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    16 avril 2011, par

    MediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
    Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
    Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
    Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...)

  • Amélioration de la version de base

    13 septembre 2013

    Jolie sélection multiple
    Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
    Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...)

  • 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 (9560)

  • html5 video crashes in Chrome and Opera but works fine in Firefox

    31 octobre 2012, par SharkTheDark

    It's 5:30AM and I'm stuck with this video problem.

    I had it all working before, and now, it has problems...
    I try to change encoding, but it's still not good... Worst part is that it works when it's runned from local file, but won't from server ( I add file types in htaccess )...

    Here is the ffmpeg encoding :

    exec('ffmpeg -i "' .$tmpName. '" -bt 50k -b 250k -ab 56k -ac 2 -s 480x320 -vcodec libx264 -vpre hq -vpre ipod640 -acodec libfaac  "upload/' .$newName. '.mp4"');
    exec('ffmpeg -i "' .$tmpName. '" -bt 50k -b 250k -ab 56k -ac 2 -s 480x320 -vcodec libvpx -acodec libvorbis -f webm "upload/' .$newName. '.webm"');
    exec('ffmpeg -i "' .$tmpName. '" -bt 50k -b 250k -ab 56k -ac 2 -s 480x320 -vcodec libtheora -acodec libvorbis -aq 100 "upload/' .$newName. '.ogv"');
    exec('ffmpeg -i "' .$tmpName. '" -bt 50k -b 250k -ab 56k -ac 2 -s 480x320 -vcodec libtheora -acodec libvorbis -aq 100 "upload/' .$newName. '.ogg"');

    or :

    exec('ffmpeg -i "' .$tmpName. '" "upload/' .$newName. '.mp4"');
    exec('ffmpeg -i "' .$tmpName. '" -vcodec libvpx -acodec libvorbis -f webm  -aq 100 "upload/' .$newName. '.webm"');
    exec('ffmpeg -i "' .$tmpName. '" -vcodec libtheora -acodec libvorbis -aq 100 "upload/' .$newName. '.ogv"');
    exec('ffmpeg -i "' .$tmpName. '" -vcodec libtheora -acodec libvorbis -aq 100 "upload/' .$newName. '.ogg"');

    or :

    exec('ffmpeg -i "' .$tmpName. '" "upload/' .$newName. '.mp4"');
    exec('ffmpeg -i "' .$tmpName. '" "upload/' .$newName. '.webm"');
    exec('ffmpeg -i "' .$tmpName. '" "upload/' .$newName. '.ogv"');
    exec('ffmpeg -i "' .$tmpName. '" "upload/' .$newName. '.ogg"');

    Nothing create good enough file to play in Chrome...

    URL of file(s) :

    http://urthots.com/homepage/upload/9b8a0c1a5af65c92936e3c6806181ec3.ogv
    http://urthots.com/homepage/upload/9b8a0c1a5af65c92936e3c6806181ec3.ogg
    http://urthots.com/homepage/upload/9b8a0c1a5af65c92936e3c6806181ec3.webm
    http://urthots.com/homepage/upload/f41355af09218d660a5d78a4012a7177.mp4

    Video element :

    <video poster="http://urthots.com/homepage/upload/images/xxx.jpg">
    <source src="http://urthots.com/homepage/upload/xxx.mp4" type="video/mp4"></source>
    <source src="http://urthots.com/homepage/upload/xxx.webm" type="video/webm"></source>
    <source src="http://urthots.com/homepage/upload/xxx.ogv" type="video/ogg; codecs=theora,vorbis"></source>
    <source src="http://urthots.com/homepage/upload/xxx.ogg" type="video/ogg; codecs=theora,vorbis"></source>
    </video>

    What can cause this ?
    I tried everything, encoding changing, HTML element changing, htaccess changing, I tried with videos from other sites, and they are working okay...
    I have no idea what is problem...

    Can someone please help...

  • Decode AAC to PCM with ffmpeg on android

    26 novembre 2012, par JeffG

    I have built ffmpeg 0.8.12 (love) with the android NDK (r8c) on ubuntu.
    I then use the generated library in another android application through JNI.

    Essentially what I want to do is pass a byte stream from java to my c jni function and use ffmpeg to decode it into a PCM audio buffer which will then be passed back to java to be played using Android's AudioTrack. I can successfully pass the buffer through to jni (have checked the values) and ffmpeg seems to initialise correctly, but when it tries to decode the first frame, it throws an error in the aac_decode_frame_int method in aacdec.c "channel element 0.0 is not allocated". The aac file plays fine and is valid.

    Here is my jni code to do the decoding

    jint Java_com_example_testffmpeg_MainActivity_decodeAacBytes(JNIEnv * env,
           jobject this, jbyteArray input, jint numBytes) {

       //copy bytes from java
       jbyte* bufferPtr = (*env)->GetByteArrayElements(env, input, NULL);
       uint8_t inputBytes[numBytes + FF_INPUT_BUFFER_PADDING_SIZE];
       memset(inputBytes, 0, numBytes + FF_INPUT_BUFFER_PADDING_SIZE);
       memcpy(inputBytes, bufferPtr, numBytes);
       (*env)->ReleaseByteArrayElements(env, input, bufferPtr, 0);

       av_register_all();

       AVCodec *codec = avcodec_find_decoder(CODEC_ID_AAC);

       if (codec == NULL) {
           LOGE("Cant find AAC codec\n");
           return 0;
       }
       LOGI("AAC codec found\n");

       AVCodecContext *avCtx = avcodec_alloc_context();

       if (avCtx == NULL) {
           LOGE("Could not allocate codec context\n");
           return 0;
       }
       LOGI("codec context allocated\n");

       if (avcodec_open2(avCtx, codec, NULL) &lt; 0) {
           LOGE("Could not open codec\n");
           return 0;
       }
       LOGI("AAC codec opened");

       //the input buffer
       AVPacket avPacket;
       av_init_packet(&amp;avPacket);

       LOGI("AVPacket initialised\n");

       avPacket.size = numBytes; //input buffer size
       avPacket.data = inputBytes; // the input buffer

       int outSize;
       int len;
       uint8_t *outbuf = malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);

       while (avPacket.size > 0) {
           outSize = AVCODEC_MAX_AUDIO_FRAME_SIZE;
           len = avcodec_decode_audio3(avCtx, (short *) outbuf, &amp;outSize,
                   &amp;avPacket);

           if (len &lt; 0) {
               LOGE("Error while decoding\n");
               return 0;
           }

           if (outSize > 0) {
               LOGI("Decoded some stuff\n");
           }

           avPacket.size -= len;
           avPacket.data += len;
       }

       LOGI("Freeing memory\n");

       av_free_packet(&amp;avPacket);
       avcodec_close(avCtx);
       av_free(avCtx);

       return 0;
    }

    The problem occurs in the call to avcodec_decode_audio3, when the decoding first occurs. I have stepped through the ffmpeg code, but can't find the problem. Any help would be greatly appreciated !

  • Adding C64 SID Music

    1er novembre 2012, par Multimedia Mike — General

    I have been working on adding support for SID files — the music format for the Commodore 64 — to the game music website for awhile. I feel a bit out of my element since I’m not that familiar with the C64. But why should I let that slow me down ? Allow me to go through the steps I have previously outlined in order to make this happen.



    I need to know what picture should represent the system in the search results page. The foregoing picture should be fine, but I’m getting way ahead of myself.

    Phase 1 is finding adequate player software. The most venerable contender in this arena is libsidplay, or so I first thought. It turns out that there’s libsidplay (originally hosted at Geocities, apparently, and no longer on the net) and also libsidplay2. Both are kind of old (libsidplay2 was last updated in 2004). I tried to compile libsidplay2 and the C++ didn’t agree with current version of g++.

    However, a recent effort named libsidplayfp is carrying on the SID emulation tradition. It works rather well, notwithstanding the fact that compiling the entire library has a habit of apparently hanging the Linux VM where I develop this stuff.

    Phase 2 is to develop a testbench app around the playback library. With the help of the libsidplayfp library maintainers, I accomplished this. The testbench app consistently requires about 15% of a single core of a fairly powerful Core i7. So I look forward to recommendations that I port that playback library to pure JavaScript.

    Phase 3 is plug into the web player. I haven’t worked on this yet. I’m confident that this will work since phase 2 worked (plus, I have a plan to combine phases 2 and 3).

    One interesting issue that has arisen is that proper operation of libsidplayfp requires that 3 C64 ROM files be present (the, ahem, KERNAL, BASIC interpreter, and character generator). While these are copyrighted ROMs, they are easily obtainable on the internet. The goal of my project is to eliminate as much friction as possible for enjoying these old tunes. To that end, I will just bake the ROM files directly into the player.

    Phase 4 is collecting a SID song corpus. This is the simplest part of the whole process thanks to the remarkable curation efforts of the High Voltage SID Collection (HVSC). Anyone can download a giant archive of every known SID file. So that’s a done deal.

    Or is it ? One small issue is that I was hoping that the first iteration of my game music website would focus on, well, game music. There is a lot of music in the HVSC that are original compositions or come from demos. The way that the archive is organized makes it difficult to automatically discern whether a particular SID file comes from a game or not.

    Phase 5 is munging the metadata. The good news here is that the files have the metadata built in. The not-so-great news is that there isn’t quite as much as I might like. Each file is tagged with title, author, and publisher/copyright. If there is more than one song in a file, they all have the same metadata. Fortunately, if I can import them all into my game music database, there is an opportunity to add a lot more metadata.

    Further, there is no play length metadata for these files. This means I will need to set each to a default length like 2 minutes and do something like I did before in order to automatically determine if any songs terminate sooner.

    Oddly, the issue I’m most concerned about is character encoding. This is the first project for which I’m making certain that I understand character encoding since I can’t reasonably get away with assuming that everything is ASCII. So far, based on the random sampling of SID files I have checked, there is a good chance of encountering metadata strings with characters that are not in the lower ASCII set. From what I have observed, these characters map to Unicode code points. So I finally get to learn about manipulating strings in such a way that it preserves the character encoding. At the very least, I need Python to rip the strings out of the binary SID files and make sure the Unicode remains intact while being inserted into an SQLite3 database.