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Médias (3)
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MediaSPIP Simple : futur thème graphique par défaut ?
26 septembre 2013, par
Mis à jour : Octobre 2013
Langue : français
Type : Video
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GetID3 - Bloc informations de fichiers
9 avril 2013, par
Mis à jour : Mai 2013
Langue : français
Type : Image
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GetID3 - Boutons supplémentaires
9 avril 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Image
Autres articles (62)
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Use, discuss, criticize
13 avril 2011, parTalk 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. -
HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
For older browsers the Flowplayer flash fallback is used.
MediaSPIP allows for media playback on major mobile platforms with the above (...) -
Support audio et vidéo HTML5
10 avril 2011MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)
Sur d’autres sites (8222)
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Unable to find audio stream
18 novembre 2011, par KurtI'm trying to adapt a code in order to add a sound capture feature (on a live stream), with the help of ffmpeg and directshow.
When i try to play with ffplay the AVIsynth file, everything works perfectly i've got the audio and video. But when i open this input file by code i only find the video stream.
The Avs file :
V = DirectShowSource("Decklink_HDMI.grf", fps=10, framecount=1000000000, seek=false, audio=false)
A = DirectShowSource("Decklink_Audio.grf", fps=1, framecount=1000000000, video=false)
AudioDub(V, A)The opening code :
ffmpeg::AVInputFormat * ifmt;
ifmt = ffmpeg::av_find_input_format("avs");
// Open input file
if(ffmpeg::avformat_open_input(&pFormatCtx, filename.toStdString().c_str(), ifmt, NULL) != 0)When i make a variable lookout on gdb just after the opening.
i'm looking at nb_streams in pFormatCtx->nb_streams and it's at 1
The only stream i can find in pFormatCtx->stream is a video one. And that's why i'm not able to capture the sound.So i'm asking if i'm doing something wrong with my opening or if i miss something.
Ps : I know the code shown is short but the problem appears at this very step !
Thank you
Kurt
— EDIT —
I've also noticed that when I dump the AVFormatContext the video stream got a 456x32 size.
And i can find this very same size of the window displayed when i try to launch a corrupted script with ffplay.The original video format when i play the correct script with ffplay is of 1920x1080
I think my problem is maybe deeper than the simple fact of not being able to get the audio stream.
I'm trying to find out how to know the error message that is displayed on this 456x32 windows
— EDIT2 —
I Find out what is written on this image and my problem is solved, badly placed avs script an old one was is place.
I'm ashamed.
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Concatenate mp4 files in Android using halfninja ffmpeg
21 février 2015, par BudiusI’ve manage to compile halfninja ffmpeg scripts for Android NDK using NDK version r5c. (Unfortunately any attempt to compile with earlier NDK generated some error), also I’m not very knowledgeable on the whole NDK process, so it’s a bit hit-n-miss for me.
His scripts are compiling ffmpeg version N-30996-gf925b24 (the specific commit he did the scripts for)
Moving forward to my actual app.
I manage to trim videos without problems, now I need to join/concatenate them but any attemp at using any and several combinations of the commands found on those 3 links (link1, link2, link3) generate errors such ascat is not valid
,> is undefinined
,unknown option filter_complex
or trying to override some of the input files.Does anyone know if it’s possible and (how to do it), to join/concatenate mp4 videos (all same codec, size, quality, etc) using half-ninja compile of ffmpeg on Android, or how to compile/get a ffmpeg for Android using latest source codes ?
I’ve also gave a quick try on the mp4Parser without much success.
ultimately I was trying to get this pseudo-method to work :
public static File concatenate(String[] inputPaths, String outputPath){
// ... do stuff do generate ffmpeg commands....
VideoKit v = new VideoKit();
v.run(cmds);
File f = new File(outputPath);
return f;
} -
Streaming raw sound with FFMPEG
30 juin 2020, par Dr_ClickI'm using the ffmpeg library to decode / encode audio in JAVA, using the Process objets. The aim is to got the raw datas decoded by ffmpeg in my JAVA code and then, to send them back to ffmpeg to generate a stream or create a file.



As JAVA sounds only allow wav audio datas, I made a first try with this command line :



ffmpeg -i http://199.180.75.118:80 -acodec pcm_s16le -ar 44100 -ac 2 "/home/dr_click/montest.wav"




It works. But, my goal is to get the frames on the fly and not a file. I thought about using a pipe but it seems it will work only with Linux and not with Windows (and no idea for MacOS)
So, I'd prefer to stream locally a wav file and to catch it with JAVA into a AudioInputStream.



I wrote this command :



ffmpeg -re -i http://199.180.75.118:80 -acodec pcm_s16le -ar 44100 -ac 2 -f rtp rtp://127.0.0.1:1234




It semms to work on the server side. But, when I enter the below command line on the server side :



ffplay rtp://127.0.0.1:1234




It doesn't work at all. I got this error :



- 

- [rtp @ 0x7f29c8000b80] Unable to receive RTP payload type 97 without an SDP file describing it





If I try the couple :



ffmpeg -i http://199.180.75.118:80 -acodec pcm_s16le -ar 44100 -ac 2 -f rtp -sdp_file /home/dr_click/audio.sdp rtp://127.0.1.1:1234




and



ffplay /home/dr_click/audio.sdp




I got the following error :



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- [rtp @ 0x7f7d00008040] Protocol 'rtp' not on whitelist 'file,crypto' ! if





And if I finally try :



fmpeg -protocol_whitelist file,http,rtp,tcp -i http://199.180.75.118:80 -acodec pcm_s16le -ar 44100 -ac 2 -f rtp -sdp_file /home/dr_click/audio.sdp rtp://127.0.1.1:1234




I still get the same error.



Which point do I miss to stream locally some raw datas and catch them back in the AudioInputStream ?



Thank you for all your replies.