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  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

  • Multilang : améliorer l’interface pour les blocs multilingues

    18 février 2011, par

    Multilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
    Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela.

  • Websites made ​​with MediaSPIP

    2 mai 2011, par

    This page lists some websites based on MediaSPIP.

Sur d’autres sites (10778)

  • 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>
  • Transcoding a video from GoToMeeting WMV to MP4 using FFMPEG on Windows locks up dos window [migrated]

    22 août 2011, par Ryan

    I've been tasked with converting some pre-recorded GoToMeeting videos from wmv to mp4 format. I've got the GoToMeeting transcoder app available from Citrix (g2mtranscoder.exe) that takes the source wmv and makes it so that ffmpeg can work with the wmv. The problem I have is when I then try and take that wmv and transcode it to mp4.

    I'm using the latest pre-compiled static version of ffmpeg for Windows from here http://ffmpeg.zeranoe.com/builds/

    Here is the command line I'm using to begin the conversion :

    "C:\ffmpeg\ffmpeg.exe" -i "999_1366_768_g2m_transcoded.wmv" -f "mp4" -b "1000k" -r "30" -ab "128k" -ar "22050" -s "1366x768" -strict "experimental" -y "999.mp4"

    Here is FFMPEG's output. The last line is where the dos window seems to stop executing/locks up (when I say lock up I mean just the dos window is locked up, not the entire machine) :

    ffmpeg version N-31932-g41bf67d, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 16 2011 18:54:12 with gcc 4.6.1
    configuration: --enable-gpl --enable-version3 --enable-memalign-hack --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
     libavutil    51. 12. 0 / 51. 12. 0
     libavcodec   53. 10. 0 / 53. 10. 0
     libavformat  53.  7. 0 / 53.  7. 0
     libavdevice  53.  3. 0 / 53.  3. 0
     libavfilter   2. 31. 1 /  2. 31. 1
     libswscale    2.  0. 0 /  2.  0. 0
     libpostproc  51.  2. 0 / 51.  2. 0
    [asf @ 01BBB600] max_analyze_duration 5000000 reached at 5194000

    Seems stream 1 codec frame rate differs from container frame rate: 1000.00 (1000/1) -> 59.92 (719/12)
    Input #0, asf, from &#39;D:\Inetpub\g2m\Transcoding\999_1366_768_g2m_transcoded.wmv&#39;:
     Metadata:
       WMFSDKVersion   : 10.00.00.4007
       WMFSDKNeeded    : 0.0.0.0000
       IsVBR           : 1
       VBR Peak        : 4403
       Buffer Average  : 1470
       WM/ToolVersion  : 4.8 Build 723
       WM/ToolName     : GoToMeeting
       BitRateFrom the writer: 1553
       Audio samples   : 6221
       Video samples   : 3455
       recording time  : Mon, 22 Aug 2011 11:48:42 Eastern Daylight Time
     Duration: 00:38:30.71, start: 0.000000, bitrate: 160 kb/s
       Stream #0.0: Audio: wmav2, 44100 Hz, 1 channels, s16, 48 kb/s
       Stream #0.1: Video: wmv3 (Main), yuv420p, 1366x768, 2154 kb/s, 59.92 tbr, 1k tbn, 1k tbc
    [buffer @ 01BBC120] w:1366 h:768 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param:
    Output #0, mp4, to &#39;\\nas4\FMS_APPLICATIONS\SVC\sk12\webinars\999.mp4&#39;:
     Metadata:
       WMFSDKVersion   : 10.00.00.4007
       WMFSDKNeeded    : 0.0.0.0000
       IsVBR           : 1
       VBR Peak        : 4403
       Buffer Average  : 1470
       WM/ToolVersion  : 4.8 Build 723
       WM/ToolName     : GoToMeeting
       BitRateFrom the writer: 1553
       Audio samples   : 6221
       Video samples   : 3455
       recording time  : Mon, 22 Aug 2011 11:48:42 Eastern Daylight Time
       encoder         : Lavf53.7.0
       Stream #0.0: Video: mpeg4, yuv420p, 1366x768, q=2-31, 1000 kb/s, 30 tbn, 30 tbc
       Stream #0.1: Audio: aac, 22050 Hz, 1 channels, s16, 128 kb/s
    Stream mapping:
     Stream #0.1 -> #0.0
     Stream #0.0 -> #0.1
    Press [q] to stop, [?] for help
    frame=  149 fps= 63 q=23.5 size=    1103kB time=00:00:04.96 bitrate=1818.6kbits/s dup=144 drop=0

    Does anyone have any idea why it's locking up the dos window and essentially failing to convert the file to mp4 ? I had a friend test it on a linux box and he was able to convert the video just fine.

    Thanks

  • Simple way to grab thumbnail of FLV in ASP.NET without changing permissions on server ?

    23 septembre 2011, par Rhys Causey

    I'm looking for a simple way to grab thumbnails of FLVs in ASP.NET, without having to change any permissions/settings on the server. Ideally, nothing is installed on the server machine, but if necessary, small tools such as FFmpeg are fine.

    I've tried FFmpeg using the command-line tool with Process.Start, but the same command that works in a Windows Forms application and from the command prompt does not work in ASP.NET (presumably because of permissions).

    I've also tried using TAO.FFmpeg, and it seems to be working most of the time, but fails randomly, and does not start working again until the machine is restarted. Even when I use the sample code (decoder.cs), it sometimes fails when I try to open multiple videos in a single request.

    If this isn't possible in a clean/straightforward way, I'm open to other suggestions.