[FFmpeg-devel] [PATCH] avformat: add tri-ACE demuxer

[Andreas Rheinhardt]

Paul B Mahol:
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
> ---
>  libavformat/Makefile     |   1 +
>  libavformat/acedec.c     | 114 +++++++++++++++++++++++++++++++++++++++
>  libavformat/allformats.c |   1 +
>  3 files changed, 116 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..c6d0c33e52
> --- /dev/null
> +++ b/libavformat/acedec.c
> @@ -0,0 +1,114 @@
> +/*
> + * 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 / 2 + 1;
> +}
> +
> +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 || nb_channels > INT32_MAX / 256)

This check is not enough to rule out overflow 2048 * par->channels, let
alone truncation when writing the 16 bit value.

> +        return AVERROR_INVALIDDATA;
> +    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 = st->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) * 1024LL;
> +    par->codec_id    = AV_CODEC_ID_ATRAC3;
> +
> +    ret = ff_alloc_extradata(par, 14);
> +    if (ret < 0)
> +        return ret;
> +
> +    AV_WL16(st->codecpar->extradata, 1);
> +    AV_WL16(st->codecpar->extradata+2, 2048 * par->channels);
> +    AV_WL16(st->codecpar->extradata+4, 0);
> +    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);
> +    AV_WL16(st->codecpar->extradata+12, 0);
> +
> +    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);

As James has mentioned: par->block_align can't be zero.

> +}
> +
> +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;
>