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Autres articles (48)
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La sauvegarde automatique de canaux SPIP
1er avril 2010, parDans le cadre de la mise en place d’une plateforme ouverte, il est important pour les hébergeurs de pouvoir disposer de sauvegardes assez régulières pour parer à tout problème éventuel.
Pour réaliser cette tâche on se base sur deux plugins SPIP : Saveauto qui permet une sauvegarde régulière de la base de donnée sous la forme d’un dump mysql (utilisable dans phpmyadmin) mes_fichiers_2 qui permet de réaliser une archive au format zip des données importantes du site (les documents, les éléments (...) -
Script d’installation automatique de MediaSPIP
25 avril 2011, parAfin de palier aux difficultés d’installation dues principalement aux dépendances logicielles coté serveur, un script d’installation "tout en un" en bash a été créé afin de faciliter cette étape sur un serveur doté d’une distribution Linux compatible.
Vous devez bénéficier d’un accès SSH à votre serveur et d’un compte "root" afin de l’utiliser, ce qui permettra d’installer les dépendances. Contactez votre hébergeur si vous ne disposez pas de cela.
La documentation de l’utilisation du script d’installation (...) -
Encodage et transformation en formats lisibles sur Internet
10 avril 2011MediaSPIP transforme et ré-encode les documents mis en ligne afin de les rendre lisibles sur Internet et automatiquement utilisables sans intervention du créateur de contenu.
Les vidéos sont automatiquement encodées dans les formats supportés par HTML5 : MP4, Ogv et WebM. La version "MP4" est également utilisée pour le lecteur flash de secours nécessaire aux anciens navigateurs.
Les documents audios sont également ré-encodés dans les deux formats utilisables par HTML5 :MP3 et Ogg. La version "MP3" (...)
Sur d’autres sites (6454)
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Anomalie #3948 : Tous les plugins se désactivent
19 mai 2017, par nico d_J’ai activé les logs verbeux de SVP, et j’ai ajouté des logs dans
- ecrire/plugins/installer.php : spip_plugin_install() et plugins_installer_dist()
- ecrire/action/desinstaller_plugin.php : action_desinstaller_plugin_dist()
- ecrire/plugins/afficher_plugin.php : plugin_desintalle()
Comme je soupçonne un problème d’accès disque, j’ai doublé les logs fichiers par des logs dans une table de la base avec un inc_log() perso, qui rappelle inc_log_dist().
J’attends maintenant la reproduction du bug, mais il y a un évènement que je n’arrive pas à logger, c’est quand un plugin se désactive parce que son répertoire n’existe plus.
J’ai testé en renommant un plugin, mais ça ne logge rien, je n’arrive pas à voir où c’est traité. -
Can't play at a good framerate with JavaAV
31 mars 2017, par TW2I have a problem in framerate when I use JavaAV (which use JavaCPP + ffmpeg). When I set my framerate as mentionned at DemuxerExample.java, the frame change each 47000ms which is too long and false. It’s better when I set 1000 / demuxer.getFrameRate() * 1000 ( 47ms) but it’s also false. I can’t have a good framerate. Here is my player class :
package gr.av;
import hoary.javaav.Audio;
import hoary.javaav.AudioFrame;
import hoary.javaav.Demuxer;
import hoary.javaav.Image;
import hoary.javaav.JavaAVException;
import hoary.javaav.MediaFrame;
import hoary.javaav.VideoFrame;
import java.awt.Color;
import java.awt.Graphics;
import java.awt.image.BufferedImage;
import java.io.ByteArrayInputStream;
import java.io.ByteArrayOutputStream;
import java.io.IOException;
import java.util.logging.Level;
import java.util.logging.Logger;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioInputStream;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.LineUnavailableException;
import javax.sound.sampled.SourceDataLine;
import javax.sound.sampled.UnsupportedAudioFileException;
import javax.swing.JPanel;
public class Player extends JPanel {
VideoThread videoTHREAD;
AudioThread audioTHREAD;
BufferedImage img = null;
public Player() {
init();
}
private void init(){
videoTHREAD = new VideoThread(this);
audioTHREAD = new AudioThread();
}
public void setImage(BufferedImage img){
this.img = img;
repaint();
}
@Override
public void paint(Graphics g){
if(img != null){
g.drawImage(img, 0, 0, null);
}else{
g.setColor(Color.blue);
g.fillRect(0, 0, getWidth(), getHeight());
}
}
public void setFilename(String filename){
videoTHREAD.setFilename(filename);
audioTHREAD.setFilename(filename);
}
public void play(){
videoTHREAD.playThread();
audioTHREAD.playThread();
}
public void stop(){
videoTHREAD.stopThread();
audioTHREAD.stopThread();
}
public static class VideoThread extends Thread {
//The video filename and a controller
String filename = null;
private volatile boolean active = false;
//Panel to see video
Player player;
public VideoThread(Player player) {
this.player = player;
}
public void setFilename(String filename){
this.filename = filename;
}
public void playThread(){
if(filename != null && active == false){
active = true;
this.start();
}
}
public void stopThread(){
if(active == true){
active = false;
this.interrupt();
}
}
public void video() throws JavaAVException, InterruptedException, IOException, UnsupportedAudioFileException{
Demuxer demuxer = new Demuxer();
demuxer.open(filename);
MediaFrame mediaFrame;
while (active && (mediaFrame = demuxer.readFrame()) != null) {
if (mediaFrame.getType() == MediaFrame.Type.VIDEO) {
VideoFrame videoFrame = (VideoFrame) mediaFrame;
player.setImage(Image.createImage(videoFrame, BufferedImage.TYPE_3BYTE_BGR));
double FPS = demuxer.getFrameRate() * 1000d;
long ms = (long)(1000d / FPS);
System.out.println("FPS = " + FPS + " ; Milliseconds = " + ms);
java.util.concurrent.TimeUnit.MILLISECONDS.sleep(ms);
}
}
demuxer.close();
}
@Override
public void run() {
if(filename != null){
try {
video();
} catch (JavaAVException | InterruptedException | IOException | UnsupportedAudioFileException ex) {
Logger.getLogger(Player.class.getName()).log(Level.SEVERE, null, ex);
}
}
}
}
public static class AudioThread extends Thread {
//The video filename and a controller
String filename = null;
private volatile boolean active = false;
//Audio
AudioFormat format = new AudioFormat(44000, 16, 2, true, false);
AudioInputStream ais;
DataLine.Info info = new DataLine.Info(SourceDataLine.class, format);
SourceDataLine soundLine;
public AudioThread() {
}
public void setFilename(String filename){
this.filename = filename;
}
public void playThread(){
if(filename != null && active == false){
active = true;
soundON();
this.start();
}
}
public void stopThread(){
if(active == true){
active = false;
soundOFF();
this.interrupt();
}
}
public void audio() throws JavaAVException, IOException {
Demuxer demuxer = new Demuxer();
demuxer.open(filename);
MediaFrame mediaFrame;
while (active && (mediaFrame = demuxer.readFrame()) != null) {
if (mediaFrame.getType() == MediaFrame.Type.AUDIO) {
AudioFrame audioFrame = (AudioFrame) mediaFrame;
byte[] bytes = Audio.getAudio16(audioFrame);
try (ByteArrayInputStream bais = new ByteArrayInputStream(bytes)) {
ais = new AudioInputStream(bais, format, bytes.length / format.getFrameSize());
try (ByteArrayOutputStream baos = new ByteArrayOutputStream()) {
int nBufferSize = bytes.length * format.getFrameSize();
byte[] abBuffer = new byte[nBufferSize];
while (true){
int nBytesRead = ais.read(abBuffer);
if (nBytesRead == -1)
break;
baos.write(abBuffer, 0, nBytesRead);
}
byte[] abAudioData = baos.toByteArray();
soundLine.write(abAudioData, 0, abAudioData.length);
}
ais.close();
}
}
}
demuxer.close();
}
@Override
public void run() {
if(filename != null){
try {
audio();
} catch (JavaAVException | IOException ex) {
Logger.getLogger(Player.class.getName()).log(Level.SEVERE, null, ex);
}
}
}
private void soundON(){
try {
soundLine = (SourceDataLine) AudioSystem.getLine(info);
soundLine.open(format);
soundLine.start();
} catch (LineUnavailableException ex) {
Logger.getLogger(Player.class.getName()).log(Level.SEVERE, null, ex);
}
}
private void soundOFF(){
soundLine.drain();
soundLine.stop();
soundLine.close();
}
}
} -
Stream RTP to FFMPEG using SDP
21 mars 2017, par Johnathan KanarekI get RTP stream from WebRTC server (I used mediasoup) using node.js and I get the decrypted RTP packets raw data from the stream.
I want to forward this RTP data to ffmpeg and from there I can save it to file, or push it as RTMP stream to other media servers.
I guess that the best way would be to create SDP file that describes both the audio and video streams and send the packets therough new sockets.The ffmpeg command is :
ffmpeg -loglevel debug -protocol_whitelist file,crypto,udp,rtp -re -vcodec libvpx -acodec opus -i test.sdp -vcodec libx264 -acodec aac -y output.mp4I tried to send the packets through UDP :
v=0
o=mediasoup 7199daf55e496b370e36cd1d25b1ef5b9dff6858 0 IN IP4 192.168.193.182
s=7199daf55e496b370e36cd1d25b1ef5b9dff6858
c=IN IP4 192.168.193.182
t=0 0
m=audio 33301 RTP/AVP 111
a=rtpmap:111 /opus/48000
a=fmtp:111 minptime=10;useinbandfec=1
a=rtcp-fb:111 transport-cc
a=sendrecv
m=video 33302 RTP/AVP 100
a=rtpmap:100 /VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=sendrecvBut I always get (removed the boring parts) :
Opening an input file: test.sdp.
[sdp @ 0x103dea0]
Format sdp probed with size=2048 and score=50
[sdp @ 0x103dea0] audio codec set to: (null)
[sdp @ 0x103dea0] audio samplerate set to: 44100
[sdp @ 0x103dea0] audio channels set to: 1
[sdp @ 0x103dea0] video codec set to: (null)
[udp @ 0x10402e0] end receive buffer size reported is 131072
[udp @ 0x10400c0] end receive buffer size reported is 131072
[sdp @ 0x103dea0] setting jitter buffer size to 500
[udp @ 0x1040740] bind failed: Address already in use
[AVIOContext @ 0x1046980] Statistics: 473 bytes read, 0 seeks
test.sdp: Invalid data found when processing inputNote that I get it even if I don’t open socket at all or send anything to this port, as if the ffmpeg itself tries to open these ports more than once.
I tried also to open two (video and audio) TCP servers and define SDP with TCP :
v=0
o=mediasoup 7199daf55e496b370e36cd1d25b1ef5b9dff6858 0 IN IP4 192.168.193.182
s=7199daf55e496b370e36cd1d25b1ef5b9dff6858
c=IN IP4 192.168.193.182
t=0 0
m=audio 33301 TCP 111
a=rtpmap:111 /opus/48000
a=fmtp:111 minptime=10;useinbandfec=1
a=rtcp-fb:111 transport-cc
a=setup:active
a=connection:new
a=sendrecv
m=video 33302 TCP 100
a=rtpmap:100 /VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=setup:active
a=connection:new
a=sendrecvHowever I don’t see any incoming connection into my TCP servers and I get the following from ffmpeg :
Opening an input file: test.sdp.
[sdp @ 0xdddea0]
Format sdp probed with size=2048 and score=50
[sdp @ 0xdddea0]
audio codec set to: (null)
[sdp @ 0xdddea0]
audio samplerate set to: 44100
[sdp @ 0xdddea0] audio channels set to: 1
[sdp @ 0xdddea0] video codec set to: (null)
[udp @ 0xde02e0] end receive buffer size reported is 131072
[udp @ 0xde00c0] end receive buffer size reported is 131072
[sdp @ 0xdddea0] setting jitter buffer size to 500
[udp @ 0xde0740] end receive buffer size reported is 131072
[udp @ 0xde0180] end receive buffer size reported is 131072
[sdp @ 0xdddea0] setting jitter buffer size to 500
[sdp @ 0xdddea0] Before avformat_find_stream_info() pos: 593 bytes read:593 seeks:0 nb_streams:2
[libvpx @ 0xdeea80] v1.3.0
[libvpx @ 0xdeea80] --target=x86_64-linux-gcc --enable-pic --disable-install-srcs --as=nasm --enable-shared --prefix=/usr --libdir=/usr/lib64
[sdp @ 0xdddea0] Could not find codec parameters for stream 1 (Video: vp8, 1 reference frame, none): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[sdp @ 0xdddea0] After avformat_find_stream_info() pos: 593 bytes read:593 seeks:0 frames:0
Input #0, sdp, from 'test.sdp':
Metadata:
title : 7199daf55e496b370e36cd1d25b1ef5b9dff6858
Duration: N/A, bitrate: N/A
Stream #0:0, 0, 1/90000: Audio: opus, 48000 Hz, mono, fltp
Stream #0:1, 0, 1/90000: Video: vp8, 1 reference frame, none, 90k tbr, 90k tbn, 90k tbc
Successfully opened the file.
Parsing a group of options: output file output.mp4.
Successfully parsed a group of options.
Opening an output file: output.mp4.
[file @ 0xde3660] Setting default whitelist 'file,crypto'
Successfully opened the file.
detected 1 logical cores
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'time_base' to value '1/48000'
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'sample_rate' to value '48000'
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'sample_fmt' to value 'fltp'
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'channel_layout' to value '0x4'
[graph 0 input from stream 0:0 @ 0xde3940] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x4
[audio format for output stream 0:0 @ 0xe37900] Setting 'sample_fmts' to value 'fltp'
[audio format for output stream 0:0 @ 0xe37900] Setting 'sample_rates' to value '96000|88200|64000|48000|44100|32000|24000|22050|16000|12000|11025|8000|7350'
[AVFilterGraph @ 0xde0220] query_formats: 4 queried, 9 merged, 0 already done, 0 delayed
Output #0, mp4, to 'output.mp4':
Metadata:
title :
7199daf55e496b370e36cd1d25b1ef5b9dff6858
encoder :
Lavf57.56.100
Stream #0:0
, 0, 1/48000
: Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, mono, fltp, delay 1024, 69 kb/s
Metadata:
encoder :
Lavc57.64.100 aac
Stream mapping:
Stream #0:0 -> #0:0 (opus (native) -> aac (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
test.sdp: Connection timed out
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[output stream 0:0 @ 0xde3b40] EOF on sink link output stream 0:0:default.
No more output streams to write to, finishing.
[aac @ 0xde2b00] Trying to remove 1024 samples, but the queue is empty
[aac @ 0xde2b00] Trying to remove 1024 more samples than there are in the queue
[mp4 @ 0xe6a540] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
[mp4 @ 0xe6a540] Encoder did not produce proper pts, making some up.
[aac @ 0xde2b00] Trying to remove 1024 samples, but the queue is empty
[aac @ 0xde2b00] Trying to remove 1024 more samples than there are in the queue
size= 1kB time=00:00:00.04 bitrate= 157.9kbits/s speed=0.00426x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 3268.000000%
Input file #0 (test.sdp):
Input stream #0:0 (audio): 0 packets read (0 bytes); 0 frames decoded (0 samples);
Input stream #0:1 (video): 0 packets read (0 bytes);
Total: 0 packets (0 bytes) demuxed
Output file #0 (output.mp4):
Output stream #0:0 (audio): 0 frames encoded (0 samples); 2 packets muxed (25 bytes);
Total: 2 packets (25 bytes) muxed
0 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0xde37a0] Statistics: 30 seeks, 25 writeouts
[aac @ 0xde2b00] Qavg: 47249.418
[AVIOContext @ 0xde6980] Statistics: 593 bytes read, 0 seeksNote to the "Connection timed out" in the log above.
I guess that both my SDPs are wrong, any suggestions ?
Alternatives to SDP are also most welcomed.
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