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Autres articles (27)

  • Les autorisations surchargées par les plugins

    27 avril 2010, par

    Mediaspip core
    autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs

  • MediaSPIP Player : problèmes potentiels

    22 février 2011, par

    Le lecteur ne fonctionne pas sur Internet Explorer
    Sur Internet Explorer (8 et 7 au moins), le plugin utilise le lecteur Flash flowplayer pour lire vidéos et son. Si le lecteur ne semble pas fonctionner, cela peut venir de la configuration du mod_deflate d’Apache.
    Si dans la configuration de ce module Apache vous avez une ligne qui ressemble à la suivante, essayez de la supprimer ou de la commenter pour voir si le lecteur fonctionne correctement : /** * GeSHi (C) 2004 - 2007 Nigel McNie, (...)

  • Keeping control of your media in your hands

    13 avril 2011, par

    The vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
    While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
    MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
    MediaSPIP aims to be accessible to as many people as possible and development is based on expanding the (...)

Sur d’autres sites (6366)

  • ffmpeg fixed video noise reduction

    19 mars 2024, par leo

    I have a camera with a fixed noise (video sample). Is it possible to remove this noise completely using ffmpeg ?

    


    I tried ffmpeg "-vf" noise reduction but didn't find any way to remove the noise by supplying a noise mask.

    


  • rtsp : Fix infinite loop in listen mode with UDP transport

    30 septembre 2020, par Martin Storsjö
    rtsp : Fix infinite loop in listen mode with UDP transport
    

    In listen mode with UDP transport, once the sender has sent
    the TEARDOWN and closed the connection, poll will indicate that
    one can read from the connection (indicating that the socket has
    reached EOF and should be closed by the receiver as well). In this
    case, parse_rtsp_message won't try to parse the command (because
    it's no longer in state STREAMING), but previously just returned
    zero.

    Prior to f6161fccf8c5720ceac1ed1df8ba60ff8fed69f5, this caused
    udp_read_packet to return zero, which is treated as EOF by
    read_packet. But after that commit, udp_read_packet would continue
    if parse_rtsp_message didn't return an explicit error code.

    To keep the original behaviour from before that commit, more
    explicitly return an error in parse_rtsp_message when in the wrong
    state.

    Fixes : #8840
    Signed-off-by : Martin Storsjö <martin@martin.st>

    • [DH] libavformat/rtsp.c
  • Libav and xaudio2 - audio not playing

    25 août 2013, par siz

    I am trying to get audio playing with libav using xaudio2. The xaudio2 code I am using works with an older ffmpeg using avcodec_decode_audio2, but that has been deprecated for avcodec_decode_audio4. I have tried following various libav examples, but can't seem to get the audio to play. Video plays fine (or rather it just plays right fast now, as I haven't coded any sync code yet).

    Firstly audio gets init, no errors, video gets init, then packet :

    while (1) {
       //is this packet from the video or audio stream?
       if (packet.stream_index == player.v_id) {
           add_video_to_queue(&amp;packet);
       } else if (packet.stream_index == player.a_id) {
           add_sound_to_queue(&amp;packet);
       } else {
           av_free_packet(&amp;packet);
       }
    }

    Then in add_sound_to_queue :

    int add_sound_to_queue(AVPacket * packet) {
    AVFrame *decoded_frame = NULL;
    int done = AVCODEC_MAX_AUDIO_FRAME_SIZE;
    int got_frame = 0;

    if (!decoded_frame) {
       if (!(decoded_frame = avcodec_alloc_frame())) {
           printf("[ADD_SOUND_TO_QUEUE] Out of memory\n");
           return -1;
       }
    } else {
       avcodec_get_frame_defaults(decoded_frame);
    }

    if (avcodec_decode_audio4(player.av_acodecctx, decoded_frame, &amp;got_frame, packet) &lt; 0)  {
       printf("[ADD_SOUND_TO_QUEUE] Error in decoding audio\n");
       av_free_packet(packet);
       //continue;
       return -1;
    }
    if (got_frame) {
       int data_size;
       if (packet->size > done) {
           data_size = done;
       } else {
           data_size = packet->size;
       }
       BYTE * snd = (BYTE *)malloc( data_size * sizeof(BYTE));

       XMemCpy(snd,
           AudioBytes,
           data_size * sizeof(BYTE)
       );

       XMemSet(&amp;g_SoundBuffer,0,sizeof(XAUDIO2_BUFFER));

       g_SoundBuffer.AudioBytes = data_size;
       g_SoundBuffer.pAudioData = snd;
       g_SoundBuffer.pContext = (VOID*)snd;

       XAUDIO2_VOICE_STATE state;
       while( g_pSourceVoice->GetState( &amp;state ), state.BuffersQueued > 60 ) {
           WaitForSingleObject( XAudio2_Notifier.hBufferEndEvent, INFINITE );
       }

       g_pSourceVoice->SubmitSourceBuffer( &amp;g_SoundBuffer );
    }
    return 0;
    }

    I can't seem to figure out the problem, I have added error messages in init, opening video, codec handling etc. As mentioned before the xaudio2 code is working with an older ffmpeg, so maybe I have missed something with the avcodec_decode_audio4 ?

    If this snappet of code isn't enough, I can post the whole code, these are just the places in the code I think the problem would be :(