[FFmpeg-devel] [PATCH] check for mod by zero (issue 2502)
Daniel Kang
daniel.d.kang
Tue Jan 11 03:43:54 CET 2011
On Mon, Jan 10, 2011 at 8:11 PM, Michael Niedermayer <michaelni at gmx.at>wrote:
> On Mon, Jan 10, 2011 at 07:01:02PM -0500, Daniel Kang wrote:
> > On Mon, Jan 10, 2011 at 3:51 PM, Michael Niedermayer <michaelni at gmx.at
> >wrote:
> >
> > > On Sat, Jan 08, 2011 at 04:39:35PM -0500, Daniel Kang wrote:
> > > > On Sat, Jan 8, 2011 at 3:45 PM, Justin Ruggles <
> justin.ruggles at gmail.com
> > > >wrote:
> > > >
> > > > > On 01/08/2011 01:52 PM, Daniel Kang wrote:
> > > > >
> > > > > > On Sat, Jan 8, 2011 at 9:33 AM, Justin Ruggles <
> > > justin.ruggles at gmail.com
> > > > > >wrote:
> > > > > >
> > > > > >> On 01/07/2011 11:33 PM, Daniel Kang wrote:
> > > > > >>> diff --git a/libavformat/vocdec.c b/libavformat/vocdec.c
> > > > > >>> index 909520c..6c37246 100644
> > > > > >>> --- a/libavformat/vocdec.c
> > > > > >>> +++ b/libavformat/vocdec.c
> > > > > >>> @@ -68,7 +68,7 @@ voc_get_packet(AVFormatContext *s, AVPacket
> *pkt,
> > > > > >> AVStream *st, int max_size)
> > > > > >>> AVCodecContext *dec = st->codec;
> > > > > >>> ByteIOContext *pb = s->pb;
> > > > > >>> VocType type;
> > > > > >>> - int size;
> > > > > >>> + int size, tmp;
> > > > > >>> int sample_rate = 0;
> > > > > >>> int channels = 1;
> > > > > >>>
> > > > > >>> @@ -90,7 +90,11 @@ voc_get_packet(AVFormatContext *s, AVPacket
> > > *pkt,
> > > > > >> AVStream *st, int max_size)
> > > > > >>> if (sample_rate)
> > > > > >>> dec->sample_rate = sample_rate;
> > > > > >>> dec->channels = channels;
> > > > > >>> - dec->codec_id = ff_codec_get_id(ff_voc_codec_tags,
> > > > > >> get_byte(pb));
> > > > > >>> + tmp = ff_codec_get_id(ff_voc_codec_tags,
> > > get_byte(pb));
> > > > > >>> + if (dec->codec_id != CODEC_ID_NONE)
> > > > > >>> + dec->codec_id =
> ff_codec_get_id(ff_voc_codec_tags,
> > > > > >> get_byte(pb));
> > > > > >>> + else
> > > > > >>> + av_log(s, AV_LOG_ERROR, "Unknown codec ID,
> > > continuing
> > > > > to
> > > > > >> decode\n");
> > > > > >>> dec->bits_per_coded_sample =
> > > > > >> av_get_bits_per_sample(dec->codec_id);
> > > > > >>> voc->remaining_size -= 2;
> > > > > >>> max_size -= 2;
> > > > > >>> @@ -113,7 +117,11 @@ voc_get_packet(AVFormatContext *s,
> AVPacket
> > > *pkt,
> > > > > >> AVStream *st, int max_size)
> > > > > >>> dec->sample_rate = get_le32(pb);
> > > > > >>> dec->bits_per_coded_sample = get_byte(pb);
> > > > > >>> dec->channels = get_byte(pb);
> > > > > >>> - dec->codec_id = ff_codec_get_id(ff_voc_codec_tags,
> > > > > >> get_le16(pb));
> > > > > >>> + tmp = ff_codec_get_id(ff_voc_codec_tags,
> > > get_byte(pb));
> > > > > >>> + if (dec->codec_id != CODEC_ID_NONE)
> > > > > >>> + dec->codec_id =
> ff_codec_get_id(ff_voc_codec_tags,
> > > > > >> get_byte(pb));
> > > > > >>> + else
> > > > > >>> + av_log(s, AV_LOG_ERROR, "Unknown codec ID,
> > > continuing
> > > > > to
> > > > > >> decode\n");
> > > > > >>> url_fskip(pb, 4);
> > > > > >>> voc->remaining_size -= 12;
> > > > > >>> max_size -= 12;
> > > > > >>
> > > > > >>
> > > > > >> This is not correct. It is reading the codec code twice. It is
> not
> > > > > >> checking to make sure the final codec_id != CODEC_ID_NONE. And
> it
> > > will
> > > > > >> never set a valid codec_id. (note that in this case codec_id is
> not
> > > set
> > > > > >> at read_header() as is normally the case for demuxers. it is set
> the
> > > > > >> first time an audio packet is read)
> > > > > >>
> > > > > >> My suggestion:
> > > > > >> switch(){
> > > > > >> ...
> > > > > >> case VOC_TYPE_VOICE_DATA:
> > > > > >> ...
> > > > > >> tmp_codec = ff_codec_get_id(ff_voc_codec_tags, get_byte(pb));
> > > > > >> if (dec->codec_id == CODEC_ID_NONE)
> > > > > >> dec->codec_id = tmp_codec;
> > > > > >> else if (tmp_codec != dec->codec_id)
> > > > > >> print AV_LOG_WARNING message about ignoring changed codec
> > > code
> > > > > >> ...
> > > > > >> case VOC_TYPE_NEW_VOICE_DATA:
> > > > > >> same as above
> > > > > >> ...
> > > > > >> }
> > > > > >> if (dec->codec_id == CODEC_ID_NONE) {
> > > > > >> print error message
> > > > > >> return AVERROR(EINVAL);
> > > > > >> }
> > > > > >>
> > > > > >> -Justin
> > > > > >>
> > > > > >
> > > > > > I have updated the patch with your comments. I am not sure if
> > > > > > sample_size can be negative or not, so I have just checked if it
> is
> > > not
> > > > > > zero.
> > > > >
> > > > >
> > > > > > + tmp_codec = ff_codec_get_id(ff_voc_codec_tags,
> > > > > get_byte(pb));
> > > > > > + if (dec->codec_id == CODEC_ID_NONE)
> > > > > > + dec->codec_id = tmp_codec;
> > > > > > + else if (dec->codec_id != tmp_codec)
> > > > > > + av_log(s, AV_LOG_WARNING, "Ignoring change in
> codec,
> > > > > continuing to decode\n");
> > > > >
> > > > >
> > > > > Technically, the demuxer does not decode, it demuxes. I would just
> say
> > > > > "Ignoring mid-stream change in audio codec".
> > > > >
> > > > > -Justin
> > > > >
> > > >
> > > > I have updated the message.
> > >
> > > > libavcodec/pcm.c | 10 ++++++++++
> > > > libavformat/vocdec.c | 18 +++++++++++++++---
> > > > 2 files changed, 25 insertions(+), 3 deletions(-)
> > > > 65c7d12fe75a01827377e1adb4dfa2a7b7015dd6 pcm_sanity_check.diff
> > > > From f9905f8b8c077d079951344c7d25646589c4c5aa Mon Sep 17 00:00:00
> 2001
> > > > From: Daniel Kang <daniel.d.kang at gmail.com>
> > > > Date: Thu, 6 Jan 2011 21:03:27 -0500
> > > > Subject: [PATCH] Add check for pcm files.
> > > >
> > > > ---
> > > > libavcodec/pcm.c | 10 ++++++++++
> > > > libavformat/vocdec.c | 18 +++++++++++++++---
> > > > 2 files changed, 25 insertions(+), 3 deletions(-)
> > > >
> > > > diff --git a/libavcodec/pcm.c b/libavcodec/pcm.c
> > > > index b6b49dc..e321a4d 100644
> > > > --- a/libavcodec/pcm.c
> > > > +++ b/libavcodec/pcm.c
> > > > @@ -272,6 +272,11 @@ static int pcm_decode_frame(AVCodecContext
> *avctx,
> > > > samples = data;
> > > > src = buf;
> > > >
> > > > + if (avctx->codec_id == CODEC_ID_NONE) {
> > > > + av_log(avctx, AV_LOG_ERROR, "invalid codec_id\n");
> > > > + return AVERROR(EINVAL);
> > > > + }
> > > > +
> > > > if (avctx->sample_fmt!=avctx->codec->sample_fmts[0]) {
> > > > av_log(avctx, AV_LOG_ERROR, "invalid sample_fmt\n");
> > > > return -1;
> > > > @@ -292,6 +297,11 @@ static int pcm_decode_frame(AVCodecContext
> *avctx,
> > > > /* we process 40-bit blocks per channel for LXF */
> > > > sample_size = 5;
> > > >
> > > > + if (sample_size == 0) {
> > > > + av_log(avctx, AV_LOG_ERROR, "Invalid sample_size\n");
> > > > + return AVERROR(EINVAL);
> > > > + }
> > >
> > > the codec_id test looks unneeded
> >
> >
> > I have removed the codec_id test.
>
> > libavcodec/pcm.c | 5 +++++
> > libavformat/vocdec.c | 18 +++++++++++++++---
> > 2 files changed, 20 insertions(+), 3 deletions(-)
> > c93b9039d5fd6aa892aa6b0fc2e876a1edd250a8 pcm_sanity_check.diff
> > From 3d4fc3a6e0740cb4bcea52515e76f7e7e425d9cd Mon Sep 17 00:00:00 2001
> > From: Daniel Kang <daniel.d.kang at gmail.com>
> > Date: Thu, 6 Jan 2011 21:03:27 -0500
> > Subject: [PATCH] Add check for pcm files.
> >
> > ---
> > libavcodec/pcm.c | 5 +++++
> > libavformat/vocdec.c | 18 +++++++++++++++---
> > 2 files changed, 20 insertions(+), 3 deletions(-)
> >
> > diff --git a/libavcodec/pcm.c b/libavcodec/pcm.c
> > index b6b49dc..533e834 100644
> > --- a/libavcodec/pcm.c
> > +++ b/libavcodec/pcm.c
> > @@ -292,6 +292,11 @@ static int pcm_decode_frame(AVCodecContext *avctx,
> > /* we process 40-bit blocks per channel for LXF */
> > sample_size = 5;
> >
> > + if (sample_size == 0) {
> > + av_log(avctx, AV_LOG_ERROR, "Invalid sample_size\n");
> > + return AVERROR(EINVAL);
> > + }
> > +
> > n = avctx->channels * sample_size;
> >
> > if(n && buf_size % n){
> > diff --git a/libavformat/vocdec.c b/libavformat/vocdec.c
> > index 909520c..835417e 100644
> > --- a/libavformat/vocdec.c
> > +++ b/libavformat/vocdec.c
> > @@ -68,7 +68,7 @@ voc_get_packet(AVFormatContext *s, AVPacket *pkt,
> AVStream *st, int max_size)
> > AVCodecContext *dec = st->codec;
> > ByteIOContext *pb = s->pb;
> > VocType type;
> > - int size;
> > + int size, tmp_codec;
> > int sample_rate = 0;
> > int channels = 1;
> >
> > @@ -90,7 +90,11 @@ voc_get_packet(AVFormatContext *s, AVPacket *pkt,
> AVStream *st, int max_size)
> > if (sample_rate)
> > dec->sample_rate = sample_rate;
> > dec->channels = channels;
> > - dec->codec_id = ff_codec_get_id(ff_voc_codec_tags,
> get_byte(pb));
> > + tmp_codec = ff_codec_get_id(ff_voc_codec_tags,
> get_byte(pb));
> > + if (dec->codec_id == CODEC_ID_NONE)
> > + dec->codec_id = tmp_codec;
> > + else if (dec->codec_id != tmp_codec)
> > + av_log(s, AV_LOG_WARNING, "Ignoring mid-stream change in
> audio codec\n");
> > dec->bits_per_coded_sample =
> av_get_bits_per_sample(dec->codec_id);
> > voc->remaining_size -= 2;
> > max_size -= 2;
> > @@ -113,7 +117,11 @@ voc_get_packet(AVFormatContext *s, AVPacket *pkt,
> AVStream *st, int max_size)
> > dec->sample_rate = get_le32(pb);
> > dec->bits_per_coded_sample = get_byte(pb);
> > dec->channels = get_byte(pb);
> > - dec->codec_id = ff_codec_get_id(ff_voc_codec_tags,
> get_le16(pb));
> > + tmp_codec = ff_codec_get_id(ff_voc_codec_tags,
> get_byte(pb));
> > + if (dec->codec_id == CODEC_ID_NONE)
> > + dec->codec_id = tmp_codec;
> > + else if (dec->codec_id != tmp_codec)
> > + av_log(s, AV_LOG_WARNING, "Ignoring mid-stream change in
> audio codec\n");
> > url_fskip(pb, 4);
> > voc->remaining_size -= 12;
> > max_size -= 12;
>
>
> > @@ -125,6 +133,10 @@ voc_get_packet(AVFormatContext *s, AVPacket *pkt,
> AVStream *st, int max_size)
> > voc->remaining_size = 0;
> > break;
> > }
> > + if (dec->codec_id == CODEC_ID_NONE) {
> > + av_log(s, AV_LOG_ERROR, "Invalid codec_id\n");
> > + return AVERROR(EINVAL);
> > + }
> > }
>
> Its possible to override the codec id see AVFormatContext.audio_codec_id
> so i think dec->codec_id == CODEC_ID_NONE is not neccesary fatal though
> surely
> not normal either, that is the av_log() is ok but i think we should
> continue
> (at least if AVFormatContext.audio_codec_id is set)
That check will only be triggered if dec->codec_id == CODEC_ID_NONE and
then the codec_id that is read in is invalid. In this case, I do not
think AVFormatContext.audio_codec_id could be set to anything valid.
I have updated the patch so that the error returns only if
s->audio_codec_id == CODEC_ID_NONE.
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