FFmpeg 5.0 "Lorentz", a new major release, is now available! For this long-overdue release, a major effort underwent to remove the old encode/decode APIs and replace them with an N:M-based API, the entire libavresample library was removed, libswscale has a new, easier to use AVframe-based API, the Vulkan code was much improved, many new filters were added, including libplacebo integration, and finally, DoVi support was added, including tonemapping and remuxing. The default AAC encoder settings were also changed to improve quality. Some of the changelog highlights:
Psn Code Generator 4.2 Plugin
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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.
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. 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.
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 128kbps 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.
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:
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.
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.
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.
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.
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.
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.
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!
In the onInit() method, update the code to appear as follows. Add the call to initialize our project configuration after the super.onInit() call. We do this after the super.onInit() to ensure that the framework has a chance to initialize anything required, and that we are setting up the library after those steps are completed.
Unreal Engine can help you implement EOS in your project through its Online Subsystem (OSS) plugin, without the need to write code using the EOS SDK directly. The Online Subsystem plugin provides a common interface across all exposed online services in Unreal Engine. To use this feature, developers must register and configure their products with Epic Account Services (EAS), and then enable and configure a few plugins to expose EOS functionality through the OSS interface.
To begin leveraging EOS in your project, you must first enable the EOS OSS plugins. You can find them by going to the Edit menu and selecting Plugins to open the Plugin Browser.
Within the Plugin Browser, locate and enable Online Subsystem EOS and EOS Shared. If you want to add voice chat support through EOS, enable the EOS RTC IVoiceChat plugin as well.
After enabling the plugins, you will need to configure them. Some of the following configuration steps require product-related settings or identifiers received after registering your product with EOS.
EOS Plus is an experimental OSS plugin that combines the features of the EOS SDK with those of another online platform (such as Xbox Live, PlayStation, Steam, and so on) into a single, crossplay-compatible Online Subsystem. 2ff7e9595c
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