virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-21 10:55:51 +02:00
Code Issues Releases Activity
master
FFmpeg/libavformat/iamf_parse.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

1120 lines
37 KiB
C
Raw Permalink Blame History

/*
* Immersive Audio Model and Formats parsing
* 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/avassert.h"
#include "libavutil/iamf.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/log.h"
#include "libavutil/mem.h"
#include "libavcodec/get_bits.h"
#include "libavcodec/flac.h"
#include "libavcodec/leb.h"
#include "libavcodec/mpeg4audio.h"
#include "libavcodec/put_bits.h"
#include "avio_internal.h"
#include "iamf_parse.h"
#include "isom.h"
static int opus_decoder_config(IAMFCodecConfig *codec_config,
AVIOContext *pb, int len)
{
int ret, left = len - avio_tell(pb);
if (left < 11 || codec_config->audio_roll_distance >= 0)
return AVERROR_INVALIDDATA;
codec_config->extradata = av_malloc(left + 8);
if (!codec_config->extradata)
return AVERROR(ENOMEM);
AV_WB32A(codec_config->extradata, MKBETAG('O','p','u','s'));
AV_WB32A(codec_config->extradata + 4, MKBETAG('H','e','a','d'));
ret = ffio_read_size(pb, codec_config->extradata + 8, left);
if (ret < 0)
return ret;
codec_config->extradata_size = left + 8;
codec_config->sample_rate = 48000;
return 0;
}
static int aac_decoder_config(IAMFCodecConfig *codec_config,
AVIOContext *pb, int len, void *logctx)
{
MPEG4AudioConfig cfg = { 0 };
int object_type_id, codec_id, stream_type;
int ret, tag, left;
if (codec_config->audio_roll_distance >= 0)
return AVERROR_INVALIDDATA;
ff_mp4_read_descr(logctx, pb, &tag);
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;
left = ff_mp4_read_descr(logctx, pb, &tag);
if (tag != MP4DecSpecificDescrTag ||
!left || left > (len - avio_tell(pb)))
return AVERROR_INVALIDDATA;
// We pad extradata here because avpriv_mpeg4audio_get_config2() needs it.
codec_config->extradata = av_malloc((size_t)left + AV_INPUT_BUFFER_PADDING_SIZE);
if (!codec_config->extradata)
return AVERROR(ENOMEM);
ret = ffio_read_size(pb, codec_config->extradata, left);
if (ret < 0)
return ret;
codec_config->extradata_size = left;
memset(codec_config->extradata + codec_config->extradata_size, 0,
AV_INPUT_BUFFER_PADDING_SIZE);
ret = avpriv_mpeg4audio_get_config2(&cfg, codec_config->extradata,
codec_config->extradata_size, 1, logctx);
if (ret < 0)
return ret;
codec_config->sample_rate = cfg.sample_rate;
return 0;
}
static int flac_decoder_config(IAMFCodecConfig *codec_config,
AVIOContext *pb, int len)
{
int ret, left;
if (codec_config->audio_roll_distance)
return AVERROR_INVALIDDATA;
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);
ret = ffio_read_size(pb, codec_config->extradata, left);
if (ret < 0)
return ret;
codec_config->extradata_size = left;
codec_config->sample_rate = AV_RB24(codec_config->extradata + 10) >> 4;
return 0;
}
static int ipcm_decoder_config(IAMFCodecConfig *codec_config,
AVIOContext *pb, int len)
{
static const enum AVCodecID 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 > 2U || codec_config->audio_roll_distance)
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(void *s, IAMFContext *c, AVIOContext *pb, int len)
{
IAMFCodecConfig **tmp, *codec_config = NULL;
FFIOContext b;
AVIOContext *pbc;
uint8_t *buf;
enum AVCodecID avcodec_id;
unsigned codec_config_id, nb_samples, codec_id;
int16_t audio_roll_distance;
int ret;
buf = av_malloc(len);
if (!buf)
return AVERROR(ENOMEM);
ret = ffio_read_size(pb, buf, len);
if (ret < 0)
goto fail;
ffio_init_context(&b, buf, len, 0, NULL, NULL, NULL, NULL);
pbc = &b.pub;
codec_config_id = ffio_read_leb(pbc);
codec_id = avio_rb32(pbc);
nb_samples = ffio_read_leb(pbc);
audio_roll_distance = avio_rb16(pbc);
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;
}
tmp = av_realloc_array(c->codec_configs, c->nb_codec_configs + 1, sizeof(*c->codec_configs));
if (!tmp) {
ret = AVERROR(ENOMEM);
goto fail;
}
c->codec_configs = tmp;
codec_config = av_mallocz(sizeof(*codec_config));
if (!codec_config) {
ret = AVERROR(ENOMEM);
goto fail;
}
codec_config->codec_config_id = codec_config_id;
codec_config->codec_id = avcodec_id;
codec_config->nb_samples = nb_samples;
codec_config->audio_roll_distance = audio_roll_distance;
switch(codec_id) {
case MKBETAG('O','p','u','s'):
ret = opus_decoder_config(codec_config, pbc, len);
break;
case MKBETAG('m','p','4','a'):
ret = aac_decoder_config(codec_config, pbc, len, s);
break;
case MKBETAG('f','L','a','C'):
ret = flac_decoder_config(codec_config, pbc, len);
break;
case MKBETAG('i','p','c','m'):
ret = ipcm_decoder_config(codec_config, pbc, len);
break;
default:
break;
}
if (ret < 0)
goto fail;
if ((codec_config->nb_samples > INT_MAX) || codec_config->nb_samples <= 0 ||
(-codec_config->audio_roll_distance > INT_MAX / codec_config->nb_samples)) {
ret = AVERROR_INVALIDDATA;
goto fail;
}
c->codec_configs[c->nb_codec_configs++] = codec_config;
len -= avio_tell(pbc);
if (len)
av_log(s, AV_LOG_WARNING, "Underread in codec_config_obu. %d bytes left at the end\n", len);
ret = 0;
fail:
av_free(buf);
if (ret < 0) {
if (codec_config)
av_free(codec_config->extradata);
av_free(codec_config);
}
return ret;
}
static int update_extradata(AVCodecParameters *codecpar)
{
GetBitContext gb;
PutBitContext pb;
int ret;
switch(codecpar->codec_id) {
case AV_CODEC_ID_OPUS:
AV_WB8(codecpar->extradata + 9, codecpar->ch_layout.nb_channels);
AV_WL16A(codecpar->extradata + 10, AV_RB16A(codecpar->extradata + 10)); // Byte swap pre-skip
AV_WL32A(codecpar->extradata + 12, AV_RB32A(codecpar->extradata + 12)); // Byte swap sample rate
AV_WL16A(codecpar->extradata + 16, AV_RB16A(codecpar->extradata + 16)); // Byte swap Output Gain
break;
case AV_CODEC_ID_AAC: {
uint8_t buf[5];
init_put_bits(&pb, buf, sizeof(buf));
ret = init_get_bits8(&gb, codecpar->extradata, 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, codecpar->ch_layout.nb_channels); // set channel config
ret = put_bits_left(&pb);
put_bits(&pb, ret, get_bits_long(&gb, ret));
flush_put_bits(&pb);
memcpy(codecpar->extradata, buf, put_bytes_output(&pb));
break;
}
case AV_CODEC_ID_FLAC: {
uint8_t buf[13];
init_put_bits(&pb, buf, sizeof(buf));
ret = init_get_bits8(&gb, codecpar->extradata, 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, codecpar->ch_layout.nb_channels - 1);
ret = put_bits_left(&pb);
put_bits(&pb, ret, get_bits(&gb, ret));
flush_put_bits(&pb);
memcpy(codecpar->extradata, buf, sizeof(buf));
break;
}
}
return 0;
}
static int scalable_channel_layout_config(void *s, AVIOContext *pb,
IAMFAudioElement *audio_element,
const IAMFCodecConfig *codec_config)
{
int nb_layers, k = 0;
nb_layers = avio_r8(pb) >> 5; // get_bits(&gb, 3);
// skip_bits(&gb, 5); //reserved
if (nb_layers > 6 || nb_layers == 0)
return AVERROR_INVALIDDATA;
audio_element->layers = av_calloc(nb_layers, sizeof(*audio_element->layers));
if (!audio_element->layers)
return AVERROR(ENOMEM);
audio_element->nb_layers = nb_layers;
for (int i = 0; i < nb_layers; i++) {
AVIAMFLayer *layer;
int loudspeaker_layout, output_gain_is_present_flag;
int substream_count, coupled_substream_count;
int ret, byte = avio_r8(pb);
layer = av_iamf_audio_element_add_layer(audio_element->element);
if (!layer)
return AVERROR(ENOMEM);
loudspeaker_layout = byte >> 4; // get_bits(&gb, 4);
output_gain_is_present_flag = (byte >> 3) & 1; //get_bits1(&gb);
if ((byte >> 2) & 1)
layer->flags |= AV_IAMF_LAYER_FLAG_RECON_GAIN;
substream_count = avio_r8(pb);
coupled_substream_count = avio_r8(pb);
if (substream_count + k > audio_element->nb_substreams)
return AVERROR_INVALIDDATA;
audio_element->layers[i].substream_count = substream_count;
audio_element->layers[i].coupled_substream_count = coupled_substream_count;
if (output_gain_is_present_flag) {
layer->output_gain_flags = avio_r8(pb) >> 2; // get_bits(&gb, 6);
layer->output_gain = av_make_q(sign_extend(avio_rb16(pb), 16), 1 << 8);
}
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 = substream_count +
coupled_substream_count };
for (int j = 0; j < substream_count; j++) {
IAMFSubStream *substream = &audio_element->substreams[k++];
substream->codecpar->ch_layout = coupled_substream_count-- > 0 ? (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO :
(AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
ret = update_extradata(substream->codecpar);
if (ret < 0)
return ret;
}
}
return 0;
}
static int ambisonics_config(void *s, AVIOContext *pb,
IAMFAudioElement *audio_element,
const IAMFCodecConfig *codec_config)
{
AVIAMFLayer *layer;
unsigned ambisonics_mode;
int output_channel_count, substream_count, order;
int ret;
ambisonics_mode = ffio_read_leb(pb);
if (ambisonics_mode > 1)
return 0;
output_channel_count = avio_r8(pb); // C
substream_count = avio_r8(pb); // N
if (audio_element->nb_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;
audio_element->layers = av_mallocz(sizeof(*audio_element->layers));
if (!audio_element->layers)
return AVERROR(ENOMEM);
audio_element->nb_layers = 1;
audio_element->layers->substream_count = substream_count;
layer = av_iamf_audio_element_add_layer(audio_element->element);
if (!layer)
return AVERROR(ENOMEM);
layer->ambisonics_mode = ambisonics_mode;
if (ambisonics_mode == 0) {
for (int i = 0; i < substream_count; i++) {
IAMFSubStream *substream = &audio_element->substreams[i];
substream->codecpar->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
ret = update_extradata(substream->codecpar);
if (ret < 0)
return ret;
}
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;
audio_element->layers->coupled_substream_count = coupled_substream_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] = av_make_q(sign_extend(avio_rb16(pb), 16), 1 << 8);
for (int i = 0; i < substream_count; i++) {
IAMFSubStream *substream = &audio_element->substreams[i];
substream->codecpar->ch_layout = coupled_substream_count-- > 0 ? (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO :
(AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
ret = update_extradata(substream->codecpar);
if (ret < 0)
return ret;
}
}
return 0;
}
static int param_parse(void *s, IAMFContext *c, AVIOContext *pb,
unsigned int type,
const IAMFAudioElement *audio_element,
AVIAMFParamDefinition **out_param_definition)
{
IAMFParamDefinition *param_definition = NULL;
AVIAMFParamDefinition *param;
unsigned int parameter_id, parameter_rate, mode;
unsigned int duration = 0, constant_subblock_duration = 0, nb_subblocks = 0;
size_t param_size;
parameter_id = ffio_read_leb(pb);
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;
}
parameter_rate = ffio_read_leb(pb);
mode = avio_r8(pb) >> 7;
if (mode == 0) {
duration = ffio_read_leb(pb);
if (!duration)
return AVERROR_INVALIDDATA;
constant_subblock_duration = ffio_read_leb(pb);
if (constant_subblock_duration == 0)
nb_subblocks = ffio_read_leb(pb);
else
nb_subblocks = duration / constant_subblock_duration;
}
param = av_iamf_param_definition_alloc(type, nb_subblocks, &param_size);
if (!param)
return AVERROR(ENOMEM);
for (int i = 0; i < nb_subblocks; i++) {
void *subblock = av_iamf_param_definition_get_subblock(param, i);
unsigned int subblock_duration = constant_subblock_duration;
if (constant_subblock_duration == 0)
subblock_duration = ffio_read_leb(pb);
switch (type) {
case AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN: {
AVIAMFMixGain *mix = subblock;
mix->subblock_duration = subblock_duration;
break;
}
case AV_IAMF_PARAMETER_DEFINITION_DEMIXING: {
AVIAMFDemixingInfo *demix = subblock;
demix->subblock_duration = subblock_duration;
// DefaultDemixingInfoParameterData
av_assert0(audio_element);
demix->dmixp_mode = avio_r8(pb) >> 5;
audio_element->element->default_w = avio_r8(pb) >> 4;
break;
}
case AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN: {
AVIAMFReconGain *recon = subblock;
recon->subblock_duration = subblock_duration;
break;
}
default:
av_free(param);
return AVERROR_INVALIDDATA;
}
}
param->parameter_id = parameter_id;
param->parameter_rate = parameter_rate;
param->duration = duration;
param->constant_subblock_duration = constant_subblock_duration;
param->nb_subblocks = nb_subblocks;
if (param_definition) {
if (param_definition->param_size != param_size || memcmp(param_definition->param, param, param_size)) {
av_log(s, AV_LOG_ERROR, "Incosistent parameters for parameter_id %u\n", parameter_id);
av_free(param);
return AVERROR_INVALIDDATA;
}
} else {
IAMFParamDefinition **tmp = av_realloc_array(c->param_definitions, c->nb_param_definitions + 1,
sizeof(*c->param_definitions));
if (!tmp) {
av_free(param);
return AVERROR(ENOMEM);
}
c->param_definitions = tmp;
param_definition = av_mallocz(sizeof(*param_definition));
if (!param_definition) {
av_free(param);
return AVERROR(ENOMEM);
}
param_definition->param = param;
param_definition->mode = !mode;
param_definition->param_size = param_size;
param_definition->audio_element = audio_element;
c->param_definitions[c->nb_param_definitions++] = param_definition;
}
av_assert0(out_param_definition);
*out_param_definition = param;
return 0;
}
static int audio_element_obu(void *s, IAMFContext *c, AVIOContext *pb, int len)
{
const IAMFCodecConfig *codec_config;
AVIAMFAudioElement *element;
IAMFAudioElement **tmp, *audio_element = NULL;
FFIOContext b;
AVIOContext *pbc;
uint8_t *buf;
unsigned audio_element_id, nb_substreams, codec_config_id, num_parameters;
int audio_element_type, ret;
buf = av_malloc(len);
if (!buf)
return AVERROR(ENOMEM);
ret = ffio_read_size(pb, buf, len);
if (ret < 0)
goto fail;
ffio_init_context(&b, buf, len, 0, NULL, NULL, NULL, NULL);
pbc = &b.pub;
audio_element_id = ffio_read_leb(pbc);
for (int i = 0; i < c->nb_audio_elements; i++)
if (c->audio_elements[i]->audio_element_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(pbc) >> 5;
if (audio_element_type > AV_IAMF_AUDIO_ELEMENT_TYPE_SCENE) {
av_log(s, AV_LOG_DEBUG, "Unknown audio_element_type referenced in an audio element. Ignoring\n");
ret = 0;
goto fail;
}
codec_config_id = ffio_read_leb(pbc);
codec_config = ff_iamf_get_codec_config(c, codec_config_id);
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;
}
tmp = av_realloc_array(c->audio_elements, c->nb_audio_elements + 1, sizeof(*c->audio_elements));
if (!tmp) {
ret = AVERROR(ENOMEM);
goto fail;
}
c->audio_elements = tmp;
audio_element = av_mallocz(sizeof(*audio_element));
if (!audio_element) {
ret = AVERROR(ENOMEM);
goto fail;
}
nb_substreams = ffio_read_leb(pbc);
audio_element->codec_config_id = codec_config_id;
audio_element->audio_element_id = audio_element_id;
audio_element->substreams = av_calloc(nb_substreams, sizeof(*audio_element->substreams));
if (!audio_element->substreams) {
ret = AVERROR(ENOMEM);
goto fail;
}
audio_element->nb_substreams = nb_substreams;
element = audio_element->element = av_iamf_audio_element_alloc();
if (!element) {
ret = AVERROR(ENOMEM);
goto fail;
}
audio_element->celement = element;
element->audio_element_type = audio_element_type;
for (int i = 0; i < audio_element->nb_substreams; i++) {
IAMFSubStream *substream = &audio_element->substreams[i];
substream->codecpar = avcodec_parameters_alloc();
if (!substream->codecpar) {
ret = AVERROR(ENOMEM);
goto fail;
}
substream->audio_substream_id = ffio_read_leb(pbc);
substream->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
substream->codecpar->codec_id = codec_config->codec_id;
substream->codecpar->frame_size = codec_config->nb_samples;
substream->codecpar->sample_rate = codec_config->sample_rate;
substream->codecpar->seek_preroll = -codec_config->audio_roll_distance * codec_config->nb_samples;
switch(substream->codecpar->codec_id) {
case AV_CODEC_ID_AAC:
case AV_CODEC_ID_FLAC:
case AV_CODEC_ID_OPUS:
substream->codecpar->extradata = av_malloc(codec_config->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE);
if (!substream->codecpar->extradata) {
ret = AVERROR(ENOMEM);
goto fail;
}
memcpy(substream->codecpar->extradata, codec_config->extradata, codec_config->extradata_size);
memset(substream->codecpar->extradata + codec_config->extradata_size, 0, AV_INPUT_BUFFER_PADDING_SIZE);
substream->codecpar->extradata_size = codec_config->extradata_size;
break;
}
}
num_parameters = ffio_read_leb(pbc);
if (num_parameters && audio_element_type != 0) {
av_log(s, AV_LOG_ERROR, "Audio Element parameter count %u is invalid"
" for Scene representations\n", num_parameters);
ret = AVERROR_INVALIDDATA;
goto fail;
}
for (int i = 0; i < num_parameters; i++) {
unsigned type;
type = ffio_read_leb(pbc);
if (type == AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN)
ret = AVERROR_INVALIDDATA;
else if (type == AV_IAMF_PARAMETER_DEFINITION_DEMIXING)
ret = param_parse(s, c, pbc, type, audio_element, &element->demixing_info);
else if (type == AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN)
ret = param_parse(s, c, pbc, type, audio_element, &element->recon_gain_info);
else {
unsigned param_definition_size = ffio_read_leb(pbc);
avio_skip(pbc, param_definition_size);
}
if (ret < 0)
goto fail;
}
if (audio_element_type == AV_IAMF_AUDIO_ELEMENT_TYPE_CHANNEL) {
ret = scalable_channel_layout_config(s, pbc, audio_element, codec_config);
if (ret < 0)
goto fail;
} else if (audio_element_type == AV_IAMF_AUDIO_ELEMENT_TYPE_SCENE) {
ret = ambisonics_config(s, pbc, audio_element, codec_config);
if (ret < 0)
goto fail;
} else {
av_assert0(0);
}
c->audio_elements[c->nb_audio_elements++] = audio_element;
len -= avio_tell(pbc);
if (len)
av_log(s, AV_LOG_WARNING, "Underread in audio_element_obu. %d bytes left at the end\n", len);
ret = 0;
fail:
av_free(buf);
if (ret < 0)
ff_iamf_free_audio_element(&audio_element);
return ret;
}
static int label_string(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(void *s, IAMFContext *c, AVIOContext *pb, int len)
{
AVIAMFMixPresentation *mix;
IAMFMixPresentation **tmp, *mix_presentation = NULL;
FFIOContext b;
AVIOContext *pbc;
uint8_t *buf;
unsigned nb_submixes, mix_presentation_id;
int ret;
buf = av_malloc(len);
if (!buf)
return AVERROR(ENOMEM);
ret = ffio_read_size(pb, buf, len);
if (ret < 0)
goto fail;
ffio_init_context(&b, buf, len, 0, NULL, NULL, NULL, NULL);
pbc = &b.pub;
mix_presentation_id = ffio_read_leb(pbc);
for (int i = 0; i < c->nb_mix_presentations; i++)
if (c->mix_presentations[i]->mix_presentation_id == mix_presentation_id) {
av_log(s, AV_LOG_ERROR, "Duplicate mix_presentation_id %d\n", mix_presentation_id);
ret = AVERROR_INVALIDDATA;
goto fail;
}
tmp = av_realloc_array(c->mix_presentations, c->nb_mix_presentations + 1, sizeof(*c->mix_presentations));
if (!tmp) {
ret = AVERROR(ENOMEM);
goto fail;
}
c->mix_presentations = tmp;
mix_presentation = av_mallocz(sizeof(*mix_presentation));
if (!mix_presentation) {
ret = AVERROR(ENOMEM);
goto fail;
}
mix_presentation->mix_presentation_id = mix_presentation_id;
mix = mix_presentation->mix = av_iamf_mix_presentation_alloc();
if (!mix) {
ret = AVERROR(ENOMEM);
goto fail;
}
mix_presentation->cmix = mix;
mix_presentation->count_label = ffio_read_leb(pbc);
mix_presentation->language_label = av_calloc(mix_presentation->count_label,
sizeof(*mix_presentation->language_label));
if (!mix_presentation->language_label) {
mix_presentation->count_label = 0;
ret = AVERROR(ENOMEM);
goto fail;
}
for (int i = 0; i < mix_presentation->count_label; i++) {
ret = label_string(pbc, &mix_presentation->language_label[i]);
if (ret < 0)
goto fail;
}
for (int i = 0; i < mix_presentation->count_label; i++) {
char *annotation = NULL;
ret = label_string(pbc, &annotation);
if (ret < 0)
goto fail;
ret = av_dict_set(&mix->annotations, mix_presentation->language_label[i], annotation,
AV_DICT_DONT_STRDUP_VAL | AV_DICT_DONT_OVERWRITE);
if (ret < 0)
goto fail;
}
nb_submixes = ffio_read_leb(pbc);
for (int i = 0; i < nb_submixes; i++) {
AVIAMFSubmix *sub_mix;
unsigned nb_elements, nb_layouts;
sub_mix = av_iamf_mix_presentation_add_submix(mix);
if (!sub_mix) {
ret = AVERROR(ENOMEM);
goto fail;
}
nb_elements = ffio_read_leb(pbc);
for (int j = 0; j < nb_elements; j++) {
AVIAMFSubmixElement *submix_element;
IAMFAudioElement *audio_element = NULL;
unsigned int rendering_config_extension_size;
submix_element = av_iamf_submix_add_element(sub_mix);
if (!submix_element) {
ret = AVERROR(ENOMEM);
goto fail;
}
submix_element->audio_element_id = ffio_read_leb(pbc);
for (int k = 0; k < c->nb_audio_elements; k++)
if (c->audio_elements[k]->audio_element_id == submix_element->audio_element_id) {
audio_element = c->audio_elements[k];
break;
}
if (!audio_element) {
av_log(s, AV_LOG_ERROR, "Invalid Audio Element with id %u referenced by Mix Parameters %u\n",
submix_element->audio_element_id, mix_presentation_id);
ret = AVERROR_INVALIDDATA;
goto fail;
}
for (int k = 0; k < mix_presentation->count_label; k++) {
char *annotation = NULL;
ret = label_string(pbc, &annotation);
if (ret < 0)
goto fail;
ret = av_dict_set(&submix_element->annotations, mix_presentation->language_label[k], annotation,
AV_DICT_DONT_STRDUP_VAL | AV_DICT_DONT_OVERWRITE);
if (ret < 0)
goto fail;
}
submix_element->headphones_rendering_mode = avio_r8(pbc) >> 6;
rendering_config_extension_size = ffio_read_leb(pbc);
avio_skip(pbc, rendering_config_extension_size);
ret = param_parse(s, c, pbc, AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN,
NULL,
&submix_element->element_mix_config);
if (ret < 0)
goto fail;
submix_element->default_mix_gain = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
}
ret = param_parse(s, c, pbc, AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN, NULL, &sub_mix->output_mix_config);
if (ret < 0)
goto fail;
sub_mix->default_mix_gain = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
nb_layouts = ffio_read_leb(pbc);
for (int j = 0; j < nb_layouts; j++) {
AVIAMFSubmixLayout *submix_layout;
int info_type;
int byte = avio_r8(pbc);
submix_layout = av_iamf_submix_add_layout(sub_mix);
if (!submix_layout) {
ret = AVERROR(ENOMEM);
goto fail;
}
submix_layout->layout_type = byte >> 6;
if (submix_layout->layout_type < AV_IAMF_SUBMIX_LAYOUT_TYPE_LOUDSPEAKERS ||
submix_layout->layout_type > AV_IAMF_SUBMIX_LAYOUT_TYPE_BINAURAL) {
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;
if (sound_system >= FF_ARRAY_ELEMS(ff_iamf_sound_system_map)) {
ret = AVERROR_INVALIDDATA;
goto fail;
}
av_channel_layout_copy(&submix_layout->sound_system, &ff_iamf_sound_system_map[sound_system].layout);
} else
submix_layout->sound_system = (AVChannelLayout)AV_CHANNEL_LAYOUT_BINAURAL;
info_type = avio_r8(pbc);
submix_layout->integrated_loudness = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
submix_layout->digital_peak = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
if (info_type & 1)
submix_layout->true_peak = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
if (info_type & 2) {
unsigned int num_anchored_loudness = avio_r8(pbc);
for (int k = 0; k < num_anchored_loudness; k++) {
unsigned int anchor_element = avio_r8(pbc);
AVRational anchored_loudness = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
if (anchor_element == IAMF_ANCHOR_ELEMENT_DIALOGUE)
submix_layout->dialogue_anchored_loudness = anchored_loudness;
else if (anchor_element <= IAMF_ANCHOR_ELEMENT_ALBUM)
submix_layout->album_anchored_loudness = anchored_loudness;
else
av_log(s, AV_LOG_DEBUG, "Unknown anchor_element. Ignoring\n");
}
}
if (info_type & 0xFC) {
unsigned int info_type_size = ffio_read_leb(pbc);
avio_skip(pbc, info_type_size);
}
}
}
c->mix_presentations[c->nb_mix_presentations++] = mix_presentation;
len -= avio_tell(pbc);
if (len)
av_log(s, AV_LOG_WARNING, "Underread in mix_presentation_obu. %d bytes left at the end\n", len);
ret = 0;
fail:
av_free(buf);
if (ret < 0)
ff_iamf_free_mix_presentation(&mix_presentation);
return ret;
}
int ff_iamf_parse_obu_header(const uint8_t *buf, int buf_size,
unsigned *obu_size, int *start_pos, enum IAMF_OBU_Type *type,
unsigned *skip_samples, unsigned *discard_padding)
{
GetBitContext gb;
int ret, extension_flag, trimming, start;
unsigned skip = 0, discard = 0;
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);
/*redundant =*/ get_bits1(&gb);
trimming = get_bits1(&gb);
extension_flag = get_bits1(&gb);
*obu_size = get_leb(&gb);
if (*obu_size > INT_MAX)
return AVERROR_INVALIDDATA;
start = get_bits_count(&gb) / 8;
if (trimming) {
discard = get_leb(&gb); // num_samples_to_trim_at_end
skip = get_leb(&gb); // num_samples_to_trim_at_start
}
if (skip_samples)
*skip_samples = skip;
if (discard_padding)
*discard_padding = discard;
if (extension_flag) {
unsigned int extension_bytes;
extension_bytes = get_leb(&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;
return size;
}
int ff_iamfdec_read_descriptors(IAMFContext *c, AVIOContext *pb,
int max_size, void *log_ctx)
{
uint8_t header[MAX_IAMF_OBU_HEADER_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
int ret;
while (1) {
unsigned obu_size;
enum IAMF_OBU_Type type;
int start_pos, len, size;
if ((ret = ffio_ensure_seekback(pb, FFMIN(MAX_IAMF_OBU_HEADER_SIZE, max_size))) < 0)
return ret;
size = avio_read(pb, header, FFMIN(MAX_IAMF_OBU_HEADER_SIZE, max_size));
if (size < 0)
return size;
memset(header + size, 0, AV_INPUT_BUFFER_PADDING_SIZE);
len = ff_iamf_parse_obu_header(header, size, &obu_size, &start_pos, &type, NULL, NULL);
if (len < 0 || obu_size > max_size) {
av_log(log_ctx, AV_LOG_ERROR, "Failed to read obu header\n");
avio_seek(pb, -size, SEEK_CUR);
return len;
}
if (type >= IAMF_OBU_IA_PARAMETER_BLOCK && type < IAMF_OBU_IA_SEQUENCE_HEADER) {
avio_seek(pb, -size, SEEK_CUR);
break;
}
avio_seek(pb, -(size - start_pos), SEEK_CUR);
switch (type) {
case IAMF_OBU_IA_CODEC_CONFIG:
ret = codec_config_obu(log_ctx, c, pb, obu_size);
break;
case IAMF_OBU_IA_AUDIO_ELEMENT:
ret = audio_element_obu(log_ctx, c, pb, obu_size);
break;
case IAMF_OBU_IA_MIX_PRESENTATION:
ret = mix_presentation_obu(log_ctx, c, pb, obu_size);
break;
default: {
int64_t offset = avio_skip(pb, obu_size);
if (offset < 0)
ret = offset;
break;
}
}
if (ret < 0) {
av_log(log_ctx, AV_LOG_ERROR, "Failed to read obu type %d\n", type);
return ret;
}
max_size -= obu_size + start_pos;
if (max_size < 0)
return AVERROR_INVALIDDATA;
if (!max_size)
break;
}
return 0;
}
Reference in New Issue View Git Blame Copy Permalink