Without this patch a user a bit absent-minded may not notice that
the connection doesn't work because the port is missing.
Signed-off-by: Martin Storsjö <martin@martin.st>
Without this patch a user a bit absent-minded may not notice that
the connection doesn't work because the port is missing.
Signed-off-by: Martin Storsjö <martin@martin.st>
commit 20e88d8618
Fix avui stream-copy.
The native decoder and MPlayer's binary decoder only need the
APRG atom, QuickTime at least requires also the ARES atom and
four additional 0 bytes padding at the end of stsd.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The native decoder and MPlayer's binary decoder only need the
APRG atom, QuickTime at least requires also the ARES atom and
four additional 0 bytes padding at the end of stsd.
The old avcodec_find_best_pix_fmt() was insufficient due to 64 pix_fmt limit.
In ffmpeg this problem has been solved long ago through avcodec_find_best_pix_fmt2()
Today libav has added a incompatible modified version of avcodec_find_best_pix_fmt2()
under the same name, thus again breaking ABI/API ...
The avcodec_find_best_pix_fmt_of_list() added in this commit here makes the libav
API available to ffmpeg users too.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Attached patch (together with demuxing patch) allows WMP/msacm G723.1 codec decode files encoded by FFmpeg.
Tested with both 6400 and 5333 mode
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
OpenJPEG doesn't have a max lowres limit, so don't enforce an arbitrary one.
Signed-off-by: Michael Bradshaw <mbradshaw@sorensonmedia.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Add credit/copyright to librtmp authors for parts of the RTMPE code
rtmp: Move the CONFIG_ condition into the if conditions
aac: Mention abbreviation as well in long_name
build: Skip compiling rtmpdh.h if ffrtmpcrypt protocol is not enabled
doc: Add Git configuration section
configure: Add a dependency on https for rtmpts
rtp: Only choose static payload types if the sample rate and channels are right
Conflicts:
doc/git-howto.texi
libavformat/rtmpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Attached patch fixes remuxing of G723.1 in wav, so the output is playable by WMP.
(It's still not enough for encoding - probably some extradata should be added to the output file
to make it playable by WMP/win codec)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This makes sure these calls are removed by dead code elimination
even if optimization is disabled. This fixes building without
crypto libraries without optimization.
Signed-off-by: Martin Storsjö <martin@martin.st>
Make it possible to use the run bisect sub command. As with all
other ffbisect commands, revisions that do not contain the needed
tools are skipped.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
In file libavcodec/x86/dsputil_mmx.c, function ff_put_pixels_clamped_mmx(), there are two assembly code blocks. In the first block (in the unrolled loop), the instructions "movq 8%3, %%mm1 \n\t" etc have problem.
For above instruction, it is clear what the programmer wants: a load from p + 8. But this assembly code doesn’t guarantee that. It only works if the compiler puts p in a register to produce an instruction like this: “movq 8(%edi), %mm1”. During compiler optimization, it is possible that the compiler will be able to constant propagate into p. Suppose p = &x[10000]. Then operand 3 can become 10000(%edi), where %edi holds &x. And the instruction becomes “movq 810000(%edx)”. That is, it will stride by 810000 instead of 8.
This will cause the segmentation fault.
This error was fixed in the second block of the assembly code, but not in the unrolled loop.
How to reproduce:
This error is exposed when we build the ffmpeg using Intel C++ Compiler, IPO+PGO optimization. The ffmpeg was crashed when decoding a mjpeg video.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The rtmpts protocol uses https implicitly, via the ffrtmphttp
protocol, but the ffrtmphttp protocol is also useable for plain
rtmpt without https, so the dependency needs to be added here instead.
Signed-off-by: Martin Storsjö <martin@martin.st>
The rtmpts protocol uses https implicitly, via the ffrtmphttp
protocol, but the ffrtmphttp protocol is also useable for plain
rtmpt without https, so the dependency needs to be added here instead.
Signed-off-by: Martin Storsjö <martin@martin.st>
If using a different sample rate or number of channels, use a dynamic
payload type instead, where the parameters are passed in the SDP.
G722 is a special case where the normal rules don't apply.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
v410dec: Implement explode mode support
zerocodec: fix direct rendering.
wav: init st to NULL to avoid a false-positive warning.
wavpack: set bits_per_raw_sample for S32 samples to properly identify 24-bit
h264: refactor NAL decode loop
RTMPTE protocol support
RTMPE protocol support
rtmp: Add ff_rtmp_calc_digest_pos()
rtmp: Rename rtmp_calc_digest to ff_rtmp_calc_digest and make it global
swscale: add missing HAVE_INLINE_ASM check.
lavfi: place x86 inline assembly under HAVE_INLINE_ASM.
vc1: Add a test for interlaced field pictures
swscale: Mark all init functions as av_cold
swscale: x86: Drop pointless _mmx suffix from filenames
lavf: use conditional notation for default codec in muxer declarations.
swscale: place inline assembly bilinear scaler under HAVE_INLINE_ASM.
dsputil: ppc: cosmetics: pretty-print
dsputil: x86: add SHUFFLE_MASK_W macro
configure: respect CC_O setting in check_cc
Conflicts:
Changelog
configure
libavcodec/v410dec.c
libavcodec/zerocodec.c
libavformat/asfenc.c
libavformat/version.h
libswscale/utils.c
libswscale/x86/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
