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Autres articles (101)

  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

  • Websites made ​​with MediaSPIP

    2 mai 2011, par

    This page lists some websites based on MediaSPIP.

  • Creating farms of unique websites

    13 avril 2011, par

    MediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
    This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...)

Sur d’autres sites (16547)

  • avcodec/huffman : beautify : add space between #include and filename.

    22 juillet 2016, par Yong Lei
    avcodec/huffman : beautify : add space between #include and filename.

    Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
    • [DH] libavcodec/huffman.c

  • ffmpeg : split mp3, encode aac and join produce artifacts and empty space

    18 juin 2016, par aganeiro

    Source mp3

        ffprobe -show_frames -select_streams a -print_format csv -show_entries  
    
 frame=index,pkt_dts_time ~/demo_files/000.orig.5352357791787324393.mp3
    
frame,0.000000
    
frame,0.026122
    
frame,0.052245
    
frame,0.078367

    every part I make with command, -ss position and -t time I got and calculate from previous ffprobe output

        /home/xxx/bin/ffmpeg -analyzeduration 50000000 -probesize 50000000  
    
-ss 0.000000 -i /home/xxx/demo_files/000.orig.5352357791787324393.mp3  
    
-s 0 -t 0.926276 -flags +global_header -c:a libfdk_aac -strict -2  
    
-b:a 64k -ac 2 -ar 44100 -vn -f mpegts -y /tmp/p0.ts
    

    
    /home/xxx/bin/ffmpeg -analyzeduration 50000000 -probesize 50000000  
    
-ss 1.018776 -i /home/xxx/demo_files/000.orig.5352357791787324393.mp 
    
-s 0 -t 0.900153 -flags +global_header -c:a libfdk_aac -strict -2  
    
-b:a 64k -ac 2 -ar 44100 -vn -f mpegts -y /tmp/p1.ts

    it’s produce

    [mp3 @ 0x39ca980] Estimating duration from bitrate, this may be inaccurate
    
    Input #0, mp3, from '/home/xxx/demo_files/000.orig.5352357791787324393.mp3':
    
    Duration: 00:05:17.20, start: 0.000000, bitrate: 320 kb/s
    
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
    
    [mpegts @ 0x39ccea0] Using AVStream.codec to pass codec  
    
parameters to muxers is deprecated, use AVStream.codecpar instead.
    
    [mpegts @ 0x39ccea0] frame size not set
    
    Output #0, mpegts, to '/tmp/p0.ts':
    
      Metadata:
    
        encoder         : Lavf57.38.100
    
        Stream #0:0: Audio: aac (libfdk_aac), 44100 Hz, stereo, s16, 64 kb/s
    
        Metadata:
    
          encoder         : Lavc57.46.100 libfdk_aac
    
    Stream mapping:
    
      Stream #0:0 -> #0:0 (mp3 (native) -> aac (libfdk_aac))
    
    Press [q] to stop, [?] for help
    
    size=      10kB time=00:00:00.92 bitrate=  92.3kbits/s speed=39.8x    
    
    video:0kB audio:8kB subtitle:0kB other streams:0kB global  
    
headers:0kB muxing overhead: 24.619143%
    
      Duration: 00:00:00.63, start: 1.400000, bitrate: 127 kb/s

    Part info

        ffmpeg -hide_banner -i /tmp/p0.ts 2>&amp;1 |grep -P 'Duration|Stream'
    
    Duration: 00:00:00.95, start: 1.400000, bitrate: 90 kb/s
    
    Stream #0:0[0x100]: Audio: aac (LC) ([15][0][0][0] / 0x000F),  
    
44100 Hz, stereo, fltp, 68 kb/s

    Then I join all parts together with

        /home/xxx/bin/ffmpeg -i /tmp/p0.ts -i /tmp/p1.ts -i /tmp/p2.ts  
    
-i /tmp/p3.ts -i /tmp/p4.ts -i /tmp/p5.ts -filter_complex  
    
"[0:a]asetpts=PTS-STARTPTS[a0];[1:a]asetpts=PTS-STARTPTS[a1];  
    
[2:a]asetpts=PTS-STARTPTS[a2];[3:a]asetpts=PTS-STARTPTS[a3];  
    
[4:a]asetpts=PTS-STARTPTS[a4];[5:a]asetpts=PTS-STARTPTS[a5];  
    
[a0][a1][a2][a3][a4][a5] concat=n=6:v=0:a=1 [a]"  
    
-map [a] -strict experimental -fflags +genpts -flags +global_header  
    
-c libfdk_aac -bsf:a aac_adtstoasc -y /tmp/res.m4a

    waveform of original and joined on the left
    i68.tinypic.com/magcnl.jpg

    So, as you can see joined have delays and waveforms starte later. Why ? maybe it depens that all encoded parts have start time 1.400000, ?? How to set start time to 0 on encode ?

    Also I tried to cut empty space on joining with filter_complex but result stil not good and contains artifacts because trim position looks different in every part.

        /home/xxx/bin/ffmpeg -i /tmp/p0.ts -i /tmp/p1.ts -i /tmp/p2.ts  
    
-i /tmp/p3.ts -i /tmp/p4.ts -i /tmp/p5.ts -filter_complex  
    
"[0:a]atrim=0.020000,asetpts=PTS-STARTPTS[a0];  
    
[1:a]atrim=0.020000,asetpts=PTS-STARTPTS[a1];  
    
[2:a]atrim=0.020000,asetpts=PTS-STARTPTS[a2];  
    
[3:a]atrim=0.020000,asetpts=PTS-STARTPTS[a3];  
    
[4:a]atrim=0.020000,asetpts=PTS-STARTPTS[a4];  
    
[5:a]atrim=0.020000,asetpts=PTS-STARTPTS[a5];  
    
[a0][a1][a2][a3][a4][a5] concat=n=6:v=0:a=1 [a]"  
    
-map [a] -strict experimental -fflags +genpts  
    
-flags +global_header -c libfdk_aac -bsf:a aac_adtstoasc  
    
-y /tmp/res.m4a

    Whyyyy and how to solve it ?

  • vf_colorspace : Add support for ycgco color space

    1er novembre 2016, par Vittorio Giovara
    vf_colorspace : Add support for ycgco color space

    Signed-off-by : Vittorio Giovara <vittorio.giovara@gmail.com>
    
Signed-off-by : Ronald S. Bultje <rsbultje@gmail.com>
    • [DH] libavfilter/vf_colorspace.c

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