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>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Pavel Koshevoy <pkoshevoy@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The filter needs input frames with color properties filled out by
the decoder. Since this is not always possible, add input options to
the filter so that user may override color space, color primaries,
transfer characteristics, and color range, as well as a generic option
to set all properties at once.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
