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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 TW2

  I 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 Kanarek

  I 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.mp4

  I 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=sendrecv

  But 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 input

  Note 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=sendrecv

  However 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 seeks

  Note 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.