]> git.sesse.net Git - ffmpeg/blobdiff - libavcodec/aacdec.c
vp9: split superframes in the filtering stage before actual decoding
[ffmpeg] / libavcodec / aacdec.c
index ca69e77e29667147403236fab8f5ebbc4c770555..22ebcdc5705699f130edbf024763785c896cd43a 100644 (file)
@@ -2,10 +2,11 @@
  * 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-ffmpeg@jannau.net>
+ * Copyright (c) 2010      Janne Grunau <janne-libav@jannau.net>
  *
  * This file is part of Libav.
  *
            Parametric Stereo.
  */
 
-
+#include "libavutil/float_dsp.h"
 #include "avcodec.h"
 #include "internal.h"
 #include "get_bits.h"
-#include "dsputil.h"
 #include "fft.h"
-#include "fmtconvert.h"
+#include "imdct15.h"
 #include "lpc.h"
 #include "kbdwin.h"
 #include "sinewin.h"
 #include "aacsbr.h"
 #include "mpeg4audio.h"
 #include "aacadtsdec.h"
+#include "libavutil/intfloat.h"
 
 #include <assert.h>
 #include <errno.h>
 #include <math.h>
+#include <stdint.h>
 #include <string.h>
 
 #if ARCH_ARM
 #   include "arm/aac.h"
 #endif
 
-union float754 {
-    float f;
-    uint32_t i;
-};
+#include "libavutil/thread.h"
 
 static VLC vlc_scalefactors;
 static VLC vlc_spectral[11];
 
 static const char overread_err[] = "Input buffer exhausted before END element found\n";
 
-static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
+static int count_channels(uint8_t (*layout)[3], int tags)
 {
-    // For PCE based channel configurations map the channels solely based on tags.
-    if (!ac->m4ac.chan_config) {
-        return ac->tag_che_map[type][elem_id];
-    }
-    // For indexed channel configurations map the channels solely based on position.
-    switch (ac->m4ac.chan_config) {
-    case 7:
-        if (ac->tags_mapped == 3 && type == TYPE_CPE) {
-            ac->tags_mapped++;
-            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
-        }
-    case 6:
-        /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
-           instead of SCE[0] CPE[0] CPE[1] LFE[0]. If we seem to have
-           encountered such a stream, transfer the LFE[0] element to the SCE[1]'s mapping */
-        if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
-            ac->tags_mapped++;
-            return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
-        }
-    case 5:
-        if (ac->tags_mapped == 2 && type == TYPE_CPE) {
-            ac->tags_mapped++;
-            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
-        }
-    case 4:
-        if (ac->tags_mapped == 2 && ac->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->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->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;
-    }
+    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;
 }
 
 /**
@@ -180,20 +142,22 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
  * @return  Returns error status. 0 - OK, !0 - error
  */
 static av_cold int che_configure(AACContext *ac,
-                                 enum ChannelPosition che_pos[4][MAX_ELEM_ID],
+                                 enum ChannelPosition che_pos,
                                  int type, int id, int *channels)
 {
-    if (che_pos[type][id]) {
+    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) {
-            ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
+            if (*channels >= MAX_CHANNELS - 2)
+                return AVERROR_INVALIDDATA;
+            ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];
             if (type == TYPE_CPE ||
-                (type == TYPE_SCE && ac->m4ac.ps == 1)) {
-                ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
+                (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
+                ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];
             }
         }
     } else {
@@ -204,103 +168,517 @@ static av_cold int che_configure(AACContext *ac,
     return 0;
 }
 
-/**
- * Configure output channel order based on the current program configuration element.
- *
- * @param   che_pos current channel position configuration
- * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
- *
- * @return  Returns error status. 0 - OK, !0 - error
- */
-static av_cold int output_configure(AACContext *ac,
-                                    enum ChannelPosition che_pos[4][MAX_ELEM_ID],
-                                    enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
-                                    int channel_config, enum OCStatus oc_type)
+static int frame_configure_elements(AVCodecContext *avctx)
 {
-    AVCodecContext *avctx = ac->avctx;
-    int i, type, channels = 0, ret;
-
-    if (new_che_pos != che_pos)
-    memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
+    AACContext *ac = avctx->priv_data;
+    int type, id, ch, ret;
 
-    if (channel_config) {
-        for (i = 0; i < tags_per_config[channel_config]; i++) {
-            if ((ret = che_configure(ac, che_pos,
-                                     aac_channel_layout_map[channel_config - 1][i][0],
-                                     aac_channel_layout_map[channel_config - 1][i][1],
-                                     &channels)))
-                return ret;
+    /* set channel pointers to internal buffers by default */
+    for (type = 0; type < 4; type++) {
+        for (id = 0; id < MAX_ELEM_ID; id++) {
+            ChannelElement *che = ac->che[type][id];
+            if (che) {
+                che->ch[0].ret = che->ch[0].ret_buf;
+                che->ch[1].ret = che->ch[1].ret_buf;
+            }
         }
+    }
+
+    /* get output buffer */
+    av_frame_unref(ac->frame);
+    ac->frame->nb_samples = 2048;
+    if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+        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;
+}
 
-        memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
+struct elem_to_channel {
+    uint64_t av_position;
+    uint8_t syn_ele;
+    uint8_t elem_id;
+    uint8_t aac_position;
+};
 
-        avctx->channel_layout = aac_channel_layout[channel_config - 1];
+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 {
-        /* Allocate or free elements depending on if they are in the
-         * current program configuration.
-         *
-         * Set up default 1:1 output mapping.
-         *
-         * For a 5.1 stream the output order will be:
-         *    [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
-         */
+        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;
+    }
+}
 
-        for (i = 0; i < MAX_ELEM_ID; i++) {
-            for (type = 0; type < 4; type++) {
-                if ((ret = che_configure(ac, che_pos, type, i, &channels)))
-                    return ret;
+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;
         }
+    }
 
-        memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
+    return layout;
+}
 
-        avctx->channel_layout = 0;
+/**
+ * Save current output configuration if and only if it has been locked.
+ */
+static void push_output_configuration(AACContext *ac) {
+    if (ac->oc[1].status == OC_LOCKED) {
+        ac->oc[0] = ac->oc[1];
     }
+    ac->oc[1].status = OC_NONE;
+}
 
-    avctx->channels = channels;
+/**
+ * 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;
+    }
+}
 
-    ac->output_configured = oc_type;
+/**
+ * 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;
+        }
+    }
+
+    avctx->channel_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;
 }
 
 /**
- * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
+ * 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));
+    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);
+
+        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);
+
+        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)) {
+            ac->tags_mapped++;
+            return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
+        }
+    case 5:
+        if (ac->tags_mapped == 2 && type == TYPE_CPE) {
+            ac->tags_mapped++;
+            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
+        }
+    case 4:
+        if (ac->tags_mapped == 2 &&
+            ac->oc[1].m4ac.chan_config == 4 &&
+            type == TYPE_SCE) {
+            ac->tags_mapped++;
+            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
+        }
+    case 3:
+    case 2:
+        if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
+            type == TYPE_CPE) {
+            ac->tags_mapped++;
+            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
+        } else if (ac->oc[1].m4ac.chan_config == 2) {
+            return NULL;
+        }
+    case 1:
+        if (!ac->tags_mapped && type == TYPE_SCE) {
+            ac->tags_mapped++;
+            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
+        }
+    default:
+        return NULL;
+    }
+}
+
+/**
+ * Decode an array of 4 bit element IDs, optionally interleaved with a
+ * stereo/mono switching bit.
  *
- * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
- * @param sce_map mono (Single Channel Element) map
  * @param type speaker type/position for these channels
  */
-static void decode_channel_map(enum ChannelPosition *cpe_map,
-                               enum ChannelPosition *sce_map,
+static void decode_channel_map(uint8_t layout_map[][3],
                                enum ChannelPosition type,
                                GetBitContext *gb, int n)
 {
     while (n--) {
-        enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
-        map[get_bits(gb, 4)] = type;
+        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
+            return;
+        }
+        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.
  *
- * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
- *
  * @return  Returns error status. 0 - OK, !0 - error
  */
 static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
-                      enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
+                      uint8_t (*layout_map)[3],
                       GetBitContext *gb)
 {
-    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
+    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");
+        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);
@@ -317,14 +695,19 @@ static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
     if (get_bits1(gb))
         skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
 
-    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
-    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
-    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK,  gb, num_back );
-    decode_channel_map(NULL,                  new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
+    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(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
+    decode_channel_map(layout_map + tags, AAC_CHANNEL_CC,    gb, num_cc);
+    tags += num_cc;
 
     align_get_bits(gb);
 
@@ -332,56 +715,10 @@ static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
     comment_len = get_bits(gb, 8) * 8;
     if (get_bits_left(gb) < comment_len) {
         av_log(avctx, AV_LOG_ERROR, overread_err);
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
     skip_bits_long(gb, comment_len);
-    return 0;
-}
-
-/**
- * Set up channel positions based on a default channel configuration
- * as specified in table 1.17.
- *
- * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
- *
- * @return  Returns error status. 0 - OK, !0 - error
- */
-static av_cold int set_default_channel_config(AVCodecContext *avctx,
-                                              enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
-                                              int channel_config)
-{
-    if (channel_config < 1 || channel_config > 7) {
-        av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
-               channel_config);
-        return -1;
-    }
-
-    /* default channel configurations:
-     *
-     * 1ch : front center (mono)
-     * 2ch : L + R (stereo)
-     * 3ch : front center + L + R
-     * 4ch : front center + L + R + back center
-     * 5ch : front center + L + R + back stereo
-     * 6ch : front center + L + R + back stereo + LFE
-     * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
-     */
-
-    if (channel_config != 2)
-        new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
-    if (channel_config > 1)
-        new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
-    if (channel_config == 4)
-        new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK;  // back center
-    if (channel_config > 4)
-        new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
-        = AAC_CHANNEL_BACK;  // back stereo
-    if (channel_config > 5)
-        new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE;   // LFE
-    if (channel_config == 7)
-        new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
-
-    return 0;
+    return tags;
 }
 
 /**
@@ -397,13 +734,15 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
                                      MPEG4AudioConfig *m4ac,
                                      int channel_config)
 {
-    enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
-    int extension_flag, ret;
+    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
-        av_log_missing_feature(avctx, "960/120 MDCT window is", 1);
-        return -1;
+        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
@@ -413,16 +752,23 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
         m4ac->object_type == AOT_ER_AAC_SCALABLE)
         skip_bits(gb, 3);     // layerNr
 
-    memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
     if (channel_config == 0) {
         skip_bits(gb, 4);  // element_instance_tag
-        if ((ret = decode_pce(avctx, m4ac, new_che_pos, gb)))
-            return ret;
+        tags = decode_pce(avctx, m4ac, layout_map, gb);
+        if (tags < 0)
+            return tags;
     } else {
-        if ((ret = set_default_channel_config(avctx, new_che_pos, channel_config)))
+        if ((ret = set_default_channel_config(avctx, layout_map,
+                                              &tags, channel_config)))
             return ret;
     }
-    if (ac && (ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
+
+    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) {
@@ -435,14 +781,86 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
         case AOT_ER_AAC_LTP:
         case AOT_ER_AAC_SCALABLE:
         case AOT_ER_AAC_LD:
-            skip_bits(gb, 3);  /* aacSectionDataResilienceFlag
-                                    * aacScalefactorDataResilienceFlag
-                                    * aacSpectralDataResilienceFlag
-                                    */
+            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;
 }
 
@@ -452,34 +870,45 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
  * @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 AVCodecContext extradata
- * @param   data_size   size of AVCCodecContext extradata
+ * @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 data_size)
+                                        const uint8_t *data, int bit_size,
+                                        int sync_extension)
 {
     GetBitContext gb;
-    int i;
+    int i, ret;
 
-    av_dlog(avctx, "extradata size %d\n", avctx->extradata_size);
+    ff_dlog(avctx, "extradata size %d\n", avctx->extradata_size);
     for (i = 0; i < avctx->extradata_size; i++)
-         av_dlog(avctx, "%02x ", avctx->extradata[i]);
-    av_dlog(avctx, "\n");
+        ff_dlog(avctx, "%02x ", avctx->extradata[i]);
+    ff_dlog(avctx, "\n");
 
-    init_get_bits(&gb, data, data_size * 8);
+    if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
+        return ret;
 
-    if ((i = avpriv_mpeg4audio_get_config(m4ac, data, data_size)) < 0)
-        return -1;
+    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 -1;
+        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;
     }
-    if (m4ac->sbr == 1 && m4ac->ps == -1)
-        m4ac->ps = 1;
 
     skip_bits_long(&gb, i);
 
@@ -487,18 +916,30 @@ static int decode_audio_specific_config(AACContext *ac,
     case AOT_AAC_MAIN:
     case AOT_AAC_LC:
     case AOT_AAC_LTP:
-        if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config))
-            return -1;
+    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:
-        av_log(avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
-               m4ac->sbr == 1? "SBR+" : "", m4ac->object_type);
-        return -1;
+        avpriv_report_missing_feature(avctx,
+                                      "Audio object type %s%d",
+                                      m4ac->sbr == 1 ? "SBR+" : "",
+                                      m4ac->object_type);
+        return AVERROR(ENOSYS);
     }
 
-    av_dlog(avctx, "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
+    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);
+            m4ac->sample_rate, m4ac->sbr,
+            m4ac->ps);
 
     return get_bits_count(&gb);
 }
@@ -512,7 +953,8 @@ static int decode_audio_specific_config(AACContext *ac,
  */
 static av_always_inline int lcg_random(int previous_val)
 {
-    return previous_val * 1664525 + 1013904223;
+    union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
+    return v.s;
 }
 
 static av_always_inline void reset_predict_state(PredictorState *ps)
@@ -555,34 +997,85 @@ static void reset_predictor_group(PredictorState *ps, int group_num)
         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]), \
+#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 av_cold void aac_static_table_init(void)
+{
+    AAC_INIT_VLC_STATIC( 0, 304);
+    AAC_INIT_VLC_STATIC( 1, 270);
+    AAC_INIT_VLC_STATIC( 2, 550);
+    AAC_INIT_VLC_STATIC( 3, 300);
+    AAC_INIT_VLC_STATIC( 4, 328);
+    AAC_INIT_VLC_STATIC( 5, 294);
+    AAC_INIT_VLC_STATIC( 6, 306);
+    AAC_INIT_VLC_STATIC( 7, 268);
+    AAC_INIT_VLC_STATIC( 8, 510);
+    AAC_INIT_VLC_STATIC( 9, 366);
+    AAC_INIT_VLC_STATIC(10, 462);
+
+    ff_aac_sbr_init();
+
+    ff_aac_tableinit();
+
+    INIT_VLC_STATIC(&vlc_scalefactors, 7,
+                    FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
+                    ff_aac_scalefactor_bits,
+                    sizeof(ff_aac_scalefactor_bits[0]),
+                    sizeof(ff_aac_scalefactor_bits[0]),
+                    ff_aac_scalefactor_code,
+                    sizeof(ff_aac_scalefactor_code[0]),
+                    sizeof(ff_aac_scalefactor_code[0]),
+                    352);
+
+
+    // window initialization
+    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();
+}
+
+static AVOnce aac_init = AV_ONCE_INIT;
+
 static av_cold int aac_decode_init(AVCodecContext *avctx)
 {
     AACContext *ac = avctx->priv_data;
-    float output_scale_factor;
+    int ret;
+
+    ret = ff_thread_once(&aac_init, &aac_static_table_init);
+    if (ret != 0)
+        return AVERROR_UNKNOWN;
 
     ac->avctx = avctx;
-    ac->m4ac.sample_rate = avctx->sample_rate;
+    ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
+
+    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
 
     if (avctx->extradata_size > 0) {
-        if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
-                                         avctx->extradata,
-                                         avctx->extradata_size) < 0)
-            return -1;
+        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;
-        enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
+        uint8_t layout_map[MAX_ELEM_ID*4][3];
+        int layout_map_tags;
 
         sr = sample_rate_idx(avctx->sample_rate);
-        ac->m4ac.sampling_index = sr;
-        ac->m4ac.channels = avctx->channels;
-        ac->m4ac.sbr = -1;
-        ac->m4ac.ps = -1;
+        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)
@@ -590,61 +1083,30 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
         if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
             i = 0;
         }
-        ac->m4ac.chan_config = i;
+        ac->oc[1].m4ac.chan_config = i;
 
-        if (ac->m4ac.chan_config) {
-            int ret = set_default_channel_config(avctx, new_che_pos, ac->m4ac.chan_config);
+        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, ac->che_pos, new_che_pos, ac->m4ac.chan_config, OC_GLOBAL_HDR);
+                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->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
-        avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
-        output_scale_factor = 1.0 / 32768.0;
-    } else {
-        avctx->sample_fmt = AV_SAMPLE_FMT_S16;
-        output_scale_factor = 1.0;
-    }
-
-    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();
-
-    dsputil_init(&ac->dsp, avctx);
-    ff_fmt_convert_init(&ac->fmt_conv, avctx);
+    avpriv_float_dsp_init(&ac->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
 
     ac->random_state = 0x1f2e3d4c;
 
-    ff_aac_tableinit();
-
-    INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
-                    ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
-                    ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
-                    352);
-
-    ff_mdct_init(&ac->mdct,       11, 1, output_scale_factor/1024.0);
-    ff_mdct_init(&ac->mdct_small,  8, 1, output_scale_factor/128.0);
-    ff_mdct_init(&ac->mdct_ltp,   11, 0, -2.0/output_scale_factor);
-    // window initialization
-    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
-    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
-    ff_init_ff_sine_windows(10);
-    ff_init_ff_sine_windows( 7);
-
-    cbrt_tableinit();
+    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;
 
     return 0;
 }
@@ -663,7 +1125,7 @@ static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
 
     if (get_bits_left(gb) < 8 * count) {
         av_log(ac->avctx, AV_LOG_ERROR, overread_err);
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
     skip_bits_long(gb, 8 * count);
     return 0;
@@ -675,12 +1137,14 @@ static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
     int sfb;
     if (get_bits1(gb)) {
         ics->predictor_reset_group = get_bits(gb, 5);
-        if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
-            av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
-            return -1;
+        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->m4ac.sampling_index]); sfb++) {
+    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;
@@ -689,7 +1153,7 @@ static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
 /**
  * Decode Long Term Prediction data; reference: table 4.xx.
  */
-static void decode_ltp(AACContext *ac, LongTermPrediction *ltp,
+static void decode_ltp(LongTermPrediction *ltp,
                        GetBitContext *gb, uint8_t max_sfb)
 {
     int sfb;
@@ -702,21 +1166,32 @@ static void decode_ltp(AACContext *ac, LongTermPrediction *ltp,
 
 /**
  * Decode Individual Channel Stream info; reference: table 4.6.
- *
- * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
  */
 static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
-                           GetBitContext *gb, int common_window)
+                           GetBitContext *gb)
 {
-    if (get_bits1(gb)) {
-        av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
-        memset(ics, 0, sizeof(IndividualChannelStream));
-        return -1;
+    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->window_sequence[1] = ics->window_sequence[0];
-    ics->window_sequence[0] = get_bits(gb, 2);
-    ics->use_kb_window[1]   = ics->use_kb_window[0];
-    ics->use_kb_window[0]   = get_bits1(gb);
     ics->num_window_groups  = 1;
     ics->group_len[0]       = 1;
     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
@@ -731,41 +1206,61 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
             }
         }
         ics->num_windows       = 8;
-        ics->swb_offset        =    ff_swb_offset_128[ac->m4ac.sampling_index];
-        ics->num_swb           =   ff_aac_num_swb_128[ac->m4ac.sampling_index];
-        ics->tns_max_bands     = ff_tns_max_bands_128[ac->m4ac.sampling_index];
+        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;
-        ics->swb_offset            =    ff_swb_offset_1024[ac->m4ac.sampling_index];
-        ics->num_swb               =   ff_aac_num_swb_1024[ac->m4ac.sampling_index];
-        ics->tns_max_bands         = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
-        ics->predictor_present     = get_bits1(gb);
-        ics->predictor_reset_group = 0;
+        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 (ac->m4ac.object_type == AOT_AAC_MAIN) {
+            if (aot == AOT_AAC_MAIN) {
                 if (decode_prediction(ac, ics, gb)) {
-                    memset(ics, 0, sizeof(IndividualChannelStream));
-                    return -1;
+                    return AVERROR_INVALIDDATA;
                 }
-            } else if (ac->m4ac.object_type == AOT_AAC_LC) {
-                av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
-                memset(ics, 0, sizeof(IndividualChannelStream));
-                return -1;
+            } 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");
+                return AVERROR_INVALIDDATA;
             } else {
+                if (aot == AOT_ER_AAC_LD) {
+                    avpriv_report_missing_feature(ac->avctx, "LTP in ER AAC LD");
+                    return AVERROR_PATCHWELCOME;
+                }
                 if ((ics->ltp.present = get_bits(gb, 1)))
-                    decode_ltp(ac, &ics->ltp, gb, ics->max_sfb);
+                    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",
+               "Number of scalefactor bands in group (%d) "
+               "exceeds limit (%d).\n",
                ics->max_sfb, ics->num_swb);
-        memset(ics, 0, sizeof(IndividualChannelStream));
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
     return 0;
@@ -793,21 +1288,22 @@ static int decode_band_types(AACContext *ac, enum BandType band_type[120],
             int sect_band_type = get_bits(gb, 4);
             if (sect_band_type == 12) {
                 av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
-                return -1;
+                return AVERROR_INVALIDDATA;
             }
-            while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
+            do {
+                sect_len_incr = get_bits(gb, bits);
                 sect_end += sect_len_incr;
-            sect_end += sect_len_incr;
-            if (get_bits_left(gb) < 0) {
-                av_log(ac->avctx, AV_LOG_ERROR, overread_err);
-                return -1;
-            }
-            if (sect_end > ics->max_sfb) {
-                av_log(ac->avctx, AV_LOG_ERROR,
-                       "Number of bands (%d) exceeds limit (%d).\n",
-                       sect_end, ics->max_sfb);
-                return -1;
-            }
+                if (get_bits_left(gb) < 0) {
+                    av_log(ac->avctx, AV_LOG_ERROR, 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;
@@ -837,22 +1333,22 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
     int offset[3] = { global_gain, global_gain - 90, 0 };
     int clipped_offset;
     int noise_flag = 1;
-    static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
     for (g = 0; g < ics->num_window_groups; g++) {
         for (i = 0; i < ics->max_sfb;) {
             int run_end = band_type_run_end[idx];
             if (band_type[idx] == ZERO_BT) {
                 for (; i < run_end; i++, idx++)
-                    sf[idx] = 0.;
-            } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
+                    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) - 60;
                     clipped_offset = av_clip(offset[2], -155, 100);
                     if (offset[2] != clipped_offset) {
-                        av_log_ask_for_sample(ac->avctx, "Intensity stereo "
-                                "position clipped (%d -> %d).\nIf you heard an "
-                                "audible artifact, there may be a bug in the "
-                                "decoder. ", 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];
                 }
@@ -864,10 +1360,10 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
                         offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
                     clipped_offset = av_clip(offset[1], -100, 155);
                     if (offset[1] != clipped_offset) {
-                        av_log_ask_for_sample(ac->avctx, "Noise gain clipped "
-                                "(%d -> %d).\nIf you heard an audible "
-                                "artifact, there may be a bug in the decoder. ",
-                                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];
                 }
@@ -876,8 +1372,8 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
                     offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
                     if (offset[0] > 255U) {
                         av_log(ac->avctx, AV_LOG_ERROR,
-                               "%s (%d) out of range.\n", sf_str[0], offset[0]);
-                        return -1;
+                               "Scalefactor (%d) out of range.\n", offset[0]);
+                        return AVERROR_INVALIDDATA;
                     }
                     sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
                 }
@@ -922,7 +1418,7 @@ static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
 {
     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->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
+    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);
@@ -932,10 +1428,11 @@ static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
                 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",
+                    av_log(ac->avctx, AV_LOG_ERROR,
+                           "TNS filter order %d is greater than maximum %d.\n",
                            tns->order[w][filt], tns_max_order);
                     tns->order[w][filt] = 0;
-                    return -1;
+                    return AVERROR_INVALIDDATA;
                 }
                 if (tns->order[w][filt]) {
                     tns->direction[w][filt] = get_bits1(gb);
@@ -963,11 +1460,12 @@ 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 < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
+        for (idx = 0; idx < max_idx; idx++)
             cpe->ms_mask[idx] = get_bits1(gb);
     } else if (ms_present == 2) {
-        memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
+        memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
     }
 }
 
@@ -999,7 +1497,7 @@ static inline float *VMUL4(float *dst, const float *v, unsigned idx,
 static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
                             unsigned sign, const float *scale)
 {
-    union float754 s0, s1;
+    union av_intfloat32 s0, s1;
 
     s0.f = s1.f = *scale;
     s0.i ^= sign >> 1 << 31;
@@ -1017,8 +1515,8 @@ static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
                             unsigned sign, const float *scale)
 {
     unsigned nz = idx >> 12;
-    union float754 s = { .f = *scale };
-    union float754 t;
+    union av_intfloat32 s = { .f = *scale };
+    union av_intfloat32 t;
 
     t.i = s.i ^ (sign & 1U<<31);
     *dst++ = v[idx    & 3] * t.f;
@@ -1031,7 +1529,7 @@ static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
     t.i = s.i ^ (sign & 1U<<31);
     *dst++ = v[idx>>4 & 3] * t.f;
 
-    sign <<= nz & 1; nz >>= 1;
+    sign <<= nz & 1;
     t.i = s.i ^ (sign & 1U<<31);
     *dst++ = v[idx>>6 & 3] * t.f;
 
@@ -1063,7 +1561,8 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
     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]));
+        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];
@@ -1088,9 +1587,9 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
                         cfo[k] = ac->random_state;
                     }
 
-                    band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
+                    band_energy = ac->fdsp.scalarproduct_float(cfo, cfo, off_len);
                     scale = sf[idx] / sqrtf(band_energy);
-                    ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
+                    ac->fdsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
                 }
             } else {
                 const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
@@ -1218,7 +1717,7 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
 
                                     if (b > 8) {
                                         av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
-                                        return -1;
+                                        return AVERROR_INVALIDDATA;
                                     }
 
                                     SKIP_BITS(re, gb, b + 1);
@@ -1236,7 +1735,7 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
                             }
                         } while (len -= 2);
 
-                        ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
+                        ac->fdsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
                     }
                 }
 
@@ -1267,7 +1766,7 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
 
 static av_always_inline float flt16_round(float pf)
 {
-    union float754 tmp;
+    union av_intfloat32 tmp;
     tmp.f = pf;
     tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
     return tmp.f;
@@ -1275,7 +1774,7 @@ static av_always_inline float flt16_round(float pf)
 
 static av_always_inline float flt16_even(float pf)
 {
-    union float754 tmp;
+    union av_intfloat32 tmp;
     tmp.f = pf;
     tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
     return tmp.f;
@@ -1283,7 +1782,7 @@ static av_always_inline float flt16_even(float pf)
 
 static av_always_inline float flt16_trunc(float pf)
 {
-    union float754 pun;
+    union av_intfloat32 pun;
     pun.f = pf;
     pun.i &= 0xFFFF0000U;
     return pun.f;
@@ -1333,14 +1832,20 @@ static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
     }
 
     if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
-        for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
-            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
+        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]);
+                        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);
+            reset_predictor_group(sce->predictor_state,
+                                  sce->ics.predictor_reset_group);
     } else
         reset_all_predictors(sce->predictor_state);
 }
@@ -1360,7 +1865,14 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
     TemporalNoiseShaping    *tns = &sce->tns;
     IndividualChannelStream *ics = &sce->ics;
     float *out = sce->coeffs;
-    int global_gain, pulse_present = 0;
+    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.
@@ -1370,39 +1882,51 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
     global_gain = get_bits(gb, 8);
 
     if (!common_window && !scale_flag) {
-        if (decode_ics_info(ac, ics, gb, 0) < 0)
-            return -1;
+        if (decode_ics_info(ac, ics, gb) < 0)
+            return AVERROR_INVALIDDATA;
     }
 
-    if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
-        return -1;
-    if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
-        return -1;
+    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 ((pulse_present = get_bits1(gb))) {
+        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 -1;
+                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 -1;
+                av_log(ac->avctx, AV_LOG_ERROR,
+                       "Pulse data corrupt or invalid.\n");
+                return AVERROR_INVALIDDATA;
             }
         }
-        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
-            return -1;
-        if (get_bits1(gb)) {
-            av_log_missing_feature(ac->avctx, "SSR", 1);
-            return -1;
+        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 implementations 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 -1;
+    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
+                                    &pulse, ics, sce->band_type) < 0)
+        return AVERROR_INVALIDDATA;
 
-    if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
+    if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
         apply_prediction(ac, sce);
 
     return 0;
@@ -1421,11 +1945,12 @@ static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
     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) {
+                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->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
-                                              ch1 + group * 128 + offsets[i],
-                                              offsets[i+1] - offsets[i]);
+                    ac->fdsp.butterflies_float(ch0 + group * 128 + offsets[i],
+                                               ch1 + group * 128 + offsets[i],
+                                               offsets[i+1] - offsets[i]);
                 }
             }
         }
@@ -1441,7 +1966,8 @@ static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
  *                      [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)
+static void apply_intensity_stereo(AACContext *ac,
+                                   ChannelElement *cpe, int ms_present)
 {
     const IndividualChannelStream *ics = &cpe->ch[1].ics;
     SingleChannelElement         *sce1 = &cpe->ch[1];
@@ -1452,7 +1978,8 @@ static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_p
     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) {
+            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);
@@ -1460,10 +1987,10 @@ static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_p
                         c *= 1 - 2 * cpe->ms_mask[idx];
                     scale = c * sce1->sf[idx];
                     for (group = 0; group < ics->group_len[g]; group++)
-                        ac->dsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
-                                                   coef0 + group * 128 + offsets[i],
-                                                   scale,
-                                                   offsets[i + 1] - offsets[i]);
+                        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];
@@ -1484,21 +2011,23 @@ static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_p
 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 = get_bits1(gb);
+    common_window = eld_syntax || get_bits1(gb);
     if (common_window) {
-        if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
-            return -1;
+        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->m4ac.object_type != AOT_AAC_MAIN))
+        if (cpe->ch[1].ics.predictor_present &&
+            (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
             if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
-                decode_ltp(ac, &cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
+                decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
         ms_present = get_bits(gb, 2);
         if (ms_present == 3) {
             av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
-            return -1;
+            return AVERROR_INVALIDDATA;
         } else if (ms_present)
             decode_mid_side_stereo(cpe, gb, ms_present);
     }
@@ -1510,7 +2039,7 @@ static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
     if (common_window) {
         if (ms_present)
             apply_mid_side_stereo(ac, cpe);
-        if (ac->m4ac.object_type == AOT_AAC_MAIN) {
+        if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
             apply_prediction(ac, &cpe->ch[0]);
             apply_prediction(ac, &cpe->ch[1]);
         }
@@ -1566,7 +2095,7 @@ static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
         int idx  = 0;
         int cge  = 1;
         int gain = 0;
-        float gain_cache = 1.;
+        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;
@@ -1621,12 +2150,10 @@ static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
 /**
  * Decode dynamic range information; reference: table 4.52.
  *
- * @param   cnt length of TYPE_FIL syntactic element in bytes
- *
  * @return  Returns number of bytes consumed.
  */
 static int decode_dynamic_range(DynamicRangeControl *che_drc,
-                                GetBitContext *gb, int cnt)
+                                GetBitContext *gb)
 {
     int n             = 1;
     int drc_num_bands = 1;
@@ -1691,25 +2218,28 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
         if (!che) {
             av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
             return res;
-        } else if (!ac->m4ac.sbr) {
+        } 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->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) {
+        } 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->m4ac.ps == -1 && ac->output_configured < OC_LOCKED && ac->avctx->channels == 1) {
-            ac->m4ac.sbr = 1;
-            ac->m4ac.ps = 1;
-            output_configure(ac, ac->che_pos, ac->che_pos, ac->m4ac.chan_config, ac->output_configured);
+        } 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->m4ac.sbr = 1;
+            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, cnt);
+        res = decode_dynamic_range(&ac->che_drc, gb);
         break;
     case EXT_FILL:
     case EXT_FILL_DATA:
@@ -1734,7 +2264,7 @@ static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
     int w, filt, m, i;
     int bottom, top, order, start, end, size, inc;
     float lpc[TNS_MAX_ORDER];
-    float tmp[TNS_MAX_ORDER];
+    float tmp[TNS_MAX_ORDER + 1];
 
     for (w = 0; w < ics->num_windows; w++) {
         bottom = ics->num_swb;
@@ -1792,15 +2322,15 @@ static void windowing_and_mdct_ltp(AACContext *ac, float *out,
     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->dsp.vector_fmul(in, in, lwindow_prev, 1024);
+        ac->fdsp.vector_fmul(in, in, lwindow_prev, 1024);
     } else {
         memset(in, 0, 448 * sizeof(float));
-        ac->dsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
+        ac->fdsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
     }
     if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
-        ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
+        ac->fdsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
     } else {
-        ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
+        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);
@@ -1853,17 +2383,17 @@ static void update_ltp(AACContext *ac, SingleChannelElement *sce)
     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
         memcpy(saved_ltp,       saved, 512 * sizeof(float));
         memset(saved_ltp + 576, 0,     448 * sizeof(float));
-        ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
+        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->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
+        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->dsp.vector_fmul_reverse(saved_ltp,       ac->buf_mdct + 512,     &lwindow[512],     512);
+        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];
     }
@@ -1904,36 +2434,121 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
      */
     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->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);
+        ac->fdsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);
     } else {
-        memcpy(                        out,               saved,            448 * sizeof(float));
+        memcpy(                         out,               saved,            448 * sizeof(float));
 
         if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
-            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, 64);
-            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      64);
-            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      64);
-            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      64);
-            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      64);
-            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
+            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->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);
-            memcpy(                    out + 576,         buf + 64,         448 * sizeof(float));
+            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->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 64);
-        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
-        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
-        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
+        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));
+        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));
+        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,"
+    // Audio, Language and Image Processing, 2008. ICALIP 2008. International Conference on
+    // 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));
 }
 
 /**
@@ -1950,7 +2565,7 @@ static void apply_dependent_coupling(AACContext *ac,
     float *dest = target->coeffs;
     const float *src = cce->ch[0].coeffs;
     int g, i, group, k, idx = 0;
-    if (ac->m4ac.object_type == AOT_AAC_LTP) {
+    if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
         av_log(ac->avctx, AV_LOG_ERROR,
                "Dependent coupling is not supported together with LTP\n");
         return;
@@ -1961,7 +2576,7 @@ static void apply_dependent_coupling(AACContext *ac,
                 const float gain = cce->coup.gain[index][idx];
                 for (group = 0; group < ics->group_len[g]; group++) {
                     for (k = offsets[i]; k < offsets[i + 1]; k++) {
-                        // XXX dsputil-ize
+                        // FIXME: SIMDify
                         dest[group * 128 + k] += gain * src[group * 128 + k];
                     }
                 }
@@ -1985,7 +2600,7 @@ static void apply_independent_coupling(AACContext *ac,
     const float gain = cce->coup.gain[index][0];
     const float *src = cce->ch[0].ret;
     float *dest = target->ret;
-    const int len = 1024 << (ac->m4ac.sbr == 1);
+    const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
 
     for (i = 0; i < len; i++)
         dest[i] += gain * src[i];
@@ -2032,13 +2647,24 @@ static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
 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 = 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) {
                 if (type <= TYPE_CPE)
                     apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
-                if (ac->m4ac.object_type == AOT_AAC_LTP) {
+                if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
                     if (che->ch[0].ics.predictor_present) {
                         if (che->ch[0].ics.ltp.present)
                             apply_ltp(ac, &che->ch[0]);
@@ -2053,15 +2679,15 @@ static void spectral_to_sample(AACContext *ac)
                 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_windowing(ac, &che->ch[0]);
-                    if (ac->m4ac.object_type == AOT_AAC_LTP)
+                    imdct_and_window(ac, &che->ch[0]);
+                    if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
                         update_ltp(ac, &che->ch[0]);
                     if (type == TYPE_CPE) {
-                        imdct_and_windowing(ac, &che->ch[1]);
-                        if (ac->m4ac.object_type == AOT_AAC_LTP)
+                        imdct_and_window(ac, &che->ch[1]);
+                        if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
                             update_ltp(ac, &che->ch[1]);
                     }
-                    if (ac->m4ac.sbr > 0) {
+                    if (ac->oc[1].m4ac.sbr > 0) {
                         ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
                     }
                 }
@@ -2076,71 +2702,160 @@ 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->output_configured != OC_LOCKED && hdr_info.chan_config) {
-            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
-            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
-            ac->m4ac.chan_config = hdr_info.chan_config;
-            if (set_default_channel_config(ac->avctx, new_che_pos, hdr_info.chan_config))
-                return -7;
-            if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
-                return -7;
-        } else if (ac->output_configured != OC_LOCKED) {
-            ac->m4ac.chan_config = 0;
-            ac->output_configured = OC_NONE;
-        }
-        if (ac->output_configured != OC_LOCKED) {
-            ac->m4ac.sbr = -1;
-            ac->m4ac.ps  = -1;
-            ac->m4ac.sample_rate     = hdr_info.sample_rate;
-            ac->m4ac.sampling_index  = hdr_info.sampling_index;
-            ac->m4ac.object_type     = hdr_info.object_type;
-        }
-        if (!ac->avctx->sample_rate)
-            ac->avctx->sample_rate = hdr_info.sample_rate;
-        if (hdr_info.num_aac_frames == 1) {
-            if (!hdr_info.crc_absent)
-                skip_bits(gb, 16);
+        if (hdr_info.num_aac_frames != 1) {
+            avpriv_report_missing_feature(ac->avctx,
+                                          "More than one AAC RDB per ADTS frame");
+            return AVERROR_PATCHWELCOME;
+        }
+        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 {
-            av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0);
-            return -1;
+            ac->oc[1].m4ac.chan_config = 0;
         }
+        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;
+        }
+        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 *data_size, GetBitContext *gb)
+                                int *got_frame_ptr, GetBitContext *gb)
 {
     AACContext *ac = avctx->priv_data;
     ChannelElement *che = NULL, *che_prev = NULL;
     enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
-    int err, elem_id, data_size_tmp;
-    int samples = 0, multiplier, audio_found = 0;
+    int err, elem_id;
+    int samples = 0, multiplier, audio_found = 0, pce_found = 0;
+
+    ac->frame = data;
 
     if (show_bits(gb, 12) == 0xfff) {
-        if (parse_adts_frame_header(ac, gb) < 0) {
+        if ((err = parse_adts_frame_header(ac, gb)) < 0) {
             av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
-            return -1;
+            goto fail;
         }
-        if (ac->m4ac.sampling_index > 12) {
-            av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
-            return -1;
+        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 (avctx->channels)
+        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->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);
-                return -1;
+                err = AVERROR_INVALIDDATA;
+                goto fail;
             }
             samples = 1024;
         }
@@ -2171,15 +2886,22 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
             break;
 
         case TYPE_PCE: {
-            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
-            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
-            if ((err = decode_pce(avctx, &ac->m4ac, new_che_pos, gb)))
+            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 (ac->output_configured > OC_TRIAL_PCE)
+            }
+            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, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
+                pop_output_configuration(ac);
+            } else {
+                err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
+                pce_found = 1;
+            }
             break;
         }
 
@@ -2188,7 +2910,8 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
                 elem_id += get_bits(gb, 8) - 1;
             if (get_bits_left(gb) < 8 * elem_id) {
                     av_log(avctx, AV_LOG_ERROR, overread_err);
-                    return -1;
+                    err = AVERROR_INVALIDDATA;
+                    goto fail;
             }
             while (elem_id > 0)
                 elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
@@ -2196,7 +2919,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
             break;
 
         default:
-            err = -1; /* should not happen, but keeps compiler happy */
+            err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
             break;
         }
 
@@ -2204,61 +2927,89 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
         elem_type_prev = elem_type;
 
         if (err)
-            return err;
+            goto fail;
 
         if (get_bits_left(gb) < 3) {
             av_log(avctx, AV_LOG_ERROR, overread_err);
-            return -1;
+            err = AVERROR_INVALIDDATA;
+            goto fail;
         }
     }
 
+    if (!avctx->channels) {
+        *got_frame_ptr = 0;
+        return 0;
+    }
+
     spectral_to_sample(ac);
 
-    multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0;
+    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->output_configured < OC_LOCKED) {
-        avctx->sample_rate = ac->m4ac.sample_rate << multiplier;
-        avctx->frame_size = samples;
-    }
 
-    data_size_tmp = samples * avctx->channels *
-                    av_get_bytes_per_sample(avctx->sample_fmt);
-    if (*data_size < data_size_tmp) {
-        av_log(avctx, AV_LOG_ERROR,
-               "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
-               *data_size, data_size_tmp);
-        return -1;
+    if (ac->oc[1].status && audio_found) {
+        avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
+        avctx->frame_size = samples;
+        ac->oc[1].status = OC_LOCKED;
     }
-    *data_size = data_size_tmp;
 
     if (samples) {
-        if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
-            ac->fmt_conv.float_interleave(data, (const float **)ac->output_data,
-                                          samples, avctx->channels);
-        else
-            ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data,
-                                                   samples, avctx->channels);
+        ac->frame->nb_samples = samples;
+        ac->frame->sample_rate = avctx->sample_rate;
     }
-
-    if (ac->output_configured && audio_found)
-        ac->output_configured = OC_LOCKED;
+    *got_frame_ptr = !!samples;
 
     return 0;
+fail:
+    pop_output_configuration(ac);
+    return err;
 }
 
 static int aac_decode_frame(AVCodecContext *avctx, void *data,
-                            int *data_size, AVPacket *avpkt)
+                            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);
+
+    if (new_extradata) {
+        av_free(avctx->extradata);
+        avctx->extradata = av_mallocz(new_extradata_size +
+                                      AV_INPUT_BUFFER_PADDING_SIZE);
+        if (!avctx->extradata)
+            return AVERROR(ENOMEM);
+        avctx->extradata_size = new_extradata_size;
+        memcpy(avctx->extradata, new_extradata, new_extradata_size);
+        push_output_configuration(ac);
+        if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
+                                         avctx->extradata,
+                                         avctx->extradata_size*8, 1) < 0) {
+            pop_output_configuration(ac);
+            return AVERROR_INVALIDDATA;
+        }
+    }
 
-    init_get_bits(&gb, buf, buf_size * 8);
+    if ((err = init_get_bits(&gb, buf, buf_size * 8)) < 0)
+        return err;
 
-    if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0)
+    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);
+    }
+    if (err < 0)
         return err;
 
     buf_consumed = (get_bits_count(&gb) + 7) >> 3;
@@ -2284,7 +3035,9 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
 
     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);
     return 0;
 }
 
@@ -2292,13 +3045,13 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
 #define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
 
 struct LATMContext {
-    AACContext      aac_ctx;             ///< containing AACContext
-    int             initialized;         ///< initilized after a valid extradata was seen
+    AACContext aac_ctx;     ///< containing AACContext
+    int initialized;        ///< initialized after a valid extradata was seen
 
     // parser data
-    int             audio_mux_version_A; ///< LATM syntax version
-    int             frame_length_type;   ///< 0/1 variable/fixed frame length
-    int             frame_length;        ///< frame length for fixed frame length
+    int audio_mux_version_A; ///< LATM syntax version
+    int frame_length_type;   ///< 0/1 variable/fixed frame length
+    int frame_length;        ///< frame length for fixed frame length
 };
 
 static inline uint32_t latm_get_value(GetBitContext *b)
@@ -2309,41 +3062,56 @@ static inline uint32_t latm_get_value(GetBitContext *b)
 }
 
 static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
-                                             GetBitContext *gb)
+                                             GetBitContext *gb, int asclen)
 {
-    AVCodecContext *avctx = latmctx->aac_ctx.avctx;
-    MPEG4AudioConfig m4ac;
-    int  config_start_bit = get_bits_count(gb);
-    int     bits_consumed, esize;
+    AACContext *ac        = &latmctx->aac_ctx;
+    AVCodecContext *avctx = ac->avctx;
+    MPEG4AudioConfig m4ac = { 0 };
+    int config_start_bit  = get_bits_count(gb);
+    int sync_extension    = 0;
+    int bits_consumed, esize;
+
+    if (asclen) {
+        sync_extension = 1;
+        asclen         = FFMIN(asclen, get_bits_left(gb));
+    } else
+        asclen         = get_bits_left(gb);
 
     if (config_start_bit % 8) {
-        av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
-                               "config not byte aligned.\n", 1);
+        avpriv_request_sample(latmctx->aac_ctx.avctx,
+                              "Non-byte-aligned audio-specific config");
+        return AVERROR_PATCHWELCOME;
+    }
+    if (asclen <= 0)
         return AVERROR_INVALIDDATA;
-    } else {
-        bits_consumed =
-            decode_audio_specific_config(NULL, avctx, &m4ac,
+    bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac,
                                          gb->buffer + (config_start_bit / 8),
-                                         get_bits_left(gb) / 8);
+                                         asclen, sync_extension);
 
-        if (bits_consumed < 0)
-            return AVERROR_INVALIDDATA;
+    if (bits_consumed < 0)
+        return AVERROR_INVALIDDATA;
+
+    if (!latmctx->initialized ||
+        ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
+        ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
+
+        av_log(avctx, AV_LOG_INFO, "audio config changed\n");
+        latmctx->initialized = 0;
 
         esize = (bits_consumed+7) / 8;
 
-        if (avctx->extradata_size <= esize) {
+        if (avctx->extradata_size < esize) {
             av_free(avctx->extradata);
-            avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
+            avctx->extradata = av_malloc(esize + AV_INPUT_BUFFER_PADDING_SIZE);
             if (!avctx->extradata)
                 return AVERROR(ENOMEM);
         }
 
         avctx->extradata_size = esize;
         memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
-        memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
-
-        skip_bits_long(gb, bits_consumed);
+        memset(avctx->extradata+esize, 0, AV_INPUT_BUFFER_PADDING_SIZE);
     }
+    skip_bits_long(gb, bits_consumed);
 
     return bits_consumed;
 }
@@ -2366,8 +3134,7 @@ static int read_stream_mux_config(struct LATMContext *latmctx,
         skip_bits(gb, 6);                       // numSubFrames
         // numPrograms
         if (get_bits(gb, 4)) {                  // numPrograms
-            av_log_missing_feature(latmctx->aac_ctx.avctx,
-                                   "multiple programs are not supported\n", 1);
+            avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
             return AVERROR_PATCHWELCOME;
         }
 
@@ -2375,18 +3142,17 @@ static int read_stream_mux_config(struct LATMContext *latmctx,
 
         // for each layer (which there is only on in DVB)
         if (get_bits(gb, 3)) {                   // numLayer
-            av_log_missing_feature(latmctx->aac_ctx.avctx,
-                                   "multiple layers are not supported\n", 1);
+            avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
             return AVERROR_PATCHWELCOME;
         }
 
         // for all but first stream: use_same_config = get_bits(gb, 1);
         if (!audio_mux_version) {
-            if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
+            if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
                 return ret;
         } else {
             int ascLen = latm_get_value(gb);
-            if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
+            if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
                 return ret;
             ascLen -= ret;
             skip_bits_long(gb, ascLen);
@@ -2480,14 +3246,15 @@ static int read_audio_mux_element(struct LATMContext *latmctx,
 }
 
 
-static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
-                             AVPacket *avpkt)
+static int latm_decode_frame(AVCodecContext *avctx, void *out,
+                             int *got_frame_ptr, AVPacket *avpkt)
 {
     struct LATMContext *latmctx = avctx->priv_data;
     int                 muxlength, err;
     GetBitContext       gb;
 
-    init_get_bits(&gb, avpkt->data, avpkt->size * 8);
+    if ((err = init_get_bits(&gb, avpkt->data, avpkt->size * 8)) < 0)
+        return err;
 
     // check for LOAS sync word
     if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
@@ -2503,13 +3270,16 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
 
     if (!latmctx->initialized) {
         if (!avctx->extradata) {
-            *out_size = 0;
+            *got_frame_ptr = 0;
             return avpkt->size;
         } else {
+            push_output_configuration(&latmctx->aac_ctx);
             if ((err = decode_audio_specific_config(
-                    &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.m4ac,
-                    avctx->extradata, avctx->extradata_size)) < 0)
+                    &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
+                    avctx->extradata, avctx->extradata_size*8, 1)) < 0) {
+                pop_output_configuration(&latmctx->aac_ctx);
                 return err;
+            }
             latmctx->initialized = 1;
         }
     }
@@ -2521,13 +3291,23 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
         return AVERROR_INVALIDDATA;
     }
 
-    if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
+    switch (latmctx->aac_ctx.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, out, got_frame_ptr, &gb);
+        break;
+    default:
+        err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb);
+    }
+    if (err < 0)
         return err;
 
     return muxlength;
 }
 
-av_cold static int latm_decode_init(AVCodecContext *avctx)
+static av_cold int latm_decode_init(AVCodecContext *avctx)
 {
     struct LATMContext *latmctx = avctx->priv_data;
     int ret = aac_decode_init(avctx);
@@ -2540,18 +3320,19 @@ av_cold static int latm_decode_init(AVCodecContext *avctx)
 
 
 AVCodec ff_aac_decoder = {
-    .name           = "aac",
-    .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = CODEC_ID_AAC,
-    .priv_data_size = sizeof(AACContext),
-    .init           = aac_decode_init,
-    .close          = aac_decode_close,
-    .decode         = aac_decode_frame,
-    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
-    .sample_fmts = (const enum AVSampleFormat[]) {
-        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+    .name            = "aac",
+    .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
+    .type            = AVMEDIA_TYPE_AUDIO,
+    .id              = AV_CODEC_ID_AAC,
+    .priv_data_size  = sizeof(AACContext),
+    .init            = aac_decode_init,
+    .close           = aac_decode_close,
+    .decode          = aac_decode_frame,
+    .sample_fmts     = (const enum AVSampleFormat[]) {
+        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
     },
-    .capabilities = CODEC_CAP_CHANNEL_CONF,
+    .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
+    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE,
     .channel_layouts = aac_channel_layout,
 };
 
@@ -2561,17 +3342,18 @@ AVCodec ff_aac_decoder = {
     To do a more complex LATM demuxing a separate LATM demuxer should be used.
 */
 AVCodec ff_aac_latm_decoder = {
-    .name = "aac_latm",
-    .type = AVMEDIA_TYPE_AUDIO,
-    .id   = CODEC_ID_AAC_LATM,
-    .priv_data_size = sizeof(struct LATMContext),
-    .init   = latm_decode_init,
-    .close  = aac_decode_close,
-    .decode = latm_decode_frame,
-    .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
-    .sample_fmts = (const enum AVSampleFormat[]) {
-        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+    .name            = "aac_latm",
+    .long_name       = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
+    .type            = AVMEDIA_TYPE_AUDIO,
+    .id              = AV_CODEC_ID_AAC_LATM,
+    .priv_data_size  = sizeof(struct LATMContext),
+    .init            = latm_decode_init,
+    .close           = aac_decode_close,
+    .decode          = latm_decode_frame,
+    .sample_fmts     = (const enum AVSampleFormat[]) {
+        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
     },
-    .capabilities = CODEC_CAP_CHANNEL_CONF,
+    .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
+    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE,
     .channel_layouts = aac_channel_layout,
 };