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  • Participer à sa traduction

    10 avril 2011

    Vous pouvez nous aider à améliorer les locutions utilisées dans le logiciel ou à traduire celui-ci dans n’importe qu’elle nouvelle langue permettant sa diffusion à de nouvelles communautés linguistiques.
    Pour ce faire, on utilise l’interface de traduction de SPIP où l’ensemble des modules de langue de MediaSPIP sont à disposition. ll vous suffit de vous inscrire sur la liste de discussion des traducteurs pour demander plus d’informations.
    Actuellement MediaSPIP n’est disponible qu’en français et (...)

  • L’utiliser, en parler, le critiquer

    10 avril 2011

    La première attitude à adopter est d’en parler, soit directement avec les personnes impliquées dans son développement, soit autour de vous pour convaincre de nouvelles personnes à l’utiliser.
    Plus la communauté sera nombreuse et plus les évolutions seront rapides ...
    Une liste de discussion est disponible pour tout échange entre utilisateurs.

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

Sur d’autres sites (11314)

  • Theatrical quality ffmpeg/x264 encoding of a high-motion 1080p video

    2 décembre 2011, par Ian

    I've been struggling with encoding videos using FFMPEG and x264. The output stutters when played back in Quicktime, while in VLC it shows a lot of compression artifacts at the same places Quicktime stutters. So it seems like Quicktime is stuttering because it's trying to suppress the corruption/artifacts.

    The videos have a lot of random motion in them, including frames where 75% of the pixels will change at a random interval (the video is software generated so it's truly pseudo-random). The compression seems to be choking in these places where it's likely detecting a "scene cut" incorrectly. It also seems to choke at regular intervals where I guess it's doing a keyframe.

    I've based my encoding preset off of the x264-hq preset that comes with FFMPEG. I've tried turning off scene cut detection, and playing with the keyint/g and keyint_min options. Setting g to 1 makes it work, but blows out the filesize. I've tried the lossless presets, but they won't playback at all in Quicktime. Oddly, I haven't had any problems when working with a lower-resolution test video (1440x810).

    Here's the preset I have right now, which works, but yields a file that's approximately 60% larger than the (non-working) hq preset yields. Is there any way to improve upon this ? The filesize doesn't matter much, I just want something that will playback anywhere and be very high quality.

    coder=1
    flags=+loop
    cmp=+chroma
    partitions=+parti8x8+parti4x4+partp8x8+partp4x4+partb8x8
    me_method=umh
    subq=8
    me_range=16
    g=1
    keyint_min=1
    sc_threshold=0
    i_qfactor=0.71
    b_strategy=1crf=20
    qcomp=0.6
    qmin=20
    qmax=51
    qdiff=4
    bf=16
    refs=4
    trellis=1
    flags2=+dct8x8+wpred+bpyramid+mixed_refs
    wpredp=2
    

    Here's the command :

    ffmpeg \
      -r 60 -i "frame-%06d.tiff" \
      -vcodec libx264 -vpre my_preset \
      -threads 0 \
      -r 60 -an -f out.mp4
    
  • lavc : drop encode() support for video.

    23 février 2012, par Anton Khirnov

    lavc : drop encode() support for video.

  • Decode audio and video and process both flows — ffmpeg, sdl, opencv

    24 février 2012, par Eric

    My goal is to proceed on audio and video of mpeg-2 file independently, and to keep synchronicity on both flows. Duration of video is about 1 or 2 minutes maximum.

    1. First, following this post "opencv for reading videos (and do the process),ffmpeg for audio , and SDL used to play both" sounds perfect. I have done some modification on the code considering recent ffmpeg naming changes. Compilation with cmake on 64-bits machine is fine. I get an error "Unsupported codec [3]" when opening codec.
      The code is following.

    2. Second, I looking for code dealing with synchronicity on both flows.


    #include "opencv/highgui.h"
    #include "opencv/cv.h"

    #ifndef INT64_C
    #define INT64_C(c) (c ## LL)
    #define UINT64_C(c) (c ## ULL)
    #endif

    extern "C"{
    #include <sdl></sdl>SDL.h>
    #include <sdl></sdl>SDL_thread.h>
    #include <libavcodec></libavcodec>avcodec.h>
    #include <libavformat></libavformat>avformat.h>
    }

    #include <iostream>
    #include
    #include

    using namespace cv;

    #define SDL_AUDIO_BUFFER_SIZE 1024

    typedef struct PacketQueue
    {
      AVPacketList *first_pkt, *last_pkt;
      int nb_packets;
      int size;
      SDL_mutex *mutex;
      SDL_cond *cond;
    } PacketQueue;
    PacketQueue audioq;

    int audioStream = -1;
    int videoStream = -1;
    int quit = 0;

    SDL_Surface* screen = NULL;
    SDL_Surface* surface = NULL;

    AVFormatContext* pFormatCtx = NULL;
    AVCodecContext* aCodecCtx = NULL;
    AVCodecContext* pCodecCtx = NULL;

    void show_frame(IplImage* img){
      if (!screen){
         screen = SDL_SetVideoMode(img->width, img->height, 0, 0);
         if (!screen){
            fprintf(stderr, "SDL: could not set video mode - exiting\n");
            exit(1);
         }
      }
      // Assuming IplImage packed as BGR 24bits
      SDL_Surface* surface = SDL_CreateRGBSurfaceFrom((void*)img->imageData,
                                                      img->width,
                                                      img->height,
                                                      img->depth * img->nChannels,
                                                      img->widthStep,
                                                      0xff0000, 0x00ff00, 0x0000ff, 0
                                                     );

      SDL_BlitSurface(surface, 0, screen, 0);
      SDL_Flip(screen);
    }

    void packet_queue_init(PacketQueue *q){
      memset(q, 0, sizeof(PacketQueue));
      q->mutex = SDL_CreateMutex();
      q->cond = SDL_CreateCond();
    }

    int packet_queue_put(PacketQueue *q, AVPacket *pkt){
      AVPacketList *pkt1;
      if (av_dup_packet(pkt) &lt; 0){
         return -1;
      }

      pkt1 = (AVPacketList*) av_malloc(sizeof(AVPacketList));
      //pkt1 = (AVPacketList*) malloc(sizeof(AVPacketList));
      if (!pkt1) return -1;
      pkt1->pkt = *pkt;
      pkt1->next = NULL;

      SDL_LockMutex(q->mutex);

      if (!q->last_pkt)
         q->first_pkt = pkt1;
      else
         q->last_pkt->next = pkt1;

      q->last_pkt = pkt1;
      q->nb_packets++;
      q->size += pkt1->pkt.size;
      SDL_CondSignal(q->cond);

      SDL_UnlockMutex(q->mutex);
      return 0;
    }

    static int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block){
      AVPacketList *pkt1;
      int ret;

      SDL_LockMutex(q->mutex);
      for (;;){
         if( quit){
            ret = -1;
            break;
         }

         pkt1 = q->first_pkt;
         if (pkt1){
            q->first_pkt = pkt1->next;
            if (!q->first_pkt)
               q->last_pkt = NULL;

            q->nb_packets--;
            q->size -= pkt1->pkt.size;
            *pkt = pkt1->pkt;
            av_free(pkt1);
            //free(pkt1);
            ret = 1;
            break;
         }

         else if (!block){
            ret = 0;
            break;
         }
         else{
            SDL_CondWait(q->cond, q->mutex);
         }
      }

      SDL_UnlockMutex(q->mutex);
      return ret;
    }

    int audio_decode_frame(AVCodecContext *aCodecCtx, uint8_t *audio_buf, int buf_size){
      static AVPacket pkt;
      static uint8_t *audio_pkt_data = NULL;
      static int audio_pkt_size = 0;

      int len1, data_size;

      for (;;){
         while (audio_pkt_size > 0){
            data_size = buf_size;
            len1 = avcodec_decode_audio3(aCodecCtx, (int16_t*)audio_buf, &amp;data_size, &amp;pkt);
            if (len1 &lt; 0){
               // if error, skip frame
               audio_pkt_size = 0;
               break;
            }
            audio_pkt_data += len1;
            audio_pkt_size -= len1;
            if (data_size &lt;= 0){
               // No data yet, get more frames
               continue;
            }
            // We have data, return it and come back for more later
            return data_size;
        }

        if (pkt.data)
           av_free_packet(&amp;pkt);
        if (quit) return -1;
        if (packet_queue_get(&amp;audioq, &amp;pkt, 1) &lt; 0) return -1;
        audio_pkt_data = pkt.data;
        audio_pkt_size = pkt.size;
     }
    }

    void audio_callback(void *userdata, Uint8 *stream, int len){
     AVCodecContext *aCodecCtx = (AVCodecContext *)userdata;
     int len1, audio_size;

     static uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
     static unsigned int audio_buf_size = 0;
     static unsigned int audio_buf_index = 0;

     while (len > 0){
        if (audio_buf_index >= audio_buf_size){
           // We have already sent all our data; get more
           audio_size = audio_decode_frame(aCodecCtx, audio_buf, sizeof(audio_buf));
           if(audio_size &lt; 0){
              // If error, output silence
              audio_buf_size = 1024; // arbitrary?
              memset(audio_buf, 0, audio_buf_size);
           }
           else{
              audio_buf_size = audio_size;
           }
           audio_buf_index = 0;
       }

       len1 = audio_buf_size - audio_buf_index;
       if (len1 > len)
          len1 = len;
       memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1);
       len -= len1;
       stream += len1;
       audio_buf_index += len1;
     }
    }

        void setup_ffmpeg(char* filename)
        {
           if (avformat_open_input(&amp;pFormatCtx, filename, NULL, NULL) != 0){
              fprintf(stderr, "FFmpeg failed to open file %s!\n", filename);
              exit(-1);
           }

           if (av_find_stream_info(pFormatCtx) &lt; 0){
              fprintf(stderr, "FFmpeg failed to retrieve stream info!\n");
              exit(-1);
           }

           // Dump information about file onto standard error
           av_dump_format(pFormatCtx, 0, filename, 0);

           // Find the first video stream
           int i = 0;
           for (i; i &lt; pFormatCtx->nb_streams; i++){
              if (pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO &amp;&amp; videoStream &lt; 0){
                 videoStream = i;
              }

              if (pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO &amp;&amp; audioStream &lt; 0){
                 audioStream = i;
              }
           }

           if (videoStream == -1){
              fprintf(stderr, "No video stream found in %s!\n", filename);
              exit(-1);
           }

           if (audioStream == -1){
              fprintf(stderr, "No audio stream found in %s!\n", filename);
              exit(-1);
           }

           // Get a pointer to the codec context for the audio stream
           aCodecCtx = pFormatCtx->streams[audioStream]->codec;

           // Set audio settings from codec info
           SDL_AudioSpec wanted_spec;
           wanted_spec.freq = aCodecCtx->sample_rate;
           wanted_spec.format = AUDIO_S16SYS;
           wanted_spec.channels = aCodecCtx->channels;
           wanted_spec.silence = 0;
           wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
           wanted_spec.callback = audio_callback;
           wanted_spec.userdata = aCodecCtx;

           SDL_AudioSpec spec;
           if (SDL_OpenAudio(&amp;wanted_spec, &amp;spec) &lt; 0){
              fprintf(stderr, "SDL_OpenAudio: %s\n", SDL_GetError());
              exit(-1);
           }

           AVCodec* aCodec = avcodec_find_decoder(aCodecCtx->codec_id);
           if (!aCodec){
              fprintf(stderr, "Unsupported codec [1]!\n");
              exit(-1);
           }
           avcodec_open(aCodecCtx, aCodec);

           // audio_st = pFormatCtx->streams[index]
           packet_queue_init(&amp;audioq);
           SDL_PauseAudio(0);

           // Get a pointer to the codec context for the video stream
           pCodecCtx = pFormatCtx->streams[videoStream]->codec;

           // Find the decoder for the video stream
           AVCodec* pCodec = avcodec_find_decoder(pCodecCtx->codec_id);
           if (pCodec == NULL){
              fprintf(stderr, "Unsupported codec [2]!\n");
              exit(-1); // Codec not found
           }

           // Open codec
           if (avcodec_open(pCodecCtx, pCodec) &lt; 0){
              fprintf(stderr, "Unsupported codec [3]!\n");
              exit(-1); // Could not open codec
           }
        }


        int main(int argc, char* argv[])
        {
           if (argc &lt; 2){
               std::cout &lt;&lt; "Usage: " &lt;&lt; argv[0] &lt;&lt; " <video>" &lt;&lt; std::endl;
               return -1;
           }

           av_register_all();

           // Init SDL
           if (SDL_Init(SDL_INIT_VIDEO | SDL_INIT_AUDIO | SDL_INIT_TIMER))
           {
              fprintf(stderr, "Could not initialize SDL - %s\n", SDL_GetError());
              return -1;
           }

           // Init ffmpeg and setup some SDL stuff related to Audio
           setup_ffmpeg(argv[1]);

           VideoCapture cap(argv[1]);
           if (!cap.isOpened()){
              std::cout &lt;&lt; "Failed to load file!" &lt;&lt; std::endl;
              return -1;
           }

           AVPacket packet;
           while (av_read_frame(pFormatCtx, &amp;packet) >= 0)
           {
              if (packet.stream_index == videoStream)
              {
                 // Actually this is were SYNC between audio/video would happen.
                 // Right now I assume that every VIDEO packet contains an entire video frame, and that&#39;s not true. A video frame can be made by multiple packets!
                 // But for the time being, assume 1 video frame == 1 video packet,
                 // so instead of reading the frame through ffmpeg, I read it through OpenCV.

                 Mat frame;
                 cap >> frame; // get a new frame from camera

                 // do some processing on the frame, either as a Mat or as IplImage.
                 // For educational purposes, applying a lame grayscale conversion
                 IplImage ipl_frame = frame;
                 for (int i = 0; i &lt; ipl_frame.width * ipl_frame.height * ipl_frame.nChannels; i += ipl_frame.nChannels)
                 {
                    ipl_frame.imageData[i] = (ipl_frame.imageData[i] + ipl_frame.imageData[i+1] + ipl_frame.imageData[i+2])/3;   //B
                    ipl_frame.imageData[i+1] = (ipl_frame.imageData[i] + ipl_frame.imageData[i+1] + ipl_frame.imageData[i+2])/3; //G
                    ipl_frame.imageData[i+2] = (ipl_frame.imageData[i] + ipl_frame.imageData[i+1] + ipl_frame.imageData[i+2])/3; //R
                 }

                 // Display it on SDL window
                 show_frame(&amp;ipl_frame);

                 av_free_packet(&amp;packet);
              }
              else if (packet.stream_index == audioStream)
              {
                 packet_queue_put(&amp;audioq, &amp;packet);
              }
              else
              {
                 av_free_packet(&amp;packet);
              }

              SDL_Event event;
              SDL_PollEvent(&amp;event);
              switch (event.type)
              {
              case SDL_QUIT:
                 SDL_FreeSurface(surface);
                 SDL_Quit();
                 break;

              default:
                 break;
              }
           }

           // the camera will be deinitialized automatically in VideoCapture destructor

           // Close the codec
           avcodec_close(pCodecCtx);

           // Close the video file
           av_close_input_file(pFormatCtx);

           return 0;
        }
    </video></iostream>