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• FFmpeg joined audio & video out of sync in some parts of the video

  20 septembre 2022, par ankur625

  I've developed a video joining service. Here is how it works in brief.

  


  Step 1 : Users upload their video (different video file types) from desktop/mobile. These videos are processed on the server, a logo is added, and converted to .mp4 format.

  


  Step 2 : All received mp4 files are then joined, a background music is added and saved.

  


  Everything is working as expected. The only problem is, when video files are joined. The final output video has audio out of sync in some parts. In step 2, I get "Non-monotonous DTS in output stream 0:1 ;" error message.

  


  Following are step 1 and step 2 codes.

  


  Step 1 :

  


  ffmpeg -y -i inputVideo -i inputImage -r 30000/1001 -video_track_timescale 30k -filter_complex "[0]scale=854:480:force_original_aspect_ratio=decrease,pad=854:480:(ow-iw)/2:(oh-ih)/2:black,setsar=1/1,setdar=16/9[v0];[1:v]format=argb,colorchannelmixer=aa=1.0,scale=71:72[overl1],[v0][overl1]overlay=297:285[v1]" -map "[v1]" -map 0:a -b:v 250k -maxrate 250k -bufsize 500k -crf 5 -preset slower -vbr 4 -movflags +faststart -b:a 96k -async 1 outputVideo.mp4


  


  Step 2 :

  


  ffmpeg -safe 0 -f concat -segment_time_metadata 1 -i videoList.txt -stream_loop -1 -i backgroundMusic.wav -filter_complex '[0:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,volume=1.5[a0];[1:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,volume=0.03[a1];[a0][a1]amerge=inputs=2[a]' -map 0:v -map '[a]' -c:v copy -ac 2 finalVideo.mp4


  


  I'm not expert with ffmpeg, and not sure, what's missing. can anyone help me fix this issue ?

  


  Step 2 (using amix suggested by kesh)

  


  ffmpeg -safe 0 -f concat -segment_time_metadata 1 -i list.txt -stream_loop -1 -i track1.mp3 -filter_complex "[0:a]volume=1.5[a0];[1:a]volume=0.03[a1];[a0][a1]amix=inputs=2:duration=first[a]" -map 0:v -map "[a]" -c:v copy -ac 2 out.mp4


  


  And, following is the full log of step 2,

  


  ffmpeg version 5.0.1-full_build-www.gyan.dev Copyright (c) 2000-2022 the FFmpeg developers
      built with gcc 11.2.0 (Rev7, Built by MSYS2 project)
      configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-libsnappy --enable-zlib --enable-librist --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libdav1d --enable-libdavs2 --enable-libuavs3d --enable-libzvbi --enable-librav1e --enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-liblensfun --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libshaderc --enable-vulkan --enable-libplacebo --enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libilbc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --enable-libsoxr --enable-chromaprint
      libavutil      57. 17.100 / 57. 17.100
      libavcodec     59. 18.100 / 59. 18.100
      libavformat    59. 16.100 / 59. 16.100
      libavdevice    59.  4.100 / 59.  4.100
      libavfilter     8. 24.100 /  8. 24.100
      libswscale      6.  4.100 /  6.  4.100
      libswresample   4.  3.100 /  4.  3.100
      libpostproc    56.  3.100 / 56.  3.100
    [mov,mp4,m4a,3gp,3g2,mj2 @ 000001ff7163ed40] Auto-inserting h264_mp4toannexb bitstream filter
    Input #0, concat, from 'list.txt':
      Duration: N/A, start: -0.023220, bitrate: 63 kb/s
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(progressive), 854x480, 61 kb/s, 29.97 fps, 29.97 tbr, 30k tbn
        Metadata:
          handler_name    : VideoHandler
          vendor_id       : [0][0][0][0]
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 1 kb/s
        Metadata:
          handler_name    : SoundHandler
          vendor_id       : [0][0][0][0]
    Input #1, mp3, from 'track1.mp3':
      Metadata:
        title           : Safety Net
        artist          : Riot
        album           : YouTube Audio Library
        genre           : Country & Folk
        encoder         : Google
      Duration: 00:01:30.88, start: 0.025057, bitrate: 128 kb/s
      Stream #1:0: Audio: mp3, 44100 Hz, stereo, fltp, 128 kb/s
        Metadata:
          encoder         : Lavf
    Stream mapping:
      Stream #0:1 (aac) -> volume:default
      Stream #1:0 (mp3float) -> volume:default
      Stream #0:0 -> #0:0 (copy)
      amix:default -> Stream #0:1 (aac)
    Press [q] to stop, [?] for help
    Output #0, mp4, to 'out.mp4':
      Metadata:
        encoder         : Lavf59.16.100
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(progressive), 854x480, q=2-31, 61 kb/s, 29.97 fps, 29.97 tbr, 30k tbn
        Metadata:
          handler_name    : VideoHandler
          vendor_id       : [0][0][0][0]
      Stream #0:1: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s
        Metadata:
          encoder         : Lavc59.18.100 aac
    frame=    3 fps=0.0 q=-1.0 size=       0kB time=00:00:00.02 bitrate=  16.5kbits/s speed=44.7x    
    [mov,mp4,m4a,3gp,3g2,mj2 @ 000001ff719acac0] Auto-inserting h264_mp4toannexb bitstream filter
    frame=  478 fps=0.0 q=-1.0 size=     256kB time=00:00:15.89 bitrate= 132.0kbits/s speed=31.7x    
    frame= 1084 fps=1083 q=-1.0 size=    1024kB time=00:00:36.11 bitrate= 232.3kbits/s speed=36.1x    
    frame= 1668 fps=1111 q=-1.0 size=    1536kB time=00:00:55.59 bitrate= 226.3kbits/s speed=  37x    
    [mov,mp4,m4a,3gp,3g2,mj2 @ 000001ff719acac0] Auto-inserting h264_mp4toannexb bitstream filter
    frame= 2282 fps=1140 q=-1.0 size=    2560kB time=00:01:16.08 bitrate= 275.6kbits/s speed=  38x    
    [mov,mp4,m4a,3gp,3g2,mj2 @ 000001ff71ce81c0] Auto-inserting h264_mp4toannexb bitstream filter
    frame= 2848 fps=1138 q=-1.0 size=    3328kB time=00:01:34.97 bitrate= 287.1kbits/s speed=37.9x    
    frame= 3443 fps=1146 q=-1.0 size=    4352kB time=00:01:54.82 bitrate= 310.5kbits/s speed=38.2x    
    [mov,mp4,m4a,3gp,3g2,mj2 @ 000001ff718f2c40] Auto-inserting h264_mp4toannexb bitstream filter
    [mov,mp4,m4a,3gp,3g2,mj2 @ 000001ff718f26c0] Auto-inserting h264_mp4toannexb bitstream filter
    [aac @ 000001ff716ed2c0] Queue input is backward in time
    [mp4 @ 000001ff71661600] Non-monotonous DTS in output stream 0:1; previous: 6224608, current: 6221946; changing to 6224609. This may result in incorrect timestamps in the output file.
    [mp4 @ 000001ff71661600] Non-monotonous DTS in output stream 0:1; previous: 6224609, current: 6222970; changing to 6224610. This may result in incorrect timestamps in the output file.
    [mp4 @ 000001ff71661600] Non-monotonous DTS in output stream 0:1; previous: 6224610, current: 6223994; changing to 6224611. This may result in incorrect timestamps in the output file.
    frame= 4336 fps=1238 q=-1.0 size=    5376kB time=00:02:24.65 bitrate= 304.4kbits/s speed=41.3x    
    frame= 4406 fps=1230 q=-1.0 Lsize=    5783kB time=00:02:27.07 bitrate= 322.1kbits/s speed=41.1x    
    video:3486kB audio:2149kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.641165%
    [aac @ 000001ff716ed2c0] Qavg: 853.082


  


• How to cut multiple parts of audio with ffmpeg ?

  27 septembre 2022, par Groovie

  How can i cut a multiple parts of single audio and save it with ffmpeg ?
I have tried -ss -to command, but instead of cutting selected part its saves it, how can i reach the result when selected part will be removed ? I want to do that for a multiple parts of audio.

  


• ffmpeg mp4 to hls but video parts size are not balanced, some are big and small

  24 décembre 2022, par kendoodoo

  here's the problem :

  


  

  


  my command :

  


  ffmpeg -i output.mp4 -start_number 0 -hls_time 10 -hls_list_size 0 -f hls 1.m3u8