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Richard Stallman et le logiciel libre
19 octobre 2011, par
Mis à jour : Mai 2013
Langue : français
Type : Texte
Autres articles (61)
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Le plugin : Podcasts.
14 juillet 2010, parLe problème du podcasting est à nouveau un problème révélateur de la normalisation des transports de données sur Internet.
Deux formats intéressants existent : Celui développé par Apple, très axé sur l’utilisation d’iTunes dont la SPEC est ici ; Le format "Media RSS Module" qui est plus "libre" notamment soutenu par Yahoo et le logiciel Miro ;
Types de fichiers supportés dans les flux
Le format d’Apple n’autorise que les formats suivants dans ses flux : .mp3 audio/mpeg .m4a audio/x-m4a .mp4 (...) -
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 -
Récupération d’informations sur le site maître à l’installation d’une instance
26 novembre 2010, parUtilité
Sur le site principal, une instance de mutualisation est définie par plusieurs choses : Les données dans la table spip_mutus ; Son logo ; Son auteur principal (id_admin dans la table spip_mutus correspondant à un id_auteur de la table spip_auteurs)qui sera le seul à pouvoir créer définitivement l’instance de mutualisation ;
Il peut donc être tout à fait judicieux de vouloir récupérer certaines de ces informations afin de compléter l’installation d’une instance pour, par exemple : récupérer le (...)
Sur d’autres sites (6367)
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Merge commit '6c916192f3d7441f5896f6c0fe151874fcd91fe4'
9 avril 2017, par Clément BœschMerge commit '6c916192f3d7441f5896f6c0fe151874fcd91fe4'
* commit '6c916192f3d7441f5896f6c0fe151874fcd91fe4' :
mimic : Convert to the new bitstream reader
metasound : Convert to the new bitstream reader
lagarith : Convert to the new bitstream reader
indeo : Convert to the new bitstream reader
imc : Convert to the new bitstream reader
webp : Convert to the new bitstream readerThis merge is a noop, see
http://ffmpeg.org/pipermail/ffmpeg-devel/2017-April/209609.htmlMerged-by : Clément Bœsch <u@pkh.me>
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FFmpeg rtp streaming opus file problems
22 avril 2017, par Yuriy Aizenbergi have the next situation.
- Have file on remote VPS server.
- I want that this file (opus codec) can be accessible through RTP on my android phone.
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I tried ffmpeg with next command :
ffmpeg -ar 44800 -i bon_jovi_loverboy.opus -acodec libopus -ac 1 -ab 96k -vn -f rtp rtp://127.0.0.1:5004 -loglevel 56
But got next error :
bon_jovi_loverboy.opus: Invalid data found when processing input
Full log :
root@cs82932 :/home/rstream/rtstream/src# ffmpeg -ar 44800 -i bon_jovi_loverboy.opus -acodec libopus -ac 1 -ab 96k -vn -f rtp rtp ://127.0.0.1:5004 -loglevel 56
ffmpeg version 2.8.11-0ubuntu0.16.04.1 Copyright (c) 2000-2017 the FFmpeg developers
built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1 16.04.4) 20160609
configuration : —prefix=/usr —extra-version=0ubuntu0.16.04.1 —build-suffix=-ffmpeg —toolchain=hardened —libdir=/usr/lib/x86_64-linux-gnu —incdir=/usr/include/x86_64-linux-gnu —cc=cc —cxx=g++ —enable-gpl —enable-shared —disable-stripping —disable-decoder=libopenjpeg —disable-decoder=libschroedinger —enable-avresample —enable-avisynth —enable-gnutls —enable-ladspa —enable-libass —enable-libbluray —enable-libbs2b —enable-libcaca —enable-libcdio —enable-libflite —enable-libfontconfig —enable-libfreetype —enable-libfribidi —enable-libgme —enable-libgsm —enable-libmodplug —enable-libmp3lame —enable-libopenjpeg —enable-libopus —enable-libpulse —enable-librtmp —enable-libschroedinger —enable-libshine —enable-libsnappy —enable-libsoxr —enable-libspeex —enable-libssh —enable-libtheora —enable-libtwolame —enable-libvorbis —enable-libvpx —enable-libwavpack —enable-libwebp —enable-libx265 —enable-libxvid —enable-libzvbi —enable-openal —enable-opengl —enable-x11grab —enable-libdc1394 —enable-libiec61883 —enable-libzmq —enable-frei0r —enable-libx264 —enable-libopencv
libavutil 54. 31.100 / 54. 31.100
libavcodec 56. 60.100 / 56. 60.100
libavformat 56. 40.101 / 56. 40.101
libavdevice 56. 4.100 / 56. 4.100
libavfilter 5. 40.101 / 5. 40.101
libavresample 2. 1. 0 / 2. 1. 0
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 2.101 / 1. 2.101
libpostproc 53. 3.100 / 53. 3.100
Splitting the commandline.
Reading option ’-ar’ ... matched as option ’ar’ (set audio sampling rate (in Hz)) with argument ’44800’.Reading option ’-i’ ... matched as input url with argument ’bon_jovi_loverboy.opus’.
Reading option ’-acodec’ ... matched as option ’acodec’ (force audio codec (’copy’ to copy stream)) with argument ’libopus’.
Reading option ’-ac’ ... matched as option ’ac’ (set number of audio channels) with argument ’1’.
Reading option ’-ab’ ... matched as option ’ab’ (audio bitrate (please use -b:a)) with argument ’96k’.
Reading option ’-vn’ ... matched as option ’vn’ (disable video) with argument ’1’.
Reading option ’-f’ ... matched as option ’f’ (force format) with argument ’rtp’.
Reading option ’rtp ://127.0.0.1:5004’ ... matched as output url.
Reading option ’-loglevel’ ... matched as option ’loglevel’ (set logging level) with argument ’56’.
Finished splitting the commandline.
Parsing a group of options : global .
Applying option loglevel (set logging level) with argument 56.Successfully parsed a group of options.
Parsing a group of options : input url bon_jovi_loverboy.opus.
Applying option ar (set audio sampling rate (in Hz)) with argument 44800.
Successfully parsed a group of options.
Opening an input file : bon_jovi_loverboy.opus.[AVIOContext @ 0x965e60] Statistics : 36389 bytes read, 0 seeks
bon_jovi_loverboy.opus : Invalid data found when processing inputWhat wrong ? Thanks
UPD
I update ffmpeg and streaming looks successfully.
ffmpeg -stream_loop -1 -i 4a6u7-ptl2w.opus -acodec libopus -ac 1 -ab
96k -vn -f rtp rtp://95.213.195.192:5004/f.opus
ffmpeg version 3.2.4-1~16.04.york1 Copyright (c) 2000-2017 the FFmpeg developers
built with gcc 5.4.1 (Ubuntu 5.4.1-5ubuntu2~16.04.york1) 20170210
configuration: --prefix=/usr --extra-version='1~16.04.york1' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil 55. 34.101 / 55. 34.101
libavcodec 57. 64.101 / 57. 64.101
libavformat 57. 56.101 / 57. 56.101
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
Input #0, ogg, from '4a6u7-ptl2w.opus':
Duration: 00:02:34.21, start: 0.000000, bitrate: 69 kb/s
Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp
Metadata:
ENCODER : Lavc57.77.100 libopus
Output #0, rtp, to 'rtp://95.213.195.192:5004/f.opus':
Metadata:
encoder : Lavf57.56.101
Stream #0:0: Audio: opus (libopus), 48000 Hz, mono, flt, 96 kb/s
Metadata:
encoder : Lavc57.64.101 libopus
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 95.213.195.192
t=0 0
a=tool:libavformat 57.56.101
m=audio 5004 RTP/AVP 97
b=AS:96
a=rtpmap:97 opus/48000/2
Stream mapping:
Stream #0:0 -> #0:0 (opus (native) -> opus (libopus))
Press [q] to stop, [?] for help
size= 44893kB time=00:52:18.12 bitrate= 117.2kbits/s speed=56.5xBut when i try to connect by rtp from local PC (VLC Player) i get the exception :
core error: socket bind error: Cannot assign requested address
core error: open of `rtp://95.213.195.192:5004/f.opus' failedWhen i try ffplay on remote pc (same where ffmpeg) :
ffplay rtp://95.213.195.192:5004/f.opus
ffplay version 3.2.4-1~16.04.york1 Copyright (c) 2003-2017 the FFmpeg developers
built with gcc 5.4.1 (Ubuntu 5.4.1-5ubuntu2~16.04.york1) 20170210
configuration: --prefix=/usr --extra-version='1~16.04.york1' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil 55. 34.101 / 55. 34.101
libavcodec 57. 64.101 / 57. 64.101
libavformat 57. 56.101 / 57. 56.101
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
Segmentation faultCan you help me ? Thanks
95.213.195.192 - IP of my remote server with file
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Progress with rtc.io
12 août 2014, par silviaAt the end of July, I gave a presentation about WebRTC and rtc.io at the WDCNZ Web Dev Conference in beautiful Wellington, NZ.
Putting that talk together reminded me about how far we have come in the last year both with the progress of WebRTC, its standards and browser implementations, as well as with our own small team at NICTA and our rtc.io WebRTC toolbox.
One of the most exciting opportunities is still under-exploited : the data channel. When I talked about the above slide and pointed out Bananabread, PeerCDN, Copay, PubNub and also later WebTorrent, that’s where I really started to get Web Developers excited about WebRTC. They can totally see the shift in paradigm to peer-to-peer applications away from the Server-based architecture of the current Web.
Many were also excited to learn more about rtc.io, our own npm nodules based approach to a JavaScript API for WebRTC.
We believe that the World of JavaScript has reached a critical stage where we can no longer code by copy-and-paste of JavaScript snippets from all over the Web universe. We need a more structured module reuse approach to JavaScript. Node with JavaScript on the back end really only motivated this development. However, we’ve needed it for a long time on the front end, too. One big library (jquery anyone ?) that does everything that anyone could ever need on the front-end isn’t going to work any longer with the amount of functionality that we now expect Web applications to support. Just look at the insane growth of npm compared to other module collections :
Packages per day across popular platforms (Shamelessly copied from : http://blog.nodejitsu.com/npm-innovation-through-modularity/) For those that – like myself – found it difficult to understand how to tap into the sheer power of npm modules as a font end developer, simply use browserify. npm modules are prepared following the CommonJS module definition spec. Browserify works natively with that and “compiles” all the dependencies of a npm modules into a single bundle.js file that you can use on the front end through a script tag as you would in plain HTML. You can learn more about browserify and module definitions and how to use browserify.
For those of you not quite ready to dive in with browserify we have prepared prepared the rtc module, which exposes the most commonly used packages of rtc.io through an “RTC” object from a browserified JavaScript file. You can also directly download the JavaScript file from GitHub.
Using rtc.io rtc JS library So, I hope you enjoy rtc.io and I hope you enjoy my slides and large collection of interesting links inside the deck, and of course : enjoy WebRTC ! Thanks to Damon, JEeff, Cathy, Pete and Nathan – you’re an awesome team !
On a side note, I was really excited to meet the author of browserify, James Halliday (@substack) at WDCNZ, whose talk on “building your own tools” seemed to take me back to the times where everything was done on the command-line. I think James is using Node and the Web in a way that would appeal to a Linux Kernel developer. Fascinating !!
The post Progress with rtc.io first appeared on ginger’s thoughts.