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• vsync flag usage in ffmpeg while filtering

  13 octobre 2022, par antortjim

  I am trying to apply a threshold to an input video with ffmpeg, but I observe the following warning emitted for every processed frame

  


  [mp4 @ 0x56360181a200] Non-monotonous DTS in output stream 0:0; previous: 182272, current: 182272; changing to 182273. This may result in incorrect timestamps in the output file.


  


  where the previous and current are always 1 less than the value to which the DTS (Decoding Time Stamp) is changed

  


  I have noticed this warning is emitted only if I set -vsync passthrough (which I changed from the original -vsync 0 which is seen in many online examples).

  


  # input.mp4 has resolution 790x790
ffmpeg -vsync passthrough  -i input.mp4  -f lavfi -i color=808080:s=790x790 -f lavfi -i color=black:s=790x790 -f lavfi -i color=white:s=790x790 -filter_complex '[0:v][1:v][2:v][3:v]threshold' -an -c:v h264_nvenc threshold.mp4


  


  Shall I leave the vsync flag set to the default (auto or -1), or is -vsync passthrough essential to guarantee the frames are displayed in the right order ? In that case, how do I handle this warning ? Some other online examples I found of users experiencing this warning are different from mine, because in their case they are concatenating videos (see 1, 2

  


  From the documentation on the -vsync flag, at the end, I see :

  


  


  With -map you can select from which stream the timestamps should be taken. You can leave either video or audio unchanged and sync the remaining stream(s) to the unchanged one

  


  


  Maybe this warning should be handled with -map ? But I don't know how.

  


  Sidenote, I keep getting the deprecation warning asserting me to change -vsync for -fps_mode, however doing so breaks the command.

  


  FFPEG Version :

  


  commit 28ac2279adb860ea8b90d3073603912bf3eb6a83 from ffmpeg master branch

  


  ffmpeg version N-108625-g28ac2279ad Copyright (c) 2000-2022 the FFmpeg developers
built with gcc 9 (Ubuntu 9.4.0-1ubuntu1~20.04.1)
configuration: --enable-nonfree --enable-cuda-nvcc --enable-libnpp --enable-gpl --extra-cflags=-I/usr/local/cuda/include --extra-ldflags=-L/usr/local/cuda/lib64 --disable-static --enable-shared
libavutil      57. 39.101 / 57. 39.101
libavcodec     59. 50.100 / 59. 50.100
libavformat    59. 34.101 / 59. 34.101
libavdevice    59.  8.101 / 59.  8.101
libavfilter     8. 49.101 /  8. 49.101
libswscale      6.  8.112 /  6.  8.112
libswresample   4.  9.100 /  4.  9.100
libpostproc    56.  7.100 / 56.  7.100


  


  OS

  


  Ubuntu 20.04.4 LTS

  


• How exactly do I extract and merge a audio file using @ffmpeg/ffmpeg

  5 octobre 2022, par jeff.gogooma

  I have a lot of issues about audio transfer to server and frontend.

  


  I am building a solution that is extracting and merging a file received from frontend.

  


  I am using local ffmpeg file with node.js child_process package.

  


  My Env :

  


  framework : nest.js

  


  runtime : node.js

  


  lang : typescript

  


  here is console.log(file : Express.Multer.File).
I am storing to S3 bucket these files. So I am not using multer's memory, disk storage.

  


  // console.log(file);
 {
  fieldname: 'file',
  originalname: 'input1.mp3',
  encoding: '7bit',
  mimetype: 'audio/mpeg'
}


  


  import cp from "child_process";

type Section = { from: number, to: number };

const files: Array = [];

export const setupAudioFileSection = async (file: Express.Multer.File, section: Section) => {
  try {
    const file1 = await cp.exec(`ffmpeg -i ${file} -y -ss 0 -t ${section.from}`);
    const file2 = await cp.exec(`ffmpeg -i ${file} -y -ss ${section.from} -t ${section.to}`);
    const file3 = await cp.exec(`ffmpeg -i ${file} -y -ss ${section.to}`);
    files.push(file1);
    files.push(file3);

    return file2;
  } catch (error) {
    console.log(error);
    throw new Error('setupAudioFileSection error');
  }


  


  But, My Local ffmpeg is printing the error. because of [object Object]. I think local ffmpeg is not recognized the audio file received from client. So, I am resolving this issue by writing a file to disk or memory directly. but I am confusing this solution whether is right or not.

  


  error Error: Command failed: ffmpeg -i [object Object] -y -ss 10 -t 20 -f mp3 ./TestOutput
ffmpeg version 5.1.1 Copyright (c) 2000-2022 the FFmpeg developers
  built with Apple clang version 13.1.6 (clang-1316.0.21.2.5)
  configuration: --prefix=/opt/homebrew/Cellar/ffmpeg/5.1.1 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox --enable-neon
  libavutil      57. 28.100 / 57. 28.100
  libavcodec     59. 37.100 / 59. 37.100
  libavformat    59. 27.100 / 59. 27.100
  libavdevice    59.  7.100 / 59.  7.100
  libavfilter     8. 44.100 /  8. 44.100
  libswscale      6.  7.100 /  6.  7.100
  libswresample   4.  7.100 /  4.  7.100
  libpostproc    56.  6.100 / 56.  6.100

[object: No such file or directory



  


  How can I write file on my server and read ? if that can be, I can resolve [object Object] of ffmpeg issue maybe. Please some answer or advise my issue. Thank you !

  


• How to correctly use h264_amf on Windows ?

  27 octobre 2022, par Harshiv

  I'm trying to use hardware encoding with FFMPEG on Windows 10 with AMD Radeon r5 M330 DGPU on laptop : HP-AC026-TX. FFMPEG version is 5.1.2-full_build-www.gyan.dev from gyan.dev. I'm using this :

  


  fmpeg ^
    -hide_banner ^
    -v verbose ^
    -i "%1" ^
    -c:v h264_amf -acodec copy -y out.mp4


  


  but it fails with :

  


  [h264_amf @ 000001fb84d80b80] AMF initialisation succeeded via D3D11.
[h264_amf @ 000001fb847ab3c0] CreateComponent(AMFVideoEncoderVCE_AVC) failed with error 36
Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height


  


  The arguments seem as generic as I could hope, FFMPEG should use sensible defaults, so I can't Understand why it is failing.
Entire FFMPEG output is :

  


  [h264 @ 000001fb84706c40] Reinit context to 1920x1088, pix_fmt: yuv420p
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '1366_pngl_20220816_161558.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: isommp42
    creation_time   : 2022-08-16T10:51:42.000000Z
    com.android.version: 11
    com.android.capture.fps: 30.000000
  Duration: 00:03:41.33, start: 0.000000, bitrate: 7517 kb/s
  Stream #0:0[0x1](eng): Video: h264 (High), 1 reference frame (avc1 / 0x31637661), yuv420p(tv, bt709, progressive, left), 1920x1080 (1920x1088), 7259 kb/s, SAR 1:1 DAR 16:9, 29.96 fps, 29.99 tbr, 90k tbn (default)
    Metadata:
      creation_time   : 2022-08-16T10:51:42.000000Z
      handler_name    : VideoHandle
      vendor_id       : [0][0][0][0]
  Stream #0:1[0x2](eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 256 kb/s (default)
    Metadata:
      creation_time   : 2022-08-16T10:51:42.000000Z
      handler_name    : SoundHandle
      vendor_id       : [0][0][0][0]
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> h264 (h264_amf))
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
[h264 @ 000001fb84787400] Reinit context to 1920x1088, pix_fmt: yuv420p
[graph 0 input from stream 0:0 @ 000001fb84cfd780] w:1920 h:1080 pixfmt:yuv420p tb:1/90000 fr:90000/3001 sar:1/1
[h264_amf @ 000001fb84d80b80] AMF initialisation succeeded via D3D11.
[h264_amf @ 000001fb847ab3c0] CreateComponent(AMFVideoEncoderVCE_AVC) failed with error 36
Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
[AVIOContext @ 000001fb846f89c0] Statistics: 0 bytes written, 0 seeks, 0 writeouts
[AVIOContext @ 000001fb846f3f40] Statistics: 371902 bytes read, 3 seeks
Conversion failed!


  


  What should I do to make hardware encoding work ?