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• FFmpeg merge two files from different sources with different bitrates

  2 mars 2023, par FamousWolluf

  I have created an intro video with Adobe Premiere Pro and I want to add the intro video to my downloaded Mixer stream videos before uploading them to YouTube, without reencoding the complete video through Adobe Premiere because that takes a long time because the videos are more then an hour.

  



  I tried it with FFmpeg ffmpeg -safe 0 -f concat -i list.txt -c copy output.mp4.
The intro is normal but the video part from Mixer is running way to fast.
  
I think it has something todo with the bitrate but I can't figure it out.

  



  Output of intro file :

  



    Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42mp41
    creation_time   : 2020-05-12T11:12:11.000000Z
  Duration: 00:00:06.06, start: 0.000000, bitrate: 2408 kb/s
    Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 2086 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
    Metadata:
      creation_time   : 2020-05-12T11:12:11.000000Z
      handler_name    : ?Mainconcept Video Media Handler
      encoder         : AVC Coding
    Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 317 kb/s (default)
    Metadata:
      creation_time   : 2020-05-12T11:12:11.000000Z
      handler_name    : #Mainconcept MP4 Sound Media Handler


  



  Output of Mixer file :

  



    Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf58.12.100
  Duration: 01:02:00.58, start: 0.000000, bitrate: 1937 kb/s
    Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720, 1800 kb/s, 29.41 fps, 29.97 tbr, 16k tbn, 32k tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
    Metadata:
      handler_name    : SoundHandler


  



  Output of the merged file :

  



    Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf58.43.100
  Duration: 00:57:03.84, start: 0.000000, bitrate: 2109 kb/s
    Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 3377 kb/s, 55.14 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
    Metadata:
      handler_name    : ?Mainconcept Video Media Handler
    Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 140 kb/s (default)
    Metadata:
      handler_name    : #Mainconcept MP4 Sound Media Handler


  



  If I need to reencode something my preference would be the intro because it's only 6 seconds.

  


• 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.