677a030b26 introduced more printable
side data types in ffprobe, however the Audio Service Type side data
'type' field that was introduced aliases an existing field of the same
name within the side data array, which can lead to JSON output like:
"side_data_list": [
{
"side_data_type": "Audio Service Type",
"type": 0
},
{
"side_data_type": "Stereo 3D",
"type": "side by side",
"inverted": 1
}
]
This, while technically valid JSON, is considered bad practice, since it
forces all downstream users to manually parse it and check all types;
it makes simple deserialization impossible. Worse, in som loosely
type languages, it can lead to silent bugs if exising code assumed
it was a different type.
As such, rename this second "type" field to "service_type".
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Adds schema validation for ffprobe XML output so that updating the
ffprobe.xsd file upon changes to ffprobe is not forgotten. This was
suggested by Marton Balint in:
http://ffmpeg.org/pipermail/ffmpeg-devel/2021-March/278428.html
The schema FATE test is only run if xmllint command is available.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
After fixing AV_PKT_DATA_SKIP_SAMPLES for reading vorbis packets from ogg,
the actual decoded samples become fewer. Three fate tests are failing:
fate-vorbis-20:
The samples in 6.ogg are not frame aligned. 6.pcm file was generated by
ffmpeg before the fix. After the fix, the decoded pcm file does not match
anymore. Ideally the ref file 6.pcm should be updated but it is probably
not worth it including another copy of the same file, only smaller.
SIZE_TOLERANCE is added for this test case.
fate-webm-dash-chapters:
The original vorbis_chapter_extension_demo.ogg is transmuxed to dash-webm.
The ref file webm-dash-chapters needs to be updated.
fate-vorbis-encode:
This exposes another bug in the vorbis encoder that initial_padding is not
correctly set. It is fixed in the previous patch.
Signed-off-by: Guangyu Sun <gsun@roblox.com>
Also use helper function to set the timestamp. Maybe we could also use
nanosecond precision, but there were some float rounding concerns.
Signed-off-by: Marton Balint <cus@passwd.hu>
Export them in UTC, not the local timezone. This way the output is
the same everywhere. The timezone information stored in the file is
still ignored, since there seems to be no simple way to export it
correctly.
Format them according to ISO 8601, which we generally use for exporting
dates.
Fixes fate-flv-demux, which was broken since
958bea5248 on some platforms.
There are no guarantees that all side data types have the same
representation on all platforms.
Tests that change output due to this:
id3v2-priv-remux, cover-art-mp3-id3v2-remux, gapless-mp3: SKIP_SAMPLES,
which is tested by fate-gapless-mp3-side-data
matroska-vp8-alpha-remux: MATROSKA_BLOCKADDITIONAL, which is tested by
remux itself (side data is written into output)
matroska-mastering-display-metadata: MASTERING_DISPLAY_METADATA and
CONTENT_LIGHT_LEVEL, which are tested by ffprobe invocation in the same
test
matroska-spherical-mono-remux: STEREO3D and SPHERICAL, which are tested
by ffprobe invocation in the same test
segment-mp4-to-ts: MPEGTS_STREAM_ID, which is tested by ts remuxing
tests
webm-webvtt-remux: WEBVTT_IDENTIFIER/SETTINGS, which is tested by the
ffprobe invocation in the same test
mxf-d10-user-comments: CPB_PROPERTIES, which is tested by mxf-probe-d10
mov-cover-image: SKIP_SAMPLES, which is tested for mov by
mov-aac-2048-priming
copy-trac3074: AUDIO_SERVICE_TYPE, which is tested by fate-hls-fmp4_ac3
Deprecated in ddef3d902f.
(The reference file of the mov-zombie test needed to be updated, because
a rotate metadata tag is no longer exported; the side-data is of course
still present.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Announced in 2e8b0446c6.
Two FATE-tests needed to be updated because the checksums of
side data containing an AVCPBProperties struct changed.
buffer_size has also been switched to 64bits because it is a bitsize.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The ASS margins are utilized to generate percentual values, as
the usage of cell-based sizing and offsetting seems to be not too
well supported by renderers.
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
Attempts to utilize the TTML cell resolution as a mapping to the
reference resolution, and maps font size to cell size. Additionally
sets the display and text alignment according to the ASS alignment
number.
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
When parsing ID3v2 tags, special (non-text) metadata is not applied
directly and unconditionally; instead it is stored in a linked list
in which elements are prepended. When traversing the list to add APICs
(or private tags) at the end, the order is reversed. The same also
happens for chapters and therefore the chapter parsing code already
reverses the chapters.
This commit changes this: By keeping pointers to both head and tail
of the linked list one can preserve the order of the entries and
remove the reordering code for chapters. Only the pointer to head
will be exported: No current caller uses a nonempty list, so exporting
both head and tail is unnecessary. This removes the functionality
to combine the lists of special metadata read from different ID3v2 tags,
but that doesn't make really much sense anyway (and would be trivial
to implement if desired) and allows to remove the now unnecessary
initializations performed by the callers.
The FATE-reference for the id3v2-priv test had to be updated
because the order of the tags read into the dict is reversed;
for id3v2-priv-remux only the md5 and not the ffprobe output
of the remuxed file changes because the order of the private tags
has up until now been reversed twice.
The references for the aiff/mp3 cover-art tests needed to be updated,
because the order of the attached pics is reversed upon reading.
It is still not correct, because the muxers write the pics in the order
in which they arrive at the muxer instead of the order given by
pkt->stream_index.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Notice that the order of the APIC tracks is currently wrong. This is
a superposition of two bugs: (i) Both muxers write the attached
pictures in the order they arrive in the muxer and not in the
stream_index order, leading to attached pictures that are copied being
written earlier because their timestamp is AV_NOPTS_VALUE, whereas the
timestamp of the encoded pictures is 0. (ii) A bug in the id3v2 parsing
code reverses the order of the parsed pictures.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Specifically test that the WebVTT flavour is correctly mapped to
the Matroska/WebM CodecID and back; and test that dispositions
unsupported by WebM are discarded even when they would be supported
by Matroska.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This makes av_read_frame() return packets with proper timestamps.
As a result, seeking now works in combination with streamcopy.
A FATE-test for this has been added.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The test sample has to have no file extension, otherwise probing
happens to work, based off file extension alone, and we want to
test the actual probing function.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Enables writing TTML documents or encoded TTML paragraphs as such
documents.
Additionally, a test for the combined TTML encoder and muxer has
been added to validate that the components still work.
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
It only got added recently, and the new name makes it consistent with
product_version_num in the next patch.
Signed-off-by: Marton Balint <cus@passwd.hu>
The FF_API macros are private and must not be used by external callers.
As the fields in question are to be removed without replacement, just
drop them.
The fields are:
AVPacket.convergence_duration
AVCodecContext.time_base
AVCodecContext.timecode_frame_start
AV_PIX_FMT_FLAG_PSEUDOPAL pixel descriptor flag
This provides coverage for writing BlockGroups with BlockAdditional
and ReferenceBlock elements. It also tests setting the hearing impaired
disposition (it fits given that this video has no audio so one needs to
be able to read lips to understand anything).
Reviewed-by: Ridley Combs <rcombs@rcombs.me>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The FATE suite already contains a file containing mastering display
and content light level metadata: Meridian-Apple_ProResProxy-HDR10.mxf
This file is used to test both the Matroska muxer and demuxer.
Reviewed-by: Ridley Combs <rcombs@rcombs.me>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The mxf_d10 muxer is very picky regarding the input it accepts:
The only video accepted is MPEG-2 with absolutely constant bitrate,
i.e. all packets need to have exactly the same size; and only a few
bitrates are accepted.
The sample file used did not abide by this: Writing the first packet
(a video packet) errors out and afterwards an audio packet from the
muxing queue has been written. That's all besides metadata (which this
test is about). The FFmpeg cli returned an error, but said error has
been ignored by the md5 test.
This commit changes the test to actually send a compliant stream to the
muxer, so that it does not error out; furthermore, the test is changed
to explicitly check the metadata instead of it only being implicitly
included in the md5 checksum. The compliant stream is created by our
encoder at runtime.
Finally, the test now also covers writing user-specified
product/company/version identification.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Also, test modifying colorspace properties and the default_mode
passthrough which is used here to create a file that has no default
track at all.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
It furthermore tests the demuxer's handling of chained SeekHeads,
level 1-elements after the Clusters and the muxer's capability of
writing huge TrackNumbers as well as expanding the Cues' length field
by one byte if necessary to fill the reserved space. It also tests
propagation of metadata.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
MJPEG does not have a single quantiser scale, so this does not fit into
the intended API use.
This removes the last use of the long-deprecated QP table API.
There is a minor bug in xbm encode which adds a trailing comma at the end
of data. This isn't a big problem, but it would be nicer to be more
technically true to an array of data (by not including the last comma).
This bug fixes the output from something like this (having 4 values):
static unsigned char image_bits[] = { 0x00, 0x11, 0x22, }
to C code that looks like this instead (having 3 values):
static unsigned char image_bits[] = { 0x00, 0x11, 0x22 }
which is the intended results.
Subject: [PATCH 1/3] avcodec/xbmenc: Do not add last comma into output array
xbm outputs c arrays of data.
Including a comma at the end means there is another value to be added.
This bug fix changes something like this:
static unsigned char image_bits[] = { 0x00, 0x11, 0x22, }
to C code like this:
static unsigned char image_bits[] = { 0x00, 0x11, 0x22 }
Signed-off-by: Joe Da Silva <digital@joescat.com>
If the edit lists remove parts of the output timeline, or add a
delay to it, this should be included in the mvhd/tkhd/mdhd durations,
which should correspond to the edit lists.
For tracks starting with pts < 0, the edit list trims out the segment
before pts=0. For tracks starting with pts > 0, a delay element is
added in the edit list, delaying the start of the track data.
In both cases, the practical effect is that the post-edit output
is as if the track had started with pts = 0. Thus calculate the range
from pts=0 to end_pts, for the purposes of mvhd/tkhd/mdhd, unless
edit lists explicitly are disabled.
mov_write_edts_tag needs to operate on the actual pts duration of
the track samples, not the duration that already takes the edit
list effect into account.
Signed-off-by: Martin Storsjö <martin@martin.st>
The AC3 encoder used to be a separate library called "Aften", which
got merged into libavcodec (literally, SVN commits and all).
The merge preserved as much features from the library as possible.
The code had two versions - a fixed point version and a floating
point version. FFmpeg had floating point DSP code used by other
codecs, the AC3 decoder including, so the floating-point DSP was
simply replaced with FFmpeg's own functions.
However, FFmpeg had no fixed-point audio code at that point. So
the encoder brought along its own fixed-point DSP functions,
including a fixed-point MDCT.
The fixed-point MDCT itself is trivially just a float MDCT with a
different type and each multiply being a fixed-point multiply.
So over time, it got refactored, and the FFT used for all other codecs
was templated.
Due to design decisions at the time, the fixed-point version of the
encoder operates at 16-bits of precision. Although convenient, this,
even at the time, was inadequate and inefficient. The encoder is noisy,
does not produce output comparable to the float encoder, and even
rings at higher frequencies due to the badly approximated winow function.
Enter MIPS (owned by Imagination Technologies at the time). They wanted
quick fixed-point decoding on their FPUless cores. So they contributed
patches to template the AC3 decoder so it had both a fixed-point
and a floating-point version. They also did the same for the AAC decoder.
They however, used 32-bit samples. Not 16-bits. And we did not have
32-bit fixed-point DSP functions, including an MDCT. But instead of
templating our MDCT to output 3 versions (float, 32-bit fixed and 16-bit fixed),
they simply copy-pasted their own MDCT into ours, and completely
ifdeffed our own MDCT code out if a 32-bit fixed point MDCT was selected.
This is also the status quo nowadays - 2 separate MDCTs, one which
produces floating point and 16-bit fixed point versions, and one
sort-of integrated which produces 32-bit MDCT.
MIPS weren't all that interested in encoding, so they left the encoder
as-is, and they didn't care much about the ifdeffery, mess or quality - it's
not their problem.
So the MDCT/FFT code has always been a thorn in anyone looking to clean up
code's eye.
Backstory over. Internally AC3 operates on 25-bit fixed-point coefficients.
So for the floating point version, the encoder simply runs the float MDCT,
and converts the resulting coefficients to 25-bit fixed-point, as AC3 is inherently
a fixed-point codec. For the fixed-point version, the input is 16-bit samples,
so to maximize precision the frame samples are analyzed and the highest set
bit is detected via ac3_max_msb_abs_int16(), and the coefficients are then
scaled up via ac3_lshift_int16(), so the input for the FFT is always at least 14 bits,
computed in normalize_samples(). After FFT, the coefficients are scaled up to 25 bits.
This patch simply changes the encoder to accept 32-bit samples, reusing
the already well-optimized 32-bit MDCT code, allowing us to clean up and drop
a large part of a very messy code of ours, as well as prepare for the future lavu/tx
conversion. The coefficients are simply scaled down to 25 bits during windowing,
skipping 2 separate scalings, as the hacks to extend precision are simply no longer
necessary. There's no point in running the MDCT always at 32 bits when you're
going to drop 6 bits off anyway, the headroom is plenty, and the MDCT rounds
properly.
This also makes the encoder even slightly more accurate over the float version,
as there's no coefficient conversion step necessary.
SIZE SAVINGS:
ARM32:
HARDCODED TABLES:
BASE - 10709590
DROP DSP - 10702872 - diff: -6.56KiB
DROP MDCT - 10667932 - diff: -34.12KiB - both: -40.68KiB
DROP FFT - 10336652 - diff: -323.52KiB - all: -364.20KiB
SOFTCODED TABLES:
BASE - 9685096
DROP DSP - 9678378 - diff: -6.56KiB
DROP MDCT - 9643466 - diff: -34.09KiB - both: -40.65KiB
DROP FFT - 9573918 - diff: -67.92KiB - all: -108.57KiB
ARM64:
HARDCODED TABLES:
BASE - 14641112
DROP DSP - 14633806 - diff: -7.13KiB
DROP MDCT - 14604812 - diff: -28.31KiB - both: -35.45KiB
DROP FFT - 14286826 - diff: -310.53KiB - all: -345.98KiB
SOFTCODED TABLES:
BASE - 13636238
DROP DSP - 13628932 - diff: -7.13KiB
DROP MDCT - 13599866 - diff: -28.38KiB - both: -35.52KiB
DROP FFT - 13542080 - diff: -56.43KiB - all: -91.95KiB
x86:
HARDCODED TABLES:
BASE - 12367336
DROP DSP - 12354698 - diff: -12.34KiB
DROP MDCT - 12331024 - diff: -23.12KiB - both: -35.46KiB
DROP FFT - 12029788 - diff: -294.18KiB - all: -329.64KiB
SOFTCODED TABLES:
BASE - 11358094
DROP DSP - 11345456 - diff: -12.34KiB
DROP MDCT - 11321742 - diff: -23.16KiB - both: -35.50KiB
DROP FFT - 11276946 - diff: -43.75KiB - all: -79.25KiB
PERFORMANCE (10min random s32le):
ARM32 - before - 39.9x - 0m15.046s
ARM32 - after - 28.2x - 0m21.525s
Speed: -30%
ARM64 - before - 36.1x - 0m16.637s
ARM64 - after - 36.0x - 0m16.727s
Speed: -0.5%
x86 - before - 184x - 0m3.277s
x86 - after - 190x - 0m3.187s
Speed: +3%
Fixes a decoding regression introduced by e9a2a87773, and as a side effect also
fixes bogus values set to certain audio frames that had some samples discarded,
where the offsets added to pts, pkt_dts and pkt_duration were not reflected in
best_effort_timestamp.
Signed-off-by: James Almer <jamrial@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>
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>
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>
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>
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.
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>
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>
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>
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>
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>
Moreover, putting the Cues in front of the Clusters by reserving space
in advance is also tested.
The new capability of using ffprobe during a remux/transcode test are
used here for information about the chapters.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Using random values for TrackUID and FileUID (as happens when the
AVFMT_FLAG_BITEXACT flag is not set) has the obvious downside of making
the output indeterministic. This commit mitigates this by writing the
potentially random values with a fixed size of eight byte, even if their
actual values would fit into less than eight bytes. This ensures that
even in non-bitexact mode, the differences between two files generated
with the same settings are restricted to a few bytes in the header.
(Namely the SegmentUID, the TrackUIDs (in Tracks as well as when
referencing them via TagTrackUID), the FileUIDs (in Attachments as
well as in TagAttachmentUID) as well as the CRC-32 checksums of the
Info, Tracks, Attachments and Tags level-1-elements.) Without this
patch, there might be an offset/a size difference between two such
files.
The FATE-tests had to be updated because the fixed-sized UIDs are also
used in bitexact mode.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If there are Attachments to write, the Matroska muxer currently
allocates two objects: An array that contains an entry for each
AttachedFile containing just the stream index of the corresponding
stream and the FileUID used for this AttachedFile; and a structure with
a pointer to said array and a counter for said array. These uids are
generated via code special to Attachments: It uses an AVLFG in the
normal and a sha of the attachment data in the bitexact case. (Said sha
requires an allocation, too.)
But now that an uid is generated for each stream in mkv_init(), there is
no need any more to use special code for generating the FileUIDs of
AttachedFiles: One can simply use the uid already generated for the
corresponding stream. And this makes the whole allocations of the
structures for AttachedFiles as well as the structures itself superfluous.
They have been removed.
In case AVFMT_FLAG_BITEXACT is set, the uids will be different from the
old ones which is the reason why the FATE-test lavf-mkv_attachment
needed to be updated. The old method had the drawback that two
AttachedFiles with the same data would have the same FileUID.
The new one doesn't.
Also notice that the dynamic buffer used to write the Attachments leaks
if an error happens when writing the buffer. By removing the
allocations potential sources of errors have been removed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Tags in the Matroska file format can be summarized as follows: There is
a level 1-element called Tags containing one or many Tag elements each
of which in turn contain a Targets element and one or many SimpleTags.
Each SimpleTag roughly corresponds to a single key-value pair similar to
an AVDictionaryEntry. The Targets meanwhile contains information to what
the metadata contained in the SimpleTags contained in the containing Tag
applies (i.e. to the file as a whole or to an individual track).
The Matroska muxer writes such metadata. It puts the metadata of every
stream into a Tag whose Targets makes it point to the corresponding
track. And if the output is seekable, then it also adds another Tag for
each track whose Targets corresponds to the track and where it reserves
space in a SimpleTag to write the duration at the end of the muxing
process into.
Yet there is no reason to write two Tag elements for a track and a few
bytes (typically 24 bytes per track) can be saved by adding the duration
SimpleTag to the other Tag of the same track (if it exists).
FATE has been updated because the output files changed. (Tests that
write to unseekable output (pipes) needn't be updated (no duration tag
has ever been written for them) and the same applies to tests without
further metadata.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
It represents the relationship between them more naturally and will be
useful in the following commits.
Allows significantly more frames in fate-h264-attachment-631 to be
decoded.
containing updated extradata, in this case a new FLAC streaminfo.
Furthermore, it also tests that the Matroska muxer is able to preserve
uncommon channel layouts by adding Vorbis comments to the CodecPrivate.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
mkvmerge versions 6.2 to 40.0 had a bug that made it not propagate the
WavPack extradata (containing the WavPack version) during remuxing from
a Matroska file; currently our demuxer would treat every WavPack block
encountered as invalid data (unless the WavPack stream is to be
discarded (i.e. the streams discard is >= AVDISCARD_ALL)) and try to
resync to the next level 1 element.
Luckily, the WavPack version is currently not really important; so we
fix this problem by assuming a version. David Bryant, the creator of
WavPack, recommended using version 0x410 (the most recent version) for
this. And this is what this commit does.
A FATE-test for this has been added.
Reviewed-by: David Bryant <david@wavpack.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until e7ddafd5, the Matroska muxer wrote two SeekHeads: One at the
beginning referencing the main level 1 elements (i.e. not the Clusters)
and one at the end, referencing the Clusters. This second SeekHead was
useless and has therefore been removed. Yet the SeekHead-related
functions and structures are still geared towards this usecase: They
are built around an allocated array of variable size that gets
reallocated every time an element is added to it although the maximum
number of Seek entries is a small compile-time constant, so that one should
rather include the array in the SeekHead structure itself; and said
structure should be contained in the MatroskaMuxContext instead of being
allocated separately.
The earlier code reserved space for a SeekHead with 10 entries, although
we currently write at most 6. Reducing said number implied that every
Matroska/Webm file will be 84 bytes smaller and required to adapt
several FATE tests; furthermore, the reserved amount overestimated the
amount needed for for the SeekHead's length field and how many bytes
need to be reserved to write a EBML Void element, bringing the total
reduction to 89 bytes.
This also fixes a potential segfault: If !mkv->is_live and if the
AVIOContext is initially unseekable when writing the header, the
SeekHead is already written when writing the header and this used to
free the SeekHead-related structures that have been allocated. But if
the AVIOContext happens to be seekable when writing the trailer, it will
be attempted to write the SeekHead again which will lead to segfaults
because the corresponding structures have already been freed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This fixes mpeg2video stream copies to mpeg muxer like this:
ffmpeg -i xdcamhd.mxf -c:v copy output.mpg
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Utilizes a subpicture sample with one decodable subpicture for the
test.
Based on a failing test case in reported by Michael in
https://ffmpeg.org/pipermail/ffmpeg-devel/2019-February/240398.html
which at the time had no test case for it.
Additionally, this is the first test case for the presentation
graphics format.
According to the H.264 specifications, the only NAL units that need to
have four byte startcodes in H.264 Annex B format are SPS/PPS units and
units that start a new access unit. Before af7e953a, the first of these
conditions wasn't upheld as already existing in-band parameter sets
would not automatically be written with a four byte startcode, but only
when they already were at the beginning of their input packets. But it
made four byte startcodes be used too often as every unit that is written
together with a parameter set that is inserted from extradata received a
four byte startcode although a three byte start code would suffice
unless the unit itself were a parameter set.
FATE has been updated to reflect the changes. Although the patch leaves
the extradata unchanged, the size of the extradata according to the FATE
reports changes. This is due to a quirk in ff_h2645_packet_split which
is used by extract_extradata: If the input is Annex B, the first zero of
a four byte startcode is considered a part of the last unit (if any).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The standard does not seem to require the counter to be zero based, but some
checker tools (MyriadBits MXFInspect, Interra Baton) have validations against 0
start...
Fixes ticket #6781.
Signed-off-by: Marton Balint <cus@passwd.hu>
RFC 3986 states that the generic syntax uses the slash ("/"), question mark
("?"), and number sign ("#") characters to delimit components that are
significant to the generic parser's hierarchical interpretation of an
identifier.
Signed-off-by: Marton Balint <cus@passwd.hu>
