One of the inputs to the fate test has an rgba pixel format which needs
to be converted to rgb32 (argb on big-endian) for the hqx filter. Because auto
scaling in the fate test is disabled, this needs a separate scale
filter.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
SMVJPEG stores frames as slices of a big JPEG image. The decoder is
implemented as a wrapper that instantiates a full internal MJPEG
decoder, then forwards the decoded frames with offset data pointers.
This is unnecessarily complex and fragile, not supporting useful decoder
capabilities like direct rendering.
Re-implement the decoder inside the MJPEG decoder, which is accomplished
by returning each decoded frame multiple times, setting cropping
information appropriately on each instance.
One peculiar aspect of the previous design is that since
- the smvjpeg decoder returns one frame per input packet
- there are multiple frames in each packets (the aformentioned slices)
the demuxer needs to return each packet multiple times.
This is now also eliminated - the demuxer now returns each packet
exactly once, with the duration set to the number of frames it decodes
to.
This also removes one of the last remaining internal uses of the old
video decoding API.
It depends on the muxer generating the timestamps, which is deprecated
and scheduled for removal on next bump.
A bunch of tests change timestamps, because of ffmpeg.c is not
generating them correctly. This should be fixed later.
Factor out the code into a separate muxing-specific function.
Stop accessing the deprecated AVStream-embedded codec context, use the
average framerate (if specified) instead.
By using the frame counter (and the video time base) for audio pts we lose some
timestamp precision but we ensure that video and audio coming from the same DV
frame are always in sync.
This patch also makes timestamps after seek consistent and it should also fix
the timestamps when the audio clock is unlocked and have a completely
indpendent clock source. (E.g. runs on fixed 48009 Hz which should have been
exact 48000 Hz)
Fixes out of sync timestamps in ticket #8762.
Signed-off-by: Marton Balint <cus@passwd.hu>
The previous threshold, 4 KB, maybe was reasonable when it was set
(in 2010), but in today's settings and with typical network speeds
and data sizes, it's pretty small. 32 KB probably is a more reasonable
default now, regardless of input.
This changes the test references for two seek tests.
When using the normal seek function, which boils down to the lseek(2)
function, a seek to an out of bounds position doesn't return an error,
but that condition is only reported when doing the subsequent read
(which returns EOF). When doing more seeks by fast forwarding, the
fact that the seeked to destination is out of bounds is noticed and
reported sooner in these cases.
Signed-off-by: Martin Storsjö <martin@martin.st>
While the FATE suite contains a sample file for Musepack 8, it did not
use it to test the decoder; it is only used in the mpc8-demux test that
tests the demuxer via streamcopy. Therefore this commit adds an actual
encoder test.
The test uses the framecrc output, because Musepack SV8 is an encoder
that returns multiple frames for a single packet, so that timing
information in the test output is valueable. Output seeking has been
used in order to limit the size of the ref file as well as to test this
codepath for the first time.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
We now have the possibility of getting AVFrames here, and we should
not touch the muxer's codecpar after writing the header.
Results of FATE tests change as the MXF and Matroska muxers actually
write down the field/frame coding type of a stream in their
respective headers. Before this change, these values in codecpar
would only be set after the muxer was initialized. Now, the
information is also available for encoder and muxer initialization.
Additionally, reap the first rewards by being able to set the
color related encoding values based on the passed AVFrame.
The only tests that seem to have changed their results with this
change seem to be the MXF tests. There, the muxer writes the
limited/full range flag to the output container if the encoder
is not set to "unspecified".
This disallows the usage of ? and # in libavformat specific scheme options
(e.g. subfile,,start,32815239,end,0,,:video.ts) but this change was considered
acceptable.
Signed-off-by: ruiquan.crq <caihaoning83@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
the warning message:
warning: using floating point absolute value function
'fabs' when argument is of integer type
use FFABS to set the absolute value.
Signed-off-by: liuqi05 <liuqi05@kuaishou.com>
These conversion appears to be exhibiting the same rounding error as the rgbf32 formats where.
I seperated the rounding value from the 16 and 128 offsets, I think it makes it a little more clear.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
av1dec should no longer attempt to output empty frames if another decoder
was used for probing and it sucessfully set a pix_fmt ever since 05872c67a4,
so we can re-add the AV_CODEC_CAP_AVOID_PROBING cap.
Signed-off-by: James Almer <jamrial@gmail.com>
changes since v1:
- made into fate test
- fixed c90 warnings
- tests more intermediate formats
- tested on BE mips too
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Filters mostly work in native endianness, but they must output
a specified endianness, usually little: that requires a final
conversion for big endian.
I do not know what's the deal with gif-deal: inserting explicitly
the filters that are implicitly inserted result in less frames in
output. Probably a strange problem of duration.
Otherwise the result of such tests will not accurately reflect the
current state.
Reviewed-by: Jan Ekström <jeebjp@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If the average bit rate cannot be calculated, such as in the case
of streamed fragmented mp4, utilize various available parameters
in priority order.
Tests are updated where the esds or btrt or ISML manifest boxes'
output changes.
This is utilized by various media ingests to figure out the bit
rate of the content you are pushing towards it, so write it for
video, audio and subtitle tracks in case at least one nonzero value
is available. It is only mentioned for timed metadata sample
descriptions in QTFF, so limit it only to ISOBMFF (MODE_MP4) mode.
Updates the FATE tests which have their results changed due to the
20 extra bytes being written per track.
SMPTE 12M timecode can only count frames up to 39, because the tens-of-frames
value is stored in 2 bit. In order to resolve this 50/60 fps SMPTE timecode is
using the field bit (which is the same bit as the phase correction bit) to
signal the least significant bit of a 50/60 fps timecode. See SMPTE ST
12-1:2014 section 12.1.
Therefore we slightly change the format of the return value of
av_timecode_get_smpte_from_framenum and AV_FRAME_DATA_S12M_TIMECODE and start
using the previously unused Phase Correction bit as Field bit. (As the SMPTE
standard suggests)
We add 50/60 fps support to av_timecode_get_smpte_from_framenum by calling the
recently added av_timecode_get_smpte function in it which already handles this
properly.
This change affects the decklink indev and the DV and MXF muxers. MXF has no
fate test for 50/60fps content, DV does, therefore the changes.
MediaInfo (a recent version) confirms that half-frame timecode must be inserted
to DV. MXFInspect confirms valid timecode insertion to the System Item of MXF
files. For MXF, also see EBU R122.
Note that for DV the field flag is not used because in the HDV specs (SMPTE
370M) it is still defined as biphase mark polarity correction flag. So it
should not matter that the DV muxer overrides the field bit.
Signed-off-by: Marton Balint <cus@passwd.hu>
This AV1 decoder is currently only used for hardware accelerated decoding.
It can be extended into a native decoder in the future, so set its name to
"av1" and temporarily give it the lowest priority in the codec list.
Signed-off-by: Fei Wang <fei.w.wang@intel.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Use pthread to multithread dnn_execute_layer_conv2d.
Can be tested with command "./ffmpeg_g -i input.png -vf \
format=yuvj420p,dnn_processing=dnn_backend=native:model= \
espcn.model:input=x:output=y:options=conv2d_threads=23 \
-y sr_native.jpg -benchmark"
before patch: utime=11.238s stime=0.005s rtime=11.248s
after patch: utime=20.817s stime=0.047s rtime=1.051s
on my 3900X 12c24t @4.2GHz
About the increase of utime, it's because that CPU HyperThreading
technology makes logical cores twice of physical cores while cpu's
counting performance improves less than double. And utime sums
all cpu's logical cores' runtime. As a result, using threads num
near cpu's logical core's number will double utime, while reduce
rtime less than half for HyperThreading CPUs.
Signed-off-by: Xu Jun <xujunzz@sjtu.edu.cn>
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Allow to set the EOF timestamp.
Also: doc/filters/testsrc*: specify the rounding of the duration option.
The changes in the ref files are right.
For filter-fps-down, the graph is testsrc2=r=7:d=3.5,fps=3.
3.5=24.5/7, so the EOF of testsrc2 will have PTS 25/7.
25/7=(10+5/7)/3, so the EOF PTS for fps should be 11/7,
and the output should contain a frame at PTS 10.
For filter-fps-up, the graph is testsrc2=r=3:d=2,fps=7,
for filter-fps-up-round-down and filter-fps-up-round-up
it is the same with explicit rounding options.
But there is no rounding: testsrc2 produces exactly 6 frames
and 2 seconds, fps converts it into exactly 14 frames.
The tests should probably be adjusted to restore them to
a useful coverage.
Explicitly insert the scale or aresample filter where it would
have been inserted by the negotiation.
Re-enable conversions if it cannot be done easily.
If a conversion is needed in a test, we want to know about it.
If the negotiation changes and makes new conversion necessary,
we want to know about it even more.
The implementation of the tag tree did not
set the correct reset value for the encoder.
This lead to inefficent tag tree being encoded.
This patch fixes the implementation of the
ff_tag_tree_zero() function.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This patch allows setting a compression ratio and to
set multiple layers. The user has to input a compression
ratio for each layer.
The per layer compression ration can be set as follows:
-layer_rates "r1,r2,...rn"
for to create 'n' layers.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The dvbsubtest_filter.ts sample is a filtered version of the Videolan
sample database (samples/sub/dvbsub/dvbsubtest.ts) using Project X. It
originates from ticket #8844.
The write_colr flag has been marked as experimental for over 5 years.
It should be safe to enable its behavior by default as follows:
- Write the colr atom by default for mp4/mov if any of the following:
- The primaries/trc/matrix are all specified, OR
- There is an ICC profile, OR
- The user specified +write_colr
- Keep the write_colr flag for situations where the user wants to
write the colr atom even if the color info is unspecified (e.g.,
http://ffmpeg.org/pipermail/ffmpeg-devel/2020-March/259334.html)
This fixes https://trac.ffmpeg.org/ticket/7961
Signed-off-by: Michael Bradshaw <mjbshaw@google.com>
The new code is analog to how it's done in our mpegaudio parser.
Acked-by: Jun Zhao <barryjzhao@tencent.com>
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Also add and update some tests.
Change the semantic a little, because for filesytem paths
symlinks complicate things.
See the comments in the code for detail.
Fix trac tickets #8813 and 8814.
Reads color_primaries, color_trc and color_space from mxf
headers. ULs are from https://registry.smpte-ra.org/ site.
Signed-off-by: Harry Mallon <harry.mallon@codex.online>
Previously, the hls-fmp4 and hls-fmp4_ac3 tests used the same file
names for init and segment files, which occasionally could cause
corruption and failed tests, if the input files for both tests are
generated in parallel, as they could overwrite each other.
This happened to work some of the time, as the fmp4_ac3 test actually
only checked the init segment file (which the fmp4 test case never
wrote, due to using the incorrect hls_segment_type option) and the
fmp4 test case always regenerated the input files due to mismatched
target and file names.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously, with the file name not matching the target, the files
were regenerated every time fate is rerun - contrary to the other
test targets in the same file. (While regenerating it every time
might be desireable, as that's what the test is about, the file
at least has a dependency on the ffmpeg executable, making them
regenerated every time the executable is updated - and this change
at least makes it consistent with the rest.)
Signed-off-by: Martin Storsjö <martin@martin.st>
Will prevet FATE from breaking once LIBAVCODEC_VERSION_MINOR is bumped to 100.
Reported-by: zane
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Add MMI & MSA runtime detection for MIPS.
Basically there are two code pathes. For systems that
natively support CPUCFG instruction or kernel emulated
that instruction, we'll sense this feature from HWCAP and
report the flags according to values grab from CPUCFG. For
systems that have no CPUCFG (or not export it in HWCAP),
we'll parse /proc/cpuinfo instead.
Signed-off-by: Jiaxun Yang <jiaxun.yang@flygoat.com>
Reviewed-by: Shiyou Yin <yinshiyou-hf@loongson.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When one of output[i] & expected_output is NAN, the unit test will always pass.
Signed-off-by: Ting Fu <ting.fu@intel.com>
Reviewed-by: Guo, Yejun <yejun.guo@intel.com>
add probeaudiostream for get audio stream's codec_name,codec_time_base,
sample_fmt,channels and channel_layout.
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Important part of this algorithm is the double threshold step: pixels
above "high" threshold being kept, pixels below "low" threshold dropped,
pixels in between (weak edges) are kept if they are neighboring "high"
pixels.
The weak edge check uses a neighboring context and should not be applied
on the plane's border. The condition was incorrect and has been fixed in
the commit.
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Reviewed-by: Andriy Gelman <andriy.gelman@gmail.com>
floating point precision will cause rgb*max generate different value on
x86_32 and x86_64. have pass fate test on x86_32 and x86_64 by using
lrintf to get the nearest integral value for rgb * max before av_clip.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Now we just use one ADTS raw frame to calculate the bit rate, it's
lead to a larger error when get the duration from bit rate, the
improvement cumulate Nth ADTS frames to get the average bit rate.
e,g used the command get the duration like:
ffprobe -show_entries format=duration -i fate-suite/aac/foo.aac
before this improvement dump the duration=2.173935
after this improvement dump the duration=1.979267
in fact, the real duration can be get by command like:
ffmpeg -i fate-suite/aac/foo.aac -f null /dev/null with time=00:00:01.97
Also update the fate-adtstoasc_ticket3715.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
This is a requirement of the AV1-ISOBMFF spec. Section 2.1.
General Requirements & Brands states:
* It SHALL have the av01 brand among the compatible brands array of the FileTypeBox
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This causes regressions in end to end timestamps with mp3s and ffmpeg.
The revert is to avoid this regression in the 4.3 release
See: [FFmpeg-devel] [PATCH] Don't adjust start time for MP3 files; packets are not adjusted.
This reverts commit 460132c998.
7546ac2fee made it so that the start_time for mp3 files is
adjusted for skip_samples. However, this appears incorrect because
subsequent packet timestamps are not adjusted and skip_samples are
applied by deleting data from a packet without changing the timestamp.
E.g., we are told the start_time is ~25ms and we get a packet with a
timestamp of 0 that has had the skip_samples discarded from it. As such
rendering engines may incorrectly discard everything prior to the
25ms thinking that is where playback should officially start. Since the
samples were deleted without adjusting timestamps though, the true
start_time is still 0.
Other formats like MP4 with edit lists will adjust both the start
time and the timestamps of subsequent packets to avoid this issue.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
A buffer whose size is not a multiple of four has been initialized using
consecutive writes of 32bits. This results in a stack-buffer-overflow
reported by ASAN in the checkasm-sw_scale FATE-test.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
changes since v1
- default behavior, no longer hidden behind decoder parameter
- updated tests to reflect change
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The Matroska muxer writes the Chapters early when chapters were already
available when writing the header; in this case any tags pertaining to
these chapters get written, too.
Yet if no chapters had been supplied before writing the header, Chapters
can also be written when writing the trailer if any are supplied. Tags
belonging to these chapters were up until now completely ignored.
This commit changes this: Writing the tags belonging to chapters has
been moved to mkv_write_chapters(). If mkv_write_tags() has not been
called yet (i.e. when chapters are written when writing the header),
the AVIOContext for writing the ordinary Tags element is used, but not
output, as this is left to mkv_write_tags() in order to only write one
Tags element. Yet if mkv_write_tags() has already been called,
mkv_write_chapters() will output a Tags element of its own which only
contains the tags for chapters.
When chapters are available initially, the corresponding tags will now
be the first tags in the Tags element; but the ordering of tags in Tags
is irrelevant anyway.
This commit also makes chapter_id_offset local to mkv_write_chapters()
as it is used only there and not reused at all.
Potentially writing a second Tags element means that the maximum number
of SeekHead entries had to be incremented. All the changes to FATE
result from the ensuing increase in the amount of space reserved for the
SeekHead (21 bytes more).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
We won't be able to seek back to write the actual duration anyway.
FATE-tests using the md5pipe command had to be updated due to this change.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Also fill x8-x17 with garbage before calling the function.
Figure out the number of stack parameters and make sure that the
value on the stack after those is untouched.
Signed-off-by: Martin Storsjö <martin@martin.st>
Figure out the number of stack parameters and make sure that the
value on the stack after those is untouched.
Signed-off-by: Martin Storsjö <martin@martin.st>
We should just use a normal bl here, and the linker will add the 'x'
bit if necessary.
This fixes calling the checkasm_fail_func on windows, where the
code is built in thumb mode (and the linker doesn't clear the 'x'
bit in the blx instruction).
Signed-off-by: Martin Storsjö <martin@martin.st>
have tested on linux x86_32/64, mingw32/64 arm & mips qemu
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Tested on x86-32/64, mingw32/64, arm & mips qemu
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
This fixes tests on 32 bit x86 mingw with clang, which uses x87
fpu by default.
In this setup, while the get_expected function is declared to
return float, the compiler is (especially given the optimization
flags set) free to keep the intermediate values (in this case,
the return value from the inlined function) in higher precision.
This results in the situation where 7.28 (which actually, as
a float, ends up as 7.2800002098), multiplied by 100, is
728.000000 when really forced into a 32 bit float, but 728.000021
when kept with higher intermediate precision.
For the multiplication case, a more suitable epsilon would e.g.
be 2*FLT_EPSILON*fabs(expected_output), but just increase the
current hardcoded threshold for now.
Signed-off-by: Martin Storsjö <martin@martin.st>
Up until now, the Matroska muxer would mark a track as default if it had
the disposition AV_DISPOSITION_DEFAULT or if there was no track with
AV_DISPOSITION_DEFAULT set; in the latter case even more than one track
of a kind (audio, video, subtitles) was marked as default which is not
sensible.
This commit changes the logic used to mark tracks as default. There are
now three modes for this:
a) In the "infer" mode the first track of every type (audio, video,
subtitles) with default disposition set will be marked as default; if
there is no such track (for a given type), then the first track of this
type (if existing) will be marked as default. This behaviour is inspired
by mkvmerge. It ensures that the default flags will be set in a sensible
way even if the input comes from containers that lack the concept of
default flags. This mode is the default mode.
b) The "infer_no_subs" mode is similar to the "infer" mode; the
difference is that if no subtitle track with default disposition exists,
no subtitle track will be marked as default at all.
c) The "passthrough" mode: Here the track will be marked as default if
and only the corresponding input stream had disposition default.
This fixes ticket #8173 (the passthrough mode is ideal for this) as
well as ticket #8416 (the "infer_no_subs" mode leads to the desired
output).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Several EBML Master elements for which a good upper bound of the final
length was available were nevertheless written without giving an
upper bound of the final length to start_ebml_master(), so that their
length fields were eight bytes long. This has been changed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Matroska muxer currently only adds CuePoints in three cases:
a) For video keyframes. b) For the first audio frame in a new Cluster if
in DASH-mode. c) For subtitles. This means that ordinary Matroska audio
files won't have any Cues which impedes seeking.
This commit changes this. For every track in a file without video track
it is checked and tracked whether a Cue entry has already been added
for said track for the current Cluster. This is used to add a Cue entry
for each first packet of each track in each Cluster.
Implements #3149.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
