Recherche avancée

Sur d’autres sites (5676)

• Revision 118606 : Version 1.4.0. - Nouveauté la plus visible : petite refacto du menu de ...

  13 novembre 2019, par Charles Razack — Log

  Version 1.4.0.
   Nouveauté la plus visible : petite refacto du menu de langues → Noms des langues en entier au lieu des codes (fr, en...), liens affichés sous formes de boutons plus grands et centrés pour être bien visibles, ajout d’un label. Petite amélioration pour éviter qu’il y ait un saut lorsque le menu passe en sticky.
   Configuration : les formulaires sur lesquels activer le script sont tous regroupés dans une même clé formulaires.
   Configuration : ajout d’une option pour ajouter des sélecteurs dans le paramètre root (exemple complètement au hasard : .formulaire_editer_noisette).
   Ajout d’un pipeline multilang_parametres pour permettre aux plugins de changer les paramètres passés au script d’init. Cas le plus courant : ajouter des formulaires à prendre en compte par le script.
   Refactorisation du script d’init afin qu’il soit plus lisible et maintenable : séparation du javascript et du php. Au passage, correction d’un bug rigolo qui faisait que quand on décochait tous les items dans la config, le script devenait actif sur *tous* les formulaires : recherche, login, etc.

• FFMPEG and FFPlay can access rtsp stream from one ip, but from other ip, it can't

  28 mai 2022, par Crear

  The situation is kind of complex. I was archiving several CCTV camera feeds (rtsp, h264, no audio) through OpenCV, which worked but the CPU utilization was too high and started to lose some frames time by time.

  


  To reduce the CPU utilization, I started to use FFMPEG to skip the decoding and encoding processes, which worked perfectly on my home machine. However, when I connected to my university VPN and tried to deploy it on our lab server, FFmpeg couldn't read any frame, ffplay couldn't get anything either. However, OpenCV, VLC Player and IINA Player could still read and display the feed.

  


  In Summary,

  


  1 FFMPEG/ffplay

  


  1.1 can only read the feed from my home network(Wi-Fi, optimum)

  


  1.2 from other two networks, the error message says : "Could not find codec parameters for stream 0 (Video : h264, none) : unspecified size
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options"

  


  2 IINA/VLC Player, OpenCV
These tools can get the video all the time.

  


  I'm wondering whether it's related to some specific port access, that the ffmpeg required but the others don't. I'd appreciate it if anyone can provide any suggestions.

  


  As references, the tested ffplay command is simple :

  


  ffplay 'the rtsp address'


  


  Thanks

  


  ---

  


  Update

  


  More tests have been performed.

  


  By specifying rtsp_transport as TCP, ffplay can play the video, but FFmpeg can't access the video. (In the beginning, when both FFmpeg and ffplay worked through my home network, it was UDP)
The FFmpeg command is as follows :

  


  ffmpeg -i rtsp://the_ip_address/axis-media/media.amp -hide_banner -c:v copy -s 1920x1080 -segment_time 00:30:00 -f segment -strftime 1 -reset_timestamps 1 -rtsp_transport tcp "%Y-%m-%d-%H-%M-%S_Test.mp4"


  


  Please help...

  


• using pocketsphinx_continuous with a .wav file

  3 avril 2013, par user2242131

  I am attempting to write an application that will allow a user to speak a small set of commands from a remote system and have them executed on my server. Using pocketsphinx to parse the spoken text. When run locally with the microphone, pocketsphinx_continuous works perfectly no matter how I slur the words. But when importing the audio file and using ffmpeg to downsample the audio to a single channel, 16 bit PCM file, it will parse the first word without difficulty. Then it will skip everything else and treat it as . I am confident that the problem is in the file format and not in the pocketsphinx configuration.

  Using command line
  ffmpeg -y -i Sound\AddSheet.wav -ac 1 -f s16le -acodec pcm_s16le -ar 16k AddTmp.wav
  in a batch file.

  The bottom of the output I get is :

  INFO: fsg_search.c(1407): Start node ADD.0:5:47
  
  INFO: fsg_search.c(1407): Start node <sil>.0:2:49
  
  INFO: fsg_search.c(1446): End node <sil>.126:128:305 (-486)
  
  INFO: fsg_search.c(1662): lattice start node <s>.0 end node <sil>.126
  
  INFO: ps_lattice.c(1352): Normalizer P(O) = alpha(<sil>:126:305) = -175371
  
  INFO: ps_lattice.c(1390): Joint P(O,S) = -176076 P(S|O) = -705
  
  000000000: ADD USER
  
  </sil></sil></s></sil></sil>

  Which is not the audio in the file. The words spoken in the file are "ADD SPREADSHEET", which works perfectly from the same microphone without the intervening .wav file.

  I have tried increasing the audio volume and decreasing the background noise using sox :

  sox -v 3.0 Sound\%1 Sound\%1-loud.wav ffmpeg -i Sound\%1-loud.wav -vn -ss 00:00:00 -t 00:00:01 -y Sound\%1-noiseaud.wav 
  
  sox Sound\%1-noiseaud.wav -n noiseprof Sound\%1-noise.prof 
  
  sox Sound\%1 Sound\%1-clean.wav noisered sound\noise.prof 0.21 
  
  ffmpeg -y -i Sound\%1-clean.wav -ac 1 -f s16le -acodec pcm_s16le -ar 16k AddTmp.wav

  with no noticeable effect on the final results.

  If you look at the output you will notice that fsg_search.c has found ADD as the start node, then silence for the remainder. Please help on this.