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• ffmpeg : xstack doesn't work when inputs are scaled to certain dimensions

  4 juin 2020, par dfriend21

  I'm using ffmpeg to create a mosaic of videos using the xstack filter. The input videos may come in varying dimensions, so I'm using the scale filter to scale them beforehand, and I'm using the force_original_aspect_ratio option and then the pad filter to keep the original aspect ratios of each video and add black bars to the sides to make each video have the correct dimensions.

  



  I have a command that's working - however, it's inconsistent. For some dimensions it works, while for others it doesn't.

  



  I'm using the fluent-ffmpeg Node.js module to call ffmpeg from Node.js. To do this, I'm passing an array of strings to the complexFilter() function.

  



  The following strings for the complex filter works :

  



  "[0:v]scale=400:250:force_original_aspect_ratio=decrease,pad=400:250:(ow-iw)/2:(oh-ih)/2 [s0]"
"[1:v]scale=400:250:force_original_aspect_ratio=decrease,pad=400:250:(ow-iw)/2:(oh-ih)/2 [s1]"
"[s0][s1]xstack=inputs=2:fill='black':layout=0_0|w0_0[v]"
"[0:a][1:a]amix=inputs=2[a]"


  



  However, if I change the output dimensions of each video to be 400:225 instead of 400:250 it fails.

  



  "[0:v]scale=400:225:force_original_aspect_ratio=decrease,pad=400:225:(ow-iw)/2:(oh-ih)/2 [s0]"
"[1:v]scale=400:225:force_original_aspect_ratio=decrease,pad=400:225:(ow-iw)/2:(oh-ih)/2 [s1]"
"[s0][s1]xstack=inputs=2:fill='black':layout=0_0|w0_0[v]"
"[0:a][1:a]amix=inputs=2[a]"


  



  The following error is given :

  



  An error occurred: ffmpeg exited with code 1: Error reinitializing filters!
Failed to inject frame into filter network: Invalid argument
Error while processing the decoded data for stream #1:1
Conversion failed!


  



  If it's relevant, the first video has dimensions of 1280x720 while the second video has dimensions of 320x240.

  



  Anyone know why one set of dimensions works while the other doesn't ?

  



  EDIT : Here is the full ffmpeg log for when it fails :

  



  ffmpeg version git-2020-05-13-b12b053 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.3.1 (GCC) 20200513
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  libavutil      56. 45.100 / 56. 45.100
  libavcodec     58. 84.100 / 58. 84.100
  libavformat    58. 43.100 / 58. 43.100
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 80.100 /  7. 80.100
  libswscale      5.  6.101 /  5.  6.101
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:/Users/user_name/Desktop/vids/vid1.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp41isom
    creation_time   : 2020-05-21T15:52:20.000000Z
  Duration: 00:00:10.76, start: 0.000000, bitrate: 8385 kb/s
    Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuvj420p(pc), 1280x720 [SAR 1:1 DAR 16:9], 8237 kb/s, 29.99 fps, 30 tbr, 30k tbn, 60 tbc (default)
    Metadata:
      creation_time   : 2020-05-21T15:52:20.000000Z
      handler_name    : VideoHandler
      encoder         : AVC Coding
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 165 kb/s (default)
    Metadata:
      creation_time   : 2020-05-21T15:52:20.000000Z
      handler_name    : SoundHandler
Input #1, mov,mp4,m4a,3gp,3g2,mj2, from 'C:/Users/user_name/Desktop/vids/vid2.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp41isom
    creation_time   : 2020-05-21T15:54:37.000000Z
  Duration: 00:00:11.01, start: 0.000000, bitrate: 836 kb/s
    Stream #1:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuvj420p(pc), 320x240 [SAR 1:1 DAR 4:3], 669 kb/s, 29.88 fps, 30 tbr, 30k tbn, 60 tbc (default)
    Metadata:
      creation_time   : 2020-05-21T15:54:37.000000Z
      handler_name    : VideoHandler
      encoder         : AVC Coding
    Stream #1:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 163 kb/s (default)
    Metadata:
      creation_time   : 2020-05-21T15:54:37.000000Z
      handler_name    : SoundHandler
Stream mapping:
  Stream #0:0 (h264) -> scale
  Stream #0:1 (aac) -> amix:input0
  Stream #1:0 (h264) -> scale
  Stream #1:1 (aac) -> amix:input1
  xstack -> Stream #0:0 (libx264)
  amix -> Stream #0:1 (aac)
Press [q] to stop, [?] for help
[swscaler @ 000001343fefc200] deprecated pixel format used, make sure you did set range correctly
[Parsed_pad_1 @ 000001343f8dc3c0] Padded dimensions cannot be smaller than input dimensions.
[Parsed_pad_1 @ 000001343f8dc3c0] Failed to configure input pad on Parsed_pad_1
Error reinitializing filters!
Failed to inject frame into filter network: Invalid argument
Error while processing the decoded data for stream #1:1
Conversion failed!

Done in 0.66s.


  


• Trim off N bytes from audio file using SoX / FFmpeg etc, on Windows ?

  17 novembre 2020, par Rinaldo Jonathan

  My team accidentally on purpose clicked NO when Audacity asked to save the recording. So I left with bunch of *.au files, after using recovery program.

  


  Some of them did have header and still open-able with audacity itself (example : this one), and some other are just complete nonsense, sometimes having the header filled with text from random javascript or HTML code (like this one). Probably hard disk half overwritten with browser cache ? I don't know. And at this point, I almost don't care.

  


  The audacity is on default settings, with sample rate 44100Hz. I can open a-113.au using audacity, from standard open files. I also was able to achieve open files using "Open RAW files" from Audacity, using this settings :

  


  

  


  so I assume header takes 12384 bytes.

  


  Now, how do I trim 12384 bytes from the file when opened as RAW, with either SoX or ffmpeg ? because if I open it as RAW with 0 offset (default settings), it will add the header as a noise.

  


  Current ffmpeg command I used : ffmpeg.exe  -f f32le -ar 44.1k -ac 1 -i  source destination
  
Current sox command I used : sox -t raw --endian little --rate 44100 -b 1 -b 32 --encoding floating-point  %%A "converted/%%~nxA.wav"
  
Both still have header as a noise in the converted files.

  


• 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