Functionality used before didn't widen the values from limited to
full range. Additionally, now the decoder uses BT.709 where it
should be used according to the video resolution.
Default for not yet set colorimetry is BT.709 due to most observed
HDMV content being HD.
BT.709 coefficients were gathered from the first two parts of BT.709
to BT.2020 conversion guide in ARIB STD-B62 (Pt. 1, Chapter 6.2.2).
They were additionally confirmed by manually calculating values.
Fixes#4637
(cherry picked from commit 9779b62624)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes out of array read
Fixes: mozilla bug 1266129
Found-by: Tyson Smith
Tested-by: Tyson Smith
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 9f36ea57ae)
Conflicts:
libavcodec/avpacket.c
Sometimes video fails to decode if H.264 configuration changes mid stream.
The reason is that configuration parser assumes that nal_ref_idc is equal to 11b
while actually some codecs but 01b there. The H.264 spec is somewhat
vague about this but it looks like it allows any non-zero nal_ref_idc for sps/pps.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 3a727606c4)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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>
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>
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)
Conflicts:
libavformat/concatdec.c
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>
Previously errors could result in random entries to be lost.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit f3ace85d88)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Sample rate of 11025 takes 16 bits but previous code would pick only 8.
Fixes assertion failure.
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Signed-off-by: Paul B Mahol <onemda@gmail.com>
(cherry picked from commit 3e7d684912)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
