Recherche avancée

Sur d’autres sites (10675)

• FFmpeg trim filter fails to use sexagisimal time specification and the output stream is empty. Is it a bug and is there a fix ?

  29 novembre 2020, par Link-akro

  With ffmpeg the trim filter (or its audio variant atrim) malfunctions whenever i try to write a sexagesimal time specification for the boundaries or duration parameters of the filter : the result is always an empty stream.

  


  According to the documentation i already linked, which was generated on November 26, 2020 so far, the time duration specification should be supported.

  


  


  start, end, and duration are expressed as time duration specifications

  


  


  Quote of the spec.

  


  


  [-][HH :]MM:SS[.m...]

  


  


  


  [-]S+[.m...][s|ms|us]

  


  


  I am working in a Windows_10-64bit and scripting in its CMD.exe command-line, should it matter.

  


  Here is an instance of trimming a video stream out of a media file with hardcoded value for simplicity. The audio is retained unless i use atrim as well.
We may or may not append the setpts filter as recommended in the documentation henceforth setpts=PTS-STARTPTS if we want to shift the stream to start at the beginning of the cut range.

  


      ffmpeg -i sample-counter.mp4 -vf "trim=start=start='1:2':end='1:5'" sample-counter-trimmed.mp4


  


  If i use decimal it works as intended.

  


      ffmpeg -i sample-counter.mp4 -vf "trim=start=start='2':end='5'" sample-counter-trimmed.mp4


  


  This is the banner of my ffmpeg build. We may see it is the latest build by gyan.dev at the moment i post this.

  


      ffmpeg version 4.3.1-2020-11-19-full_build-www.gyan.dev Copyright (c) 2000-2020 the FFmpeg developers
      built with gcc 10.2.0 (Rev5, 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-lzma --enable-libsnappy --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libdav1d --enable-libzvbi --enable-librav1e --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --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-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


  


  The incorrect stream is always reported as 0kB weight in the output of ffmpeg. I confirmed with ffprobe Movie_Countdown-trim-2-5.mov -show_streams -show_entries format=duration that the incorrect stream is duration zero.

  


  Should i report it as a bug ? Is there some correction or workaround ?

  


  I would rather a solution with the trim filter itself, but if not possible a CMD batch scripting.
We should not need a different filter like select or seeking option like there are tutorials and questions/answers everywhere already. Scripting would be straight-forward in a proper *nix shell+distrib and taught everywhere so it is not worth caring while CMD answers on the other hand are rare so a scripting workaround for CMD would have some microcosmic worth since making one is considerably less straight-forward than a shell with modern mathematics and parsing abilities built-in or packaged.

  


• How to embed an MP4 inside a PDF ?

  19 août 2020, par malat

  I am a happy user of img2pdf. This tool does the minimal amount of work to put a series of JPEG 2000/JPEG/PNG images into a PDF "enveloppe". However I am now faced with a new challenge : embed a MP4 file into a PDF "enveloppe".

  


  I see that commercial tool can do it, as seen at :

  


  

  • Add audio, video, and interactive objects to PDFs

  


  


  Here is one such sample PDF file (no Flash required on windows in this sample) :

  


  

  • https://gitlab.com/agrahn/media9/-/issues/9#note_345903962
** https://gitlab.com/agrahn/media9/uploads/90fddd777e0ec514c39c924cd8d3b688/video_test.pdf

  


  


  It seems to have been introduced in ISO 32000-1 (PDF 1.7 Extension Level 5)

  


  I am looking for a solution which will use the Rich Media annotation inside the PDF stream.

  


  There are dozen of duplicated questions on superuser/stackoverflow, which all pretty much refer to imagemagick/convert command line tool. But in my case, convert expand the images into a multi-page PDF (which is not my desired behavior) :

  


  $ convert input.mp4 output.pdf
$ pdfinfo output.pdf 
Title:          out
Producer:       https://imagemagick.org
CreationDate:   Wed Aug 19 15:38:01 2020 CEST
ModDate:        Wed Aug 19 15:38:01 2020 CEST
Tagged:         no
UserProperties: no
Suspects:       no
Form:           none
JavaScript:     no
Pages:          1601
Encrypted:      no
Page size:      352 x 288 pts
Page rot:       0
File size:      534407296 bytes
Optimized:      no
PDF version:    1.3


  


  with :

  


  $ convert --version
Version: ImageMagick 6.9.10-23 Q16 x86_64 20190101 https://imagemagick.org
Copyright: © 1999-2019 ImageMagick Studio LLC
License: https://imagemagick.org/script/license.php
Features: Cipher DPC Modules OpenMP 
Delegates (built-in): bzlib djvu fftw fontconfig freetype jbig jng jpeg lcms lqr ltdl lzma openexr pangocairo png tiff webp wmf x xml zlib


  


  and

  


  $ file input.mp4 
input.mp4: ISO Media, MP4 Base Media v1 [IS0 14496-12:2003]
$ ffprobe -v quiet -print_format json  -show_streams input.mp4 | grep codec_long_name
            "codec_long_name": "H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10",


  


• Using ffmpeg to convert the audio of an RTSP stream to PCM alaw and hosting an RTSP stream as the output

  24 octobre 2020, par Sirisian

  I have an RTSP stream on my network that is H264 for video and aac for audio. I need the audio converted to PCM alaw with the video just passed through.

  


  I tried :

  


  ffmpeg -rtsp_transport udp -i rtsp://10.0.0.10/main -vcodec copy -acodec pcm_alaw -rtsp_flags listen -f rtsp rtsp://localhost:8001/main


  


  The output I get is :

  


  ffmpeg version 4.3.1-2020-10-01-essentials_build-www.gyan.dev Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 10.2.0 (Rev3, 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-lzma --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-libass --enable-libfreetype --enable-libfribidi --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-libgme --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-librubberband
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
[rtsp @ 0000025ebdf1db00] Interleaved RTP mode is not supported yet.
Input #0, rtsp, from 'rtsp://10.0.0.10/main':
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 180k tbc
    Stream #0:1: Audio: aac (LC), 48000 Hz, stereo, fltp
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (aac (native) -> pcm_alaw (native))
Press [q] to stop, [?] for help


  


  It's running, but it's not listening on port 8001. I've tried both UDP and TCP, so I think I am misunderstanding what "-rtsp_flags listen" does for ffmpeg ? My goal is to start an "RTSP server" locally and allow say VLC or any other player to connect and play it. Is this something ffmpeg can do or do I need to send the stream to something else to allow VLC or other media players to connect to it locally ? (The hardware I'm running isn't terribly fast, so I'd like to do this with the least amount of operations possible).

  


  For reference if I run :

  


  ffplay -rtsp_flags listen rtsp://localhost:8001/main


  


  Then run :

  


  ffmpeg -rtsp_transport udp -i rtsp://10.0.0.10/main -vcodec copy -acodec pcm_alaw -f rtsp rtsp://localhost:8001/main


  


  It plays fine in the ffplay application. So it's like the "-rtsp_flags listen" only works with the input portion of ffplay and ffmpeg.