[FFmpeg-devel] [PATCH] libavformat/svs.c: Numeric Truncation in svs.c:57. Added a checker for valid sample_rate value.
From: headshog --- libavformat/svs.c | 7 ++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/libavformat/svs.c b/libavformat/svs.c index b91d29f5a6..bdfb856184 100644 --- a/libavformat/svs.c +++ b/libavformat/svs.c @@ -42,6 +42,7 @@ static int svs_read_header(AVFormatContext *s) { AVStream *st; uint32_t pitch; +int64_t rescale_val; st = avformat_new_stream(s, NULL); if (!st) @@ -51,10 +52,14 @@ static int svs_read_header(AVFormatContext *s) pitch = avio_rl32(s->pb); avio_skip(s->pb, 12); +rescale_val = av_rescale_rnd(pitch, 48000, 4096, AV_ROUND_INF); +if (rescale_val > INT_MAX) +return AVERROR(ERANGE); + st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; st->codecpar->codec_id = AV_CODEC_ID_ADPCM_PSX; st->codecpar->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO; -st->codecpar->sample_rate= av_rescale_rnd(pitch, 48000, 4096, AV_ROUND_INF); +st->codecpar->sample_rate= rescale_val; st->codecpar->block_align= 32; st->start_time = 0; if (s->pb->seekable & AVIO_SEEKABLE_NORMAL) -- 2.25.1 ___ 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".
[FFmpeg-devel] [PATCH] libavformat/svs.c Fixed, now it is in the right place.
From: headshog --- libavformat/svs.c | 7 ++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/libavformat/svs.c b/libavformat/svs.c index b91d29f5a6..bdfb856184 100644 --- a/libavformat/svs.c +++ b/libavformat/svs.c @@ -42,6 +42,7 @@ static int svs_read_header(AVFormatContext *s) { AVStream *st; uint32_t pitch; +int64_t rescale_val; st = avformat_new_stream(s, NULL); if (!st) @@ -51,10 +52,14 @@ static int svs_read_header(AVFormatContext *s) pitch = avio_rl32(s->pb); avio_skip(s->pb, 12); +rescale_val = av_rescale_rnd(pitch, 48000, 4096, AV_ROUND_INF); +if (rescale_val > INT_MAX) +return AVERROR(ERANGE); + st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; st->codecpar->codec_id = AV_CODEC_ID_ADPCM_PSX; st->codecpar->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO; -st->codecpar->sample_rate= av_rescale_rnd(pitch, 48000, 4096, AV_ROUND_INF); +st->codecpar->sample_rate= rescale_val; st->codecpar->block_align= 32; st->start_time = 0; if (s->pb->seekable & AVIO_SEEKABLE_NORMAL) -- 2.25.1 ___ 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".
[FFmpeg-devel] [PATCH] libavformat\svs.c: Fixed, not it is in the right place
From: headshog --- 0001-Fixed-not-it-is-in-the-right-place.patch | 89 +++ 0001-Numeric-truncation-in-svs.c-57.patch | 41 + libavformat/svs.c | 7 +- 3 files changed, 136 insertions(+), 1 deletion(-) create mode 100644 0001-Fixed-not-it-is-in-the-right-place.patch create mode 100644 0001-Numeric-truncation-in-svs.c-57.patch diff --git a/0001-Fixed-not-it-is-in-the-right-place.patch b/0001-Fixed-not-it-is-in-the-right-place.patch new file mode 100644 index 00..8373c56851 --- /dev/null +++ b/0001-Fixed-not-it-is-in-the-right-place.patch @@ -0,0 +1,89 @@ +From ea89215a6b5627f73c370381f07a87f1ccb146ea Mon Sep 17 00:00:00 2001 +From: headshog +Date: Mon, 25 Sep 2023 14:38:04 +0300 +Subject: [PATCH] Fixed, not it is in the right place. + +--- + 0001-Numeric-truncation-in-svs.c-57.patch | 41 +++ + libavformat/svs.c | 7 +++- + 2 files changed, 47 insertions(+), 1 deletion(-) + create mode 100644 0001-Numeric-truncation-in-svs.c-57.patch + +diff --git a/0001-Numeric-truncation-in-svs.c-57.patch b/0001-Numeric-truncation-in-svs.c-57.patch +new file mode 100644 +index 00..dbfdf07e6f +--- /dev/null b/0001-Numeric-truncation-in-svs.c-57.patch +@@ -0,0 +1,41 @@ ++From a9b235e392024f1877d5c43fc0129e9eebb6aea2 Mon Sep 17 00:00:00 2001 ++From: headshog ++Date: Mon, 25 Sep 2023 12:48:22 +0300 ++Subject: [PATCH] Numeric truncation in svs.c:57 ++ ++Hi! We've been fuzzing `ffmpeg` with [sydr-fuzz](https://github.com/ispras/oss-sydr-fuzz) security predicates and we found numeric truncation error in `svs.c:57`. ++In function `svs_read_header` on line 57 field `st->codecpar->sample_rate` has type `int`, the type of return value in `av_rescale_rnd` function is `int64_t`, so the numeric truncation may occur here. ++Then value of `st->codecpar->sample_rate` is passed to `avpriv_set_pts_info` function parameter `unsgined int pts_den`. ++In a way not to break API/ABI, I've added a checker for valid `sample_rate` value. ++--- ++ libavformat/svs.c | 7 ++- ++ 1 file changed, 6 insertions(+), 1 deletion(-) ++ ++diff --git a/libavformat/svs.c b/libavformat/svs.c ++index b91d29f5a6..54f24f539c 100644 ++--- a/libavformat/svs.c + b/libavformat/svs.c ++@@ -42,6 +42,11 @@ static int svs_read_header(AVFormatContext *s) ++ { ++ AVStream *st; ++ uint32_t pitch; +++int64_t rescale_val; +++ +++rescale_val = av_rescale_rnd(pitch, 48000, 4096, AV_ROUND_INF); +++if (rescale_val > INT_MAX) +++return AVERROR(ERANGE); ++ ++ st = avformat_new_stream(s, NULL); ++ if (!st) ++@@ -54,7 +59,7 @@ static int svs_read_header(AVFormatContext *s) ++ st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; ++ st->codecpar->codec_id = AV_CODEC_ID_ADPCM_PSX; ++ st->codecpar->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO; ++-st->codecpar->sample_rate= av_rescale_rnd(pitch, 48000, 4096, AV_ROUND_INF); +++st->codecpar->sample_rate= rescale_val; ++ st->codecpar->block_align= 32; ++ st->start_time = 0; ++ if (s->pb->seekable & AVIO_SEEKABLE_NORMAL) ++-- ++2.25.1 ++ +diff --git a/libavformat/svs.c b/libavformat/svs.c +index b91d29f5a6..bdfb856184 100644 +--- a/libavformat/svs.c b/libavformat/svs.c +@@ -42,6 +42,7 @@ static int svs_read_header(AVFormatContext *s) + { + AVStream *st; + uint32_t pitch; ++int64_t rescale_val; + + st = avformat_new_stream(s, NULL); + if (!st) +@@ -51,10 +52,14 @@ static int svs_read_header(AVFormatContext *s) + pitch = avio_rl32(s->pb); + avio_skip(s->pb, 12); + ++rescale_val = av_rescale_rnd(pitch, 48000, 4096, AV_ROUND_INF); ++if (rescale_val > INT_MAX) ++return AVERROR(ERANGE); ++ + st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; + st->codecpar->codec_id = AV_CODEC_ID_ADPCM_PSX; + st->codecpar->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO; +-st->codecpar->sample_rate= av_rescale_rnd(pitch, 48000, 4096, AV_ROUND_INF); ++st->codecpar->sample_rate= rescale_val; + st->codecpar->block_align= 32; + st->start_time = 0; + if (s->pb->seekable & AVIO_SEEKABLE_NORMAL) +-- +2.25.1 + diff --git a/0001-Numeric-truncation-in-svs.c-57.patch b/0001-Numeric-truncation-in-svs.c-57.patch new file mode 100644 index 00..dbfdf07e6f --- /dev/null +++ b/0001-Numeric-truncation-in-svs.c-57.patch @@ -0,0 +1,41 @@ +From a9b235e392024f1877d5c43fc0129e9eebb6aea2 Mon Sep 17 00:00:00 2001 +From: headshog +Date: Mon, 25 Sep 2023 12:48:22 +0300 +Subject: [PATCH] Numeric truncation in svs.c:57 + +Hi! We've been fuzzing `ffmpeg` with [sydr-fuzz](https://github.com/ispras/oss-sydr-fuzz) security predicates and we found numeric truncation error in `svs.c:57`. +In function `svs_read_header` on line 57 field `st->codecpar->sample_rate` has type `int`, the type of return value in
[FFmpeg-devel] [PATCH] Numeric truncation in svs.c:57
From: headshog Hi! We've been fuzzing `ffmpeg` with [sydr-fuzz](https://github.com/ispras/oss-sydr-fuzz) security predicates and we found numeric truncation error in `svs.c:57`. In function `svs_read_header` on line 57 field `st->codecpar->sample_rate` has type `int`, the type of return value in `av_rescale_rnd` function is `int64_t`, so the numeric truncation may occur here. Then value of `st->codecpar->sample_rate` is passed to `avpriv_set_pts_info` function parameter `unsgined int pts_den`. In a way not to break API/ABI, I've added a checker for valid `sample_rate` value. --- libavformat/svs.c | 7 ++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/libavformat/svs.c b/libavformat/svs.c index b91d29f5a6..54f24f539c 100644 --- a/libavformat/svs.c +++ b/libavformat/svs.c @@ -42,6 +42,11 @@ static int svs_read_header(AVFormatContext *s) { AVStream *st; uint32_t pitch; +int64_t rescale_val; + +rescale_val = av_rescale_rnd(pitch, 48000, 4096, AV_ROUND_INF); +if (rescale_val > INT_MAX) +return AVERROR(ERANGE); st = avformat_new_stream(s, NULL); if (!st) @@ -54,7 +59,7 @@ static int svs_read_header(AVFormatContext *s) st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; st->codecpar->codec_id = AV_CODEC_ID_ADPCM_PSX; st->codecpar->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO; -st->codecpar->sample_rate= av_rescale_rnd(pitch, 48000, 4096, AV_ROUND_INF); +st->codecpar->sample_rate= rescale_val; st->codecpar->block_align= 32; st->start_time = 0; if (s->pb->seekable & AVIO_SEEKABLE_NORMAL) -- 2.25.1 ___ 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".
Re: [FFmpeg-devel] [PATCH] Hi! We've been fuzzing `ffmpeg` with [sydr-fuzz](https://github.com/ispras/oss-sydr-fuzz) security predicates and we found numeric truncation error in `svs.c:57`.
On 2023-09-20 15:29, Paul B Mahol wrote: Unacceptable code changes as that Breaks ABI/API. ___ 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". Maybe then a checker for valid sample_rate value should be added to svs_read_header function? ___ 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".
[FFmpeg-devel] [PATCH] Hi! We've been fuzzing `ffmpeg` with [sydr-fuzz](https://github.com/ispras/oss-sydr-fuzz) security predicates and we found numeric truncation error in `svs.c:57`.
From: headshog In function `svs_read_header` on line 57 field `st->codecpar->sample_rate` has type `int`, the type of return value in `av_rescale_rnd` function is `uint64_t`, so the numeric truncation may occur here. Then value of `st->codecpar->sample_rate` is passed to `avpriv_set_pts_info` function parameter `unsgined int pts_den`. In this function `pts_den` is used only in passing its value to parameter `int64_t den` in function `av_reduce`. So we suggest to change the type of field `sample_rate` to `int64_t` and to change the type of `pts_den` to `uint64_t` in `avpriv_set_pts_info` function. The other way to solve this is to add a checker for `sample_rate` valid value. - OS: ubuntu 20.04 - commit: f225f8d7464569c7b917015c26ad30a37a5fbbe2 ``` libavformat/svs.c:57:36: runtime error: implicit conversion from type 'int64_t' (aka 'long') of value 6321554672 (64-bit, signed) to type 'int' changed the value to 2026587376 (32-bit, signed) SUMMARY: UndefinedBehaviorSanitizer: undefined-behavior libavformat/svs.c:57:36 ``` --- libavcodec/codec_par.h | 2 +- libavformat/internal.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/libavcodec/codec_par.h b/libavcodec/codec_par.h index add90fdb1e..2bbb7ba4e7 100644 --- a/libavcodec/codec_par.h +++ b/libavcodec/codec_par.h @@ -175,7 +175,7 @@ typedef struct AVCodecParameters { /** * Audio only. The number of audio samples per second. */ -int sample_rate; +int64_t sample_rate; /** * Audio only. The number of bytes per coded audio frame, required by some * formats. diff --git a/libavformat/internal.h b/libavformat/internal.h index 901a8b51c6..6e1fce41aa 100644 --- a/libavformat/internal.h +++ b/libavformat/internal.h @@ -584,7 +584,7 @@ const struct AVCodec *ff_find_decoder(AVFormatContext *s, const AVStream *st, * @param pts_den time base denominator */ void avpriv_set_pts_info(AVStream *st, int pts_wrap_bits, - unsigned int pts_num, unsigned int pts_den); + unsigned int pts_num, uint64_t pts_den); /** * Set the timebase for each stream from the corresponding codec timebase and -- 2.25.1 ___ 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".