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  • Websites made ​​with MediaSPIP

    2 mai 2011, par

    This page lists some websites based on MediaSPIP.

  • Contribute to a better visual interface

    13 avril 2011

    MediaSPIP is based on a system of themes and templates. Templates define the placement of information on the page, and can be adapted to a wide range of uses. Themes define the overall graphic appearance of the site.
    Anyone can submit a new graphic theme or template and make it available to the MediaSPIP community.

  • Submit enhancements and plugins

    13 avril 2011

    If you have developed a new extension to add one or more useful features to MediaSPIP, let us know and its integration into the core MedisSPIP functionality will be considered.
    You can use the development discussion list to request for help with creating a plugin. As MediaSPIP is based on SPIP - or you can use the SPIP discussion list SPIP-Zone.

Sur d’autres sites (9937)

  • libswresample : swr_convert() returns empty buffer

    11 septembre 2019, par Герман Лиманський

    I try to convert audio in format AV_SAMPLE_FMT_S32. I use swr_convert(), but out buffer still empty.

    // frame is decoded frame, rframe - is empty frame(out buffer)

        if (!main_context->resampler) {
    
                    main_context->resampler = 
    
                    swr_alloc_set_opts(main_context->resampler,
    
                    AV_CH_LAYOUT_STEREO,                    // output
    
                    AV_SAMPLE_FMT_S32,                      // output
    
                    44100,                                  // output
    
                    audio_codec_context->channel_layout,    // input
    
                    audio_codec_context->sample_fmt,        // input
    
                    audio_codec_context->sample_rate,       // input
    
                    0,
    
                    nullptr);
    
                swr_init(main_context->resampler);
    

    
            }
    

    
            //int in_samples = frame->nb_samples;
    
            int out_samples = av_rescale_rnd(swr_get_delay(
    
                main_context->resampler, 44100) + 44100,
    
                44100,
    
                44100,
    
                AV_ROUND_UP);
    

    
            size_t buffSize = av_samples_alloc(rframe->data, NULL,audio_codec_context->channels, out_samples, AV_SAMPLE_FMT_S32, 0);
    
            int len = swr_convert(main_context->resampler, rframe->data, frame->nb_samples, (const uint8_t * *)frame->data, frame->nb_samples);
    
//here.. rframe->data should have some data, but its empty
    
            while (len > 0)
    
            {
    
                size_t size_ = rframe->nb_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S32);
    
                main_context->audio_buf.write(rframe->data[0], size_, 1);
    

    
                len = swr_convert(main_context->resampler, rframe->data, frame->nb_samples, NULL, NULL);
    
            }

  • avformat/rtpdec : int overflow in start_time_realtime

    8 janvier, par Jonathan Baudanza
    avformat/rtpdec : int overflow in start_time_realtime

    This was previously adjusted by me in 6b3f9c2e92b.
    
Unfortunately, I traded one integer overflow bug for
    
another.
    

    Currently, NTP timestamps that exceed INT64_MAX
    
( Jan 20, 1968) will cause an overflow when passed
    
to av_rescale.
    

    This patch replaces av_rescale, which operates on
    
int64_t, with ff_parse_ntp_time, which operates on
    
uint64_t. This will give the correct values for
    
timestamps back around the NTP epoch and present day
    
timestamps.
    

    Fixes ticket #11388.
    

    Signed-off-by : Martin Storsjö <martin@martin.st>
    • [DH] libavformat/rtsp.c

  • Revision 6d44dad4aa : Merge "vp8 : common : postproc : fix signed overflow of statement of (X +c) >= X wh

    17 septembre 2014, par Johann

    Merge "vp8 : common : postproc : fix signed overflow of statement of (X +c) >= X
    when ’-Werror=strict-overflow’ is set."

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