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• looking for Settings for FFmpeg, xTeVe, Plex [closed]

  3 août 2022, par Chris

  I am not sure if I am in the right place. But maybe someone can help me.

  


  Sorry for my bad english.

  


  I have a Qnap NAS, on this is insalled FFmepg, xTeVe and Plex.
I try to see IPTV Streams on Plex.

  


  Here the log file from xTeVe :

  


  ---

  


  2022-02-28 21:57:42 [xTeVe] Buffer : true [ffmpeg]

  


  2022-02-28 21:57:42 [xTeVe] Buffer Size : 4096 KB

  


  2022-02-28 21:57:42 [xTeVe] Channel Name : VIP SW : SRF 1 FHD

  


  2022-02-28 21:57:42 [xTeVe] Client User-Agent : Lavf/58.65.101

  


  2022-02-28 21:57:43 [xTeVe] Streaming Status : Playlist : Schweiz Neu - Tuner : 1 / 1

  


  2022-02-28 21:57:43 [xTeVe] FFMPEG path : /usr/bin/ffmpeg

  


  2022-02-28 21:57:43 [xTeVe] Streaming URL : http://tunestream.me:8080/XXXXXXX
/XXXXXXXX/9165

  


  2022-02-28 21:57:43 [xTeVe] FFMPEG : Processing data

  


  2022-02-28 21:57:43 [xTeVe] Streaming Status : Receive data from FFMPEG

  


  2022-02-28 21:57:43 [xTeVe] FFMPEG log : [mpegts @ 0x55e0732fe560] dimensions
not set

  


  2022-02-28 21:57:43 [xTeVe] [ERROR] FFMPEG error (Streaming was stopped by third party
transcoder (FFmpeg / VLC)) - EC : 1204

  


  2022-02-28 21:57:43 [xTeVe] FFMPEG log : Could not write header for output file
#0 (incorrect codec parameters ?) : Invalid argument

  


  2022-02-28 21:57:43 [xTeVe] FFMPEG log :

  


  2022-02-28 21:57:43 [xTeVe] Streaming Status : Client has terminated the connection

  


  2022-02-28 21:57:43 [xTeVe] Streaming Status : Channel : VIP SW : SRF 1 FHD (Clients : 0)

  


  2022-02-28 21:57:44 [xTeVe] Streaming Status : Channel : VIP SW : SRF 1 FHD - No client
is using this channel anymore. Streaming Server connection has ended

  


  2022-02-28 21:57:44 [xTeVe] Streaming Status : Playlist : Schweiz Neu - Tuner : 0 / 1

  


  ---

  


  on xTeVe i set this settings :

  


  Buffer Size : 5 MB

  


  Timeout for new client connections : 500

  


  FFmpeg Options : -hide_banner -loglevel error -i [URL] -c copy -f mpegts pipe:1

  


  If i put the Streaming URL on VLC is working.
But not with FFmpeg, xTeVe and Plex.

  


  Thanks for your help.

  


  Greetings
Chris

  


• streaming an mkv file while processing with ffmpeg

  2 avril 2019, par Phani Rithvij

  What I want to do :

  • I want to play an mkv video in Firefox.
  • But Firefox doesn’t support the mkv format.

  So I’ve searched a lot and found that,
  I could stream instead of playing the video following these steps.

  • mkv can be converted to an m3u8 using FFmpeg
  • The m3u8 points towards ts segment files
  • then use hls.js on the browser side to play the video

  But the catch is I want to do this programmatically.

  What I actually want to do :

  Steps

  • the client uploads a huge (>1 GB) mkv file to the client’s server (server is the client’s machine itself)

  • after the upload is done,

  • client requests to play the video.

  • the server starts transcoding and sends the client the m3u8 stream immediately instead of making the client wait for the transcoding to complete.

  • the client should be able to seek the video. (the most IMPORTANT part)

  It is possible as Emby and Plex both have implemented it.

  I was able to get this to work in chrome as it supports playing some mkv files.
  I wrote a node js server that accepts Range header and pseudo-streams the video.

  A gist

  But as I’ve mentioned Firefox says NO to mkv.

  So I tried the hls thing but I couldn’t quite get the command to generate the stream and also play on the fly.

  I started this on a command line

  ffmpeg -i ../stream.mkv -hls_list_size 0 -acodec copy -vcodec copy file.m3u8

  and a simple http-server on another shell instance

  My index.html file

      
  
      
  
          <code class="echappe-js">&lt;script src='http://stackoverflow.com/feeds/tag/hls.js'&gt;&lt;/script&gt;
          
  
  &lt;script&gt;<br />
  
              var video = document.getElementById('video');<br />
  
              if(Hls.isSupported()) {<br />
  
                  var hls = new Hls();<br />
  
                  hls.loadSource('file.m3u8');<br />
  
                  hls.attachMedia(video);<br />
  
                  hls.on(Hls.Events.MANIFEST_PARSED,function() {<br />
  
                      video.play();<br />
  
                  });<br />
  
              } else if (video.canPlayType('application/vnd.apple.mpegurl')) {<br />
  
                  video.src = 'file.m3u8';<br />
  
                  video.addEventListener('loadedmetadata',function() {<br />
  
                      video.play();<br />
  
                  });<br />
  
              }<br />
  
          &lt;/script&gt;
  

  and while it was running I went ahead and requested the server.

  I was able to get the video but It only seeks as far it’s converted into the ts files.

  And it’s random and the video length keeps increasing. It won’t play sometimes and after the FFmpeg is done converting to m3u8 the video plays if I refresh the webpage.

  I think this has to do with the continuous overwriting of the m3u8 file.
  is there a way to predetermine the file contents of m3u8 and fill it up ?

  I want to be able to seek even further and somehow spawn another FFmpeg process
  to start from that timestamp of the video ? How could I approach the seeking part ?

  So what I’d like to do again is

  • I want to request the server to play a video file
  • It spawns a child process FFmpeg that does the transcoding
  • Sends the client the stream
  • The client should be able to seek to the end and it should play the thing.

• Demuxing RTSP stream using GStreamer and sending to FFmpeg using MPEG-TS via TCP

  25 janvier 2023, par Brian Schrameck

  I am trying to run a pipeline that will read an h264/aac stream from RTSP and push it to an FFmpeg TCP socket (that FFmpeg instance will re-publish as another RTSP stream, I know it's odd). Requirements :

  


  

  • The RTSP client at the start of this pipeline MUST be GStreamer (i.e. I can't use FFmpeg to read the RTSP stream, only to publish).

  


  • The RTSP client at the end of the pipeline MUST be FFmpeg.

  


  


  My pipeline works for video but adding audio has been a challenge. Here's the current GStreamer pipeline :

  


  gst-launch-1.0 rtspsrc location=rtsp://{camIp}/live name=cam \
  cam. ! rtph264depay ! h264parse ! queue ! mux. \
  cam. ! rtpmp4gdepay ! aacparse ! queue ! mux. \
  mpegtsmux name=mux ! tcpclientsink host=${ip} port=${port} sync=false


  


  Then, FFmpeg is listening like this :

  


  ffmpeg -f mpegts -i tcp://${ip}:${port} \
  -rtsp_transport tcp -vcodec copy -an -f rtsp rtsp://${rtspIp}:${rtspPort}/d8a5bb19e63326d7


  


  This pipeline works if I remove the audio by removing cam. ! rtpmp4gdepay ! aacparse ! queue ! mux. in my GStreamer chain. However, with audio added FFmpeg won't publish the data to my RTSP client ; FFmpeg starts probing the GStreamer output and then exits for no apparent reason.