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• FFMPEG merge one audio and one video file [duplicate]

  29 mars 2020, par Karl R

  I am trying to get this to work for a few hours and am having no luck. I have two .mp4 files - one is audio and one is video. All I want to do is merge them into one file so the audio and video play together. I know I am close but I keep getting stuck on "Cannot find a matching stream for unlabeled input pad 1 on filter Parsed_hstack_0".

  My command I am using is :

  .\ffmpeg.exe -y -i .\camera.mp4 -i .\screen.mp4 -filter_complex hstack output.mp4

  Output of each file type is attached.

  Thank you for your help - we found a bug in a screen recording app on our AMD laptops and this will helps us with a temporary fix. I work at a tech college and we are scrambling to get online videos going for the students to learn remotely.
  pic1
  pic2

• Video streaming through HAProxy

  21 janvier 2015, par n00bie

  I want to stream video from my webcam to many clients (all clients use html 5 video player).

  Now i have this :

  Server :

  sudo gst-launch-0.10 tcpserversrc port = 1234 ! oggparse ! tcpserversink port = 1235

  Sender :

  ffmpeg -f video4linux2 -s 320x240 -i /dev/mycam -f alsa -i hw:1 -codec:v libtheora -qscale:v 5 -codec:a libvorbis -qscale:a 5 -f ogg http://localhost:1234

  Receiver :

  <video width="320" height="240" autoplay="autoplay">
  
    <source src="http://localhost:1235" type="video/ogg">
  
    Your browser does not support the video tag.
  
  </source></video>

  It works.

  Now i want to increase count of web cameras. Therefore, i need to increase count of gstreamer’s. But, i want to use only port 80 to communucate between server and clients, therefore i try to use HAProxy.

  HAProxy config : (only one web camera)

  global
  
          maxconn 4096
  
          user workshop-staff
  
          group workshop-staff
  
          daemon
  
          log 127.0.0.1 local0 debug
  
  
  
  defaults
  
          log     global
  
          mode    http
  
          option  httplog
  
          option  dontlognull
  
          retries 3
  
          option redispatch
  
          option http-server-close
  
          option forwardfor
  
          maxconn 2000
  
          timeout connect 5s
  
          timeout client  15min
  
          timeout server  15min
  
          option http-no-delay
  
  
  
  frontend public
  
          bind *:80
  
          use_backend stream_input if { path_beg /stream_input }        
  
          use_backend stream_output if { path_beg /stream_output }
  
  
  
  backend stream_input
  
          server stream_input1 localhost:1234
  
  
  
  backend stream_output
  
          server stream_output1 localhost:1235

  Server :

  sudo gst-launch-0.10 tcpserversrc port = 1234 ! oggparse ! tcpserversink port = 1235

  Sender :

  ffmpeg -f video4linux2 -s 320x240 -i /dev/mycam -f alsa -i hw:1 -codec:v libtheora -qscale:v 5 -codec:a libvorbis -qscale:a 5 -f ogg http://localhost/stream_input

  Receiver :

  <video width="320" height="240" autoplay="autoplay">
  
    <source src="http://localhost/stream_output" type="video/ogg">
  
    Your browser does not support the video tag.
  
  </source></video>

  But, in this case, HTML5 video player shows nothing.

  If i change receiver to : (i.e. use localhost:1235 instead of localhost/stream_output)

  <video width="320" height="240" autoplay="autoplay">
  
    <source src="http://localhost:1235" type="video/ogg">
  
    Your browser does not support the video tag.
  
  </source></video>

  It works. Could someone help me ?

• FFmpeg exact bitrate

  17 avril 2018, par WhiteNinja

  So, I’m experimenting with audio formats/params using FFmpeg. I’m using this code to downgrade audio bitrate and sample rate (original is 128kbs, 44100hz respectively) :

  ffmpeg -i test.m4a -ab 16k -ar 16k output.m4a

  After finishing, I’m checking results :

  ffprobe -show_streams -show_format output.m4a

  I omit the whole output for brevity, lets look for only interesting part :

  sample_rate=16000
  
  bit_rate=16539
  
  max_bit_rate=16539

  So, sample rate is perfect (its always exact for me, I was checking with different settings). But bitrate is not what I was asking for, although looks similar.

  The question is how can I get exact 16000 bitrate for my audio ?
  Thanks is advance !