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  • Personnaliser les catégories

    21 June 2013, by

    Formulaire de création d’une catégorie
    Pour ceux qui connaissent bien SPIP, une catégorie peut être assimilée à une rubrique.
    Dans le cas d’un document de type catégorie, les champs proposés par défaut sont : Texte
    On peut modifier ce formulaire dans la partie :
    Administration > Configuration des masques de formulaire.
    Dans le cas d’un document de type média, les champs non affichés par défaut sont : Descriptif rapide
    Par ailleurs, c’est dans cette partie configuration qu’on peut indiquer le (...)

  • Publier sur MédiaSpip

    13 June 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

  • La sauvegarde automatique de canaux SPIP

    1 April 2010, by

    Dans le cadre de la mise en place d’une plateforme ouverte, il est important pour les hébergeurs de pouvoir disposer de sauvegardes assez régulières pour parer à tout problème éventuel.
    Pour réaliser cette tâche on se base sur deux plugins SPIP : Saveauto qui permet une sauvegarde régulière de la base de donnée sous la forme d’un dump mysql (utilisable dans phpmyadmin) mes_fichiers_2 qui permet de réaliser une archive au format zip des données importantes du site (les documents, les éléments (...)

On other websites (5887)

  • Connecting to RTSP stream via FFMpeg - 'could not find codec parameters for stream 0'

    14 November 2013, by user2992545

    I'm a beginner in the FFMpeg world, so please excuse me my overall level of knowledge and any mistakes.

    What I have here is a DVR, made by ITX Security. I have an SDK for it and currently I'm trying to connect to its RTSP using FFMPEG. I've been partially succesfull (i think), the best what I got was:

    `(...)bin>ffplay -user-agent "ITX Security" rtsp://ADMIN:1234@192.168.2.130:5554/live/
    

    
[rtsp @ 02a242e0] UDP timeout, retrying with TCPB sq=0B f=0/0
    
[rtsp @ 02a242e0] method PAUSE failed: 501 Not Implemented
    
[rtsp @ 02a242e0] Could not find codec parameters for stream 0 (Video: h264):unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options
    
rtsp://ADMIN:1234@192.168.2.130:5554/live/: could not find codec parameters
    

    
nan:  0.000 fd=0 aq=    0KB vq=    0KB sq=    0B f=0/0`

    Is it really indicating that it has trouble in playing the stream (because it lacks a proper codec?) - or am I just not getting the point?

    I've captured traffic between my computer and the DVR in question, and something is happening (at least that's what Wireshark says).

    https://www.dropbox.com/s/j65lige244kg8jt/rtsp_ffmpeg.pcap

    What am I doing wrong?

    Regards.

  • ffmpeg: Dynamically set output duration based on sliding text width

    18 January 2016, by John Whiteman

    I need to create a smooth ’news ticker’ on a low powered android device. Unfortunately this is impossible at runtime using HTML or native code as there is always some stutter or glitch.

    I’ve created a solution that gives me a smooth result by encoding an mp4 for each message and displaying one video after the other. This is the code I’m using:

    ffmpeg -f lavfi -i color=c=black:s=1280x100 -vf "drawtext=BebasNeue.otf:fontsize=60:fontcolor=white:y=h-line_h-30:x=-(4*n)+1280:text='Hello world'" -t 10 output.mp4

    Problem:
    I need to set the video’s duration dynamically so that the video stops when the text has completed it’s journey from right to left. The messages will be of varying lengths and I need each message to scroll at a constant speed (ie. an mp4 with a longer message would have a longer duration).

    Is this possible via an expression? If not is there some clever way I can calculate this outside of ffmpeg and pass it to the ’-t’ (duration) parameter?

    ** Edit **
    To calculate outside of ffmpeg I can do a calculation like video_width + text_width / video_fps (ie. 1280 + 262 / 25) to give me the duration. So now I’m just looking to see if this is possible within the ffmpeg command line itself. t

    Many thanks

  • armv6: Accelerate ff_fft_calc for general case (nbits != 4)

    16 July 2014, by Ben Avison
    armv6: Accelerate ff_fft_calc for general case (nbits != 4)

    The previous implementation targeted DTS Coherent Acoustics, which only
    
requires nbits == 4 (fft16()). This case was (and still is) linked directly
    
rather than being indirected through ff_fft_calc_vfp(), but now the full
    
range from radix-4 up to radix-65536 is available. This benefits other codecs
    
such as AAC and AC3.
    

    The implementaion is based upon the C version, with each routine larger than
    
radix-16 calling a hierarchy of smaller FFT functions, then performing a
    
post-processing pass. This pass benefits a lot from loop unrolling to
    
counter the long pipelines in the VFP. A relaxed calling standard also
    
reduces the overhead of the call hierarchy, and avoiding the excessive
    
inlining performed by GCC probably helps with I-cache utilisation too.
    

    I benchmarked the result by measuring the number of gperftools samples that
    
hit anywhere in the AAC decoder (starting from aac_decode_frame()) or
    
specifically in the FFT routines (fft4() to fft512() and pass()) for the
    
same sample AAC stream:
    

                  Before          After
    
              Mean   StdDev   Mean   StdDev  Confidence  Change
    
Audio decode  2245.5 53.1     1599.6 43.8    100.0%      +40.4%
    
FFT routines  940.6  22.0     348.1  20.8    100.0%      +170.2%
    

    Signed-off-by: Martin Storsjö <martin@martin.st>
    • [DH] libavcodec/arm/fft_init_arm.c
    • [DH] libavcodec/arm/fft_vfp.S

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