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  • Encoding and processing into web-friendly formats

    13 avril 2011, par

    MediaSPIP 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 (...)

  • Use, discuss, criticize

    13 avril 2011, par

    Talk to people directly involved in MediaSPIP’s development, or to people around you who could use MediaSPIP to share, enhance or develop their creative projects.
    The bigger the community, the more MediaSPIP’s potential will be explored and the faster the software will evolve.
    A discussion list is available for all exchanges between users.

  • Supporting all media types

    13 avril 2011, par

    Unlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)

Sur d’autres sites (4709)

  • FFMPEG ALSA xrun crash

    13 décembre 2017, par Liam Martens

    I’m running a YouTube RTMP stream using FFMPEG with x11grab and an alsa loopback device but sometimes after let’s say 20 hours there is an ALSA xrun and then the ffmpeg command crashes, but I’m not sure why or how this happens. (mind you the ffmpeg command does not run continuously it gets restarted automatically every so often, but the xrun makes the command crash causing the stream to go offline sometimes because a crash restart is not fast enough)

    I’m using thread_queue_size and I’ve even manually compiled ffmpeg with a higher ALSA BUFFER SIZE, but the issue appears to persist still. Besides this I’ve also scoured many posts with people having similar issues but these never really seem to end up resolved.

    This is the stream command

    ffmpeg -loglevel verbose -f alsa -thread_queue_size 12288 -ac 2 -i hw:Loopback,1,0 \
    
         -probesize 10M -f x11grab -field_order tt -thread_queue_size 12288 -video_size 1280x720 -r 30 -i :1.1 \
    
        -c:v libx264 -c:a libmp3lame -shortest -tune fastdecode -tune zerolatency \
    
        -crf 26 -pix_fmt yuv420p -threads 0 -maxrate 2500k -bufsize 2500k -pass 1 -af aresample=async=1 \
    
        -movflags +faststart -flags +global_header -preset ultrafast -r 30 -g 60 -b:v 2000k -b:a 192k -ar 44100 \
    
        -f flv -rtmp_live live rtmp://a.rtmp.youtube.com/live2/{KEY}

    Log excerpt

    ffmpeg version N-89463-gc7a5e80 Copyright (c) 2000-2017 the FFmpeg developers
    
  built with gcc 6.3.0 (Debian 6.3.0-18) 20170516
    
  configuration: --prefix=/usr --enable-avresample --enable-avfilter --enable-gpl --enable-libmp3lame --enable-librtmp --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libtheora --enable-postproc --enable-pic --enable-pthreads --enable-shared --disable-stripping --disable-static --enable-vaapi --enable-libopus --enable-libfreetype --enable-libfontconfig --enable-libpulse --disable-debug
    
  libavutil      56.  5.100 / 56.  5.100
    
  libavcodec     58.  6.103 / 58.  6.103
    
  libavformat    58.  3.100 / 58.  3.100
    
  libavdevice    58.  0.100 / 58.  0.100
    
  libavfilter     7.  7.100 /  7.  7.100
    
  libavresample   4.  0.  0 /  4.  0.  0
    
  libswscale      5.  0.101 /  5.  0.101
    
  libswresample   3.  0.101 /  3.  0.101
    
  libpostproc    55.  0.100 / 55.  0.100
    
Guessed Channel Layout for Input Stream #0.0 : stereo
    
Input #0, alsa, from 'hw:Loopback,1,0':
    
  Duration: N/A, start: 1513163617.594224, bitrate: 1536 kb/s
    
    Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
    
Input #1, x11grab, from ':1.1':
    
  Duration: N/A, start: 1513163617.632434, bitrate: N/A
    
    Stream #1:0: Video: rawvideo, 1 reference frame (BGR[0] / 0x524742), bgr0(top first), 854x480, 30 fps, 30 tbr, 1000k tbn, 1000k tbc
    
Parsing...
    
Parsed protocol: 0
    
Parsed host    : a.rtmp.youtube.com
    
Parsed app     : live2
    
RTMP_Connect1, ... connected, handshaking
    
HandShake: Type Answer   : 03
    
HandShake: Server Uptime : 0
    
HandShake: FMS Version   : 4.0.0.1
    
HandShake: Handshaking finished....
    
RTMP_Connect1, handshaked
    
Invoking connect
    
HandleServerBW: server BW = 2500000
    
HandleClientBW: client BW = 10000000 2
    
HandleChangeChunkSize, received: chunk size change to 256
    
RTMP_ClientPacket, received: invoke 240 bytes
    
(object begin)
    
Property: 
    
Property: 
    
Property: 
    
(object begin)
    
Property: 3,5,3,824>
    
Property: 
    
Property: 
    
(object end)
    
Property: 
    
(object begin)
    
Property: 
    
Property: 
    
Property: 
    
Property: 
    
Property: 
    
(object begin)
    
Property: 
    
(object end)
    
(object end)
    
(object end)
    
HandleInvoke, server invoking <_result>
    
HandleInvoke, received result for method call <connect>
    
Invoking releaseStream
    
Invoking FCPublish
    
Invoking createStream
    
RTMP_ClientPacket, received: invoke 21 bytes
    
(object begin)
    
Property: 
    
Property: 
    
Property: NULL
    
(object end)
    
HandleInvoke, server invoking <onbwdone>
    
Invoking _checkbw
    
RTMP_ClientPacket, received: invoke 29 bytes
    
(object begin)
    
Property: 
    
Property: 
    
Property: NULL
    
Property: 
    
(object end)
    
HandleInvoke, server invoking &lt;_result>
    
HandleInvoke, received result for method call <createstream>
    
Invoking publish
    
RTMP_ClientPacket, received: invoke 73 bytes
    
(object begin)
    
Property: 
    
Property: 
    
Property: NULL
    
Property: 
    
(object begin)
    
Property: 
    
Property: 
    
(object end)
    
(object end)
    
HandleInvoke, server invoking <onstatus>
    
HandleInvoke, onStatus: NetStream.Publish.Start
    
Stream mapping:
    
  Stream #1:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
    
  Stream #0:0 -> #0:1 (pcm_s16le (native) -> mp3 (libmp3lame))
    
Press [q] to stop, [?] for help
    
[graph 0 input from stream 1:0 @ 0x5607d087e060] w:854 h:480 pixfmt:bgr0 tb:1/30 fr:30/1 sar:0/1 sws_param:flags=2
    
[auto_scaler_0 @ 0x5607d087d800] w:iw h:ih flags:'bicubic' interl:0
    
[format @ 0x5607d087ed40] auto-inserting filter 'auto_scaler_0' between the filter 'Parsed_null_0' and the filter 'format'
    
[auto_scaler_0 @ 0x5607d087d800] w:854 h:480 fmt:bgr0 sar:0/1 -> w:854 h:480 fmt:yuv420p sar:0/1 flags:0x4
    
[swscaler @ 0x5607d0880260] Warning: data is not aligned! This can lead to a speed loss
    
[libx264 @ 0x5607d08684e0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
    
[libx264 @ 0x5607d08684e0] profile Constrained Baseline, level 3.1
    
[libx264 @ 0x5607d08684e0] 264 - core 148 r2748 97eaef2 - H.264/MPEG-4 AVC codec - Copyleft 2003-2016 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=2 lookahead_threads=2 sliced_threads=1 slices=2 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=60 keyint_min=6 scenecut=0 intra_refresh=0 rc_lookahead=0 rc=crf mbtree=0 crf=26.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=1500 vbv_bufsize=1500 crf_max=0.0 nal_hrd=none filler=0 ip_ratio=1.40 aq=0
    
[graph_1_in_0_0 @ 0x5607d091c840] tb:1/48000 samplefmt:s16 samplerate:48000 chlayout:0x3
    
[Parsed_aresample_0 @ 0x5607d0916b40] ch:2 chl:stereo fmt:s16 r:48000Hz -> ch:2 chl:stereo fmt:s16p r:44100Hz
    
Output #0, flv, to 'rtmp://a.rtmp.youtube.com/live2/{KEY}':
    
  Metadata:
    
    encoder         : Lavf58.3.100
    
    Stream #0:0: Video: h264 (libx264), 1 reference frame ([7][0][0][0] / 0x0007), yuv420p(top coded first (swapped)), 854x480, q=-1--1, 1000 kb/s, 30 fps, 1k tbn, 30 tbc
    
    Metadata:
    
      encoder         : Lavc58.6.103 libx264
    
    Side data:
    
      cpb: bitrate max/min/avg: 1500000/0/1000000 buffer size: 1500000 vbv_delay: -1
    
    Stream #0:1: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 44100 Hz, stereo, s16p, delay 1105, 192 kb/s
    
    Metadata:
    
      encoder         : Lavc58.6.103 libmp3lame
    
frame=   29 fps=0.0 q=17.0 size=     146kB time=00:00:00.94 bitrate=1267.3kbits/s speed=1.86x    
    
frame=   44 fps= 44 q=18.0 size=     168kB time=00:00:01.46 bitrate= 942.4kbits/s speed=1.45x    
    
frame=   60 fps= 40 q=16.0 size=     191kB time=00:00:01.96 bitrate= 794.8kbits/s speed= 1.3x    
    
...
    
frame= 2740 fps= 30 q=17.0 size=    7993kB time=00:01:31.32 bitrate= 717.0kbits/s speed=   1x    
    
frame= 2755 fps= 30 q=18.0 size=    8013kB time=00:01:31.82 bitrate= 714.9kbits/s speed=   1x    
    
[alsa @ 0x5607d084d7e0] ALSA buffer xrun.
    
</onstatus></createstream></onbwdone></connect>

  • ffmpeg : unsync audio after processing

    18 novembre 2013, par QuickSilver

    I am recording a video and using RecordRTC : WebRTC . After receiving the webm video and wav audio at server, I'm encoding it to a mp4 file using ffmpeg(executing shell command via php). But after encoding process, the audio is unsync with video (audio ends before video). How can I fix this ?

    js code is here

    ffmpeg command used is :

    ffmpeg -y -i 166890589.wav -i 166890589.webm -vcodec libx264 166890589.mp4

    Console output :

    ffmpeg version 0.8.9-6:0.8.9-0ubuntu0.13.04.1, Copyright (c) 2000-2013 the Libav developers
    
  built on Nov  9 2013 19:15:52 with gcc 4.7.3
    
*** THIS PROGRAM IS DEPRECATED ***
    
This program is only provided for compatibility and will be removed in a future release. Please use avconv instead.
    
[wav @ 0x81caa60] max_analyze_duration reached
    
Input #0, wav, from &#39;166890589.wav&#39;:
    
  Duration: 00:00:07.05, bitrate: 1411 kb/s
    
    Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
    
[matroska,webm @ 0x823c340] Unknown entry 0x63C5
    
[matroska,webm @ 0x823c340] Estimating duration from bitrate, this may be inaccurate
    

    
Seems stream 0 codec frame rate differs from container frame rate: 1000.00 (1000/1) -> 10.00 (10/1)
    
Input #1, matroska,webm, from &#39;166890589.webm&#39;:
    
  Duration: 00:00:08.40, start: 0.000000, bitrate: N/A
    
    Stream #1.0: Video: vp8, yuv420p, 320x240, PAR 1:1 DAR 4:3, 10 tbr, 1k tbn, 1k tbc (default)
    
[buffer @ 0x8245620] w:320 h:240 pixfmt:yuv420p
    
[libx264 @ 0x82618a0] using SAR=1/1
    
[libx264 @ 0x82618a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX
    
[libx264 @ 0x82618a0] profile Main, level 1.1
    
[libx264 @ 0x82618a0] 264 - core 123 r2189 35cf912 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=0 b_adapt=1 b_bias=0 direct=1 weightb=0 open_gop=1 weightp=2 keyint=250 keyint_min=10 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.25 aq=1:1.00
    
Output #0, mp4, to &#39;166890589.mp4&#39;:
    
  Metadata:
    
    encoder         : Lavf53.21.1
    
    Stream #0.0: Video: libx264, yuv420p, 320x240 [PAR 1:1 DAR 4:3], q=-1--1, 10 tbn, 10 tbc (default)
    
    Stream #0.1: Audio: libvo_aacenc, 44100 Hz, 2 channels, s16, 200 kb/s
    
Stream mapping:
    
  Stream #1.0 -> #0.0
    
  Stream #0.0 -> #0.1
    
Press ctrl-c to stop encoding
    
frame=   84 fps=  0 q=25.0 Lsize=     260kB time=7.06 bitrate= 301.3kbits/s    
    
video:83kB audio:172kB global headers:0kB muxing overhead 1.783102%
    
frame I:1     Avg QP:17.52  size:  6554
    
[libx264 @ 0x82618a0] frame P:41    Avg QP:19.07  size:  1555
    
[libx264 @ 0x82618a0] frame B:42    Avg QP:20.01  size:   325
    
[libx264 @ 0x82618a0] consecutive B-frames: 33.3%  0.0%  0.0% 66.7%
    
[libx264 @ 0x82618a0] mb I  I16..4: 36.7%  0.0% 63.3%
    
[libx264 @ 0x82618a0] mb P  I16..4:  3.8%  0.0%  5.0%  P16..4: 34.3%  9.8%  7.1%  0.0%  0.0%    skip:40.0%
    
[libx264 @ 0x82618a0] mb B  I16..4:  1.4%  0.0%  0.1%  B16..8: 37.5%  5.5%  0.4%  direct: 2.5%  skip:52.5%  L0:41.4% L1:51.5% BI: 7.2%
    
[libx264 @ 0x82618a0] coded y,uvDC,uvAC intra: 40.7% 76.2% 26.8% inter: 10.9% 22.5% 2.7%
    
[libx264 @ 0x82618a0] i16 v,h,dc,p: 28% 34% 23% 15%
    
[libx264 @ 0x82618a0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 29% 28% 21%  3%  4%  4%  5%  3%  3%
    
[libx264 @ 0x82618a0] i8c dc,h,v,p: 49% 20% 26%  4%
    
[libx264 @ 0x82618a0] Weighted P-Frames: Y:0.0% UV:0.0%
    
[libx264 @ 0x82618a0] ref P L0: 62.7%  4.6% 16.3% 16.4%
    
[libx264 @ 0x82618a0] ref B L0: 79.0% 21.0%
    
[libx264 @ 0x82618a0] kb/s:80.00

  • Multiple Video Streams in one Feed ffmpeg

    1er février 2014, par trueblue

    We are trying to send live stream from two webcams as below :

    ffmpeg -f video4linux2 -i /dev/video0 -f video4linux2 -i /dev/video1 http://127.0.0.1:8090/feed1.ffm

    We want to play both the streams using any players available. When we use VLC, to open the stream, we get only one stream( from /dev/video0). The command is as below :

    vlc http://127.0.0.1:8090/test.mpg

    Here I am running ffserver in my machine and trying to access as localhost. My ffserver config is as below :

    Port 8090
    
BindAddress 0.0.0.0
    
MaxHTTPConnections 2000
    
MaxClients 1000
    
MaxBandwidth 1000
    
CustomLog -
    
NoDaemon
    

    
<feed>
    
File /tmp/feed1.ffm
    
FileMaxSize 1G
    
ACL allow 127.0.0.1
    
</feed>
    

    
<stream>
    

    
# coming from live feed &#39;feed1&#39;
    
Feed feed1.ffm
    
Format mpeg
    
VideoBufferSize 40000
    
VideoSize 1280x720
    
VideoCodec mpeg1video
    
NoAudio
    
ACL ALLOW 127.0.0.1
    
</stream>
    

    
<stream>
    
Format status
    

    
# Only allow local people to get the status
    
ACL allow localhost
    
ACL allow 192.168.0.0 192.168.255.255
    

    
#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
    
</stream>
    

    

    
# Redirect index.html to the appropriate site
    

    
<redirect>
    
URL http://www.ffmpeg.org/
    
</redirect>

    If we try the below command to save two streams onto the file, we are getting two instances of VLC player and both Streams can be seen :

    ffmpeg -f video4linux2 -i /dev/video0 -f video4linux2 -i /dev/video1 /home/2Streams.mpg

    Its a strange behavior I am able to save two Video Streams as a file but I am unable to send Two Video Streams in one Single feed. Kindly help me out in achieving the same.

    Regards

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