We don't need to print the tags here because they're added as dict
elements to AVFrame->metadata and are printed elsewhere with ffprobe
-show_frames.
Signed-off-by: Leo Izen <leo.izen@gmail.com>
This commit will cause TIFF files to store their tags in the EXIF
struct so tags such as orientation can be transfered to other formats
(such as PNG) in a way that doesn't corrupt the IFD.
Signed-off-by: Leo Izen <leo.izen@gmail.com>
Add support to write EXIF profiles using the new EXIF framework, namely
ff_exif_get_buffer, and writing them into eXIf chunks.
Signed-off-by: Leo Izen <leo.izen@gmail.com>
This commit adds a structure to contain parsed EXIF metadata, as well
as code to read and write that struct from/to binary EXIF buffers. Some
internal functions have been moved to exif_internal.h. Code to read
from this new struct and write to an AVDictionary **dict has been added
as well in order to preserve interoperability with existing callers.
The only codec changes so far as of this commit are to call these
interop functions, but in future commits there will be codec changes to
use the new parsing routines instead.
Signed-off-by: Leo Izen <leo.izen@gmail.com>
cbs.mak is meant to contain tests strictly for the CBS framework, not for any
bsf that happens to use it under the hood.
Signed-off-by: James Almer <jamrial@gmail.com>
Previously, these tests failed when running on Windows, if the
system is configured with a time zone east of Greenwich, i.e.
with a positive GMT offset.
The muxer converts the creation_date given by the user using
av_parse_time to unix time, as a time_t. The creation_date is
interpreted as a local time, i.e. according to the current time
zone. (This time_t value is then converted back to a broken out
local time form with localtime_r.)
The given reference date/time, "1970-01-01T00:00:00", is the
origin point for unix time, corresponding to time_t zero. However
when interpreted as local time, this doesn't map to exactly zero.
Time zones east of Greenwich reached this time a number of hours
before the point of zero time_t - so the corresponding time_t
value essentially is minus the GMT offset, in seconds.
Windows mktime returns an error, returning (time_t)-1, when given
such a "struct tm", while e.g. glibc mktime happily returns a
negative time_t. av_parse_time doesn't check the return value of
mktime for potential errors.
This is observable with the following test snippet:
struct tm tm = { 0 };
tm.tm_year = 70;
tm.tm_isdst = -1;
tm.tm_mday = 1;
tm.tm_hour = 0;
time_t t = mktime(&tm);
printf("%d-%02d-%02d %02d:%02d:%02d\n", tm.tm_year + 1900, tm.tm_mon + 1, tm.tm_mday, tm.tm_hour, tm.tm_min, tm.tm_sec);
printf("t %d\n", (int)t);
By varying the value of tm_hour and the system time zone, one
can observe that Windows mktime returns -1 for all time_t values
that would have been negative.
This range limit is also documented by Microsoft in detail at
https://learn.microsoft.com/en-us/cpp/c-runtime-library/reference/mktime-mktime32-mktime64.
To avoid the issue, pick a different, arbitrary reference time,
which should have a nonnegative time_t for all time zones.
Don't overwrite the bitstream values when updating the top-level loop
filter and segmentation state, instead do the update separately at the
end of the frame parsing.
This also reverts the change to the passthrough tests which made them
have output not matching the input.
Initially, avcodec/srtenc.c was outputting CRLF [1]. Later, a real SRT
muxer was added [2], which outputs LF. The original srtenc.c was
converted to use the muxer [3], changing its output to LF, except for
newline characters within subtitle text.
Fix this to avoid producing SRT files with mixed line endings.
[1] 8e43b6fed9
[2] 9e63c30daa
[3] 55180b3299
Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
This was introduced in commit 9c43703, to support a codec "extension"
in the prores_aw encoder.
This removes the chroma fill loop, and instead performs the inverse
transform on null coefficients, which achieves the same result and
fixes an off-by-one in the chroma values produced.
Updated test to reflect this change.
The first sample in the stsc box may not refer to the first stsd entry.
This is the case in h264/thezerotheorem-cut.mp4, and as such the
fate-h264_redundant_pps-side_data test is updated accordingly.
Signed-off-by: James Almer <jamrial@gmail.com>
The new logic should be easier to follow.
It also uses ff_inlink_consume_frame() for all simple passthrough operations
making custom get_audio_buffer callback unnecessary.
Fate changes are because the new logic does not repacketize input audio up
until the crossfade. Content is the same.
Signed-off-by: Marton Balint <cus@passwd.hu>
- Changes in mov_write_video_tag function to handle APV elementary stream
- Provided structure APVDecoderConfigurationRecord that specifies the decoder configuration information for APV video content
Co-Authored-by: James Almer <jamrial@gmail.com>
Signed-off-by: Dawid Kozinski <d.kozinski@samsung.com>
Signed-off-by: James Almer <jamrial@gmail.com>
A frame graph activation might not produce a frame in the requested sink, so
keep on requesting a frame there unless we encounter a filter activation with
buffersrc empty error.
This makes av_buffersink_get_frame(_flags) work according to its documentation
which claims that EAGAIN is only returned if additional frames must be inserted
into the graph.
Fate changes are because audio frames will have different sizes at segment
boundaries, but content is the same.
Signed-off-by: Marton Balint <cus@passwd.hu>
Before this patch, the DFPWM1a encoder was marked as supporting
variable frame sizes. The DFPWM1a format converts eight bytes
of input into one output byte and so it simply padded the number
of data output by
frame->nb_samples * frame->ch_layout.nb_channels / 8 +
(frame->nb_samples % 8 > 0 ? 1 : 0)
This has several bugs:
a) The additional byte leads to eight additional input byte being
read; this can read into the frame's padding, i.e. the data can
be uninitialized.
b) The criterion for whether one should pad is wrong:
nb_samples * nb_channels should be tested for divisibility by eight.
c) The created frames can be undecodable (at least with our decoder):
Our decoder requires the number of bits per frame to divisible by
the number of channels, yet the above approach does not guarantee this.
d) The padding will be added in the middle of the stream (potentially
for every packet).
This commit fixes all of this by removing the variable frame size cap
and using AVCodecInternal.pad_samples to pad the last frame so that
nb_samples * nb_channels is always a multiple of eight.
The lavf-dfpwm FATE-test was affected by a). The frames originated from
lavfi and were part of an audio frame pool, so that the padding
contained data from an earlier (bigger) frame. Now the last frame is
properly filled with silence.
Reported-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The way streams are coded in an IAMF struct follows a scalable model where the
channel layouts for each layer may not match the channel order our API can
represent in a Native order layout.
For example, an audio element may have six coded streams in the form of two
stereo streams, followed by two mono streams, and then by another two stereo
streams, for a total of 10 channels, and define for them four scalable layers
with loudspeaker_layout values "Stereo", "5.1ch", "5.1.2ch", and "5.1.4ch".
The first layer references the first stream, and each following layer will
reference all previous streams plus extra ones.
In this case, the "5.1ch" layer will reference four streams (the first two
stereo and the two mono) to encompass six channels, which does not match out
native layout 5.1(side) given that FC and LFE come after FL+FR but before
SL+SR, and here, they are at the end.
For this reason, we need to build Custom order layouts that properly represent
what we're exporting.
----
Before:
Stream group #0:0[0x12c]: IAMF Audio Element:
Layer 0: stereo
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Layer 1: 5.1(side)
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Layer 2: 5.1.2
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:4[0x4]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Layer 3: 5.1.4
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:4[0x4]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:5[0x5]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
----
AFter:
Stream group #0:0[0x12c]: IAMF Audio Element:
Layer 0: stereo
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Layer 1: 6 channels (FL+FR+SL+SR+FC+LFE)
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Layer 2: 8 channels (FL+FR+SL+SR+FC+LFE+TFL+TFR)
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:4[0x4]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Layer 3: 10 channels (FL+FR+SL+SR+FC+LFE+TFL+TFR+TBL+TBR)
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:4[0x4]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:5[0x5]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Signed-off-by: James Almer <jamrial@gmail.com>
Follows the proper order defined by the spec, even if mostly cosmetic, and is
also preparation for a following change.
Signed-off-by: James Almer <jamrial@gmail.com>
Tests both the Media 100 decoder (using the media100_to_mjpegb BSF
implicitly) as well as using said BSF, followed by the MJPEGB decoder.
(We currently hit a bug when remuxing: The demuxer treats compressorname
as encoded in a Mac character encoding (Mac OS Roman?) and converts
it to UTF-8, yet the muxer just writes it.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This allows detecting changes and regressions in side data related code, same as what
framecrc does for before already for packet data itself.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
As described in section F.6.1 from ETSI TS 102 366.
Found-by: nyanmisaka
Reviewed-by: Baptiste Coudurier <baptiste.coudurier@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Improves compatibility with Resolume products by adding an additional
hashtable for DXT color+LUT combinations, and padding the DXT texture
dimensions to the next largest multiple of 16. Produces identical
packets to Resolume Alley in manual tests.
Signed-off-by: Emma Worley <emma@emma.gg>
Up until now, our muxer wrote FFV1 in video-for-windows
compatibility mode out of concern for old demuxers that
only support that (whereas the demuxer accepts V_FFV1).
This commit switches to using native mode, because
a) V_FFV1 is around long enough so that old demuxers
should not be an issue (support in FFmpeg has been added
in commit 9ae762da7e
in March 2017/FFmpeg 3.3),
b) using native mode uses fewer bytes for the CodecPrivate,
c) the VfW extradata is zero-padded to an even length
if necessary, but our demuxer forgot to undo the padding
until very recently (92e310eb82),
so that there are many versions of our demuxer around that
are buggy wrt VFW, but not V_FFV1.
This affects the FFV1 extradata checksums, specifically
the (experimental) version 4 files with error check version 2*
as created by
ffmpeg -i ../fate-suite/mpeg2/sony-ct3.bs -c:v ffv1 \
-slices 16 -frames 1 -level 4 -strict experimental ffv1.mkv
VFW files like the above created by this muxer before this patch
would not work with an old demuxer.
*: Without error check version 2, the CRC for the whole extradata
is zero, which is not changed by appending a zero byte.
Reviewed-by: compn <ff@hawaiiantel.net>
Reviewed-by: Dave Rice <dave@dericed.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
