[FFmpeg-devel,3/2,WIP,RFC] avformat: Immersive Audio Model and Formats demuxer

diff --git a/libavcodec/avpacket.c b/libavcodec/avpacket.c
index e29725c2d2..0f8c9b77ae 100644
--- a/libavcodec/avpacket.c
+++ b/libavcodec/avpacket.c
@@ -301,6 +301,9 @@  const char *av_packet_side_data_name(enum AVPacketSideDataType type)
     case AV_PKT_DATA_DOVI_CONF:                  return "DOVI configuration record";
     case AV_PKT_DATA_S12M_TIMECODE:              return "SMPTE ST 12-1:2014 timecode";
     case AV_PKT_DATA_DYNAMIC_HDR10_PLUS:         return "HDR10+ Dynamic Metadata (SMPTE 2094-40)";
+    case AV_PKT_DATA_IAMF_MIX_GAIN_PARAM:        return "IAMF Mix Gain Parameter Data";
+    case AV_PKT_DATA_IAMF_DEMIXING_INFO_PARAM:   return "IAMF Demixing Info Parameter Data";
+    case AV_PKT_DATA_IAMF_RECON_GAIN_INFO_PARAM: return "IAMF Recon Gain Info Parameter Data";
     }
     return NULL;
 }
diff --git a/libavcodec/packet.h b/libavcodec/packet.h
index b19409b719..2c57d262c6 100644
--- a/libavcodec/packet.h
+++ b/libavcodec/packet.h
@@ -299,6 +299,30 @@  enum AVPacketSideDataType {
      */
     AV_PKT_DATA_DYNAMIC_HDR10_PLUS,
 
+    /**
+     * IAMF Mix Gain Parameter Data associated with the audio frame. This metadata
+     * is in the form of the AVIAMFParamDefinition struct and contains information
+     * defined in sections 3.6.1 and 3.8.1 of the Immersive Audio Model and
+     * Formats standard.
+     */
+    AV_PKT_DATA_IAMF_MIX_GAIN_PARAM,
+
+    /**
+     * IAMF Demixing Info Parameter Data associated with the audio frame. This
+     * metadata is in the form of the AVIAMFParamDefinition struct and contains
+     * information defined in sections 3.6.1 and 3.8.2 of the Immersive Audio Model
+     * and Formats standard.
+     */
+    AV_PKT_DATA_IAMF_DEMIXING_INFO_PARAM,
+
+    /**
+     * IAMF Recon Gain Info Parameter Data associated with the audio frame. This
+     * metadata is in the form of the AVIAMFParamDefinition struct and contains
+     * information defined in sections 3.6.1 and 3.8.3 of the Immersive Audio Model
+     * and Formats standard.
+     */
+    AV_PKT_DATA_IAMF_RECON_GAIN_INFO_PARAM,
+
     /**
      * The number of side data types.
      * This is not part of the public API/ABI in the sense that it may
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 329055ccfd..364bc417a3 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -3,6 +3,7 @@  DESC = FFmpeg container format library
 
 HEADERS = avformat.h                                                    \
           avio.h                                                        \
+          iamf.h                                                        \
           version.h                                                     \
           version_major.h                                               \
 
@@ -258,6 +259,7 @@  OBJS-$(CONFIG_EVC_MUXER)                 += rawenc.o
 OBJS-$(CONFIG_HLS_DEMUXER)               += hls.o hls_sample_encryption.o
 OBJS-$(CONFIG_HLS_MUXER)                 += hlsenc.o hlsplaylist.o avc.o
 OBJS-$(CONFIG_HNM_DEMUXER)               += hnm.o
+OBJS-$(CONFIG_IAMF_DEMUXER)              += iamfdec.o iamf.o
 OBJS-$(CONFIG_ICO_DEMUXER)               += icodec.o
 OBJS-$(CONFIG_ICO_MUXER)                 += icoenc.o
 OBJS-$(CONFIG_IDCIN_DEMUXER)             += idcin.o
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index d4b505a5a3..63ca44bacd 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -212,6 +212,7 @@  extern const FFOutputFormat ff_hevc_muxer;
 extern const AVInputFormat  ff_hls_demuxer;
 extern const FFOutputFormat ff_hls_muxer;
 extern const AVInputFormat  ff_hnm_demuxer;
+extern const AVInputFormat  ff_iamf_demuxer;
 extern const AVInputFormat  ff_ico_demuxer;
 extern const FFOutputFormat ff_ico_muxer;
 extern const AVInputFormat  ff_idcin_demuxer;
diff --git a/libavformat/avformat.c b/libavformat/avformat.c
index 99cda56c2f..caba3308c2 100644
--- a/libavformat/avformat.c
+++ b/libavformat/avformat.c
@@ -37,6 +37,7 @@ 
 #include "avformat.h"
 #include "avio.h"
 #include "demux.h"
+#include "iamf.h"
 #include "mux.h"
 #include "internal.h"
 
@@ -90,7 +91,14 @@  void ff_free_stream_group(AVStreamGroup **pstg)
     av_freep(&stg->streams);
     av_freep(&stg->priv_data);
     switch (stg->type) {
-    // Structs in the union are freed here
+    case AV_STREAM_GROUP_PARAMS_IAMF_AUDIO_ELEMENT: {
+        avformat_iamf_audio_element_free(&stg->params.iamf_audio_element);
+        break;
+    }
+    case AV_STREAM_GROUP_PARAMS_IAMF_MIX_PRESENTATION: {
+        avformat_iamf_mix_presentation_free(&stg->params.iamf_mix_presentation);
+        break;
+    }
     default:
         break;
     }
diff --git a/libavformat/avformat.h b/libavformat/avformat.h
index f045084c8d..455e07333e 100644
--- a/libavformat/avformat.h
+++ b/libavformat/avformat.h
@@ -1020,8 +1020,13 @@  typedef struct AVStream {
 
 enum AVStreamGroupParamsType {
     AV_STREAM_GROUP_PARAMS_NONE,
+    AV_STREAM_GROUP_PARAMS_IAMF_AUDIO_ELEMENT,
+    AV_STREAM_GROUP_PARAMS_IAMF_MIX_PRESENTATION,
 };
 
+struct AVIAMFAudioElement;
+struct AVIAMFMixPresentation;
+
 typedef struct AVStreamGroup {
     /**
      * A class for @ref avoptions. Set on group creation.
@@ -1055,7 +1060,8 @@  typedef struct AVStreamGroup {
      * Group-specific type parameters
      */
     union {
-        uintptr_t dummy; // Placeholder
+        struct AVIAMFAudioElement *iamf_audio_element;
+        struct AVIAMFMixPresentation *iamf_mix_presentation;
     } params;
 
     /**
diff --git a/libavformat/dump.c b/libavformat/dump.c
index c0868a1bb3..f48afdf679 100644
--- a/libavformat/dump.c
+++ b/libavformat/dump.c
@@ -38,6 +38,7 @@ 
 #include "libavcodec/avcodec.h"
 
 #include "avformat.h"
+#include "iamf.h"
 #include "internal.h"
 
 #define HEXDUMP_PRINT(...)                                                    \
@@ -509,7 +510,7 @@  static void dump_sidedata(void *ctx, const AVStream *st, const char *indent)
 
 /* "user interface" functions */
 static void dump_stream_format(const AVFormatContext *ic, int i,
-                               int index, int is_output)
+                               int group_index, int index, int is_output)
 {
     char buf[256];
     int flags = (is_output ? ic->oformat->flags : ic->iformat->flags);
@@ -517,6 +518,8 @@  static void dump_stream_format(const AVFormatContext *ic, int i,
     const FFStream *const sti = cffstream(st);
     const AVDictionaryEntry *lang = av_dict_get(st->metadata, "language", NULL, 0);
     const char *separator = ic->dump_separator;
+    const char *group_indent = group_index >= 0 ? "    " : "";
+    const char *extra_indent = group_index >= 0 ? "        " : "      ";
     AVCodecContext *avctx;
     int ret;
 
@@ -543,7 +546,10 @@  static void dump_stream_format(const AVFormatContext *ic, int i,
     avcodec_string(buf, sizeof(buf), avctx, is_output);
     avcodec_free_context(&avctx);
 
-    av_log(NULL, AV_LOG_INFO, "  Stream #%d:%d", index, i);
+    av_log(NULL, AV_LOG_INFO, "%s  Stream #%d", group_indent, index);
+    if (group_index >= 0)
+        av_log(NULL, AV_LOG_INFO, ":%d", group_index);
+    av_log(NULL, AV_LOG_INFO, ":%d", i);
 
     /* the pid is an important information, so we display it */
     /* XXX: add a generic system */
@@ -621,9 +627,61 @@  static void dump_stream_format(const AVFormatContext *ic, int i,
         av_log(NULL, AV_LOG_INFO, " (non-diegetic)");
     av_log(NULL, AV_LOG_INFO, "\n");
 
-    dump_metadata(NULL, st->metadata, "    ");
+    dump_metadata(NULL, st->metadata, extra_indent);
 
-    dump_sidedata(NULL, st, "    ");
+    dump_sidedata(NULL, st, extra_indent);
+}
+
+static void dump_stream_group(const AVFormatContext *ic, uint8_t *printed,
+                              int i, int index, int is_output)
+{
+    const AVStreamGroup *stg = ic->stream_groups[i];
+    char buf[512];
+    int ret;
+
+    av_log(NULL, AV_LOG_INFO, "  Stream group #%d:%d:", index, i);
+
+    switch (stg->type) {
+    case AV_STREAM_GROUP_PARAMS_IAMF_AUDIO_ELEMENT: {
+        AVIAMFAudioElement *iamf = stg->params.iamf_audio_element;
+        int substream_count = 0;
+        av_log(NULL, AV_LOG_INFO, " IAMF Audio Element\n");
+        for (int j = 0; j < iamf->num_layers; j++) {
+            AVIAMFLayer *layer = iamf->layers[j];
+            substream_count += layer->substream_count;
+            av_log(NULL, AV_LOG_INFO, "    Layer %d:", j);
+            ret = av_channel_layout_describe(&layer->ch_layout, buf, sizeof(buf));
+            if (ret >= 0)
+                av_log(NULL, AV_LOG_INFO, " %s", buf);
+            av_log(NULL, AV_LOG_INFO, "\n");
+            for (int k = 0; k < substream_count && k < stg->nb_streams; k++) {
+                dump_stream_format(ic, stg->streams[k]->index, i, index, is_output);
+                printed[stg->streams[k]->index] = 1;
+            }
+        }
+        break;
+    }
+    case AV_STREAM_GROUP_PARAMS_IAMF_MIX_PRESENTATION: {
+        AVIAMFMixPresentation *mix_presentation = stg->params.iamf_mix_presentation;
+        av_log(NULL, AV_LOG_INFO, " IAMF Mix Presentation\n");
+        for (int j = 0; j < mix_presentation->num_sub_mixes; j++) {
+            AVIAMFSubmixPresentation *sub_mix = mix_presentation->sub_mixes[j];
+            av_log(NULL, AV_LOG_INFO, "    Submix %d:\n", j);
+            for (int k = 0; k < sub_mix->num_submix_layouts; k++) {
+                AVIAMFSubmixLayout *submix_layout = sub_mix->submix_layouts[k];
+                av_log(NULL, AV_LOG_INFO, "      Layout %d", k);
+                if (submix_layout->layout_type == 2) {
+                    ret = av_channel_layout_describe(&submix_layout->sound_system, buf, sizeof(buf));
+                    if (ret >= 0)
+                        av_log(NULL, AV_LOG_INFO, " %s", buf);
+                } else if (submix_layout->layout_type == 3)
+                    av_log(NULL, AV_LOG_INFO, " Binaural");
+                av_log(NULL, AV_LOG_INFO, "\n");
+            }
+        }
+        break;
+    }
+    }
 }
 
 void av_dump_format(AVFormatContext *ic, int index,
@@ -699,7 +757,7 @@  void av_dump_format(AVFormatContext *ic, int index,
             dump_metadata(NULL, program->metadata, "    ");
             for (k = 0; k < program->nb_stream_indexes; k++) {
                 dump_stream_format(ic, program->stream_index[k],
-                                   index, is_output);
+                                   -1, index, is_output);
                 printed[program->stream_index[k]] = 1;
             }
             total += program->nb_stream_indexes;
@@ -708,9 +766,12 @@  void av_dump_format(AVFormatContext *ic, int index,
             av_log(NULL, AV_LOG_INFO, "  No Program\n");
     }
 
+    for (i = 0; i < ic->nb_stream_groups; i++)
+         dump_stream_group(ic, printed, i, index, is_output);
+
     for (i = 0; i < ic->nb_streams; i++)
         if (!printed[i])
-            dump_stream_format(ic, i, index, is_output);
+            dump_stream_format(ic, i, -1, index, is_output);
 
     av_free(printed);
 }
diff --git a/libavformat/iamf.c b/libavformat/iamf.c
new file mode 100644
index 0000000000..2d6f84a073
--- /dev/null
+++ b/libavformat/iamf.c
@@ -0,0 +1,336 @@ 
+/*
+ * Immersive Audio Model and Formats helper functions and defines
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <limits.h>
+#include <stddef.h>
+#include <stdint.h>
+
+#include "libavutil/error.h"
+#include "libavutil/mem.h"
+
+#include "iamf.h"
+#include "iamf_internal.h"
+
+const AVChannelLayout ff_iamf_scalable_ch_layouts[10] = {
+    AV_CHANNEL_LAYOUT_MONO,
+    AV_CHANNEL_LAYOUT_STEREO,
+    // "Loudspeaker configuration for Sound System B"
+    AV_CHANNEL_LAYOUT_5POINT1_BACK,
+    // "Loudspeaker configuration for Sound System C"
+    AV_CHANNEL_LAYOUT_7POINT1_TOP_BACK,
+    // "Loudspeaker configuration for Sound System D"
+    {
+        .nb_channels = 10,
+        .order       = AV_CHANNEL_ORDER_NATIVE,
+        .u.mask      = AV_CH_LAYOUT_7POINT1_TOP_BACK | AV_CH_TOP_BACK_LEFT | AV_CH_TOP_BACK_RIGHT,
+    },
+    // "Loudspeaker configuration for Sound System I"
+    AV_CHANNEL_LAYOUT_7POINT1,
+    // "Loudspeaker configuration for Sound System I" + Ltf + Rtf
+    {
+        .nb_channels = 10,
+        .order       = AV_CHANNEL_ORDER_NATIVE,
+        .u.mask      = AV_CH_LAYOUT_7POINT1 | AV_CH_TOP_FRONT_LEFT | AV_CH_TOP_FRONT_RIGHT,
+    },
+    // "Loudspeaker configuration for Sound System J"
+    {
+        .nb_channels = 12,
+        .order       = AV_CHANNEL_ORDER_NATIVE,
+        .u.mask      = AV_CH_LAYOUT_7POINT1 | AV_CH_TOP_FRONT_LEFT | AV_CH_TOP_FRONT_RIGHT |
+                                              AV_CH_TOP_BACK_LEFT  | AV_CH_TOP_BACK_RIGHT,
+    },
+    // Front subset of "Loudspeaker configuration for Sound System J"
+    {
+        .nb_channels = 6,
+        .order       = AV_CHANNEL_ORDER_NATIVE,
+        .u.mask      = AV_CH_LAYOUT_3POINT1 | AV_CH_TOP_FRONT_LEFT | AV_CH_TOP_FRONT_RIGHT,
+    },
+    // Binaural
+    AV_CHANNEL_LAYOUT_STEREO,
+};
+
+const struct IAMFSoundSystemMap ff_iamf_sound_system_map[13] = {
+    { SOUND_SYSTEM_A_0_2_0, AV_CHANNEL_LAYOUT_STEREO },
+    { SOUND_SYSTEM_B_0_5_0, AV_CHANNEL_LAYOUT_5POINT1_BACK },
+    { SOUND_SYSTEM_C_2_5_0, AV_CHANNEL_LAYOUT_7POINT1_TOP_BACK },
+    { SOUND_SYSTEM_D_4_5_0,
+        {
+            .nb_channels = 10,
+            .order       = AV_CHANNEL_ORDER_NATIVE,
+            .u.mask      = AV_CH_LAYOUT_7POINT1_TOP_BACK | AV_CH_TOP_BACK_LEFT | AV_CH_TOP_BACK_RIGHT,
+        },
+    },
+    { SOUND_SYSTEM_E_4_5_1,
+        {
+            .nb_channels = 11,
+            .order       = AV_CHANNEL_ORDER_NATIVE,
+            .u.mask      = AV_CH_LAYOUT_7POINT1_TOP_BACK | AV_CH_TOP_BACK_LEFT | AV_CH_TOP_BACK_RIGHT | AV_CH_BOTTOM_FRONT_CENTER,
+        },
+    },
+    { SOUND_SYSTEM_F_3_7_0,
+        {
+            .nb_channels = 12,
+            .order       = AV_CHANNEL_ORDER_NATIVE,
+            .u.mask      = AV_CH_LAYOUT_7POINT1 | AV_CH_TOP_FRONT_LEFT | AV_CH_TOP_FRONT_RIGHT | AV_CH_TOP_BACK_CENTER | AV_CH_LOW_FREQUENCY_2,
+        },
+    },
+    { SOUND_SYSTEM_G_4_9_0,
+        {
+            .nb_channels = 14,
+            .order       = AV_CHANNEL_ORDER_NATIVE,
+            .u.mask      = AV_CH_LAYOUT_7POINT1 | AV_CH_TOP_FRONT_LEFT | AV_CH_TOP_FRONT_RIGHT | AV_CH_TOP_BACK_LEFT | AV_CH_TOP_BACK_RIGHT |
+                                                  AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER,
+        },
+    },
+    { SOUND_SYSTEM_H_9_10_3, AV_CHANNEL_LAYOUT_22POINT2 },
+    { SOUND_SYSTEM_I_0_7_0, AV_CHANNEL_LAYOUT_7POINT1 },
+    { SOUND_SYSTEM_J_4_7_0,
+        {
+            .nb_channels = 12,
+            .order       = AV_CHANNEL_ORDER_NATIVE,
+            .u.mask      = AV_CH_LAYOUT_7POINT1 | AV_CH_TOP_FRONT_LEFT | AV_CH_TOP_FRONT_RIGHT |
+                                                  AV_CH_TOP_BACK_LEFT  | AV_CH_TOP_BACK_RIGHT,
+        },
+    },
+    { SOUND_SYSTEM_10_2_7_0,
+        {
+            .nb_channels = 10,
+            .order       = AV_CHANNEL_ORDER_NATIVE,
+            .u.mask      = AV_CH_LAYOUT_7POINT1 | AV_CH_TOP_FRONT_LEFT | AV_CH_TOP_FRONT_RIGHT,
+        },
+    },
+    { SOUND_SYSTEM_11_2_3_0,
+        {
+            .nb_channels = 6,
+            .order       = AV_CHANNEL_ORDER_NATIVE,
+            .u.mask      = AV_CH_LAYOUT_3POINT1 | AV_CH_TOP_FRONT_LEFT | AV_CH_TOP_FRONT_RIGHT,
+        },
+    },
+    { SOUND_SYSTEM_12_0_1_0, AV_CHANNEL_LAYOUT_MONO },
+};
+
+AVIAMFAudioElement *avformat_iamf_audio_element_alloc()
+{
+    return av_mallocz(sizeof(AVIAMFAudioElement));
+}
+
+int avformat_iamf_audio_element_add_layer(AVIAMFAudioElement *audio_element)
+{
+    AVIAMFLayer **layers;
+
+    if (audio_element->num_layers == UINT_MAX)
+        return AVERROR(EINVAL);
+
+    layers = av_realloc_array(audio_element->layers, audio_element->num_layers + 1,
+                              sizeof(*audio_element->layers));
+    if (!layers)
+        return AVERROR(ENOMEM);
+
+    audio_element->layers = layers;
+
+    audio_element->layers[audio_element->num_layers] = av_mallocz(sizeof(AVIAMFLayer));
+    if (!audio_element->layers[audio_element->num_layers])
+        return AVERROR(ENOMEM);
+
+    av_channel_layout_uninit(&audio_element->layers[audio_element->num_layers++]->ch_layout);
+
+    return 0;
+}
+
+void avformat_iamf_audio_element_free(AVIAMFAudioElement **paudio_element)
+{
+    AVIAMFAudioElement *audio_element = *paudio_element;
+
+    if (!audio_element)
+        return;
+
+    for (int i; i < audio_element->num_layers; i++) {
+        AVIAMFLayer *layer = audio_element->layers[i];
+        av_channel_layout_uninit(&layer->ch_layout);
+        av_free(layer->demixing_matrix);
+        av_free(layer);
+    }
+    av_free(audio_element->layers);
+
+    av_free(audio_element->demixing_info);
+    av_free(audio_element->recon_gain_info);
+    av_freep(paudio_element);
+}
+
+AVIAMFMixPresentation *avformat_iamf_mix_presentation_alloc()
+{
+    return av_mallocz(sizeof(AVIAMFMixPresentation));
+}
+
+int avformat_iamf_mix_presentation_add_submix(AVIAMFMixPresentation *mix_presentation,
+                                              unsigned int num_submix_elements,
+                                              unsigned int num_submix_layouts)
+{
+    AVIAMFSubmixPresentation **sub_mixes, *sub_mix;
+
+    if (mix_presentation->num_sub_mixes == UINT_MAX)
+        return AVERROR(EINVAL);
+
+    sub_mixes = av_realloc_array(mix_presentation->sub_mixes, mix_presentation->num_sub_mixes + 1,
+                                 sizeof(*mix_presentation->sub_mixes));
+    if (!sub_mixes)
+        return AVERROR(ENOMEM);
+
+    mix_presentation->sub_mixes = sub_mixes;
+
+    sub_mix = av_mallocz(sizeof(*sub_mix));
+    if (!sub_mix)
+        return AVERROR(ENOMEM);
+
+    sub_mix->submix_elements = av_calloc(num_submix_elements, sizeof(*sub_mix->submix_elements));
+    if (!sub_mix->submix_elements)
+        goto fail;
+
+    sub_mix->submix_layouts = av_calloc(num_submix_layouts, sizeof(*sub_mix->submix_layouts));
+    if (!sub_mix->submix_layouts)
+        goto fail;
+
+    for (int i = 0; i < num_submix_elements; i++) {
+        AVIAMFSubmixElement *submix_element = av_mallocz(sizeof(*submix_element));
+        if (!submix_element)
+            goto fail;
+
+        sub_mix->submix_elements[sub_mix->num_submix_elements++] = submix_element;
+    }
+
+    for (int i = 0; i < num_submix_layouts; i++) {
+        AVIAMFSubmixLayout *submix_layout = av_mallocz(sizeof(*submix_layout));
+        if (!submix_layout)
+            goto fail;
+
+        av_channel_layout_uninit(&submix_layout->sound_system);
+        sub_mix->submix_layouts[sub_mix->num_submix_layouts++] = submix_layout;
+    }
+
+    mix_presentation->sub_mixes[mix_presentation->num_sub_mixes++] = sub_mix;
+
+    return 0;
+fail:
+    for (int i = 0; i < sub_mix->num_submix_elements; i++) {
+        av_free(sub_mix->submix_elements[i]->element_mix_config);
+        av_free(sub_mix->submix_elements[i]);
+    }
+    for (int i = 0; i < sub_mix->num_submix_layouts; i++)
+        av_free(sub_mix->submix_layouts[i]);
+    av_free(sub_mix->submix_elements);
+    av_free(sub_mix->submix_layouts);
+    av_free(sub_mix);
+
+    return AVERROR(ENOMEM);
+}
+
+void avformat_iamf_mix_presentation_free(AVIAMFMixPresentation **pmix_presentation)
+{
+    AVIAMFMixPresentation *mix_presentation = *pmix_presentation;
+
+    if (!mix_presentation)
+        return;
+
+    for (int i; i < mix_presentation->num_sub_mixes; i++) {
+        AVIAMFSubmixPresentation *sub_mix = mix_presentation->sub_mixes[i];
+        for (int j; j < sub_mix->num_submix_elements; j++) {
+            AVIAMFSubmixElement *submix_element = sub_mix->submix_elements[j];
+            for (int k; k < mix_presentation->count_label; k++)
+                av_free(submix_element->mix_presentation_element_annotations[k]);
+            av_free(submix_element->mix_presentation_element_annotations);
+            av_free(submix_element->element_mix_config);
+            av_free(submix_element);
+        }
+        av_free(sub_mix->submix_elements);
+        for (int j; j < sub_mix->num_submix_layouts; j++) {
+            AVIAMFSubmixLayout *submix_layout = sub_mix->submix_layouts[j];
+            av_channel_layout_uninit(&submix_layout->sound_system);
+            av_free(submix_layout);
+        }
+        av_free(sub_mix->submix_layouts);
+        av_free(sub_mix->output_mix_config);
+        av_free(sub_mix);
+    }
+    for (int i; i < mix_presentation->count_label; i++) {
+        av_free(mix_presentation->language_label[i]);
+        av_free(mix_presentation->mix_presentation_annotations[i]);
+    }
+    av_free(mix_presentation->sub_mixes);
+    av_free(mix_presentation->language_label);
+    av_free(mix_presentation->mix_presentation_annotations);
+
+    av_freep(pmix_presentation);
+}
+
+AVIAMFParamDefinition *avformat_iamf_param_definition_alloc(enum AVIAMFParamDefinitionType type,
+                                                            unsigned int num_subblocks, size_t *out_size)
+{
+
+    struct MixGainStruct {
+        AVIAMFParamDefinition p;
+        AVIAMFMixGainParameterData m;
+    };
+    struct DemixStruct {
+        AVIAMFParamDefinition p;
+        AVIAMFDemixingInfoParameterData d;
+    };
+    struct ReconGainStruct {
+        AVIAMFParamDefinition p;
+        AVIAMFReconGainParameterData r;
+    };
+    size_t subblocks_offset, subblock_size;
+    size_t size;
+    AVIAMFParamDefinition *par;
+
+    switch (type) {
+    case AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN:
+        subblocks_offset = offsetof(struct MixGainStruct, m);
+        subblock_size = sizeof(AVIAMFMixGainParameterData);
+        break;
+    case AV_IAMF_PARAMETER_DEFINITION_DEMIXING:
+        subblocks_offset = offsetof(struct DemixStruct, d);
+        subblock_size = sizeof(AVIAMFDemixingInfoParameterData);
+        break;
+    case AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN:
+        subblocks_offset = offsetof(struct ReconGainStruct, r);
+        subblock_size = sizeof(AVIAMFReconGainParameterData);
+        break;
+    default:
+        return NULL;
+    }
+
+    size = subblocks_offset;
+    if (num_subblocks > (SIZE_MAX - size) / subblock_size)
+        return NULL;
+    size += subblock_size * num_subblocks;
+
+    par = av_mallocz(size);
+    if (!par)
+        return NULL;
+
+    par->param_definition_type = type;
+    par->num_subblocks = num_subblocks;
+    par->subblock_size = subblock_size;
+    par->subblocks_offset = subblocks_offset;
+    if (out_size)
+        *out_size = size;
+
+    return par;
+}
diff --git a/libavformat/iamf.h b/libavformat/iamf.h
new file mode 100644
index 0000000000..2743fd0c07
--- /dev/null
+++ b/libavformat/iamf.h
@@ -0,0 +1,228 @@ 
+/*
+ * Immersive Audio Model and Formats helper functions and defines
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVFORMAT_IAMF_H
+#define AVFORMAT_IAMF_H
+
+#include <stdint.h>
+#include <stddef.h>
+
+#include "libavutil/attributes.h"
+#include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
+
+struct AVStreamGroup;
+
+enum AVIAMFAudioElementType {
+    AV_IAMF_AUDIO_ELEMENT_TYPE_CHANNEL,
+    AV_IAMF_AUDIO_ELEMENT_TYPE_SCENE,
+};
+
+enum AVIAMFParamDefinitionType {
+    AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN,
+    AV_IAMF_PARAMETER_DEFINITION_DEMIXING,
+    AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN,
+};
+
+/**
+ * Parameters as defined in 3.6.1
+ */
+typedef struct AVIAMFParamDefinition {
+    size_t subblocks_offset;
+    size_t subblock_size;
+    unsigned int parameter_id;
+    enum AVIAMFParamDefinitionType param_definition_type;
+    unsigned int parameter_rate;
+    unsigned int param_definition_mode;
+    unsigned int duration;
+    unsigned int constant_subblock_duration;
+    unsigned int num_subblocks;
+} AVIAMFParamDefinition;
+
+AVIAMFParamDefinition *avformat_iamf_param_definition_alloc(enum AVIAMFParamDefinitionType param_definition_type,
+                                                            unsigned int num_subblocks, size_t *size);
+
+/**
+ * Get the subblock at the specified {@code idx}. Must be between 0 and num_subblocks.
+ * The @ref AVIAMFParamDefinition.param_definition_type "param definition type" defines
+ * the struct type of the returned pointer.
+ */
+static av_always_inline void*
+avformat_iamf_param_definition_get_subblock(AVIAMFParamDefinition *par, unsigned int idx)
+{
+    av_assert0(idx < par->num_subblocks);
+    return (void *)((uint8_t *)par + par->subblocks_offset + idx * par->subblock_size);
+}
+
+enum AVIAMFAnimationType {
+    AV_IAMF_ANIMATION_TYPE_STEP,
+    AV_IAMF_ANIMATION_TYPE_LINEAR,
+    AV_IAMF_ANIMATION_TYPE_BEZIER,
+};
+
+/**
+ * Mix Gain Parameter Data as defined in 3.8.1
+ *
+ * Subblocks in AVIAMFParamDefinition use this struct when the value or
+ * @ref AVIAMFParamDefinition.param_definition_type param_definition_type is
+ * AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN.
+ */
+typedef struct AVIAMFMixGainParameterData {
+    unsigned int subblock_duration;
+    enum AVIAMFAnimationType animation_type;
+    int start_point_value;
+    int end_point_value;
+    int control_point_value;
+    unsigned int control_point_relative_time;
+} AVIAMFMixGainParameterData;
+
+/**
+ * Demixing Info Parameter Data as defined in 3.8.2
+ *
+ * Subblocks in AVIAMFParamDefinition use this struct when the value or
+ * @ref AVIAMFParamDefinition.param_definition_type param_definition_type is
+ * AV_IAMF_PARAMETER_DEFINITION_DEMIXING.
+ */
+typedef struct AVIAMFDemixingInfoParameterData {
+    unsigned int subblock_duration;
+    unsigned int dmixp_mode;
+} AVIAMFDemixingInfoParameterData;
+
+/**
+ * Recon Gain Info Parameter Data as defined in 3.8.3
+ *
+ * Subblocks in AVIAMFParamDefinition use this struct when the value or
+ * @ref AVIAMFParamDefinition.param_definition_type param_definition_type is
+ * AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN.
+ */
+typedef struct AVIAMFReconGainParameterData {
+    unsigned int subblock_duration;
+} AVIAMFReconGainParameterData;
+
+typedef struct AVIAMFLayer {
+    AVChannelLayout ch_layout;
+    unsigned int substream_count;
+
+    unsigned int recon_gain_is_present;
+    /**
+     * Output gain flags as defined in 3.6.2
+     *
+     * This field is defined only if audio_element_type is
+     * AV_IAMF_AUDIO_ELEMENT_TYPE_CHANNEL, must be 0 otherwise.
+     */
+    unsigned int output_gain_flags;
+    /**
+     * Output gain as defined in 3.6.2
+     *
+     * Must be 0 if @ref output_gain_flags is 0.
+     */
+    int output_gain;
+    /**
+     * Ambisonics mode as defined in 3.6.3
+     *
+     * This field is defined only if audio_element_type is
+     * AV_IAMF_AUDIO_ELEMENT_TYPE_SCENE, must be 0 otherwise.
+     *
+     * If 0, channel_mapping is defined implicitly (Ambisonic Order)
+     * or explicitly (Custom Order with ambi channels) in @ref ch_layout.
+     * If 1, @ref demixing_matrix must be set.
+     */
+    unsigned int ambisonics_mode;
+    /**
+     * Demixing matrix as defined in 3.6.3
+     *
+     * Set only if @ref ambisonics_mode == 1, must be NULL otherwise.
+     */
+    int16_t *demixing_matrix;
+} AVIAMFLayer;
+
+typedef struct AVIAMFAudioElement {
+    AVIAMFLayer **layers;
+    unsigned int num_layers;
+
+    unsigned int codec_config_id;
+
+    AVIAMFParamDefinition *demixing_info;
+    AVIAMFParamDefinition *recon_gain_info;
+
+    /**
+     * Audio element type as defined in 3.6
+     */
+    enum AVIAMFAudioElementType audio_element_type;
+
+    /**
+     * Default weight value as defined in 3.6
+     */
+    unsigned int default_w;
+} AVIAMFAudioElement;
+
+AVIAMFAudioElement *avformat_iamf_audio_element_alloc(void);
+
+int avformat_iamf_audio_element_add_layer(AVIAMFAudioElement *audio_element);
+
+void avformat_iamf_audio_element_free(AVIAMFAudioElement **audio_element);
+
+typedef struct AVIAMFSubmixElement {
+    const struct AVStreamGroup *audio_element;
+
+    char **mix_presentation_element_annotations;
+
+    unsigned int headphones_rendering_mode;
+    AVIAMFParamDefinition *element_mix_config;
+    int default_mix_gain;
+} AVIAMFSubmixElement;
+
+typedef struct AVIAMFSubmixLayout {
+    unsigned int layout_type;
+    AVChannelLayout sound_system;
+
+    int integrated_loudness;
+    int digital_peak;
+} AVIAMFSubmixLayout;
+
+typedef struct AVIAMFSubmixPresentation {
+    AVIAMFSubmixElement **submix_elements;
+    unsigned int num_submix_elements;
+
+    AVIAMFSubmixLayout **submix_layouts;
+    unsigned int num_submix_layouts;
+
+    AVIAMFParamDefinition *output_mix_config;
+    int default_mix_gain;
+} AVIAMFSubmixPresentation;
+
+typedef struct AVIAMFMixPresentation {
+    AVIAMFSubmixPresentation **sub_mixes;
+    unsigned int num_sub_mixes;
+
+    unsigned int count_label;
+    char **language_label;
+    char **mix_presentation_annotations;
+} AVIAMFMixPresentation;
+
+AVIAMFMixPresentation *avformat_iamf_mix_presentation_alloc(void);
+
+int avformat_iamf_mix_presentation_add_submix(AVIAMFMixPresentation *mix_presentation,
+                                              unsigned int num_submix_elements,
+                                              unsigned int num_submix_layouts);
+
+void avformat_iamf_mix_presentation_free(AVIAMFMixPresentation **mix_presentation);
+
+#endif /* AVFORMAT_IAMF_H */
diff --git a/libavformat/iamf_internal.h b/libavformat/iamf_internal.h
new file mode 100644
index 0000000000..e4cfbcae33
--- /dev/null
+++ b/libavformat/iamf_internal.h
@@ -0,0 +1,86 @@ 
+/*
+ * Immersive Audio Model and Formats helper functions and defines
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVFORMAT_IAMF_INTERNAL_H
+#define AVFORMAT_IAMF_INTERNAL_H
+
+#include <stdint.h>
+
+#include "libavutil/channel_layout.h"
+
+#define MAX_IAMF_OBU_HEADER_SIZE (1 + 8 * 3)
+
+// OBU types (section 3.2).
+enum IAMF_OBU_Type {
+    // 0 reserved.
+    IAMF_OBU_IA_CODEC_CONFIG        = 0,
+    IAMF_OBU_IA_AUDIO_ELEMENT       = 1,
+    IAMF_OBU_IA_MIX_PRESENTATION    = 2,
+    IAMF_OBU_IA_PARAMETER_BLOCK     = 3,
+    IAMF_OBU_IA_TEMPORAL_DELIMITER  = 4,
+    IAMF_OBU_IA_AUDIO_FRAME         = 5,
+    IAMF_OBU_IA_AUDIO_FRAME_ID0     = 6,
+    IAMF_OBU_IA_AUDIO_FRAME_ID1     = 7,
+    IAMF_OBU_IA_AUDIO_FRAME_ID2     = 8,
+    IAMF_OBU_IA_AUDIO_FRAME_ID3     = 9,
+    IAMF_OBU_IA_AUDIO_FRAME_ID4     = 10,
+    IAMF_OBU_IA_AUDIO_FRAME_ID5     = 11,
+    IAMF_OBU_IA_AUDIO_FRAME_ID6     = 12,
+    IAMF_OBU_IA_AUDIO_FRAME_ID7     = 13,
+    IAMF_OBU_IA_AUDIO_FRAME_ID8     = 14,
+    IAMF_OBU_IA_AUDIO_FRAME_ID9     = 15,
+    IAMF_OBU_IA_AUDIO_FRAME_ID10    = 16,
+    IAMF_OBU_IA_AUDIO_FRAME_ID11    = 17,
+    IAMF_OBU_IA_AUDIO_FRAME_ID12    = 18,
+    IAMF_OBU_IA_AUDIO_FRAME_ID13    = 19,
+    IAMF_OBU_IA_AUDIO_FRAME_ID14    = 20,
+    IAMF_OBU_IA_AUDIO_FRAME_ID15    = 21,
+    IAMF_OBU_IA_AUDIO_FRAME_ID16    = 22,
+    IAMF_OBU_IA_AUDIO_FRAME_ID17    = 23,
+    // 24~30 reserved.
+    IAMF_OBU_IA_SEQUENCE_HEADER     = 31,
+};
+
+enum IAMF_Sound_System {
+    SOUND_SYSTEM_A_0_2_0  = 0,  // "Loudspeaker configuration for Sound System A"
+    SOUND_SYSTEM_B_0_5_0  = 1,  // "Loudspeaker configuration for Sound System B"
+    SOUND_SYSTEM_C_2_5_0  = 2,  // "Loudspeaker configuration for Sound System C"
+    SOUND_SYSTEM_D_4_5_0  = 3,  // "Loudspeaker configuration for Sound System D"
+    SOUND_SYSTEM_E_4_5_1  = 4,  // "Loudspeaker configuration for Sound System E"
+    SOUND_SYSTEM_F_3_7_0  = 5,  // "Loudspeaker configuration for Sound System F"
+    SOUND_SYSTEM_G_4_9_0  = 6,  // "Loudspeaker configuration for Sound System G"
+    SOUND_SYSTEM_H_9_10_3 = 7,  // "Loudspeaker configuration for Sound System H"
+    SOUND_SYSTEM_I_0_7_0  = 8,  // "Loudspeaker configuration for Sound System I"
+    SOUND_SYSTEM_J_4_7_0  = 9, // "Loudspeaker configuration for Sound System J"
+    SOUND_SYSTEM_10_2_7_0 = 10, // "Loudspeaker configuration for Sound System I" + Ltf + Rtf
+    SOUND_SYSTEM_11_2_3_0 = 11, // Front subset of "Loudspeaker configuration for Sound System J"
+    SOUND_SYSTEM_12_0_1_0 = 12, // Mono
+};
+
+extern const AVChannelLayout ff_iamf_scalable_ch_layouts[10];
+
+struct IAMFSoundSystemMap {
+    enum IAMF_Sound_System id;
+    AVChannelLayout layout;
+};
+
+extern const struct IAMFSoundSystemMap ff_iamf_sound_system_map[13];
+
+#endif /* AVFORMAT_IAMF_INTERNAL_H */
diff --git a/libavformat/iamfdec.c b/libavformat/iamfdec.c
new file mode 100644
index 0000000000..799addbc8c
--- /dev/null
+++ b/libavformat/iamfdec.c
@@ -0,0 +1,1646 @@ 
+/*
+ * Immersive Audio Model and Formats demuxer
+ * Copyright (c) 2023 James Almer <jamrial@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "config_components.h"
+
+#include "libavutil/avassert.h"
+#include "libavutil/common.h"
+#include "libavutil/intreadwrite.h"
+#include "libavutil/opt.h"
+#include "libavcodec/get_bits.h"
+#include "libavcodec/flac.h"
+#include "libavcodec/mpeg4audio.h"
+#include "libavcodec/put_bits.h"
+#include "avformat.h"
+#include "avio_internal.h"
+#include "demux.h"
+#include "iamf.h"
+#include "iamf_internal.h"
+#include "internal.h"
+#include "isom.h"
+
+typedef struct IAMFCodecConfig {
+    unsigned codec_config_id;
+    enum AVCodecID codec_id;
+    unsigned nb_samples;
+    int seek_preroll;
+    uint8_t *extradata;
+    int extradata_size;
+    int sample_rate;
+} IAMFCodecConfig;
+
+typedef struct IAMFAudioElement {
+    AVStreamGroup *stream_group;
+
+    AVStream **audio_substreams;
+    int num_substreams;
+} IAMFAudioElement;
+
+typedef struct IAMFMixPresentation {
+    AVStreamGroup *stream_group;
+} IAMFMixPresentation;
+
+typedef struct IAMFParamDefinition {
+    const AVIAMFAudioElement *audio_element;
+    AVIAMFParamDefinition *param;
+} IAMFParamDefinition;
+
+typedef struct IAMFDemuxContext {
+    IAMFCodecConfig *codec_configs;
+    int nb_codec_configs;
+    IAMFAudioElement *audio_elements;
+    int nb_audio_elements;
+    IAMFMixPresentation *mix_presentations;
+    int nb_mix_presentations;
+    IAMFParamDefinition *param_definitions;
+    int nb_param_definitions;
+
+    // Packet side data
+    AVIAMFParamDefinition *mix;
+    size_t mix_size;
+    AVIAMFParamDefinition *demix;
+    size_t demix_size;
+    AVIAMFParamDefinition *recon;
+    size_t recon_size;
+} IAMFDemuxContext;
+
+static inline unsigned get_leb128(GetBitContext *gb) {
+    int more, i = 0;
+    unsigned len = 0;
+
+    do {
+        unsigned bits;
+        int byte = get_bits(gb, 8);
+        more = byte & 0x80;
+        bits = byte & 0x7f;
+        if (i <= 3 || (i == 4 && bits < (1 << 4)))
+            len |= bits << (i * 7);
+        else if (bits)
+            return AVERROR_INVALIDDATA;
+        if (++i == 8 && more)
+            return AVERROR_INVALIDDATA;
+    } while (more);
+
+    return len;
+}
+
+static int parse_obu_header(const uint8_t *buf, int buf_size,
+                            unsigned *obu_size, int *start_pos, enum IAMF_OBU_Type *type)
+{
+    GetBitContext gb;
+    int ret, extension_flag, trimming, start;
+    unsigned size;
+
+    ret = init_get_bits8(&gb, buf, FFMIN(buf_size, MAX_IAMF_OBU_HEADER_SIZE));
+    if (ret < 0)
+        return ret;
+
+    *type          = get_bits(&gb, 5);
+    av_log(NULL, AV_LOG_DEBUG, "OBU type %d\n", *type);
+    /*redundant      =*/ get_bits1(&gb);
+    trimming       = get_bits1(&gb);
+    extension_flag = get_bits1(&gb);
+
+    *obu_size = get_leb128(&gb);
+    if (*obu_size > INT_MAX)
+        return AVERROR_INVALIDDATA;
+
+    start = get_bits_count(&gb) / 8;
+
+    if (trimming) {
+        get_leb128(&gb); // num_samples_to_trim_at_end
+        get_leb128(&gb); // num_samples_to_trim_at_start
+    }
+
+    if (extension_flag) {
+        unsigned extension_bytes = get_leb128(&gb);
+        if (extension_bytes > INT_MAX / 8)
+            return AVERROR_INVALIDDATA;
+        skip_bits_long(&gb, extension_bytes * 8);
+    }
+
+    if (get_bits_left(&gb) < 0)
+        return AVERROR_INVALIDDATA;
+
+    size = *obu_size + start;
+    if (size > INT_MAX)
+        return AVERROR_INVALIDDATA;
+
+    *obu_size -= get_bits_count(&gb) / 8 - start;
+    *start_pos = size - *obu_size;
+
+    av_log(NULL, AV_LOG_DEBUG, "OBU size %u\n", *obu_size);
+    return size;
+}
+
+//return < 0 if we need more data
+static int get_score(const uint8_t *buf, int buf_size, enum IAMF_OBU_Type type, int *seq)
+{
+    if (type == IAMF_OBU_IA_SEQUENCE_HEADER) {
+        if (buf_size < 4 || AV_RB32(buf) != MKBETAG('i','a','m','f'))
+            return 0;
+        *seq = 1;
+        return -1;
+    }
+    if (type >= IAMF_OBU_IA_CODEC_CONFIG && type <= IAMF_OBU_IA_TEMPORAL_DELIMITER)
+        return *seq ? -1 : 0;
+    if (type >= IAMF_OBU_IA_AUDIO_FRAME && type <= IAMF_OBU_IA_AUDIO_FRAME_ID17)
+        return *seq ? AVPROBE_SCORE_EXTENSION + 1 : 0;
+    return 0;
+}
+
+static int iamf_probe(const AVProbeData *p)
+{
+    unsigned obu_size;
+    enum IAMF_OBU_Type type;
+    int seq = 0, cnt = 0, start_pos;
+    int ret;
+
+    while (1) {
+        int size = parse_obu_header(p->buf + cnt, p->buf_size - cnt,
+                                    &obu_size, &start_pos, &type);
+        if (size < 0)
+            return 0;
+
+        ret = get_score(p->buf + cnt + start_pos,
+                        p->buf_size - cnt - start_pos,
+                        type, &seq);
+        if (ret >= 0)
+            return ret;
+
+        cnt += FFMIN(size, p->buf_size - cnt);
+    }
+    return 0;
+}
+
+static inline int leb(AVIOContext *pb, unsigned *len) {
+    int more, i = 0;
+    *len = 0;
+
+    do {
+        unsigned bits;
+        int byte = avio_r8(pb);
+        if (pb->error)
+            return pb->error;
+        if (pb->eof_reached)
+            return  AVERROR_INVALIDDATA;
+        more = byte & 0x80;
+        bits = byte & 0x7f;
+        if (i <= 3 || (i == 4 && bits < (1 << 4)))
+            *len |= bits << (i * 7);
+        else if (bits)
+            return AVERROR_INVALIDDATA;
+        if (++i == 8 && more)
+            return AVERROR_INVALIDDATA;
+    } while (more);
+
+    return i;
+}
+
+static int opus_decoder_config(AVFormatContext *s, AVIOContext *pb, int len,
+                               IAMFCodecConfig *codec_config)
+{
+    int left = len - avio_tell(pb);
+
+    if (left < 11)
+        return AVERROR_INVALIDDATA;
+
+    codec_config->extradata = av_malloc(left + 8);
+    if (!codec_config->extradata)
+        return AVERROR(ENOMEM);
+
+    AV_WB32(codec_config->extradata, MKBETAG('O','p','u','s'));
+    AV_WB32(codec_config->extradata + 4, MKBETAG('H','e','a','d'));
+    codec_config->extradata_size = avio_read(pb, codec_config->extradata + 8, left);
+    if (codec_config->extradata_size < left)
+        return AVERROR_INVALIDDATA;
+
+    codec_config->extradata_size += 8;
+    codec_config->sample_rate = 48000;
+
+    return 0;
+}
+
+static int aac_decoder_config(AVFormatContext *s, AVIOContext *pb, int len,
+                              IAMFCodecConfig *codec_config)
+{
+    MPEG4AudioConfig cfg = { 0 };
+    int object_type_id, codec_id, stream_type;
+    int ret, tag, left;
+
+    tag = avio_r8(pb);
+    if (tag != MP4DecConfigDescrTag)
+        return AVERROR_INVALIDDATA;
+
+    object_type_id = avio_r8(pb);
+    if (object_type_id != 0x40)
+        return AVERROR_INVALIDDATA;
+
+    stream_type = avio_r8(pb);
+    if (((stream_type >> 2) != 5) || ((stream_type >> 1) & 1))
+        return AVERROR_INVALIDDATA;
+
+    avio_skip(pb, 3); // buffer size db
+    avio_skip(pb, 4); // rc_max_rate
+    avio_skip(pb, 4); // avg bitrate
+
+    codec_id = ff_codec_get_id(ff_mp4_obj_type, object_type_id);
+    if (codec_id && codec_id != codec_config->codec_id)
+        return AVERROR_INVALIDDATA;
+
+    tag = avio_r8(pb);
+    if (tag != MP4DecSpecificDescrTag)
+        return AVERROR_INVALIDDATA;
+
+    left = len - avio_tell(pb);
+    if (left <= 0)
+        return AVERROR_INVALIDDATA;
+
+    codec_config->extradata = av_malloc(left);
+    if (!codec_config->extradata)
+        return AVERROR(ENOMEM);
+
+    codec_config->extradata_size = avio_read(pb, codec_config->extradata, left);
+    if (codec_config->extradata_size < left)
+        return AVERROR_INVALIDDATA;
+
+    ret = avpriv_mpeg4audio_get_config2(&cfg, codec_config->extradata,
+                                        codec_config->extradata_size, 1, s);
+    if (ret < 0)
+        return ret;
+
+    codec_config->sample_rate = cfg.sample_rate;
+
+    return 0;
+}
+
+static int flac_decoder_config(AVFormatContext *s, AVIOContext *pb, int len,
+                               IAMFCodecConfig *codec_config)
+{
+    int left;
+
+    avio_skip(pb, 4); // METADATA_BLOCK_HEADER
+
+    left = len - avio_tell(pb);
+    if (left < FLAC_STREAMINFO_SIZE)
+        return AVERROR_INVALIDDATA;
+
+    codec_config->extradata = av_malloc(left);
+    if (!codec_config->extradata)
+        return AVERROR(ENOMEM);
+
+    codec_config->extradata_size = avio_read(pb, codec_config->extradata, left);
+    if (codec_config->extradata_size < left)
+        return AVERROR_INVALIDDATA;
+
+    codec_config->sample_rate = AV_RB24(codec_config->extradata + 10) >> 4;
+
+    return 0;
+}
+
+static int ipcm_decoder_config(AVFormatContext *s, AVIOContext *pb, int len,
+                               IAMFCodecConfig *codec_config)
+{
+    static const enum AVSampleFormat sample_fmt[2][3] = {
+        { AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S24BE, AV_CODEC_ID_PCM_S32BE },
+        { AV_CODEC_ID_PCM_S16LE, AV_CODEC_ID_PCM_S24LE, AV_CODEC_ID_PCM_S32LE },
+    };
+    int sample_format = avio_r8(pb); // 0 = BE, 1 = LE
+    int sample_size = (avio_r8(pb) / 8 - 2); // 16, 24, 32
+    if (sample_format > 1 || sample_size > 2)
+        return AVERROR_INVALIDDATA;
+
+    codec_config->codec_id = sample_fmt[sample_format][sample_size];
+    codec_config->sample_rate = avio_rb32(pb);
+
+    if (len - avio_tell(pb))
+        return AVERROR_INVALIDDATA;
+
+    return 0;
+}
+
+static int codec_config_obu(AVFormatContext *s, int len)
+{
+    IAMFDemuxContext *const c = s->priv_data;
+    IAMFCodecConfig *codec_config = NULL;
+    FFIOContext b;
+    AVIOContext *pb;
+    uint8_t *buf;
+    enum AVCodecID avcodec_id;
+    unsigned codec_config_id, nb_samples, codec_id;
+    int16_t seek_preroll;
+    int ret;
+
+    buf = av_malloc(len);
+    if (!buf)
+        return AVERROR(ENOMEM);
+
+    ret = avio_read(s->pb, buf, len);
+    if (ret != len) {
+        if (ret >= 0)
+            ret = AVERROR_INVALIDDATA;
+        goto fail;
+    }
+
+    ffio_init_context(&b, buf, len, 0, NULL, NULL, NULL, NULL);
+    pb = &b.pub;
+
+    ret = leb(pb, &codec_config_id);
+    if (ret < 0)
+        goto fail;
+
+    codec_id = avio_rb32(pb);
+    ret = leb(pb, &nb_samples);
+    if (ret < 0)
+        goto fail;
+
+    seek_preroll = avio_rb16(pb);
+
+    switch(codec_id) {
+    case MKBETAG('O','p','u','s'):
+        avcodec_id = AV_CODEC_ID_OPUS;
+        break;
+    case MKBETAG('m','p','4','a'):
+        avcodec_id = AV_CODEC_ID_AAC;
+        break;
+    case MKBETAG('f','L','a','C'):
+        avcodec_id = AV_CODEC_ID_FLAC;
+        break;
+    default:
+        avcodec_id = AV_CODEC_ID_NONE;
+        break;
+    }
+
+    for (int i = 0; i < c->nb_codec_configs; i++)
+        if (c->codec_configs[i].codec_config_id == codec_config_id) {
+            ret = AVERROR_INVALIDDATA;
+            goto fail;
+        }
+
+    codec_config = av_dynarray2_add_nofree((void **)&c->codec_configs, &c->nb_codec_configs,
+                                            sizeof(*c->codec_configs), NULL);
+    if (!codec_config) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    memset(codec_config, 0, sizeof(*codec_config));
+
+    codec_config->codec_config_id = codec_config_id;
+    codec_config->codec_id = avcodec_id;
+    codec_config->nb_samples = nb_samples;
+    codec_config->seek_preroll = seek_preroll;
+
+    switch(codec_id) {
+    case MKBETAG('O','p','u','s'):
+        ret = opus_decoder_config(s, pb, len, codec_config);
+        break;
+    case MKBETAG('m','p','4','a'):
+        ret = aac_decoder_config(s, pb, len, codec_config);
+        break;
+    case MKBETAG('f','L','a','C'):
+        ret = flac_decoder_config(s, pb, len, codec_config);
+        break;
+    case MKBETAG('i','p','c','m'):
+        ret = ipcm_decoder_config(s, pb, len, codec_config);
+        break;
+    default:
+        break;
+    }
+    if (ret < 0)
+        goto fail;
+
+    av_log(s, AV_LOG_DEBUG, "%"PRId64" bytes left at the end of codec_config_obu\n", len - avio_tell(pb));
+
+    ret = 0;
+fail:
+    av_free(buf);
+    return ret;
+}
+
+static int update_extradata(AVFormatContext *s, AVStream *st)
+{
+    GetBitContext gb;
+    PutBitContext pb;
+    int ret;
+
+    switch(st->codecpar->codec_id) {
+    case AV_CODEC_ID_OPUS:
+        AV_WB8(st->codecpar->extradata + 9, st->codecpar->ch_layout.nb_channels);
+        break;
+    case AV_CODEC_ID_AAC: {
+        uint8_t buf[5];
+
+        init_put_bits(&pb, buf, sizeof(buf));
+        ret = init_get_bits8(&gb, st->codecpar->extradata, st->codecpar->extradata_size);
+        if (ret < 0)
+            return ret;
+
+        ret = get_bits(&gb, 5);
+        put_bits(&pb, 5, ret);
+        if (ret == AOT_ESCAPE) // violates section 3.11.2, but better check for it
+            put_bits(&pb, 6, get_bits(&gb, 6));
+        ret = get_bits(&gb, 4);
+        put_bits(&pb, 4, ret);
+        if (ret == 0x0f)
+            put_bits(&pb, 24, get_bits(&gb, 24));
+
+        skip_bits(&gb, 4);
+        put_bits(&pb, 4, st->codecpar->ch_layout.nb_channels); // set channel config
+        ret = put_bits_left(&pb);
+        put_bits(&pb, ret, get_bits(&gb, ret));
+        flush_put_bits(&pb);
+
+        memcpy(st->codecpar->extradata, buf, sizeof(buf));
+        break;
+    }
+    case AV_CODEC_ID_FLAC: {
+        uint8_t buf[13];
+
+        init_put_bits(&pb, buf, sizeof(buf));
+        ret = init_get_bits8(&gb, st->codecpar->extradata, st->codecpar->extradata_size);
+        if (ret < 0)
+            return ret;
+
+        put_bits32(&pb, get_bits_long(&gb, 32)); // min/max blocksize
+        put_bits64(&pb, 48, get_bits64(&gb, 48)); // min/max framesize
+        put_bits(&pb, 20, get_bits(&gb, 20)); // samplerate
+        skip_bits(&gb, 3);
+        put_bits(&pb, 3, st->codecpar->ch_layout.nb_channels - 1);
+        ret = put_bits_left(&pb);
+        put_bits(&pb, ret, get_bits(&gb, ret));
+        flush_put_bits(&pb);
+
+        memcpy(st->codecpar->extradata, buf, sizeof(buf));
+        break;
+    }
+    }
+
+    return 0;
+}
+
+static int scalable_channel_layout_config(AVFormatContext *s, AVIOContext *pb,
+                                          IAMFAudioElement *audio_element,
+                                          const IAMFCodecConfig *codec_config)
+{
+    AVStreamGroup *stg = audio_element->stream_group;
+    int num_layers, k = 0;
+
+    num_layers = avio_r8(pb) >> 5; // get_bits(&gb, 3);
+    // skip_bits(&gb, 5); //reserved
+
+    if (num_layers > 6)
+        return AVERROR_INVALIDDATA;
+
+    for (int i = 0; i < num_layers; i++) {
+        AVIAMFLayer *layer;
+        int loudspeaker_layout, output_gain_is_present_flag;
+        int coupled_substream_count;
+        int ret, byte = avio_r8(pb);
+
+        ret = avformat_iamf_audio_element_add_layer(stg->params.iamf_audio_element);
+        if (ret < 0)
+            return ret;
+
+        loudspeaker_layout = byte >> 4; // get_bits(&gb, 4);
+        output_gain_is_present_flag = (byte >> 3) & 1; //get_bits1(&gb);
+        layer = stg->params.iamf_audio_element->layers[i];
+        layer->recon_gain_is_present = (byte >> 2) & 1;
+        layer->substream_count = avio_r8(pb);
+        coupled_substream_count = avio_r8(pb);
+
+        if (output_gain_is_present_flag) {
+            layer->output_gain_flags = avio_r8(pb) >> 2;  // get_bits(&gb, 6);
+            layer->output_gain = sign_extend(avio_rb16(pb), 16);
+        }
+
+        if (loudspeaker_layout < 10)
+            av_channel_layout_copy(&layer->ch_layout, &ff_iamf_scalable_ch_layouts[loudspeaker_layout]);
+        else
+            layer->ch_layout = (AVChannelLayout){ .order = AV_CHANNEL_ORDER_UNSPEC,
+                                                          .nb_channels = layer->substream_count +
+                                                                         coupled_substream_count };
+
+        for (int j = 0; j < layer->substream_count; j++) {
+            AVStream *st = audio_element->audio_substreams[k++];
+
+            ret = avformat_stream_group_add_stream(stg, st);
+            if (ret < 0)
+                return ret;
+
+            st->codecpar->ch_layout = coupled_substream_count-- > 0 ? (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO :
+                                                                      (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
+
+            ret = update_extradata(s, st);
+            if (ret < 0)
+                return ret;
+
+            avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
+        }
+
+    }
+
+    return 0;
+}
+
+static int ambisonics_config(AVFormatContext *s, AVIOContext *pb,
+                             IAMFAudioElement *audio_element,
+                             const IAMFCodecConfig *codec_config)
+{
+    AVStreamGroup *stg = audio_element->stream_group;
+    AVIAMFLayer *layer;
+    unsigned ambisonics_mode;
+    int output_channel_count, substream_count, order;
+    int ret;
+
+    ret = leb(pb, &ambisonics_mode);
+    if (ret < 0)
+        return ret;
+
+    if (ambisonics_mode > 1)
+        return 0;
+
+    output_channel_count = avio_r8(pb);  // C
+    substream_count = avio_r8(pb);  // N
+    if (audio_element->num_substreams != substream_count)
+        return AVERROR_INVALIDDATA;
+
+    order = floor(sqrt(output_channel_count - 1));
+    /* incomplete order - some harmonics are missing */
+    if ((order + 1) * (order + 1) != output_channel_count)
+        return AVERROR_INVALIDDATA;
+
+    ret = avformat_iamf_audio_element_add_layer(stg->params.iamf_audio_element);
+    if (ret < 0)
+        return ret;
+
+    layer = stg->params.iamf_audio_element->layers[0];
+    layer->ambisonics_mode = ambisonics_mode;
+    layer->substream_count = substream_count;
+    if (ambisonics_mode == 0) {
+        for (int i = 0; i < substream_count; i++) {
+            AVStream *st = audio_element->audio_substreams[i];
+
+            st->codecpar->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
+
+            ret = avformat_stream_group_add_stream(stg, st);
+            if (ret < 0)
+                return ret;
+
+            ret = update_extradata(s, st);
+            if (ret < 0)
+                return ret;
+
+            avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
+        }
+
+        layer->ch_layout.order = AV_CHANNEL_ORDER_CUSTOM;
+        layer->ch_layout.nb_channels = output_channel_count;
+        layer->ch_layout.u.map = av_calloc(output_channel_count, sizeof(*layer->ch_layout.u.map));
+        if (!layer->ch_layout.u.map)
+            return AVERROR(ENOMEM);
+
+        for (int i = 0; i < output_channel_count; i++)
+            layer->ch_layout.u.map[i].id = avio_r8(pb) + AV_CHAN_AMBISONIC_BASE;
+    } else {
+        int coupled_substream_count = avio_r8(pb);  // M
+        int nb_demixing_matrix = substream_count + coupled_substream_count;
+        int demixing_matrix_size = nb_demixing_matrix * output_channel_count;
+
+        layer->ch_layout = (AVChannelLayout){ .order = AV_CHANNEL_ORDER_AMBISONIC, .nb_channels = output_channel_count };
+        layer->demixing_matrix = av_malloc_array(demixing_matrix_size, sizeof(*layer->demixing_matrix));
+        if (!layer->demixing_matrix)
+            return AVERROR(ENOMEM);
+
+        for (int i = 0; i < demixing_matrix_size; i++)
+            layer->demixing_matrix[i] = sign_extend(avio_rb16(pb), 16);
+
+        for (int i = 0; i < substream_count; i++) {
+            AVStream *st = audio_element->audio_substreams[i];
+
+            st->codecpar->ch_layout = coupled_substream_count-- > 0 ? (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO :
+                                                                      (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
+
+            ret = avformat_stream_group_add_stream(stg, st);
+            if (ret < 0)
+                return ret;
+
+            ret = update_extradata(s, st);
+            if (ret < 0)
+                return ret;
+
+            avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
+        }
+    }
+
+    return 0;
+}
+
+static int param_parse(AVFormatContext *s, AVIOContext *pb,
+                       unsigned int param_definition_type,
+                       const AVIAMFAudioElement *audio_element,
+                       AVIAMFParamDefinition **out_param_definition)
+{
+    IAMFDemuxContext *const c = s->priv_data;
+    IAMFParamDefinition *param_definition;
+    const IAMFParamDefinition *old_param = NULL;
+    unsigned int parameter_id, parameter_rate, param_definition_mode;
+    unsigned int duration, constant_subblock_duration, num_subblocks = 0;
+    int nb_param_definitions = c->nb_param_definitions, ret;
+
+    ret = leb(pb, &parameter_id);
+    if (ret < 0)
+        return ret;
+
+    for (int i = 0; i < c->nb_param_definitions; i++)
+        if (c->param_definitions[i].param->parameter_id == parameter_id) {
+            old_param = param_definition = &c->param_definitions[i];
+            break;
+        }
+
+    if (!old_param) {
+        param_definition = av_dynarray2_add_nofree((void **)&c->param_definitions, &nb_param_definitions,
+                                                   sizeof(*c->param_definitions), NULL);
+        if (!param_definition)
+            return AVERROR(ENOMEM);
+
+        memset(param_definition, 0, sizeof(*param_definition));
+    }
+
+    ret = leb(pb, &parameter_rate);
+    if (ret < 0)
+        return ret;
+
+    param_definition_mode = avio_r8(pb) >> 7;
+
+    if (old_param && (param_definition_mode != old_param->param->param_definition_mode ||
+                      param_definition_type != old_param->param->param_definition_type)) {
+        av_log(s, AV_LOG_ERROR, "Inconsistent param_definition_mode or param_definition_type values "
+                                "for parameter_id %d\n", parameter_id);
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (param_definition_mode == 0) {
+        ret = leb(pb, &duration);
+        if (ret < 0)
+            return ret;
+
+        ret = leb(pb, &constant_subblock_duration);
+        if (ret < 0)
+            return ret;
+
+        if (constant_subblock_duration == 0) {
+            ret = leb(pb, &num_subblocks);
+            if (ret < 0)
+                return ret;
+        } else
+            num_subblocks = duration / constant_subblock_duration;
+    }
+
+    if (old_param) {
+        if (num_subblocks != old_param->param->num_subblocks) {
+            av_log(s, AV_LOG_ERROR, "Inconsistent num_subblocks values for parameter_id %d\n", parameter_id);
+            return AVERROR_INVALIDDATA;
+        }
+    } else {
+        param_definition->param = avformat_iamf_param_definition_alloc(param_definition_type, num_subblocks, NULL);
+        if (!param_definition->param)
+            return AVERROR(ENOMEM);
+        param_definition->audio_element = audio_element;
+    }
+
+    for (int i = 0; i < num_subblocks; i++) {
+        void *subblock = avformat_iamf_param_definition_get_subblock(param_definition->param, i);
+        unsigned int subblock_duration = constant_subblock_duration;
+
+        if (constant_subblock_duration == 0) {
+            ret = leb(pb, &subblock_duration);
+            if (ret < 0) {
+                if (!old_param)
+                    av_freep(&param_definition->param);
+                return ret;
+            }
+        }
+
+        switch (param_definition_type) {
+        case AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN: {
+            AVIAMFMixGainParameterData *mix = subblock;
+            mix->subblock_duration = subblock_duration;
+            break;
+        }
+        case AV_IAMF_PARAMETER_DEFINITION_DEMIXING: {
+            AVIAMFDemixingInfoParameterData *demix = subblock;
+            demix->subblock_duration = subblock_duration;
+            // DemixingInfoParameterData
+            demix->dmixp_mode = avio_r8(pb) >> 5;
+            break;
+        }
+        case AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN: {
+            AVIAMFReconGainParameterData *recon = subblock;
+            recon->subblock_duration = subblock_duration;
+            break;
+        }
+        default:
+            if (!old_param)
+                av_freep(&param_definition->param);
+            return AVERROR_INVALIDDATA;
+        }
+    }
+
+    param_definition->param->parameter_id = parameter_id;
+    param_definition->param->parameter_rate = parameter_rate;
+    param_definition->param->param_definition_mode = param_definition_mode;
+    param_definition->param->duration = duration;
+    param_definition->param->constant_subblock_duration = constant_subblock_duration;
+    param_definition->param->num_subblocks = num_subblocks;
+
+    av_assert0(out_param_definition);
+    *out_param_definition = param_definition->param;
+
+    if (!old_param)
+        c->nb_param_definitions = nb_param_definitions;
+
+    return 0;
+}
+
+static int audio_element_obu(AVFormatContext *s, int len)
+{
+    IAMFDemuxContext *const c = s->priv_data;
+    const IAMFCodecConfig *codec_config = NULL;
+    AVIAMFAudioElement *avaudio_element;
+    IAMFAudioElement *audio_element;
+    FFIOContext b;
+    AVIOContext *pb;
+    uint8_t *buf;
+    unsigned audio_element_id, codec_config_id, num_substreams, num_parameters;
+    int audio_element_type, ret;
+
+    buf = av_malloc(len);
+    if (!buf)
+        return AVERROR(ENOMEM);
+
+    ret = avio_read(s->pb, buf, len);
+    if (ret != len) {
+        if (ret >= 0)
+            ret = AVERROR_INVALIDDATA;
+        goto fail;
+    }
+
+    ffio_init_context(&b, buf, len, 0, NULL, NULL, NULL, NULL);
+    pb = &b.pub;
+
+    ret = leb(pb, &audio_element_id);
+    if (ret < 0)
+        goto fail;
+
+    for (int i = 0; i < c->nb_audio_elements; i++)
+        if (c->audio_elements[i].stream_group->id == audio_element_id) {
+            av_log(s, AV_LOG_ERROR, "Duplicate audio_element_id %d\n", audio_element_id);
+            ret = AVERROR_INVALIDDATA;
+            goto fail;
+        }
+
+    audio_element_type = avio_r8(pb) >> 5;
+
+    ret = leb(pb, &codec_config_id);
+    if (ret < 0)
+        goto fail;
+
+    for (int i = 0; i < c->nb_codec_configs; i++) {
+        if (c->codec_configs[i].codec_config_id == codec_config_id) {
+            codec_config = &c->codec_configs[i];
+            break;
+        }
+    }
+
+    if (!codec_config) {
+        av_log(s, AV_LOG_ERROR, "Non existant codec config id %d referenced in an audio element\n", codec_config_id);
+        ret = AVERROR_INVALIDDATA;
+        goto fail;
+    }
+
+    if (codec_config->codec_id == AV_CODEC_ID_NONE) {
+        av_log(s, AV_LOG_DEBUG, "Unknown codec id referenced in an audio element. Ignoring\n");
+        ret = 0;
+        goto fail;
+    }
+
+    ret = leb(pb, &num_substreams);
+    if (ret < 0)
+        goto fail;
+
+    audio_element = av_dynarray2_add_nofree((void **)&c->audio_elements, &c->nb_audio_elements,
+                                            sizeof(*c->audio_elements), NULL);
+    if (!audio_element) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    memset(audio_element, 0, sizeof(*audio_element));
+
+    audio_element->audio_substreams = av_calloc(num_substreams, sizeof(*audio_element->audio_substreams));
+    if (!audio_element->audio_substreams) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    audio_element->stream_group = avformat_stream_group_create(s, AV_STREAM_GROUP_PARAMS_IAMF_AUDIO_ELEMENT);
+    if (!audio_element->stream_group)
+        return AVERROR(ENOMEM);
+    audio_element->stream_group->id = audio_element_id;
+    avaudio_element = audio_element->stream_group->params.iamf_audio_element;
+    avaudio_element->codec_config_id = codec_config_id;
+    avaudio_element->audio_element_type = audio_element_type;
+
+    audio_element->num_substreams = num_substreams;
+
+    for (int i = 0; i < num_substreams; i++) {
+        AVStream *st = audio_element->audio_substreams[i] = avformat_new_stream(s, NULL);
+        unsigned audio_substream_id;
+
+        if (!st) {
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
+
+        ret = leb(pb, &audio_substream_id);
+        if (ret < 0)
+            goto fail;
+
+        st->id = audio_substream_id;
+        st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
+        st->codecpar->codec_id   = codec_config->codec_id;
+        st->codecpar->frame_size = codec_config->nb_samples;
+        st->codecpar->sample_rate = codec_config->sample_rate;
+        st->codecpar->seek_preroll = codec_config->seek_preroll;
+        ffstream(st)->need_parsing = AVSTREAM_PARSE_HEADERS;
+
+        switch(st->codecpar->codec_id) {
+        case AV_CODEC_ID_AAC:
+        case AV_CODEC_ID_FLAC:
+        case AV_CODEC_ID_OPUS:
+            st->codecpar->extradata = av_malloc(codec_config->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE);
+            if (!st->codecpar->extradata) {
+                ret = AVERROR(ENOMEM);
+                goto fail;
+            }
+            memcpy(st->codecpar->extradata, codec_config->extradata, codec_config->extradata_size);
+            memset(st->codecpar->extradata + codec_config->extradata_size, 0, AV_INPUT_BUFFER_PADDING_SIZE);
+            st->codecpar->extradata_size = codec_config->extradata_size;
+            break;
+        }
+    }
+
+    ret = leb(pb, &num_parameters);
+    if (ret < 0)
+        goto fail;
+
+    for (int i = 0; i < num_parameters; i++) {
+        unsigned param_definition_type;
+
+        ret = leb(pb, &param_definition_type);
+        if (ret < 0)
+            goto fail;
+
+        if (param_definition_type == 0) {
+            ret = AVERROR_INVALIDDATA;
+            goto fail;
+        } else if (param_definition_type == 1) {
+            ret = param_parse(s, pb, param_definition_type, avaudio_element, &avaudio_element->demixing_info);
+            if (ret < 0)
+                goto fail;
+
+            avaudio_element->default_w = avio_r8(pb) >> 4;
+        } else if (param_definition_type == 2) {
+            ret = param_parse(s, pb, param_definition_type, avaudio_element, &avaudio_element->recon_gain_info);
+            if (ret < 0)
+                goto fail;
+        } else {
+            unsigned param_definition_size;
+            ret = leb(pb, &param_definition_size);
+            if (ret < 0)
+                goto fail;
+
+            avio_skip(pb, param_definition_size);
+        }
+    }
+
+    if (audio_element_type == AV_IAMF_AUDIO_ELEMENT_TYPE_CHANNEL) {
+        ret = scalable_channel_layout_config(s, pb, audio_element, codec_config);
+        if (ret < 0)
+            goto fail;
+    } else if (audio_element_type == AV_IAMF_AUDIO_ELEMENT_TYPE_SCENE) {
+        ret = ambisonics_config(s, pb, audio_element, codec_config);
+        if (ret < 0)
+            goto fail;
+    } else {
+        unsigned audio_element_config_size;
+        ret = leb(pb, &audio_element_config_size);
+        if (ret < 0)
+            goto fail;
+    }
+
+    av_log(s, AV_LOG_DEBUG, "%"PRId64" bytes left at the end of audio_element_obu\n", len - avio_tell(pb));
+
+    ret = 0;
+fail:
+    av_free(buf);
+
+    return ret;
+}
+
+static int label_string(AVFormatContext *s, AVIOContext *pb, char **label)
+{
+    uint8_t buf[128];
+
+    avio_get_str(pb, sizeof(buf), buf, sizeof(buf));
+
+    if (pb->error)
+        return pb->error;
+    if (pb->eof_reached)
+        return AVERROR_INVALIDDATA;
+    *label = av_strdup(buf);
+    if (!*label)
+        return AVERROR(ENOMEM);
+
+    return 0;
+}
+
+static int mix_presentation_obu(AVFormatContext *s, int len)
+{
+    IAMFDemuxContext *const c = s->priv_data;
+    AVIAMFMixPresentation *mix_presentation;
+    IAMFMixPresentation *mixi;
+    FFIOContext b;
+    AVIOContext *pb;
+    uint8_t *buf;
+    unsigned mix_presentation_id;
+    int ret;
+
+    buf = av_malloc(len);
+    if (!buf)
+        return AVERROR(ENOMEM);
+
+    ret = avio_read(s->pb, buf, len);
+    if (ret != len) {
+        if (ret >= 0)
+            ret = AVERROR_INVALIDDATA;
+        goto fail;
+    }
+
+    ffio_init_context(&b, buf, len, 0, NULL, NULL, NULL, NULL);
+    pb = &b.pub;
+
+    ret = leb(pb, &mix_presentation_id);
+    if (ret < 0)
+        goto fail;
+
+    for (int i = 0; i < c->nb_mix_presentations; i++)
+        if (c->mix_presentations[i].stream_group->id == mix_presentation_id) {
+            av_log(s, AV_LOG_ERROR, "Duplicate mix_presentation_id %d\n", mix_presentation_id);
+            ret = AVERROR_INVALIDDATA;
+            goto fail;
+        }
+
+    mixi = av_dynarray2_add_nofree((void **)&c->mix_presentations, &c->nb_mix_presentations,
+                                   sizeof(*c->mix_presentations), NULL);
+    if (!mixi) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    memset(mixi, 0, sizeof(*mixi));
+    mixi->stream_group = avformat_stream_group_create(s, AV_STREAM_GROUP_PARAMS_IAMF_MIX_PRESENTATION);
+    if (!mixi->stream_group) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    mixi->stream_group->id = mix_presentation_id;
+    mix_presentation = mixi->stream_group->params.iamf_mix_presentation;
+
+    ret = leb(pb, &mix_presentation->count_label);
+    if (ret < 0)
+        goto fail;
+
+    mix_presentation->language_label = av_calloc(mix_presentation->count_label, sizeof(*mix_presentation->language_label));
+    if (!mix_presentation->language_label) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    for (int i = 0; i < mix_presentation->count_label; i++) {
+        ret = label_string(s, pb, &mix_presentation->language_label[i]);
+        if (ret < 0)
+            goto fail;
+    }
+
+    mix_presentation->mix_presentation_annotations = av_calloc(mix_presentation->count_label, sizeof(*mix_presentation->mix_presentation_annotations));
+    if (!mix_presentation->mix_presentation_annotations) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    for (int i = 0; i < mix_presentation->count_label; i++) {
+        ret = label_string(s, pb, &mix_presentation->mix_presentation_annotations[i]);
+        if (ret < 0)
+            goto fail;
+    }
+
+    ret = leb(pb, &mix_presentation->num_sub_mixes);
+    if (ret < 0)
+        goto fail;
+
+    mix_presentation->sub_mixes = av_calloc(mix_presentation->num_sub_mixes, sizeof(*mix_presentation->sub_mixes));
+    if (!mix_presentation->sub_mixes) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    for (int i = 0; i < mix_presentation->num_sub_mixes; i++) {
+        AVIAMFSubmixPresentation *sub_mix;
+
+        sub_mix = mix_presentation->sub_mixes[i] = av_mallocz(sizeof(*sub_mix));
+        if (!sub_mix) {
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
+
+        ret = leb(pb, &sub_mix->num_submix_elements);
+        if (ret < 0)
+            goto fail;
+
+        sub_mix->submix_elements = av_calloc(sub_mix->num_submix_elements, sizeof(*sub_mix->submix_elements));
+        if (!sub_mix->submix_elements) {
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
+
+        for (int j = 0; j < sub_mix->num_submix_elements; j++) {
+            AVIAMFSubmixElement *submix_element;
+            IAMFAudioElement *audio_element = NULL;
+            unsigned int audio_element_id, rendering_config_extension_size;
+
+            submix_element = sub_mix->submix_elements[j] = av_mallocz(sizeof(*submix_element));
+            if (!submix_element) {
+                ret = AVERROR(ENOMEM);
+                goto fail;
+            }
+
+            ret = leb(pb, &audio_element_id);
+            if (ret < 0)
+                goto fail;
+
+            for (int k = 0; k < c->nb_audio_elements; k++)
+                if (c->audio_elements[k].stream_group->id == audio_element_id) {
+                    audio_element = &c->audio_elements[k];
+                    submix_element->audio_element = audio_element->stream_group;
+                }
+
+            if (!audio_element) {
+                av_log(s, AV_LOG_ERROR, "Invalid Audio Element with id %u referenced by Mix Parameters %u\n", audio_element_id, mix_presentation_id);
+                ret = AVERROR_INVALIDDATA;
+                goto fail;
+            }
+
+            for (int k = 0; k < audio_element->num_substreams; k++) {
+                ret = avformat_stream_group_add_stream(mixi->stream_group, audio_element->audio_substreams[k]);
+                if (ret < 0 && ret != AVERROR(EEXIST))
+                    goto fail;
+            }
+
+            submix_element->mix_presentation_element_annotations = av_calloc(mix_presentation->count_label, sizeof(*submix_element->mix_presentation_element_annotations));
+            if (!submix_element->mix_presentation_element_annotations) {
+                ret = AVERROR(ENOMEM);
+                goto fail;
+            }
+
+            for (int k = 0; k < mix_presentation->count_label; k++) {
+                ret = label_string(s, pb, &submix_element->mix_presentation_element_annotations[k]);
+                if (ret < 0)
+                    goto fail;
+            }
+
+            submix_element->headphones_rendering_mode = avio_r8(pb) >> 6;
+
+            ret = leb(pb, &rendering_config_extension_size);
+            if (ret < 0)
+                goto fail;
+            avio_skip(pb, rendering_config_extension_size);
+
+            ret = param_parse(s, pb, AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN, NULL, &submix_element->element_mix_config);
+            if (ret < 0)
+                goto fail;
+            submix_element->default_mix_gain = sign_extend(avio_rb16(pb), 16);
+        }
+        ret = param_parse(s, pb, AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN, NULL, &sub_mix->output_mix_config);
+        if (ret < 0)
+            goto fail;
+        sub_mix->default_mix_gain = sign_extend(avio_rb16(pb), 16);
+
+        ret = leb(pb, &sub_mix->num_submix_layouts);
+        if (ret < 0)
+            goto fail;
+
+        sub_mix->submix_layouts = av_calloc(sub_mix->num_submix_layouts, sizeof(*sub_mix->submix_layouts));
+        if (!sub_mix->submix_layouts) {
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
+
+        for (int j = 0; j < sub_mix->num_submix_layouts; j++) {
+            AVIAMFSubmixLayout *submix_layout;
+            int info_type;
+            int byte = avio_r8(pb);
+
+            submix_layout = sub_mix->submix_layouts[j] = av_mallocz(sizeof(*submix_layout));
+            if (!submix_layout) {
+                ret = AVERROR(ENOMEM);
+                goto fail;
+            }
+
+            submix_layout->layout_type = byte >> 6;
+            if (submix_layout->layout_type < 2 && submix_layout->layout_type > 3) {
+                av_log(s, AV_LOG_ERROR, "Invalid Layout type %u in a submix from Mix Presentation %u\n", submix_layout->layout_type, mix_presentation_id);
+                ret = AVERROR_INVALIDDATA;
+                goto fail;
+            }
+            if (submix_layout->layout_type == 2) {
+                int sound_system;
+                sound_system = (byte >> 2) & 0xF;
+                av_channel_layout_copy(&submix_layout->sound_system, &ff_iamf_sound_system_map[sound_system].layout);
+            }
+
+            info_type = avio_r8(pb);
+            submix_layout->integrated_loudness = sign_extend(avio_rb16(pb), 16);
+            submix_layout->digital_peak = sign_extend(avio_rb16(pb), 16);
+
+            if (info_type & 1)
+                sign_extend(avio_rb16(pb), 16); // true_peak
+
+            if (info_type & 2) {
+                unsigned int num_anchored_loudness = avio_r8(pb);
+
+                for (int k = 0; k < num_anchored_loudness; k++) {
+                    avio_r8(pb); // anchor_element
+                    sign_extend(avio_rb16(pb), 16); // anchored_loudness
+                }
+            }
+
+            if (info_type & 0xFC) {
+                unsigned int info_type_size;
+                ret = leb(pb, &info_type_size);
+                if (ret < 0)
+                    goto fail;
+
+                avio_skip(pb, info_type_size);
+            }
+        }
+    }
+
+    av_log(s, AV_LOG_DEBUG, "%"PRId64" bytes left at the end of mixing_parameters_obu\n", len - avio_tell(pb));
+
+    ret = 0;
+fail:
+    av_free(buf);
+
+    return ret;
+}
+
+static int iamf_read_header(AVFormatContext *s)
+{
+    IAMFDemuxContext *const c = s->priv_data;
+    uint8_t header[MAX_IAMF_OBU_HEADER_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    int ret;
+
+    av_log(s, AV_LOG_DEBUG, "HEADER\n");
+    while (1) {
+        unsigned obu_size;
+        enum IAMF_OBU_Type type;
+        int start_pos, len, size;
+
+        if ((ret = ffio_ensure_seekback(s->pb, MAX_IAMF_OBU_HEADER_SIZE)) < 0)
+            return ret;
+        size = avio_read(s->pb, header, MAX_IAMF_OBU_HEADER_SIZE);
+        if (size < 0)
+            return size;
+
+        len = parse_obu_header(header, size, &obu_size, &start_pos, &type);
+        if (len < 0) {
+            av_log(s, AV_LOG_ERROR, "Failed to read obu\n");
+            return len;
+        }
+
+        if (type >= IAMF_OBU_IA_PARAMETER_BLOCK && type < IAMF_OBU_IA_SEQUENCE_HEADER) {
+            avio_seek(s->pb, -size, SEEK_CUR);
+            break;
+        }
+
+        avio_seek(s->pb, -(size - start_pos), SEEK_CUR);
+        switch (type) {
+        case IAMF_OBU_IA_CODEC_CONFIG:
+            ret = codec_config_obu(s, obu_size);
+            break;
+        case IAMF_OBU_IA_AUDIO_ELEMENT:
+            ret = audio_element_obu(s, obu_size);
+            break;
+        case IAMF_OBU_IA_MIX_PRESENTATION:
+            ret = mix_presentation_obu(s, obu_size);
+            break;
+        case IAMF_OBU_IA_TEMPORAL_DELIMITER:
+            av_freep(&c->mix);
+            c->mix_size = 0;
+            av_freep(&c->demix);
+            c->demix_size = 0;
+            av_freep(&c->recon);
+            c->recon_size = 0;
+            break;
+        default: {
+            int64_t offset = avio_skip(s->pb, obu_size);
+            if (offset < 0)
+                ret = offset;
+            break;
+        }
+        }
+        if (ret < 0)
+            return ret;
+    }
+
+    return 0;
+}
+
+static AVStream *find_stream_by_id(AVFormatContext *s, int id)
+{
+    for (int i = 0; i < s->nb_streams; i++)
+        if (s->streams[i]->id == id)
+            return s->streams[i];
+
+    av_log(s, AV_LOG_ERROR, "Invalid stream id %d\n", id);
+    return NULL;
+}
+
+static int audio_frame_obu(AVFormatContext *s, AVPacket *pkt, int len,
+                           enum IAMF_OBU_Type type, int id_in_bitstream)
+{
+    const IAMFDemuxContext *const c = s->priv_data;
+    AVStream *st;
+    int ret, audio_substream_id;
+
+    if (id_in_bitstream) {
+        unsigned explicit_audio_substream_id;
+        ret = leb(s->pb, &explicit_audio_substream_id);
+        if (ret < 0)
+            return ret;
+        len -= ret;
+        audio_substream_id = explicit_audio_substream_id;
+    } else
+        audio_substream_id = type - IAMF_OBU_IA_AUDIO_FRAME_ID0;
+
+    st = find_stream_by_id(s, audio_substream_id);
+    if (!st)
+        return AVERROR_INVALIDDATA;
+
+    ret = av_get_packet(s->pb, pkt, len);
+    if (ret < 0)
+        return ret;
+    if (ret != len)
+        return AVERROR_INVALIDDATA;
+
+    if (c->mix) {
+        uint8_t *side_data = av_packet_new_side_data(pkt, AV_PKT_DATA_IAMF_MIX_GAIN_PARAM, c->mix_size);
+        if (!side_data)
+            return AVERROR(ENOMEM);
+        memcpy(side_data, c->mix, c->mix_size);
+    }
+    if (c->demix) {
+        uint8_t *side_data = av_packet_new_side_data(pkt, AV_PKT_DATA_IAMF_DEMIXING_INFO_PARAM, c->demix_size);
+        if (!side_data)
+            return AVERROR(ENOMEM);
+        memcpy(side_data, c->demix, c->demix_size);
+    }
+    if (c->recon) {
+        uint8_t *side_data = av_packet_new_side_data(pkt, AV_PKT_DATA_IAMF_RECON_GAIN_INFO_PARAM, c->recon_size);
+        if (!side_data)
+            return AVERROR(ENOMEM);
+        memcpy(side_data, c->recon, c->recon_size);
+    }
+
+    pkt->stream_index = st->index;
+    return 0;
+}
+
+static const IAMFParamDefinition *get_param_definition(AVFormatContext *s, unsigned int parameter_id)
+{
+    const IAMFDemuxContext *const c = s->priv_data;
+    const IAMFParamDefinition *param_definition = NULL;
+
+    for (int i = 0; i < c->nb_param_definitions; i++)
+        if (c->param_definitions[i].param->parameter_id == parameter_id) {
+            param_definition = &c->param_definitions[i];
+            break;
+        }
+
+    return param_definition;
+}
+
+static int parameter_block_obu(AVFormatContext *s, int len)
+{
+    IAMFDemuxContext *const c = s->priv_data;
+    const IAMFParamDefinition *param_definition;
+    const AVIAMFParamDefinition *param;
+    AVIAMFParamDefinition *out_param = NULL;
+    FFIOContext b;
+    AVIOContext *pb;
+    uint8_t *buf;
+    unsigned int duration, constant_subblock_duration;
+    unsigned int num_subblocks;
+    unsigned int parameter_id;
+    size_t out_param_size;
+    int ret;
+
+    buf = av_malloc(len);
+    if (!buf)
+        return AVERROR(ENOMEM);
+
+    ret = avio_read(s->pb, buf, len);
+    if (ret != len) {
+        if (ret >= 0)
+            ret = AVERROR_INVALIDDATA;
+        goto fail;
+    }
+
+    ffio_init_context(&b, buf, len, 0, NULL, NULL, NULL, NULL);
+    pb = &b.pub;
+
+    ret = leb(pb, &parameter_id);
+    if (ret < 0)
+        goto fail;
+
+    param_definition = get_param_definition(s, parameter_id);
+    if (!param_definition) {
+        av_log(s, AV_LOG_VERBOSE, "Non existant parameter_id %d referenced in a parameter block. Ignoring\n", parameter_id);
+        ret = 0;
+        goto fail;
+    }
+
+    param = param_definition->param;
+    if (param->param_definition_mode) {
+        ret = leb(pb, &duration);
+        if (ret < 0)
+            goto fail;
+
+        ret = leb(pb, &constant_subblock_duration);
+        if (ret < 0)
+            goto fail;
+
+        if (constant_subblock_duration == 0) {
+            ret = leb(pb, &num_subblocks);
+            if (ret < 0)
+                goto fail;
+        } else
+            num_subblocks = duration / constant_subblock_duration;
+    } else {
+        duration = param->duration;
+        constant_subblock_duration = param->constant_subblock_duration;
+        num_subblocks = param->num_subblocks;
+        if (!num_subblocks)
+            num_subblocks = duration / constant_subblock_duration;
+    }
+
+    out_param = avformat_iamf_param_definition_alloc(param->param_definition_type, num_subblocks, &out_param_size);
+    if (!out_param) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    out_param->parameter_id = param->parameter_id;
+    out_param->param_definition_type = param->param_definition_type;
+    out_param->parameter_rate = param->parameter_rate;
+    out_param->param_definition_mode = param->param_definition_mode;
+    out_param->duration = duration;
+    out_param->constant_subblock_duration = constant_subblock_duration;
+    out_param->num_subblocks = num_subblocks;
+
+    for (int i = 0; i < num_subblocks; i++) {
+        void *subblock = avformat_iamf_param_definition_get_subblock(out_param, i);
+        unsigned int subblock_duration;
+
+        if (param->param_definition_mode && !constant_subblock_duration) {
+            ret = leb(pb, &subblock_duration);
+            if (ret < 0)
+                goto fail;
+
+        } else {
+            switch (param->param_definition_type) {
+            case AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN:
+                subblock_duration = ((AVIAMFMixGainParameterData *)subblock)->subblock_duration;
+                break;
+            case AV_IAMF_PARAMETER_DEFINITION_DEMIXING:
+                subblock_duration = ((AVIAMFDemixingInfoParameterData *)subblock)->subblock_duration;
+                break;
+            case AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN:
+                subblock_duration = ((AVIAMFReconGainParameterData *)subblock)->subblock_duration;
+                break;
+            default:
+                return AVERROR_INVALIDDATA;
+            }
+        }
+
+        switch (param->param_definition_type) {
+        case AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN: {
+            AVIAMFMixGainParameterData *mix = subblock;
+
+            ret = leb(pb, &mix->animation_type);
+            if (ret < 0)
+                goto fail;
+
+            if (mix->animation_type > AV_IAMF_ANIMATION_TYPE_BEZIER) {
+                ret = 0;
+                av_free(out_param);
+                goto fail;
+            }
+
+            mix->start_point_value = sign_extend(avio_rb16(pb), 16);
+            if (mix->animation_type >= AV_IAMF_ANIMATION_TYPE_LINEAR)
+                mix->end_point_value = sign_extend(avio_rb16(pb), 16);
+            if (mix->animation_type == AV_IAMF_ANIMATION_TYPE_BEZIER) {
+                mix->control_point_value = sign_extend(avio_rb16(pb), 16);
+                mix->control_point_relative_time = avio_r8(pb);
+            }
+            mix->subblock_duration = subblock_duration;
+            break;
+        }
+        case AV_IAMF_PARAMETER_DEFINITION_DEMIXING: {
+            AVIAMFDemixingInfoParameterData *demix = subblock;
+
+            demix->dmixp_mode = avio_r8(pb) >> 5;
+            demix->subblock_duration = subblock_duration;
+            break;
+        }
+        case AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN: {
+            const AVIAMFAudioElement *audio_element = param_definition->audio_element;
+
+            av_assert0(audio_element);
+            for (int i = 0; i < audio_element->num_layers; i++) {
+                const AVIAMFLayer *layer = audio_element->layers[i];
+                if (layer->recon_gain_is_present) {
+                    unsigned int recon_gain_flags, bitcount;
+                    ret = leb(pb, &recon_gain_flags);
+                    if (ret < 0)
+                        goto fail;
+
+                    bitcount = 7 + 5 * !!(recon_gain_flags & 0x80);
+                    recon_gain_flags = (recon_gain_flags & 0x7F) | ((recon_gain_flags & 0xFF00) >> 1);
+                    for (int j = 0; j < bitcount; j++) {
+                        if (recon_gain_flags & (1 << j)) {
+                            avio_r8(pb); // recon_gain
+                        }
+                    }
+                }
+            }
+            break;
+        }
+        default: {
+            unsigned parameter_data_size;
+            ret = leb(pb, &parameter_data_size);
+            if (ret < 0)
+                goto fail;
+
+            avio_skip(pb, parameter_data_size);
+            break;
+        }
+        }
+    }
+
+    av_log(s, AV_LOG_DEBUG, "%"PRId64" bytes left at the end of parameter_block_obu\n", len - avio_tell(pb));
+
+    switch (param->param_definition_type) {
+    case AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN:
+        av_free(c->mix);
+        c->mix = out_param;
+        c->mix_size = out_param_size;
+        break;
+    case AV_IAMF_PARAMETER_DEFINITION_DEMIXING:
+        av_free(c->demix);
+        c->demix = out_param;
+        c->demix_size = out_param_size;
+        break;
+    case AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN: // TODO
+        ret = 0;
+        av_free(out_param);
+        goto fail;
+    default:
+        return AVERROR_INVALIDDATA;
+    }
+
+    ret = 0;
+fail:
+    if (ret < 0)
+        av_free(out_param);
+    av_free(buf);
+
+    return ret;
+}
+
+static int iamf_read_packet(AVFormatContext *s, AVPacket *pkt)
+{
+    uint8_t header[MAX_IAMF_OBU_HEADER_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    unsigned obu_size;
+    int ret;
+
+    av_log(s, AV_LOG_DEBUG, "PACKET\n");
+    while (1) {
+        enum IAMF_OBU_Type type;
+        int len, size, start_pos;
+
+        if ((ret = ffio_ensure_seekback(s->pb, MAX_IAMF_OBU_HEADER_SIZE)) < 0)
+            return ret;
+        size = avio_read(s->pb, header, MAX_IAMF_OBU_HEADER_SIZE);
+        if (size < 0)
+            return size;
+
+        len = parse_obu_header(header, size, &obu_size, &start_pos, &type);
+        if (len < 0) {
+            av_log(s, AV_LOG_ERROR, "Failed to read obu\n");
+            return len;
+        }
+        avio_seek(s->pb, -(size - start_pos), SEEK_CUR);
+
+        if (type == IAMF_OBU_IA_AUDIO_FRAME)
+            return audio_frame_obu(s, pkt, obu_size, type, 1);
+        else if (type >= IAMF_OBU_IA_AUDIO_FRAME_ID0 && type <= IAMF_OBU_IA_AUDIO_FRAME_ID17)
+            return audio_frame_obu(s, pkt, obu_size, type, 0);
+        else if (type == IAMF_OBU_IA_PARAMETER_BLOCK) {
+            ret = parameter_block_obu(s, obu_size);
+            if (ret < 0)
+                return ret;
+        } else {
+            int64_t offset = avio_skip(s->pb, obu_size);
+            if (offset < 0)
+                ret = offset;
+            break;
+        }
+    }
+
+    return ret;
+}
+
+static int iamf_read_close(AVFormatContext *s)
+{
+    IAMFDemuxContext *const c = s->priv_data;
+
+    for (int i = 0; i < c->nb_codec_configs; i++)
+        av_free(c->codec_configs[i].extradata);
+    av_freep(&c->codec_configs);
+    c->nb_codec_configs = 0;
+
+    for (int i = 0; i < c->nb_audio_elements; i++)
+        av_free(c->audio_elements[i].audio_substreams);
+    av_freep(&c->audio_elements);
+    c->nb_audio_elements = 0;
+
+    av_freep(&c->mix_presentations);
+    c->nb_mix_presentations = 0;
+
+    av_freep(&c->param_definitions);
+    c->nb_param_definitions = 0;
+
+    av_freep(&c->mix);
+    c->mix_size = 0;
+    av_freep(&c->demix);
+    c->demix_size = 0;
+    av_freep(&c->recon);
+    c->recon_size = 0;
+    return 0;
+}
+
+const AVInputFormat ff_iamf_demuxer = {
+    .name           = "iamf",
+    .long_name      = NULL_IF_CONFIG_SMALL("Raw Immersive Audio Model and Formats"),
+    .priv_data_size = sizeof(IAMFDemuxContext),
+    .flags_internal = FF_FMT_INIT_CLEANUP,
+    .read_probe     = iamf_probe,
+    .read_header    = iamf_read_header,
+    .read_packet    = iamf_read_packet,
+    .read_close     = iamf_read_close,
+    .extensions     = "iamf",
+    .flags          = AVFMT_GENERIC_INDEX | AVFMT_NO_BYTE_SEEK | AVFMT_NOTIMESTAMPS | AVFMT_SHOW_IDS,
+};
diff --git a/libavformat/options.c b/libavformat/options.c
index 09eb13e97a..115f95d48a 100644
--- a/libavformat/options.c
+++ b/libavformat/options.c
@@ -20,6 +20,7 @@ 
 #include "avformat.h"
 #include "avio_internal.h"
 #include "demux.h"
+#include "iamf.h"
 #include "internal.h"
 
 #include "libavcodec/avcodec.h"
@@ -345,7 +346,16 @@  AVStreamGroup *avformat_stream_group_create(AVFormatContext *s,
 
     stg->type = type;
     switch (type) {
-    // Structs in the union are allocated here
+    case AV_STREAM_GROUP_PARAMS_IAMF_AUDIO_ELEMENT:
+        stg->params.iamf_audio_element = avformat_iamf_audio_element_alloc();
+        if (!stg->params.iamf_audio_element)
+            goto fail;
+        break;
+    case AV_STREAM_GROUP_PARAMS_IAMF_MIX_PRESENTATION:
+        stg->params.iamf_mix_presentation = avformat_iamf_mix_presentation_alloc();
+        if (!stg->params.iamf_mix_presentation)
+            goto fail;
+        // fall-through
     default:
         break;
     }
@@ -356,6 +366,9 @@  AVStreamGroup *avformat_stream_group_create(AVFormatContext *s,
     s->stream_groups[s->nb_stream_groups++] = stg;
 
     return stg;
+fail:
+    ff_free_stream_group(&stg);
+    return NULL;
 }
 
 static int stream_group_add_stream(AVStreamGroup *stg, const AVStream *st)