Currently, if the movie source filter is used and a seek_point is
specified on a file that has a negative start time, ffmpeg will fail.
An easy way to reproduce this is as follows:
$ ffmpeg -vsync passthrough -filter_complex 'color=d=10,setpts=PTS-1/TB' test.mp4
$ ffmpeg -filter_complex 'movie=filename=test.mp4:seek_point=2' -f null -
The problem is caused by checking for int64_t overflow the wrong way.
In general, to check whether a + b overflows, it is not enough to do:
a > INT64_MAX - b
because b might be negative; the correct way is:
b > 0 && > a > INT64_MAX - b
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c1f9734f97)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes segfault
Fixes Ticket5333
Regression since bfc8a4dabe
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 8f2a1990c0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This is ~2x faster for y not an integer on Haswell+GCC, and should
generally be faster due to the fact that anyway powf essentially does
this under the hood. Made an inline function in lavu/internal.h for this
purpose.
Note that there are some accuracy differences, that should generally be
negligible. In particular, FATE still passes on this platform.
Results in ~ 7% speedup in aac encoding with -march=native, Haswell+GCC.
before:
ffmpeg -i sin.flac -acodec aac -y sin_new.aac 6.05s user 0.06s system 104% cpu 5.821 total
after:
ffmpeg -i sin.flac -acodec aac -y sin_new.aac 5.67s user 0.03s system 105% cpu 5.416 total
This is also faster than an alternative approach that pulls in powf, gets rid of
the crufty NaN checks and other special cases, exploits knowledge about the intervals, etc.
This of course does not exclude smarter approaches; just suggests that
there would need to be significant work on this front of lower utility than
searches for hotspots elsewhere.
Reviewed-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Ganesh Ajjanagadde <gajjanag@gmail.com>
(cherry picked from commit bccc81dfa0)
This ensures gcc does not create unnecessary
loads or stores and possibly even does not vectorize
the negation.
Speeds up mp3 to aac transcoding with default settings
by 10% when using "gcc (Debian 5.3.1-10) 5.3.1 20160224".
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
(cherry picked from commit b60dfae7af)
I cannot see any point whatsoever to use
double here instead of float, the results
are likely identical in all cases..
Using float allows for much more
efficient use of SIMD.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
(cherry picked from commit 0a04c2885f)
It makes no sense whatsoever to do this at each function call; we
already have a table for this.
Yields a 2x improvement in find_min_book (x86-64, Haswell+GCC):
ffmpeg -i sin.flac -acodec aac -y sin.aac
find_min_book
old
605 decicycles in find_min_book, 8388453 runs, 155 skips.9x
606 decicycles in find_min_book,16776912 runs, 304 skips.9x
607 decicycles in find_min_book,33553819 runs, 613 skips.2x
607 decicycles in find_min_book,67107668 runs, 1196 skips.3x
607 decicycles in find_min_book,134215360 runs, 2368 skips3x
new
359 decicycles in find_min_book, 8388552 runs, 56 skips.3x
360 decicycles in find_min_book,16777112 runs, 104 skips.1x
361 decicycles in find_min_book,33554218 runs, 214 skips.4x
361 decicycles in find_min_book,67108381 runs, 483 skips.5x
361 decicycles in find_min_book,134216725 runs, 1003 skips5x
and more importantly a non-negligible speedup (~ 8%) to overall AAC encoding:
old:
ffmpeg -i sin.flac -acodec aac -strict -2 -y sin_new.aac 6.82s user 0.03s system 104% cpu 6.565 total
new:
ffmpeg -i sin.flac -acodec aac -strict -2 -y sin_old.aac 6.24s user 0.03s system 104% cpu 5.993 total
This also improves accuracy of the expression by ~ 2 ulp in some cases.
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Signed-off-by: Ganesh Ajjanagadde <gajjanag@gmail.com>
(cherry picked from commit bd9c58756a)
Reviewed-by: maintainer
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 0cd9ff4e3a)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Original mail and my own followup on ffmpeg-user earlier today:
I have a device sending out a MJPEG/RTP stream on a low quality setting.
Decoding and displaying the video with libavformat results in a washed
out, low contrast, greyish image. Playing the same stream with VLC results
in proper color representation.
Screenshots for comparison:
http://zevv.nl/div/libav/shot-ffplay.jpghttp://zevv.nl/div/libav/shot-vlc.jpg
A pcap capture of a few seconds of video and SDP file for playing the
stream are available at
http://zevv.nl/div/libav/mjpeg.pcaphttp://zevv.nl/div/libav/mjpeg.sdp
I believe the problem might be in the calculation of the quantization
tables in the function create_default_qtables(), the attached patch
solves the issue for me.
The problem is that the argument 'q' is of the type uint8_t. According to the
JPEG standard, if 1 <= q <= 50, the scale factor 'S' should be 5000 / Q.
Because the create_default_qtables() reuses the variable 'q' to store the
result of this calculation, for small values of q < 19, q wil subsequently
overflow and give wrong results in the calculated quantization tables. The
patch below uses a new variable 'S' (same name as in RFC2435) with the proper
range to store the result of the division.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit e3e6a2cff4)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Simplifies code
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 50d017a281)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes Ticket5287
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit efa98cdc2f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes Ticket5244
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 068026b0f7)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit fbec157ea0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes Ticket5345
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 50ef7361cb)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Rename luma table to delta table and change how it is used.
CC: libav-stable@libav.org
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
(cherry picked from commit f8c34f4b8d)
(cherry picked from commit 73f3c8f73e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit b2ab3398f5)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This should theoretically improve the randomness slightly
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2540d884f3)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Trying to make heads and tails out of DTS 6.1 I can across this typo.
I also noticed that this wiki page is incorrect or misleading, the
channel order for 6.1 given does not match the source code. At the
least it should be clarified that the layout given does not apply to
DTS. https://trac.ffmpeg.org/wiki/AudioChannelManipulation
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 73d1398f0c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This is safer, as a selected demuxer could still mean that it was auto-detected
by a user application
Reviewed-previously-by: Nicolas George <george@nsup.org>
Reviewed-previously-by: Andreas Cadhalpun <andreas.cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 689211d572)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit de1de49324)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
RTCP synchronization packet was broken since commit in ffmpeg version > 2.8.3
(commit: e04b039b15) Since this commit (2e814d0329)
"rtpenc: Simplify code by introducing a macro for rescaling NTP timestamps", NTP_TO_RTP_FORMAT
uses av_rescale_rnd() function to add the data to the packet.
This causes an overflow in the av_rescale_rnd() function and it will return INT64_MIN.
Causing the NTP stamp in the RTCP packet to have an invalid value.
Github: Closes#182
Reverting commit '2e814d0329aded98c811d0502839618f08642685' solves the problem.
(cherry picked from commit 1109ed7973)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The first X96 channel set can have more channels than core, causing X96
decoding to be skipped. Clear the number of decoded X96 channels to zero
in this rudimentary case.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit db44b59980)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c78a726717)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Found-by: jamrial
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 554f6e930c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: ebd58db6-dc86-11e5-91c2-59daeddf50c7.jpg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c6f4720b86)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes part of Ticket5264
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 67e5bd0c50)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes part of Ticket5264
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit df36257a53)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 1ec7a70380)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: BBB
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit f6492a2ea8)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: BBB
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit d07f6e5f1c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 346ec91764)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This zeroes the WebPAnimEncoderOptions.verbose field, silencing library info messages
printed to stderr.
Reviewed-by: James Zern <jzern@google.com>
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 626b6b769c)
I discovered that ffserver streaming was broken (it seems like it has been since 20th November) and I opened a ticket for this (https://trac.ffmpeg.org/ticket/5250 <https://trac.ffmpeg.org/ticket/5250>).
I spent yesterday learning git bisect (with the kind help of cehoyos) to painstakingly track down the cause. This was made more difficult due to the presence of a segfault in ffserver during the period where the bug was introduced so I first had to identify when and how that was fixed and then retrospectively apply that fix again for each step of the second git bisect to find the actual bug.
Anyway, the fruits of my labour are the innocent looking patch below to correct a couple of typos and define a valid range for two variables.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit a2f8beef2d)
