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• ffmpeg record pulseaudio has some crackles

  8 janvier 2021, par boygiandi

  I'm trying to record audio from pulseaudio (on Centos 7), using ffmpeg. But the audio is not smooth, it has some crackles, noise after few seconds
Please check this video link : https://www.facebook.com/watch/live/?v=692754594726722&ref=watch_permalink

  


  The ffmpeg command looks like this

  


  ffmpeg -fflags +igndts -framerate 30 -s 1920x1130 -draw_mouse 0 -f x11grab -i :1085.0+nomouse -f pulse -i sinkgstvDViJbLFIaUBLdTxTeLtr.monitor -c:v libx264 -filter:v crop=1920:1080:0:50 -g 60 -r 30 -bufsize 2M -b:v 7M -vbsf h264_mp4toannexb -pix_fmt yuv420p -strict experimental -preset ultrafast -c:a aac -async 1 -vsync 1 -b:a 128k -bsf:a aac_adtstoasc -metadata comment=gstvDViJbLFIaUBLdTxTeLtr -f flv "rtmps://live-api-s.facebook.com:443/rtmp/4248189238527878?s_bl=1&s_psm=1&s_sc=4248189291861206&s_sw=0&s_vt=api-s&a=Abx_pMiM4_ccqBcS"


  


  I found this article but it didn't fix my problem https://forum.level1techs.com/t/improving-linux-audio-updated/134511 .
Please help

  


• ffmpeg cannot open connection tcp ://a.rtmp.youtube.com

  13 mars 2024, par Hiji Deui

  I want to live stream using ffmpeg, when live on Facebook it runs normally, but when I live on YouTube there is an error, is there anything wrong with the command I entered ? even though the command is the same as live on Facebook, but only the RTMP link has been changed

  


  


  


  ffmpeg -re -i out.mp4 -c:v copy -c:a aac -ar 44100 -ab 128k -ac 2 -strict -2 -flags +global_header -bsf:a aac_adtstoasc -bufsize 3000k -f flv "rtmp://a.rtmp.youtube.com/live2/my-key-streaming"

  


  


  



  and the output is

  


  


  


  ffmpeg version N-55112-g7eb9cf593e-static https://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 8 (Debian 8.3.0-6)
  configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg
  libavutil      56. 61.100 / 56. 61.100
  libavcodec     58.114.100 / 58.114.100
  libavformat    58. 64.100 / 58. 64.100
  libavdevice    58. 11.103 / 58. 11.103
  libavfilter     7. 91.100 /  7. 91.100
  libswscale      5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc    55.  8.100 / 55.  8.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'out.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: isommp42
    creation_time   : 2020-12-26T11:13:27.000000Z
    com.android.version: 10
  Duration: 00:00:03.27, start: 0.000000, bitrate: 21344 kb/s
    Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, unknown/bt470bg/unknown), 1920x1080, 20225 kb/s, SAR 1:1 DAR 16:9, 29.99 fps, 30.01 tbr, 90k tbn, 180k tbc (default)
    Metadata:
      rotate          : 90
      creation_time   : 2020-12-26T11:13:27.000000Z
      handler_name    : VideoHandle
    Side data:
      displaymatrix: rotation of -90.00 degrees
    Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, mono, fltp, 128 kb/s (default)
    Metadata:
      creation_time   : 2020-12-26T11:13:27.000000Z
      handler_name    : SoundHandle
[tcp @ 0x58bf880] Connection to tcp://a.rtmp.youtube.com:1935 failed: Connection timed out
[rtmp @ 0x5893140] Cannot open connection tcp://a.rtmp.youtube.com:1935
rtmp://a.rtmp.youtube.com/live2/my-key: Connection timed out

  


  


  



  how to fix this, btw i use vps, sorry, my english so bad and this is the first time i asked on this website

  


• Setting up RTP on Nginx

  2 février 2021, par Swap

  I'm trying to use Janus Media Server to relay WebRTC streams to a particular RTP host/port, from where ffmpeg can pick it up as an input and convert it further to an rtmp stream, which can then be used to broadcast to various social media platforms (such as, YouTube, Twitch, Facebook, etc.)

  


  My inspiration for this has been the following blog - https://www.meetecho.com/blog/firefox-webrtc-youtube-kinda/

  


  Specifically, I'm trying to replicate the following architecture -

  


  

  


  And Janus, as per their documentation, has a very neat API for doing it -

  


  {&#xA;    "request" : "rtp_forward",&#xA;    "room" : <unique numeric="numeric" of="of" the="the" room="room" publisher="publisher" is="is" in="in">,&#xA;    "publisher_id" : <unique numeric="numeric" of="of" the="the" publisher="publisher" to="to" relay="relay" externally="externally">,&#xA;    "host" : "<host address="address" to="to" forward="forward" the="the" rtp="rtp" and="and" packets="packets">",&#xA;    "host_family" : "",&#xA;    "audio_port" : <port to="to" forward="forward" the="the" audio="audio" rtp="rtp" packets="packets">,&#xA;    "audio_ssrc" : <audio ssrc="ssrc" to="to" use="use" when="when" optional="optional">,&#xA;    "audio_pt" : <audio payload="payload" type="type" to="to" use="use" when="when" optional="optional">,&#xA;    "audio_rtcp_port" : <port to="to" contact="contact" receive="receive" audio="audio" rtcp="rtcp" feedback="feedback" from="from" the="the" and="and" currently="currently" unused="unused" for="for">,&#xA;    "video_port" : <port to="to" forward="forward" the="the" video="video" rtp="rtp" packets="packets">,&#xA;    "video_ssrc" : <video ssrc="ssrc" to="to" use="use" when="when" optional="optional">,&#xA;    "video_pt" : <video payload="payload" type="type" to="to" use="use" when="when" optional="optional">,&#xA;    "video_rtcp_port" : <port to="to" contact="contact" receive="receive" video="video" rtcp="rtcp" feedback="feedback" from="from" the="the" optional="optional">,&#xA;    "simulcast" : ,&#xA;    "video_port_2" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" port="port" to="to" forward="forward" the="the" video="video" rtp="rtp" packets="packets" from="from" second="second" substream="substream"></if>layer to>,&#xA;    "video_ssrc_2" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" video="video" ssrc="ssrc" to="to" use="use" the="the" second="second" substream="substream"></if>layer; optional>,&#xA;    "video_pt_2" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" video="video" payload="payload" type="type" to="to" use="use" the="the" second="second" substream="substream"></if>layer; optional>,&#xA;    "video_port_3" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" port="port" to="to" forward="forward" the="the" video="video" rtp="rtp" packets="packets" from="from" third="third" substream="substream"></if>layer to>,&#xA;    "video_ssrc_3" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" video="video" ssrc="ssrc" to="to" use="use" the="the" third="third" substream="substream"></if>layer; optional>,&#xA;    "video_pt_3" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" video="video" payload="payload" type="type" to="to" use="use" the="the" third="third" substream="substream"></if>layer; optional>,&#xA;    "data_port" : <port to="to" forward="forward" the="the" messages="messages">,&#xA;    "srtp_suite" : <length of="of" authentication="authentication" tag="tag" or="or" optional="optional">,&#xA;    "srtp_crypto" : "<key to="to" use="use" as="as" crypto="crypto" encoded="encoded" key="key" in="in" optional="optional">"&#xA;}&#xA;</key></length></port></port></video></video></port></port></audio></audio></port></host></unique></unique>

  


  For this, I've setup a Nginx server, where I've also installed Janus and everything's been running smoothly so far. But I'm quite clueless as to how to setup my Nginx server so that it accepts RTP connections (which will be forwarded as RTMP using ffmpeg).

  


  Please guide me to any relevant resources that would help me achieve this. Thanks in advance !