

To celebrate, Dascha (daschasara) answered FFmpeg's call for non-AI artwork and drew this piece for us:

Artwork by Dascha (daschasara)
]]>This release features a lot of internal changes and bugfixes. The groundwork for the upcoming swscale rewrite is progressing. The Vulkan compute-based codecs, and a few filters, no longer depend on runtime GLSL compilation, which speeds up their initialization.
A companion post about the Vulkan Compute-based codec implementations has been published on the Khronos blog, featuring technical details on the implementations and future plans.
We recommend users, distributors, and system integrators to upgrade unless they use current git master.
]]>
A new class of decoders and encoders based on pure Vulkan compute implementation have been added.
Vulkan is a cross-platform, open standard set of APIs that allows programs to use GPU hardware in various ways,
from drawing on screen, to doing calculations, to decoding video via custom hardware accelerators.
Rather than using a custom hardware accelerator present, these codecs are based on compute shaders, and work
on any implementation of Vulkan 1.3.
Decoders use the same hwaccel API and commands, so users do not need to do anything special to enable them,
as enabling Vulkan decoding is sufficient to use them.
Encoders, like our hardware accelerated encoders, require specifying a new encoder (ffv1_vulkan).
Currently, the only codecs supported are: FFv1 (encoding and decoding) and ProRes RAW (decode only).
ProRes (encode+decode) and VC-2 (encode+decode) implementations are complete and currently in review,
to be merged soon and available with the next minor release.
Only codecs specifically designed for parallelized decoding can be implemented in such a way, with
more mainstream codecs not being planned for support.
Depending on the hardware, these new codecs can provide very significant speedups, and open up
possibilities to work with them for situations like non-linear video editors and
lossless screen recording/streaming, so we are excited to learn what our downstream users can make with them.
The project has recently started to modernize its infrastructure. Our mailing list servers have been fully upgraded, and we have recently started to accept contributions via a new forge, available on code.ffmpeg.org, running a Forgejo instance.
As usual, we recommend that users, distributors, and system integrators to upgrade unless they use current git master.
]]>
The more important highlights of the release are that the VVC decoder, merged as experimental in version 7.0,
has had enough time to mature and be optimized enough to be declared as stable. The codec is starting to gain
traction with broadcast standardization bodies.
Support has been added for a native AAC USAC (part of the xHE-AAC coding system) decoder, with the format starting
to be adopted by streaming websites, due to its extensive volume normalization metadata.
MV-HEVC decoding is now supported. This is a stereoscopic coding tool that begun to be shipped and generated
by recent phones and VR headsets.
LC-EVC decoding, an enhancement metadata layer to attempt to improve the quality of codecs, is now supported via an
external library.
Support for Vulkan encoding, with H264 and HEVC was merged. This finally allows fully Vulkan-based decode-filter-encode pipelines, by having a sink for Vulkan frames, other than downloading or displaying them. The encoders have feature-parity with their VAAPI implementation counterparts. Khronos has announced that support for AV1 encoding is also coming soon to Vulkan, and FFmpeg is aiming to have day-one support.
In addition to the above, this release has had a lot of important internal work done. By far, the standout internally
are the improvements made for full-range images. Previously, color range data had two paths, no negotiation,
and was unreliably forwarded to filters, encoders, muxers. Work on cleaning the system up started more than 10
years ago, however this stalled due to how fragile the system was, and that breaking behaviour would be unacceptable.
The new system fixes this, so now color range is forwarded correctly and consistently everywhere needed, and also
laid the path for more advanced forms of negotiation.
Cropping metadata is now supported with Matroska and MP4 formats. This metadata is important not only for archival,
but also with AV1, as hardware encoders require its signalling due to the codec not natively supporting one.
As usual, we recommend that users, distributors, and system integrators to upgrade unless they use current git master.
]]>ffmpeg CLI tool.
This release is not backwards compatible, removing APIs deprecated before 6.0.
The biggest change for most library callers will be the removal of the old bitmask-based
channel layout API, replaced by the AVChannelLayout API allowing such
features as custom channel ordering, or Ambisonics. Certain deprecated ffmpeg
CLI options were also removed, and a C11-compliant compiler is now required to build
the code.
As usual, there is also a number of new supported formats and codecs, new filters, APIs,
and countless smaller features and bugfixes. Compared to 6.1, the git repository
contains almost ∼2000 new commits by ∼100 authors, touching >100000 lines in
∼2000 files — thanks to everyone who contributed. See the
Changelog,
APIchanges,
and the git log for more comprehensive lists of changes.
libavcodec library now contains a native VVC (Versatile Video Coding)
decoder, supporting a large subset of the codec's features. Further optimizations and
support for more features are coming soon. The code was written by Nuo Mi, Xu Mu,
Frank Plowman, Shaun Loo, and Wu Jianhua.
]]>libavformat library can now read and write IAMF
(Immersive Audio) files. The ffmpeg CLI tool can configure IAMF structure with the new
-stream_group option. IAMF support was written by James Almer.
]]>ffmpeg CLI toolffmpeg command-line tool, all the major
components of the transcoding pipeline (demuxers, decoders, filters, encodes, muxers) now
run in parallel. This should improve throughput and CPU utilization, decrease latency,
and open the way to other exciting new features.
Note that you should not expect significant performance improvements in cases where almost all computational time is spent in a single component (typically video encoding).
]]>This release had been overdue for at least half a year, but due to constant activity in the repository, had to be delayed, and we were finally able to branch off the release recently, before some of the large changes scheduled for 7.0 were merged.
Internally, we have had a number of changes too. The FFT, MDCT, DCT and DST implementation used for codecs
and filters has been fully replaced with the faster libavutil/tx (full article about it coming soon).
This also led to a reduction in the the size of the compiled binary, which can be noticeable in small builds.
There was a very large reduction in the total amount of allocations being done on each frame throughout video decoders,
reducing overhead.
RISC-V optimizations for many parts of our DSP code have been merged, with mainly the large decoders being left.
There was an effort to improve the correctness of timestamps and frame durations of each packet, increasing the
accurracy of variable frame rate video.
Next major release will be version 7.0, scheduled to be released in February. We will attempt to better stick to the new release schedule we announced at the start of this year.
We strongly recommend users, distributors, and system integrators to upgrade unless they use current git master.
]]>Those wishing to test the code can read our documentation page. For those who would like to integrate FFmpeg's Vulkan code to demux, parse, decode, and receive a VkImage to present or manipulate, documentation and examples are available in our source tree. Currently, using the latest available git checkout of our repository is required. The functionality will be included in stable branches with the release of version 6.1, due to be released soon.
As this is also the first practical implementation of the specifications, bugs may be present, particularly in drivers, and, although passing verification, the implementation itself. New codecs, and encoding support are also being worked on, by both the Khronos organization for standardizing, and us as implementing it, and giving feedback on improving.
]]>New decoders featured are Bonk, RKA, Radiance, SC-4, APAC, VQC, WavArc and a few ADPCM formats. QSV and NVenc now support AV1 encoding. The FFmpeg CLI (we usually reffer to it as ffmpeg.c to avoid confusion) has speed-up improvements due to threading, as well as statistics options, and the ability to pass option values for filters from a file. There are quite a few new audio and video filters, such as adrc, showcwt, backgroundkey and ssim360, with a few hardware ones too. Finally, the release features many behind-the-scenes changes, including a new FFT and MDCT implementation used in codecs (expect a blog post about this soon), numerous bugfixes, better ICC profile handling and colorspace signalling improvement, introduction of a number of RISC-V vector and scalar assembly optimized routines, and a few new improved APIs, which can be viewed in the doc/APIchanges file in our tree. A few submitted features, such as the Vulkan improvements and more FFT optimizations will be in the next minor release, 6.1, which we plan to release soon, in line with our new release schedule. Some highlights are:
We strongly recommend users, distributors, and system integrators to upgrade unless they use current git master.
]]>We strongly recommend users, distributors, and system integrators to upgrade unless they use current git master.
]]>We strongly recommend users, distributors, and system integrators to upgrade unless they use current git master.
]]>We strongly recommend users, distributors, and system integrators to upgrade unless they use current git master.
]]>We strongly recommend users, distributors, and system integrators to upgrade unless they use current git master.
]]>Note that this filter is not FDA approved, nor are we medical professionals. Nor has this filter been tested with anyone who has photosensitive epilepsy. FFmpeg and its photosensitivity filter are not making any medical claims.
That said, this is a new video filter that may help photosensitive people watch tv, play video games or even be used with a VR headset to block out epiletic triggers such as filtered sunlight when they are outside. Or you could use it against those annoying white flashes on your tv screen. The filter fails on some input, such as the Incredibles 2 Screen Slaver scene. It is not perfect. If you have other clips that you want this filter to work better on, please report them to us on our trac.
See for yourself. Example was made with -vf photosensitivity=20:0.8
We are not professionals. Please use this in your medical studies to advance epilepsy research. If you decide to use this in a medical setting, or make a hardware hdmi input output realtime tv filter, or find another use for this, please let me know. This filter was a feature request of mine since 2013.
]]>We strongly recommend users, distributors, and system integrators to upgrade unless they use current git master.
]]>We strongly recommend users, distributors, and system integrators to upgrade unless they use current git master.
]]>We strongly recommend users, distributors, and system integrators to upgrade unless they use current git master.
]]>We strongly recommend users, distributors, and system integrators to upgrade unless they use current git master.
]]>We strongly recommend users, distributors, and system integrators to upgrade unless they use current git master.
]]>Without further ado, here's the silver-lining for each one of the projects we sought to complete during this Summer of Code season:
Stanislav Dolganov designed and implemented experimental support for motion estimation and compensation in the lossless FFV1 codec. The design and implementation is based on the snow video codec, which uses OBMC. Stanislav's work proved that significant compression gains can be achieved with inter frame compression. FFmpeg welcomes Stanislav to continue working beyond this proof of concept and bring its advances into the official FFV1 specification within the IETF.
Petru Rares Sincraian added several self-tests to FFmpeg and successfully went through the in-some-cases tedious process of fine tuning tests parameters to avoid known and hard to avoid problems, like checksum mismatches due to rounding errors on the myriad of platforms we support. His work has improved the code coverage of our self tests considerably.
Umair Khan updated and integrated the ALS encoder to fit in the current FFmpeg codebase. He also implemented a missing feature for the ALS decoder that enables floating-point sample decoding. FFmpeg support for MPEG-4 ALS has been improved significantly by Umair's work. We welcome him to keep maintaining his improvements and hope for great contributions to come.
Ján Sebechlebský's generic goal was to improve the tee muxer so it tolerated blocking IO and allowed transparent error recovery. During the design phase it turned out that this functionality called for a separate muxer, so Ján spent his summer working on the so-called FIFO muxer, gradually fixing issues all over the codebase. He succeeded in his task, and the FIFO muxer is now part of the main repository, alongside several other improvements he made in the process.
Jai Luthra's objective was to update the out-of-tree and pretty much abandoned MLP (Meridian Lossless Packing) encoder for libavcodec and improve it to enable encoding to the TrueHD format. For the qualification period the encoder was updated such that it was usable and throughout the summer, successfully improved adding support for multi-channel audio and TrueHD encoding. Jai's code has been merged into the main repository now. While a few problems remain with respect to LFE channel and 32 bit sample handling, these are in the process of being fixed such that effort can be finally put in improving the encoder's speed and efficiency.
Davinder Singh investigated existing motion estimation and interpolation approaches from the available literature and previous work by our own: Michael Niedermayer, and implemented filters based on this research. These filters allow motion interpolating frame rate conversion to be applied to a video, for example, to create a slow motion effect or change the frame rate while smoothly interpolating the video along the motion vectors. There's still work to be done to call these filters 'finished', which is rather hard all things considered, but we are looking optimistically at their future.
And that's it. We are happy with the results of the program and immensely thankful for the opportunity of working with such an amazing set of students. We can be a tough crowd but our mentors did an amazing job at hand holding our interns through their journey. Thanks also to Google for this wonderful program and to everyone that made room in their busy lives to help making GSoC2016 a success. See you in 2017!
]]>We recommend users, distributors, and system integrators, to upgrade unless they use current git master.
]]>We strongly recommend users, distributors, and system integrators, especially those who experienced issues upgrading from 3.0, to upgrade unless they use current git master.
]]>We strongly recommend users, distributors, and system integrators to upgrade unless they use current git master.
]]>We strongly recommend users, distributors, and system integrators to upgrade unless they use current git master.
]]>Even before marking our internal AAC encoder as stable, it was known that libvo-aacenc was of an inferior quality compared to our native one for most samples. However, the VisualOn encoder was used extensively by the Android Open Source Project, and we would like to have a tested-and-true stable option in our code base.
When first committed in 2011, libaacplus filled in the gap of encoding High Efficiency AAC formats (HE-AAC and HE-AACv2), which was not supported by any of the encoders in FFmpeg at that time.
The circumstances for both have changed. After the work spearheaded by Rostislav Pehlivanov and Claudio Freire, the now-stable FFmpeg native AAC encoder is ready to compete with much more mature encoders. The Fraunhofer FDK AAC Codec Library for Android was added in 2012 as the fourth supported external AAC encoder, and the one with the best quality and the most features supported, including HE-AAC and HE-AACv2.
Therefore, we have decided that it is time to remove libvo-aacenc and
libaacplus. If you are currently using libvo-aacenc, prepare to transition
to the native encoder (aac) when updating to the next version
of FFmpeg. In most cases it is as simple as merely swapping the encoder
name. If you are currently using libaacplus, start using FDK AAC
(libfdk_aac) with an appropriate profile option
to select the exact AAC profile that fits your needs. In both cases, you
will enjoy an audible quality improvement and as well as fewer licensing
headaches.
Enjoy!
]]>We recommend users, distributors and system integrators to upgrade unless they use current git master.
]]>Licensing has always been an issue with encoding AAC audio as most of the encoders have had a license making FFmpeg unredistributable if compiled with support for them. The fact that there now exists a fully open and truly free AAC encoder integrated directly within the project means a lot to those who wish to use accepted and widespread standards.
The majority of the work done to bring the encoder up to quality was started during this year's GSoC by developer Claudio Freire and Rostislav Pehlivanov. Both continued to work on the encoder with the latter joining as a developer and mainainer, working on other parts of the project as well. Also, thanks to Kamedo2 who does comparisons and tests, the original authors and all past and current contributors to the encoder. Users are suggested and encouraged to use the encoder and provide feedback or breakage reports through our bug tracker.
]]>Telepoint is the biggest carrier-neutral data center in Bulgaria. Located in the heart of Sofia on a cross-road of many Bulgarian and International networks, the facility is a fully featured Tier 3 data center that provides flexible customer-oriented colocation solutions (ranging from a server to a private collocation hall) and a high level of security.
MediaHub Ltd. is a Bulgarian IPTV platform and services provider which uses FFmpeg heavily since it started operating a year ago. "Donating to help keep FFmpeg online is our way of giving back to the community" .
Thanks Telepoint and MediaHub for their support!
]]>We want to thank Google, the participating students, and especially the mentors who joined this effort. We're looking forward to participating in the next GSoC edition!
Below you can find a brief description of the final outcome of each single project.
Stephan Holljes's project for this session of Google Summer of Code was to implement basic HTTP server features for libavformat, to complement the already present HTTP client and RTMP and RTSP server code.
The first part of the project was to make the HTTP code capable of accepting a single client; it was completed partly during the qualification period and partly during the first week of the summer. Thanks to this work, it is now possible to make a simple HTTP stream using the following commands:
ffmpeg -i /dev/video0 -listen 1 -f matroska \
-c:v libx264 -preset fast -tune zerolatency http://:8080
ffplay http://localhost:8080/
The next part of the project was to extend the code to be able to accept several clients, simultaneously or consecutively. Since libavformat did not have an API for that kind of task, it was necessary to design one. This part was mostly completed before the midterm and applied shortly afterwards. Since the ffmpeg command-line tool is not ready to serve several clients, the test ground for that new API is an example program serving hard-coded content.
The last and most ambitious part of the project was to update ffserver to make use of the new API. It would prove that the API is usable to implement real HTTP servers, and expose the points where more control was needed. By the end of the summer, a first working patch series was undergoing code review.
Mariusz finished an API prepared by the FFmpeg community and implemented Samba directory listing as qualification task.
During the program he extended the API with the possibility to remove and rename files on remote servers. He completed the implementation of these features for file, Samba, SFTP, and FTP protocols.
At the end of the program, Mariusz provided a sketch of an implementation for HTTP directory listening.
Mate was working on directshow input from digital video sources. He got working input from ATSC input sources, with specifiable tuner.
The code has not been committed, but a patch of it was sent to the ffmpeg-devel mailing list for future use.
The mentor plans on cleaning it up and committing it, at least for the ATSC side of things. Mate and the mentor are still working trying to finally figure out how to get DVB working.
Niklesh's project was to expand our support for 3GPP Timed Text subtitles. This is the native subtitle format for mp4 containers, and is interesting because it's usually the only subtitle format supported by the stock playback applications on iOS and Android devices.
ffmpeg already had basic support for these subtitles which ignored all formatting information - it just provided basic plain-text support.
Niklesh did work to add support on both the encode and decode side for text formatting capabilities, such as font size/colour and effects like bold/italics, highlighting, etc.
The main challenge here is that Timed Text handles formatting in a very different way from most common subtitle formats. It uses a binary encoding (based on mp4 boxes, naturally) and stores information separately from the text itself. This requires additional work to track which parts of the text formatting applies to, and explicitly dealing with overlapping formatting (which other formats support but Timed Text does not) so it requires breaking the overlapping sections into separate non-overlapping ones with different formatting.
Finally, Niklesh had to be careful about not trusting any size information in the subtitles - and that's no joke: the now infamous Android stagefright bug was in code for parsing Timed Text subtitles.
All of Niklesh's work is committed and was released in ffmpeg 2.8.
Pedro Arthur has modularized the vertical and horizontal scalers. To do this he designed and implemented a generic filter framework and moved the existing scaler code into it. These changes now allow easily adding removing, splitting or merging processing steps. The implementation was benchmarked and several alternatives were tried to avoid speed loss.
He also added gamma corrected scaling support. An example to use gamma corrected scaling would be:
ffmpeg -i input -vf scale=512:384:gamma=1 output
Pedro has done impressive work considering the short time available, and he is a FFmpeg committer now. He continues to contribute to FFmpeg, and has fixed some bugs in libswscale after GSoC has ended.
Rostislav Pehlivanov has implemented PNS, TNS, I/S coding and main prediction on the native AAC encoder. Of all those extensions, only TNS was left in a less-than-usable state, but the implementation has been pushed (disabled) anyway since it's a good basis for further improvements.
PNS replaces noisy bands with a single scalefactor representing the energy of that band, gaining in coding efficiency considerably, and the quality improvements on low bitrates are impressive for such a simple feature.
TNS still needs some polishing, but has the potential to reduce coding artifacts by applying noise shaping in the temporal domain (something that is a source of annoying, notable distortion on low-entropy bands).
Intensity Stereo coding (I/S) can double coding efficiency by exploiting strong correlation between stereo channels, most effective on pop-style tracks that employ panned mixing. The technique is not as effective on classic X-Y recordings though.
Finally, main prediction improves coding efficiency by exploiting correlation among successive frames. While the gains have not been huge at this point, Rostislav has remained active even after the GSoC, and is polishing both TNS and main prediction, as well as looking for further improvements to make.
In the process, the MIPS port of the encoder was broken a few times, something he's also working to fix.
Donny Yang implemented basic keyframe only APNG encoder as the qualification task. Later he wrote interframe compression via various blend modes. The current implementation tries all blend modes and picks one which takes the smallest amount of memory.
Special care was taken to make sure that the decoder plays correctly all files found in the wild and that the encoder produces files that can be played in browsers that support APNG.
During his work he was tasked to fix any encountered bug in the decoder due to the fact that it doesn't match APNG specifications. Thanks to this work, a long standing bug in the PNG decoder has been fixed.
For latter work he plans to continue working on the encoder, making it possible to select which blend modes will be used in the encoding process. This could speed up encoding of APNG files.
]]>We recommend users, distributors and system integrators to upgrade unless they use current git master.
]]>The resignation of Michael Niedermayer as leader of FFmpeg yesterday has come by surprise. He has worked tirelessly on the FFmpeg project for many years and we must thank him for the work that he has done. We hope that in the future he will continue to contribute to the project. In the coming weeks, the FFmpeg project will be managed by the active contributors.
The last four years have not been easy for our multimedia community - both contributors and users. We should now look to the future, try to find solutions to these issues, and to have reconciliation between the forks, which have split the community for so long.
Unfortunately, much of the disagreement has taken place in inappropriate venues so far, which has made finding common ground and solutions difficult. We aim to discuss this in our communities online over the coming weeks, and in person at the VideoLAN Developer Days in Paris in September: a neutral venue for the entire open source multimedia community.
The FFmpeg project.
]]>After graciously hosting our projects (FFmpeg, MPlayer and rtmpdump) for 4 years, Arpi (our hoster) has informed us that we have to secure a new host somewhere else immediately.
If you want to host an open source project, please let us know, either on ffmpeg-devel mailing list or irc.freenode.net #ffmpeg-devel.
We use about 4TB of storage and at least 4TB of bandwidth / month for various mailing lists, trac, samples repo, svn, etc.
]]>We recommend users, distributors and system integrators to upgrade unless they use current git master.
]]>More information can be found here
We demonstrate usage of FFmpeg, answer your questions and listen to your problems and wishes. If you have media files that cannot be processed correctly with FFmpeg, be sure to have a sample with you so we can have a look!
For the first time in our CLT history, there will be an FFmpeg workshop! You can read the details here. The workshop is targeted at FFmpeg beginners. First the basics of multimedia will be covered. Thereafter you will learn how to use that knowledge and the FFmpeg CLI tools to analyse and process media files. The workshop is in German language only and prior registration is necessary. The workshop will be on Saturday starting at 10 o'clock.
We are looking forward to meet you (again)!
]]>We recommend users, distributors and system integrators to upgrade unless they use current git master.
]]>Unfortunately that was already the easy part of this news. The bad news is the packages probably won't migrate to Debian testing to be in the upcoming release codenamed jessie. Read the argumentation over at Debian.
However things will come out in the end, we hope for your continued remarkable support!
]]>We all know FFmpeg is used by the industry, but even while there are countless products building on our code, it is not at all common for companies to step up and help us out when needed. So a big thank-you to Samsung and the OPW program committee!
If you are thinking on participating in OPW as an intern, please take a look at our OPW wiki page for some initial guidelines. The page is still a work in progress, but there should be enough information there to get you started. If you, on the other hand, are thinking on sponsoring work on FFmpeg through the OPW program, please get in touch with us at opw@ffmpeg.org. With your help, we might be able to secure some extra intern spots for this round!
]]>We recommend users, distributors and system integrators to upgrade unless they use current git master.
]]>We recommend users, distributors and system integrators to upgrade unless they use current git master.
]]>We recommend users, distributors and system integrators to upgrade unless they use current git master.
]]>We have set up an email address you can use to contact us about donations and general inquires regarding our participation in the program. Hope to hear from you soon!
]]>We recommend users, distributors and system integrators to upgrade unless they use current git master.
]]>We will have a shared booth with XBMC and VideoLAN. If you have media files that cannot be processed correctly with FFmpeg, be sure to have a sample with you so we can have a look!
More information about LinuxTag can be found here
We are looking forward to see you in Berlin!
]]>We encourage you to read up on "OpenSSL heartbleed". It is possible that login data for the issue tracker was exposed to people exploiting this security hole. You might want to change your password in the tracker and everywhere else you used that same password.
]]>
- HNM version 4 demuxer and video decoder
- Live HDS muxer
- setsar/setdar filters now support variables in ratio expressions
- elbg filter
- string validation in ffprobe
- support for decoding through VDPAU in ffmpeg (the -hwaccel option)
- complete Voxware MetaSound decoder
- remove mp3_header_compress bitstream filter
- Windows resource files for shared libraries
- aeval filter
- stereoscopic 3d metadata handling
- WebP encoding via libwebp
- ATRAC3+ decoder
- VP8 in Ogg demuxing
- side & metadata support in NUT
- framepack filter
- XYZ12 rawvideo support in NUT
- Exif metadata support in WebP decoder
- OpenGL device
- Use metadata_header_padding to control padding in ID3 tags (currently used in
MP3, AIFF, and OMA files), FLAC header, and the AVI "junk" block.
- Mirillis FIC video decoder
- Support DNx444
- libx265 encoder
- dejudder filter
- Autodetect VDA like all other hardware accelerations
We recommend users, distributors and system integrators to upgrade unless they use current git master.
]]>More information can be found here
We invite you to visit us at our booth located in the Linux-Live area! There we will demonstrate usage of FFmpeg, answer your questions and listen to your problems and wishes.
If you have media files that cannot be processed correctly with FFmpeg, be sure to have a sample with you so we can have a look!
We are looking forward to meet you (again)!
]]>We recommend all users to change their passwords. Especially users who use a password on Trac that they also use elsewhere, should change that password at least elsewhere.
]]>Rogério Brito, a Debian developer, has proposed a Request For Package (RFP) in the Debian bug tracking system.
Please let the Debian and Ubuntu developers know that you support packaging of the real FFmpeg! See Debian ticket #729203 for more details.
]]>
- aecho filter
- perspective filter ported from libmpcodecs
- ffprobe -show_programs option
- compand filter
- RTMP seek support
- when transcoding with ffmpeg (i.e. not streamcopying), -ss is now accurate
even when used as an input option. Previous behavior can be restored with
the -noaccurate_seek option.
- ffmpeg -t option can now be used for inputs, to limit the duration of
data read from an input file
- incomplete Voxware MetaSound decoder
- read EXIF metadata from JPEG
- DVB teletext decoder
- phase filter ported from libmpcodecs
- w3fdif filter
- Opus support in Matroska
- FFV1 version 1.3 is stable and no longer experimental
- FFV1: YUVA(444,422,420) 9, 10 and 16 bit support
- changed DTS stream id in lavf mpeg ps muxer from 0x8a to 0x88, to be
more consistent with other muxers.
- adelay filter
- pullup filter ported from libmpcodecs
- ffprobe -read_intervals option
- Lossless and alpha support for WebP decoder
- Error Resilient AAC syntax (ER AAC LC) decoding
- Low Delay AAC (ER AAC LD) decoding
- mux chapters in ASF files
- SFTP protocol (via libssh)
- libx264: add ability to encode in YUVJ422P and YUVJ444P
- Fraps: use BT.709 colorspace by default for yuv, as reference fraps decoder does
- make decoding alpha optional for prores, ffv1 and vp6 by setting
the skip_alpha flag.
- ladspa wrapper filter
- native VP9 decoder
- dpx parser
- max_error_rate parameter in ffmpeg
- PulseAudio output device
- ReplayGain scanner
- Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support)
- Linux framebuffer output device
- HEVC decoder, raw HEVC demuxer, HEVC demuxing in TS, Matroska and MP4
- mergeplanes filter
We recommend users, distributors and system integrators to upgrade unless they use current git master.
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