av_ts_make_time_string() used "%.6g" format, but this format was losing
precision even when the timestamp to be printed was not that large. For example
for 3 hours (10800) seconds, only 1 decimal digit was printed, which made this
format inaccurate when it was used in e.g. the silencedetect filter. Other
detection filters printing timestamps had similar issues. Also time base
parameter of the function was *AVRational instead of AVRational.
Resolve these problems by introducing a new function, av_ts_make_time_string2().
We change the used format to "%.*f", use a precision of 6, except when printing
values near 0, in which case we calculate the precision dynamically to aim for
a similar precision in normal form as with %.6g. No longer using scientific
representation can make parsing the timestamp easier for the users, we can
safely do this because the theoretical maximum of INT64_MAX*INT32_MAX still
fits into the string buffer in normal form.
We somewhat imitate %g by trimming ending zeroes and the potential decimal
point characters. In order not to trim "inf" as well, we assume that the
decimal point string does not contain the letter "f". Note that depending on
printf %f implementation, we might trim "infinity" to "inf".
Thanks for Allan Cady for bringing up this issue.
Signed-off-by: Marton Balint <cus@passwd.hu>
Common utility function that can be used by all codecs to select the
right (any valid) film grain parameter set. In particular, this is
useful for AFGS1, which has support for multiple parameters.
However, it also performs parameter validation for H274.
This is needed for AV1 film grain as well, when using AFGS1 streams.
Also add extra width/height and subsampling information, which AFGS1
cares about, as part of the same API bump. (And in principle, H274
should also expose this information, since it is needed downstream to
correctly adjust the chroma grain frequency to the subsampling ratio)
Deprecate the equivalent H274-exclusive fields. To avoid breaking ABI,
add the new fields after the union; but with enough of a paper trail to
hopefully re-order them on the next bump.
This will allow users to pass the Android ApplicationContext which is mandatory
to retrieve the ContentResolver responsible to resolve/open Android content URIS.
av_frame_side_data_get() has a const AVFrameSideData * const *sd
parameter; so calling it with an AVFramesSideData **sd like
AVCodecContext.decoded_side_data (or with a AVFramesSideData * const
*sd) is safe, but the conversion is not performed automatically
in C. All users of this function therefore resort to a cast.
This commit changes this: av_frame_side_data_get() is renamed
to av_frame_side_data_get_c(); furthermore, a static inline
wrapper for it name av_frame_side_data_get() is added
that accepts an AVFramesSideData * const * and converts this
to const AVFramesSideData * const * in a Wcast-qual safe way.
This also allows to remove the casts from the current users.
Reviewed-by: Jan Ekström <jeebjp@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The documentation correctly states that the rdiv is a multiplier but incorrectly states the default behavior is to multiply by the sum of all matrix elements - it multiplies by 1/sum.
This changes the documentation to match the code.
Address trac #10889
Signed-off-by: Marton Balint <cus@passwd.hu>
The samples I found all have 2000 sample packets, and by forcing the packet
size with a bsf we could automagically make muxing work for packets containing
more than 3640 samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
The allocators have been superseded by av_vdpau_bind_context().
The latter have only been added "to allow multiple forks to add
fields to the structure without breaking ABI" [1], but libav
is no more, so this is not needed any longer.
[1]: https://ffmpeg.org/pipermail/ffmpeg-devel/2013-August/146954.html
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Frame-level side data attributes are printed with the same key/value
structure as packet-level side data attributes, but this is not
reflected in the XSD.
The wav demuxer by default tried to demux 4096-byte packets which caused
packets with very few number of samples for files with high channel count.
This caused a significant overhead especially since the latest ffmpeg.c
threading changes.
So let's use a similar approach for selecting audio frame size which is already
used in the PCM demuxer, which is to read 25 times per second but at most 1024
samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
