A lot of changes happen at the same time:
- Add a framequeue fifo to AVFilterLink.
- split AVFilterLink.status into status_in and status_out: requires
changes to the few filters and programs that use it directly
(f_interleave, split, filtfmts).
- Add a field ready to AVFilterContext, marking when the filter is ready
and its activation priority.
- Add flags to mark blocked links.
- Change ff_filter_frame() to enqueue the frame.
- Change all filtering functions to update the ready field and the
blocked flags.
- Update ff_filter_graph_run_once() to use the ready field.
- buffersrc: always push the frame immediately.
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>
All copyright holders have agreed to the relicensing.
Approved-by: Andreas Cadhalpun <andreas.cadhalpun@googlemail.com>
Approved-by: David Sedacca <sedacca@comcast.net>
Approved-by: Ganesh Ajjanagadde <gajjanag@mit.edu>
Approved-by: Jean First <jeanfirst@gmail.com>
Approved-by: Kyle Swanson <k@ylo.ph>
Approved-by: Michael Niedermayer <michael@niedermayer.cc>
Approved-by: Nicolas George <george@nsup.org>
Approved-by: Paul B Mahol <onemda@gmail.com>
Approved-by: Thilo Borgmann <thilo.borgmann@mail.de>
Expands the parser to also accept the separator '-' in addition to
'+', and take the negative sign into consideration.
The optional sign for the first factor in the expression is already
covered by parsing for an integer.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'c3f113d58488df7594a489bdbb993a69ad47063c':
vf_hwdownload: allocate the destination frame for the pool size
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
AVFilterLink.frame_count is supposed to count the number of frames
that were passed on the link, but with min_samples, that number is
not always the same for the source and destination filters.
With the addition of a FIFO on the link, the difference will become
more significant.
Split the variable in two: frame_count_in counts the number of
frames that entered the link, frame_count_out counts the number
of frames that were sent to the destination filter.
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>
2-channels convolution using complex fft
improves speed significantly
not sure if it should be enabled by default
so disable it by default
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
Thanks to Mathieu Malaterre <malat@debian.org> for reporting the
Que/Queue typo. (https://bugs.debian.org/839542)
Reviewed-by: Lou Logan <lou@lrcd.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
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>
The parser for the outdef will accept a negative value for the first
named channel's gain. As negative values effectively only invert the
phase of the signal, and not negate the level, the gains' absolute
values must be used to correctly accumulate the levels.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '5b63b15663d31f50ce45d980b904a68795ad3f7a':
lavfi: set the link hwframes context before configuring the dst input
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
As all known valid HDCD sample formats and sample rates are now handled
by the filter, remove the scan that "invades the privacy" of the filter graph
and turn off autoconvert by default as requested by Nicolas George.
http://ffmpeg.org/pipermail/ffmpeg-devel/2016-August/197571.html
Signed-off-by: Burt P <pburt0@gmail.com>
I don't have any legitimate 20 or 24-bit HDCD to test. It is known
that the PM Model Two would insert packets into 20 and 24-bit output,
but I have no idea what differences in behavior existed when decoding
20 or 24-bit. For now, as with 16-bit, PE (if enabled) will expand the
top 3dB into 9dB and LLE (gain adjust) will be applied if signaled.
Signed-off-by: Burt P <pburt0@gmail.com>
New versions of hdcd_scan() and hdcd_integrate() that also do the
work of hdcd_scan_stereo() and hdcd_integrate_stereo().
Some code split into previously separate functions to remove
duplication is now merged back into each function in the single
place where it is used.
Signed-off-by: Burt P <pburt0@gmail.com>
The buffer is already being copied anyway, so interlace the planar
format during the copy and remove one use of auto-convert.
Signed-off-by: Burt P <pburt0@gmail.com>
The PM Model Two could output HDCD-encoded audio in CD and all
DVD-Audio sample rates. (44100, 48000, 88200, 96000, 176400, and
192000 Hz)
Signed-off-by: Burt P <pburt0@gmail.com>
This is the assumption that is made in pixel format conversion do
throughout the code (in particular swscale), and BT-specifications
mandate.
Add a warning to inform the user that an automatic selection is being
made.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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>
Allows to use values returned from API and from ffprobe directly.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
