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• Converting a PCM file from Discord.js call to MP3 or WAV using ffmpeg in Node.js causes file deletion without saving it. How can I solve this issue ?

  28 mai 2023, par ItsChriss

  I am trying to convert a pcm file, which I get from a discord call into a mp3 or wav file.
I saw a example with ffmpeg :

  


  const ffmpeg = require('ffmpeg');

try {
  var process = new ffmpeg('path/to/pcm/file');
  process.then(function (audio) {
    audio.fnExtractSoundToMP3('path/to/new/file.mp3', function (error, file) {
      if (!error) console.log('Audio File: ' + file);
    });
  }, function (err) {
    console.log('Error: ' + err);      
  });
} catch (e) {
  console.log(e);
}


  


  But it's not saving or creating the file, it just deletes it without an error message. how can I fixx this ?

  


  I tried multiple methodes with other ffmpeg modules like "fluent-ffmpeg" but it didn't work eiter.

  


  This is the code I am using to get the PCM data :

  


  const reciever = connection.receiver.subscribe(message.author.id,
{             
  mode: "pcm",
  end: {                 
    behavior: EndBehaviorType.AfterSilence,                 
    duration: 1000             
  }         
})


  


• FFMPEG convert rtp stream to rtmp - bind failed address already in use

  27 août 2020, par Leo

  I setup a server with Janus gateway and using videoroom plugin I'm trying to forward locally the rtp stream using port 5002 for audio and 5004 for video.
This is the videoroom plugin configuration

  



      room-1234: {
        description = "Demo Room"
        secret = "adminpwd"
        publishers = 6
        bitrate = 128000
        fir_freq = 1
        #fir_freq = 10
        audiocodec = "opus"
        videocodec = "vp8"
        #videocodec = "h264"
        record = false
        #rec_dir = "/path/to/recordings-folder"
}


  



  After the RTP forward I would like to convert the video to rtmp to get the video remotely using OBS Studio and I set up an nginx server with rtmp plugin. Using ffmpeg I'm trying to make this conversion and I created the sdp file with this content :

  



  v=0
o=- 0 0 IN IP4 127.0.0.1
s=RTP Video
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 58.29.100
m=audio 5002 RTP/AVP 111
a=rtpmap:111 OPUS/48000/2
m=video 5004 RTP/AVP 100
a=rtpmap:100 VP8/90000
a=fmtp:100


  



  And then I launched the command

  



  ffmpeg -protocol_whitelist rtp,udp,file -loglevel trace -analyzeduration 300M -probesize 300M -i config.sdp -c:v copy -c:a aac -ar 16k -ac 1 -preset ultrafast -tune zerolatency rtmp://127.0.0.1/live/1234


  



  But I got back the error bind failed address already in use. Below the complete output

  



  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. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
Splitting the commandline.
Reading option '-protocol_whitelist' ... matched as AVOption 'protocol_whitelist' with argument 'rtp,udp,file'.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'trace'.
Reading option '-analyzeduration' ... matched as AVOption 'analyzeduration' with argument '300M'.
Reading option '-probesize' ... matched as AVOption 'probesize' with argument '300M'.
Reading option '-i' ... matched as input url with argument 'config.sdp'.
Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'copy'.
Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'aac'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '16k'.
Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '1'.
Reading option '-preset' ... matched as AVOption 'preset' with argument 'ultrafast'.
Reading option '-tune' ... matched as AVOption 'tune' with argument 'zerolatency'.
Reading option 'rtmp://127.0.0.1/live/1234' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument trace.
Successfully parsed a group of options.
Parsing a group of options: input url config.sdp.
Successfully parsed a group of options.
Opening an input file: config.sdp.
[NULL @ 0x5594280] Opening 'config.sdp' for reading
Probing sdp score:50 size:205
[sdp @ 0x5594280] Format sdp probed with size=2048 and score=50
[sdp @ 0x5594280] sdp: v='0'
[sdp @ 0x5594280] sdp: o='- 0 0 IN IP4 127.0.0.1'
[sdp @ 0x5594280] sdp: s='RTP Video'
[sdp @ 0x5594280] sdp: c='IN IP4 127.0.0.1'
[sdp @ 0x5594280] sdp: t='0 0'
[sdp @ 0x5594280] sdp: a='tool:libavformat 58.29.100'
[sdp @ 0x5594280] sdp: m='audio 5002 RTP/AVP 111'
[sdp @ 0x5594280] sdp: a='rtpmap:111 OPUS/48000/2'
[sdp @ 0x5594280] audio codec set to: opus
[sdp @ 0x5594280] audio samplerate set to: 48000
[sdp @ 0x5594280] audio channels set to: 2
[sdp @ 0x5594280] sdp: m='video 5004 RTP/AVP 100'
[sdp @ 0x5594280] sdp: a='rtpmap:100 VP8/90000'
[sdp @ 0x5594280] video codec set to: vp8
[sdp @ 0x5594280] sdp: a='fmtp:100'
[udp @ 0x5597980] bind failed: Address already in use
[AVIOContext @ 0x559d580] Statistics: 205 bytes read, 0 seeks
config.sdp: Invalid data found when processing input


  



  I did a lot of searches and tries but I'm really not able to figure out what's wrong. Could you please help me to understand the error ?

  



  Thank you !

  


• rtpproto : Support more than one SSM include address, support excludes

  26 juillet 2013, par Ed Torbett

  rtpproto : Support more than one SSM include address, support excludes
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DBH] libavformat/rtpproto.c