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• How to chose a stream from all stream in ffmpeg

  27 février 2016, par combo_ci

  I try to convert a UDP stream (that genrated from DVB signal) to HLS m3u8 file with this code :

  ffmpeg -i udp://239.1.2.1:60001 -acodec aac -strict -2 -vcodec libx264 -hls_wrap 100 -f hls /var/www/html/ts/1.m3u8 

  UDP stream contain 1 channel (in this case IRIB-TV1).

  When i run this code ffmpeg detect all of service and channel that streamed from DVB card with this message :

  Input #0, mpegts, from 'udp://239.1.2.1:60001':
  
  Duration: N/A, start: 77906.812644, bitrate: N/A
  
  Program 101 
  
  Metadata:
  
    service_name    : IRIB-TV1
  
    service_provider: IRIB
  
  Stream #0:0[0x3f2]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(tv, bt470bg), 720x576 [SAR 12:11 DAR 15:11], 25 fps, 50 tbr, 90k tbn, 50 tbc
  
  Stream #0:1[0x3f3](per): Audio: aac_latm ([17][0][0][0] / 0x0011), 48000 Hz, stereo, fltp
  
  Program 102 
  
  Metadata:
  
    service_name    : IRIB-TV2
  
    service_provider: IRIB
  
  Program 103 
  
  Metadata:
  
    service_name    : IRIB-TV3
  
    service_provider: IRIB
  
  Program 104 
  
  Metadata:
  
    service_name    : IRIB-TV4
  
    service_provider: IRIB

  As you see ffmpeg finf 4 channel in UDP stream, But VLC play only channel 1(IRIB-TV1).

  Now i have have 2 question :

  1-Can I get all channel and service via this ffmpeg code ?

  2-Can i choose a spesial stream from this ffmpeg code ?(i know that ffmpeg can choose a stream with -map otion but i want to choose other service_name that in output log)

• Reduce RTMP buffer while decoding using ffmpeg

  15 janvier 2019, par Alberto Padilla

  I’m setting up a new server using nginx and nginx-rtmp-module. I installed a DeckLink Duo2 card to output SDI signals comming from the RTMP stream.

  Everything works really great using this simple nginx.conf configuration :

  rtmp {
  
      server {
  
              listen 1935;
  
              chunk_size 4096;
  
  
  
              wait_video on;
  
              wait_key on;
  
  
  
              sync 10ms;
  
  
  
              application live {
  
                  live on;
  
                  record off;
  
  
  
                  exec_push /home/myuser/bin/ffmpeg -i rtmp://127.0.0.1/live/mystreamname -f decklink -format_code pal -pix_fmt uyvy422 'DeckLink Duo (1)';
  
              }  
  
     }
  
  }

  My goal is to achieve the absolute minimum delay between the received stream and the SDI output.

  I can achieve 3 seconds "waving hand" delay using the configuration that I mentioned. However, using ffplay I observe 1.5s delay using the -fflags nobuffer option.

  As ffplay cannot output to a decklink device, I want ffmpeg to reduce the buffer to the minimum, but I can’t find the way to do it. All the -rtmp_buffer or -fflag nobuffer or any other options have no impact on the delay.

  The stream is being received in the same machine ffmpeg is running, that is why it makes sense to reduce the ffmpeg buffering to the minimum if I want to achieve low delay.

  Any tips ?

• Can you think of a reason why windows might not enable audio if noone is logged in ?

  3 juillet 2017, par Caius Jard

  I’m having a bizarre problem with some virtual servers created to record podcasts. They run on amazon AWS as windows server 2012 instances and a small c# app tells FFMPEG to do the heavy lifting of capturing from the virtual screen and reading from the virtual sound card (Virtual Audio Cable : https://en.wikipedia.org/wiki/Virtual_Audio_Cable) via DirectShow filters

  The problem I have is if I leave the machine to do its stuff unattended, the recordings are sometimes silent. If I log in via VNC and watch it doing its stuff the audio is recorded just fine. All other aspects of the test op are the same, and the virtual machine is shut down between successive recordings so each one should theoretically be a clean slate. The app runs under a logged in session (hence the use of VNC rather than RDP)

  I’m now wondering if there is some optimisation of the windows sound engine whereby it doesn’t bother playing audio if it thinks noone is listening. The confusing thing to me is that not every virtual machine suffers these problems ; some of them record fine (and they’re all created from the same seed virtual hard disk image) in unattended mode

  I’m asking this question with the aim of getting together a list of things I can check/look into/debug.. I don’t have much knowledge of how MME/DirectSound/WASAPI work internally...