This matches a similar cap on the number of automatic threads
in libavcodec/pthread_slice.c.
On systems with lots of cores, this fixes a couple fate failures
in 32 bit mode on such machines (where spawning a huge number of
threads runs out of address space).
Signed-off-by: Martin Storsjö <martin@martin.st>
Instead of the potentially adjusted ones. Otherwise, if config_props() is
called again and if using force_original_aspect_ratio, the already adjusted
values could be altered again.
Example command line
scale=size=1920x1000:force_original_aspect_ratio=decrease:force_divisible_by=2
user value 1920x1000 -> 1920x798 on init_dict() -> 1918x798 on frame
change when eval_mode == EVAL_MODE_INIT, which after e645a1ddb9 could be at the
very first frame.
Signed-off-by: James Almer <jamrial@gmail.com>
This state is not refcounted, so make sure it always has a well-defined
owner.
Remove the block added in 091341f2ab, as
this commit also solves that issue in a more general way.
Mention:
- that it is legacy and optional (every hwaccel that uses it can also
work with hwcontext, though some optional information can only be
signalled throught hwaccel_context)
- that it can be used for encoders (only qsvenc currently)
- ownership and lifetime
This commit implements an iMDCT in pure assembly.
This is capable of processing any mod-8 transforms, rather than just
power of two, but since power of two is all we have assembly for
currently, that's what's supported.
It would really benefit if we could somehow use the C code to decide
which function to jump into, but exposing function labels from assebly
into C is anything but easy.
The post-transform loop could probably be improved.
This was somewhat annoying to write, as we must support arbitrary
strides during runtime. There's a fast branch for stride == 4 bytes
and a slower one which uses vgatherdps.
Zen 3 benchmarks for stride == 4 for old (av_imdct_half) vs new (av_tx):
128pt:
2811 decicycles in av_tx (imdct),16775916 runs, 1300 skips
3082 decicycles in av_imdct_half,16776751 runs, 465 skips
256pt:
4920 decicycles in av_tx (imdct),16775820 runs, 1396 skips
5378 decicycles in av_imdct_half,16776411 runs, 805 skips
512pt:
9668 decicycles in av_tx (imdct),16775774 runs, 1442 skips
10626 decicycles in av_imdct_half,16775647 runs, 1569 skips
1024pt:
19812 decicycles in av_tx (imdct),16777144 runs, 72 skips
23036 decicycles in av_imdct_half,16777167 runs, 49 skips
Inside a function, the second ';' in ";;" is just a null statement,
but it is actually illegal outside of functions. Compilers
nevertheless accept it without warning, except when in -pedantic
mode when e.g. Clang emits a -Wextra-semi warning. Therefore
remove the unnecessary ';'.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The field is not specific to Opus.
The mp2fixed encoder signals initial_padding and is used
by both the matroska-encoding-delay test as well as
the lavf-mkv tests which necessitated several FATE ref changes.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Matroska requires pts to be >= 0 with a slight exception:
It has a mechanism to deal with codec delay, i.e. with
the data added at the beginning that does not correspond
to actual input data and should be discarded by the player.
Only the audio actually intended to be output needs to have
a timestamp >= 0.
In order to avoid unnecessary timestamp shifting, this patch
allows muxers to inform the shifting code about this so that
it can take it into account.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Matroska generally requires timestamps to be nonnegative, but
there is an exception: Data that corresponds to encoder delay
and is not supposed to be output anyway can have a negative
timestamp. This is achieved by using the CodecDelay header
field: The demuxer has to subtract this value from the raw
(nonnegative) timestamps of the corresponding track.
Therefore the muxer has to add this value first to write
this raw timestamp.
Support for writing CodecDelay has been added in FFmpeg commit
d92b1b1bab and in Libav commit
a1aa37dd0b. The former simply
wrote the header field and did not apply any timestamp offsets,
leading to desynchronisation (if one uses multiple tracks).
The latter applied it at two places, but not at the one where
it actually matters, namely in mkv_write_block(), leading to
the same desynchronisation as with the former commit. It furthermore
used the wrong stream timebase to convert the delay to the
stream's timebase, as the conversion used the timebase from
before avpriv_set_pts_info().
When the latter was merged in 82e4f39883,
it was only done in a deactivated state that still did not
offset the timestamps when muxing due to "assertion failures
and av sync errors". a1aa37dd0b
made it definitely more likely to run into assertion failures
(namely if the relative block timestamp doesn't fit into an int16_t).
Yet all of the above issues have been fixed (in commits
962d631573,
5d3953a5dc and
4ebeab15b0. This commit therefore
enables applying CodecDelay, fixing ticket #7182.
There is just one slight regression from this: If one has input
with encoder delay where the first timestamp is negative, but
the pts of the part of the data that is actually intended to be
output is nonnegative, then the timestamps will currently by default
be shifted to make them nonnegative before they reach the muxer;
the muxer will then ensure that the shifted timestamps are retained.
Before this commit, the muxer did not ensure this; instead the
timestamps that the demuxer will output were shifted and
if the first timestamp of the actually intended output was zero
before shifting, then this unintentional shift just cancels
the shift performed before the packet reached the muxer.
(But notice that this only applies if all the tracks use the same
CodecDelay, or the relative sync between tracks will be impaired.)
This happens in the matroska-opus-remux and matroska-ogg-opus-remux
FATE tests. Future commits will forward the information that
the Matroska muxer has a limited capability to handle negative
timestamps so that the shifting in libavformat can take advantage
of it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Opus can be decoded to multiple samplerates (namely 48kHz, 24KHz,
16Khz, 12 KHz and 8Khz); libopus as well as our encoder wrapper
support these sample rates. The OpusHead contains a field for
this original samplerate. Yet the pre-skip (and the granule-position
in the Ogg-Opus mapping in general) are always in the 48KHz clock,
irrespective of the original sample rate.
Before commit c3c22bee63, our libopus
encoder was buggy: It did not account for the fact that the pre-skip
field is always according to a 48kHz clock and wrote a too small
value in case one uses the encoder with a sample rate other than 48kHz;
this discrepancy between CodecDelay and OpusHead led to Firefox
rejecting such streams.
In order to account for that, said commit made the muxer always use
48kHz instead of the actual sample rate to convert the initial_padding
(in samples in the stream's sample rate) to ns. This meant that both
fields are now off by the same factor, so Firefox was happy.
Then commit f4bdeddc3c fixed the issue
in libopusenc; so the OpusHead is correct, but the CodecDelay is
still off*. This commit fixes this by effectively reverting
c3c22bee63.
*: Firefox seems to no longer abort when CodecDelay and OpusHead
are off.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It is possible for the trailing padding to be zero, namely
e.g. if the AV_PKT_DATA_SKIP_SAMPLES side data is used
for leading padding. Matroska supports this (use a negative
DiscardPadding), but players do not; at least Firefox refuses
to play such a file. So for now only write DiscardPadding
if it is trailing padding and nonzero.
The fate-matroska-ogg-opus-remux was affected by this.
(I wish CodecDelay would not exist and DiscardPadding would
be used to instead trim the codec delay away (with the Block
timestamp corresponding to the time at which the actually
output audio is output).)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Creating CFHD RL VLC tables works by first extending
the codes by the sign, followed by creating a VLC,
followed by deriving the RL VLC from this VLC (which
is then discarded). Extending the codes uses stack arrays.
The tables used to initialize the VLC are already sorted
from left-to-right in the tree. This means that the
corresponding VLC entries are generally also ascending,
but not always: Entries from subtables always follow
the corresponding main table although it is possible
for the right-most node to fit into the main table.
This suggests that one can try to use the final destination
buffer as scratch buffer for the tables with sign included.
Unfortunately it works for neither of the tables if one
uses the right-most part of the RL VLC buffer as scratch buffer;
using the left-most part of the RL VLC buffer as scratch buffer
might work if one traverses the VLC entries from end to start.
But it works only for the little RL VLC (table 9), not for table 18.
Therefore this patch uses the RL VLC buffer for table 9
as scratch buffer for creating the bigger table 18.
Afterwards the left part of the buffer for table 9 is
used as scratch buffer to create table 9.
This fixes the cfhd part of ticket #9399 (if it is not already fixed).
Notice that I do not consider the previous stack usage excessive.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The VLC is only used to initialize RL VLC.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
cfhddata.c initializes a RL VLC table via code tables and
corresponding tables for length, run and level. code and length
tables are used to initialize a VLC, no symbol table is used.
Afterwards the symbols of said VLC are just the indices of
the corresponding entries in the code and length table that
were used for initialization; they can therefore be used
to get the matching level and run entry and they are not used
for anything else. Therefore one can just permute these tables
without changing the resulting RL VLC tables.
This commit does just this. It permutes these tables so that
the code tables are ordered from left to right in the resulting
tree and then switches to ff_init_vlc_from_lengths(), which
allows to remove the codes table altogether.
Given that these tables are constructed on the stack, this
also reduces stack usage, potentially fixing part of #9399.
(The size of the tables on the stack decreases from 4752 to
2640.)
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
cfhd.c checked for level being equal to a certain codebook-
dependent constant and to run being two. The first check is
actually redundant, as all codebooks contain only one (real)
entry with run == 2 (as is usual with VLCs, this one real entry
has several corresponding entries in the table). But given
that no entry has a run of zero (except incomplete entries
which just signal that one needs to do another round of parsing),
one can actually use that as sentinel. This patch does so.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Demuxers are not allowed to do this and few callers, if any, will handle
this correctly. Send the AV_SIDE_DATA_PARAM_CHANGE_SAMPLE_RATE side data
instead.
Demuxers are not supposed to update AVCodecParameters after the stream
was seen by the caller. This value is not important enough to support
dynamic updates for.
The mov demuxer only returns DV audio, video packets are discarded.
It first reads the data to be parsed into a packet. Then both this
packet and the pointer to its data are passed together to
avpriv_dv_produce_packet(), which parses the data and partially
overwrites the packet. This is confusing and potentially dangerous, so
just pass NULL and avoid pointless packet modification.
The struct is quite small and the decoder and the encoder use different
fields from it, so benefits from reusing it are small.
This allows making the buf field const.
The function contains only two assignments, setting DVVideoContext.avctx
and AVCodecContext.chroma_sample_location. However, the decoder does not
use the former, and the encoder should not be setting the latter.
Therefore move the first assignment to dvenc and the second to dvdec.
Make the encoder warn if the user-signalled chroma sample location does
not match the supported one, and return an error on higher compliance
levels.
Martin Storsjö
James Almer
Anton Khirnov
Lynne
Andreas Rheinhardt
Paul B Mahol