The earlier code distinguished between a partial reset
(yae_clear()) and a complete reset (yae_release_buffers()
which also releases the buffers); this separation existed
to avoid allocations, as buffers were reallocated on reconfigs.
Yet it is pointless since a5704659e3,
so simply use yae_release_buffers() everywhere.
Reviewed-by: Pavel Koshevoy <pkoshevoy@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.
Keep it for external users in order to not cause breakages.
Also improve the other headers a bit while just at it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
If one looks at the many query_formats callbacks in existence,
one will immediately recognize that there is one type of default
callback for video and a slightly different default callback for
audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);"
for video with a filter-specific pix_fmts list. For audio, it is
the same with a filter-specific sample_fmts list together with
ff_set_common_all_samplerates() and ff_set_common_all_channel_counts().
This commit allows to remove the boilerplate query_formats callbacks
by replacing said callback with a union consisting the old callback
and pointers for pixel and sample format arrays. For the not uncommon
case in which these lists only contain a single entry (besides the
sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also
added to the union to store them directly in the AVFilter,
thereby avoiding a relocation.
The state of said union will be contained in a new, dedicated AVFilter
field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t
in order to create a hole for this new field; this is no problem, as
the maximum of all the nb_inputs is four; for nb_outputs it is only
two).
The state's default value coincides with the earlier default of
query_formats being unset, namely that the filter accepts all formats
(and also sample rates and channel counts/layouts for audio)
provided that these properties agree coincide for all inputs and
outputs.
By using different union members for audio and video filters
the type-unsafety of using the same functions for audio and video
lists will furthermore be more confined to formats.c than before.
When the new fields are used, they will also avoid allocations:
Currently something nearly equivalent to ff_default_query_formats()
is called after every successful call to a query_formats callback;
yet in the common case that the newly allocated AVFilterFormats
are not used at all (namely if there are no free links) these newly
allocated AVFilterFormats are freed again without ever being used.
Filters no longer using the callback will not exhibit this any more.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Up until now, an AVFilter's lists of input and output AVFilterPads
were terminated by a sentinel and the only way to get the length
of these lists was by using avfilter_pad_count(). This has two
drawbacks: first, sizeof(AVFilterPad) is not negligible
(i.e. 64B on 64bit systems); second, getting the size involves
a function call instead of just reading the data.
This commit therefore changes this. The sentinels are removed and new
private fields nb_inputs and nb_outputs are added to AVFilter that
contain the number of elements of the respective AVFilterPad array.
Given that AVFilter.(in|out)puts are the only arrays of zero-terminated
AVFilterPads an API user has access to (AVFilterContext.(in|out)put_pads
are not zero-terminated and they already have a size field) the argument
to avfilter_pad_count() is always one of these lists, so it just has to
find the filter the list belongs to and read said number. This is slower
than before, but a replacement function that just reads the internal numbers
that users are expected to switch to will be added soon; and furthermore,
avfilter_pad_count() is probably never called in hot loops anyway.
This saves about 49KiB from the binary; notice that these sentinels are
not in .bss despite being zeroed: they are in .data.rel.ro due to the
non-sentinels.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Several combinations of functions happen quite often in query_format
functions; e.g. ff_set_common_formats(ctx, ff_make_format_list(sample_fmts))
is very common. This commit therefore adds functions that are equivalent
to commonly used function combinations in order to reduce code
duplication.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This is possible now that the next-API is gone.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: James Almer <jamrial@gmail.com>
yae_set_tempo was overlooked when max tempo limit was raised to 100.
tested with:
./ffmpeg_g -i Delerium/SemanticSpaces/Gateway.mp3 \
-af asendcmd=f=asendcmd.cfg,atempo=1.0 -y /tmp/asendcmd-atempo.wav
where asendcmd.cfg was:
15.0-45.0 [enter] atempo tempo 2.0,
[leave] atempo tempo 0.5;
60.0-300.0 [enter] atempo tempo 4.0,
[leave] atempo tempo 1.0;
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Pavel Koshevoy <pkoshevoy@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Current method for constraining fragment position drift suffers from
round-off error build up.
Instead of calculating cumulative drift as a sum of input fragment
position corrections, it is more accurate to calculate drift as the
difference between current fragment position and the ideal position
specified by the tempo scale factor.
Signed-off-by: Pavel Koshevoy <pkoshevoy@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
valgrind reported uninitialized memory access which was caused by
incorrect number of samples being passed to push_samples(..)
Signed-off-by: Pavel Koshevoy <pkoshevoy@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is mostly automated global search and replace
The deprecated aconvert filter is disabled, if it still has users
it should be updated
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Memory obtained from av_realloc is not aligned enough for AVX.
The other similar allocations are changed too because they use
the same macro. The buffers were cleared afterwards anyway.
Fix trac ticket #1692.
* qatar/master:
mss3: use standard zigzag table
mss3: split DSP functions that are used in MTS2(MSS4) into separate file
motion-test: do not use getopt()
tcp: add initial timeout limit for incoming connections
configure: Change the rdtsc check to a linker check
avconv: propagate fatal errors from lavfi.
lavfi: add error handling to filter_samples().
fate-run: make avconv() properly deal with multiple inputs.
asplit: don't leak the input buffer.
af_resample: fix request_frame() behavior.
af_asyncts: fix request_frame() behavior.
libx264: support aspect ratio switching
matroskadec: honor error_recognition when encountering unknown elements.
lavr: resampling: add support for s32p, fltp, and dblp internal sample formats
lavr: resampling: add filter type and Kaiser window beta to AVOptions
lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format
lavr: mix: validate internal sample format in ff_audio_mix_init()
Conflicts:
ffmpeg.c
ffplay.c
libavcodec/libx264.c
libavfilter/audio.c
libavfilter/split.c
libavformat/tcp.c
tests/fate-run.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'a5e8c41c28f907d98d2a739db08f7aef4cbfcf3a':
lavfi: remove 'opaque' parameter from AVFilter.init()
mov: do not try to read total disc/track number if data atom is too short.
avconv: fix -force_key_frames
dxva2_h264: fix signaling of mbaff frames
x86: fft: elf64: fix PIC build
Conflicts:
ffmpeg.c
libavcodec/v210dec.h
libavfilter/asrc_anullsrc.c
libavfilter/buffersrc.c
libavfilter/src_movie.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_overlay.c
libavfilter/vsrc_color.c
libavfilter/vsrc_testsrc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
libavfilter API was designed in order to be clarly distinguished from the
libavcodec API, including avcodec.h in avfilter.h is not going to help to
stick to this principle.
The inclusion of libavutil/audioconvert.h in many files was required
because avcodec.h includes audioconvert.h.
libavfilter/avcodec.h is where the lavc/lavfi interface should be
entirely placed.
* qatar/master:
x86: Only use optimizations with cmov if the CPU supports the instruction
x86: Add CPU flag for the i686 cmov instruction
x86: remove unused inline asm macros from dsputil_mmx.h
x86: move some inline asm macros to the only places they are used
lavfi: Add the af_channelmap audio channel mapping filter.
lavfi: add join audio filter.
lavfi: allow audio filters to request a given number of samples.
lavfi: support automatically inserting the fifo filter when needed.
lavfi/audio: eliminate ff_default_filter_samples().
Conflicts:
Changelog
libavcodec/x86/h264dsp_mmx.c
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/avfilter.h
libavfilter/avfiltergraph.c
libavfilter/version.h
libavutil/x86/cpu.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
