On 9/24/2020 1:42 PM, Paul B Mahol wrote: > Signed-off-by: Paul B Mahol <one...@gmail.com> > --- > libavformat/Makefile | 1 + > libavformat/acedec.c | 113 +++++++++++++++++++++++++++++++++++++++ > libavformat/allformats.c | 1 + > 3 files changed, 115 insertions(+) > create mode 100644 libavformat/acedec.c > > diff --git a/libavformat/Makefile b/libavformat/Makefile > index 213004cb45..e82af409b7 100644 > --- a/libavformat/Makefile > +++ b/libavformat/Makefile > @@ -71,6 +71,7 @@ OBJS-$(CONFIG_AAC_DEMUXER) += aacdec.o > apetag.o img2.o rawdec.o > OBJS-$(CONFIG_AAX_DEMUXER) += aaxdec.o > OBJS-$(CONFIG_AC3_DEMUXER) += ac3dec.o rawdec.o > OBJS-$(CONFIG_AC3_MUXER) += rawenc.o > +OBJS-$(CONFIG_ACE_DEMUXER) += acedec.o > OBJS-$(CONFIG_ACM_DEMUXER) += acm.o rawdec.o > OBJS-$(CONFIG_ACT_DEMUXER) += act.o > OBJS-$(CONFIG_ADF_DEMUXER) += bintext.o sauce.o > diff --git a/libavformat/acedec.c b/libavformat/acedec.c > new file mode 100644 > index 0000000000..aeef343b74 > --- /dev/null > +++ b/libavformat/acedec.c > @@ -0,0 +1,113 @@ > +/* > + * ACE demuxer > + * Copyright (c) 2020 Paul B Mahol > + * > + * 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/intreadwrite.h" > +#include "avformat.h" > +#include "internal.h" > + > +static int ace_probe(const AVProbeData *p) > +{ > + uint32_t asc; > + > + if (AV_RB32(p->buf) != MKBETAG('A','A','C',' ')) > + return 0; > + if (p->buf_size < 0x44) > + return 0; > + asc = AV_RB32(p->buf + 0x40); > + if (asc < 0x44 || asc > p->buf_size - 4) > + return 0; > + if (AV_RB32(p->buf + asc) != MKBETAG('A','S','C',' ')) > + return 0; > + > + return AVPROBE_SCORE_MAX; > +} > + > +static int ace_read_header(AVFormatContext *s) > +{ > + AVIOContext *pb = s->pb; > + AVCodecParameters *par; > + int ret, codec, rate, nb_channels; > + uint32_t asc_pos, size; > + AVStream *st; > + > + avio_skip(pb, 0x40); > + asc_pos = avio_rb32(pb); > + if (asc_pos < 0x44) > + return AVERROR_INVALIDDATA; > + avio_skip(pb, asc_pos - 0x44); > + if (avio_rb32(pb) != MKBETAG('A','S','C',' ')) > + return AVERROR_INVALIDDATA; > + avio_skip(pb, 0xec); > + codec = avio_rb32(pb); > + nb_channels = avio_rb32(pb); > + if (nb_channels <= 0) > + return AVERROR_INVALIDDATA;
You should also ensure the amount of channels is sane (Atrac3 seems to be 8 channels max, so limit it to that). Otherwise the block_align line below can overflow. > + size = avio_rb32(pb); > + if (size == 0) > + return AVERROR_INVALIDDATA; > + rate = avio_rb32(pb); > + if (rate <= 0) > + return AVERROR_INVALIDDATA; > + avio_skip(pb, 16); > + > + st = avformat_new_stream(s, NULL); > + if (!st) > + return AVERROR(ENOMEM); > + st->start_time = 0; > + par = s->streams[0]->codecpar; > + par->codec_type = AVMEDIA_TYPE_AUDIO; > + par->channels = nb_channels; > + par->sample_rate = rate; > + par->block_align = (codec == 4 ? 0x60 : codec == 5 ? 0x98 : 0xC0) * > nb_channels; > + st->duration = (size / par->block_align) * 1024; > + par->codec_id = AV_CODEC_ID_ATRAC3; > + > + ret = ff_alloc_extradata(st->codecpar, 14); > + if (ret < 0) > + return ret; > + > + memset(st->codecpar->extradata, 0, st->codecpar->extradata_size); > + AV_WL16(st->codecpar->extradata, 1); > + AV_WL16(st->codecpar->extradata+2, 2048 * st->codecpar->channels); > + AV_WL16(st->codecpar->extradata+6, codec == 4 ? 1 : 0); > + AV_WL16(st->codecpar->extradata+8, codec == 4 ? 1 : 0); > + AV_WL16(st->codecpar->extradata+10, 1); > + > + avpriv_set_pts_info(st, 64, 1, par->sample_rate); > + > + return 0; > +} > + > +static int ace_read_packet(AVFormatContext *s, AVPacket *pkt) > +{ > + AVCodecParameters *par = s->streams[0]->codecpar; > + > + return av_get_packet(s->pb, pkt, par->block_align ? par->block_align : > 1024 * par->channels); When would par->block_align be 0? > +} > + > +AVInputFormat ff_ace_demuxer = { > + .name = "ace", > + .long_name = NULL_IF_CONFIG_SMALL("tri-Ace Audio Container"), > + .read_probe = ace_probe, > + .read_header = ace_read_header, > + .read_packet = ace_read_packet, > + .flags = AVFMT_GENERIC_INDEX, > +}; > diff --git a/libavformat/allformats.c b/libavformat/allformats.c > index 3a6b8e6dac..9d85b8ccc3 100644 > --- a/libavformat/allformats.c > +++ b/libavformat/allformats.c > @@ -34,6 +34,7 @@ extern AVInputFormat ff_aac_demuxer; > extern AVInputFormat ff_aax_demuxer; > extern AVInputFormat ff_ac3_demuxer; > extern AVOutputFormat ff_ac3_muxer; > +extern AVInputFormat ff_ace_demuxer; > extern AVInputFormat ff_acm_demuxer; > extern AVInputFormat ff_act_demuxer; > extern AVInputFormat ff_adf_demuxer; > _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".