Do not build from scratch the section header for each section, but build
using the previous level buffer, thus improving efficiency and fix some
few corner cases which are exposed by the pending disposition patch.
Regular section fields nested in a regular section are now prefixed by
the nested section name.
This is required by the pending change related to disposition.
Regular section fields nested in a regular section are now prefixed by
the nested section name.
This is required by the pending change related to disposition.
This prevents decoding happening on a half initialized context.
Fixes CVE-2012-2779
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Our decoder does not support changing w/h.
Fixes CVE-2012-2777 and CVE-2012-2784.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This practice is not supported by the MinGW developers, and even requires
patching the MinGW runtimes in newer versions. Furthermore, we now support
build with MSVC, so this section is rendered useless.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This practice is not supported by the MinGW developers, and even requires
patching the MinGW runtimes in newer versions. Furthermore, we now support
build with MSVC, so this section is rendered useless.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
It is not correct in all cases and it is less predictable than a skip of 0
for user applications.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The value should be always 3, as it follows from the specification.
Fix a stack buffer overflow in exponents_from_scale_factors as reported
by asan. Thanks to Dale Curtis for the sample vector.
* qatar/master:
libopus: Remap channels using libopus' internal remapping.
Opus decoder using libopus
avcodec: document the use of AVCodecContext.delay for audio decoding
vc1dec: add flush function for WMV9 and VC-1 decoders
http: Increase buffer sizes to cope with longer URIs
nutenc: const correctness for ff_put_v_trace/put_s_trace function arguments
h264_refs: Fix debug tprintf argument types
golomb: const correctness for get_ue()/get_se() function arguments
get_bits: const correctness for get_bits_trace()/get_xbits_trace() arguments
Conflicts:
Changelog
libavcodec/Makefile
libavcodec/version.h
libavformat/http.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This way we can directly remap channels from Opus' channel order to
libav's internal channel order, instead of mapping channels from
Opus' order to Vorbis' order then to libav's order.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
* qatar/master:
dwt: Drop unused functions spatial_compose{53|97}i()
nutdec: Remove unused and broken debug function stub
avcodec: Drop long-deprecated imgconvert.h header
Add Opus support to the Ogg muxer.
Add Opus codec id and codec description.
avformat: Identify anonymous AVIO typedef structs.
Conflicts:
libavcodec/avcodec.h
libavcodec/codec_desc.c
libavcodec/imgconvert.h
libavcodec/version.h
libavformat/oggenc.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
I was sadly unable to find a non fuzzed mp3 that uses the
feature that contained the bug (and i searched hard ...), thus
while this fixes the security issue. It may or may not fix
mixed blocks in 8khz mp3s, i cant say due to lack of samples to test.
Security issue exists since: b37d945dd4
Reported-by: Dale Curtis <dalecurtis@google.com>
(Probably) Found-by: inferno@chromium.org
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Use the MAX_URL_SIZE define where applicable. Increase buffer
sizes for all buffers that need to fit a long pathname - buffers
that need to fit only the hostname (and other short strings, but
not the pathname - such as "headers" in http_connect) are kept
at 1024 bytes for now.
Also increase the max line length in http_read_header, since it
might need to contain a full url for Location: redirects.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes some DNXHD files generated by AVID TM, where codec UL was set to A-law
meanwhile the real audio codec was PCM S16. According to SMPTE RP 224, A-law is
the default value for sound essence parameters therefore we should handle it
specially.
Signed-off-by: Marton Balint <cus@passwd.hu>
Reviewed-by: Tomas Härdin <tomas.hardin@codemill.se>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Stefano Sabatini
Michael Niedermayer
Anton Khirnov
Derek Buitenhuis
Luca Barbato
jamal
Nathan Caldwell
Nicolas George
Kostya Shishkov
Duncan Salerno
Marton Balint