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  • Keeping control of your media in your hands

    13 April 2011, by

    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 (...)

  • Creating farms of unique websites

    13 April 2011, by

    MediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
    This allows (among other things): implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...)

  • Submit bugs and patches

    13 April 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 (...)

On other websites (5690)

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

    4 December 2015, by 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 June 2014, by blueimp
    Updated version number for last commit.

  • Updated version number for last merge.

    29 August 2014, by blueimp
    Updated version number for last merge.

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