* qatar/master:
qdm2: remove broken and disabled dump_context() debug function
x86: h264_intrapred: use newly introduced SPLAT* and PSHUFLW macros
x86inc: add SPLATB_LOAD, SPLATB_REG, PSHUFLW macros
x86inc: modify ALIGN to not generate long nops on i586
x86: h264_intrapred: port to cpuflag macros
avplay: update input filter pointer when the filtergraph is reset.
avconv: fix parsing of -force_key_frames option.
h264: use templates to avoid excessive inlining
xtea: Make the count parameter match the documentation
blowfish: Make the count parameter match the documentation
mpegvideo: Don't use ff_mspel_motion() for vc1
xtea: invert branch and loop precedence
blowfish: invert branch and loop precedence
flvdec: optionally trust the metadata
avconv: Set audio filter time base to the sample rate
vp8: Add ifdef guards around the sse2 loopfilter in the sse2slow branch too
Conflicts:
ffmpeg.c
ffplay.c
libavcodec/h264.c
libavcodec/mpegvideo_common.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This avoids creating new AVStreams for them when switching between
different variants of them, since we can handle changes between
different sample rates of nellymoser within the same stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
* 'mxfenc_improvements' of https://github.com/mbouron/FFmpeg:
mxfenc: support dnxhd codec
mxfenc: support smpte dv codec
Merged-by: Michael Niedermayer <michaelni@gmx.at>
After some internal talks it seems the code is similar to what is in FFmbc
by Baptiste Coudurier; Baptiste accepted to relicense the similiar chunks
from GPL to LGPL.
Instead of inlining everything into ff_h264_hl_decode_mb(), use
explicit templating to create versions of the called functions
with constant parameters filled in. This greatly speeds up
compilation of h264.c and reduces the code size without any
measurable impact on performance.
Compilation time for h264.c on an i7 goes from 30s to 5.5s.
Code size is reduced by 430kB.
Signed-off-by: Mans Rullgard <mans@mansr.com>
MP3 fixed and floating point decoders are optimized
for MIPS architecture.
Signed-off-by: Nedeljko Babic <nbabic@mips.com>
Reviewed-by: Vitor Sessak <vitor1001@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* git://github.com/mjbshaw/FFmpeg-OpenJPEG-J2K-Encoder:
libopenjpegdec: add support for decoding YUV420/422/444P12/14
libopenjpegenc: cosmetics: reorder pix_fmts
libopenjpegenc: add support for YUV420/422/444P12/14
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Previously it was interpreted as number of bytes, while the
documentation stated that it was the number of 8 byte blocks.
This makes it behave similarly to the existing AES code.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously it was interpreted as number of bytes, while the
documentation stated that it was the number of 8 byte blocks.
This makes it behave similarly to the existing AES code.
Signed-off-by: Martin Storsjö <martin@martin.st>
Using ff_mspel_motion assumes that s (a MpegEncContext
poiinter) really is a Wmv2Context.
This fixes crashes in error resilience on vc1/wmv3 videos.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
It might be easier to understand for some people and is a bit more
in line with e.g. OpenSSH documentation. The meaning of the text stays
exactly the same.
If the output frame size is smaller than the input sample rate,
and the input stream time base corresponds exactly to the input
frame size (getting input packet timestamps like 0, 1, 2, 3, 4 etc),
the output timestamps from the filter will be like
0, 1, 2, 3, 4, 4, 5 ..., leadning to non-monotone timestamps later.
A concrete example is input mp3 data having frame sizes of 1152
samples, transcoded to aac with 1024 sample frames.
By setting the audio filter time base to the sample rate, we will
get sensible timestamps for all output packets, regardless of
the ratio between the input and output frame sizes.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (29 commits)
lavfi: reclassify showfiltfmts as a TESTPROG
graph2dot: fix printf format specifier
swscale: yuv2planeX 8bit >=sse2 functions need aligned stack on x86-32.
vp8: loopfilter >=sse2 functions need aligned stack on x86-32.
amr: remove shift out of the AMR_BIT() macro.
dsputilenc: group yasm and inline asm function pointer assignment.
mov: use forward declaration of a function instead of a table.
Clarify Doxygen comment for FF_API_* #defines.
configure: simplify get_version()
Create version.h headers for libraries that lack them
gitignore: Use full path instead of relative path to specify patterns
mpegvideo: remove VLAs
Add XTEA encryption support in libavutil
Add Blowfish encryption support in libavutil
eval: Add the isinf() function and tests for it
flacdec: move lpc filter to flacdsp
flacdec: split off channel decorrelation as flacdsp
avplay: Add an option for not limiting the input buffer size
FATE: add a test for WMA cover art.
FATE: add a test for apetag cover art
...
Conflicts:
.gitignore
configure
ffplay.c
libavcodec/Makefile
libavcodec/error_resilience.c
libavcodec/mpegvideo.c
libavcodec/ratecontrol.c
libavdevice/avdevice.h
libavfilter/Makefile
libavfilter/filtfmts.c
libavfilter/version.h
libavformat/mov.c
libavformat/version.h
libavutil/Makefile
libavutil/avutil.h
libavutil/version.h
libswscale/swscale.h
libswscale/x86/swscale_mmx.c
tests/fate/libavutil.mak
tests/lavfi-regression.sh
tools/graph2dot.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Paul B Mahol
Stefano Sabatini
Mans Rullgard
Michael Niedermayer
Martin Storsjö
Diego Biurrun
Loren Merritt
Carl Eugen Hoyos
Matthieu Bouron
Anton Khirnov
Reimar Döffinger
Georg Lippitsch
Nedeljko Babic
Samuel Pitoiset
Alexander Strasser
Luca Barbato