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  • 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 (...)

  • Multilang : améliorer l’interface pour les blocs multilingues

    18 février 2011, par

    Multilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
    Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela.

  • 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.

Sur d’autres sites (8441)

  • How to solve Jiiter Buffer problem in receiving audio RTP stream (bad sound quality) in PJSIP ?

    1er août 2019, par Mayur Patel

    I’m a newbie to pjsip and want to build an RTP stream receiver using pjsip.

    Setup :

    I want to use specific L16/16000/1 codec and have also enabled it in "config_site.h" during compiling the pjsip project and checked that its available

    Receiver :

    • BeagleBone
    • CrossCompiled Pjsip and Installed all req. libs and sample apps

    Sender :

    • Another Windows PC in the same Network using FFmpeg to transmit Audio Stream via Multicast

    I got to know about streamutil.c(pjsip sample-apps) which does similar things to send and receive both. Now for the sake of easyness, I’m using the same Cross-Compiled binary streamutil.

    SENDER :

    ..\ffmpeg -re -stream_loop -1 -i test.mp3 -ar 16000 -acodec pcm_s16be -b:a 128k -ac 1 -payload_type 123 -f rtp udp://239.255.255.211:5500?pkt_size=652
    

    
......
    
Output #0, rtp, to 'udp://239.255.255.211:5500?pkt_size=652':
    
  Metadata:
    
    title           : -----
    
    artist          : --------
    
    album           : -------
    
    date            : 2019
    
    track           : 1
    
    encoder         : Lavf58.20.100
    
    Stream #0:0: Audio: pcm_s16be, 16000 Hz, mono, s16, 256 kb/s
    
    Metadata:
    
      encoder         : Lavc58.35.100 pcm_s16be
    
SDP:
    
v=0
    
o=- 0 0 IN IP4 127.0.0.1
    
s=GREATEST HITS (2) [1 HOUR 20 MINUTES LONG]
    
c=IN IP4 239.255.255.211/5
    
t=0 0
    
a=tool:libavformat 58.26.101
    
m=audio 5500 RTP/AVP 123
    
b=AS:256
    
a=rtpmap:97 L16/16000/1
    
a=rtpmap:123 L16/16000/1
    
a=control:streamid=
    

    
size=     833kB time=00:00:25.91 bitrate= 263.4kbits/s speed=   1x

    RECIEVER LOG :

    ./streamutil --mcast-addr=239.255.255.211 --recv-only --codec=L16/16000/1
    
...
    
...
    
17:05:05.178     strm0x55dee1537f48  Jitter buffer starts returning normal frames (after 1 empty/lost)
    
17:05:05.246     strm0x55dee1537f48  Jitter buffer empty (prefetch=0), plc invoked
    
17:05:05.266     strm0x55dee1537f48  Jitter buffer starts returning normal frames (after 1 empty/lost)
    
17:05:05.325     strm0x55dee1537f48  Jitter buffer empty (prefetch=0), plc invoked
    
17:05:05.344     strm0x55dee1537f48  Jitter buffer starts returning normal frames (after 1 empty/lost)
    
17:05:05.422     strm0x55dee1537f48  Jitter buffer empty (prefetch=0), plc invoked

    Tried So far :

    • set different payload_type
    • set specific codec in streamutil as parameter
    • all other parameters in FFmpeg ex. bitrate, clockrate, channels

    Check working stream

    I am facing no issue, if I use a *.sdp file to receive RTP stream in VLC.

    SDP file :

    v=0
    
o=- 0 0 IN IP4 127.0.0.1
    
s=GREATEST HITS (2) [1 HOUR 20 MINUTES LONG]
    
c=IN IP4 239.255.255.211/5
    
t=0 0
    
a=tool:libavformat 58.26.101
    
m=audio 5500 RTP/AVP 123
    
b=AS:256
    
a=rtpmap:97 PCMU/8000/1
    
a=rtpmap:123 PCMU/8000/1
    
a=control:streamid=

    I have googled a lot but stuck now at this problem.
    So finally my question is that,
    How can I get the same Output via Pjsip without this Jitter Buffer logging and dropped sound ?

    Any help would be greatly appreciated.!

  • Revision 71faaa5b58 : Merge "One-pass rate control fixes and cleanups"

    4 février 2014, par Deb Mukherjee

    Merge "One-pass rate control fixes and cleanups"

  • Revision 090574c0be : Merge "vp8 : Code cleanup for control of denoiser mode."

    9 août 2014, par Marco Paniconi

    Merge "vp8 : Code cleanup for control of denoiser mode."

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