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Autres articles (62)

  • 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.

  • La sauvegarde automatique de canaux SPIP

    1er avril 2010, par

    Dans le cadre de la mise en place d’une plateforme ouverte, il est important pour les hébergeurs de pouvoir disposer de sauvegardes assez régulières pour parer à tout problème éventuel.
    Pour réaliser cette tâche on se base sur deux plugins SPIP : Saveauto qui permet une sauvegarde régulière de la base de donnée sous la forme d’un dump mysql (utilisable dans phpmyadmin) mes_fichiers_2 qui permet de réaliser une archive au format zip des données importantes du site (les documents, les éléments (...)

  • Script d’installation automatique de MediaSPIP

    25 avril 2011, par

    Afin de palier aux difficultés d’installation dues principalement aux dépendances logicielles coté serveur, un script d’installation "tout en un" en bash a été créé afin de faciliter cette étape sur un serveur doté d’une distribution Linux compatible.
    Vous devez bénéficier d’un accès SSH à votre serveur et d’un compte "root" afin de l’utiliser, ce qui permettra d’installer les dépendances. Contactez votre hébergeur si vous ne disposez pas de cela.
    La documentation de l’utilisation du script d’installation (...)

Sur d’autres sites (7598)

  • nginx RTMP to HLS : FFMPG error when trying multiple bitrate output [closed]

    28 mai 2014, par user3685074

    I’m currently trying to convert my RTMP Livestream into a HLS with 3 quality-settings.

    I followed this guide

    I’ve compiled my own FFMPEG and it’s working if I just convert 1 file.
    It seems libx264 isn’t able to do multiple encodings at the same time ?

    I’m using these command :

            exec /usr/local/bin/ffmpeg -i rtmp://localhost/src/$name 
    
        -c:a libfdk_aac -b:a 32k   -c:v libx264 -b:v 128K -f flv rtmp://localhost/hls/$name_low 
    
        -c:a libfdk_aac -b:a 64k   -c:v libx264 -b:v 256K -f flv rtmp://localhost/hls/$name_mid 
    
        -c:a libfdk_aac -b:a 128k  -c:v libx264 -b:v 512K -f flv rtmp://localhost/hls/$name_hi  2>>/tmp/ffmpeg.log;

    this is the output :

        ffmpeg version N-63519-g61917a1 Copyright (c) 2000-2014 the FFmpeg developers
    
      built on May 28 2014 18:06:42 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1)
    
      configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-x11grab --enable-libvpx --enable-libmp3lame --enable-librtmp --enable-libspeex --enable-libfdk_aac
    
      libavutil      52. 87.100 / 52. 87.100
    
      libavcodec     55. 65.100 / 55. 65.100
    
      libavformat    55. 41.100 / 55. 41.100
    
      libavdevice    55. 13.101 / 55. 13.101
    
      libavfilter     4.  5.100 /  4.  5.100
    
      libswscale      2.  6.100 /  2.  6.100
    
      libswresample   0. 19.100 /  0. 19.100
    
      libpostproc    52.  3.100 / 52.  3.100
    
    Metadata:
    
      Server                NGINX RTMP (github.com/arut/nginx-rtmp-module)
    
      width                 1280.00
    
      height                720.00
    
      displayWidth          1280.00
    
      displayHeight         720.00
    
      duration              0.00
    
      framerate             25.00
    
      fps                   25.00
    
      videodatarate         390.00
    
      videocodecid          0.00
    
      audiodatarate         27.00
    
      audiocodecid          11.00
    
    Input #0, flv, from 'rtmp://localhost/src/test':
    
      Metadata:
    
        Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
    
        displayWidth    : 1280
    
        displayHeight   : 720
    
        fps             : 25
    
        profile         : 
    
        level           : 
    
      Duration: 00:00:00.00, start: 0.080000, bitrate: N/A
    
        Stream #0:0: Video: h264 (High), yuv420p, 1280x720, 399 kb/s, 25 fps, 25 tbr, 1k tbn, 50 tbc
    
        Stream #0:1: Audio: speex, 16000 Hz, mono, s16, 27 kb/s
    
    [libx264 @ 0x5260380] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
    
    [libx264 @ 0x5260380] profile High, level 3.1
    
    [libx264 @ 0x5260380] 264 - core 142 r2431 f23da7c - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=24 lookahead_threads=4 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=128 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
    
    [libx264 @ 0x525a920] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
    
    Output #0, flv, to 'rtmp://localhost/hls/test_low':
    
      Metadata:
    
        Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
    
        displayWidth    : 1280
    
        displayHeight   : 720
    
        fps             : 25
    
        profile         : 
    
        level           : 
    
        Stream #0:0: Video: h264 (libx264), yuv420p, 1280x720, q=-1--1, 128 kb/s, 25 fps, 90k tbn, 25 tbc
    
        Metadata:
    
          encoder         : Lavc55.65.100 libx264
    
        Stream #0:1: Audio: aac (libfdk_aac), 16000 Hz, mono, s16, 32 kb/s
    
        Metadata:
    
          encoder         : Lavc55.65.100 libfdk_aac
    
    Output #1, flv, to 'rtmp://localhost/hls/test_mid':
    
      Metadata:
    
        Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
    
        displayWidth    : 1280
    
        displayHeight   : 720
    
        fps             : 25
    
        profile         : 
    
        level           : 
    
        Stream #1:0: Video: h264, yuv420p, 1280x720, q=-1--1, 256 kb/s, 25 fps, 90k tbn, 25 tbc
    
        Metadata:
    
          encoder         : Lavc55.65.100 libx264
    
        Stream #1:1: Audio: aac, 16000 Hz, mono, s16
    
        Metadata:
    
          encoder         : Lavc55.65.100 libfdk_aac
    
    Output #2, flv, to 'rtmp://localhost/hls/test_hi':
    
      Metadata:
    
        Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
    
        displayWidth    : 1280
    
        displayHeight   : 720
    
        fps             : 25
    
        profile         : 
    
        level           : 
    
        Stream #2:0: Video: h264, yuv420p, 1280x720, q=-1--1, 25 fps, 90k tbn, 25 tbc
    
        Metadata:
    
          encoder         : Lavc55.65.100 libx264
    
        Stream #2:1: Audio: aac, 16000 Hz, mono, s16
    
        Metadata:
    
          encoder         : Lavc55.65.100 libfdk_aac
    
    Stream mapping:
    
      Stream #0:0 -> #0:0 (h264 -> libx264)
    
      Stream #0:1 -> #0:1 (libspeex -> libfdk_aac)
    
      Stream #0:0 -> #1:0 (h264 -> libx264)
    
      Stream #0:1 -> #1:1 (libspeex -> libfdk_aac)
    
      Stream #0:0 -> #2:0 (h264 -> libx264)
    
      Stream #0:1 -> #2:1 (libspeex -> libfdk_aac)
    
    Error while opening encoder for output stream #1:0 - maybe incorrect parameters such as bit_rate, rate, width or height

    I hope you can help me and sorry for my bad english.

    Greetz
    Kevin

  • FFplay requesting video via RTSP :// but receiving on multicast address

    28 mai 2014, par DavidG

    First of all, I apologize for how long the supporting information will be in this post. This is my first post on this forum.

    My issue is I need to run the command line version of ffmpeg to capture a video stream. However, as a proof of concept I’m first attempting to capture and view the video using ffplay (BTW, I have not had any success using ffmpeg or ffprobe). I’m running the ffplay command to read video from a Coretec video encoder which has multicast enabled.

    Unicast address:   172.30.18.50
    
Multicast address: 239.130.18.50:4002

    My question is how can I request the Unicast address, but receive the video on the multicast address ? (BTW, the ffplay operation does not work even if I replace the Unicast address with the Multicast address below)

    NOTE : After looking at the Wireshark trace, I see the video data has GSMTAP in the protocol column. When I do "ffmpeg -protocols : I see there is a Decoder "gsm" which decodes raw gsm. however, when I use ffplay -f gsm ... I get "Protocol not found".

    I am able to use VLC to view the video using the following command :

    VLC rtsp://172.30.18.50

    It appears from the Wireshark trace that the session is initiated on the Unicast address, but the video is streamed on the Multicast address. VLC is able to determine this and perform the appropriate operation. I don’t know what to add to ffplay to let it know that another stream will be carrying the video.

    I am UNABLE to perform the following ffplay commands (none of them work) :

    ffplay -v debug rtsp://172.30.18.50
    
ffplay -v debug -rtsp_transport udp rtsp://172.30.18.50
    
ffplay -v debug -rtsp_transport udp_multicast rtsp://172.30.18.50

    NOTE : I am able to get ffplay to launch, but the video is garbled badly. Maybe this bit of information will ring a bell for someone ? The command I used was :

    ffplay -v debug -i udp://239.130.18.50:4002?sources=172.30.18.50

    The version of ffplay I’m using is :

    ffplay version N-63439-g96470ca Copyright (c) 2003-2014 the FFmpeg developers
    
  built on May 25 2014 22:09:07 with gcc 4.8.2 (GCC)
    
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
    
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
    
le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
    
e --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-
    
libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libope
    
njpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsox
    
r --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab -
    
-enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx
    
--enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-
    
libxavs --enable-libxvid --enable-decklink --enable-zlib
    
  libavutil      52. 86.100 / 52. 86.100
    
  libavcodec     55. 65.100 / 55. 65.100
    
  libavformat    55. 41.100 / 55. 41.100
    
  libavdevice    55. 13.101 / 55. 13.101
    
  libavfilter     4.  5.100 /  4.  5.100
    
  libswscale      2.  6.100 /  2.  6.100
    
  libswresample   0. 19.100 /  0. 19.100
    
  libpostproc    52.  3.100 / 52.  3.100

    The debug output for ffplay -v debug rtsp ://172.30.18.50 is :

    [rtsp @ 0000000002a8be80] SDP:=    0KB vq=    0KB sq=    0B f=0/0
    
v=0
    
o=- 1 1 IN IP4 50.18.30.172 
    
s=Test
    
a=type:broadcast
    
t=0 0
    
c=IN IP4 239.130.18.50/63
    
m=video 4002 RTP/AVP 96
    
a=rtpmap:96 MP4V-ES/90000
    
a=fmtp:96 profile-level-id=245;config=000001B0F5000001B509000001000000012000C8F8
    
A058BA9860FA616087828307a=control:track1
    

    
[rtsp @ 0000000002a8be80] video codec set to: mpeg4
    
[udp @ 0000000002a8bac0] end receive buffer size reported is 65536
    
[udp @ 0000000002aa1600] end receive buffer size reported is 65536
    
[rtsp @ 0000000002a8be80] Nonmatching transport in server reply/0
    
rtsp://172.30.18.50: Invalid data found when processing input

    And the Wireshark trace output is :

    OPTIONS rtsp://172.30.18.50:554 RTSP/1.0
    
CSeq: 1
    
User-Agent: Lavf55.41.100
    

    
RTSP/1.0 200 OK
    
CSeq: 1
    
Public: DESCRIBE, SETUP, TEARDOWN, PLAY
    

    
DESCRIBE rtsp://172.30.18.50:554 RTSP/1.0
    
Accept: application/sdp
    
CSeq: 2
    
User-Agent: Lavf55.41.100
    

    
RTSP/1.0 200 OK
    
CSeq: 2 Content-Type: application/sdp
    
Content-Length: 270
    

    
v=0
    
o=- 1 1 IN IP4 50.18.30.172
    
s=Test
    
a=type:broadcast
    
t=0 0
    
c=IN IP4 239.130.18.50/63
    
m=video 4002 RTP/AVP 96
    
a=rtpmap:96 MP4V-ES/90000
    
a=fmtp:96 profile-level-id=245;config=000001B0F5000001B509000001000000012000C8F8A058BA9860FA616087828307a=control:track1
    

    
SETUP rtsp://172.30.18.50:554 RTSP/1.0
    
Transport: RTP/AVP/UDP;unicast;client_port=9574-9575
    
CSeq: 3
    
User-Agent: Lavf55.41.100
    

    
RTSP/1.0 200 OK
    
CSeq: 3
    
Session: test
    
Transport: RTP/AVP;multicast;destination=;port=4002-4003;ttl=63

    The debug output for ffplay -v debug -rtsp_transport udp rtsp ://172.30.18.50 is :

    [rtsp @ 0000000002c5c0a0] SDP:=    0KB vq=    0KB sq=    0B f=0/0
    
v=0
    
o=- 1 1 IN IP4 50.18.30.172
    
s=Test
    
a=type:broadcast
    
t=0 0
    
c=IN IP4 239.130.18.50/63
    
m=video 4002 RTP/AVP 96
    
a=rtpmap:96 MP4V-ES/90000
    
a=fmtp:96 profile-level-id=245;config=000001B0F5000001B509000001000000012000C8F8
    
A058BA9860FA616087828307a=control:track1
    

    

    
[rtsp @ 0000000002c5c0a0] video codec set to: mpeg4
    
[udp @ 0000000002c62420] end receive buffer size reported is 65536
    
[udp @ 0000000002c726a0] end receive buffer size reported is 65536
    
[rtsp @ 0000000002c5c0a0] Nonmatching transport in server reply/0
    
rtsp://172.30.18.50: Invalid data found when processing input

    And the Wireshark trace output is :

    OPTIONS rtsp://172.30.18.50:554 RTSP/1.0
    
CSeq: 1
    
User-Agent: Lavf55.41.100
    

    
RTSP/1.0 200 OK
    
CSeq: 1
    
Public: DESCRIBE, SETUP, TEARDOWN, PLAY
    

    
DESCRIBE rtsp://172.30.18.50:554 RTSP/1.0
    
Accept: application/sdp
    
CSeq: 2
    
User-Agent: Lavf55.41.100
    

    
RTSP/1.0 200 OK
    
CSeq: 2
    
Content-Type: application/sdp
    
Content-Length: 270
    

    
v=0
    
o=- 1 1 IN IP4 50.18.30.172
    
s=Test
    
a=type:broadcast
    
t=0 0
    
c=IN IP4 239.130.18.50/63
    
m=video 4002 RTP/AVP 96
    
a=rtpmap:96 MP4V-ES/90000 a=fmtp:96 profile-level-id=245;config=000001B0F5000001B509000001000000012000C8F8A058BA9860FA616087828307a=control:track1
    

    
SETUP rtsp://172.30.18.50:554 RTSP/1.0
    
Transport: RTP/AVP/UDP;unicast;client_port=22332-22333
    
CSeq: 3
    
User-Agent: Lavf55.41.100
    

    
RTSP/1.0 200 OK
    
CSeq: 3
    
Session: test
    
Transport: RTP/AVP;multicast;destination=239.130.18.50;port=4002-4003;ttl=63

    The debug output for ffplay -v debug -rtsp_transport udp_multicast is :

    [rtsp @ 00000000002fc100] SDP:=    0KB vq=    0KB sq=    0B f=0/0
    
v=0
    
o=- 1 1 IN IP4 50.18.30.172
    
s=Test
    
a=type:broadcast
    
t=0 0
    
c=IN IP4 239.130.18.50/63
    
m=video 4002 RTP/AVP 96
    
a=rtpmap:96 MP4V-ES/90000
    
a=fmtp:96 profile-level-id=245;config=000001B0F5000001B509000001000000012000C8F8
    
A058BA9860FA616087828307a=control:track1
    

    
[rtsp @ 00000000002fc100] video codec set to: mpeg4
    
    nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0

    And the Wireshark trace output is :

    OPTIONS rtsp://172.30.18.50:554
    
RTSP/1.0
    
CSeq: 1
    
User-Agent: Lavf55.41.100
    

    
RTSP/1.0 200 OK
    
CSeq: 1
    
Public: DESCRIBE, SETUP, TEARDOWN, PLAY
    

    
DESCRIBE rtsp://172.30.18.50:554 RTSP/1.0
    
Accept: application/sdp
    
CSeq: 2
    
User-Agent: Lavf55.41.100
    

    
RTSP/1.0 200 OK
    
CSeq: 2
    
Content-Type: application/sdp
    
Content-Length: 270
    

    
v=0
    
o=- 1 1 IN IP4 50.18.30.172
    
s=Test
    
a=type:broadcast
    
t=0 0
    
c=IN IP4 239.130.18.50/63
    
m=video 4002 RTP/AVP 96
    
a=rtpmap:96 MP4V-ES/90000
    
a=fmtp:96 profile-level-id=245;config=000001B0F5000001B509000001000000012000C8F8A058BA9860FA616087828307a=control:track1
    

    
SETUP rtsp://172.30.18.50:554 RTSP/1.0
    
Transport: RTP/AVP/UDP;multicast
    
CSeq: 3
    
User-Agent: Lavf55.41.100

    Thank you in advance to whomever is willing to tackle this.
    - DavidG

  • FFmpeg - wrong duration

    28 mai 2014, par miss_tais

    I have a command, it get’s first 30 seconds of audio file :

    ffmpeg -ss 0 -t 30 -i AUDIO_FILE -acodec copy -f FORMAT_NAME NEW_AUDIO_FILE

    With mp3 files is works right, but with wmv or wav files it is not correct.

    The result :

    Guessed Channel Layout for  Input Stream #0.0 : stereo
    
Input #0, asf, from '':
    
  Metadata:
    
    WM/Track        : 0
    
    WM/MediaPrimaryClassID: {D1607DBC-E323-4BE2-86A1-48A42A28441E}
    
    WMFSDKVersion   : 9.00.00.4509
    
    WMFSDKNeeded    : 0.0.0.0000
    
    album           : Álbum desconocido (19/12/2013 11:32:53 a.m.)
    
    track           : 1
    
    WM/EncodingTime : 18446744072125569792
    
    WM/UniqueFileIdentifier: ;
    
    IsVBR           : 0
    
    DeviceConformanceTemplate: L1
    
    WM/WMADRCPeakReference: 32096
    
    WM/WMADRCAverageReference: 10814
    
    title           : Pista 1
    
  Duration: 00:02:45.47, start: 0.000000, bitrate: 129 kb/s
    
    Stream #0:0(eng): Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, stereo, fltp, 128 kb/s
    
Output #0, asf, to '':
    
  Metadata:
    
    WM/Track        : 0
    
    WM/MediaPrimaryClassID: {D1607DBC-E323-4BE2-86A1-48A42A28441E}
    
    WMFSDKVersion   : 9.00.00.4509
    
    WMFSDKNeeded    : 0.0.0.0000
    
    WM/AlbumTitle   : Álbum desconocido (19/12/2013 11:32:53 a.m.)
    
    WM/TrackNumber  : 1
    
    WM/EncodingTime : 18446744072125569792
    
    WM/UniqueFileIdentifier: ;
    
    IsVBR           : 0
    
    DeviceConformanceTemplate: L1
    
    WM/WMADRCPeakReference: 32096
    
    WM/WMADRCAverageReference: 10814
    
    title           : Pista 1
    
    WM/EncodingSettings: Lavf55.37.101
    
    Stream #0:0(eng): Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, stereo, 128 kb/s
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (copy)
    
Press [q] to stop, [?] for help
    
size=      33kB time=00:00:01.35 bitrate= 196.9kbits/s    

    FFmpeg version :

    ffmpeg version 2.2.git Copyright (c) 2000-2014 the FFmpeg developers

    How can I fix command for getting right duration ?

    Thank you

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