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Autres articles (105)
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Encoding and processing into web-friendly formats
13 avril 2011, parMediaSPIP automatically converts uploaded files to internet-compatible formats.
Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
All uploaded files are stored online in their original format, so you can (...) -
Contribute to a better visual interface
13 avril 2011MediaSPIP 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. -
Création définitive du canal
12 mars 2010, parLorsque votre demande est validée, vous pouvez alors procéder à la création proprement dite du canal. Chaque canal est un site à part entière placé sous votre responsabilité. Les administrateurs de la plateforme n’y ont aucun accès.
A la validation, vous recevez un email vous invitant donc à créer votre canal.
Pour ce faire il vous suffit de vous rendre à son adresse, dans notre exemple "http://votre_sous_domaine.mediaspip.net".
A ce moment là un mot de passe vous est demandé, il vous suffit d’y (...)
Sur d’autres sites (4510)
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AWS Lambda tmp disk limit with ffmpeg
21 avril 2022, par AlmajuI am using AWS Lambda to render videos using ffmpeg. It is a basic function that downloads external clips, merges those, then uploads the result to S3.


This function needs to be able to scale quickly because I need to generate a lot of videos in a short time period.


The problem is that I reach the
/tmpdisk limit of 512MB because the same lambda environment is reused by the different requests. I have increased that memory limit to temporarily solve the problem but it does not feel scalable.

I have looked into EFS but the requirement to use a VPC and a NAT to have internet access sounds really heavy for my simple needs.


Is there a way to output the result of ffmpeg directly in S3 (thus not needing the temporary folder) ? Or perhaps force to respawn a clean lambda environment if the memory is about to be full ?


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FFMPEG conversion failed at random times on live stream to youtube
3 avril 2022, par good karmaI am running a bash script using ubuntu server on arm64 that live streams video and audio to youtube, heres the script below


FFMPEG is the lateset version available in the ubuntu repo


#! /bin/bash

VBR="8000k"
FPS="24"
QUAL="superfast"

YOUTUBE_URL="rtmp://a.rtmp.youtube.com/live2"
KEY="****"

VIDEO_SOURCE="/mnt/disk1/test/****"

AUDIO_SOURCE="****"

ffmpeg \
 -re -f lavfi -i "movie=filename=$VIDEO_SOURCE:loop=0, setpts=N/(FRAME_RATE*TB)" \
 -thread_queue_size 512 -i "$AUDIO_SOURCE" \
 -map 0:v:0 -map 1:a:0 \
 -map_metadata:g 1:g \
 -vcodec libx264 -pix_fmt yuv420p -preset $QUAL -r $FPS -g $(($FPS * 2)) -b:v $VBR \
 -acodec libmp3lame -ar 44100 -threads 4 -qscale:v 3 -b:a 320000 -bufsize 512k \
 -report -loglevel fatal -nostats \
 -f flv "$YOUTUBE_URL/$KEY"


The stream works fine, however after a seemingly random period of time, I get a conversion failed error


first after 64 hours of streaming
second after 12 hours of streaming
third after 8 hours of streaming


The server itsef is fine cpu and memory usage are extremly low and there are no spikes


from what i can guess the error has something to do with libx264 judging by the error logs, although I do not really know. After some guidance on what to do try next. Here is the tail end of the ffmpeg log report


[libx264 @ 0xaaab23ad3e30] frame=1351975 QP=16.89 NAL=2 Slice:P Poc:14 I:6461 P:1224 SKIP:475 size=26602 bytes
av_interleaved_write_frame(): Broken pipe
No more output streams to write to, finishing.
[libx264 @ 0xaaab23ad3e30] frame=1351976 QP=17.90 NAL=2 Slice:B Poc:10 I:2395 P:3322 SKIP:2443 size=20127 bytes
[libx264 @ 0xaaab23ad3e30] frame=1351977 QP=17.89 NAL=0 Slice:B Poc:12 I:1848 P:3235 SKIP:3077 size=17313 bytes
[libx264 @ 0xaaab23ad3e30] frame=1351978 QP=15.65 NAL=2 Slice:P Poc:22 I:6458 P:1171 SKIP:531 size=26073 bytes
[libx264 @ 0xaaab23ad3e30] frame=1351979 QP=17.19 NAL=2 Slice:B Poc:18 I:2766 P:3514 SKIP:1880 size=19633 bytes
[libx264 @ 0xaaab23ad3e30] frame=1351980 QP=18.35 NAL=0 Slice:B Poc:16 I:1233 P:2876 SKIP:4051 size=11769 bytes
[libx264 @ 0xaaab23ad3e30] frame=1351981 QP=17.28 NAL=0 Slice:B Poc:20 I:822 P:3197 SKIP:4141 size=10699 bytes
[libx264 @ 0xaaab23ad3e30] frame=1351982 QP=17.08 NAL=2 Slice:P Poc:30 I:6351 P:1443 SKIP:366 size=29120 bytes
[libx264 @ 0xaaab23ad3e30] frame=1351983 QP=17.29 NAL=2 Slice:B Poc:26 I:2060 P:4211 SKIP:1889 size=18465 bytes
[libx264 @ 0xaaab23ad3e30] frame=1351984 QP=18.00 NAL=0 Slice:B Poc:24 I:472 P:3225 SKIP:4463 size=8893 bytes
[libx264 @ 0xaaab23ad3e30] frame=1351985 QP=17.98 NAL=0 Slice:B Poc:28 I:1339 P:3681 SKIP:3140 size=13378 bytes
[libx264 @ 0xaaab23ad3e30] frame=1351986 QP=15.57 NAL=2 Slice:P Poc:32 I:6270 P:1584 SKIP:306 size=33703 bytes
av_interleaved_write_frame(): Broken pipe
[libmp3lame @ 0xaaab23ac06f0] Trying to remove 47 more samples than there are in the queue
[flv @ 0xaaab23ad2ff0] Failed to update header with correct duration.
[flv @ 0xaaab23ad2ff0] Failed to update header with correct filesize.
Error writing trailer of rtmp://a.rtmp.youtube.com/live2/****: Broken pipe
frame=1351987 fps= 24 q=22.0 Lsize=37296521kB time=15:38:52.22 bitrate=5423.8kbits/s dup=0 drop=56332 speed= 1x 
video:35035969kB audio:2200479kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.161329%
Input file #0 (movie=filename=/mnt/disk1/test/****.mp4:loop=0, setpts=N/(FRAME_RATE*TB)):
 Input stream #0:0 (video): 1408319 packets read (4380435417600 bytes); 1408319 frames decoded; 
 Total: 1408319 packets (4380435417600 bytes) demuxed
Input file #1 (https://****.mp3):
 Input stream #1:0 (audio): 2156470 packets read (901316441 bytes); 2156470 frames decoded (2484253440 samples); 
 Total: 2156470 packets (901316441 bytes) demuxed
Output file #0 (rtmp://a.rtmp.youtube.com/live2/****):
 Output stream #0:0 (video): 1351987 frames encoded; 1351976 packets muxed (35876832486 bytes); 
 Output stream #0:1 (audio): 2156470 frames encoded (2484253440 samples); 2156469 packets muxed (2253290057 bytes); 
 Total: 3508445 packets (38130122543 bytes) muxed
3564789 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0xaaab23ac4ef0] Statistics: 0 seeks, 3823215 writeouts
[rtmp @ 0xaaab23dab6a0] UnPublishing stream...
[rtmp @ 0xaaab23dab6a0] Deleting stream...
[libx264 @ 0xaaab23ad3e30] frame I:53166 Avg QP:16.31 size: 39514
[libx264 @ 0xaaab23ad3e30] frame P:500115 Avg QP:17.79 size: 32831
[libx264 @ 0xaaab23ad3e30] frame B:798706 Avg QP:19.67 size: 21732
[libx264 @ 0xaaab23ad3e30] consecutive B-frames: 13.9% 15.2% 20.2% 50.7%
[libx264 @ 0xaaab23ad3e30] mb I I16..4: 45.9% 44.7% 9.4%
[libx264 @ 0xaaab23ad3e30] mb P I16..4: 40.6% 37.1% 2.2% P16..4: 16.5% 0.0% 0.0% 0.0% 0.0% skip: 3.7%
[libx264 @ 0xaaab23ad3e30] mb B I16..4: 16.3% 16.2% 0.2% B16..8: 25.0% 0.0% 0.0% direct:19.7% skip:22.5% L0:48.7% L1:48.4% BI: 2.9%
[libx264 @ 0xaaab23ad3e30] 8x8 transform intra:47.5% inter:31.9%
[libx264 @ 0xaaab23ad3e30] coded y,uvDC,uvAC intra: 15.8% 78.7% 25.0% inter: 2.0% 60.3% 1.8%
[libx264 @ 0xaaab23ad3e30] i16 v,h,dc,p: 50% 20% 9% 21%
[libx264 @ 0xaaab23ad3e30] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 19% 16% 31% 4% 7% 5% 5% 6% 6%
[libx264 @ 0xaaab23ad3e30] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 36% 20% 33% 2% 2% 2% 1% 2% 2%
[libx264 @ 0xaaab23ad3e30] i8c dc,h,v,p: 29% 18% 18% 35%
[libx264 @ 0xaaab23ad3e30] Weighted P-Frames: Y:3.0% UV:2.2%
[libx264 @ 0xaaab23ad3e30] kb/s:5095.01
[AVIOContext @ 0xaaab22460eb0] Statistics: 42281852862 bytes read, 2731 seeks
[AVIOContext @ 0xaaab23ac4de0] Statistics: 901390389 bytes read, 0 seeks
Conversion failed!


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FFPlay : How to manually select video quality during playing the mpd stream ?
19 août 2023, par Brian_wuI used ffplay to play mpd stream and successed,


Here the MPD file :


<?xml version="1.0" encoding="utf-8"?>
<mpd xmlns="urn:mpeg:dash:schema:mpd:2011" profiles="urn:mpeg:dash:profile:isoff-live:2011" type="static" mediapresentationduration="PT1M8.7S" maxsegmentduration="PT5.0S" minbuffertime="PT12.5S">
 <programinformation>
 </programinformation>
 <servicedescription>
 </servicedescription>
 <period start="PT0.0S">
 <adaptationset contenttype="video" startwithsap="1" segmentalignment="true" bitstreamswitching="true" framerate="24000/1001" maxwidth="1280" maxheight="720" par="16:9">
 <representation mimetype="video/mp4" codecs="avc1.4d401f" bandwidth="10237" width="480" height="270" sar="1:1">
 <segmenttemplate timescale="24000" initialization="init-stream$RepresentationID$.m4s" media="chunk-stream$RepresentationID$-$Number%05d$.m4s" startnumber="1">
 <segmenttimeline>
 <s t="0" d="150150" r="9"></s>
 <s d="149149"></s>
 </segmenttimeline>
 </segmenttemplate>
 </representation>
 <representation mimetype="video/mp4" codecs="avc1.4d401f" bandwidth="60882" width="1280" height="720" sar="1:1">
 <segmenttemplate timescale="24000" initialization="init-stream$RepresentationID$.m4s" media="chunk-stream$RepresentationID$-$Number%05d$.m4s" startnumber="1">
 <segmenttimeline>
 <s t="0" d="150150" r="9"></s>
 <s d="149149"></s>
 </segmenttimeline>
 </segmenttemplate>
 </representation>
 </adaptationset>
 <adaptationset contenttype="audio" startwithsap="1" segmentalignment="true" bitstreamswitching="true" lang="und">
 <representation mimetype="audio/mp4" codecs="mp4a.40.2" bandwidth="128000" audiosamplingrate="32000">
 <audiochannelconfiguration schemeiduri="urn:mpeg:dash:23003:3:audio_channel_configuration:2011" value="2"></audiochannelconfiguration>
 <segmenttemplate timescale="32000" initialization="init-stream$RepresentationID$.m4s" media="chunk-stream$RepresentationID$-$Number%05d$.m4s" startnumber="1">
 <segmenttimeline>
 <s t="0" d="159744"></s>
 <s d="160768" r="11"></s>
 <s d="111915"></s>
 </segmenttimeline>
 </segmenttemplate>
 </representation>
 <representation mimetype="audio/mp4" codecs="mp4a.40.2" bandwidth="128000" audiosamplingrate="44100">
 <audiochannelconfiguration schemeiduri="urn:mpeg:dash:23003:3:audio_channel_configuration:2011" value="2"></audiochannelconfiguration>
 <segmenttemplate timescale="44100" initialization="init-stream$RepresentationID$.m4s" media="chunk-stream$RepresentationID$-$Number%05d$.m4s" startnumber="1">
 <segmenttimeline>
 <s t="0" d="220160"></s>
 <s d="221184" r="11"></s>
 <s d="158713"></s>
 </segmenttimeline>
 </segmenttemplate>
 </representation>
 </adaptationset>
</period>
</mpd>


this dash stream contains 2 vindeos(480P and 720P), and 2 audios.The videos always starts low quality(480P), I want to change the video resolution to the high quality(720P) during playing process, what should i do ?


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