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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 FFmpeg 4. Some of the highlights:. We strongly recommend users, distributors, and system integrators to upgrade unless they use current git master.

FFmpeg 3. This has been a long time coming but we wanted to give a proper closure to our participation in this run of the program and it takes time.

Sometimes it's just to get the final report for each project trimmed down, others, is finalizing whatever was still in progress when the program finished: final patches need to be merged, TODO lists stabilized, future plans agreed; you name it.

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.

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. He also implemented a missing feature for the ALS decoder that enables floating-point sample decoding.

We welcome him to keep maintaining his improvements and hope for great contributions to come. 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 GSoC a success.

See you in ! Support for the SDL1 library has been dropped, due to it no longer being maintained as of January, and it being superseded by the SDL2 library.

Both the ffplay and opengl output devices have been updated to support SDL2. It fixes several bugs. We recommend users, distributors, and system integrators, to upgrade unless they use current git master.

After thorough deliberation, we're announcing that we're about to drop the ffserver program from the project starting with the next release.

Furthermore the program has been hard for users to deploy and run due to reliability issues, lack of knowledgable people to help and confusing configuration file syntax.

Current users and members of the community are invited to write a replacement program to fill the same niche that ffserver did using the new APIs and to contact us so we may point users to test and contribute to its development.

It mainly deals with a few ABI issues introduced in the previous release. We strongly recommend users, distributors, and system integrators, especially those who experienced issues upgrading from 3.

FFmpeg has been accepted as a Google Summer of Code open source organization. If you wish to participate as a student see our project ideas page.

You can already get in contact with mentors and start working on qualification tasks as well as register at google and submit your project proposal draft.

Good luck! 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.

The circumstances for both have changed. 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. In both cases, you will enjoy an audible quality improvement and as well as fewer licensing headaches.

We have made several new point releases 2. Please see the changelog for each release for more details. We recommend users, distributors and system integrators to upgrade unless they use current git master.

After seven years the native FFmpeg AAC encoder has had its experimental flag removed and declared as ready for general use.

The encoder is transparent at kbps for most samples tested with artifacts only appearing in extreme cases. Subjective quality tests put the encoder to be of equal or greater quality than most of the other encoders available to the public.

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.

A big thank you note goes to our newest supporters: MediaHub and Telepoint. Both companies have donated a dedicated server with free of charge internet connectivity.

Here is a little bit about them in their own words:. 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. FFmpeg got a total of 8 assigned projects, and 7 of them were successful. 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! 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:. 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.

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. 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. 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.

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.

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:. Pedro has done impressive work considering the short time available, and he is a FFmpeg committer now.

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.

As a member of this team, you will be responsible of following:. At FFM we need to find a balance between quality and speed to provide working, stable solutions in shortest time.

We expect you to have experience and knowledge of majority of the following:. Please send CV to jobs ffm-online. Hola MWC19! FFM team is heading to Barcelona!

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