    if (show_bits(gb, 12) == 0xfff) {
-        if ((err = parse_adts_frame_header(ac, gb)) < 0) {
-            av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
-            goto fail;
-        }
-        if (ac->oc[1].m4ac.sampling_index > 12) {
-            av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
-            err = AVERROR_INVALIDDATA;
-            goto fail;
-        }
-    }
-
-    if ((err = frame_configure_elements(avctx)) < 0)
-        goto fail;
-
-    // The FF_PROFILE_AAC_* defines are all object_type - 1
-    // This may lead to an undefined profile being signaled
-    ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
-
-    ac->tags_mapped = 0;
-    // parse
-    while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
-        elem_id = get_bits(gb, 4);
-
-        if (avctx->debug & FF_DEBUG_STARTCODE)
-            av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
-
-        if (!avctx->channels && elem_type != TYPE_PCE) {
-            err = AVERROR_INVALIDDATA;
-            goto fail;
-        }
-
-        if (elem_type < TYPE_DSE) {
-            if (!(che=get_che(ac, elem_type, elem_id))) {
-                av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
-                       elem_type, elem_id);
-                err = AVERROR_INVALIDDATA;
-                goto fail;
-            }
-            samples = 1024;
-            che->present = 1;
-        }
-
-        switch (elem_type) {
-
-        case TYPE_SCE:
-            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
-            audio_found = 1;
-            sce_count++;
-            break;
-
-        case TYPE_CPE:
-            err = decode_cpe(ac, gb, che);
-            audio_found = 1;
-            break;
-
-        case TYPE_CCE:
-            err = decode_cce(ac, gb, che);
-            break;
-
-        case TYPE_LFE:
-            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
-            audio_found = 1;
-            break;
-
-        case TYPE_DSE:
-            err = skip_data_stream_element(ac, gb);
-            break;
-
-        case TYPE_PCE: {
-            uint8_t layout_map[MAX_ELEM_ID*4][3];
-            int tags;
-            push_output_configuration(ac);
-            tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb);
-            if (tags < 0) {
-                err = tags;
-                break;
-            }
-            if (pce_found) {
-                av_log(avctx, AV_LOG_ERROR,
-                       "Not evaluating a further program_config_element as this construct is dubious at best.\n");
-            } else {
-                err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
-                if (!err)
-                    ac->oc[1].m4ac.chan_config = 0;
-                pce_found = 1;
-            }
-            break;
-        }
-
-        case TYPE_FIL:
-            if (elem_id == 15)
-                elem_id += get_bits(gb, 8) - 1;
-            if (get_bits_left(gb) < 8 * elem_id) {
-                    av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
-                    err = AVERROR_INVALIDDATA;
-                    goto fail;
-            }
-            while (elem_id > 0)
-                elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
-            err = 0; /* FIXME */
-            break;
-
-        default:
-            err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
-            break;
-        }
-
-        che_prev       = che;
-        elem_type_prev = elem_type;
-
-        if (err)
-            goto fail;
-
-        if (get_bits_left(gb) < 3) {
-            av_log(avctx, AV_LOG_ERROR, overread_err);
-            err = AVERROR_INVALIDDATA;
-            goto fail;
-        }
-    }
-
-    if (!avctx->channels) {
-        *got_frame_ptr = 0;
-        return 0;
-    }
-
-    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;
-
-    if (ac->oc[1].status && audio_found) {
-        avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
-        avctx->frame_size = samples;
-        ac->oc[1].status = OC_LOCKED;
-    }
-
-    if (multiplier) {
-        int side_size;
-        const uint8_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);
-        if (side && side_size>=4)
-            AV_WL32(side, 2*AV_RL32(side));
-    }
-
-    if (!ac->frame->data[0] && samples) {
-        av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
-        err = AVERROR_INVALIDDATA;
-        goto fail;
-    }
-
-    if (samples) {
-        ac->frame->nb_samples = samples;
-        ac->frame->sample_rate = avctx->sample_rate;
-    } else
-        av_frame_unref(ac->frame);
-    *got_frame_ptr = !!samples;
-
-    /* for dual-mono audio (SCE + SCE) */
-    is_dmono = ac->dmono_mode && sce_count == 2 &&
-               ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
-    if (is_dmono) {
-        if (ac->dmono_mode == 1)
-            ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
-        else if (ac->dmono_mode == 2)
-            ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
-    }
-
-    return 0;
-fail:
-    pop_output_configuration(ac);
-    return err;
-}
-
-static int aac_decode_frame(AVCodecContext *avctx, void *data,
-                            int *got_frame_ptr, AVPacket *avpkt)
-{
-    AACContext *ac = avctx->priv_data;
-    const uint8_t *buf = avpkt->data;
-    int buf_size = avpkt->size;
-    GetBitContext gb;
-    int buf_consumed;
-    int buf_offset;
-    int err;
-    int new_extradata_size;
-    const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
-                                       AV_PKT_DATA_NEW_EXTRADATA,
-                                       &new_extradata_size);
-    int jp_dualmono_size;
-    const uint8_t *jp_dualmono   = av_packet_get_side_data(avpkt,
-                                       AV_PKT_DATA_JP_DUALMONO,
-                                       &jp_dualmono_size);
-
-    if (new_extradata && 0) {
-        av_free(avctx->extradata);
-        avctx->extradata = av_mallocz(new_extradata_size +
-                                      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;
-
-    if ((err = init_get_bits(&gb, buf, buf_size * 8)) < 0)
-        return err;
-
-    switch (ac->oc[1].m4ac.object_type) {
-    case AOT_ER_AAC_LC:
-    case AOT_ER_AAC_LTP:
-    case AOT_ER_AAC_LD:
-    case AOT_ER_AAC_ELD:
-        err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
-        break;
-    default:
-        err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
-    }
-    if (err < 0)
-        return err;
-
-    buf_consumed = (get_bits_count(&gb) + 7) >> 3;
-    for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
-        if (buf[buf_offset])
-            break;
-
-    return buf_size > buf_offset ? buf_consumed : buf_size;
-}
-
-static av_cold int aac_decode_close(AVCodecContext *avctx)
-{
-    AACContext *ac = avctx->priv_data;
-    int i, type;
-
-    for (i = 0; i < MAX_ELEM_ID; i++) {
-        for (type = 0; type < 4; type++) {
-            if (ac->che[type][i])
-                ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
-            av_freep(&ac->che[type][i]);
-        }
-    }
-
-    ff_mdct_end(&ac->mdct);
-    ff_mdct_end(&ac->mdct_small);
-    ff_mdct_end(&ac->mdct_ld);
-    ff_mdct_end(&ac->mdct_ltp);
-    ff_imdct15_uninit(&ac->mdct480);
-    av_freep(&ac->fdsp);
-    return 0;
-}
-
+#include "aacdec_template.c"
 
 #define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
 
@@ -3505,53 +534,6 @@ static av_cold int latm_decode_init(AVCodecContext *avctx)
     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,
-};
-
-static const AVProfile profiles[] = {
-    { FF_PROFILE_AAC_MAIN,  "Main"     },
-    { FF_PROFILE_AAC_LOW,   "LC"       },
-    { FF_PROFILE_AAC_SSR,   "SSR"      },
-    { FF_PROFILE_AAC_LTP,   "LTP"      },
-    { FF_PROFILE_AAC_HE,    "HE-AAC"   },
-    { FF_PROFILE_AAC_HE_V2, "HE-AACv2" },
-    { FF_PROFILE_AAC_LD,    "LD"       },
-    { FF_PROFILE_AAC_ELD,   "ELD"      },
-    { FF_PROFILE_UNKNOWN },
-};
-
 AVCodec ff_aac_decoder = {
     .name            = "aac",
     .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
diff --git a/libavcodec/aacdec_template.c b/libavcodec/aacdec_template.c
new file mode 100644 (file)
index 0000000..1b2b2fc
--- /dev/null
@@ -0,0 +1,3052 @@
+/*
+ * AAC decoder
+ * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
+ * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+ * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
+ *
+ * AAC LATM decoder
+ * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
+ * Copyright (c) 2010      Janne Grunau <janne-libav@jannau.net>
+ *
+ * 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
+ * Y  (not in fixed point code) Enhanced AAC Low Delay (ER AAC ELD)
+ *
+ * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
+ *       - HE AAC v2 comprises LC AAC with Spectral Band Replication and
+           Parametric Stereo.
+ */
+
+static VLC vlc_scalefactors;
+static VLC vlc_spectral[11];
+
+static int output_configure(AACContext *ac,
+                            uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
+                            enum OCStatus oc_type, int get_new_frame);
+
+#define overread_err "Input buffer exhausted before END element found\n"
+
+static int count_channels(uint8_t (*layout)[3], int tags)
+{
+    int i, sum = 0;
+    for (i = 0; i < tags; i++) {
+        int syn_ele = layout[i][0];
+        int pos     = layout[i][2];
+        sum += (1 + (syn_ele == TYPE_CPE)) *
+               (pos != AAC_CHANNEL_OFF && pos != AAC_CHANNEL_CC);
+    }
+    return sum;
+}
+
+/**
+ * Check for the channel element in the current channel position configuration.
+ * If it exists, make sure the appropriate element is allocated and map the
+ * channel order to match the internal FFmpeg channel layout.
+ *
+ * @param   che_pos current channel position configuration
+ * @param   type channel element type
+ * @param   id channel element id
+ * @param   channels count of the number of channels in the configuration
+ *
+ * @return  Returns error status. 0 - OK, !0 - error
+ */
+static av_cold int che_configure(AACContext *ac,
+                                 enum ChannelPosition che_pos,
+                                 int type, int id, int *channels)
+{
+    if (*channels >= MAX_CHANNELS)
+        return AVERROR_INVALIDDATA;
+    if (che_pos) {
+        if (!ac->che[type][id]) {
+            if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
+                return AVERROR(ENOMEM);
+            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);
+    if (!avctx->channels)
+        return 1;
+
+    ac->frame->nb_samples = 2048;
+    if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
+        return ret;
+
+    /* map output channel pointers to AVFrame data */
+    for (ch = 0; ch < avctx->channels; ch++) {
+        if (ac->output_element[ch])
+            ac->output_element[ch]->ret = (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;
+
+    if (num_side_channels == 0 && num_back_channels >= 4) {
+        num_side_channels = 2;
+        num_back_channels -= 2;
+    }
+
+    i = 0;
+    if (num_front_channels & 1) {
+        e2c_vec[i] = (struct elem_to_channel) {
+            .av_position  = AV_CH_FRONT_CENTER,
+            .syn_ele      = TYPE_SCE,
+            .elem_id      = layout_map[i][1],
+            .aac_position = AAC_CHANNEL_FRONT
+        };
+        i++;
+        num_front_channels--;
+    }
+    if (num_front_channels >= 4) {
+        i += assign_pair(e2c_vec, layout_map, i,
+                         AV_CH_FRONT_LEFT_OF_CENTER,
+                         AV_CH_FRONT_RIGHT_OF_CENTER,
+                         AAC_CHANNEL_FRONT);
+        num_front_channels -= 2;
+    }
+    if (num_front_channels >= 2) {
+        i += assign_pair(e2c_vec, layout_map, i,
+                         AV_CH_FRONT_LEFT,
+                         AV_CH_FRONT_RIGHT,
+                         AAC_CHANNEL_FRONT);
+        num_front_channels -= 2;
+    }
+    while (num_front_channels >= 2) {
+        i += assign_pair(e2c_vec, layout_map, i,
+                         UINT64_MAX,
+                         UINT64_MAX,
+                         AAC_CHANNEL_FRONT);
+        num_front_channels -= 2;
+    }
+
+    if (num_side_channels >= 2) {
+        i += assign_pair(e2c_vec, layout_map, i,
+                         AV_CH_SIDE_LEFT,
+                         AV_CH_SIDE_RIGHT,
+                         AAC_CHANNEL_FRONT);
+        num_side_channels -= 2;
+    }
+    while (num_side_channels >= 2) {
+        i += assign_pair(e2c_vec, layout_map, i,
+                         UINT64_MAX,
+                         UINT64_MAX,
+                         AAC_CHANNEL_SIDE);
+        num_side_channels -= 2;
+    }
+
+    while (num_back_channels >= 4) {
+        i += assign_pair(e2c_vec, layout_map, i,
+                         UINT64_MAX,
+                         UINT64_MAX,
+                         AAC_CHANNEL_BACK);
+        num_back_channels -= 2;
+    }
+    if (num_back_channels >= 2) {
+        i += assign_pair(e2c_vec, layout_map, i,
+                         AV_CH_BACK_LEFT,
+                         AV_CH_BACK_RIGHT,
+                         AAC_CHANNEL_BACK);
+        num_back_channels -= 2;
+    }
+    if (num_back_channels) {
+        e2c_vec[i] = (struct elem_to_channel) {
+            .av_position  = AV_CH_BACK_CENTER,
+            .syn_ele      = TYPE_SCE,
+            .elem_id      = layout_map[i][1],
+            .aac_position = AAC_CHANNEL_BACK
+        };
+        i++;
+        num_back_channels--;
+    }
+
+    if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
+        e2c_vec[i] = (struct elem_to_channel) {
+            .av_position  = AV_CH_LOW_FREQUENCY,
+            .syn_ele      = TYPE_LFE,
+            .elem_id      = layout_map[i][1],
+            .aac_position = AAC_CHANNEL_LFE
+        };
+        i++;
+    }
+    while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
+        e2c_vec[i] = (struct elem_to_channel) {
+            .av_position  = UINT64_MAX,
+            .syn_ele      = TYPE_LFE,
+            .elem_id      = layout_map[i][1],
+            .aac_position = AAC_CHANNEL_LFE
+        };
+        i++;
+    }
+
+    // Must choose a stable sort
+    total_non_cc_elements = n = i;
+    do {
+        int next_n = 0;
+        for (i = 1; i < n; i++)
+            if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
+                FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]);
+                next_n = i;
+            }
+        n = next_n;
+    } while (n > 0);
+
+    layout = 0;
+    for (i = 0; i < total_non_cc_elements; i++) {
+        layout_map[i][0] = e2c_vec[i].syn_ele;
+        layout_map[i][1] = e2c_vec[i].elem_id;
+        layout_map[i][2] = e2c_vec[i].aac_position;
+        if (e2c_vec[i].av_position != UINT64_MAX) {
+            layout |= e2c_vec[i].av_position;
+        }
+    }
+
+    return layout;
+}
+
+/**
+ * Save current output configuration if and only if it has been locked.
+ */
+static void push_output_configuration(AACContext *ac) {
+    if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) {
+        ac->oc[0] = ac->oc[1];
+    }
+    ac->oc[1].status = OC_NONE;
+}
+
+/**
+ * Restore the previous output configuration if and only if the current
+ * configuration is unlocked.
+ */
+static void pop_output_configuration(AACContext *ac) {
+    if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
+        ac->oc[1] = ac->oc[0];
+        ac->avctx->channels = ac->oc[1].channels;
+        ac->avctx->channel_layout = ac->oc[1].channel_layout;
+        output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
+                         ac->oc[1].status, 0);
+    }
+}
+
+/**
+ * Configure output channel order based on the current program
+ * configuration element.
+ *
+ * @return  Returns error status. 0 - OK, !0 - error
+ */
+static int output_configure(AACContext *ac,
+                            uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
+                            enum OCStatus oc_type, int get_new_frame)
+{
+    AVCodecContext *avctx = ac->avctx;
+    int i, channels = 0, ret;
+    uint64_t layout = 0;
+    uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }};
+    uint8_t type_counts[TYPE_END] = { 0 };
+
+    if (ac->oc[1].layout_map != layout_map) {
+        memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
+        ac->oc[1].layout_map_tags = tags;
+    }
+    for (i = 0; i < tags; i++) {
+        int type =         layout_map[i][0];
+        int id =           layout_map[i][1];
+        id_map[type][id] = type_counts[type]++;
+    }
+    // Try to sniff a reasonable channel order, otherwise output the
+    // channels in the order the PCE declared them.
+    if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE)
+        layout = sniff_channel_order(layout_map, tags);
+    for (i = 0; i < tags; i++) {
+        int type =     layout_map[i][0];
+        int id =       layout_map[i][1];
+        int iid =      id_map[type][id];
+        int position = layout_map[i][2];
+        // Allocate or free elements depending on if they are in the
+        // current program configuration.
+        ret = che_configure(ac, position, type, iid, &channels);
+        if (ret < 0)
+            return ret;
+        ac->tag_che_map[type][id] = ac->che[type][iid];
+    }
+    if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
+        if (layout == AV_CH_FRONT_CENTER) {
+            layout = AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT;
+        } else {
+            layout = 0;
+        }
+    }
+
+    if (layout) avctx->channel_layout = layout;
+                            ac->oc[1].channel_layout = layout;
+    avctx->channels       = ac->oc[1].channels       = channels;
+    ac->oc[1].status = oc_type;
+
+    if (get_new_frame) {
+        if ((ret = frame_configure_elements(ac->avctx)) < 0)
+            return ret;
+    }
+
+    return 0;
+}
+
+static void flush(AVCodecContext *avctx)
+{
+    AACContext *ac= avctx->priv_data;
+    int type, i, j;
+
+    for (type = 3; type >= 0; type--) {
+        for (i = 0; i < MAX_ELEM_ID; i++) {
+            ChannelElement *che = ac->che[type][i];
+            if (che) {
+                for (j = 0; j <= 1; j++) {
+                    memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved));
+                }
+            }
+        }
+    }
+}
+
+/**
+ * Set up channel positions based on a default channel configuration
+ * as specified in table 1.17.
+ *
+ * @return  Returns error status. 0 - OK, !0 - error
+ */
+static int set_default_channel_config(AVCodecContext *avctx,
+                                      uint8_t (*layout_map)[3],
+                                      int *tags,
+                                      int channel_config)
+{
+    if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
+        channel_config > 12) {
+        av_log(avctx, AV_LOG_ERROR,
+               "invalid default channel configuration (%d)\n",
+               channel_config);
+        return AVERROR_INVALIDDATA;
+    }
+    *tags = tags_per_config[channel_config];
+    memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
+           *tags * sizeof(*layout_map));
+
+    /*
+     * AAC specification has 7.1(wide) as a default layout for 8-channel streams.
+     * However, at least Nero AAC encoder encodes 7.1 streams using the default
+     * channel config 7, mapping the side channels of the original audio stream
+     * to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD
+     * decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
+     * the incorrect streams as if they were correct (and as the encoder intended).
+     *
+     * As actual intended 7.1(wide) streams are very rare, default to assuming a
+     * 7.1 layout was intended.
+     */
+    if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
+        av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
+               " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
+               " according to the specification instead.\n", FF_COMPLIANCE_STRICT);
+        layout_map[2][2] = AAC_CHANNEL_SIDE;
+    }
+
+    return 0;
+}
+
+static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
+{
+    /* For PCE based channel configurations map the channels solely based
+     * on tags. */
+    if (!ac->oc[1].m4ac.chan_config) {
+        return ac->tag_che_map[type][elem_id];
+    }
+    // Allow single CPE stereo files to be signalled with mono configuration.
+    if (!ac->tags_mapped && type == TYPE_CPE &&
+        ac->oc[1].m4ac.chan_config == 1) {
+        uint8_t layout_map[MAX_ELEM_ID*4][3];
+        int layout_map_tags;
+        push_output_configuration(ac);
+
+        av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");
+
+        if (set_default_channel_config(ac->avctx, layout_map,
+                                       &layout_map_tags, 2) < 0)
+            return NULL;
+        if (output_configure(ac, layout_map, layout_map_tags,
+                             OC_TRIAL_FRAME, 1) < 0)
+            return NULL;
+
+        ac->oc[1].m4ac.chan_config = 2;
+        ac->oc[1].m4ac.ps = 0;
+    }
+    // And vice-versa
+    if (!ac->tags_mapped && type == TYPE_SCE &&
+        ac->oc[1].m4ac.chan_config == 2) {
+        uint8_t layout_map[MAX_ELEM_ID * 4][3];
+        int layout_map_tags;
+        push_output_configuration(ac);
+
+        av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");
+
+        if (set_default_channel_config(ac->avctx, layout_map,
+                                       &layout_map_tags, 1) < 0)
+            return NULL;
+        if (output_configure(ac, layout_map, layout_map_tags,
+                             OC_TRIAL_FRAME, 1) < 0)
+            return NULL;
+
+        ac->oc[1].m4ac.chan_config = 1;
+        if (ac->oc[1].m4ac.sbr)
+            ac->oc[1].m4ac.ps = -1;
+    }
+    /* For indexed channel configurations map the channels solely based
+     * on position. */
+    switch (ac->oc[1].m4ac.chan_config) {
+    case 12:
+    case 7:
+        if (ac->tags_mapped == 3 && type == TYPE_CPE) {
+            ac->tags_mapped++;
+            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
+        }
+    case 11:
+        if (ac->tags_mapped == 2 &&
+            ac->oc[1].m4ac.chan_config == 11 &&
+            type == TYPE_SCE) {
+            ac->tags_mapped++;
+            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
+        }
+    case 6:
+        /* Some streams incorrectly code 5.1 audio as
+         * SCE[0] CPE[0] CPE[1] SCE[1]
+         * instead of
+         * SCE[0] CPE[0] CPE[1] LFE[0].
+         * If we seem to have encountered such a stream, transfer
+         * the LFE[0] element to the SCE[1]'s mapping */
+        if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
+            if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) {
+                av_log(ac->avctx, AV_LOG_WARNING,
+                   "This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n",
+                   type == TYPE_SCE ? "SCE" : "LFE", elem_id);
+                ac->warned_remapping_once++;
+            }
+            ac->tags_mapped++;
+            return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
+        }
+    case 5:
+        if (ac->tags_mapped == 2 && type == TYPE_CPE) {
+            ac->tags_mapped++;
+            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
+        }
+    case 4:
+        /* Some streams incorrectly code 4.0 audio as
+         * SCE[0] CPE[0] LFE[0]
+         * instead of
+         * SCE[0] CPE[0] SCE[1].
+         * If we seem to have encountered such a stream, transfer
+         * the SCE[1] element to the LFE[0]'s mapping */
+        if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
+            if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) {
+                av_log(ac->avctx, AV_LOG_WARNING,
+                   "This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
+                   type == TYPE_SCE ? "SCE" : "LFE", elem_id);
+                ac->warned_remapping_once++;
+            }
+            ac->tags_mapped++;
+            return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1];
+        }
+        if (ac->tags_mapped == 2 &&
+            ac->oc[1].m4ac.chan_config == 4 &&
+            type == TYPE_SCE) {
+            ac->tags_mapped++;
+            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
+        }
+    case 3:
+    case 2:
+        if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
+            type == TYPE_CPE) {
+            ac->tags_mapped++;
+            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
+        } else if (ac->oc[1].m4ac.chan_config == 2) {
+            return NULL;
+        }
+    case 1:
+        if (!ac->tags_mapped && type == TYPE_SCE) {
+            ac->tags_mapped++;
+            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
+        }
+    default:
+        return NULL;
+    }
+}
+
+/**
+ * Decode an array of 4 bit element IDs, optionally interleaved with a
+ * stereo/mono switching bit.
+ *
+ * @param type speaker type/position for these channels
+ */
+static void decode_channel_map(uint8_t layout_map[][3],
+                               enum ChannelPosition type,
+                               GetBitContext *gb, int n)
+{
+    while (n--) {
+        enum RawDataBlockType syn_ele;
+        switch (type) {
+        case AAC_CHANNEL_FRONT:
+        case AAC_CHANNEL_BACK:
+        case AAC_CHANNEL_SIDE:
+            syn_ele = get_bits1(gb);
+            break;
+        case AAC_CHANNEL_CC:
+            skip_bits1(gb);
+            syn_ele = TYPE_CCE;
+            break;
+        case AAC_CHANNEL_LFE:
+            syn_ele = TYPE_LFE;
+            break;
+        default:
+            // AAC_CHANNEL_OFF has no channel map
+            av_assert0(0);
+        }
+        layout_map[0][0] = syn_ele;
+        layout_map[0][1] = get_bits(gb, 4);
+        layout_map[0][2] = type;
+        layout_map++;
+    }
+}
+
+/**
+ * Decode program configuration element; reference: table 4.2.
+ *
+ * @return  Returns error status. 0 - OK, !0 - error
+ */
+static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
+                      uint8_t (*layout_map)[3],
+                      GetBitContext *gb)
+{
+    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
+    int sampling_index;
+    int comment_len;
+    int tags;
+
+    skip_bits(gb, 2);  // object_type
+
+    sampling_index = get_bits(gb, 4);
+    if (m4ac->sampling_index != sampling_index)
+        av_log(avctx, AV_LOG_WARNING,
+               "Sample rate index in program config element does not "
+               "match the sample rate index configured by the container.\n");
+
+    num_front       = get_bits(gb, 4);
+    num_side        = get_bits(gb, 4);
+    num_back        = get_bits(gb, 4);
+    num_lfe         = get_bits(gb, 2);
+    num_assoc_data  = get_bits(gb, 3);
+    num_cc          = get_bits(gb, 4);
+
+    if (get_bits1(gb))
+        skip_bits(gb, 4); // mono_mixdown_tag
+    if (get_bits1(gb))
+        skip_bits(gb, 4); // stereo_mixdown_tag
+
+    if (get_bits1(gb))
+        skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
+
+    if (get_bits_left(gb) < 4 * (num_front + num_side + num_back + num_lfe + num_assoc_data + num_cc)) {
+        av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
+        return -1;
+    }
+    decode_channel_map(layout_map       , AAC_CHANNEL_FRONT, gb, num_front);
+    tags = num_front;
+    decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE,  gb, num_side);
+    tags += num_side;
+    decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK,  gb, num_back);
+    tags += num_back;
+    decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE,   gb, num_lfe);
+    tags += num_lfe;
+
+    skip_bits_long(gb, 4 * num_assoc_data);
+
+    decode_channel_map(layout_map + tags, AAC_CHANNEL_CC,    gb, num_cc);
+    tags += num_cc;
+
+    align_get_bits(gb);
+
+    /* comment field, first byte is length */
+    comment_len = get_bits(gb, 8) * 8;
+    if (get_bits_left(gb) < comment_len) {
+        av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
+        return AVERROR_INVALIDDATA;
+    }
+    skip_bits_long(gb, comment_len);
+    return tags;
+}
+
+/**
+ * Decode GA "General Audio" specific configuration; reference: table 4.1.
+ *
+ * @param   ac          pointer to AACContext, may be null
+ * @param   avctx       pointer to AVCCodecContext, used for logging
+ *
+ * @return  Returns error status. 0 - OK, !0 - error
+ */
+static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
+                                     GetBitContext *gb,
+                                     MPEG4AudioConfig *m4ac,
+                                     int channel_config)
+{
+    int extension_flag, ret, ep_config, res_flags;
+    uint8_t layout_map[MAX_ELEM_ID*4][3];
+    int tags = 0;
+
+    if (get_bits1(gb)) { // frameLengthFlag
+        avpriv_request_sample(avctx, "960/120 MDCT window");
+        return AVERROR_PATCHWELCOME;
+    }
+    m4ac->frame_length_short = 0;
+
+    if (get_bits1(gb))       // dependsOnCoreCoder
+        skip_bits(gb, 14);   // coreCoderDelay
+    extension_flag = get_bits1(gb);
+
+    if (m4ac->object_type == AOT_AAC_SCALABLE ||
+        m4ac->object_type == AOT_ER_AAC_SCALABLE)
+        skip_bits(gb, 3);     // layerNr
+
+    if (channel_config == 0) {
+        skip_bits(gb, 4);  // element_instance_tag
+        tags = decode_pce(avctx, m4ac, layout_map, gb);
+        if (tags < 0)
+            return tags;
+    } else {
+        if ((ret = set_default_channel_config(avctx, layout_map,
+                                              &tags, channel_config)))
+            return ret;
+    }
+
+    if (count_channels(layout_map, tags) > 1) {
+        m4ac->ps = 0;
+    } else if (m4ac->sbr == 1 && m4ac->ps == -1)
+        m4ac->ps = 1;
+
+    if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
+        return ret;
+
+    if (extension_flag) {
+        switch (m4ac->object_type) {
+        case AOT_ER_BSAC:
+            skip_bits(gb, 5);    // numOfSubFrame
+            skip_bits(gb, 11);   // layer_length
+            break;
+        case AOT_ER_AAC_LC:
+        case AOT_ER_AAC_LTP:
+        case AOT_ER_AAC_SCALABLE:
+        case AOT_ER_AAC_LD:
+            res_flags = get_bits(gb, 3);
+            if (res_flags) {
+                avpriv_report_missing_feature(avctx,
+                                              "AAC data resilience (flags %x)",
+                                              res_flags);
+                return AVERROR_PATCHWELCOME;
+            }
+            break;
+        }
+        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
+    }
+    switch (m4ac->object_type) {
+    case AOT_ER_AAC_LC:
+    case AOT_ER_AAC_LTP:
+    case AOT_ER_AAC_SCALABLE:
+    case AOT_ER_AAC_LD:
+        ep_config = get_bits(gb, 2);
+        if (ep_config) {
+            avpriv_report_missing_feature(avctx,
+                                          "epConfig %d", ep_config);
+            return AVERROR_PATCHWELCOME;
+        }
+    }
+    return 0;
+}
+
+static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
+                                     GetBitContext *gb,
+                                     MPEG4AudioConfig *m4ac,
+                                     int channel_config)
+{
+    int ret, ep_config, res_flags;
+    uint8_t layout_map[MAX_ELEM_ID*4][3];
+    int tags = 0;
+    const int ELDEXT_TERM = 0;
+
+    m4ac->ps  = 0;
+    m4ac->sbr = 0;
+
+    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(avctx, layout_map,
+                                          &tags, channel_config)))
+        return ret;
+
+    if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
+        return ret;
+
+    ep_config = get_bits(gb, 2);
+    if (ep_config) {
+        avpriv_report_missing_feature(avctx,
+                                      "epConfig %d", ep_config);
+        return AVERROR_PATCHWELCOME;
+    }
+    return 0;
+}
+
+/**
+ * Decode audio specific configuration; reference: table 1.13.
+ *
+ * @param   ac          pointer to AACContext, may be null
+ * @param   avctx       pointer to AVCCodecContext, used for logging
+ * @param   m4ac        pointer to MPEG4AudioConfig, used for parsing
+ * @param   data        pointer to buffer holding an audio specific config
+ * @param   bit_size    size of audio specific config or data in bits
+ * @param   sync_extension look for an appended sync extension
+ *
+ * @return  Returns error status or number of consumed bits. <0 - error
+ */
+static int decode_audio_specific_config(AACContext *ac,
+                                        AVCodecContext *avctx,
+                                        MPEG4AudioConfig *m4ac,
+                                        const uint8_t *data, int bit_size,
+                                        int sync_extension)
+{
+    GetBitContext gb;
+    int i, ret;
+
+    ff_dlog(avctx, "audio specific config size %d\n", bit_size >> 3);
+    for (i = 0; i < bit_size >> 3; i++)
+        ff_dlog(avctx, "%02x ", data[i]);
+    ff_dlog(avctx, "\n");
+
+    if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
+        return ret;
+
+    if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size,
+                                          sync_extension)) < 0)
+        return AVERROR_INVALIDDATA;
+    if (m4ac->sampling_index > 12) {
+        av_log(avctx, AV_LOG_ERROR,
+               "invalid sampling rate index %d\n",
+               m4ac->sampling_index);
+        return AVERROR_INVALIDDATA;
+    }
+    if (m4ac->object_type == AOT_ER_AAC_LD &&
+        (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
+        av_log(avctx, AV_LOG_ERROR,
+               "invalid low delay sampling rate index %d\n",
+               m4ac->sampling_index);
+        return AVERROR_INVALIDDATA;
+    }
+
+    skip_bits_long(&gb, i);
+
+    switch (m4ac->object_type) {
+    case AOT_AAC_MAIN:
+    case AOT_AAC_LC:
+    case AOT_AAC_LTP:
+    case AOT_ER_AAC_LC:
+    case AOT_ER_AAC_LD:
+        if ((ret = decode_ga_specific_config(ac, avctx, &gb,
+                                            m4ac, m4ac->chan_config)) < 0)
+            return ret;
+        break;
+    case AOT_ER_AAC_ELD:
+        if ((ret = decode_eld_specific_config(ac, avctx, &gb,
+                                              m4ac, m4ac->chan_config)) < 0)
+            return ret;
+        break;
+    default:
+        avpriv_report_missing_feature(avctx,
+                                      "Audio object type %s%d",
+                                      m4ac->sbr == 1 ? "SBR+" : "",
+                                      m4ac->object_type);
+        return AVERROR(ENOSYS);
+    }
+
+    ff_dlog(avctx,
+            "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
+            m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
+            m4ac->sample_rate, m4ac->sbr,
+            m4ac->ps);
+
+    return get_bits_count(&gb);
+}
+
+/**
+ * linear congruential pseudorandom number generator
+ *
+ * @param   previous_val    pointer to the current state of the generator
+ *
+ * @return  Returns a 32-bit pseudorandom integer
+ */
+static av_always_inline int lcg_random(unsigned previous_val)
+{
+    union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
+    return v.s;
+}
+
+static void reset_all_predictors(PredictorState *ps)
+{
+    int i;
+    for (i = 0; i < MAX_PREDICTORS; i++)
+        reset_predict_state(&ps[i]);
+}
+
+static int sample_rate_idx (int rate)
+{
+         if (92017 <= rate) return 0;
+    else if (75132 <= rate) return 1;
+    else if (55426 <= rate) return 2;
+    else if (46009 <= rate) return 3;
+    else if (37566 <= rate) return 4;
+    else if (27713 <= rate) return 5;
+    else if (23004 <= rate) return 6;
+    else if (18783 <= rate) return 7;
+    else if (13856 <= rate) return 8;
+    else if (11502 <= rate) return 9;
+    else if (9391  <= rate) return 10;
+    else                    return 11;
+}
+
+static void reset_predictor_group(PredictorState *ps, int group_num)
+{
+    int i;
+    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
+        reset_predict_state(&ps[i]);
+}
+
+#define AAC_INIT_VLC_STATIC(num, size)                                     \
+    INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num],     \
+         ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]),  \
+                                    sizeof(ff_aac_spectral_bits[num][0]),  \
+        ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
+                                    sizeof(ff_aac_spectral_codes[num][0]), \
+        size);
+
+static void aacdec_init(AACContext *ac);
+
+static av_cold int aac_decode_init(AVCodecContext *avctx)
+{
+    AACContext *ac = avctx->priv_data;
+    int ret;
+
+    ac->avctx = avctx;
+    ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
+
+    aacdec_init(ac);
+
+    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 * 8,
+                                                1)) < 0)
+            return ret;
+    } else {
+        int sr, i;
+        uint8_t layout_map[MAX_ELEM_ID*4][3];
+        int layout_map_tags;
+
+        sr = sample_rate_idx(avctx->sample_rate);
+        ac->oc[1].m4ac.sampling_index = sr;
+        ac->oc[1].m4ac.channels = avctx->channels;
+        ac->oc[1].m4ac.sbr = -1;
+        ac->oc[1].m4ac.ps = -1;
+
+        for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
+            if (ff_mpeg4audio_channels[i] == avctx->channels)
+                break;
+        if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
+            i = 0;
+        }
+        ac->oc[1].m4ac.chan_config = i;
+
+        if (ac->oc[1].m4ac.chan_config) {
+            int ret = set_default_channel_config(avctx, layout_map,
+                &layout_map_tags, ac->oc[1].m4ac.chan_config);
+            if (!ret)
+                output_configure(ac, layout_map, layout_map_tags,
+                                 OC_GLOBAL_HDR, 0);
+            else if (avctx->err_recognition & AV_EF_EXPLODE)
+                return AVERROR_INVALIDDATA;
+        }
+    }
+
+    if (avctx->channels > MAX_CHANNELS) {
+        av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    AAC_INIT_VLC_STATIC( 0, 304);
+    AAC_INIT_VLC_STATIC( 1, 270);
+    AAC_INIT_VLC_STATIC( 2, 550);
+    AAC_INIT_VLC_STATIC( 3, 300);
+    AAC_INIT_VLC_STATIC( 4, 328);
+    AAC_INIT_VLC_STATIC( 5, 294);
+    AAC_INIT_VLC_STATIC( 6, 306);
+    AAC_INIT_VLC_STATIC( 7, 268);
+    AAC_INIT_VLC_STATIC( 8, 510);
+    AAC_INIT_VLC_STATIC( 9, 366);
+    AAC_INIT_VLC_STATIC(10, 462);
+
+    ff_aac_sbr_init();
+
+    ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
+    if (!ac->fdsp) {
+        return AVERROR(ENOMEM);
+    }
+
+    ac->random_state = 0x1f2e3d4c;
+
+    ff_aac_tableinit();
+
+    INIT_VLC_STATIC(&vlc_scalefactors, 7,
+                    FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
+                    ff_aac_scalefactor_bits,
+                    sizeof(ff_aac_scalefactor_bits[0]),
+                    sizeof(ff_aac_scalefactor_bits[0]),
+                    ff_aac_scalefactor_code,
+                    sizeof(ff_aac_scalefactor_code[0]),
+                    sizeof(ff_aac_scalefactor_code[0]),
+                    352);
+
+    ff_mdct_init(&ac->mdct,       11, 1, 1.0 / (32768.0 * 1024.0));
+    ff_mdct_init(&ac->mdct_ld,    10, 1, 1.0 / (32768.0 * 512.0));
+    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);
+    ret = ff_imdct15_init(&ac->mdct480, 5);
+    if (ret < 0)
+        return ret;
+
+    // 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( 9);
+    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 AVERROR_INVALIDDATA;
+    }
+    skip_bits_long(gb, 8 * count);
+    return 0;
+}
+
+static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
+                             GetBitContext *gb)
+{
+    int sfb;
+    if (get_bits1(gb)) {
+        ics->predictor_reset_group = get_bits(gb, 5);
+        if (ics->predictor_reset_group == 0 ||
+            ics->predictor_reset_group > 30) {
+            av_log(ac->avctx, AV_LOG_ERROR,
+                   "Invalid Predictor Reset Group.\n");
+            return AVERROR_INVALIDDATA;
+        }
+    }
+    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
+        ics->prediction_used[sfb] = get_bits1(gb);
+    }
+    return 0;
+}
+
+/**
+ * Decode Long Term Prediction data; reference: table 4.xx.
+ */
+static void decode_ltp(LongTermPrediction *ltp,
+                       GetBitContext *gb, uint8_t max_sfb)
+{
+    int sfb;
+
+    ltp->lag  = get_bits(gb, 11);
+    ltp->coef = ltp_coef[get_bits(gb, 3)];
+    for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
+        ltp->used[sfb] = get_bits1(gb);
+}
+
+/**
+ * Decode Individual Channel Stream info; reference: table 4.6.
+ */
+static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
+                           GetBitContext *gb)
+{
+    const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
+    const int aot = m4ac->object_type;
+    const int sampling_index = m4ac->sampling_index;
+    if (aot != AOT_ER_AAC_ELD) {
+        if (get_bits1(gb)) {
+            av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
+            if (ac->avctx->err_recognition & AV_EF_BITSTREAM)
+                return AVERROR_INVALIDDATA;
+        }
+        ics->window_sequence[1] = ics->window_sequence[0];
+        ics->window_sequence[0] = get_bits(gb, 2);
+        if (aot == AOT_ER_AAC_LD &&
+            ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
+            av_log(ac->avctx, AV_LOG_ERROR,
+                   "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
+                   "window sequence %d found.\n", ics->window_sequence[0]);
+            ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
+            return AVERROR_INVALIDDATA;
+        }
+        ics->use_kb_window[1]   = ics->use_kb_window[0];
+        ics->use_kb_window[0]   = get_bits1(gb);
+    }
+    ics->num_window_groups  = 1;
+    ics->group_len[0]       = 1;
+    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+        int i;
+        ics->max_sfb = get_bits(gb, 4);
+        for (i = 0; i < 7; i++) {
+            if (get_bits1(gb)) {
+                ics->group_len[ics->num_window_groups - 1]++;
+            } else {
+                ics->num_window_groups++;
+                ics->group_len[ics->num_window_groups - 1] = 1;
+            }
+        }
+        ics->num_windows       = 8;
+        ics->swb_offset        =    ff_swb_offset_128[sampling_index];
+        ics->num_swb           =   ff_aac_num_swb_128[sampling_index];
+        ics->tns_max_bands     = ff_tns_max_bands_128[sampling_index];
+        ics->predictor_present = 0;
+    } else {
+        ics->max_sfb           = get_bits(gb, 6);
+        ics->num_windows       = 1;
+        if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
+            if (m4ac->frame_length_short) {
+                ics->swb_offset    =     ff_swb_offset_480[sampling_index];
+                ics->num_swb       =    ff_aac_num_swb_480[sampling_index];
+                ics->tns_max_bands =  ff_tns_max_bands_480[sampling_index];
+            } else {
+                ics->swb_offset    =     ff_swb_offset_512[sampling_index];
+                ics->num_swb       =    ff_aac_num_swb_512[sampling_index];
+                ics->tns_max_bands =  ff_tns_max_bands_512[sampling_index];
+            }
+            if (!ics->num_swb || !ics->swb_offset)
+                return AVERROR_BUG;
+        } else {
+            ics->swb_offset    =    ff_swb_offset_1024[sampling_index];
+            ics->num_swb       =   ff_aac_num_swb_1024[sampling_index];
+            ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index];
+        }
+        if (aot != AOT_ER_AAC_ELD) {
+            ics->predictor_present     = get_bits1(gb);
+            ics->predictor_reset_group = 0;
+        }
+        if (ics->predictor_present) {
+            if (aot == AOT_AAC_MAIN) {
+                if (decode_prediction(ac, ics, gb)) {
+                    goto fail;
+                }
+            } else if (aot == AOT_AAC_LC ||
+                       aot == AOT_ER_AAC_LC) {
+                av_log(ac->avctx, AV_LOG_ERROR,
+                       "Prediction is not allowed in AAC-LC.\n");
+                goto fail;
+            } else {
+                if (aot == AOT_ER_AAC_LD) {
+                    av_log(ac->avctx, AV_LOG_ERROR,
+                           "LTP in ER AAC LD not yet implemented.\n");
+                    return AVERROR_PATCHWELCOME;
+                }
+                if ((ics->ltp.present = get_bits(gb, 1)))
+                    decode_ltp(&ics->ltp, gb, ics->max_sfb);
+            }
+        }
+    }
+
+    if (ics->max_sfb > ics->num_swb) {
+        av_log(ac->avctx, AV_LOG_ERROR,
+               "Number of scalefactor bands in group (%d) "
+               "exceeds limit (%d).\n",
+               ics->max_sfb, ics->num_swb);
+        goto fail;
+    }
+
+    return 0;
+fail:
+    ics->max_sfb = 0;
+    return AVERROR_INVALIDDATA;
+}
+
+/**
+ * Decode band types (section_data payload); reference: table 4.46.
+ *
+ * @param   band_type           array of the used band type
+ * @param   band_type_run_end   array of the last scalefactor band of a band type run
+ *
+ * @return  Returns error status. 0 - OK, !0 - error
+ */
+static int decode_band_types(AACContext *ac, enum BandType band_type[120],
+                             int band_type_run_end[120], GetBitContext *gb,
+                             IndividualChannelStream *ics)
+{
+    int g, idx = 0;
+    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
+    for (g = 0; g < ics->num_window_groups; g++) {
+        int k = 0;
+        while (k < ics->max_sfb) {
+            uint8_t sect_end = k;
+            int sect_len_incr;
+            int sect_band_type = get_bits(gb, 4);
+            if (sect_band_type == 12) {
+                av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
+                return AVERROR_INVALIDDATA;
+            }
+            do {
+                sect_len_incr = get_bits(gb, bits);
+                sect_end += sect_len_incr;
+                if (get_bits_left(gb) < 0) {
+                    av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
+                    return AVERROR_INVALIDDATA;
+                }
+                if (sect_end > ics->max_sfb) {
+                    av_log(ac->avctx, AV_LOG_ERROR,
+                           "Number of bands (%d) exceeds limit (%d).\n",
+                           sect_end, ics->max_sfb);
+                    return AVERROR_INVALIDDATA;
+                }
+            } while (sect_len_incr == (1 << bits) - 1);
+            for (; k < sect_end; k++) {
+                band_type        [idx]   = sect_band_type;
+                band_type_run_end[idx++] = sect_end;
+            }
+        }
+    }
+    return 0;
+}
+
+/**
+ * Decode scalefactors; reference: table 4.47.
+ *
+ * @param   global_gain         first scalefactor value as scalefactors are differentially coded
+ * @param   band_type           array of the used band type
+ * @param   band_type_run_end   array of the last scalefactor band of a band type run
+ * @param   sf                  array of scalefactors or intensity stereo positions
+ *
+ * @return  Returns error status. 0 - OK, !0 - error
+ */
+static int decode_scalefactors(AACContext *ac, 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 - NOISE_OFFSET, 0 };
+    int clipped_offset;
+    int noise_flag = 1;
+    for (g = 0; g < ics->num_window_groups; g++) {
+        for (i = 0; i < ics->max_sfb;) {
+            int run_end = band_type_run_end[idx];
+            if (band_type[idx] == ZERO_BT) {
+                for (; i < run_end; i++, idx++)
+                    sf[idx] = 0.0;
+            } else if ((band_type[idx] == INTENSITY_BT) ||
+                       (band_type[idx] == INTENSITY_BT2)) {
+                for (; i < run_end; i++, idx++) {
+                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
+                    clipped_offset = av_clip(offset[2], -155, 100);
+                    if (offset[2] != clipped_offset) {
+                        avpriv_request_sample(ac->avctx,
+                                              "If you heard an audible artifact, there may be a bug in the decoder. "
+                                              "Clipped intensity stereo position (%d -> %d)",
+                                              offset[2], clipped_offset);
+                    }
+                    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, NOISE_PRE_BITS) - NOISE_PRE;
+                    else
+                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
+                    clipped_offset = av_clip(offset[1], -100, 155);
+                    if (offset[1] != clipped_offset) {
+                        avpriv_request_sample(ac->avctx,
+                                              "If you heard an audible artifact, there may be a bug in the decoder. "
+                                              "Clipped noise gain (%d -> %d)",
+                                              offset[1], clipped_offset);
+                    }
+                    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) - 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;
+                    }
+                    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] >= swb_offset[num_swb])
+        return -1;
+    pulse->amp[0]    = get_bits(gb, 4);
+    for (i = 1; i < pulse->num_pulse; i++) {
+        pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
+        if (pulse->pos[i] >= swb_offset[num_swb])
+            return -1;
+        pulse->amp[i] = get_bits(gb, 4);
+    }
+    return 0;
+}
+
+/**
+ * Decode Temporal Noise Shaping data; reference: table 4.48.
+ *
+ * @return  Returns error status. 0 - OK, !0 - error
+ */
+static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
+                      GetBitContext *gb, const IndividualChannelStream *ics)
+{
+    int w, filt, i, coef_len, coef_res, coef_compress;
+    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
+    const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
+    for (w = 0; w < ics->num_windows; w++) {
+        if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
+            coef_res = get_bits1(gb);
+
+            for (filt = 0; filt < tns->n_filt[w]; filt++) {
+                int tmp2_idx;
+                tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
+
+                if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
+                    av_log(ac->avctx, AV_LOG_ERROR,
+                           "TNS filter order %d is greater than maximum %d.\n",
+                           tns->order[w][filt], tns_max_order);
+                    tns->order[w][filt] = 0;
+                    return AVERROR_INVALIDDATA;
+                }
+                if (tns->order[w][filt]) {
+                    tns->direction[w][filt] = get_bits1(gb);
+                    coef_compress = get_bits1(gb);
+                    coef_len = coef_res + 3 - coef_compress;
+                    tmp2_idx = 2 * coef_compress + coef_res;
+
+                    for (i = 0; i < tns->order[w][filt]; i++)
+                        tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
+                }
+            }
+        }
+    }
+    return 0;
+}
+
+/**
+ * Decode Mid/Side data; reference: table 4.54.
+ *
+ * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
+ *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
+ *                      [3] reserved for scalable AAC
+ */
+static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
+                                   int ms_present)
+{
+    int idx;
+    int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
+    if (ms_present == 1) {
+        for (idx = 0; idx < max_idx; idx++)
+            cpe->ms_mask[idx] = get_bits1(gb);
+    } else if (ms_present == 2) {
+        memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
+    }
+}
+
+/**
+ * Decode spectral data; reference: table 4.50.
+ * Dequantize and scale spectral data; reference: 4.6.3.3.
+ *
+ * @param   coef            array of dequantized, scaled spectral data
+ * @param   sf              array of scalefactors or intensity stereo positions
+ * @param   pulse_present   set if pulses are present
+ * @param   pulse           pointer to pulse data struct
+ * @param   band_type       array of the used band type
+ *
+ * @return  Returns error status. 0 - OK, !0 - error
+ */
+static int decode_spectrum_and_dequant(AACContext *ac, 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 AVERROR_INVALIDDATA;
+                                    }
+
+                                    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;
+}
+
+/**
+ * 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, eld_syntax, er_syntax, pulse_present = 0;
+    int ret;
+
+    eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
+    er_syntax  = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
+                 ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
+                 ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
+                 ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
+
+    /* This assignment is to silence a GCC warning about the variable being used
+     * uninitialized when in fact it always is.
+     */
+    pulse.num_pulse = 0;
+
+    global_gain = get_bits(gb, 8);
+
+    if (!common_window && !scale_flag) {
+        if (decode_ics_info(ac, ics, gb) < 0)
+            return AVERROR_INVALIDDATA;
+    }
+
+    if ((ret = decode_band_types(ac, sce->band_type,
+                                 sce->band_type_run_end, gb, ics)) < 0)
+        return ret;
+    if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
+                                  sce->band_type, sce->band_type_run_end)) < 0)
+        return ret;
+
+    pulse_present = 0;
+    if (!scale_flag) {
+        if (!eld_syntax && (pulse_present = get_bits1(gb))) {
+            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+                av_log(ac->avctx, AV_LOG_ERROR,
+                       "Pulse tool not allowed in eight short sequence.\n");
+                return AVERROR_INVALIDDATA;
+            }
+            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
+                av_log(ac->avctx, AV_LOG_ERROR,
+                       "Pulse data corrupt or invalid.\n");
+                return AVERROR_INVALIDDATA;
+            }
+        }
+        tns->present = get_bits1(gb);
+        if (tns->present && !er_syntax)
+            if (decode_tns(ac, tns, gb, ics) < 0)
+                return AVERROR_INVALIDDATA;
+        if (!eld_syntax && get_bits1(gb)) {
+            avpriv_request_sample(ac->avctx, "SSR");
+            return AVERROR_PATCHWELCOME;
+        }
+        // I see no textual basis in the spec for this occurring after SSR gain
+        // control, but this is what both reference and real implmentations do
+        if (tns->present && er_syntax)
+            if (decode_tns(ac, tns, gb, ics) < 0)
+                return AVERROR_INVALIDDATA;
+    }
+
+    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
+                                    &pulse, ics, sce->band_type) < 0)
+        return AVERROR_INVALIDDATA;
+
+    if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
+        apply_prediction(ac, sce);
+
+    return 0;
+}
+
+/**
+ * Mid/Side stereo decoding; reference: 4.6.8.1.3.
+ */
+static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
+{
+    const IndividualChannelStream *ics = &cpe->ch[0].ics;
+    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;
+    int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
+
+    common_window = eld_syntax || get_bits1(gb);
+    if (common_window) {
+        if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
+            return AVERROR_INVALIDDATA;
+        i = cpe->ch[1].ics.use_kb_window[0];
+        cpe->ch[1].ics = cpe->ch[0].ics;
+        cpe->ch[1].ics.use_kb_window[1] = i;
+        if (cpe->ch[1].ics.predictor_present &&
+            (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
+            if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
+                decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
+        ms_present = get_bits(gb, 2);
+        if (ms_present == 3) {
+            av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
+            return AVERROR_INVALIDDATA;
+        } else if (ms_present)
+            decode_mid_side_stereo(cpe, gb, ms_present);
+    }
+    if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
+        return ret;
+    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
+        return ret;
+
+    if (common_window) {
+        if (ms_present)
+            apply_mid_side_stereo(ac, cpe);
+        if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
+            apply_prediction(ac, &cpe->ch[0]);
+            apply_prediction(ac, &cpe->ch[1]);
+        }
+    }
+
+    apply_intensity_stereo(ac, cpe, ms_present);
+    return 0;
+}
+
+static const float cce_scale[] = {
+    1.09050773266525765921, //2^(1/8)
+    1.18920711500272106672, //2^(1/4)
+    M_SQRT2,
+    2,
+};
+
+/**
+ * Decode coupling_channel_element; reference: table 4.8.
+ *
+ * @return  Returns error status. 0 - OK, !0 - error
+ */
+static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
+{
+    int num_gain = 0;
+    int c, g, sfb, ret;
+    int sign;
+    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 = 1.0;
+        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;
+    int type = get_bits(gb, 4);
+
+    if (ac->avctx->debug & FF_DEBUG_STARTCODE)
+        av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
+
+    switch (type) { // extension type
+    case EXT_SBR_DATA_CRC:
+        crc_flag++;
+    case EXT_SBR_DATA:
+        if (!che) {
+            av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
+            return res;
+        } else if (!ac->oc[1].m4ac.sbr) {
+            av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
+            skip_bits_long(gb, 8 * cnt - 4);
+            return res;
+        } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
+            av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
+            skip_bits_long(gb, 8 * cnt - 4);
+            return res;
+        } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
+            ac->oc[1].m4ac.sbr = 1;
+            ac->oc[1].m4ac.ps = 1;
+            ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
+            output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
+                             ac->oc[1].status, 1);
+        } else {
+            ac->oc[1].m4ac.sbr = 1;
+            ac->avctx->profile = FF_PROFILE_AAC_HE;
+        }
+        res = 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));
+    }
+}
+
+static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
+{
+    IndividualChannelStream *ics = &sce->ics;
+    float *in    = sce->coeffs;
+    float *out   = sce->ret;
+    float *saved = sce->saved;
+    float *buf  = ac->buf_mdct;
+
+    // imdct
+    ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
+
+    // window overlapping
+    if (ics->use_kb_window[1]) {
+        // AAC LD uses a low overlap sine window instead of a KBD window
+        memcpy(out, saved, 192 * sizeof(float));
+        ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, ff_sine_128, 64);
+        memcpy(                     out + 320, buf + 64, 192 * sizeof(float));
+    } else {
+        ac->fdsp->vector_fmul_window(out, saved, buf, ff_sine_512, 256);
+    }
+
+    // buffer update
+    memcpy(saved, buf + 256, 256 * sizeof(float));
+}
+
+static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
+{
+    float *in    = sce->coeffs;
+    float *out   = sce->ret;
+    float *saved = sce->saved;
+    float *buf  = ac->buf_mdct;
+    int i;
+    const int n  = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
+    const int n2 = n >> 1;
+    const int n4 = n >> 2;
+    const float *const window = n == 480 ? ff_aac_eld_window_480 :
+                                           ff_aac_eld_window_512;
+
+    // Inverse transform, mapped to the conventional IMDCT by
+    // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
+    // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
+    // International Conference on Audio, Language and Image Processing, ICALIP 2008.
+    // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
+    for (i = 0; i < n2; i+=2) {
+        float temp;
+        temp =  in[i    ]; in[i    ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
+        temp = -in[i + 1]; in[i + 1] =  in[n - 2 - i]; in[n - 2 - i] = temp;
+    }
+    if (n == 480)
+        ac->mdct480->imdct_half(ac->mdct480, buf, in, 1, -1.f/(16*1024*960));
+    else
+        ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
+    for (i = 0; i < n; i+=2) {
+        buf[i] = -buf[i];
+    }
+    // Like with the regular IMDCT at this point we still have the middle half
+    // of a transform but with even symmetry on the left and odd symmetry on
+    // the right
+
+    // window overlapping
+    // The spec says to use samples [0..511] but the reference decoder uses
+    // samples [128..639].
+    for (i = n4; i < n2; i ++) {
+        out[i - n4] =    buf[n2 - 1 - i]       * window[i       - n4] +
+                       saved[      i + n2]     * window[i +   n - n4] +
+                      -saved[  n + n2 - 1 - i] * window[i + 2*n - n4] +
+                      -saved[2*n + n2 + i]     * window[i + 3*n - n4];
+    }
+    for (i = 0; i < n2; i ++) {
+        out[n4 + i] =    buf[i]               * window[i + n2       - n4] +
+                      -saved[      n - 1 - i] * window[i + n2 +   n - n4] +
+                      -saved[  n + i]         * window[i + n2 + 2*n - n4] +
+                       saved[2*n + n - 1 - i] * window[i + n2 + 3*n - n4];
+    }
+    for (i = 0; i < n4; i ++) {
+        out[n2 + n4 + i] =    buf[      i + n2]     * window[i +   n - n4] +
+                           -saved[      n2 - 1 - i] * window[i + 2*n - n4] +
+                           -saved[  n + n2 + i]     * window[i + 3*n - n4];
+    }
+
+    // buffer update
+    memmove(saved + n, saved, 2 * n * sizeof(float));
+    memcpy( saved,       buf,     n * sizeof(float));
+}
+
+/**
+ * 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;
+    void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
+    switch (ac->oc[1].m4ac.object_type) {
+    case AOT_ER_AAC_LD:
+        imdct_and_window = imdct_and_windowing_ld;
+        break;
+    case AOT_ER_AAC_ELD:
+        imdct_and_window = imdct_and_windowing_eld;
+        break;
+    default:
+        imdct_and_window = ac->imdct_and_windowing;
+    }
+    for (type = 3; type >= 0; type--) {
+        for (i = 0; i < MAX_ELEM_ID; i++) {
+            ChannelElement *che = ac->che[type][i];
+            if (che && che->present) {
+                if (type <= TYPE_CPE)
+                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, 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) {
+                    imdct_and_window(ac, &che->ch[0]);
+                    if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
+                        ac->update_ltp(ac, &che->ch[0]);
+                    if (type == TYPE_CPE) {
+                        imdct_and_window(ac, &che->ch[1]);
+                        if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
+                            ac->update_ltp(ac, &che->ch[1]);
+                    }
+                    if (ac->oc[1].m4ac.sbr > 0) {
+                        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);
+                che->present = 0;
+            } else if (che) {
+                av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
+            }
+        }
+    }
+}
+
+static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
+{
+    int size;
+    AACADTSHeaderInfo hdr_info;
+    uint8_t layout_map[MAX_ELEM_ID*4][3];
+    int layout_map_tags, ret;
+
+    size = avpriv_aac_parse_header(gb, &hdr_info);
+    if (size > 0) {
+        if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
+            // This is 2 for "VLB " audio in NSV files.
+            // See samples/nsv/vlb_audio.
+            avpriv_report_missing_feature(ac->avctx,
+                                          "More than one AAC RDB per ADTS frame");
+            ac->warned_num_aac_frames = 1;
+        }
+        push_output_configuration(ac);
+        if (hdr_info.chan_config) {
+            ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
+            if ((ret = set_default_channel_config(ac->avctx,
+                                                  layout_map,
+                                                  &layout_map_tags,
+                                                  hdr_info.chan_config)) < 0)
+                return ret;
+            if ((ret = output_configure(ac, layout_map, layout_map_tags,
+                                        FFMAX(ac->oc[1].status,
+                                              OC_TRIAL_FRAME), 0)) < 0)
+                return ret;
+        } else {
+            ac->oc[1].m4ac.chan_config = 0;
+            /**
+             * dual mono frames in Japanese DTV can have chan_config 0
+             * WITHOUT specifying PCE.
+             *  thus, set dual mono as default.
+             */
+            if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {
+                layout_map_tags = 2;
+                layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
+                layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
+                layout_map[0][1] = 0;
+                layout_map[1][1] = 1;
+                if (output_configure(ac, layout_map, layout_map_tags,
+                                     OC_TRIAL_FRAME, 0))
+                    return -7;
+            }
+        }
+        ac->oc[1].m4ac.sample_rate     = hdr_info.sample_rate;
+        ac->oc[1].m4ac.sampling_index  = hdr_info.sampling_index;
+        ac->oc[1].m4ac.object_type     = hdr_info.object_type;
+        ac->oc[1].m4ac.frame_length_short = 0;
+        if (ac->oc[0].status != OC_LOCKED ||
+            ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
+            ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
+            ac->oc[1].m4ac.sbr = -1;
+            ac->oc[1].m4ac.ps  = -1;
+        }
+        if (!hdr_info.crc_absent)
+            skip_bits(gb, 16);
+    }
+    return size;
+}
+
+static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
+                               int *got_frame_ptr, GetBitContext *gb)
+{
+    AACContext *ac = avctx->priv_data;
+    const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
+    ChannelElement *che;
+    int err, i;
+    int samples = m4ac->frame_length_short ? 960 : 1024;
+    int chan_config = m4ac->chan_config;
+    int aot = m4ac->object_type;
+
+    if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
+        samples >>= 1;
+
+    ac->frame = data;
+
+    if ((err = frame_configure_elements(avctx)) < 0)
+        return err;
+
+    // The FF_PROFILE_AAC_* defines are all object_type - 1
+    // This may lead to an undefined profile being signaled
+    ac->avctx->profile = aot - 1;
+
+    ac->tags_mapped = 0;
+
+    if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
+        avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
+                              chan_config);
+        return AVERROR_INVALIDDATA;
+    }
+    for (i = 0; i < tags_per_config[chan_config]; i++) {
+        const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
+        const int elem_id   = aac_channel_layout_map[chan_config-1][i][1];
+        if (!(che=get_che(ac, elem_type, elem_id))) {
+            av_log(ac->avctx, AV_LOG_ERROR,
+                   "channel element %d.%d is not allocated\n",
+                   elem_type, elem_id);
+            return AVERROR_INVALIDDATA;
+        }
+        che->present = 1;
+        if (aot != AOT_ER_AAC_ELD)
+            skip_bits(gb, 4);
+        switch (elem_type) {
+        case TYPE_SCE:
+            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+            break;
+        case TYPE_CPE:
+            err = decode_cpe(ac, gb, che);
+            break;
+        case TYPE_LFE:
+            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+            break;
+        }
+        if (err < 0)
+            return err;
+    }
+
+    spectral_to_sample(ac);
+
+    ac->frame->nb_samples = samples;
+    ac->frame->sample_rate = avctx->sample_rate;
+    *got_frame_ptr = 1;
+
+    skip_bits_long(gb, get_bits_left(gb));
+    return 0;
+}
+
+static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
+                                int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
+{
+    AACContext *ac = avctx->priv_data;
+    ChannelElement *che = NULL, *che_prev = NULL;
+    enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
+    int err, elem_id;
+    int samples = 0, multiplier, audio_found = 0, pce_found = 0;
+    int is_dmono, sce_count = 0;
+
+    ac->frame = data;
+
+    if (show_bits(gb, 12) == 0xfff) {
+        if ((err = parse_adts_frame_header(ac, gb)) < 0) {
+            av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
+            goto fail;
+        }
+        if (ac->oc[1].m4ac.sampling_index > 12) {
+            av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
+            err = AVERROR_INVALIDDATA;
+            goto fail;
+        }
+    }
+
+    if ((err = frame_configure_elements(avctx)) < 0)
+        goto fail;
+
+    // The FF_PROFILE_AAC_* defines are all object_type - 1
+    // This may lead to an undefined profile being signaled
+    ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
+
+    ac->tags_mapped = 0;
+    // parse
+    while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
+        elem_id = get_bits(gb, 4);
+
+        if (avctx->debug & FF_DEBUG_STARTCODE)
+            av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
+
+        if (!avctx->channels && elem_type != TYPE_PCE) {
+            err = AVERROR_INVALIDDATA;
+            goto fail;
+        }
+
+        if (elem_type < TYPE_DSE) {
+            if (!(che=get_che(ac, elem_type, elem_id))) {
+                av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
+                       elem_type, elem_id);
+                err = AVERROR_INVALIDDATA;
+                goto fail;
+            }
+            samples = 1024;
+            che->present = 1;
+        }
+
+        switch (elem_type) {
+
+        case TYPE_SCE:
+            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+            audio_found = 1;
+            sce_count++;
+            break;
+
+        case TYPE_CPE:
+            err = decode_cpe(ac, gb, che);
+            audio_found = 1;
+            break;
+
+        case TYPE_CCE:
+            err = decode_cce(ac, gb, che);
+            break;
+
+        case TYPE_LFE:
+            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+            audio_found = 1;
+            break;
+
+        case TYPE_DSE:
+            err = skip_data_stream_element(ac, gb);
+            break;
+
+        case TYPE_PCE: {
+            uint8_t layout_map[MAX_ELEM_ID*4][3];
+            int tags;
+            push_output_configuration(ac);
+            tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb);
+            if (tags < 0) {
+                err = tags;
+                break;
+            }
+            if (pce_found) {
+                av_log(avctx, AV_LOG_ERROR,
+                       "Not evaluating a further program_config_element as this construct is dubious at best.\n");
+            } else {
+                err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
+                if (!err)
+                    ac->oc[1].m4ac.chan_config = 0;
+                pce_found = 1;
+            }
+            break;
+        }
+
+        case TYPE_FIL:
+            if (elem_id == 15)
+                elem_id += get_bits(gb, 8) - 1;
+            if (get_bits_left(gb) < 8 * elem_id) {
+                    av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
+                    err = AVERROR_INVALIDDATA;
+                    goto fail;
+            }
+            while (elem_id > 0)
+                elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
+            err = 0; /* FIXME */
+            break;
+
+        default:
+            err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
+            break;
+        }
+
+        che_prev       = che;
+        elem_type_prev = elem_type;
+
+        if (err)
+            goto fail;
+
+        if (get_bits_left(gb) < 3) {
+            av_log(avctx, AV_LOG_ERROR, overread_err);
+            err = AVERROR_INVALIDDATA;
+            goto fail;
+        }
+    }
+
+    if (!avctx->channels) {
+        *got_frame_ptr = 0;
+        return 0;
+    }
+
+    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;
+
+    if (ac->oc[1].status && audio_found) {
+        avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
+        avctx->frame_size = samples;
+        ac->oc[1].status = OC_LOCKED;
+    }
+
+    if (multiplier) {
+        int side_size;
+        const uint8_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);
+        if (side && side_size>=4)
+            AV_WL32(side, 2*AV_RL32(side));
+    }
+
+    if (!ac->frame->data[0] && samples) {
+        av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
+        err = AVERROR_INVALIDDATA;
+        goto fail;
+    }
+
+    if (samples) {
+        ac->frame->nb_samples = samples;
+        ac->frame->sample_rate = avctx->sample_rate;
+    } else
+        av_frame_unref(ac->frame);
+    *got_frame_ptr = !!samples;
+
+    /* for dual-mono audio (SCE + SCE) */
+    is_dmono = ac->dmono_mode && sce_count == 2 &&
+               ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
+    if (is_dmono) {
+        if (ac->dmono_mode == 1)
+            ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
+        else if (ac->dmono_mode == 2)
+            ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
+    }
+
+    return 0;
+fail:
+    pop_output_configuration(ac);
+    return err;
+}
+
+static int aac_decode_frame(AVCodecContext *avctx, void *data,
+                            int *got_frame_ptr, AVPacket *avpkt)
+{
+    AACContext *ac = avctx->priv_data;
+    const uint8_t *buf = avpkt->data;
+    int buf_size = avpkt->size;
+    GetBitContext gb;
+    int buf_consumed;
+    int buf_offset;
+    int err;
+    int new_extradata_size;
+    const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
+                                       AV_PKT_DATA_NEW_EXTRADATA,
+                                       &new_extradata_size);
+    int jp_dualmono_size;
+    const uint8_t *jp_dualmono   = av_packet_get_side_data(avpkt,
+                                       AV_PKT_DATA_JP_DUALMONO,
+                                       &jp_dualmono_size);
+
+    if (new_extradata && 0) {
+        av_free(avctx->extradata);
+        avctx->extradata = av_mallocz(new_extradata_size +
+                                      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;
+
+    if ((err = init_get_bits(&gb, buf, buf_size * 8)) < 0)
+        return err;
+
+    switch (ac->oc[1].m4ac.object_type) {
+    case AOT_ER_AAC_LC:
+    case AOT_ER_AAC_LTP:
+    case AOT_ER_AAC_LD:
+    case AOT_ER_AAC_ELD:
+        err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
+        break;
+    default:
+        err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
+    }
+    if (err < 0)
+        return err;
+
+    buf_consumed = (get_bits_count(&gb) + 7) >> 3;
+    for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
+        if (buf[buf_offset])
+            break;
+
+    return buf_size > buf_offset ? buf_consumed : buf_size;
+}
+
+static av_cold int aac_decode_close(AVCodecContext *avctx)
+{
+    AACContext *ac = avctx->priv_data;
+    int i, type;
+
+    for (i = 0; i < MAX_ELEM_ID; i++) {
+        for (type = 0; type < 4; type++) {
+            if (ac->che[type][i])
+                ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
+            av_freep(&ac->che[type][i]);
+        }
+    }
+
+    ff_mdct_end(&ac->mdct);
+    ff_mdct_end(&ac->mdct_small);
+    ff_mdct_end(&ac->mdct_ld);
+    ff_mdct_end(&ac->mdct_ltp);
+    ff_imdct15_uninit(&ac->mdct480);
+    av_freep(&ac->fdsp);
+    return 0;
+}
+
+static void aacdec_init(AACContext *c)
+{
+    c->imdct_and_windowing                      = imdct_and_windowing;
+    c->apply_ltp                                = apply_ltp;
+    c->apply_tns                                = apply_tns;
+    c->windowing_and_mdct_ltp                   = windowing_and_mdct_ltp;
+    c->update_ltp                               = update_ltp;
+
+    if(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,
+};
+
+static const AVProfile profiles[] = {
+    { FF_PROFILE_AAC_MAIN,  "Main"     },
+    { FF_PROFILE_AAC_LOW,   "LC"       },
+    { FF_PROFILE_AAC_SSR,   "SSR"      },
+    { FF_PROFILE_AAC_LTP,   "LTP"      },
+    { FF_PROFILE_AAC_HE,    "HE-AAC"   },
+    { FF_PROFILE_AAC_HE_V2, "HE-AACv2" },
+    { FF_PROFILE_AAC_LD,    "LD"       },
+    { FF_PROFILE_AAC_ELD,   "ELD"      },
+    { FF_PROFILE_UNKNOWN },
+};
index e0a8e63..59918ae 100644 (file)
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
-
-#include <stdlib.h>
-#define CONFIG_HARDCODED_TABLES 0
-#include "cbrt_tablegen.h"
-#include "tableprint.h"
-
-int main(void)
-{
-    cbrt_tableinit();
-
-    write_fileheader();
-
-    WRITE_ARRAY("static const", uint32_t, cbrt_tab);
-
-    return 0;
-}
diff --git a/libavcodec/cbrt_tablegen_template.c b/libavcodec/cbrt_tablegen_template.c
new file mode 100644 (file)
index 0000000..e0a8e63
--- /dev/null
@@ -0,0 +1,37 @@
+/*
+ * Generate a header file for hardcoded AAC cube-root table
+ *
+ * Copyright (c) 2010 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
+ *
+ * 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
+ */
+
+#include <stdlib.h>
+#define CONFIG_HARDCODED_TABLES 0
+#include "cbrt_tablegen.h"
+#include "tableprint.h"
+
+int main(void)
+{
+    cbrt_tableinit();
+
+    write_fileheader();
+
+    WRITE_ARRAY("static const", uint32_t, cbrt_tab);
+
+    return 0;
+}
index 561ae3e..2013b95 100644 (file)
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
-
-#include <stdlib.h>
-#define CONFIG_HARDCODED_TABLES 0
-#define SINETABLE_CONST
-#define SINETABLE(size) \
-    float ff_sine_##size[size]
-#define FF_ARRAY_ELEMS(a) (sizeof(a) / sizeof((a)[0]))
-#include "sinewin_tablegen.h"
-#include "tableprint.h"
-
-int main(void)
-{
-    int i;
-
-    write_fileheader();
-
-    for (i = 5; i <= 13; i++) {
-        ff_init_ff_sine_windows(i);
-        printf("SINETABLE(%4i) = {\n", 1 << i);
-        write_float_array(ff_sine_windows[i], 1 << i);
-        printf("};\n");
-    }
-
-    return 0;
-}
diff --git a/libavcodec/sinewin_tablegen_template.c b/libavcodec/sinewin_tablegen_template.c
new file mode 100644 (file)
index 0000000..561ae3e
--- /dev/null
@@ -0,0 +1,46 @@
+/*
+ * Generate a header file for hardcoded sine windows
+ *
+ * Copyright (c) 2009 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
+ *
+ * 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
+ */
+
+#include <stdlib.h>
+#define CONFIG_HARDCODED_TABLES 0
+#define SINETABLE_CONST
+#define SINETABLE(size) \
+    float ff_sine_##size[size]
+#define FF_ARRAY_ELEMS(a) (sizeof(a) / sizeof((a)[0]))
+#include "sinewin_tablegen.h"
+#include "tableprint.h"
+
+int main(void)
+{
+    int i;
+
+    write_fileheader();
+
+    for (i = 5; i <= 13; i++) {
+        ff_init_ff_sine_windows(i);
+        printf("SINETABLE(%4i) = {\n", 1 << i);
+        write_float_array(ff_sine_windows[i], 1 << i);
+        printf("};\n");
+    }
+
+    return 0;
+}