User icon An illustration of a person's head and chest. Sign up Log in. Web icon An illustration of a computer application window Wayback Machine Texts icon An illustration of an open book. Books Video icon An illustration of two cells of a film strip.
Video Audio icon An illustration of an audio speaker. Audio Software icon An illustration of a 3. Software Images icon An illustration of two photographs. Images Donate icon An illustration of a heart shape Donate Ellipses icon An illustration of text ellipses. It can be loaded quickly and an Output Profile window will pop up, giving a wide range of output video format codec options. Step 2 : Select a video format you want to convert to. MP4 is a widely accepted video format.
And then click "Done". Cross-platform avconv can convert between arbitrary sample rates and resize video audio file in a simple way. FFmpeg vs alternative avconv from Libav, FFmpeg gives more codecs, formats, devices and filters options. The faster FFmpeg transcoding alternative, in fact, is a fork of the FFmpeg project.
If you receive FFmpeg error message like "This program is only provided for compatibility and will be removed in a future release. 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 Support for the SDL1 library has been dropped, due to it no longer being maintained as of January, and it being superseded by the SDL2 library.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:.
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