lance.lmw...@gmail.com: > On Tue, May 11, 2021 at 03:45:51AM +0200, Andreas Rheinhardt wrote: >> lance.lmw...@gmail.com: >>> On Tue, May 11, 2021 at 01:27:09AM +0200, Andreas Rheinhardt wrote: >>>> lance.lmw...@gmail.com: >>>>> On Mon, May 10, 2021 at 09:52:48PM +0200, Andreas Rheinhardt wrote: >>>>>> lance.lmw...@gmail.com: >>>>>>> From: Limin Wang <lance.lmw...@gmail.com> >>>>>>> >>>>>>> This prevents OOM in case of data buffer size is insufficient. >>>>>> >>>>>> OOM? >>>>> Yes, it's invalid Out Of Memory access, not no memory available. >>>>> What's your suggestion? >>>>> >>>> >>>> What you mean is commonly called "buffer overflow"; OOM exclusively >>>> means "no memory available". >>>> I already told you what I think should be done below. >>> >>> Sorry, I didn't notice that. >>> >>>> >>>>>> >>>>>>> >>>>>>> Signed-off-by: Limin Wang <lance.lmw...@gmail.com> >>>>>>> --- >>>>>>> libavformat/hls.c | 4 +++- >>>>>>> libavformat/internal.h | 6 ++++-- >>>>>>> libavformat/rtpdec_latm.c | 6 ++++-- >>>>>>> libavformat/rtpdec_mpeg4.c | 6 ++++-- >>>>>>> libavformat/utils.c | 7 +++++-- >>>>>>> 5 files changed, 20 insertions(+), 9 deletions(-) >>>>>>> >>>>>>> diff --git a/libavformat/hls.c b/libavformat/hls.c >>>>>>> index 8fc6924..d7d0387 100644 >>>>>>> --- a/libavformat/hls.c >>>>>>> +++ b/libavformat/hls.c >>>>>>> @@ -800,7 +800,9 @@ static int parse_playlist(HLSContext *c, const char >>>>>>> *url, >>>>>>> if (!strcmp(info.method, "SAMPLE-AES")) >>>>>>> key_type = KEY_SAMPLE_AES; >>>>>>> if (!av_strncasecmp(info.iv, "0x", 2)) { >>>>>>> - ff_hex_to_data(iv, info.iv + 2); >>>>>>> + ret = ff_hex_to_data(iv, sizeof(iv), info.iv + 2); >>>>>>> + if (ret < 0) >>>>>>> + goto fail; >>>>>>> has_iv = 1; >>>>>>> } >>>>>>> av_strlcpy(key, info.uri, sizeof(key)); >>>>>>> diff --git a/libavformat/internal.h b/libavformat/internal.h >>>>>>> index d57e63c..3192aca 100644 >>>>>>> --- a/libavformat/internal.h >>>>>>> +++ b/libavformat/internal.h >>>>>>> @@ -423,10 +423,12 @@ char *ff_data_to_hex(char *buf, const uint8_t >>>>>>> *src, int size, int lowercase); >>>>>>> * digits is ignored. >>>>>>> * >>>>>>> * @param data if non-null, the parsed data is written to this pointer >>>>>>> + * @param data_size the data buffer size >>>>>>> * @param p the string to parse >>>>>>> - * @return the number of bytes written (or to be written, if data is >>>>>>> null) >>>>>>> + * @return the number of bytes written (or to be written, if data is >>>>>>> null), >>>>>>> + * or < 0 in case data_size is insufficient if data isn't null. >>>>>>> */ >>>>>>> -int ff_hex_to_data(uint8_t *data, const char *p); >>>>>>> +int ff_hex_to_data(uint8_t *data, int data_size, const char *p); >>>>>> >>>>>> This is unnecessary, as none of the callers need it: The rtpdec users >>>>>> call ff_hex_to_data() twice, once to get the necessary size, once to >>>>>> write the data. And the hls buffer is already big enough. I only see two >>> >>> Yes, most of caller is call ff_hex_to_data() twice, but hls is using iv[16], >>> We can't assume the string is 128bit only as cracker can change the m3u8 and >>> make the buffer overflow. For the data may be array, so I prefer to add the >>> memory overflow check. In theory, it's big enough, but we can't assume it. >>> >> >> How could this happen? Even if the m3u8 is changed concurrently, >> ff_parse_key_value() will always write a NUL-terminated string into >> info.iv and said string won't change even if the m3u8 is changed. > > I'm not talk about change the m3u8 after playing, in fact it can be changed > before play the m3u8. > For example below is one sample, the size of IV is 16 always, but if cracker > will change with > extra data, I think it'll make memory overflow. > #EXT-X-KEY:METHOD=AES-128,URI="https://license-server/video.key",IV=0x7b84a718bbac5e2053d64b3295ca2dce > -> add extra aaaa > #EXT-X-KEY:METHOD=AES-128,URI="https://license-server/video.key",IV=0x7b84a718bbac5e2053d64b3295ca2dceaaaa
ff_parse_key_value() already ensures not to write beyond the end of the buffer of info.iv and to zero-terminate the buffer (if it is used at all): In your second example, info.iv will be truncated; the "aaaa" won't be copied; if it were otherwise, then there would already be a buffer overflow in ff_parse_key_value(). (If you want to check for truncation, you would need to increase the size of info.iv by one and check whether the second-to-last-element of the array is != NUL.) > > >> >>> >>>>>> things that could be improved: Return size_t in ff_hex_to_data() as >>>>>> that's the proper type (this includes checks in the callers for whether >>>>>> the return type fit into the type of the extradata size)) and making the >>>>>> size of the iv automatically match the needed size of (struct >>>>>> keyinfo).iv. >>> >>> Maybe I think it's better to alloc in ff_hex_to_data function and return the >>> buffer directly. >>> >> >> This has several drawbacks: The user might know an upper bound of the >> buffer size in advance, so that an allocation is unnecessary; the user >> might not want a huge buffer at all (this happens with the rtp users: >> they should reject lengths that don't fit into an int (anything that >> comes close to that bound is probably bullshit anyway)); or the user has >> special needs (this also happens with rtp: they need it as extradata, >> i.e. it needs to be padded). >> >>>>>> >>>>>>> >>>>>>> /** >>>>>>> * Add packet to an AVFormatContext's packet_buffer list, determining >>>>>>> its >>>>>>> diff --git a/libavformat/rtpdec_latm.c b/libavformat/rtpdec_latm.c >>>>>>> index 104a00a..cf1d581 100644 >>>>>>> --- a/libavformat/rtpdec_latm.c >>>>>>> +++ b/libavformat/rtpdec_latm.c >>>>>>> @@ -91,7 +91,7 @@ static int latm_parse_packet(AVFormatContext *ctx, >>>>>>> PayloadContext *data, >>>>>>> >>>>>>> static int parse_fmtp_config(AVStream *st, const char *value) >>>>>>> { >>>>>>> - int len = ff_hex_to_data(NULL, value), i, ret = 0; >>>>>>> + int len = ff_hex_to_data(NULL, 0, value), i, ret = 0; >>>>>>> GetBitContext gb; >>>>>>> uint8_t *config; >>>>>>> int audio_mux_version, same_time_framing, num_programs, num_layers; >>>>>>> @@ -100,7 +100,9 @@ static int parse_fmtp_config(AVStream *st, const >>>>>>> char *value) >>>>>>> config = av_mallocz(len + AV_INPUT_BUFFER_PADDING_SIZE); >>>>>>> if (!config) >>>>>>> return AVERROR(ENOMEM); >>>>>>> - ff_hex_to_data(config, value); >>>>>>> + ret = ff_hex_to_data(config, len, value); >>>>>>> + if (ret < 0) >>>>>>> + return ret; >>>>>>> init_get_bits(&gb, config, len*8); >>>>>>> audio_mux_version = get_bits(&gb, 1); >>>>>>> same_time_framing = get_bits(&gb, 1); >>>>>>> diff --git a/libavformat/rtpdec_mpeg4.c b/libavformat/rtpdec_mpeg4.c >>>>>>> index 34c7950..27751df 100644 >>>>>>> --- a/libavformat/rtpdec_mpeg4.c >>>>>>> +++ b/libavformat/rtpdec_mpeg4.c >>>>>>> @@ -112,11 +112,13 @@ static void close_context(PayloadContext *data) >>>>>>> static int parse_fmtp_config(AVCodecParameters *par, const char *value) >>>>>>> { >>>>>>> /* decode the hexa encoded parameter */ >>>>>>> - int len = ff_hex_to_data(NULL, value), ret; >>>>>>> + int len = ff_hex_to_data(NULL, 0, value), ret; >>>>>>> >>>>>>> if ((ret = ff_alloc_extradata(par, len)) < 0) >>>>>>> return ret; >>>>>>> - ff_hex_to_data(par->extradata, value); >>>>>>> + ret = ff_hex_to_data(par->extradata, par->extradata_size, value); >>>>>>> + if (ret < 0) >>>>>>> + return ret;> return 0; >>>>>>> } >>>>>>> >>>>>>> diff --git a/libavformat/utils.c b/libavformat/utils.c >>>>>>> index 9228313..dfe9f4c 100644 >>>>>>> --- a/libavformat/utils.c >>>>>>> +++ b/libavformat/utils.c >>>>>>> @@ -4768,7 +4768,7 @@ char *ff_data_to_hex(char *buff, const uint8_t >>>>>>> *src, int s, int lowercase) >>>>>>> return buff; >>>>>>> } >>>>>>> >>>>>>> -int ff_hex_to_data(uint8_t *data, const char *p) >>>>>>> +int ff_hex_to_data(uint8_t *data, int data_size, const char *p) >>>>>>> { >>>>>>> int c, len, v; >>>>>>> >>>>>>> @@ -4787,8 +4787,11 @@ int ff_hex_to_data(uint8_t *data, const char *p) >>>>>>> break; >>>>>>> v = (v << 4) | c; >>>>>>> if (v & 0x100) { >>>>>>> - if (data) >>>>>>> + if (data) { >>>>>>> + if (len >= data_size) >>>>>>> + return AVERROR(ERANGE); >>>>>>> data[len] = v; >>>>>>> + } >>>>>>> len++; >>>>>>> v = 1; >>>>>>> } >>>>>>> >>>>>> >> _______________________________________________ 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".