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  • 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 (...)

  • List of compatible distributions

    26 avril 2011, par

    The table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
    If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...)

  • Submit bugs and patches

    13 avril 2011

    Unfortunately a software is never perfect.
    If you think you have found a bug, report it using our ticket system. Please to help us to fix it by providing the following information : the browser you are using, including the exact version as precise an explanation as possible of the problem if possible, the steps taken resulting in the problem a link to the site / page in question
    If you think you have solved the bug, fill in a ticket and attach to it a corrective patch.
    You may also (...)

Sur d’autres sites (5802)

  • encode audio ffmpeg c++ with different number of samples between input and output

    4 décembre 2015, par Victor Canezin de Oliveira

    I am trying to make an audio encoder to encode a live stream. I get my audio stream from a webrtc source. The properties for the source audio buffer is(AND I CANNOT CHANGE IT) :

    number of samples: 480 
    
sample size: 2 bytes 
    
sample rate: 44100Hz 
    
number of channels: 1

    I am using MP2 codec to encode the audio. It expects an audio number of samples of 1152(CAN’T CHANGE THAT EITHER), which is different from the source(480)

    I generate the audio frame using :

    frame->nb_samples = 480;
    
avcodec_fill_audio_frame(frame, nb_channels(=1), sample_fmt(=AVCodecContext sample_fmt), temp_audio_buffer(=source), 480, 0);

    And I am getting a "chopped" sound. From what I know, It is because of the difference between number of samples in each frame.

    Is there a way to fill the entire frame(1152 samples) somehow ? Will I be able to encode this live stream ?

    thanks

  • Updated version number for last commit.

    29 juin 2014, par blueimp
    Updated version number for last commit.

  • Updated version number for last merge.

    29 août 2014, par blueimp
    Updated version number for last merge.

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