
Recherche avancée
Autres articles (60)
-
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 (...) -
Organiser par catégorie
17 mai 2013, parDans MédiaSPIP, une rubrique a 2 noms : catégorie et rubrique.
Les différents documents stockés dans MédiaSPIP peuvent être rangés dans différentes catégories. On peut créer une catégorie en cliquant sur "publier une catégorie" dans le menu publier en haut à droite ( après authentification ). Une catégorie peut être rangée dans une autre catégorie aussi ce qui fait qu’on peut construire une arborescence de catégories.
Lors de la publication prochaine d’un document, la nouvelle catégorie créée sera proposée (...) -
Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs
Sur d’autres sites (11777)
-
FFMPEG on MACOSX is complaining "no matches found"
15 juin 2022, par Byte PlayerI have a FFMPEG command line function that works perfectly on Windows but on Mac produces the following error :


"no matches found [1:a]adelay=15000:all=1[aud2]"

"no matches found [2:a]adelay=5000:all=1[aud3]"

Here is the command (less the full paths which just made it very hard to read). I've verified that the files exist at the paths specified by copying the file path from the command line and going into terminal, typing "open" then pasting in the copied path and pressing enter. In all cases they played.


ffmpeg -loglevel warning -hide_banner -y -i "file1.mp3" -t 5 -i "file2.mp3" -i "file3.mp3" -i "file4.mp3" -ss 25 -t 15 -i "file5.mp3" -ss 15 -t 5 -i "file6.mp3" -filter_complex [1:a]adelay=15000:all=1[aud1];[2:a]adelay=5000:all=1[aud2];[4:a]adelay=25000:all=1[aud4];[5:a]adelay=5000:all=1[aud5];[aud1][aud2][aud4][aud5]amix=inputs=6:duration=longest:dropout_transition=0:normalize=0 "output.mp3"



I know the immediate response (as it should be), is update your FFMPEG but I got the latest build (built on 2022-06-12) and here's what it reports...


ffmpeg version N-107092-g843c4346b1-tessus Copyright (c) 2000-2022 the FFmpeg developers
 built with Apple clang version 11.0.0 (clang-1100.0.33.17)
 configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvmaf --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-version3 --pkg-config-flags=--static --disable-ffplay
 libavutil 57. 26.100 / 57. 26.100
 libavcodec 59. 33.100 / 59. 33.100
 libavformat 59. 24.100 / 59. 24.100
 libavdevice 59. 6.100 / 59. 6.100
 libavfilter 8. 40.100 / 8. 40.100
 libswscale 6. 6.100 / 6. 6.100
 libswresample 4. 6.100 / 4. 6.100
 libpostproc 56. 5.100 / 56. 5.100
Hyper fast Audio and Video encoder
usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}...



Any insight or help would be GREATLY appreciated.


-
How can I convert a mp4 file to webm using java only ?
31 octobre 2023, par mir-shakirI am trying to convert a file from mp4 to webm. I am trying to use the JAVE wrapper of FFmpeg. I am getting the error.
Here is my code :


private String ConvertVideo(URL url) {
 File target =null;
 MultimediaObject multimediaObject = new MultimediaObject(url);
 try {
 target = File.createTempFile("target", ".webm");

 } catch (IOException e1) {
 e1.printStackTrace();
 }
 AudioAttributes audio = new AudioAttributes();
 audio.setCodec(AudioAttributes.DIRECT_STREAM_COPY);
 audio.setBitRate(new Integer(128000));
 audio.setSamplingRate(new Integer(44100));
 audio.setChannels(new Integer(2));
 VideoAttributes video = new VideoAttributes();
 video.setBitRate(new Integer(160000));
 video.setFrameRate(new Integer(15));
 video.setCodec("libvpx-vp9");
 video.setCodec(VideoAttributes.DIRECT_STREAM_COPY);
 EncodingAttributes attrs = new EncodingAttributes();
 attrs.setFormat("webm");
 attrs.setAudioAttributes(audio);
 attrs.setVideoAttributes(video);
 
 try {
 Encoder encoder = new Encoder(); 
 encoder.encode(multimediaObject, target, attrs);
 } catch (Exception e) { 
 e.printStackTrace();
 }
 
 return "success";
}



I am getting the below error :


2022-Jun-13 11:12:55 AM [qtp1914526580-175] ERROR ws.schild.jave.Encoder - Process exit code: 1 to target2636257785060285182.webm
ws.schild.jave.EncoderException: Exit code of ffmpeg encoding run is 1



What am I doing wrong here. Is there any other way around it ? I want to do it only using java.


-
VLC RTSP HTML5 transcoding
30 mai 2022, par PierogiI'm trying to get audio streaming on an HTML page from an RTSP server.


The RTSP server is the rtsp-simple-server running a command line below.

./rtsp-simple-server rtsp-simple-server.yml
.

The configure file is the default.

The stream player is FFmpeg running a command line below.

ffmpeg -re -stream_loop -1 -i myaudio.mp3 -c copy -f rtsp -rtsp_transport tcp rtsp://localhost:8554/mystream


The console log at the time the rtsp-simple-server and the ffmpeg are started is below.


2022/05/29 19:06:38 INF rtsp-simple-server v0.18.4
2022/05/29 19:06:38 INF [RTSP] listener opened on :8554 (TCP), :8000 (UDP/RTP), :8001 (UDP/RTCP)
2022/05/29 19:06:38 INF [RTMP] listener opened on :1935
2022/05/29 19:06:38 INF [HLS] listener opened on :8888
2022/05/29 19:09:16 INF [RTSP] [conn [::1]:62737] opened
2022/05/29 19:09:16 INF [RTSP] [session 271690815] created by [::1]:62737
2022/05/29 19:09:16 INF [RTSP] [session 271690815] is publishing to path 'mystream', 1 track with TCP



And the time the rtsp path(rtsp ://localhost:8554/mystream) is opened by VLC, the contents can be played properly. The additional console log at the time is below.


2022/05/29 19:13:19 INF [RTSP] [conn 127.0.0.1:62780] opened
2022/05/29 19:13:19 INF [RTSP] [session 734209460] created by 127.0.0.1:62780
2022/05/29 19:13:19 INF [RTSP] [session 734209460] is reading from path 'mystream', 1 track with UDP
2022/05/29 19:13:29 INF [RTSP] [session 734209460] destroyed (teared down by 127.0.0.1:62780)
2022/05/29 19:13:29 INF [RTSP] [conn 127.0.0.1:62780] closed (EOF)
2022/05/29 19:13:29 INF [RTSP] [conn 127.0.0.1:62781] opened
2022/05/29 19:13:29 INF [RTSP] [session 445756113] created by 127.0.0.1:62781
2022/05/29 19:13:29 INF [RTSP] [session 445756113] is reading from path 'mystream', 1 track with TCP



However, I open the rtsp streaming from the VLC's "Network" tab like below,



and configure the "Stream output" like below,



and I tried to get this streaming from an HTML page like below,




 
 
 
 
 
 <h1>transcode test</h1>
 <audio src="http://localhost:9999/mystream" autoplay="autoplay"></audio>
 




the browser console displays
Failed to load resource: the server responded with a status of 404 (Not found)
. I've already tried other ports(etc. 8080).

So, how can I get the rtsp stream from the RTSP server on an HTML page.
Any idea ?


My environment.


- 

- Browser : Microsoft edge
- OS : MacOS 11.6.5
- rtsp-simple-server : 0.18.4
- FFmpeg : 5.0.1
- VLC : 3.0.17.3