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  • 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 (...)

  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

  • MediaSPIP v0.2

    21 juin 2013, par

    MediaSPIP 0.2 is the first MediaSPIP stable release.
    Its official release date is June 21, 2013 and is announced here.
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

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  • Streaming from Icecast to Facebook Live with ffmpeg on Ubuntu 16.04

    24 juillet 2017, par Matthieu

    I have a webradio streamed by Liquidsoap+Icecast on a DigitalOcean droplet (Ubuntu 16.04), and I want to combine this audio stream with a simple jpeg image with ffmpeg, transform it to a video stream and send it to Facebook live.

    Facebook Live specifications :

    Video Format :

    We accept video in maximum 720p (1280 x 720) resolution, at 30 frames
    per second. (or 1 key frame every 2 seconds). You must send an I-frame
    (keyframe) at least once every two seconds throughout the stream..
    Recommended max bit rate is 4000 Kbps. Titles must be less than 255
    characters otherwise the stream will fail. The Live API accepts H264
    encoded video and AAC encoded audio only.

    Video Length :

    240 minute maximum length, with the exception of continuous live (see
    above). 240 minute maximum length for preview streams (either through
    Live dialog or publisher tools). After 240 minutes, a new stream key
    must be generated.

    Advanced Settings :

    Pixel Aspect Ratio : Square. Frame Types : Progressive Scan. Audio
    Sample Rate : 44.1 KHz. Audio Bitrate : 128 Kbps stereo. Bitrate
    Encoding : CBR.

    And the ffmpeg command I tried :

    ffmpeg -loop 1 -i radio-background.jpg -thread_queue_size 20480 -i http://localhost:8000/radio -framerate 30 -r 30 -acodec aac -strict -2 -c:v libx264 -strict experimental -b:a 128k -pix_fmt yuvj444p -x264-params keyint=60 -b:v 256k -minrate 128k -maxrate 512k -bufsize 768k -f flv 'rtmp://rtmp-api.facebook.com:80/rtmp/'

    This is actually working, as Facebook receives the live video and allows me to publish it. But I can’t figured out why there is a lag almost every 2 or 3 seconds. I asked different people to watch the test video, and everyone gets the same problem : every 2 or 3 seconds the playing "freezes" for half a second and seems to load the video, I even can see the loading icon spinning on the screen.

    I tried different combinations of values for the following options : -thread_queue_size / -b:v / -minrate / -maxrate / -bufsize. Nothing seems to produce any change.

    Video streaming is new for me, I’m not really confortable with the options listed before, so I think I’m missing something here...

    Also, note that the icecast audio stream perfectly works, and according to DigitalOcean graphs, the server is not overloaded. So I think my ffmpeg command is wrong.

    What ffmpeg parameters would be working for that case ?

  • Live streaming from FFMPEG : output a window m3u8 and also an all-segments m3u8

    1er décembre 2022, par coder_uk

    I've been experimenting with using FFMPEG to take an incoming RTMP stream, transcode into a selection of bitrates, and output it as HLS. It works.

    


    I wanted to store the live stream as a VOD. And found by adding the -hls_list_size 0 flag, sure enough, all segment are in the .m3u8. Making it super easy to turn into a VOD afterwards. So far, so good.

    


    But the obvious consequence of using -hls_list_size 0 is that now the m3u8 is huge during the live stream. That's fine for a VOD where it is only requested once, but less good during a live stream where it is requested over and over.

    


    So ... my question : without re-transcoding, can FFMPEG output both an all-segments all.m3u8 (to keep internally for making a VOD afterwards, ie using -hls_list_size 0) and also output a sliding-window style latest.m3u8 (of only the last X segments, ie using -hls_list_size 3) ?

    


    That way, viewers of the live stream could be served that little latest.m3u8, as a tiny file, with only the last few segments in. And after the event ends, I'd ditch that little latest.m3u8 and only keep the all.m3u8 to make a VOD version of the stream ?

    


    Thanks !

    


  • Error while opening encoder for output stream #0:1 Webm VP9 live streaming

    29 août 2019, par Salem

    The source video I use is H264 m3u8 live stream and this is the command I tried

    ffmpeg -re -i "http://sorce.com/live.m3u8" -c:v libvpx-vp9 -s 480x360 -keyint_min 60\
    -g 60 -speed 5 -tile-columns 4 -frame-parallel 1 -threads 8 -static-thresh 0 \
    -max-intra-rate 300 -deadline realtime -lag-in-frames 0 -error-resilient 1 \
    -b:v 300k -c:a libvorbis  -b:a 64k -ar 44100 -f webm_chunk -audio_chunk_duration 2000 \
    -header "/var/www/example.com/live/glass_360.hdr" -chunk_start_index 1  \
    /var/www/example.com/live/glass_360_%d.chk  

    I pickup this code from wkiki.webmproject.org
    most of the time I got this error message

    [libvorbis @ 0x5617bae0c240] encoder setup failed Error initializing
    output stream 0:1 -- Error while opening encoder for output stream
    #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height

    Here is FFMPEG command output

    Input #0, mpegts, from 'http://sorce.com/live.m3u8':
     Duration: N/A, start: 1369.000978, bitrate: N/A
     Program 1
       Metadata:
         service_name    : Service01
         service_provider: FFmpeg
       Stream #0:0[0x100]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1280x720 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc
       Stream #0:1[0x101]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, mono, fltp, 127 kb/s
    Stream mapping:
     Stream #0:0 -> #0:0 (h264 (native) -> vp9 (libvpx-vp9))
     Stream #0:1 -> #0:1 (aac (native) -> vorbis (libvorbis))

    I already add video map before video and audio coding argue but I got same error

    -map  0:v:0 -c:v libvpx-vp9 and -map 0:a:0 -c:a libvorbis

    I tried new broadcast command it works first after I disable the audio

    VP9_DASH_PARAMS="-tile-columns 4 -frame-parallel 1 -speed 6" && \
    ffmpeg -y -re -i http://sorce.com/live.m3u8 -c:v libvpx-vp9 -s 480x360 -b:v 150k \
    -keyint_min 150 -g 150 ${VP9_DASH_PARAMS} -an -f webm -dash 1 \
    video_1280x720_500k.webm && sleep 1 && ffmpeg  -f webm_dash_manifest \
    -i video_1280x720_500k.webm -c copy -f webm_dash_manifest - \ adaptation_sets "id=0"  manifest.mpd

    this command didn’t create manifest.mpd it created only video_1280x720_500k.webm