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  • Support de tous types de médias

    10 avril 2011

    Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)

  • Supporting all media types

    13 avril 2011, par

    Unlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)

  • 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 (3238)

  • Cracking Aztec Game Audio

    7 juin 2011, par Multimedia Mike — Game Hacking

    Here’s a mild multimedia-related reverse engineering challenge for you. It’s pretty straightforward for those skilled in the art.

    The Setup
    One side effect of running this ridiculously niche interest blog at the intersection of multimedia, reverse engineering, and game hacking is that people occasionally contact me for assistance on those very matters. So it was when one of my MobyGames peers asked if I can help to extract some music from a game called Aztec Wars. The game consists of 2 discs, each with a music.xbe file that contains multiple tunes and is hundreds of megabytes large.



    That’s all the data I received from the first email. At first I’m wondering what makes people think I have some magical insight into cracking these formats with such little information. Ordinarily, I would need to have the entire data file to work with and possibly the game binaries. But I didn’t want to ask him to upload hundreds of megabytes of data and I didn’t feel like downloading it ; commitment issues and all.

    But then I gathered a little confidence and remembered that the .xbe files are probably just Game Resource Archive Formats (GRAF) which are, traditionally, absurdly simple. I asked my colleague to send me a hexdump of the first kilobyte of one of the .xbe GRAFs ('hexdump -C -n 1024 music.xbe > file') as well as the total file size of the GRAF.

    The Hexdump
    The first music.xbe file is 192817376 bytes large. These are the first 1024 144 bytes (more than enough) :

    00000000  01 00 00 00 60 04 00 00  14 00 00 00 01 00 00 00  |....`...........|
    00000010  0d 00 00 00 48 00 00 00  94 39 63 01 1c a4 21 03  |....H....9c..¤ !.|
    00000020  7a d2 54 04 04 28 ad 05  d8 88 fd 06 d8 88 fd 06  |zÒT..(­.Ø.ý.Ø.ý.|
    00000030  2a 6e 46 08 2a 6e 46 08  2a 6e 46 08 2a 6e 46 08  |*nF.*nF.*nF.*nF.|
    00000040  50 13 2f 0a e0 28 7e 0b  52 49 46 46 44 39 63 01  |P./.à( .RIFFD9c.|
    00000050  57 41 56 45 66 6d 74 20  10 00 00 00 01 00 02 00  |WAVEfmt ........|
    00000060  44 ac 00 00 10 b1 02 00  04 00 10 00 64 61 74 61  |D¬...±......data|
    00000070  fc 13 63 01 00 00 00 00  00 00 00 00 00 00 00 00  |ü.c.............|
    00000080  00 00 00 00 00 00 00 00  00 00 00 00 00 00 00 00  |................|
    

    The Challenge
    Armed with only the information in the foregoing section, figure out a method for extracting all the audio files in that file and advise on their playback/conversion. Ideally, this method should require minimal effort from both you and the person on the other end of the conversation.

    The Resolution
    The reason I ask is because I came up with a solution but knew, deep down, that there must be a slightly easier way. How would you solve this ?

    The music files in question are now preserved on YouTube (until they see fit to remove them for one reason or another).

  • Android : How to play MP3 files with ID3 tags using FFMPEG

    16 juillet 2012, par Vipul Purohit

    I am trying to play various sound files using FFMPEG lib in android. I have managed to play most files but some file Example : MP3 files with ID3 tag are not playing. I tried playing files by taking reference of various example but didn't get any success.

    Here is my native side code :

    #include
    #include
    #include
    #include <android></android>log.h>

    #include "libavcodec/avcodec.h"
    #include "libavformat/avformat.h"

    #define LOG_TAG "mylib"
    #define LOGI(...)  __android_log_print(ANDROID_LOG_INFO, LOG_TAG, __VA_ARGS__)
    #define LOGE(...)  __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__)

    #define AUDIO_INBUF_SIZE 20480
    #define AUDIO_REFILL_THRESH 4096

    void Java_ru_dzakhov_ffmpeg_test_MainActivity_createEngine(JNIEnv* env, jclass clazz)
       {
           avcodec_init();

           av_register_all();


       }

       jstring Java_ru_dzakhov_ffmpeg_test_MainActivity_loadFile(JNIEnv* env, jobject obj,jstring file,jbyteArray array)
       {
           {
                   jboolean            isfilenameCopy;
                   const char *        filename = (*env)->GetStringUTFChars(env, file, &amp;isfilenameCopy);
                   int audioStreamIndex;
                   AVCodec *codec;
                   AVCodecContext *c= NULL;
                   AVFormatContext * pFormatCtx;
                   AVCodecContext * aCodecCtx;
                   int out_size, len, audioStream=-1, i, err;
                   FILE *f, *outfile;
                   uint8_t *outbuf;
                   uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
                   AVPacket avpkt;
                   jclass              cls = (*env)->GetObjectClass(env, obj);
                   jmethodID           play = (*env)->GetMethodID(env, cls, "playSound", "([BI)V");//At the begining of your main function

                   LOGE("source file name is %s", filename);

                      avcodec_init();

                      av_register_all();

                      LOGE("Stage 1");
                      /* get format somthing of source file to AVFormatContext */
                      int lError;
                      if ((lError = av_open_input_file(&amp;pFormatCtx, filename, NULL, 0, NULL)) !=0 ) {
                          LOGE("Error open source file: %d", lError);
                          exit(1);
                      }
                      if ((lError = av_find_stream_info(pFormatCtx)) &lt; 0) {
                          LOGE("Error find stream information: %d", lError);
                          exit(1);
                      }
                      LOGE("Stage 1.5");
                      LOGE("audio format: %s", pFormatCtx->iformat->name);
                      LOGE("audio bitrate: %d", pFormatCtx->bit_rate);



                      audioStreamIndex = av_find_best_stream(pFormatCtx, AVMEDIA_TYPE_AUDIO, -1, -1, &amp;codec, 0);

    //                 audioStreamIndex=-1;
    //                 LOGE("nb_stream %d", pFormatCtx->nb_streams);
    //                     for(i=0; inb_streams; i++)
    //                     {
    //                         LOGE("nb_stream %d", pFormatCtx->nb_streams);
    //                         if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO)
    //                         {
    //                             audioStream=i;
    //                             break;
    //                         }
    //                     }
    //
    //                     if(audioStream==-1)
    //                     {
    //                         LOGE("Cannot find audio");
    //                     }

                         LOGE("audioStreamIndex %d", audioStreamIndex);
    //                    if (audioStreamIndex == AVERROR_STREAM_NOT_FOUND) {
    //                        LOGE(1, "cannot find a audio stream");
    //                        exit(1);
    //                    } else if (audioStreamIndex == AVERROR_DECODER_NOT_FOUND) {
    //                        LOGE(1, "audio stream found, but no decoder is found!");
    //                        exit(1);
    //                    }
                         LOGE("audio codec: %s", codec->name);

                         /* get codec somthing of audio stream to AVCodecContext */
                         aCodecCtx = pFormatCtx->streams[audioStreamIndex]->codec;
                         if (avcodec_open(aCodecCtx, codec) &lt; 0) {
                         LOGE("cannot open the audio codec!");
                             exit(1);
                         }

                   printf("Audio decoding\n");
                   LOGE("Stage 1.7");
                   LOGE("S");
                   codec = avcodec_find_decoder(aCodecCtx->codec_id);
                   LOGE("Stage 1.8");
                   if (!codec) {
                       LOGE("codec not found\n");
                       exit(1);
                   }
                   LOGE("Stage 2");
    //              c= avcodec_alloc_context();
                   LOGE("Stage 3");
                   /* open it */
                   if (avcodec_open(aCodecCtx, codec) &lt; 0) {
                       LOGE("could upper");
                       fprintf(stderr, "could not open codec\n");
                       LOGE("could not open codec");

                   }
                   LOGE("Stage 4");
                   outbuf = malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);

                   f = fopen(filename, "rb");
                   if (!f) {
                       fprintf(stderr, "could not open %s\n", filename);
                       LOGE("could not open");
                       exit(1);
                   }

                   /* decode until eof */
                   avpkt.data = inbuf;
                   avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
                   LOGE("Stage 5");


                   while (avpkt.size > 0) {
    //                  LOGE("Stage 6");
                       out_size = (AVCODEC_MAX_AUDIO_FRAME_SIZE/3)*2;
                       len = avcodec_decode_audio3(aCodecCtx, (int16_t *)outbuf, &amp;out_size, &amp;avpkt);
                       LOGE("data_size %d len %d", out_size, len);
                       if (len &lt; 0) {
                           fprintf(stderr, "Error while decoding\n");
                           LOGE("DECODING ERROR");
                           LOGE("DECODING ERROR %d", len);
                           exit(1);
                       }
    //                  LOGE("Stage 7");
                       if (out_size > 0) {
                           /* if a frame has been decoded, output it */
    //                      LOGE("Stage 8");
                           jbyte *bytes = (*env)->GetByteArrayElements(env, array, NULL);
                           memcpy(bytes, outbuf, out_size); //
                           (*env)->ReleaseByteArrayElements(env, array, bytes, 0);
                           (*env)->CallVoidMethod(env, obj, play, array, out_size);
    //                      LOGE("DECODING ERROR5");
                       }
                       LOGE("Stage 9");
                       avpkt.size -= len;
                       avpkt.data += len;
                       if (avpkt.size &lt; AUDIO_REFILL_THRESH) {
                           /* Refill the input buffer, to avoid trying to decode
                            * incomplete frames. Instead of this, one could also use
                            * a parser, or use a proper container format through
                            * libavformat. */
                           memmove(inbuf, avpkt.data, avpkt.size);
                           avpkt.data = inbuf;
                           len = fread(avpkt.data + avpkt.size, 1,
                                       AUDIO_INBUF_SIZE - avpkt.size, f);
                           if (len > 0)
                               avpkt.size += len;
                       }
                   }
                   LOGE("Stage 12");
                   fclose(f);
                   free(outbuf);

                   avcodec_close(c);
                   av_free(c);

           }
       }

    Sorry for a rough code but I am testing on this code.

    I found the reason that the mp3 file has ID3 info, but I don't handle this, all the ID3 info data is decode as frame data, so the right frame header sign 0xFFE00000 can't be found and the method "avcodec_decode_audio3" return -1.

    I found this in question mp3 decoding using ffmpeg API (Header missing)

    If anyone can help me handle and play file with ID3 tags in MP3 and similar for other formats.

  • Working way to make video from images in C#

    29 août 2011, par Jim Mischel

    Does anybody have a known reliable way to create a video from a series of image files ? Before you mod me down for not searching for the answer before posting the question, and before you fire off a simple message like "use FFMPEG," read the rest of this message.

    I'm trying to create a video, it doesn't matter too much what format as long as it's widely supported, from a series of images (.jpg, .bmp, etc.). My platform is Windows Server 2008, 64-bit. If I can make the video from within my C# program, that's great, but I'm not averse to writing a series of image files to a directory and then firing off an external program to make a video from those images.

    The only constraints are : it must work on my Windows Server 2008 system, and be scriptable. That is, no GUI programs that require operator intervention.

    I found a number of similar questions on StackOverflow, and have tried several of the solutions, all with varying degrees of frustration and none with anything like success.

    FFMPEG looks like a great program. Maybe it is, on Linux. The two Windows builds I downloaded are broken. Given this command line :

     ffmpeg -r 1 -f image2 -i jpeg\*.jpg video.avi

    One of the builds reads the images and then crashes due to data execution prevention. The other reads the first file and then spits out an error message that says "cannot find suitable codec for file jpeg/image2.jpg". Helpful, that. In any case, FFMPEG looks like a non-starter under Windows.

    One answer to a previous posting recommended Splicer . It looks like pretty good code. I compiled the samples and tried to run, but got some cryptic error message about a file not found. It looks like a COM class isn't registered. I suppose I need to install something (DirectShow, maybe, although I thought that was already installed ?). Depending on what's required, I might have a difficult time justifying its installation on a server. ("What ? Why do you need that on a server ?")

    Another answer suggested the AviFile library from Code Project. That looks simple enough : a wrapper around the Windows AviFile subsystem. Except that the AVI files the package creates appear to have all of the frames, but only the first frame shows when I play the AVI in Windows Media Player. Well, that and if you try to create a compressed video, the program throws an exception.

    So, I'm left wondering if there is a good, reliable way to do what I want : on a Windows system, create an AVI or other common video file format from a series of images, either through a .NET API or using an external program. Any help ?