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  • Pas question de marché, de cloud etc...

    10 avril 2011

    Le vocabulaire utilisé sur ce site essaie d’éviter toute référence à la mode qui fleurit allègrement
    sur le web 2.0 et dans les entreprises qui en vivent.
    Vous êtes donc invité à bannir l’utilisation des termes "Brand", "Cloud", "Marché" etc...
    Notre motivation est avant tout de créer un outil simple, accessible à pour tout le monde, favorisant
    le partage de créations sur Internet et permettant aux auteurs de garder une autonomie optimale.
    Aucun "contrat Gold ou Premium" n’est donc prévu, aucun (...)

  • Keeping control of your media in your hands

    13 avril 2011, par

    The vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
    While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
    MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
    MediaSPIP aims to be accessible to as many people as possible and development is based on expanding the (...)

  • Le plugin : Podcasts.

    14 juillet 2010, par

    Le problème du podcasting est à nouveau un problème révélateur de la normalisation des transports de données sur Internet.
    Deux formats intéressants existent : Celui développé par Apple, très axé sur l’utilisation d’iTunes dont la SPEC est ici ; Le format "Media RSS Module" qui est plus "libre" notamment soutenu par Yahoo et le logiciel Miro ;
    Types de fichiers supportés dans les flux
    Le format d’Apple n’autorise que les formats suivants dans ses flux : .mp3 audio/mpeg .m4a audio/x-m4a .mp4 (...)

Sur d’autres sites (4832)

  • "File descriptor in bad state" error while running ffmpeg on android device and selecting an input device

    25 août 2012, par user1545779

    Below is the output of the ffmpeg command :# ./ffmpeg -y -f s16le -i /dev/snd/pcmC3D0c 1640.wmv -to create an audio file from a Logitech webcam on an android device.

    As shown in the output, I received a File descriptor in bad state error for referring to the mic input as /dev/snd/pcmC3D0c I determined the value of the device (webcam mic) by reviewing the contents of /proc/asound. The webcam mic was card3 and its STREAM0 file indicated that the mic has an audio format of format S16_LE

    It was also confirmed that it is a capture device and its' pcm id was pcmC3D0c (C3 being the card number and D0 being the Device number. I then confirmed the correct device by checking the /dev/snd/ directory to confirm its proper and full description. The /dev/snd folder confirmed that the mic was /dev/snd/pcmC3D0c

    I then checked the permissions and ownership to make sure that I could use that device. Hence as far as identifying the correct device to used I do believe that /dev/snd/pcmC3D0c is the correct device. I do believe this error could possibly have something to do with the OS, however after all these checks, I still cannot figure out what is giving the bad file descriptor state error.

    Please note that I tested for different output formats, etc and that did not make any difference. Any leads or suggestions ?

    # ./ffmpeg -y -f s16le -i /dev/snd/pcmC3D0c 1640.wmv

    ffmpeg version N-43170-gd84dd35 Copyright (c) 2000-2012 the FFmpeg developers
    built on Aug 24 2012 09:16:05 with gcc 4.4.3 (GCC) configuration : —enable-cross-compile —arch=arm —cpu=cortex-a9 —target-os=linux —enable-runtime-cpudetect —prefix=/output —enable-pic —cross-prefix=/home/jasongipsyblues/Desktop/apps/android-ndk-r8b/toolchains/arm-linux-androideabi-4.4.3/prebuilt/linux-x86/bin/arm-linux-androideabi- —sysroot=/home/jasongipsyblues/Desktop/apps/android-ndk-r8b/platforms/android-14/arch-arm —enable-version3 —enable-gpl —enable-memalign-hack —disable-doc —enable-yasm —enable-libx264 —enable-zlib —extra-cflags=-I../x264 —extra-ldflags='-L../x264 -lc'

    libavutil 51. 66.100 / 51. 66.100
    libavcodec 54. 48.100 / 54. 48.100
    libavformat 54. 22.100 / 54. 22.100
    libavdevice 54. 2.100 / 54. 2.100
    libavfilter 3. 5.102 / 3. 5.102
    libswscale 2. 1.100 / 2. 1.100
    libswresample 0. 15.100 / 0. 15.100
    libpostproc 52. 0.100 / 52. 0.100

    [s16le @ 0xfd84f0] Invalid sample rate 0 specified using default of 44100
    [s16le @ 0xfd84f0] Estimating duration from bitrate, this may be inaccurate
    Guessed Channel Layout for Input Stream #0.0 : mono
    Input #0, s16le, from '/dev/snd/pcmC3D0c' :
    Duration : N/A, bitrate : 705 kb/s
    Stream #0:0 : Audio : pcm_s16le, 44100 Hz, mono, s16, 705 kb/s
    Output #0, asf, to '1640.wmv' :
    Metadata :
    WM/EncodingSettings : Lavf54.22.100
    Stream #0:0 : Audio : wmav2 (a[1][0][0] / 0x0161), 44100 Hz, mono, s16, 128 kb/s
    Stream mapping :
    Stream #0:0 -> #0:0 (pcm_s16le -> wmav2)
    Press [q] to stop, [?] for help

    /dev/snd/pcmC3D0c : File descriptor in bad state

    size= 1kB time=00:00:00.00 bitrate= 0.0kbits/s
    video:0kB audio:0kB subtitle:0 global headers:0kB muxing overhead 5340.000000%

  • Grayscale streaming from raw video using FFMPEG

    21 janvier 2015, par user1657208

    I have a raw video file (testvideo_1000f.raw) that I am trying to stream in gray scale using ffmpeg and output the grayscale video to output.swf. The command I am using to do this is :

    ffmpeg/ffmpeg -qmin 2 -qmax 31 -s 320x240 -f rawvideo -flags gray -pix_fmt:output gray -an -i testvideo_1000f.raw output.swf

    However, the result from this command is a video stream that is in gray scale but still contains some of the chrominance data. The output from this command is pasted below :

       3 [volta]/home/student/elliott> ffmpeg/ffmpeg -qmin 2 -qmax 31 -s 320x240 -f rawvideo -flags gray -pix_fmt:output gray -an -i testvideo_1000f.raw output.swf
    ffmpeg version N-41632-g2b1fc56 Copyright (c) 2000-2012 the FFmpeg developers
     built on Jul 29 2012 10:27:26 with gcc 4.1.2 20080704 (Red Hat 4.1.2-51)
     configuration:
     libavutil      51. 58.100 / 51. 58.100
     libavcodec     54. 25.100 / 54. 25.100
     libavformat    54.  6.101 / 54.  6.101
     libavdevice    54.  0.100 / 54.  0.100
     libavfilter     2. 80.100 /  2. 80.100
     libswscale      2.  1.100 /  2.  1.100
     libswresample   0. 15.100 /  0. 15.100
    *** CHOOSING 8
    [rawvideo @ 0xdda9660] Estimating duration from bitrate, this may be inaccurate
    Input #0, rawvideo, from 'testvideo_1000f.raw':
     Duration: N/A, start: 0.000000, bitrate: N/A
      Stream #0:0: Video: rawvideo (Y800 / 0x30303859), gray, 320x240, 25 tbr, 25 tbn, 25 tbc
    File 'output.swf' already exists. Overwrite ? [y/N] y
    w:320 h:240 pixfmt:gray tb:1/25 fr:25/1 sar:0/1 sws_param:flags=2
    [ffmpeg_buffersink @ 0xddb7b40] No opaque field provided
    [format @ 0xddb7d40] auto-inserting filter 'auto-inserted scaler 0' between the filter 'Parsed_null_0' and the filter 'format'
    [auto-inserted scaler 0 @ 0xddb7920] w:320 h:240 fmt:gray sar:0/1 -> w:320 h:240 fmt:yuv420p sar:0/1 flags:0x4
    *** CHOOSING 8
    Output #0, swf, to 'output.swf':
     Metadata:
       encoder         : Lavf54.6.101
      Stream #0:0: Video: flv1, yuv420p, 320x240, q=2-31, 200 kb/s, 90k tbn, 25 tbc
    Stream mapping:
     Stream #0:0 -> #0:0 (rawvideo -> flv)
    Press [q] to stop, [?] for help
    Truncating packet of size 76800 to 1 2875kB time=00:00:40.84 bitrate= 576.7kbits/s    
    frame= 1500 fps=1035 q=24.8 Lsize=    4194kB time=00:01:00.00 bitrate= 572.6kbits/s    
    video:4166kB audio:0kB global headers:0kB muxing overhead 0.669245%

    I am fairly new to FFMPEG and I am afraid I am using either the wrong syntax or the wrong parameters in my command line. For some reason, the format of the output is yuv420p. I have tried searching for this answer all over but have had no luck. Could anyone please help me and tell me why the output is being formatted in yuv420p when I am giving the command for it to be in 8bit grayscale ? Any help would be greatly appreciated. Thank you.

    Marc Elliott

  • Burn subtitles Ffmpeg Segmantation Fault error

    5 septembre 2012, par Batuhan Topbaş

    I have vps and ffmpeg latest version. I wanna burn subtitles on videos then i use this input

    ffmpeg -i ohd.avi -report -acodec copy -strict -2 -vcodec libx264 -acodec aac -sameq -vf ass=ohd.ass outputass.mp4

    But this is give me error. Segmantation Fault. See in output..

       ffmpeg started on 2012-09-05 at 10:30:37
    Report written to "ffmpeg-20120905-103037.log"
    ffmpeg version N-44073-g54ca7e3 Copyright (c) 2000-2012 the FFmpeg developers
     built on Aug 31 2012 16:15:40 with gcc 4.1.2 (GCC) 20080704 (Red Hat 4.1.2-52)
     configuration: --enable-libfreetype --enable-libx264 --enable-gpl --enable-libfaac --enable-libass --enable-nonfree
     libavutil      51. 70.100 / 51. 70.100
     libavcodec     54. 55.100 / 54. 55.100
     libavformat    54. 25.104 / 54. 25.104
     libavdevice    54.  2.100 / 54.  2.100
     libavfilter     3. 15.101 /  3. 15.101
     libswscale      2.  1.101 /  2.  1.101
     libswresample   0. 15.100 /  0. 15.100
     libpostproc    52.  0.100 / 52.  0.100
    Input #0, avi, from 'ohd.avi':
     Metadata:
       encoder         : Lavf54.25.104
     Duration: 00:01:00.22, start: 0.000000, bitrate: 4538 kb/s
       Stream #0:0: Video: h264 (High) (H264 / 0x34363248), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 23.98 tbn, 47.95 tbc
       Stream #0:1: Audio: ac3 ([0] [0][0] / 0x2000), 44100 Hz, stereo, s16, 192 kb/s
    File 'outputass.mp4' already exists. Overwrite ? [y/N] y
    FreeType library version: 2.2.1
    [Parsed_ass_0 @ 0xa6b4b00] FreeType headers version: 2.2.1
    [Parsed_ass_0 @ 0xa6b4b00] Init
    [Parsed_ass_0 @ 0xa6b4b00] File size: 622
    [Parsed_ass_0 @ 0xa6b4b00] [0xa6b7420] Style: Default,FreeSerif,24,&H00FFFFFF,&H00FFFFFF,&H00FFFFFF,&H00C0C0C0,-1,0,0,0,100,100,0,0.00,1,2,3,2,20,20,20,1
    [Parsed_ass_0 @ 0xa6b4b00] Added subtitle file: 'ohd.ass' (1 styles, 1 events)
    [Parsed_ass_0 @ 0xa6b4b00] Fontconfig disabled, only default font will be used.
    [graph 0 input from stream 0:0 @ 0xa6cd900] w:1280 h:720 pixfmt:yuv420p tb:1001/24000 fr:24000/1001 sar:1/1 sws_param:flags=2
    [graph 1 input from stream 0:1 @ 0xa6b0ae0] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
    [audio format for output stream 0:1 @ 0xa6b0c60] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:1'
    [auto-inserted resampler 0 @ 0xa6b17c0] chl:stereo fmt:s16 r:44100Hz -> chl:stereo fmt:flt r:44100Hz
    [libx264 @ 0xa6bfa20] using SAR=1/1
    [libx264 @ 0xa6bfa20] using cpu capabilities: MMX2 SSE2Fast FastShuffle SSEMisalign LZCNT
    [libx264 @ 0xa6bfa20] profile High, level 3.1
    [libx264 @ 0xa6bfa20] 264 - core 125 r2208 d9d2288 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=23 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
    Output #0, mp4, to 'outputass.mp4':
     Metadata:
       encoder         : Lavf54.25.104
       Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 24k tbn, 23.98 tbc
       Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, stereo, flt, 128 kb/s
    Stream mapping:
     Stream #0:0 -> #0:0 (h264 -> libx264)
     Stream #0:1 -> #0:1 (ac3 -> aac)
    Press [q] to stop, [?] for help
    Segmentation fault

    What is wrong my input code ? i can't find because i am amateur.