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Code Issues Releases Activity

Do not crash for illegal sample size, fixes issue 2502.

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Patch by Daniel Kang, daniel.d.kang at gmail

Originally committed as revision 26309 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Daniel Kang 2011-01-11 14:08:45 +00:00 committed by Carl Eugen Hoyos
parent 440d761e40
commit e048a9cab1
2 changed files with 20 additions and 3 deletions
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5
libavcodec/pcm.c
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@ -292,6 +292,11 @@ static int pcm_decode_frame(AVCodecContext *avctx,
/* we process 40-bit blocks per channel for LXF */ /* we process 40-bit blocks per channel for LXF */
sample_size = 5; 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; n = avctx->channels * sample_size;
if(n && buf_size % n){ if(n && buf_size % n){

18
libavformat/vocdec.c
View File

@ -68,7 +68,7 @@ voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
AVCodecContext *dec = st->codec; AVCodecContext *dec = st->codec;
ByteIOContext *pb = s->pb; ByteIOContext *pb = s->pb;
VocType type; VocType type;
int size; int size, tmp_codec;
int sample_rate = 0; int sample_rate = 0;
int channels = 1; int channels = 1;
@ -90,7 +90,11 @@ voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
if (sample_rate) if (sample_rate)
dec->sample_rate = sample_rate; dec->sample_rate = sample_rate;
dec->channels = channels; 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); dec->bits_per_coded_sample = av_get_bits_per_sample(dec->codec_id);
voc->remaining_size -= 2; voc->remaining_size -= 2;
max_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->sample_rate = get_le32(pb);
dec->bits_per_coded_sample = get_byte(pb); dec->bits_per_coded_sample = get_byte(pb);
dec->channels = 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); url_fskip(pb, 4);
voc->remaining_size -= 12; voc->remaining_size -= 12;
max_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; voc->remaining_size = 0;
break; break;
} }
if (dec->codec_id == CODEC_ID_NONE) {
av_log(s, AV_LOG_ERROR, "Invalid codec_id\n");
if (s->audio_codec_id == CODEC_ID_NONE) return AVERROR(EINVAL);
}
} }
dec->bit_rate = dec->sample_rate * dec->bits_per_coded_sample; dec->bit_rate = dec->sample_rate * dec->bits_per_coded_sample;