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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 (...)

  • 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 (...)

  • Encoding and processing into web-friendly formats

    13 avril 2011, par

    MediaSPIP automatically converts uploaded files to internet-compatible formats.
    Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
    Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
    Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
    All uploaded files are stored online in their original format, so you can (...)

Sur d’autres sites (3639)

  • w32threads : Use newer thread synchronization functions when targeting Vista

    6 août 2014, par Martin Storsjö
    w32threads : Use newer thread synchronization functions when targeting Vista
    

    When explicitly targeting Vista or newer (which only happens if the
    caller explicitly sets _WIN32_WINNT to a high enough value via the
    extra cflags option - otherwise configure script sets
    - D_WIN32_WINNT=0x0502), we already unconditionally link to the
    ConditionVariable functions, since 4622f11f9.

    Similarly use the newer -Ex versions of CreateEvent, CreateSemaphore,
    InitializeCriticalSection and WaitForSingleObject, that all appeared
    in Vista. When building Windows Store applications, the older versions
    of these functions aren’t available, only the -Ex functions. When
    doing such a build, the user can set -D_WIN32_WINNT=0x0600 to
    forcibly use the newer functions instead.

    Signed-off-by : Martin Storsjö <martin@martin.st>

    • [DBH] compat/w32pthreads.h
  • samplerate conversion function fails to produce an audible sound but only small pieces of audio

    2 juillet 2014, par user3749290

    playmp3() using libmpg123

    if (isPaused==0 &amp;&amp; mpg123_read(mh, buffer, buffer_size, &amp;done) == MPG123_OK)
    {
       char * resBuffer=&amp;buffer[0]; //22100=0,5s
       buffer = resample(resBuffer,22050,22050); // I think the result is 1/2 of audio speed
       if((ao_play(dev, (char*)buffer, done)==0)){
           return 1;
    }

    resample() Using avcodec from ffmpeg

    #define LENGTH_MS 500       // how many milliseconds of speech to store 0,5s:x=1:44100 x=22050 sample to store
    #define RATE 44100      // the sampling rate (input)
    #define FORMAT PA_SAMPLE_S16NE  // sample size: 8 or 16 bits
    #define CHANNELS 2      // 1 = mono 2 = stereo

    struct AVResampleContext* audio_cntx = 0;
    //(LENGTH_MS*RATE*16*CHANNELS)/8000

       void resample(char in_buffer[],int out_rate,int nsamples,char out_buffer[])
       {
           //char out_buffer[ sizeof( in_buffer ) * 4];
           audio_cntx = av_resample_init( out_rate, //out rate
               RATE, //in rate
               16, //filter length
               10, //phase count
               0, //linear FIR filter
               1.0 ); //cutoff frequency
           assert( audio_cntx &amp;&amp; "Failed to create resampling context!");
           int samples_consumed;
           //*out_buffer = malloc(sizeof(in_buffer));
           int samples_output = av_resample( audio_cntx, //resample context
               (short*)out_buffer, //buffout
               (short*)in_buffer,  //buffin
               &amp;samples_consumed,  //&amp;consumed
               nsamples,       //nb_samples
               sizeof(out_buffer)/2,//lenout sizeof(out_buffer)/2
               0);//is_last
           assert( samples_output > 0 &amp;&amp; "Error calling av_resample()!" );
           av_resample_close( audio_cntx );    
       }

    When I run this code, the application part, the problem is that I hear the sound jerky, why ?
    The size of the array I think is right, I calculated considering that in the second half should be 22050 samples from store.

  • Segmentation fault in samplerate conversion function

    17 juin 2014, par user3749290

    playmp3() using libmpg123

    if (isPaused==0 &amp;&amp; mpg123_read(mh, buffer, buffer_size, &amp;done) == MPG123_OK)
    {
       char * resBuffer=&amp;buffer[0]; //22100=0,5s
       buffer = resample(resBuffer,22100,22100);
       if((ao_play(dev, (char*)buffer, done)==0)){
           return 1;
    }

    resample() Using avcodec from ffmpeg

    #define LENGTH_MS 500       // how many milliseconds of speech to store
    #define RATE 44100      // the sampling rate (input)
    #define FORMAT PA_SAMPLE_S16NE  // sample size: 8 or 16 bits
    #define CHANNELS 2      // 1 = mono 2 = stereo

    struct AVResampleContext* audio_cntx = 0;

    char * resample(char in_buffer[(LENGTH_MS*RATE*16*CHANNELS)/8000],int out_rate,int nsamples)
    {
       char out_buffer[ sizeof( in_buffer ) * 4];
       audio_cntx = av_resample_init( out_rate, //out rate
           RATE, //in rate
           16, //filter length
           10, //phase count
           0, //linear FIR filter
           1.0 ); //cutoff frequency
       assert( audio_cntx &amp;&amp; "Failed to create resampling context!");
       int samples_consumed;
       int samples_output = av_resample( audio_cntx, //resample context
           (short*)out_buffer, //buffout
           (short*)in_buffer,  //buffin
           &amp;samples_consumed,  //&amp;consumed
           nsamples,       //nb_samples
           sizeof(out_buffer)/2,//lenout
           0);//is_last
       assert( samples_output > 0 &amp;&amp; "Error calling av_resample()!" );
       av_resample_close( audio_cntx );
       //*resample = malloc(sizeof(out_buffer));
       return &amp;out_buffer[0];  
    }

    When i run this code i get 3393 Segmentation fault (core dump created). Why ?

    For example, the use of pointers is correct ?
    and 22100 are the samples that are contained in 0.5 seconds of the song ?