ffmpeg/libavformat/iamfdec.c

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/*
* 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 "libavutil/avassert.h"
#include "libavutil/intreadwrite.h"
#include "avformat.h"
#include "demux.h"
#include "iamf.h"
#include "iamf_reader.h"
#include "iamf_parse.h"
#include "internal.h"
//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 = ff_iamf_parse_obu_header(p->buf + cnt, p->buf_size - cnt,
&obu_size, &start_pos, &type,
NULL, NULL);
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 int iamf_read_header(AVFormatContext *s)
{
IAMFDemuxContext *const c = s->priv_data;
IAMFContext *const iamf = &c->iamf;
int ret;
ret = ff_iamfdec_read_descriptors(iamf, s->pb, INT_MAX, s);
if (ret < 0)
return ret;
for (int i = 0; i < iamf->nb_audio_elements; i++) {
IAMFAudioElement *audio_element = iamf->audio_elements[i];
AVStreamGroup *stg = avformat_stream_group_create(s, AV_STREAM_GROUP_PARAMS_IAMF_AUDIO_ELEMENT, NULL);
if (!stg)
return AVERROR(ENOMEM);
av_iamf_audio_element_free(&stg->params.iamf_audio_element);
stg->id = audio_element->audio_element_id;
/* Transfer ownership */
stg->params.iamf_audio_element = audio_element->element;
audio_element->element = NULL;
for (int j = 0; j < audio_element->nb_substreams; j++) {
IAMFSubStream *substream = &audio_element->substreams[j];
AVStream *st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
ret = avformat_stream_group_add_stream(stg, st);
if (ret < 0)
return ret;
ret = avcodec_parameters_copy(st->codecpar, substream->codecpar);
if (ret < 0)
return ret;
if (!i && !j && audio_element->layers[0].substream_count == 1)
st->disposition |= AV_DISPOSITION_DEFAULT;
else
st->disposition |= AV_DISPOSITION_DEPENDENT;
st->id = substream->audio_substream_id;
avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
}
}
for (int i = 0; i < iamf->nb_mix_presentations; i++) {
IAMFMixPresentation *mix_presentation = iamf->mix_presentations[i];
AVStreamGroup *stg = avformat_stream_group_create(s, AV_STREAM_GROUP_PARAMS_IAMF_MIX_PRESENTATION, NULL);
const AVIAMFMixPresentation *mix = mix_presentation->cmix;
if (!stg)
return AVERROR(ENOMEM);
av_iamf_mix_presentation_free(&stg->params.iamf_mix_presentation);
stg->id = mix_presentation->mix_presentation_id;
/* Transfer ownership */
stg->params.iamf_mix_presentation = mix_presentation->mix;
mix_presentation->mix = NULL;
for (int j = 0; j < mix->nb_submixes; j++) {
const AVIAMFSubmix *sub_mix = mix->submixes[j];
for (int k = 0; k < sub_mix->nb_elements; k++) {
const AVIAMFSubmixElement *submix_element = sub_mix->elements[k];
AVStreamGroup *audio_element = NULL;
for (int l = 0; l < s->nb_stream_groups; l++)
if (s->stream_groups[l]->type == AV_STREAM_GROUP_PARAMS_IAMF_AUDIO_ELEMENT &&
s->stream_groups[l]->id == submix_element->audio_element_id) {
audio_element = s->stream_groups[l];
break;
}
av_assert0(audio_element);
for (int l = 0; l < audio_element->nb_streams; l++) {
ret = avformat_stream_group_add_stream(stg, audio_element->streams[l]);
if (ret < 0 && ret != AVERROR(EEXIST))
return ret;
}
}
}
}
if (!s->nb_streams)
return AVERROR_INVALIDDATA;
return 0;
}
static int iamf_read_packet(AVFormatContext *s, AVPacket *pkt)
{
IAMFDemuxContext *const c = s->priv_data;
int ret;
ret = ff_iamf_read_packet(s, c, s->pb, INT_MAX, pkt);
if (ret < 0)
return ret;
return 0;
}
static int iamf_read_close(AVFormatContext *s)
{
IAMFDemuxContext *const c = s->priv_data;
ff_iamf_read_deinit(c);
return 0;
}
const FFInputFormat ff_iamf_demuxer = {
.p.name = "iamf",
.p.long_name = NULL_IF_CONFIG_SMALL("Raw Immersive Audio Model and Formats"),
.p.extensions = "iamf",
.p.flags = AVFMT_GENERIC_INDEX | AVFMT_NO_BYTE_SEEK | AVFMT_NOTIMESTAMPS | AVFMT_SHOW_IDS,
.priv_data_size = sizeof(IAMFDemuxContext),
.flags_internal = FF_INFMT_FLAG_INIT_CLEANUP,
.read_probe = iamf_probe,
.read_header = iamf_read_header,
.read_packet = iamf_read_packet,
.read_close = iamf_read_close,
};