The LC3 audio codec is the default codec of Bluetooth LE audio.
This is a wrapper over the liblc3 library (https://github.com/google/liblc3).
Signed-off-by: Antoine Soulier <asoulier@google.com>
* SMPTE ST 2128 IPT-C2 defines the coefficients utilized in DoVi
Profile 5. Profile 5 can thus now be represented in VUI as
{AVCOL_RANGE_JPEG, AVCOL_PRI_BT2020, AVCOL_TRC_SMPTE2084,
AVCOL_SPC_IPT_C2, AVCHROMA_LOC_LEFT} (although other chroma
sample locations are allowed). AVCOL_TRC_SMPTE2084 should in
this case be interpreted as 'PQ with reshaping'.
* YCgCo-Re and YCgCo-Ro define the bitexact YCgCo-R, where the
number of bits added to a source RGB bit depth is 2 (i.e., even)
and 1 (i.e., odd), respectively.
As well as accessors plus a function for allocating this struct with
extension blocks,
Definitions generously taken from quietvoid/dovi_tool, which is
assembled as a collection of various patent fragments, as well as output
by the official Dolby Vision bitstream verifier tool.
The NLQ pivots are not documented but should be present in the header
for profile 7 RPU format. It has been verified using Dolby's
verification toolkit.
Signed-off-by: quietvoid <tcChlisop0@gmail.com>
Signed-off-by: Niklas Haas <git@haasn.dev>
RCWT (Raw Captions With Time) is a format native to ccextractor,
a commonly used OSS tool for processing 608/708 Closed Captions (CC).
RCWT can be used to archive the original extracted CC bitstream.
The muxer was added in January 2024. In this commit, add the demuxer.
One can now demux RCWT files for rendering in ccaption_dec or interop
with ccextractor (which produces RCWT). Using the muxer/demuxer combo,
the CC bits can be kept for processing or rendering with either tool.
This can be an effective way to backup an original CC stream, including
format extensions like EIA-708 and overall original presentation.
Signed-off-by: Marth64 <marth64@proxyid.net>
Yet another probesize used to get the durations when
estimate_timings_from_pts is required. It is aimed at users interested
in better durations probing for itself, or because using
avformat_find_stream_info indirectly and requiring exact values: for
concatdec for example, especially if streamcopying above it.
The current code is a performance trade-off that can fail to get video
stream durations in a scenario with high bitrates and buffering for
files ending cleanly (as opposed to live captures): the physical gap
between the last video packet and the last audio packet is very high in
such a case.
Default behaviour is unchanged: 250k up to 250k << 6 (step by step).
Setting this new option has two effects:
- override the maximum probesize (currently 250k << 6)
- reduce the number of steps to 1 instead of 6, this is to avoid
detecting the audio "too early" and failing to reach a video packet.
Even if a single audio stream duration is found but not the other
audio/video stream durations, there will be a retry, so at the end the
full user-overriden probesize will be used as expected by the user.
Signed-off-by: Nicolas Gaullier <nicolas.gaullier@cji.paris>
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.
Keep it for external users in order to not cause breakages.
Also improve the other headers a bit while just at it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Depending on the filters used, the filtergraph may produce trailing data
after feeding it the last input frame. Update the example to include the
necessary loop for draining the filtergraph.
Reviewed-by: Stefano Sabatini <stefasab@gmail.com>
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
Depending on the filters used, the filtergraph may produce trailing data
after feeding it the last input frame. Update the example to include the
necessary loop for draining the filtergraph.
Reviewed-by: Stefano Sabatini <stefasab@gmail.com>
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
av_ts_make_time_string() used "%.6g" format, but this format was losing
precision even when the timestamp to be printed was not that large. For example
for 3 hours (10800) seconds, only 1 decimal digit was printed, which made this
format inaccurate when it was used in e.g. the silencedetect filter. Other
detection filters printing timestamps had similar issues. Also time base
parameter of the function was *AVRational instead of AVRational.
Resolve these problems by introducing a new function, av_ts_make_time_string2().
We change the used format to "%.*f", use a precision of 6, except when printing
values near 0, in which case we calculate the precision dynamically to aim for
a similar precision in normal form as with %.6g. No longer using scientific
representation can make parsing the timestamp easier for the users, we can
safely do this because the theoretical maximum of INT64_MAX*INT32_MAX still
fits into the string buffer in normal form.
We somewhat imitate %g by trimming ending zeroes and the potential decimal
point characters. In order not to trim "inf" as well, we assume that the
decimal point string does not contain the letter "f". Note that depending on
printf %f implementation, we might trim "infinity" to "inf".
Thanks for Allan Cady for bringing up this issue.
Signed-off-by: Marton Balint <cus@passwd.hu>
Common utility function that can be used by all codecs to select the
right (any valid) film grain parameter set. In particular, this is
useful for AFGS1, which has support for multiple parameters.
However, it also performs parameter validation for H274.
This is needed for AV1 film grain as well, when using AFGS1 streams.
Also add extra width/height and subsampling information, which AFGS1
cares about, as part of the same API bump. (And in principle, H274
should also expose this information, since it is needed downstream to
correctly adjust the chroma grain frequency to the subsampling ratio)
Deprecate the equivalent H274-exclusive fields. To avoid breaking ABI,
add the new fields after the union; but with enough of a paper trail to
hopefully re-order them on the next bump.
This will allow users to pass the Android ApplicationContext which is mandatory
to retrieve the ContentResolver responsible to resolve/open Android content URIS.
av_frame_side_data_get() has a const AVFrameSideData * const *sd
parameter; so calling it with an AVFramesSideData **sd like
AVCodecContext.decoded_side_data (or with a AVFramesSideData * const
*sd) is safe, but the conversion is not performed automatically
in C. All users of this function therefore resort to a cast.
This commit changes this: av_frame_side_data_get() is renamed
to av_frame_side_data_get_c(); furthermore, a static inline
wrapper for it name av_frame_side_data_get() is added
that accepts an AVFramesSideData * const * and converts this
to const AVFramesSideData * const * in a Wcast-qual safe way.
This also allows to remove the casts from the current users.
Reviewed-by: Jan Ekström <jeebjp@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The documentation correctly states that the rdiv is a multiplier but incorrectly states the default behavior is to multiply by the sum of all matrix elements - it multiplies by 1/sum.
This changes the documentation to match the code.
Address trac #10889
Signed-off-by: Marton Balint <cus@passwd.hu>
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 allocators have been superseded by av_vdpau_bind_context().
The latter have only been added "to allow multiple forks to add
fields to the structure without breaking ABI" [1], but libav
is no more, so this is not needed any longer.
[1]: https://ffmpeg.org/pipermail/ffmpeg-devel/2013-August/146954.html
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Frame-level side data attributes are printed with the same key/value
structure as packet-level side data attributes, but this is not
reflected in the XSD.
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>
This BSF is supposed to be used in conjunction with mp3_header_compress,
which has been removed more than ten years ago in commit
c6080d8900. It mangled the headers
by removing the CRC field as well as fields that are supposed
to stay constant for the entirety of a stream (which are put into
extradata). This made these files unplayable; they need to be
decompressed with the BSF first (which does not happen automatically).
Even in this case the CRC does not get restored.
I am not aware that such compressed files exist at all; therefore
this commit removes the BSF completely.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
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>
It used to be used with preallocated packet buffers with
the old encode API, but said API is no more and therefore
there is no reason for this to be public any more.
So deprecate it and use an internal replacement
for the encoders using it as an upper bound for the
size of their headers.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It should be available in all relevant modern compilers and will allow
us to use features like anonymous unions.
Note that stdatomic.h is still emulated on MSVC, as current versions
require the /experimental:c11atomics, and do not support
ATOMIC_VAR_INIT() anyway.
This exposes VP8E_SET_SCREEN_CONTENT_MODE option from libvpx.
Co-authored-by: Erik Språng <sprang@webrtc.org>
Signed-off-by: Dariusz Marcinkiewicz <darekm@google.com>
Signed-off-by: James Zern <jzern@google.com>
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>
It implements BSD-specific support for very old analog capture cards,
which are highly unlikely to be useful today. After being added in 2005,
there were never any commits to it beyond compilation fixes and generic
maintenance. There have also been zero trac tickets for this device, and
the only related web search result I found concludes that it does not
work.
The code also does some unacceptable things, like messing with signal
handlers and storing its state in global variables.
Sort options by name, review formatting, apply consistency fixes and
fill the gaps (e.g. missing value for constants or flags), and review
and extend content.
The C library function double atan2(double y, double x) takes y as the first
parameter and x as the second parameter.
Signed-off-by: Haixia Shi <hshi@meta.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 was accidentally removed in a prior revision of the series,
alongside the corresponding (separate) version bump. Instead coalesce it
into the follow-up commit's entry, since that's the lowest version
actually supporting the new fields.
Move section to the top of the file, use table in place of subsection
to list the comprising muxers, and show media type information and
extensions in the item entry names.
