On 2015-09-27 14:37, Stefano Sabatini wrote: > --- > src/index | 234 > ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > 1 file changed, 234 insertions(+) > > diff --git a/src/index b/src/index > index 9d12a7b..8937172 100644 > --- a/src/index > +++ b/src/index > @@ -37,6 +37,240 @@ > News > </h1> > > + <a id="gsoc2015_result"></a><h3>September 27th, 2015, GSoC 2015 > results</h3> > + > + <p> > + FFmpeg participated to the latest edition of > + the <a > href="http://www.google-melange.com/gsoc/homepage/google/gsoc2015";>Google > + Summer of Code</a> Project. FFmpeg got a total of 8 assigned > + projects, and 7 of them were successfull. ^^^^^^^^^^^ successful
> + </p> > + > + <p>We want to thank <a href="https://www.google.com";>Google</a>, the > + participating students, and especially the mentors who joined this > + effort. We're looking forward participating to the next GSoC > + edition! > + </p> > + > + <p> > + Below you can find a brief description of the final outcome of > + each single project. > + </p> > + > + <h4>Basic servers for network protocols, mentee: klaxa, mentor: Nicolas > George</h4> > + > + <p> > + Stephan Holljes's project for this session of Google Summer of Code was > to Are you deliberately using an IRC nick and a name separately here? > + implement basic HTTP server features for libavformat, to complement the > + already present HTTP client and RTMP and RTSP server code. > + </p> > + > + <p> > + 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: > + <pre> > + ffmpeg -i /dev/video0 -listen 1 -f matroska \ > + -c:v libx264 -preset fast -tune zerolatency http://:8080 > + ffplay http://localhost:8080/ > + </pre> > + </p> > + > + <p> > + 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 on. This > part "it was necessary to design on" does not read very well. Perhaps it should be "it was necessary to design _it_" > + 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. > + </p> > + > + <p> > + 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. > + </p> > + > + <h4>Browsing content on the server, mentee: Mariusz Szczepańczyk, mentor: > Lukasz Marek</h4> > + > + <p> > + Mariusz finished an API prepared by FFmpeg community and implemented > + Samba directory listing as qualification task. > + </p> > + > + <p> > + During the program he extended the API with possibility to remove and > + rename files on remote servers. He implemented/finished the > + implementation of these features for file, Samba, SFTP, and FTP > + protocols. > + </p> > + > + <p> > + At the end of the program, Mariusz provided a sketch of implementation > + for HTTP directory listening. > + </p> > + > + <h4>Directshow digital video capture, mentee: Mate Sebok, mentor: Roger > Pack</h4> > + > + <p> > + Mate was working on directshow input from digital video sources. He > + got working input from ATSC input sources, with specifiable tuner. > + </p> > + > + <p> > + The code has not been committed, but a patch of it was sent to the > + ffmpeg-devel mailing list for future use. > + </p> > + > + <p> > + The mentor plans on cleaning it up and committing up, 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. > + </p> > + > + <h4>Implementing full support for 3GPP Timed Text Subtitles, mentee: > Niklesh Lalwani, mentors: Philip Langdale</h4> > + > + <p> > + 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. > + </p> > + > + <p> > + ffmpeg already had basic support for these subtitles which ignored all > + formatting information - it just provided basic plain-text support. > + </p> > + > + <p> > + 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. > + </p> > + > + <p> > + 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. > + </p> > + > + <p> > + 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. > + </p> > + > + <p> > + All of Niklesh's work is committed and was released in ffmpeg 2.8. > + </p> > + > +<h4>libswscale refactoring, mentee: Pedro Arthur, mentors: Michael > Niedermayer, Ramiro Polla</h4> > + > + <p> > + Pedro Arthur has modlularized 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 where ^^^^^ were > + tried to avoid speedloss. > + </p> > + > + <p> > + He also added gamma correct scaling support. > + An example to use gamma corrected scaling would be: > + <pre> > + ffmpeg -i input -vf scale=512:384:gamma=1 output > + </pre> > + </p> > + > + <p> > + Pedro has done impressive work considering the short time available, > + and he is a FFmpeg comitter now. He continues to contribute to ^^^^^^^^ If "committer" is an acceptable new word then it should have two 'M's > + FFmpeg, and has fixed some bugs in libswscale after GSoC has > + ended. > + </p> > + > + <h4>AAC Encoder Improvements, mentee: Rostislav Pehlivanov, mentor: > Claudio Freire</h4> > + > + <p> > + 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. > + </p> > + > + <p> > + 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. > + </p> > + > + <p> > + TNS still needs some polishing, but has the potential to reduce coding > + artifacts by applying noise shaping in the temporal domain (something ^^^^^^^^^ artefacts > + that is a source of annoying, notable distortion on low-entropy > + bands). > + </p> > + > + <p> > + 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. > + </p> > + > + <p> > + 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 > + furhter improvements to make. ^^^^^^^ further > + </p> > + > + <p> > + In the process, the MIPS port of the encoder was broken a few times, > + something he's also working to fix. > + </p> > + > + <h4>Animated Portable Network Graphics (APNG), mentee: Donny Yang, mentor: > Paul B Mahol</h4> > + > + <p> > + Donny Yang implemented basic keyframe only APNG encoder as qualification > + task, later he wrote kind of interframe compression via various blend > + modes. Current implementation tries all blend modes and picks one which > + takes the smallest amount of memory. > + </p> > + > + <p> > + Special care was taken to make sure that the decoder plays correctly > + all files found in wild and that the encoder produces files that can > + be played in browsers that support APNG. > + </p> > + > + <p> > + 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, the also long standing bug in the > + PNG decoder has been fixed. > + </p> > + > + <p> > + For latter work he plans to continue working on the encoder, making > + possible to select which blend modes will be used in the encoding > + process, which could speed up encoding of APNG files. > + </p> > + > <a id="pr2.8"></a><h3>September 9th, 2015, FFmpeg 2.8</h3> > <p> > We published release <b><a href="download.html#release_2.8">2.8</a></b> > as new major version. > There are many more technical terms that my spell checker insists should be hyphenated--inter-frame, intra-frame, key-frame--but I always see them written as one word.
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