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ffmpeg/libavformat/iamf_writer.c
James Almer edc7b67508 avformat/iamf: use the new Binaural channel layout 
Signed-off-by: James Almer <jamrial@gmail.com>
2024-11-13 12:38:04 -03:00

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/*
* Immersive Audio Model and Formats muxing helpers and structs
* 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 "libavutil/channel_layout.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/iamf.h"
#include "libavutil/mem.h"
#include "libavcodec/get_bits.h"
#include "libavcodec/put_bits.h"
#include "avformat.h"
#include "avio_internal.h"
#include "iamf.h"
#include "iamf_writer.h"
static int update_extradata(IAMFCodecConfig *codec_config)
{
GetBitContext gb;
PutBitContext pb;
int ret;
switch(codec_config->codec_id) {
case AV_CODEC_ID_OPUS:
if (codec_config->extradata_size != 19)
return AVERROR_INVALIDDATA;
codec_config->extradata_size -= 8;
AV_WB8(codec_config->extradata + 0, AV_RL8(codec_config->extradata + 8)); // version
AV_WB8(codec_config->extradata + 1, 2); // set channels to stereo
AV_WB16A(codec_config->extradata + 2, AV_RL16A(codec_config->extradata + 10)); // Byte swap pre-skip
AV_WB32A(codec_config->extradata + 4, AV_RL32A(codec_config->extradata + 12)); // Byte swap sample rate
AV_WB16A(codec_config->extradata + 8, 0); // set Output Gain to 0
AV_WB8(codec_config->extradata + 10, AV_RL8(codec_config->extradata + 18)); // Mapping family
break;
case AV_CODEC_ID_FLAC: {
uint8_t buf[13];
init_put_bits(&pb, buf, sizeof(buf));
ret = init_get_bits8(&gb, codec_config->extradata, codec_config->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, 1); // set channels to stereo
ret = put_bits_left(&pb);
put_bits(&pb, ret, get_bits(&gb, ret));
flush_put_bits(&pb);
memcpy(codec_config->extradata, buf, sizeof(buf));
break;
}
default:
break;
}
return 0;
}
static int populate_audio_roll_distance(IAMFCodecConfig *codec_config)
{
switch (codec_config->codec_id) {
case AV_CODEC_ID_OPUS:
if (!codec_config->nb_samples)
return AVERROR(EINVAL);
// ceil(3840 / nb_samples)
codec_config->audio_roll_distance = -(1 + ((3840 - 1) / codec_config->nb_samples));
break;
case AV_CODEC_ID_AAC:
codec_config->audio_roll_distance = -1;
break;
case AV_CODEC_ID_FLAC:
case AV_CODEC_ID_PCM_S16BE:
case AV_CODEC_ID_PCM_S24BE:
case AV_CODEC_ID_PCM_S32BE:
case AV_CODEC_ID_PCM_S16LE:
case AV_CODEC_ID_PCM_S24LE:
case AV_CODEC_ID_PCM_S32LE:
codec_config->audio_roll_distance = 0;
break;
default:
return AVERROR(EINVAL);
}
return 0;
}
static int fill_codec_config(IAMFContext *iamf, const AVStreamGroup *stg,
IAMFCodecConfig *codec_config)
{
const AVStream *st = stg->streams[0];
IAMFCodecConfig **tmp;
int j, ret = 0;
codec_config->codec_id = st->codecpar->codec_id;
codec_config->sample_rate = st->codecpar->sample_rate;
codec_config->codec_tag = st->codecpar->codec_tag;
codec_config->nb_samples = st->codecpar->frame_size;
populate_audio_roll_distance(codec_config);
if (st->codecpar->extradata_size) {
codec_config->extradata = av_memdup(st->codecpar->extradata, st->codecpar->extradata_size);
if (!codec_config->extradata)
return AVERROR(ENOMEM);
codec_config->extradata_size = st->codecpar->extradata_size;
ret = update_extradata(codec_config);
if (ret < 0)
goto fail;
}
for (j = 0; j < iamf->nb_codec_configs; j++) {
if (!memcmp(iamf->codec_configs[j], codec_config, offsetof(IAMFCodecConfig, extradata)) &&
(!codec_config->extradata_size || !memcmp(iamf->codec_configs[j]->extradata,
codec_config->extradata, codec_config->extradata_size)))
break;
}
if (j < iamf->nb_codec_configs) {
av_free(iamf->codec_configs[j]->extradata);
av_free(iamf->codec_configs[j]);
iamf->codec_configs[j] = codec_config;
return j;
}
tmp = av_realloc_array(iamf->codec_configs, iamf->nb_codec_configs + 1, sizeof(*iamf->codec_configs));
if (!tmp) {
ret = AVERROR(ENOMEM);
goto fail;
}
iamf->codec_configs = tmp;
iamf->codec_configs[iamf->nb_codec_configs] = codec_config;
codec_config->codec_config_id = iamf->nb_codec_configs;
return iamf->nb_codec_configs++;
fail:
av_freep(&codec_config->extradata);
return ret;
}
static int add_param_definition(IAMFContext *iamf, AVIAMFParamDefinition *param,
const IAMFAudioElement *audio_element, void *log_ctx)
{
IAMFParamDefinition **tmp, *param_definition;
IAMFCodecConfig *codec_config = NULL;
tmp = av_realloc_array(iamf->param_definitions, iamf->nb_param_definitions + 1,
sizeof(*iamf->param_definitions));
if (!tmp)
return AVERROR(ENOMEM);
iamf->param_definitions = tmp;
if (audio_element)
codec_config = iamf->codec_configs[audio_element->codec_config_id];
if (!param->parameter_rate) {
if (!codec_config) {
av_log(log_ctx, AV_LOG_ERROR, "parameter_rate needed but not set for parameter_id %u\n",
param->parameter_id);
return AVERROR(EINVAL);
}
param->parameter_rate = codec_config->sample_rate;
}
if (codec_config) {
if (!param->duration)
param->duration = codec_config->nb_samples;
if (!param->constant_subblock_duration)
param->constant_subblock_duration = codec_config->nb_samples;
}
param_definition = av_mallocz(sizeof(*param_definition));
if (!param_definition)
return AVERROR(ENOMEM);
param_definition->mode = !!param->duration;
param_definition->param = param;
param_definition->audio_element = audio_element;
iamf->param_definitions[iamf->nb_param_definitions++] = param_definition;
return 0;
}
int ff_iamf_add_audio_element(IAMFContext *iamf, const AVStreamGroup *stg, void *log_ctx)
{
const AVIAMFAudioElement *iamf_audio_element;
IAMFAudioElement **tmp, *audio_element;
IAMFCodecConfig *codec_config;
int ret;
if (stg->type != AV_STREAM_GROUP_PARAMS_IAMF_AUDIO_ELEMENT)
return AVERROR(EINVAL);
iamf_audio_element = stg->params.iamf_audio_element;
if (iamf_audio_element->audio_element_type == AV_IAMF_AUDIO_ELEMENT_TYPE_SCENE) {
const AVIAMFLayer *layer = iamf_audio_element->layers[0];
if (iamf_audio_element->nb_layers != 1) {
av_log(log_ctx, AV_LOG_ERROR, "Invalid amount of layers for SCENE_BASED audio element. Must be 1\n");
return AVERROR(EINVAL);
}
if (layer->ch_layout.order != AV_CHANNEL_ORDER_CUSTOM &&
layer->ch_layout.order != AV_CHANNEL_ORDER_AMBISONIC) {
av_log(log_ctx, AV_LOG_ERROR, "Invalid channel layout for SCENE_BASED audio element\n");
return AVERROR(EINVAL);
}
if (layer->ambisonics_mode >= AV_IAMF_AMBISONICS_MODE_PROJECTION) {
av_log(log_ctx, AV_LOG_ERROR, "Unsuported ambisonics mode %d\n", layer->ambisonics_mode);
return AVERROR_PATCHWELCOME;
}
for (int i = 0; i < stg->nb_streams; i++) {
if (stg->streams[i]->codecpar->ch_layout.nb_channels > 1) {
av_log(log_ctx, AV_LOG_ERROR, "Invalid amount of channels in a stream for MONO mode ambisonics\n");
return AVERROR(EINVAL);
}
}
} else
for (int j, i = 0; i < iamf_audio_element->nb_layers; i++) {
const AVIAMFLayer *layer = iamf_audio_element->layers[i];
for (j = 0; j < FF_ARRAY_ELEMS(ff_iamf_scalable_ch_layouts); j++)
if (!av_channel_layout_compare(&layer->ch_layout, &ff_iamf_scalable_ch_layouts[j]))
break;
if (j >= FF_ARRAY_ELEMS(ff_iamf_scalable_ch_layouts)) {
av_log(log_ctx, AV_LOG_ERROR, "Unsupported channel layout in stream group #%d\n", i);
return AVERROR(EINVAL);
}
}
for (int i = 0; i < iamf->nb_audio_elements; i++) {
if (stg->id == iamf->audio_elements[i]->audio_element_id) {
av_log(log_ctx, AV_LOG_ERROR, "Duplicated Audio Element id %"PRId64"\n", stg->id);
return AVERROR(EINVAL);
}
}
codec_config = av_mallocz(sizeof(*codec_config));
if (!codec_config)
return AVERROR(ENOMEM);
ret = fill_codec_config(iamf, stg, codec_config);
if (ret < 0) {
av_free(codec_config);
return ret;
}
audio_element = av_mallocz(sizeof(*audio_element));
if (!audio_element)
return AVERROR(ENOMEM);
audio_element->celement = stg->params.iamf_audio_element;
audio_element->audio_element_id = stg->id;
audio_element->codec_config_id = ret;
audio_element->substreams = av_calloc(stg->nb_streams, sizeof(*audio_element->substreams));
if (!audio_element->substreams) {
ret = AVERROR(ENOMEM);
goto fail;
}
audio_element->nb_substreams = stg->nb_streams;
audio_element->layers = av_calloc(iamf_audio_element->nb_layers, sizeof(*audio_element->layers));
if (!audio_element->layers) {
ret = AVERROR(ENOMEM);
goto fail;
}
for (int i = 0, j = 0; i < iamf_audio_element->nb_layers; i++) {
int nb_channels = iamf_audio_element->layers[i]->ch_layout.nb_channels;
IAMFLayer *layer = &audio_element->layers[i];
if (i)
nb_channels -= iamf_audio_element->layers[i - 1]->ch_layout.nb_channels;
for (; nb_channels > 0 && j < stg->nb_streams; j++) {
const AVStream *st = stg->streams[j];
IAMFSubStream *substream = &audio_element->substreams[j];
substream->audio_substream_id = st->id;
layer->substream_count++;
layer->coupled_substream_count += st->codecpar->ch_layout.nb_channels == 2;
nb_channels -= st->codecpar->ch_layout.nb_channels;
}
if (nb_channels) {
av_log(log_ctx, AV_LOG_ERROR, "Invalid channel count across substreams in layer %u from stream group %u\n",
i, stg->index);
ret = AVERROR(EINVAL);
goto fail;
}
}
for (int i = 0; i < audio_element->nb_substreams; i++) {
for (int j = i + 1; j < audio_element->nb_substreams; j++)
if (audio_element->substreams[i].audio_substream_id ==
audio_element->substreams[j].audio_substream_id) {
av_log(log_ctx, AV_LOG_ERROR, "Duplicate id %u in streams %u and %u from stream group %u\n",
audio_element->substreams[i].audio_substream_id, i, j, stg->index);
ret = AVERROR(EINVAL);
goto fail;
}
}
if (iamf_audio_element->demixing_info) {
AVIAMFParamDefinition *param = iamf_audio_element->demixing_info;
const IAMFParamDefinition *param_definition = ff_iamf_get_param_definition(iamf, param->parameter_id);
if (param->nb_subblocks != 1) {
av_log(log_ctx, AV_LOG_ERROR, "nb_subblocks in demixing_info for stream group %u is not 1\n", stg->index);
ret = AVERROR(EINVAL);
goto fail;
}
if (!param_definition) {
ret = add_param_definition(iamf, param, audio_element, log_ctx);
if (ret < 0)
goto fail;
}
}
if (iamf_audio_element->recon_gain_info) {
AVIAMFParamDefinition *param = iamf_audio_element->recon_gain_info;
const IAMFParamDefinition *param_definition = ff_iamf_get_param_definition(iamf, param->parameter_id);
if (param->nb_subblocks != 1) {
av_log(log_ctx, AV_LOG_ERROR, "nb_subblocks in recon_gain_info for stream group %u is not 1\n", stg->index);
ret = AVERROR(EINVAL);
goto fail;
}
if (!param_definition) {
ret = add_param_definition(iamf, param, audio_element, log_ctx);
if (ret < 0)
goto fail;
}
}
tmp = av_realloc_array(iamf->audio_elements, iamf->nb_audio_elements + 1, sizeof(*iamf->audio_elements));
if (!tmp) {
ret = AVERROR(ENOMEM);
goto fail;
}
iamf->audio_elements = tmp;
iamf->audio_elements[iamf->nb_audio_elements++] = audio_element;
return 0;
fail:
ff_iamf_free_audio_element(&audio_element);
return ret;
}
int ff_iamf_add_mix_presentation(IAMFContext *iamf, const AVStreamGroup *stg, void *log_ctx)
{
IAMFMixPresentation **tmp, *mix_presentation;
int ret;
if (stg->type != AV_STREAM_GROUP_PARAMS_IAMF_MIX_PRESENTATION)
return AVERROR(EINVAL);
for (int i = 0; i < iamf->nb_mix_presentations; i++) {
if (stg->id == iamf->mix_presentations[i]->mix_presentation_id) {
av_log(log_ctx, AV_LOG_ERROR, "Duplicate Mix Presentation id %"PRId64"\n", stg->id);
return AVERROR(EINVAL);
}
}
mix_presentation = av_mallocz(sizeof(*mix_presentation));
if (!mix_presentation)
return AVERROR(ENOMEM);
mix_presentation->cmix = stg->params.iamf_mix_presentation;
mix_presentation->mix_presentation_id = stg->id;
for (int i = 0; i < mix_presentation->cmix->nb_submixes; i++) {
const AVIAMFSubmix *submix = mix_presentation->cmix->submixes[i];
AVIAMFParamDefinition *param = submix->output_mix_config;
IAMFParamDefinition *param_definition;
if (!param) {
av_log(log_ctx, AV_LOG_ERROR, "output_mix_config is not present in submix %u from "
"Mix Presentation ID %"PRId64"\n", i, stg->id);
ret = AVERROR(EINVAL);
goto fail;
}
param_definition = ff_iamf_get_param_definition(iamf, param->parameter_id);
if (!param_definition) {
ret = add_param_definition(iamf, param, NULL, log_ctx);
if (ret < 0)
goto fail;
}
for (int j = 0; j < submix->nb_elements; j++) {
const AVIAMFSubmixElement *element = submix->elements[j];
param = element->element_mix_config;
if (!param) {
av_log(log_ctx, AV_LOG_ERROR, "element_mix_config is not present for element %u in submix %u from "
"Mix Presentation ID %"PRId64"\n", j, i, stg->id);
ret = AVERROR(EINVAL);
goto fail;
}
param_definition = ff_iamf_get_param_definition(iamf, param->parameter_id);
if (!param_definition) {
ret = add_param_definition(iamf, param, NULL, log_ctx);
if (ret < 0)
goto fail;
}
}
}
tmp = av_realloc_array(iamf->mix_presentations, iamf->nb_mix_presentations + 1, sizeof(*iamf->mix_presentations));
if (!tmp) {
ret = AVERROR(ENOMEM);
goto fail;
}
iamf->mix_presentations = tmp;
iamf->mix_presentations[iamf->nb_mix_presentations++] = mix_presentation;
return 0;
fail:
ff_iamf_free_mix_presentation(&mix_presentation);
return ret;
}
static int iamf_write_codec_config(const IAMFContext *iamf,
const IAMFCodecConfig *codec_config,
AVIOContext *pb)
{
uint8_t header[MAX_IAMF_OBU_HEADER_SIZE];
AVIOContext *dyn_bc;
uint8_t *dyn_buf = NULL;
PutBitContext pbc;
int dyn_size;
int ret = avio_open_dyn_buf(&dyn_bc);
if (ret < 0)
return ret;
ffio_write_leb(dyn_bc, codec_config->codec_config_id);
avio_wl32(dyn_bc, codec_config->codec_tag);
ffio_write_leb(dyn_bc, codec_config->nb_samples);
avio_wb16(dyn_bc, codec_config->audio_roll_distance);
switch(codec_config->codec_id) {
case AV_CODEC_ID_OPUS:
avio_write(dyn_bc, codec_config->extradata, codec_config->extradata_size);
break;
case AV_CODEC_ID_AAC:
return AVERROR_PATCHWELCOME;
case AV_CODEC_ID_FLAC:
avio_w8(dyn_bc, 0x80);
avio_wb24(dyn_bc, codec_config->extradata_size);
avio_write(dyn_bc, codec_config->extradata, codec_config->extradata_size);
break;
case AV_CODEC_ID_PCM_S16LE:
avio_w8(dyn_bc, 1);
avio_w8(dyn_bc, 16);
avio_wb32(dyn_bc, codec_config->sample_rate);
break;
case AV_CODEC_ID_PCM_S24LE:
avio_w8(dyn_bc, 1);
avio_w8(dyn_bc, 24);
avio_wb32(dyn_bc, codec_config->sample_rate);
break;
case AV_CODEC_ID_PCM_S32LE:
avio_w8(dyn_bc, 1);
avio_w8(dyn_bc, 32);
avio_wb32(dyn_bc, codec_config->sample_rate);
break;
case AV_CODEC_ID_PCM_S16BE:
avio_w8(dyn_bc, 0);
avio_w8(dyn_bc, 16);
avio_wb32(dyn_bc, codec_config->sample_rate);
break;
case AV_CODEC_ID_PCM_S24BE:
avio_w8(dyn_bc, 0);
avio_w8(dyn_bc, 24);
avio_wb32(dyn_bc, codec_config->sample_rate);
break;
case AV_CODEC_ID_PCM_S32BE:
avio_w8(dyn_bc, 0);
avio_w8(dyn_bc, 32);
avio_wb32(dyn_bc, codec_config->sample_rate);
break;
default:
break;
}
init_put_bits(&pbc, header, sizeof(header));
put_bits(&pbc, 5, IAMF_OBU_IA_CODEC_CONFIG);
put_bits(&pbc, 3, 0);
flush_put_bits(&pbc);
dyn_size = avio_get_dyn_buf(dyn_bc, &dyn_buf);
avio_write(pb, header, put_bytes_count(&pbc, 1));
ffio_write_leb(pb, dyn_size);
avio_write(pb, dyn_buf, dyn_size);
ffio_free_dyn_buf(&dyn_bc);
return 0;
}
static inline int rescale_rational(AVRational q, int b)
{
return av_clip_int16(av_rescale(q.num, b, q.den));
}
static int scalable_channel_layout_config(const IAMFAudioElement *audio_element,
AVIOContext *dyn_bc)
{
const AVIAMFAudioElement *element = audio_element->celement;
uint8_t header[MAX_IAMF_OBU_HEADER_SIZE];
PutBitContext pb;
init_put_bits(&pb, header, sizeof(header));
put_bits(&pb, 3, element->nb_layers);
put_bits(&pb, 5, 0);
flush_put_bits(&pb);
avio_write(dyn_bc, header, put_bytes_count(&pb, 1));
for (int i = 0; i < element->nb_layers; i++) {
const AVIAMFLayer *layer = element->layers[i];
int layout;
for (layout = 0; layout < FF_ARRAY_ELEMS(ff_iamf_scalable_ch_layouts); layout++) {
if (!av_channel_layout_compare(&layer->ch_layout, &ff_iamf_scalable_ch_layouts[layout]))
break;
}
init_put_bits(&pb, header, sizeof(header));
put_bits(&pb, 4, layout);
put_bits(&pb, 1, !!layer->output_gain_flags);
put_bits(&pb, 1, !!(layer->flags & AV_IAMF_LAYER_FLAG_RECON_GAIN));
put_bits(&pb, 2, 0); // reserved
put_bits(&pb, 8, audio_element->layers[i].substream_count);
put_bits(&pb, 8, audio_element->layers[i].coupled_substream_count);
if (layer->output_gain_flags) {
put_bits(&pb, 6, layer->output_gain_flags);
put_bits(&pb, 2, 0);
put_bits(&pb, 16, rescale_rational(layer->output_gain, 1 << 8));
}
flush_put_bits(&pb);
avio_write(dyn_bc, header, put_bytes_count(&pb, 1));
}
return 0;
}
static int ambisonics_config(const IAMFAudioElement *audio_element,
AVIOContext *dyn_bc)
{
const AVIAMFAudioElement *element = audio_element->celement;
const AVIAMFLayer *layer = element->layers[0];
ffio_write_leb(dyn_bc, 0); // ambisonics_mode
ffio_write_leb(dyn_bc, layer->ch_layout.nb_channels); // output_channel_count
ffio_write_leb(dyn_bc, audio_element->nb_substreams); // substream_count
if (layer->ch_layout.order == AV_CHANNEL_ORDER_AMBISONIC)
for (int i = 0; i < layer->ch_layout.nb_channels; i++)
avio_w8(dyn_bc, i);
else
for (int i = 0; i < layer->ch_layout.nb_channels; i++)
avio_w8(dyn_bc, layer->ch_layout.u.map[i].id);
return 0;
}
static int param_definition(const IAMFContext *iamf,
const IAMFParamDefinition *param_def,
AVIOContext *dyn_bc, void *log_ctx)
{
const AVIAMFParamDefinition *param = param_def->param;
ffio_write_leb(dyn_bc, param->parameter_id);
ffio_write_leb(dyn_bc, param->parameter_rate);
avio_w8(dyn_bc, param->duration ? 0 : 1 << 7);
if (param->duration) {
ffio_write_leb(dyn_bc, param->duration);
ffio_write_leb(dyn_bc, param->constant_subblock_duration);
if (param->constant_subblock_duration == 0) {
ffio_write_leb(dyn_bc, param->nb_subblocks);
for (int i = 0; i < param->nb_subblocks; i++) {
const void *subblock = av_iamf_param_definition_get_subblock(param, i);
switch (param->type) {
case AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN: {
const AVIAMFMixGain *mix = subblock;
ffio_write_leb(dyn_bc, mix->subblock_duration);
break;
}
case AV_IAMF_PARAMETER_DEFINITION_DEMIXING: {
const AVIAMFDemixingInfo *demix = subblock;
ffio_write_leb(dyn_bc, demix->subblock_duration);
break;
}
case AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN: {
const AVIAMFReconGain *recon = subblock;
ffio_write_leb(dyn_bc, recon->subblock_duration);
break;
}
}
}
}
}
return 0;
}
static int iamf_write_audio_element(const IAMFContext *iamf,
const IAMFAudioElement *audio_element,
AVIOContext *pb, void *log_ctx)
{
const AVIAMFAudioElement *element = audio_element->celement;
const IAMFCodecConfig *codec_config = iamf->codec_configs[audio_element->codec_config_id];
uint8_t header[MAX_IAMF_OBU_HEADER_SIZE];
AVIOContext *dyn_bc;
uint8_t *dyn_buf = NULL;
PutBitContext pbc;
int param_definition_types = AV_IAMF_PARAMETER_DEFINITION_DEMIXING, dyn_size;
int ret = avio_open_dyn_buf(&dyn_bc);
if (ret < 0)
return ret;
ffio_write_leb(dyn_bc, audio_element->audio_element_id);
init_put_bits(&pbc, header, sizeof(header));
put_bits(&pbc, 3, element->audio_element_type);
put_bits(&pbc, 5, 0);
flush_put_bits(&pbc);
avio_write(dyn_bc, header, put_bytes_count(&pbc, 1));
ffio_write_leb(dyn_bc, audio_element->codec_config_id);
ffio_write_leb(dyn_bc, audio_element->nb_substreams);
for (int i = 0; i < audio_element->nb_substreams; i++)
ffio_write_leb(dyn_bc, audio_element->substreams[i].audio_substream_id);
if (element->nb_layers == 1)
param_definition_types &= ~AV_IAMF_PARAMETER_DEFINITION_DEMIXING;
if (element->nb_layers > 1)
param_definition_types |= AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN;
if (codec_config->codec_tag == MKTAG('f','L','a','C') ||
codec_config->codec_tag == MKTAG('i','p','c','m'))
param_definition_types &= ~AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN;
ffio_write_leb(dyn_bc, av_popcount(param_definition_types)); // num_parameters
if (param_definition_types & 1) {
const AVIAMFParamDefinition *param = element->demixing_info;
const IAMFParamDefinition *param_def;
const AVIAMFDemixingInfo *demix;
if (!param) {
av_log(log_ctx, AV_LOG_ERROR, "demixing_info needed but not set in Stream Group #%u\n",
audio_element->audio_element_id);
return AVERROR(EINVAL);
}
demix = av_iamf_param_definition_get_subblock(param, 0);
ffio_write_leb(dyn_bc, AV_IAMF_PARAMETER_DEFINITION_DEMIXING); // type
param_def = ff_iamf_get_param_definition(iamf, param->parameter_id);
ret = param_definition(iamf, param_def, dyn_bc, log_ctx);
if (ret < 0)
return ret;
avio_w8(dyn_bc, demix->dmixp_mode << 5); // dmixp_mode
avio_w8(dyn_bc, element->default_w << 4); // default_w
}
if (param_definition_types & 2) {
const AVIAMFParamDefinition *param = element->recon_gain_info;
const IAMFParamDefinition *param_def;
if (!param) {
av_log(log_ctx, AV_LOG_ERROR, "recon_gain_info needed but not set in Stream Group #%u\n",
audio_element->audio_element_id);
return AVERROR(EINVAL);
}
ffio_write_leb(dyn_bc, AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN); // type
param_def = ff_iamf_get_param_definition(iamf, param->parameter_id);
ret = param_definition(iamf, param_def, dyn_bc, log_ctx);
if (ret < 0)
return ret;
}
if (element->audio_element_type == AV_IAMF_AUDIO_ELEMENT_TYPE_CHANNEL) {
ret = scalable_channel_layout_config(audio_element, dyn_bc);
if (ret < 0)
return ret;
} else {
ret = ambisonics_config(audio_element, dyn_bc);
if (ret < 0)
return ret;
}
init_put_bits(&pbc, header, sizeof(header));
put_bits(&pbc, 5, IAMF_OBU_IA_AUDIO_ELEMENT);
put_bits(&pbc, 3, 0);
flush_put_bits(&pbc);
dyn_size = avio_get_dyn_buf(dyn_bc, &dyn_buf);
avio_write(pb, header, put_bytes_count(&pbc, 1));
ffio_write_leb(pb, dyn_size);
avio_write(pb, dyn_buf, dyn_size);
ffio_free_dyn_buf(&dyn_bc);
return 0;
}
static int iamf_write_mixing_presentation(const IAMFContext *iamf,
const IAMFMixPresentation *mix_presentation,
AVIOContext *pb, void *log_ctx)
{
uint8_t header[MAX_IAMF_OBU_HEADER_SIZE];
const AVIAMFMixPresentation *mix = mix_presentation->cmix;
const AVDictionaryEntry *tag = NULL;
PutBitContext pbc;
AVIOContext *dyn_bc;
uint8_t *dyn_buf = NULL;
int dyn_size;
int ret = avio_open_dyn_buf(&dyn_bc);
if (ret < 0)
return ret;
ffio_write_leb(dyn_bc, mix_presentation->mix_presentation_id); // mix_presentation_id
ffio_write_leb(dyn_bc, av_dict_count(mix->annotations)); // count_label
while ((tag = av_dict_iterate(mix->annotations, tag)))
avio_put_str(dyn_bc, tag->key);
while ((tag = av_dict_iterate(mix->annotations, tag)))
avio_put_str(dyn_bc, tag->value);
ffio_write_leb(dyn_bc, mix->nb_submixes);
for (int i = 0; i < mix->nb_submixes; i++) {
const AVIAMFSubmix *sub_mix = mix->submixes[i];
const IAMFParamDefinition *param_def;
ffio_write_leb(dyn_bc, sub_mix->nb_elements);
for (int j = 0; j < sub_mix->nb_elements; j++) {
const IAMFAudioElement *audio_element = NULL;
const AVIAMFSubmixElement *submix_element = sub_mix->elements[j];
for (int k = 0; k < iamf->nb_audio_elements; k++)
if (iamf->audio_elements[k]->audio_element_id == submix_element->audio_element_id) {
audio_element = iamf->audio_elements[k];
break;
}
av_assert0(audio_element);
ffio_write_leb(dyn_bc, submix_element->audio_element_id);
if (av_dict_count(submix_element->annotations) != av_dict_count(mix->annotations)) {
av_log(log_ctx, AV_LOG_ERROR, "Inconsistent amount of labels in submix %d from Mix Presentation id #%u\n",
j, audio_element->audio_element_id);
return AVERROR(EINVAL);
}
while ((tag = av_dict_iterate(submix_element->annotations, tag)))
avio_put_str(dyn_bc, tag->value);
init_put_bits(&pbc, header, sizeof(header));
put_bits(&pbc, 2, submix_element->headphones_rendering_mode);
put_bits(&pbc, 6, 0); // reserved
flush_put_bits(&pbc);
avio_write(dyn_bc, header, put_bytes_count(&pbc, 1));
ffio_write_leb(dyn_bc, 0); // rendering_config_extension_size
param_def = ff_iamf_get_param_definition(iamf, submix_element->element_mix_config->parameter_id);
ret = param_definition(iamf, param_def, dyn_bc, log_ctx);
if (ret < 0)
return ret;
avio_wb16(dyn_bc, rescale_rational(submix_element->default_mix_gain, 1 << 8));
}
param_def = ff_iamf_get_param_definition(iamf, sub_mix->output_mix_config->parameter_id);
ret = param_definition(iamf, param_def, dyn_bc, log_ctx);
if (ret < 0)
return ret;
avio_wb16(dyn_bc, rescale_rational(sub_mix->default_mix_gain, 1 << 8));
ffio_write_leb(dyn_bc, sub_mix->nb_layouts); // nb_layouts
for (int i = 0; i < sub_mix->nb_layouts; i++) {
const AVIAMFSubmixLayout *submix_layout = sub_mix->layouts[i];
int layout, info_type;
int dialogue = submix_layout->dialogue_anchored_loudness.num &&
submix_layout->dialogue_anchored_loudness.den;
int album = submix_layout->album_anchored_loudness.num &&
submix_layout->album_anchored_loudness.den;
if (submix_layout->layout_type == AV_IAMF_SUBMIX_LAYOUT_TYPE_LOUDSPEAKERS) {
for (layout = 0; layout < FF_ARRAY_ELEMS(ff_iamf_sound_system_map); layout++) {
if (!av_channel_layout_compare(&submix_layout->sound_system, &ff_iamf_sound_system_map[layout].layout))
break;
}
if (layout == FF_ARRAY_ELEMS(ff_iamf_sound_system_map)) {
av_log(log_ctx, AV_LOG_ERROR, "Invalid Sound System value in a submix\n");
return AVERROR(EINVAL);
}
} else if (submix_layout->layout_type != AV_IAMF_SUBMIX_LAYOUT_TYPE_BINAURAL) {
av_log(log_ctx, AV_LOG_ERROR, "Unsupported Layout Type value in a submix\n");
return AVERROR(EINVAL);
}
init_put_bits(&pbc, header, sizeof(header));
put_bits(&pbc, 2, submix_layout->layout_type); // layout_type
if (submix_layout->layout_type == AV_IAMF_SUBMIX_LAYOUT_TYPE_LOUDSPEAKERS) {
put_bits(&pbc, 4, ff_iamf_sound_system_map[layout].id); // sound_system
put_bits(&pbc, 2, 0); // reserved
} else
put_bits(&pbc, 6, 0); // reserved
flush_put_bits(&pbc);
avio_write(dyn_bc, header, put_bytes_count(&pbc, 1));
info_type = (submix_layout->true_peak.num && submix_layout->true_peak.den);
info_type |= (dialogue || album) << 1;
avio_w8(dyn_bc, info_type);
avio_wb16(dyn_bc, rescale_rational(submix_layout->integrated_loudness, 1 << 8));
avio_wb16(dyn_bc, rescale_rational(submix_layout->digital_peak, 1 << 8));
if (info_type & 1)
avio_wb16(dyn_bc, rescale_rational(submix_layout->true_peak, 1 << 8));
if (info_type & 2) {
avio_w8(dyn_bc, dialogue + album); // num_anchored_loudness
if (dialogue) {
avio_w8(dyn_bc, IAMF_ANCHOR_ELEMENT_DIALOGUE);
avio_wb16(dyn_bc, rescale_rational(submix_layout->dialogue_anchored_loudness, 1 << 8));
}
if (album) {
avio_w8(dyn_bc, IAMF_ANCHOR_ELEMENT_ALBUM);
avio_wb16(dyn_bc, rescale_rational(submix_layout->album_anchored_loudness, 1 << 8));
}
}
}
}
init_put_bits(&pbc, header, sizeof(header));
put_bits(&pbc, 5, IAMF_OBU_IA_MIX_PRESENTATION);
put_bits(&pbc, 3, 0);
flush_put_bits(&pbc);
dyn_size = avio_get_dyn_buf(dyn_bc, &dyn_buf);
avio_write(pb, header, put_bytes_count(&pbc, 1));
ffio_write_leb(pb, dyn_size);
avio_write(pb, dyn_buf, dyn_size);
ffio_free_dyn_buf(&dyn_bc);
return 0;
}
int ff_iamf_write_descriptors(const IAMFContext *iamf, AVIOContext *pb, void *log_ctx)
{
int ret;
// Sequence Header
avio_w8(pb, IAMF_OBU_IA_SEQUENCE_HEADER << 3);
ffio_write_leb(pb, 6);
avio_wb32(pb, MKBETAG('i','a','m','f'));
avio_w8(pb, iamf->nb_audio_elements > 1); // primary_profile
avio_w8(pb, iamf->nb_audio_elements > 1); // additional_profile
for (int i = 0; i < iamf->nb_codec_configs; i++) {
ret = iamf_write_codec_config(iamf, iamf->codec_configs[i], pb);
if (ret < 0)
return ret;
}
for (int i = 0; i < iamf->nb_audio_elements; i++) {
ret = iamf_write_audio_element(iamf, iamf->audio_elements[i], pb, log_ctx);
if (ret < 0)
return ret;
}
for (int i = 0; i < iamf->nb_mix_presentations; i++) {
ret = iamf_write_mixing_presentation(iamf, iamf->mix_presentations[i], pb, log_ctx);
if (ret < 0)
return ret;
}
return 0;
}
static int write_parameter_block(const IAMFContext *iamf, AVIOContext *pb,
const AVIAMFParamDefinition *param, void *log_ctx)
{
uint8_t header[MAX_IAMF_OBU_HEADER_SIZE];
const IAMFParamDefinition *param_definition = ff_iamf_get_param_definition(iamf, param->parameter_id);
PutBitContext pbc;
AVIOContext *dyn_bc;
uint8_t *dyn_buf = NULL;
int dyn_size, ret;
if (param->type > AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN) {
av_log(log_ctx, AV_LOG_DEBUG, "Ignoring side data with unknown type %u\n",
param->type);
return 0;
}
if (!param_definition) {
av_log(log_ctx, AV_LOG_ERROR, "Non-existent Parameter Definition with ID %u referenced by a packet\n",
param->parameter_id);
return AVERROR(EINVAL);
}
if (param->type != param_definition->param->type) {
av_log(log_ctx, AV_LOG_ERROR, "Inconsistent values for Parameter Definition "
"with ID %u in a packet\n",
param->parameter_id);
return AVERROR(EINVAL);
}
ret = avio_open_dyn_buf(&dyn_bc);
if (ret < 0)
return ret;
// Sequence Header
init_put_bits(&pbc, header, sizeof(header));
put_bits(&pbc, 5, IAMF_OBU_IA_PARAMETER_BLOCK);
put_bits(&pbc, 3, 0);
flush_put_bits(&pbc);
avio_write(pb, header, put_bytes_count(&pbc, 1));
ffio_write_leb(dyn_bc, param->parameter_id);
if (!param_definition->mode) {
ffio_write_leb(dyn_bc, param->duration);
ffio_write_leb(dyn_bc, param->constant_subblock_duration);
if (param->constant_subblock_duration == 0)
ffio_write_leb(dyn_bc, param->nb_subblocks);
}
for (int i = 0; i < param->nb_subblocks; i++) {
const void *subblock = av_iamf_param_definition_get_subblock(param, i);
switch (param->type) {
case AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN: {
const AVIAMFMixGain *mix = subblock;
if (!param_definition->mode && param->constant_subblock_duration == 0)
ffio_write_leb(dyn_bc, mix->subblock_duration);
ffio_write_leb(dyn_bc, mix->animation_type);
avio_wb16(dyn_bc, rescale_rational(mix->start_point_value, 1 << 8));
if (mix->animation_type >= AV_IAMF_ANIMATION_TYPE_LINEAR)
avio_wb16(dyn_bc, rescale_rational(mix->end_point_value, 1 << 8));
if (mix->animation_type == AV_IAMF_ANIMATION_TYPE_BEZIER) {
avio_wb16(dyn_bc, rescale_rational(mix->control_point_value, 1 << 8));
avio_w8(dyn_bc, av_clip_uint8(av_rescale(mix->control_point_relative_time.num, 1 << 8,
mix->control_point_relative_time.den)));
}
break;
}
case AV_IAMF_PARAMETER_DEFINITION_DEMIXING: {
const AVIAMFDemixingInfo *demix = subblock;
if (!param_definition->mode && param->constant_subblock_duration == 0)
ffio_write_leb(dyn_bc, demix->subblock_duration);
avio_w8(dyn_bc, demix->dmixp_mode << 5);
break;
}
case AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN: {
const AVIAMFReconGain *recon = subblock;
const AVIAMFAudioElement *audio_element = param_definition->audio_element->celement;
if (!param_definition->mode && param->constant_subblock_duration == 0)
ffio_write_leb(dyn_bc, recon->subblock_duration);
if (!audio_element) {
av_log(log_ctx, AV_LOG_ERROR, "Invalid Parameter Definition with ID %u referenced by a packet\n", param->parameter_id);
return AVERROR(EINVAL);
}
for (int j = 0; j < audio_element->nb_layers; j++) {
const AVIAMFLayer *layer = audio_element->layers[j];
if (layer->flags & AV_IAMF_LAYER_FLAG_RECON_GAIN) {
unsigned int recon_gain_flags = 0;
int k = 0;
for (; k < 7; k++)
recon_gain_flags |= (1 << k) * !!recon->recon_gain[j][k];
for (; k < 12; k++)
recon_gain_flags |= (2 << k) * !!recon->recon_gain[j][k];
if (recon_gain_flags >> 8)
recon_gain_flags |= (1 << k);
ffio_write_leb(dyn_bc, recon_gain_flags);
for (k = 0; k < 12; k++) {
if (recon->recon_gain[j][k])
avio_w8(dyn_bc, recon->recon_gain[j][k]);
}
}
}
break;
}
default:
av_assert0(0);
}
}
dyn_size = avio_get_dyn_buf(dyn_bc, &dyn_buf);
ffio_write_leb(pb, dyn_size);
avio_write(pb, dyn_buf, dyn_size);
ffio_free_dyn_buf(&dyn_bc);
return 0;
}
int ff_iamf_write_parameter_blocks(const IAMFContext *iamf, AVIOContext *pb,
const AVPacket *pkt, void *log_ctx)
{
AVIAMFParamDefinition *mix =
(AVIAMFParamDefinition *)av_packet_get_side_data(pkt,
AV_PKT_DATA_IAMF_MIX_GAIN_PARAM,
NULL);
AVIAMFParamDefinition *demix =
(AVIAMFParamDefinition *)av_packet_get_side_data(pkt,
AV_PKT_DATA_IAMF_DEMIXING_INFO_PARAM,
NULL);
AVIAMFParamDefinition *recon =
(AVIAMFParamDefinition *)av_packet_get_side_data(pkt,
AV_PKT_DATA_IAMF_RECON_GAIN_INFO_PARAM,
NULL);
if (mix) {
int ret = write_parameter_block(iamf, pb, mix, log_ctx);
if (ret < 0)
return ret;
}
if (demix) {
int ret = write_parameter_block(iamf, pb, demix, log_ctx);
if (ret < 0)
return ret;
}
if (recon) {
int ret = write_parameter_block(iamf, pb, recon, log_ctx);
if (ret < 0)
return ret;
}
return 0;
}
static IAMFAudioElement *get_audio_element(const IAMFContext *c,
unsigned int audio_substream_id)
{
for (int i = 0; i < c->nb_audio_elements; i++) {
IAMFAudioElement *audio_element = c->audio_elements[i];
for (int j = 0; j < audio_element->nb_substreams; j++) {
IAMFSubStream *substream = &audio_element->substreams[j];
if (substream->audio_substream_id == audio_substream_id)
return audio_element;
}
}
return NULL;
}
int ff_iamf_write_audio_frame(const IAMFContext *iamf, AVIOContext *pb,
unsigned audio_substream_id, const AVPacket *pkt)
{
uint8_t header[MAX_IAMF_OBU_HEADER_SIZE];
PutBitContext pbc;
AVIOContext *dyn_bc;
const uint8_t *side_data;
uint8_t *dyn_buf = NULL;
unsigned int skip_samples = 0, discard_padding = 0;
size_t side_data_size;
int dyn_size, type = audio_substream_id <= 17 ?
audio_substream_id + IAMF_OBU_IA_AUDIO_FRAME_ID0 : IAMF_OBU_IA_AUDIO_FRAME;
int ret;
if (!pkt->size) {
const IAMFAudioElement *audio_element;
IAMFCodecConfig *codec_config;
size_t new_extradata_size;
const uint8_t *new_extradata = av_packet_get_side_data(pkt,
AV_PKT_DATA_NEW_EXTRADATA,
&new_extradata_size);
if (!new_extradata)
return AVERROR_INVALIDDATA;
audio_element = get_audio_element(iamf, audio_substream_id);
if (!audio_element)
return AVERROR(EINVAL);
codec_config = ff_iamf_get_codec_config(iamf, audio_element->codec_config_id);
if (!codec_config)
return AVERROR(EINVAL);
av_free(codec_config->extradata);
codec_config->extradata = av_memdup(new_extradata, new_extradata_size);
if (!codec_config->extradata) {
codec_config->extradata_size = 0;
return AVERROR(ENOMEM);
}
codec_config->extradata_size = new_extradata_size;
return update_extradata(codec_config);
}
side_data = av_packet_get_side_data(pkt, AV_PKT_DATA_SKIP_SAMPLES,
&side_data_size);
if (side_data && side_data_size >= 10) {
skip_samples = AV_RL32(side_data);
discard_padding = AV_RL32(side_data + 4);
}
ret = avio_open_dyn_buf(&dyn_bc);
if (ret < 0)
return ret;
init_put_bits(&pbc, header, sizeof(header));
put_bits(&pbc, 5, type);
put_bits(&pbc, 1, 0); // obu_redundant_copy
put_bits(&pbc, 1, skip_samples || discard_padding);
put_bits(&pbc, 1, 0); // obu_extension_flag
flush_put_bits(&pbc);
avio_write(pb, header, put_bytes_count(&pbc, 1));
if (skip_samples || discard_padding) {
ffio_write_leb(dyn_bc, discard_padding);
ffio_write_leb(dyn_bc, skip_samples);
}
if (audio_substream_id > 17)
ffio_write_leb(dyn_bc, audio_substream_id);
dyn_size = avio_get_dyn_buf(dyn_bc, &dyn_buf);
ffio_write_leb(pb, dyn_size + pkt->size);
avio_write(pb, dyn_buf, dyn_size);
ffio_free_dyn_buf(&dyn_bc);
avio_write(pb, pkt->data, pkt->size);
return 0;
}
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