mirror of https://git.ffmpeg.org/ffmpeg.git
196 lines
6.5 KiB
C
196 lines
6.5 KiB
C
/*
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* Immersive Audio Model and Formats demuxer
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* Copyright (c) 2023 James Almer <jamrial@gmail.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/avassert.h"
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#include "libavutil/intreadwrite.h"
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#include "avformat.h"
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#include "demux.h"
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#include "iamf.h"
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#include "iamf_reader.h"
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#include "iamf_parse.h"
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#include "internal.h"
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//return < 0 if we need more data
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static int get_score(const uint8_t *buf, int buf_size, enum IAMF_OBU_Type type, int *seq)
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{
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if (type == IAMF_OBU_IA_SEQUENCE_HEADER) {
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if (buf_size < 4 || AV_RB32(buf) != MKBETAG('i','a','m','f'))
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return 0;
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*seq = 1;
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return -1;
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}
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if (type >= IAMF_OBU_IA_CODEC_CONFIG && type <= IAMF_OBU_IA_TEMPORAL_DELIMITER)
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return *seq ? -1 : 0;
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if (type >= IAMF_OBU_IA_AUDIO_FRAME && type <= IAMF_OBU_IA_AUDIO_FRAME_ID17)
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return *seq ? AVPROBE_SCORE_EXTENSION + 1 : 0;
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return 0;
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}
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static int iamf_probe(const AVProbeData *p)
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{
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unsigned obu_size;
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enum IAMF_OBU_Type type;
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int seq = 0, cnt = 0, start_pos;
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int ret;
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while (1) {
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int size = ff_iamf_parse_obu_header(p->buf + cnt, p->buf_size - cnt,
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&obu_size, &start_pos, &type,
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NULL, NULL);
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if (size < 0)
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return 0;
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ret = get_score(p->buf + cnt + start_pos,
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p->buf_size - cnt - start_pos,
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type, &seq);
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if (ret >= 0)
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return ret;
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cnt += FFMIN(size, p->buf_size - cnt);
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}
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return 0;
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}
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static int iamf_read_header(AVFormatContext *s)
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{
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IAMFDemuxContext *const c = s->priv_data;
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IAMFContext *const iamf = &c->iamf;
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int ret;
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ret = ff_iamfdec_read_descriptors(iamf, s->pb, INT_MAX, s);
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if (ret < 0)
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return ret;
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for (int i = 0; i < iamf->nb_audio_elements; i++) {
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IAMFAudioElement *audio_element = iamf->audio_elements[i];
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AVStreamGroup *stg = avformat_stream_group_create(s, AV_STREAM_GROUP_PARAMS_IAMF_AUDIO_ELEMENT, NULL);
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if (!stg)
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return AVERROR(ENOMEM);
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av_iamf_audio_element_free(&stg->params.iamf_audio_element);
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stg->id = audio_element->audio_element_id;
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/* Transfer ownership */
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stg->params.iamf_audio_element = audio_element->element;
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audio_element->element = NULL;
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for (int j = 0; j < audio_element->nb_substreams; j++) {
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IAMFSubStream *substream = &audio_element->substreams[j];
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AVStream *st = avformat_new_stream(s, NULL);
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if (!st)
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return AVERROR(ENOMEM);
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ret = avformat_stream_group_add_stream(stg, st);
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if (ret < 0)
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return ret;
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ret = avcodec_parameters_copy(st->codecpar, substream->codecpar);
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if (ret < 0)
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return ret;
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if (!i && !j && audio_element->layers[0].substream_count == 1)
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st->disposition |= AV_DISPOSITION_DEFAULT;
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else
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st->disposition |= AV_DISPOSITION_DEPENDENT;
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st->id = substream->audio_substream_id;
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avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
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}
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}
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for (int i = 0; i < iamf->nb_mix_presentations; i++) {
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IAMFMixPresentation *mix_presentation = iamf->mix_presentations[i];
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AVStreamGroup *stg = avformat_stream_group_create(s, AV_STREAM_GROUP_PARAMS_IAMF_MIX_PRESENTATION, NULL);
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const AVIAMFMixPresentation *mix = mix_presentation->cmix;
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if (!stg)
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return AVERROR(ENOMEM);
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av_iamf_mix_presentation_free(&stg->params.iamf_mix_presentation);
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stg->id = mix_presentation->mix_presentation_id;
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/* Transfer ownership */
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stg->params.iamf_mix_presentation = mix_presentation->mix;
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mix_presentation->mix = NULL;
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for (int j = 0; j < mix->nb_submixes; j++) {
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const AVIAMFSubmix *sub_mix = mix->submixes[j];
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for (int k = 0; k < sub_mix->nb_elements; k++) {
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const AVIAMFSubmixElement *submix_element = sub_mix->elements[k];
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AVStreamGroup *audio_element = NULL;
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for (int l = 0; l < s->nb_stream_groups; l++)
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if (s->stream_groups[l]->type == AV_STREAM_GROUP_PARAMS_IAMF_AUDIO_ELEMENT &&
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s->stream_groups[l]->id == submix_element->audio_element_id) {
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audio_element = s->stream_groups[l];
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break;
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}
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av_assert0(audio_element);
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for (int l = 0; l < audio_element->nb_streams; l++) {
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ret = avformat_stream_group_add_stream(stg, audio_element->streams[l]);
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if (ret < 0 && ret != AVERROR(EEXIST))
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return ret;
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}
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}
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}
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}
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if (!s->nb_streams)
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return AVERROR_INVALIDDATA;
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return 0;
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}
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static int iamf_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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IAMFDemuxContext *const c = s->priv_data;
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int ret;
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ret = ff_iamf_read_packet(s, c, s->pb, INT_MAX, pkt);
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if (ret < 0)
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return ret;
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return 0;
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}
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static int iamf_read_close(AVFormatContext *s)
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{
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IAMFDemuxContext *const c = s->priv_data;
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ff_iamf_read_deinit(c);
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return 0;
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}
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const FFInputFormat ff_iamf_demuxer = {
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.p.name = "iamf",
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.p.long_name = NULL_IF_CONFIG_SMALL("Raw Immersive Audio Model and Formats"),
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.p.extensions = "iamf",
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.p.flags = AVFMT_GENERIC_INDEX | AVFMT_NO_BYTE_SEEK | AVFMT_NOTIMESTAMPS | AVFMT_SHOW_IDS,
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.priv_data_size = sizeof(IAMFDemuxContext),
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.flags_internal = FF_INFMT_FLAG_INIT_CLEANUP,
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.read_probe = iamf_probe,
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.read_header = iamf_read_header,
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.read_packet = iamf_read_packet,
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.read_close = iamf_read_close,
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};
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