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Head down (wav version)
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Autres articles (97)
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List of compatible distributions
26 avril 2011, parThe 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 enhancements and plugins
13 avril 2011If 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. -
HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
For older browsers the Flowplayer flash fallback is used.
MediaSPIP allows for media playback on major mobile platforms with the above (...)
Sur d’autres sites (12199)
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What makes the bitrate of dash stream output low as compared to bitrate of input mp4 video
7 janvier 2021, par Chinmaey ShendeI used ffmpeg to convert bbb.mp4 to dash stream. I used the ffprobe to get the bitrates of input and output (shown below)
For the same resolution video 1920x1080 input has bit rate 2998 kb/s where as output has bitate of 20 kb/s.
My profile is set to "-b:v:0 200K -s:v:0 640x360 -b:v:1 600K -s:v:1 852x480 -b:v:2 4000K -s:v:2 1920x1080".
I am trying to understand what determines the output bit rate for the dash stream ?
Why does my output different from the input profile set in ffmpeg command



Input video stream - bbb.mp4



''' Duration: 00:10:34.53, start: 0.000000, bitrate: 3481 kb/s
 Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 2998 kb/s, 30 fps, 30 tbr, 30k tbn, 60 tbc (default)'''




Output :



''' Duration: 00:10:34.60, start: 630.000000, bitrate: 20 kb/s
 Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 20 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)

 major_brand : iso5
 minor_version : 512
 compatible_brands: iso5iso6mp41
 encoder : Lavf58.42.100
 Duration: 00:10:34.60, start: 630.000000, bitrate: 4 kb/s
 Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 852x480 [SAR 640:639 DAR 16:9], 4 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)

 major_brand : iso5
 minor_version : 512
 compatible_brands: iso5iso6mp41
 encoder : Lavf58.42.100
 Duration: 00:10:34.60, start: 630.000000, bitrate: 1 kb/s
 Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 640x360 [SAR 1:1 DAR 16:9], 1 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)
'''




I tried to look in ffprobe documentation I could not find the definition of the bit_rate field output in ffprobe.



Can you please help me to understand this ?


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Ffmpeg Windows split 1 video to chunks with random length (no re-encode)
24 janvier 2021, par Anna LoneI have 1 big video file and I need to split it to random length chunks without re-encode.


For example chunks from 130 to 240 seconds.


For windows ffmpeg


Tried this and. Nothing in output folder.


$ times=$(ruby -e 's=[]; d=0; while d < 150 do t=rand(15..50); s << (d+t); d=d+t end; puts s.join(",")')
$ echo $times
15,53,96,124,168
$ ffmpeg -i lutherceleb.mp4 -f segment -segment_times $times -c copy -reset_timestamps 1 -map 0 OUTPUT%d.mp4



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Converting AAC stream to DASH MP4 with high fragment length precision
5 mars 2017, par vdudouytFor my HTML5 project I need to create a fragmented MP4 file with a single audio stream (no video), each fragment of which has a duration of exactly 0.1 second.
Accordingly to ffmpeg docs, you can accomplish that by passing a value in microseconds with ’-frag_duration’ - which I found to be working and playable with HTML5 MediaSource API :
$ ffmpeg -y -i input.aac -c:a libfdk_aac -b:a 64k -level:v 13 -r 25 -strict experimental -movflags empty_moov+default_base_moof -frag_duration 100000 output.mp4
As we have a 210 second audio split up by 0.1s fragments, I expect that in output.mp4 we’d have 2100 fragments, hence 2100 moof atoms. But, upon inspecting it I’ve figured out that we only have 1811 moof atoms - which means that some (or maybe even all) fragments are bigger than expected :
$ python ~/git/mp4viewer/src/showboxes.py output.mp4 |grep moof|wc -l
1811Could anybody tell me what’s wrong, and how could I accomplish what I want ?
Right now my assumption is that during an encoding I have AAC frame length which is not a multiple of 0.1s, hence ffmpeg has no chance to produce the fragments that are strictly equal to 0.1s but I’m not sure. If somebody can confirm that - and let me know a way to explicitly set AAC frame_size in FFMPEG (I couldn’t find anything like that in the docs), or completely disprove this - it would be also highly appreciated.