Recherche avancée

Sur d’autres sites (9209)

• How to set the destination folder of a Node.js fluent-ffmpeg screenshot to your AWS S3 bucket using getSignedUrl() ?

  10 juillet 2017, par Madhavi Mohoni

  I’m writing a program to generate .png thumbnails (with the same name, in the same folder) for a set of .mp4 videos in my Amazon S3 bucket. For this example, I’m going to create a /folder/file.png for a /folder/file.mp4 in the bucket. I’ve managed to set the source URL using the s3 object and getSignedUrl as follows :

  var srcurl = s3.getSignedUrl('getObject', {
  
          Bucket: 'bucket-name',
  
          Key: '/folder/file.mp4'
  
        });

  and

  new ffmpeg({ source: srcurl })
  
       .screenshots({
  
          count: 1, 
  
          filename: '%f'.substr(0, '%f'.indexOf('.')) + '.png', 
  
          /* To shorten the long string that's returned */ 
  
          folder: desturl,
  
          size: MAX_WIDTH + 'x' + MAX_HEIGHT
  
        });

  The destination URL has to be the same folder as the source. So I set it as follows :

  var desturl = s3.getSignedUrl('putObject', {
  
          Bucket: 'bucket-name',
  
          Key: '/folder/file' + '.png'
  
        });

  This combination doesn’t work - is there a way to do this correctly ?

• What is the best way to merge .mkv and .mka files using ffmpeg ?

  28 juin 2017, par Robert

  I’m using ffmpeg to merge .mkv and .mka files into .mp4 files. My current command looks like this :

  ffmpeg -i video.mkv -i audio.mka output_path.mp4

  The audio and video files are pre-signed urls from Amazon S3. Even on a server with sufficient resources, this process is going very slowly. I’ve researched situations where you can tell ffmpeg to skip re-encoding each frame, but I think that in my situation it actually does need to re-encode each frame.

  I’ve downloaded 2 sample files to my macbook pro and have installed ffmpeg locally via homebrew. When I run the command

  ffmpeg -i video.mkv -i audio.mka -c copy output.mp4

  I get the following output :

  ffmpeg version 3.3.2 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with Apple LLVM version 8.1.0 (clang-802.0.42)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/3.3.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-vda
  
    libavutil      55. 58.100 / 55. 58.100
  
    libavcodec     57. 89.100 / 57. 89.100
  
    libavformat    57. 71.100 / 57. 71.100
  
    libavdevice    57.  6.100 / 57.  6.100
  
    libavfilter     6. 82.100 /  6. 82.100
  
    libavresample   3.  5.  0 /  3.  5.  0
  
    libswscale      4.  6.100 /  4.  6.100
  
    libswresample   2.  7.100 /  2.  7.100
  
    libpostproc    54.  5.100 / 54.  5.100
  
  Input #0, matroska,webm, from '319_audio_1498590673766.mka':
  
    Metadata:
  
      encoder         : GStreamer matroskamux version 1.8.1.1
  
      creation_time   : 2017-06-27T19:10:58.000000Z
  
    Duration: 00:00:03.53, start: 2.831000, bitrate: 50 kb/s
  
      Stream #0:0(eng): Audio: opus, 48000 Hz, stereo, fltp (default)
  
      Metadata:
  
        title           : Audio
  
  Input #1, matroska,webm, from '319_video_1498590673766.mkv':
  
    Metadata:
  
      encoder         : GStreamer matroskamux version 1.8.1.1
  
      creation_time   : 2017-06-27T19:10:58.000000Z
  
    Duration: 00:00:03.97, start: 2.851000, bitrate: 224 kb/s
  
      Stream #1:0(eng): Video: vp8, yuv420p(progressive), 640x480, SAR 1:1 DAR 4:3, 30 tbr, 1k tbn, 1k tbc (default)
  
      Metadata:
  
        title           : Video
  
  [mp4 @ 0x7fa4f0806800] Could not find tag for codec vp8 in stream #0, codec not currently supported in container
  
  Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
  
  Stream mapping:
  
    Stream #1:0 -> #0:0 (copy)
  
    Stream #0:0 -> #0:1 (copy)
  
      Last message repeated 1 times

  So it appears that the specific encodings I’m working with are vp8 videos and opus audio files, which I believe are incompatible with the .mp4 output container. I would appreciate answers that cover ways of optimally merging vp8 and opus into .mp4 output or answers that point me in the direction of output media formats that are both compatible with vp8 & opus and are playable on web and mobile devices so that I can bypass the re-encoding step altogether.

  EDIT :

  Just wanted to provide a benchmark after following LordNeckbeard’s advice :

  4 min 41 second video transcoded locally on my mac
  
  
  
  LordNeckbeard’s approach : 15 mins 55 seconds (955 seconds)
  
  Current approach : 18 mins 49 seconds (1129 seconds)
  
  
  
  18% speed increase

• Can you think of a reason why windows might not enable audio if noone is logged in ?

  3 juillet 2017, par Caius Jard

  I’m having a bizarre problem with some virtual servers created to record podcasts. They run on amazon AWS as windows server 2012 instances and a small c# app tells FFMPEG to do the heavy lifting of capturing from the virtual screen and reading from the virtual sound card (Virtual Audio Cable : https://en.wikipedia.org/wiki/Virtual_Audio_Cable) via DirectShow filters

  The problem I have is if I leave the machine to do its stuff unattended, the recordings are sometimes silent. If I log in via VNC and watch it doing its stuff the audio is recorded just fine. All other aspects of the test op are the same, and the virtual machine is shut down between successive recordings so each one should theoretically be a clean slate. The app runs under a logged in session (hence the use of VNC rather than RDP)

  I’m now wondering if there is some optimisation of the windows sound engine whereby it doesn’t bother playing audio if it thinks noone is listening. The confusing thing to me is that not every virtual machine suffers these problems ; some of them record fine (and they’re all created from the same seed virtual hard disk image) in unattended mode

  I’m asking this question with the aim of getting together a list of things I can check/look into/debug.. I don’t have much knowledge of how MME/DirectSound/WASAPI work internally...