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>
This tool uses lavfi internal symbols not accessible in shared
libraries. TESTPROGS are linked statically to allow them use of
library internals not normally exported.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This commit is based on libav's implementation and
makes sure to compare output timestamps together.
It also reduces the differences with avconv.
The changes to the test reference files are caused
by an additional packet at the end, the timestamp
of the frame encoded by this packet is always
strictly below the limit stated by the -t option.
This code is intended for errors in external libraries
when no corresponding error code can be found.
AVERROR_UNKNOWN is too vague for that and looks like AVERROR_BUG.
For reading from normal files on disk, the queue limits for
demuxed data work fine, but for reading data from realtime
streams, they mean we're not reading from the input stream
at all once the queue limit has been reached. For TCP streams,
this means that writing to the socket from the peer side blocks
(potentially leading to the peer dropping data), and for UDP
streams it means that our kernel might drop data.
For some protocols/servers, the server initially sends a
large burst with data to fill client side buffers, but once
filled, we should keep reading to avoid dropping data.
For all realtime streams, it IMO makes sense to just buffer
as much as we get (rather in buffers in avplay.c than in
OS level buffers). With this option set, the input thread
should always be blocking waiting for more input data,
never sleeping waiting for the decoder to consume data.
Signed-off-by: Martin Storsjö <martin@martin.st>
