Avoids .got entries for ff_iamf_scalable_ch_layouts and
ff_iamf_sound_system_map (whether they would have been
created otherwise depends upon the compiler and compiler
options).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This makes the wav and pcm demuxer demux bigger packets, which is more
efficient.
As a side effect of the bigger packets, audio durations can become less exact
for command lines such as "ffmpeg -i $INPUT -c:a copy -t 1.0 $OUTPUT".
Signed-off-by: Marton Balint <cus@passwd.hu>
- Remove the 1024 cap on the number of samples, for high sample rate audio it
was suboptimal, calculate the low neighbour power of two for the number of
samples (audio blocks) instead.
- Make the function work correctly also for non-pcm codecs by using the stream
bitrate to estimate the target packet size. A previous version of this patch
used av_get_audio_frame_duration2() the estimate the desired packet size, but
for some codecs that returns the duration of a single audio frame regardless
of frame_bytes.
- Fallback to 4096/block_align*block_align if bitrate is not available.
Signed-off-by: Marton Balint <cus@passwd.hu>
All versions of MSVC that support C11 (namely >= v19.27)
also support the restrict keyword, therefore av_restrict
is no longer necessary since 75697836b1.
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The samples I found all have 2000 sample packets, and by forcing the packet
size with a bsf we could automagically make muxing work for packets containing
more than 3640 samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
The 's', ';', and '<' itemtypes are used for audio and video by
Gophernicus and Gopher+.
Signed-off-by: Christian Lee Seibold <christian.seibold32@outlook.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
This muxer solely exists to test the fifo muxer via a dedicated
test tool in libavformat/tests/fifo_muxer.c. It fulfills no
other role and it is only designed with this role in mind.
The latter can be seen in two facts: The muxer uses printf
for logging and it simply presumes the packets' data to contain
a FailingMuxerPacketData (a struct duplicated in fifo_test.c
and tests/fifo_muxer.c.); in particular, it presumes packets
to have data at all, but this need not be true with side-data
only packets and a segfault can easily be triggered by e.g.
encoding flac (our native encoder sends a side-data only packet
with updated extradata at the end of encoding).
This patch fixes this by moving the test muxer into the fifo
test tool, making it inaccessible via the API (and actually
removing it from libavformat.so and libavformat.a).
While this muxer was accessible via e.g. av_guess_format(),
it was not really usable for an API user as FailingMuxerPacketData
was not public. Therefore this is not considered a breaking change.
In order to continue to use the test muxer in the test tool,
the ordinary fifo muxer had to be overridden: fifo_muxer.c
includes lavf/fifo.c but with FIFO_TEST defined which makes
it support the fifo_test muxer. This is possible because
test tools are always linked statically to their respective
library.
Reviewed-by: Stefano Sabatini <stefasab@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The formal title of the muxer according to the specification
is "RCWT (Raw Captions With Time)", so canonize this
in the long name of the codec and docs.
In the documentation section, point #2 was wrong: ccextractor
extracts the Closed Captions data and stores normalized bits
similarly to this muxer.
Signed-off-by: Marth64 <marth64@proxyid.net>
Fixes: signed integer overflow: -93000000 - 9223372036839000000 cannot be represented in type 'long'
Fixes: 64546/clusterfuzz-testcase-minimized-ffmpeg_dem_CONCAT_fuzzer-5110813828186112
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit does for AVInputFormat what commit
59c9dc82f4 did for AVOutputFormat:
It adds a new type FFInputFormat, moves all the internals
of AVInputFormat to it and adds a now reduced AVInputFormat
as first member.
This does not affect/improve extensibility of both public
or private fields for demuxers (it is still a mess due to lavd).
This is possible since 50f34172e0
(which removed the last usage of an internal field of AVInputFormat
in fftools).
(Hint: tools/probetest.c accesses the internals of FFInputFormat
as well, but given that it is a testing tool this is not considered
a problem.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Do not propagate the return value of avformat_write_header(),
as it contains the information whether the output had
already been initialized in avformat_init_output(),
but this is set generically; the return value of
FFOutputFormat.write_header is not documented at all
(and is currently ignored if >= 0), but it is more future-proof
to simply return 0 on success.
Reviewed-by: Liu Steven <lq@chinaffmpeg.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This is intended to avoid -Wformat= warnings on systems
where %s might not be supported (and also generally emitted
by GCC with -pedantic).
Reviewed-by: Liu Steven <lq@chinaffmpeg.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It is undefined behaviour to use a different type for a call
than the actual type of the function.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This separates the URL-layer adjacent parts of the code
from the parts that are also usable with custom IO.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Some encoders, like flac, propagate updated extradata at the end of encoding
as packet side data. Use it to update the relevant codec_config.
Signed-off-by: James Almer <jamrial@gmail.com>
If it's the primary item, then it's expected to be ready for presentation even
outside of the grid it belongs to.
Signed-off-by: James Almer <jamrial@gmail.com>
Rebased on top of recently merged fixes (should apply correctly now).
In merged DVD patch, -pgc and -pg options were broken. While these are
rather advanced options, they are the only means to get content for
some strangely authored discs.
Signed-off-by: Marth64 <marth64@proxyid.net>
The mpegts code historically tries to strip (the first) metadata access unit
header from synchronous KLV metadata, but the detection for such streams was
unreliable causing strips of asynchronous metadata or ID3 as well.
MISB ST 1402 specifies required stream type, stream id and registration
descriptor (which eventually maps to the codec ID) so let's use all of these
for reliable detection.
Fixes a regression caused by 468615f204.
Fixes ticket #10828, #10883.
Signed-off-by: Marton Balint <cus@passwd.hu>
The inner AVInputFormat* of the inner mpegps-demuxer
is only used once (in avformat_open_input()), so
don't even store it. In fact, just use ff_mpegps_demuxer
directly, as this demuxer has a configure dependency
on it.
Reviewed-by: Marth64 <marth64@proxyid.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Don't "return ret" even when ret is zero on success.
Reviewed-by: Marth64 <marth64@proxyid.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The wav demuxer by default tried to demux 4096-byte packets which caused
packets with very few number of samples for files with high channel count.
This caused a significant overhead especially since the latest ffmpeg.c
threading changes.
So let's use a similar approach for selecting audio frame size which is already
used in the PCM demuxer, which is to read 25 times per second but at most 1024
samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
We typically are only interesed in a single type of metadata set, so it is
better to keep them separated instead of always filtering for them.
Also use av_dynarray_add for increasing their array.
Signed-off-by: Marton Balint <cus@passwd.hu>
UUIDs do not have to be unique if their type sets them apart, so avoid using
AnyType, since we are only interested in specific types.
Signed-off-by: Marton Balint <cus@passwd.hu>
It is undefined behaviour.
Fixes many failed tests with UBSan and GCC 13 like
"src/libavformat/mov.c:4229:44: runtime error: store to address
0x5572abe20f80 with insufficient space for an object of type 'struct
MOVIndexRange'"
(The line number does not refer to the line where &entry[-1]
is assigned.)
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This reverts commit eee3b7e2fb.
It has been made in an attempt to fix UBSan test failures with
GCC 13 (see e.g. [1]), but it did not help at all. So revert it,
but use av_malloc_array() instead of going back to av_malloc().
[1]: https://fate.ffmpeg.org/report.cgi?slot=x86_64-archlinux-gcc-ubsan&time=20240226182430
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Export each tile as its own stream, and the grid information as a Stream Group
of type TILE_GRID.
This also enables exporting other stream items like thumbnails, which may be
present in non tiled HEIF images too. For those, the primary stream will be
tagged with the default disposition.
Based on a patch by Swaraj Hota
Signed-off-by: James Almer <jamrial@gmail.com>
This is the proper poll mode for waiting for an incoming connection according
to the SRT API docs.
Fixes ticket #9142.
Signed-off-by: Marton Balint <cus@passwd.hu>
The `entries` value is read directly from the stream and used to
allocate memory. This change clamps `entries` to however many are
possible in the remaining atom or file size (whichever is smallest).
Fixes https://crbug.com/1429357
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Also fixes a Clang warning:
"overlapping comparisons always evaluate to false
[-Wtautological-overlap-compare]"
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
By default the option "flv_metadata" (internally using the field
name "trust_metadata") is set to 0, meaning that we don't allocate
streams based on information in the metadata, only based on
actual streams we encounter. However the "datastream" metadata field
still would allocate a subtitle stream.
When muxing, the "datastream" field is added if either a data stream
or subtitle stream is present - but the same metadata field is used
to preemtively create a subtitle stream only. Thus, if the field
was added due to a data stream, not a subtitle stream, the demuxer
would create a stream which won't get any actual packets.
If there was such an extra, empty subtitle stream, running
avformat_find_stream_info still used to terminate within reasonable
time before 3749eede66. After that
commit, it no longer would terminate until it reaches the max
analyze duration, which is 90 seconds for flv streams (see
e6a084641a,
24fdf7334d and
f58e011a1f).
Before that commit (which removed the deprecated AVStream.codec), the
"st->codecpar->codec_id = AV_CODEC_ID_TEXT", set within the demuxer,
would get propagated into st->codec->codec_id by numerous
avcodec_parameters_to_context(st->codec, st->codecpar), then further
into st->internal->avctx->codec_id by update_stream_avctx within
read_frame_internal in libavformat/utils.c (demux.c these days).
Signed-off-by: Martin Storsjö <martin@martin.st>
if there's an audio layer with a single stream that can be rendered alone, mark it
as default. Otherwise, mark every stream as dependent.
Signed-off-by: James Almer <jamrial@gmail.com>
By using AnyType for resolving a strong reference we searched among all types,
not just the ones which can be the target of the reference, which in some cases
caused to find the wrong type, if the metadata set UUIDs were not unique.
UUIDs do not have to be unique if their type sets them apart, SMPTE 377M says:
> StrongRef: 'One to One’ relationship between sets and implemented in MXF
> with UUIDs. Strong References are typed which means that the definition
> identifies the kind of set which is the target of the reference.
Fixes ticket #10865.
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes Coverity issues #1559544 and #1559547.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Surprisingly the return value of add_param_definition()
(a pointer) has only been used to check for success
and not to actually access the pointee; nonsuccess
was equated with ENOMEM, although there is a non-enomem
error path in this function.
Change this by returning an int.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
IAMFAudioElement and IAMFMixPresentation currently contain
pointers to independently allocated objects that are sometimes
owned by said structures and sometimes not.
More precisely, upon success the demuxer transfers ownership
of these other objects newly created AVStreamGroups, but it
keeps its pointers. iamf_read_close() therefore always resets
these pointers (because the cleanup code always treats them
as ownership pointers). This leads to memory leaks in case
iamf_read_header() without having attached all of these
objects to stream groups.
The muxer has a similar issue: It also clears these pointers
(pointing to objects owned by stream groups created by the user)
in its deinit function.
This commit fixes this memleak by explicitly adding non-ownership
pointers; this also allows to remove the code to reset the
ownership pointers.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This has been allocated via av_calloc() a few lines above.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Checking whether a pointer to an element of an array is NULL
makes no sense, as the pointer addition involved in getting
the address would be undefined behaviour already if the array
were NULL.
In this case the array allocation has already been checked
a few lines before.
Fixes Coverity issue #1559548.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Fix this by postponing the allocation.
Fixes Coverity issue #1559545.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It's a read only exported option, and not meant to be set by the user.
Also, move it to MPEGTS_OPTIONS while at it to avoid duplication.
Signed-off-by: James Almer <jamrial@gmail.com>
A lot of changes and fixes to channel layout parsing, notably
- get rid of dynamic allocation of channel positions
- signal unimplemented speaker positions as unknown instead of failure, but
warn the user about it
- native order, and that a single channel only appears once was always assumed
for less than 64 channels, obviously this was incorrect
Signed-off-by: Marton Balint <cus@passwd.hu>
Makes it robust against adding fields before it, which will be useful in
following commits.
Majority of the patch generated by the following Coccinelle script:
@@
typedef AVOption;
identifier arr_name;
initializer list il;
initializer list[8] il1;
expression tail;
@@
AVOption arr_name[] = { il, { il1,
- tail
+ .unit = tail
}, ... };
with some manual changes, as the script:
* has trouble with options defined inside macros
* sometimes does not handle options under an #else branch
* sometimes swallows whitespace
The existing (and upcoming) available group types are meant to combine several
streams for presentation, with the result being treated as if it was a stream
itself.
For example, a file could export two stream groups of the same type with one of
them as the "default".
Signed-off-by: James Almer <jamrial@gmail.com>
Previously, we produced output with either \r\n or mixed line endings.
This was undesirable unto itself, but also made working with patches affecting
FATE output particularly challenging, especially via the mailing list.
Everything that consumes the SSA/ASS format is line-ending-agnostic,
so \n is selected to simplify git/ML usage in FATE.
Extra \r characters at the end of a packet are dropped. These are always
ignored by the renderer anyway.
The channel designation metadata should not override the number of channels.
Let's warn the user if it is inconsistent, and keep the channel layout
unspecified.
Before the conversion to the channel layout API the code only set the mask, but
never overridden the channel count, so this restores the old behaviour.
Signed-off-by: Marton Balint <cus@passwd.hu>
Existing code could have caused wrong channel order signalling or reduced
channel count if a channel designation appeared multiple times. This is
actually an old bug, but the conversion to the new channel layout API made it
visible, because now the code overrides the proper channel count with the one
calculated from the mask.
Signed-off-by: Marton Balint <cus@passwd.hu>
The demuxer opens an internal parser instance in read_timestamp(), which
requires a codec context. There is no need for it to access the FFStream
one which is used for other purposes, it can allocate its own internal
one.
This check has survived the transition to AVCodecParameters, but is no
longer relevant after it, since the codec context is no longer updated
or accessed at all from the demuxer.
It does not use the AVFormatContext at all.
Reviewed-by: Marth64 <marth64@proxyid.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Resetting the counter of used elements is enough as nothing is
ever read from the currently unused elements.
Reviewed-by: Marth64 <marth64@proxyid.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The rcwt muxer uses several counters for how much data
it has already cached: One byte counter and one counter
for how many complete blocks (of three bytes each).
These counters can become inconsistent when the muxer is
fed incomplete blocks as the muxer presumes that it is
about to write a new block at the start of each write_packet
call. E.g. sending 65535*3+1 1-byte packets (with data[0] e.g. 0x03)
will trigger an out-of-bounds write.
This patch fixes this by processing the data in complete blocks
only. This also allows to simplify the code, e.g. to remove one of
the counters.
Reviewed-by: Marth64 <marth64@proxyid.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The muxer's AVCodecContext is currently used for exactly one thing:
To store a time base in it that has been derived via heuristics
in avformat_transfer_internal_stream_timing_info(); said time base
can then be read back via av_stream_get_codec_timebase().
But one does not need a whole AVCodecContext for that, a simple
AVRational is enough.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
For muxers, the internal AVCodecContext is basically unused
except in avformat_transfer_internal_stream_timing_info()
(which sets time_base and ticks_per_frame) and
av_stream_get_codec_timebase() (a getter for time_base).
This makes ticks_per_frame write-only, so don't set it.
Also remove an always-false check for the AVCodecContext's
codec_tag.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Currently, when writing PCMA or PCMU tracks with FLV or RTMP, the
stereo flag and sample rate flag inside RTMP audio messages are
overridden, making impossible to distinguish between mono and stereo
tracks. This patch fixes the issue by restoring the same flag mechanism
of all other codecs, that takes into consideration the right channel
count and sample rate.
Signed-off-by: Alessandro Ros <aler9.dev@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
There are 6 deprecated ISO language codes that are still valid for DVDs.
This patch allows avlanguage to recognize them correctly. The codes are:
(1) "in" - legacy code for Indonesian, mapped to the modern code
(2) "iw" - legacy code for Hebrew, mapped to the modern code
(3) "ji" - legacy code for Yiddish, mapped to the modern code
(4) "jw" - legacy code for Javanese, published and used as a typoed version of "jv"
(5) "mo" - legacy code for Moldavian, mapped to the inclusive code
(6) "sh" - legacy code for Serbo-Croatian, no modern inclusive code so it is left alone
All of this can be verified from several sources including:
https://en.wikipedia.org/wiki/List_of_ISO_639_language_codes
Signed-off-by: Marth64 <marth64@proxyid.net>
Use avio_get_dyn_buf()+ffio_free_dyn_buf() instead of
avio_close_dyn_buf()+av_free(). This saves an allocation
(and memcpy) in case all the data fits in the AVIOContext's
write buffer.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Fixes a regression since d9fed9df2a, where the single animated stream would
be exported twice as two independent streams.
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes segfaults when trying to map a group index with a value equal to the
amount of groups in the file.
Signed-off-by: James Almer <jamrial@gmail.com>
Probably an artifact of a rebase, as this check is done below.
Fixes "Conditional jump or move depends on uninitialised value(s)" errors as
reported by Valgrind.
Signed-off-by: James Almer <jamrial@gmail.com>
The AVIAMFAudioElement and AVIAMFMixPresentation that are ultimately used
are allocated by ff_iamfdec_read_descriptors().
Fixes some memory leaks reported by Valgrind.
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes server compatibility issues with rtspclientsink GStreamer plugin.
>From specification:
RFC 7826 "Real-Time Streaming Protocol Version 2.0" (https://datatracker.ietf.org/doc/html/rfc7826), section 18.54:
mode: The mode parameter indicates the methods to be supported for
this session. The currently defined valid value is "PLAY". If
not provided, the default is "PLAY". The "RECORD" value was
defined in RFC 2326; in this specification, it is unspecified
but reserved. RECORD and other values may be specified in the
future.
RFC 2326 "Real Time Streaming Protocol (RTSP)" (https://datatracker.ietf.org/doc/html/rfc2326), section 12.39:
mode:
The mode parameter indicates the methods to be supported for
this session. Valid values are PLAY and RECORD. If not
provided, the default is PLAY.
mode=receive was always like this, from the initial commit 'a8ad6ffa rtsp: Add listen mode'.
For comparison, Wowza was used to push RTSP stream to. Both GStreamer and FFmpeg had no issues.
Here is the capture of Wowza responding to SETUP request:
200 OK
CSeq: 3
Server: Wowza Streaming Engine 4.8.26+4 build20231212155517
Cache-Control: no-cache
Expires: Mon, 15 Jan 2024 19:40:31 GMT
Transport: RTP/AVP/UDP;unicast;client_port=11640-11641;mode=record;source=172.17.0.2;server_port=6976-6977
Date: Mon, 15 Jan 2024 19:40:31 GMT
Session: 1401457689;timeout=60
Test setup:
Server: ffmpeg -loglevel trace -y -rtsp_flags listen -i rtsp://0.0.0.0:30800/live.stream t.mp4
FFmpeg client: ffmpeg -re -i "Big Buck Bunny - FULL HD 30FPS.mp4" -c:v libx264 -f rtsp rtsp://127.0.0.1:30800/live.stream
GStreamer client: gst-launch-1.0 videotestsrc is-live=true pattern=smpte ! queue ! videorate ! videoscale ! video/x-raw,width=640,height=360,framerate=60/1 ! timeoverlay font-desc="Sans, 84" halignment=center valignment=center ! queue ! videoconvert ! tee name=t t. ! x264enc bitrate=9000 pass=cbr speed-preset=ultrafast byte-stream=false key-int-max=15 threads=1 ! video/x-h264,profile=baseline ! queue ! rsink. audiotestsrc ! voaacenc ! queue ! rsink. t. ! queue ! autovideosink rtspclientsink name=rsink location=rtsp://localhost:30800/live.stream
Test results:
modified FFmpeg client -> stock server : ok
stock FFmpeg client -> modified server : ok
modified FFmpeg client -> modified server : ok
GStreamer client -> modified server : ok
Signed-off-by: Paul Orlyk <paul.orlyk@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes server compatibility issues with rtspclientsink GStreamer plugin
Signed-off-by: Paul Orlyk <paul.orlyk@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Parse iprp and iinf boxes and its child boxes to get the actual codec used
(AV1 for avif, HEVC for heic), and properly export extradata and other
properties in a generic way.
The avif tests reference files are updated as the extradata is now exported.
Based on a patch by Swaraj Hota
Co-authored-by: Swaraj Hota <swarajhota353@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Marth64 <marth64@proxyid.net>
Raw Captions With Time (RCWT) is a format native to ccextractor, a commonly
used open source tool for processing 608/708 closed caption (CC) sources.
It can be used to archive the original, raw CC bitstream and to produce
a source file file for later CC processing or conversion. As a result,
it also allows for interopability with ccextractor for processing CC data
extracted via ffmpeg. The format is simple to parse and can be used
to retain all lines and variants of CC.
A free specification of RCWT can be found here:
https://github.com/CCExtractor/ccextractor/blob/master/docs/BINARY_FILE_FORMAT.TXT
This muxer implements the specification as of 01/05/2024, which has
been stable and unchanged for 10 years as of this writing.
This muxer will have some nuances from the way that ccextractor muxes RCWT.
No compatibility issues when processing the output with ccextractor
have been observed as a result of this so far, but mileage may vary
and outputs will not be a bit-exact match.
Specifically, the differences are:
(1) This muxer will identify as "FF" as the writing program identifier, so
as to be honest about the output's origin.
(2) ffmpeg's MPEG-1/2, H264, HEVC, etc. decoders extract closed captioning
data differently than ccextractor from embedded SEI/user data.
For example, DVD captioning bytes will be translated to ATSC A53 format.
This allows ffmpeg to handle 608/708 in a consistant way downstream.
This is a lossless conversion and the meaningful data is retained.
(3) This muxer will not alter the extracted data except to remove invalid
packets in between valid CC blocks. On the other hand, ccextractor
will by default remove mid-stream padding, and add padding at the end
of the stream (in order to convey the end time of the source video).
This second patch fixes the following error at the end of a .STR stream conversion:
[in#0/psxstr @ 0000000000681e80] Error during demuxing: I/O error
It's been a bit of trial and error as I've never used ffmpeg, but returning AVERROR_EOF appears to be the way to go (doesn't complain anymore).
Signed-off-by: aybe <aybe@users.noreply.github.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This third patch fixes warnings that are false positives (still on STRv1).
That's because these sectors are simply empty ones as can be read in "System Description CD-ROM XA, May 1991,
4.3.2.3".
Haven't attempted significant refactoring as it just works, left a comment instead about the situation.
The result is that there are no more false warnings when converting.
Signed-off-by: aybe <aybe@users.noreply.github.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
They are similar to AVIF images (both use the HEIF container).
The only additional work needed is to parse the hvcC box and put
it in the extradata.
With this patch applied, ffmpeg (when built with an HEVC decoder)
is able to decode the files in
https://github.com/nokiatech/heif/tree/gh-pages/content/images
Also add a couple of fate tests with samples from
https://github.com/nokiatech/heif_conformance/tree/master/conformance_files
Partially fixes trac ticket #6521.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The gcrypt definition of `bn_new` used to use the return statement
on errors, with an AVERROR return value, regardless of the signature
of the function where the macro is used - it is called in
`dh_generate_key` and `ff_dh_init` which return pointers. As a result,
compiling with gcrypt and the ffrtmpcrypt protocol resulted in an
int-conversion warning. GCC 14 may upgrade these to errors [1].
This patch fixes the problem by changing the macro to remove `AVERROR`
and instead set `bn` to null if the allocation fails. This is the
behaviour of all the other `bn_new` implementations and so the result is
already checked at all the callsites. AFAICT, this should be the only
change needed to get ffmpeg off Fedora's naughty list of projects with
warnings which may be upgraded to errors in GCC 14 [2].
[1]: https://gcc.gnu.org/pipermail/gcc/2023-May/241264.html
[2]: https://www.mail-archive.com/devel@lists.fedoraproject.org/msg196024.html
Signed-off-by: Frank Plowman <post@frankplowman.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This patch populates the third entry for HLS codec attribute using the
AAC profile.
The HLS specifications[1] require this value to be the Object Type ID as
referred to in table 1.3 of ISO/IEC 14496-3:2009[2].
The numerical constants in the code refer to these OTIs minus one, as
documented in commit 372597e[3], confirmed by comparing the values in the
code with the values in the table mentioned above.
Links:
1: https://datatracker.ietf.org/doc/html/rfc6381#section-3.3
2: https://csclub.uwaterloo.ca/~ehashman/ISO14496-3-2009.pdf
3: 372597e538
Changes in this version:
- Default value set to "mp4a.40.2" when profile is unknown for backward
compatibility.
Signed-off-by: Steven Liu <liuqi05@kuaishou.com>
Fixes: signed integer overflow: 2314885530818453536 - -7412889664301817824 cannot be represented in type 'long'
Fixes: 64296/clusterfuzz-testcase-minimized-ffmpeg_dem_MOV_fuzzer-6304027146846208
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Unnecessary since acf63d5350adeae551d412db699f8ca03f7e76b9;
also avoids relocations.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Reduce false positives for VVC files by adding additional checks in
`vvc_probe`. Specifically, `nuh_temporal_id_plus1` is tested for valid
values in extra cases depending on the NAL unit type, as per ITU-T H.266
section 7.4.2.2.
Resolves trac #10703.
Signed-off-by: Frank Plowman <post@frankplowman.com>
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
Avoids a -Wstringop-truncation warning by using av_strlcopy instead of
strncpy. Additionally, prints a warning to the log context if this
truncation occurred.
Signed-off-by: Leo Izen <leo.izen@gmail.com>
The earlier code writes the file and then tries to patch up
the size later. This is avoidable for the common case of
a single image because one can know the complete size
in advance and write it.
Fixes ticket #4609.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This reverts commit bec6dfcd5c.
The patch is NOP since ffurl_open_whitelist copy options from parent
automatically.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
When encoders don't support global header like MediaCodec, FLV
muxer needs to add extract_extradata bsf automatically. The codec
list doesn't include VP9 since it's not supported by
extract_extradata.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
It is common for subtitle streams to have large gaps between packets.
When the caller is interleaving packets from multiple files, it can
easily happen that two successive subtitle packets trigger this limit,
even though no excessive buffering is happening.
Should fix#7064
Andreas Rheinhardt