Recherche avancée

Médias (0)

Mot : - Tags -/flash

Aucun média correspondant à vos critères n’est disponible sur le site.

Autres articles (44)

  • Personnaliser en ajoutant son logo, sa bannière ou son image de fond

    5 septembre 2013, par

    Certains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;

  • Ecrire une actualité

    21 juin 2013, par

    Présentez les changements dans votre MédiaSPIP ou les actualités de vos projets sur votre MédiaSPIP grâce à la rubrique actualités.
    Dans le thème par défaut spipeo de MédiaSPIP, les actualités sont affichées en bas de la page principale sous les éditoriaux.
    Vous pouvez personnaliser le formulaire de création d’une actualité.
    Formulaire de création d’une actualité Dans le cas d’un document de type actualité, les champs proposés par défaut sont : Date de publication ( personnaliser la date de publication ) (...)

  • Publier sur MédiaSpip

    13 juin 2013

    Puis-je poster des contenus à partir d’une tablette Ipad ?
    Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir

Sur d’autres sites (7053)

  • FFmpeg - Max rtbufsize via dshow ?

    14 septembre 2018, par Nimble

    I recently added an additional 4K capture card to my setup and now I’m dropping frames while initializing and ending recordings. In the past I was encoding a 1080P60 stream and a 4K60 stream simultaneously and had no issues, but now that I am trying to encode 2 4K60 streams at once I’m dropping frames as mentioned before.

    The error displays as :

    [dshow @ 000001499bb17180] real-time buffer [Video (00 Pro Capture HDMI 4K+)] [video input] too full or near too full (62% of size: 2147480000 [rtbufsize parameter])! frame dropped!

    or

    [dshow @ 00000149944e7080] real-time buffer [AVerMedia HD Capture GC573 1] [video input] too full or near too full (62% of size: 2147480000 [rtbufsize parameter])! frame dropped!

    10 - 20 times when starting a recording or ending a recording.

    You’d think the solution would be simply increasing my rtbufsize but when I do I just get another error :

    [dshow @ 00000250df6c7080] Value 3000000000.000000 for parameter 'rtbufsize' out of range [0 - 2.14748e+09]
    [dshow @ 00000250df6c7080] Error setting option rtbufsize to value 3000M.
    video=AVerMedia HD Capture GC573 1:audio=SPDIF/ADAT (1+2) (RME Fireface UC): Result too large

    This same error seems to appear if I try to increase the rtbufsize past 2147.48M on any input so I assume it’s a limitation of FFmpeg and not my hardware ? If it is a baked in limitation of FFmpeg what would be the reasoning ? Any way to bypass or other possible solutions ?

    Full command :

    ffmpeg -y -hide_banner -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -rtbufsize 2147.48M `
    -i audio="Analog (1+2) (RME Fireface UC)" `
    -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -rtbufsize 2147.48M `
    -i audio="ADAT (5+6) (RME Fireface UC)" `
    -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -video_size 3840x2160 -rtbufsize 2147.48M `
    -framerate 60 -pixel_format nv12 -i video="Video (00 Pro Capture HDMI 4K+)":audio="ADAT (3+4) (RME Fireface UC)" `
    -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -video_size 3840x2160 -rtbufsize 2147.48M `
    -framerate 60 -pixel_format nv12 -i video="AVerMedia HD Capture GC573 1":audio="SPDIF/ADAT (1+2) (RME Fireface UC)" `
    -thread_queue_size 9999 -indexmem 9999 -r 25 -f lavfi -rtbufsize 2147.48M -i color=c=black:s=50x50 `
    -map 4,0 -map 0 -c:v libx264 -r 25 -rc-lookahead 50 -forced-idr 1 -sc_threshold 0 -flags +cgop `
    -force_key_frames "expr:gte(t,n_forced*2)" -preset ultrafast -pix_fmt nv12 -b:v 16K -minrate 16K -maxrate 16K -bufsize 16k `
    -c:a aac -ar 44100 -b:a 384k -ac 2 -af "aresample=async=250" -vsync 1 -ss 00:00:01.768 `
    -max_muxing_queue_size 9999 -f segment -segment_time 600 -segment_wrap 9 -reset_timestamps 1 `
    -segment_format_options max_delay=0 C:\Users\djcim\Videos\Main\Discord\Discord%02d.ts `
    -map 4,1 -map 1 -c:v libx264 -r 25 -rc-lookahead 50 -forced-idr 1 -sc_threshold 0 -flags +cgop `
    -force_key_frames "expr:gte(t,n_forced*2)" -preset ultrafast -pix_fmt nv12 -b:v 16K -minrate 16K -maxrate 16K -bufsize 16k `
    -c:a aac -ar 44100 -b:a 384k -ac 2 -af "aresample=async=250" -vsync 1 -ss 00:00:01.071 `
    -max_muxing_queue_size 9999 -f segment -segment_time 600 -segment_wrap 9 -reset_timestamps 1 `
    -segment_format_options max_delay=0 C:\Users\djcim\Videos\Main\Soundboard\Soundboard%02d.ts `
    -map 2:0,2:1 -map 2:1 -c:v h264_nvenc -r 60 -rc-lookahead 120 -forced-idr 1 -strict_gop 1 -sc_threshold 0 -flags +cgop `
    -force_key_frames "expr:gte(t,n_forced*2)" -preset: llhp -pix_fmt nv12 -b:v 250M -minrate 250M -maxrate 250M -bufsize 250M `
    -c:a aac -ar 44100 -b:a 384k -ac 2 -af "atrim=0.086, asetpts=PTS-STARTPTS, aresample=async=250" -vsync 1 -ss 00:00:00.102 `
    -max_muxing_queue_size 9999 -f segment -segment_time 600 -segment_wrap 9 -reset_timestamps 1 `
    -segment_format_options max_delay=0 C:\Users\djcim\Videos\Main\Magewell\Magewell%02d.ts `
    -map 3:0,3:1 -map 3:1 -c:v h264_nvenc -r 60 -rc-lookahead 120 -forced-idr 1 -strict_gop 1 -sc_threshold 0 -flags +cgop `
    -force_key_frames "expr:gte(t,n_forced*2)" -preset: llhp -pix_fmt nv12 -b:v 250M -minrate 250M -maxrate 250M -bufsize 250M `
    -c:a aac -ar 44100 -b:a 384k -ac 2 -af "pan=mono|c0=c0, aresample=async=250" -vsync 1 `
    -max_muxing_queue_size 9999 -f segment -segment_time 600 -segment_wrap 9 -reset_timestamps 1 `
    -segment_format_options max_delay=0 C:\Users\djcim\Videos\Main\Camera\Camera%02d.ts

    EDIT : Also worth mentioning that I only drop frames when starting and ending recording, everything is fine in the middle. Wonder if I could like "ease" the recording in or something ?

    (09/13/2018) : I was able to stop frames from dropping while starting a recording by re-arranging inputs and outputs, however no matter how I list things I still drop frames ending recordings.

  • FFMpeg command to convert from avi to mp4 in php script

    16 septembre 2018, par user27240

    ffmpeg command to convert from avi file to mp4 file below does create mp4 file, but the quality of it is not all right at all. Fist, the sizes of the files are different, the mp4 is much smaller and about 1/10 of the avi file.
    Second, the playing time of the newly created mp4 is short lasting about 0.1 seconds where the avi file lasts about 60 seconds.

    Can anyone please help me out for the ffmpeg command to create mp4 file from avi under the directory like below ?

    for i in /xxxxxxx/xxxxxxx/*.avi; do
       ffmpeg -i "$i" -frames:v 1  "/xxxxxxx/xxxxxxx/$(basename "$i" .avi).mp4"
    done    

    Log after the attempt :

    [root@server-xxxxxxx-x ~]# for i in /xxxxxx/xxxxxxx/xxxxxxx/*.avi; do ffmpeg -i "$i" -pix_fmt yuv420p -movflags +faststart "/xxxxxx/xxxxxxx/xxxxxxx/$(basename "$i" .avi).mp4"; done
    ffmpeg version 3.4 Copyright (c) 2000-2017 the FFmpeg developers
     built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-18)
     configuration:
     libavutil      55. 78.100 / 55. 78.100
     libavcodec     57.107.100 / 57.107.100
     libavformat    57. 83.100 / 57. 83.100
     libavdevice    57. 10.100 / 57. 10.100
     libavfilter     6.107.100 /  6.107.100
     libswscale      4.  8.100 /  4.  8.100
     libswresample   2.  9.100 /  2.  9.100
    Guessed Channel Layout for Input Stream #0.1 : mono
    Input #0, avi, from '/xxxxxx/xxxxxxx/xxxxxxx/09141944_5b9b910308858.avi':
     Metadata:
       creation_time   : 2018-09-14 00:18:07
       encoder         : CanonMVI06
      Duration: 00:00:04.20, start: 0.000000, bitrate: 12999 kb/s
       Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc,    bt470bg/unknown/unknown), 640x480, 12384 kb/s, 30 fps, 30 tbr, 30 tbn, 30 tbc
        Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
    File '/xxxxxx/xxxxxxx/xxxxxxx/09141944_5b9b910308858.mp4' already exists.    Overwrite ? [y/N]

                                                                                                                             y
    Stream mapping:
     Stream #0:0 -> #0:0 (mjpeg (native) -> mpeg4 (native))
     Stream #0:1 -> #0:1 (pcm_s16le (native) -> aac (native))
     Press [q] to stop, [?] for help
     [swscaler @ 0x3d75fa0] deprecated pixel format used, make sure you did set   range correctly
    Output #0, mp4, to '/xxxxxx/xxxxxxx/xxxxxxx/09141944_5b9b910308858.mp4':
     Metadata:
       encoder         : Lavf57.83.100
       Stream #0:0: Video: mpeg4 (mp4v / 0x7634706D), yuv420p, 640x480, q=2-31, 200 kb/s, 30 fps, 65521 tbn, 30 tbc
       Metadata:
         encoder         : Lavc57.107.100 mpeg4
        Side data:
         cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: -1
        Stream #0:1: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 69 kb/s
        Metadata:
          encoder         : Lavc57.107.100 aac
     [mp4 @ 0x3d0e6e0] Starting second pass: moving the moov atom to the  beginning of the file17x
     frame=  126 fps=0.0 q=11.8 Lsize=     310kB time=00:00:04.20 bitrate= 603.3kbits/s speed=6.21x
     video:270kB audio:35kB subtitle:0kB other streams:0kB global headers:0kB  muxing overhead: 1.605194%
     [aac @ 0x3d36f40] Qavg: 2283.534
     ffmpeg version 3.4 Copyright (c) 2000-2017 the FFmpeg developers
      built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-18)
      configuration:
      libavutil      55. 78.100 / 55. 78.100
      libavcodec     57.107.100 / 57.107.100
      libavformat    57. 83.100 / 57. 83.100
      libavdevice    57. 10.100 / 57. 10.100
      libavfilter     6.107.100 /  6.107.100
      libswscale      4.  8.100 /  4.  8.100
      libswresample   2.  9.100 /  2.  9.100
      Guessed Channel Layout for Input Stream #0.1 : mono
      Input #0, avi, from '/xxxxxx/xxxxxxx/xxxxxxx/09141945_5b9b913e87834.avi':
       Metadata:
        creation_time   : 2018-09-14 00:18:07
        encoder         : CanonMVI06
       Duration: 00:00:04.20, start: 0.000000, bitrate: 12999 kb/s
       Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc,      bt470bg/unknown/unknown), 640x480, 12384 kb/s, 30 fps, 30 tbr, 30 tbn, 30 tbc
       Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono,    s16, 705 kb/s
  • FFMPEG convert from MP3 to M4A- Getting errors

    4 juin 2021, par JK81

    I'm working on Windows 10 x64, build ffmpeg-20180916-5109c38-win64-static.

    



    I have a folder of MP3 audio files that I want to convert to M4A.

    



    This is the command that I'm running :

    



    FOR /F "tokens=*" %G IN ('dir /b *.mp3') DO ffmpeg -i "%G" -map_metadata 0 -c:a aac -b:a 192000 "%~nG.m4a"

    



    This scans my bin filder and converts all files. This works great for converting video files from MOV to MP4 (using a different code string) but I keep getting errors such as :
[ipod @ 0000025507991b40] Frame rate very high for a muxer not efficiently supporting it.
Please consider specifying a lower framerate, a different muxer or -vsync 2
[libx264 @ 0000025507e45180] using SAR=1/1
[libx264 @ 0000025507e45180] MB rate (129960000) > level limit (16711680)
[libx264 @ 0000025507e45180] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
[libx264 @ 0000025507e45180] profile High 4:4:4 Predictive, level 6.2, 4:4:4, 8-bit
[libx264 @ 0000025507e45180] 264 - core 157 r2932 303c484 - H.264/MPEG-4 AVC codec - Copyleft 2003-2018 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=4 threads=6 lookahead_threads=1 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=25 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
[ipod @ 0000025507991b40] Could not find tag for codec h264 in stream #0, codec not currently supported in container
Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
Error initializing output stream 0:1 --
[aac @ 0000025507e60100] Qavg: nan
Conversion failed!

    



    What's the best way to convert to M4A ? I'm doing this to reduce disk space.