Previously alac encoder was used, from a first glance I thought it is bitexact,
but it turns out it is using floating point arithmetic as well, so probably it
is not. Fixes fate failures on mingw32/64.
Signed-off-by: Marton Balint <cus@passwd.hu>
Otherwise the frame size of the codec is not set in the buffersink.
Fixes ticket #6603 and the following simpler case:
ffmpeg -c aac -filter_complex "sine=d=0.1,asetnsamples=1025" out.aac
Signed-off-by: Marton Balint <cus@passwd.hu>
Since af1761f7b5 ffmpeg waits for a frame in each
stream before writing the output header. If we are using threaded decoding for
attached pictures, we have to read till EOF to be able to finally flush the
decoder and output the decoded frame. This essentially makes ffmpeg buffer all
non-attached picture packets, which will cause a "Too many packets buffered for
output stream" eventually.
By forcing single threaded decoding, we get a frame from a single packet as
well and we can avoid the error.
Fixes part of ticket #6375:
ffmpeg -i 46564100.mp3 -acodec libmp3lame -ab 128k -ac 2 out.mp3
Reviewed-by: Hendrik Leppkes <h.leppkes@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
When the background had an alpha channel, the old code in blend_plane
calculated premultiplied alpha from the destination plane colors instead of the
destination alpha.
Also the calculation of the output alpha should only happen after the color
planes are already finished.
Fixes output of:
ffplay -f lavfi "testsrc2=alpha=32[a];color=black[b];[b][a]overlay[out0]"
Signed-off-by: Marton Balint <cus@passwd.hu>
v2:
- use uint16_t instead of int to store 10-bit ancillary data
- fix ancillary line numbers for 1080p
- some comments and clarifications as requested by Aaron Levinson
Signed-off-by: Marton Balint <cus@passwd.hu>
This also add supports for 4K DeckLink cards because they always output the
ancillary data in 10-bit.
v2:
- only try teletext decoding for 576i PAL mode
- some comments as requested by Aaron Levinson
Signed-off-by: Marton Balint <cus@passwd.hu>
Buffering more than one packet can be a huge performance improvement for
encoding files with small packets (e.g. wav) over SMB/CIFS.
Acked-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Marton Balint <cus@passwd.hu>
If flushing is not disabled, then mux.c will signal the end of the packets with
an AVIO_DATA_MARKER_FLUSH_POINT, and aviobuf will be able to decide to flush or
not based on the preferred minimum packet size set by the used protocol.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Marton Balint <cus@passwd.hu>
This patch makes aviobuf work more like traditinal file IO, which is how people
think about it.
For example, in the past, aviobuf only flushed buffers until the current buffer
position, even if more data was written to it previously, and a backward seek
was used to reposition the IO context.
From now, aviobuf will keep track of the written data, so no explicit seek will
be required till the end of the buffer, or till the end of file before flushing.
This fixes at least one regression, fate-vsynth3-flv was broken if
flush_packets option was set to false, an explicit seek was removed in
4e3cc4bdd8.
Also from now on, if a forward seek in the write buffer were to cause a gap
between the already written data and the new file position, a flush will
happen.
The must_flush varable is also removed, which might have caused needless
flushes with multiple seeks whithin the write buffer. Since we know the amount
of data written to it, we will know when to flush.
Signed-off-by: Marton Balint <cus@passwd.hu>
The md5 protocol has no seek support, but some tests use seeks. This changes
the fate tests to actually create the output files and calculate the md5 on the
written files, which also makes the tests independent of the size of the output
buffers and output buffering in general.
A new md5pipe fate test method is also introduced to keep the old functionality
for tests where using a non-seekable output was intentional, and matroska md5
tests are changed to use that.
Signed-off-by: Marton Balint <cus@passwd.hu>
The custom callback can cause significant CPU usage on Windows for some large
files with many index entries for some reason.
v2: Move check after parsing options.
Signed-off-by: Marton Balint <cus@passwd.hu>
Needed for the C+11 atomics. Also change add_cxxflags to check_cxxflags.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
This ensures that the wrapped avframe will not get reallocated later, which
would invalidate internal references such as extended data.
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Detecting a leap second depends on a lot of things, segment time, segment
offset, system leap second implementation, the removed part is a huge
simplification which can be misleading, so it is best to remove it.
Signed-off-by: Marton Balint <cus@passwd.hu>
Not starting a new segment if the elapsed microsecs since the start of the day
equals the the elapsed microsecs since the start of the day at the time of the
last cut seems plain wrong to me, Deti do you remember the original reason
behind this check?
Signed-off-by: Marton Balint <cus@passwd.hu>
Current code returned the number of channels as channel layout in that case,
and if nret is not set then unknown layouts are typically not supported.
Also use the common parsing code. Use a temporary workaround to parse an
unknown channel layout such as '13c', after a 1 year grace period only '13C'
will work.
Signed-off-by: Marton Balint <cus@passwd.hu>
Return a channel layout and the number of channels based on the specified name.
This function is similar to av_get_channel_layout(), but can also parse unknown
channel layout specifications.
Unknown channel layout specifications are a decimal number and a capital 'C'
suffix, in order to not break compatibility with the lowercase 'c' suffix,
which is used for a guessed channel layout with the specified number of
channels.
Signed-off-by: Marton Balint <cus@passwd.hu>
As I used simple RGBA formats for subtitles and for the video texture if
avfilter is disabled I kind of assumed that sws_scale won't access data
pointers and strides above index 0, but apparently that is not the case.
Fixes Coverity CID 1396737, 1396738, 1396739, 1396740.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Marton Balint <cus@passwd.hu>
Since the default in the libav fork is to only allow known layouts, making
unknown layouts allowed by default here can be a security risk for filters
directly merged from libav. However, usually it is simple to detect such cases,
use of av_get_channel_layout_nb_channels is a good indicator, so I suggest we
change this regardless.
See http://ffmpeg.org/pipermail/ffmpeg-devel/2016-November/203204.html.
This patch indirectly adds unknown channel layout support for filters where
query_formats is not specified:
abench
afifo
ainterleave
anullsink
apad
aperms
arealtime
aselect
asendcmd
asetnsamples
asetpts
asettb
ashowinfo
azmq
It introduces a query_formats callback for the asyncts filter, which only
supports known channel layouts since it is using libavresample.
And it removes .query_formats callback from filters where it was only there to
support unknown layouts, as this is now the default:
aloop
ametadata
anull
asidedata
asplit
atrim
Acked-by: Nicolas George <george@nsup.org>
Signed-off-by: Marton Balint <cus@passwd.hu>
Stream timebase should be set using avpriv_set_pts_info, otherwise
avctx->pkt_timebase is not correct, leading to A/V desync.
Signed-off-by: Marton Balint <cus@passwd.hu>
Reviewed-by: Stephen Hutchinson <qyot27@gmail.com>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Also contains the following changes to the library:
- add ff_ prefix to functions
- remove cplusplus defines.
- add FF_ prefix to contants and some structs
- remove true peak calculation feature, since it uses its own resampler, and
af_loudnorm does not need it.
- remove version info and some fprintf(stderr) functions
- convert to use av_malloc
- always use histogram mode for LRA calculation, otherwise LRA data is slowly
consuming memory making af_loudnorm unfit for 24/7 operation. It also uses a
BSD style linked list implementation which is probably not available on all
platforms. So let's just remove the classic mode which not uses histogram.
- add ff_thread_once for calculating static histogram tables
- convert some functions to void which cannot fail
- remove intrinsics and some unused headers
- add support for planar audio
- remove channel / sample rate changer function, in ffmpeg usually we simply
alloc a new context
- convert some static variables to defines
- declare static histogram variables as aligned
- convert some initalizations to mallocz
- add window size parameter to init function and remove window size setter
function
- convert return codes to AVERROR
- fix indentation
Signed-off-by: Marton Balint <cus@passwd.hu>
This is a similar filter to f_metadata, only it works on side data. Since
adding side data from a user provided arbitrary binary string is unsafe,
because current code assumes that a side data of a certain kind has the proper
size, this filter only implements selection and deletion. Also, no value
matching support is implemented yet, because there is no uniform way to specify
a side data textually.
Signed-off-by: Marton Balint <cus@passwd.hu>
This commit also drops SDL1 support for ffplay.
Tested-by: James Almer <jamrial@gmail.com> (Windows, mingw-w64)
Signed-off-by: Josh de Kock <josh@itanimul.li>
Useful when the amerge filter parameters are generated from a script based on
the number of input streams, by allowing 1 input it does not have to be handled
specially.
The split filter also allows 1 output, so it is more consistent to allow
merging 1 input as well.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Marton Balint <cus@passwd.hu>
Use proper ISO 8601 timestamps which also signal that they are in UTC.
This changes the format of creation_time and modification_date metadata values
from 2016-06-01 22:30:00 to 2016-01-01T22:30:00.000000Z
Fixes ticket #5673.
Signed-off-by: Marton Balint <cus@passwd.hu>
Marton Balint
