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Websites made with MediaSPIP
2 mai 2011, parThis page lists some websites based on MediaSPIP.
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Ecrire une actualité
21 juin 2013, parPrésentez les changements dans votre MédiaSPIP ou les actualités de vos projets sur votre MédiaSPIP grâce à la rubrique actualités.
Dans le thème par défaut spipeo de MédiaSPIP, les actualités sont affichées en bas de la page principale sous les éditoriaux.
Vous pouvez personnaliser le formulaire de création d’une actualité.
Formulaire de création d’une actualité Dans le cas d’un document de type actualité, les champs proposés par défaut sont : Date de publication ( personnaliser la date de publication ) (...) -
Publier sur MédiaSpip
13 juin 2013Puis-je poster des contenus à partir d’une tablette Ipad ?
Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir
Sur d’autres sites (5628)
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Merge commit '3794062ab1a13442b06f6d76c54dce51ffa54697'
9 avril 2017, par Clément BœschMerge commit '3794062ab1a13442b06f6d76c54dce51ffa54697'
* commit '3794062ab1a13442b06f6d76c54dce51ffa54697' :
Remove Plan 9 supportMerged-by : Clément Bœsch <u@pkh.me>
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VLC - Could someone assist me into improving latency in streaming to web based app ?
19 janvier 2017, par zyeekI have been looking for solutions in which I can stream an IP camera’s stream to
HTML 5
. Currently as is it doesn’t supportRTSP
so easily.I am trying to be able to view the camera’s stream as live as possible. I was hoping someone could help me achieve that. I have been playing with it to get something workable, but at the moment I get a 5s delay stream. It is smooth, but wish to get it hopefully within <1-2s delay if possible.
My current setup goes from taking my IP camera’s stream in
RTSP
and converting it to awebm
and streaming it to a url, which then I plan on using that to put else where in a web app.What I would like to achieve
Use a protocol that has low latency with audio was well. Webm was used as test, but I can’t seem to get other commands to get the proper stream to be going.
I would like to use DASH, but from reading
FFMPEG
currently doesn’t support it. I was thinking maybeRTMP
would be good enough for now, being both low latency andHTTP 5
compatible. I am just unable to figure out how to getFFMPEG
to transcode theRTSP
toRTMP
.SETUP :
I am using ffserver and ffmpeg. Overall scope : trying to pull IP camera stream and put it on a web app.
Framework I am use is Meteor JS. So, I am trying to few plugins or outside complex setups as I want to be able to deploy this Meteor app on mobile devices as well. So, I want to stay within the boundaries of what
HTML 5
can support.My current ffserver setup is
ffserver.conf
(this was taking from bunch of different place :HTTPPort 8090 # Port to bind the server to
HTTPBindAddress 0.0.0.0
MaxHTTPConnections 2000
MaxClients 1000
MaxBandwidth 10000 # Maximum bandwidth per client
# set this high enough to exceed stream bitrate
CustomLog -
<feed>
File /tmp/feed.ffm
FileMaxSize 100K
ACL allow 127.0.0.1
</feed>
<stream>
Format webm
Feed feed.ffm
NoAudio
VideoCodec libvpx
VideoFrameRate 24
VideoBitRate 1024
VideoSize 480x270
VideoBufferSize 1024
AVOptionVideo flags +global_header
StartSendOnKey
</stream>
<stream> # Server status URL
Format status
# Only allow local people to get the status
ACL allow localhost
</stream>
<redirect> # Just an URL redirect for index
# Redirect index.html to the appropriate site
URL url/
</redirect>Works normally :
ffserver version 3.2.2 Copyright (c) 2000-2016 the FFmpeg developers
built with Apple LLVM version 5.1 (clang-503.0.40) (based on LLVM 3.4svn)
configuration: --prefix=/usr/local/Cellar/ffmpeg/3.2.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-frei0r --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-libopenjpeg --disable-decoder=jpeg2000 --extra-cflags=-I/usr/local/Cellar/openjpeg/2.1.2/include/openjpeg-2.1 --enable-nonfree --enable-vda
libavutil 55. 34.100 / 55. 34.100
libavcodec 57. 64.101 / 57. 64.101
libavformat 57. 56.100 / 57. 56.100
libavdevice 57. 1.100 / 57. 1.100
libavfilter 6. 65.100 / 6. 65.100
libavresample 3. 1. 0 / 3. 1. 0
libswscale 4. 2.100 / 4. 2.100
libswresample 2. 3.100 / 2. 3.100
libpostproc 54. 1.100 / 54. 1.100
/etc/ffserver.conf:27: Setting default value for video bit rate tolerance = 256000. Use NoDefaults to disable it.
/etc/ffserver.conf:27: Setting default value for video rate control equation = tex^qComp. Use NoDefaults to disable it.
/etc/ffserver.conf:27: Setting default value for video max rate = 2048000. Use NoDefaults to disable it.
Wed Jan 18 17:04:30 2017 FFserver started.Now I give life to the feed with
ffmpeg
. Command I use :ffmpeg -vsync 2 -i rtsp://admin:password@192.168.2.165:88/videoMain -map 0 http://localhost:8090/feed.ffm
which gives the result :
ffmpeg version 3.2.2 Copyright (c) 2000-2016 the FFmpeg developers
built with Apple LLVM version 5.1 (clang-503.0.40) (based on LLVM 3.4svn)
configuration: --prefix=/usr/local/Cellar/ffmpeg/3.2.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-frei0r --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-libopenjpeg --disable-decoder=jpeg2000 --extra-cflags=-I/usr/local/Cellar/openjpeg/2.1.2/include/openjpeg-2.1 --enable-nonfree --enable-vda
libavutil 55. 34.100 / 55. 34.100
libavcodec 57. 64.101 / 57. 64.101
libavformat 57. 56.100 / 57. 56.100
libavdevice 57. 1.100 / 57. 1.100
libavfilter 6. 65.100 / 6. 65.100
libavresample 3. 1. 0 / 3. 1. 0
libswscale 4. 2.100 / 4. 2.100
libswresample 2. 3.100 / 2. 3.100
libpostproc 54. 1.100 / 54. 1.100
Guessed Channel Layout for Input Stream #0.1 : mono
Input #0, rtsp, from 'rtsp://admin:password@192.168.2.165:88/videoMain':
Metadata:
title : IP Camera Video
comment : videoMain
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0: Video: h264 (Main), yuv420p(progressive), 1280x720, 90k tbr, 90k tbn, 180k tbc
Stream #0:1: Audio: pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s
[libvpx @ 0x7fd58184a600] v1.6.0
Output #0, ffm, to 'http://localhost:8090/feed.ffm':
Metadata:
title : IP Camera Video
comment : videoMain
creation_time : now
encoder : Lavf57.56.100
Stream #0:0: Video: vp8 (libvpx), yuv420p, 480x270, q=-1--1, 1024 kb/s, 90k fps, 1000k tbn, 24 tbc
Metadata:
encoder : Lavc57.64.101 libvpx
Side data:
cpb: bitrate max/min/avg: 0/0/0 buffer size: 8388608 vbv_delay: -1
Stream mapping:
Stream #0:0 -> #0:0 (h264 (native) -> vp8 (libvpx))
Press [q] to stop, [?] for help
[rtsp @ 0x7fd581000000] max delay reached. need to consume packet
[rtsp @ 0x7fd581000000] RTP: missed 5 packets
[h264 @ 0x7fd5818ae800] Increasing reorder buffer to 1
frame= 139 fps= 18 q=0.0 Lsize= 440kB time=00:00:09.25 bitrate= 389.7kbits/s speed=1.19x
video:429kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.663893% -
FFMPEG send RTP audio at 8k bytes/sec
10 mai, par MuzzaI'm trying to use FFMPEG to mimick a device that transmits G711U audio over UDP/RTP at 8k bytes per second.
The device im mimicking sends rtp packets every 20ms with 160byte payload.


I've had limited success using the following command


ffmpeg -f dshow -i audio="Microphone (Realtek(R) Audio)" -ac 1 -ar 8000 -ab 8 -acodec pcm_mulaw -f rtp rtp://127.0.0.1:12345?pkt_size=160



This sends G711U encoded audio, in 160byte chunks, but streams at 64kB/s, not the 8kB/s that my device is expected, so the device errors out ?


Any idea's would be massively appreciated !


Thank you


Log from FFMPEG


>ffmpeg -f dshow -i audio="Microphone (Realtek(R) Audio)" -ac 1 -ar 8000 -ab 8 -acodec pcm_mulaw -f rtp rtp://127.0.0.1:12345?pkt_size=160
ffmpeg version 2025-04-23-git-25b0a8e295-essentials_build-www.gyan.dev Copyright (c) 2000-2025 the FFmpeg developers
 built with gcc 14.2.0 (Rev3, Built by MSYS2 project)
 configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-libfreetype --enable-libfribidi --enable-libharfbuzz --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-dxva2 --enable-d3d11va --enable-d3d12va --enable-ffnvcodec --enable-libvpl --enable-nvdec --enable-nvenc --enable-vaapi --enable-libgme --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-librubberband
 libavutil 60. 2.100 / 60. 2.100
 libavcodec 62. 0.101 / 62. 0.101
 libavformat 62. 0.100 / 62. 0.100
 libavdevice 62. 0.100 / 62. 0.100
 libavfilter 11. 0.100 / 11. 0.100
 libswscale 9. 0.100 / 9. 0.100
 libswresample 6. 0.100 / 6. 0.100
 libpostproc 59. 1.100 / 59. 1.100
[aist#0:0/pcm_s16le @ 00000198256b73c0] Guessed Channel Layout: stereo
Input #0, dshow, from 'audio=Microphone (Realtek(R) Audio)':
 Duration: N/A, start: 135470.702000, bitrate: 1411 kb/s
 Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s, Start-Time 135470.702s
Stream mapping:
 Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_mulaw (native))
Press [q] to stop, [?] for help
[pcm_mulaw @ 00000198256cf240] Bitrate 8 is extremely low, maybe you mean 8k
Output #0, rtp, to 'rtp://127.0.0.1:12345?pkt_size=160':
 Metadata:
 encoder : Lavf62.0.100
 Stream #0:0: Audio: pcm_mulaw, 8000 Hz, mono, s16 (8 bit), 64 kb/s
 Metadata:
 encoder : Lavc62.0.101 pcm_mulaw
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 62.0.100
m=audio 12345 RTP/AVP 0
b=AS:64

[out#0/rtp @ 00000198256cdd00] video:0KiB audio:973KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 8.467470%
size= 1055KiB time=00:02:04.51 bitrate= 69.4kbits/s speed= 1x
Exiting normally, received signal 2.



Wireshark :
Wireshark Log


Shows packets being sent every 0.20ms