Recherche avancée

Sur d’autres sites (5289)

• FFMPEG Re-muxing .MKV to MP4\MV4 while retaining Dolby Digital Plus JOC [closed]

  31 décembre 2020, par David F

  My ask is, does anyone have any knowledge of how to remux and keep Enhanced AC-3 with Joint Object Control with FFMPEG.

  


  I would be grateful if anyone could help me out here. I'm remuxing a number of video files from .MKV to .MP4\M4V so they play on the Apple platform (including the TV App). It's important NOT to convert, but to remux.

  


  I have achieved the ability to remux 4K HDR with Dolby Digital Plus in a .MKV to 4K HDR with Dolby Digtial 5.1. MP4. The issue is retaining the Dolby Digital Plus. This contains the ATMOS information for object based sound.

  


  The media players on Apple reports Dolby Digital Plus for the .MKV and Dolby Digital 5.1 for .MP4. I know Dolby Labs to allow free remuxing of Dolby Digital Plus/ Atmos with sample code and information they have uploaded onto Github for developers.

  


  I'm using FFMPEG with the following command :-

  


  ffmpeg -i original.mkv -c:v copy -c:a eac3 -tag:v hvc1 output.mp4

  


  Resulting output from MediaInfo (note .MKV contains only 1 video and 1 audio track)

  


  Audio -----Original.mkv

  


  

  • ID : 2

  


  • Format : E-AC-3 JOC

  


  • Format/Info : Enhanced AC-3 with Joint
Object Coding

  


  • Commercial name : Dolby Digital Plus with Dolby
Atmos

  


  • Codec ID : **A_**EAC3

  


  • Duration : 51 min 51 s

  


  • Bit rate mode : Constant

  


  


  Audio ----- Remuxed.mp4

  


  

  • ID : 2

  


  • Format : E-AC-3

  


  • Format/Info : Enhanced AC-3

  


  • Commercial name : Dolby Digital Plus

  


  • Codec ID : ec-3

  


  • Duration : 51 min 51 s

  


  • Bit rate mode : Constant

  


  


  The remux operation finished in about 1 minute for a 8Gb 4k video file, so I know there is no reencoding.

  


  I can see that .mp4 can take E-AC-3 Joint Object Control (JOC) with the demo files from Dolby labs.

  


  I've also remuxed an MKV to MKV and able to see the same result. The JOC markings have been removed with FFMPEG

  


  Thanks in advanced.

  


  David

  


• Encode iOS compatible h264 audio stream with FFMPEG

  11 octobre 2014, par beamformer

  I am using FFMPEG with Python to encode a low bit rate version of videos that I import. I would like the output to be playable on several devices, including Roku and iOS. This link states that Apple supports H264 High profile, level 4.1 and MP3 audio.

  ffmpeg -preset veryslow -y -profile:v high -level 4.0 -movflags +faststart -codec:a libmp3lame -qscale:a 2 -s 1280x720 out.mp4

  The resulting videos play fine in an HTML5 browser on a computer. However, on the iPhone the video appears but the audio stream does not play. Is there an issue with the mp3 settings ?

  From ffprobe :

  [STREAM]
  
  index=1
  
  codec_name=mp3
  
  codec_long_name=MP3 (MPEG audio layer 3)
  
  profile=unknown
  
  codec_type=audio
  
  codec_time_base=1/44100
  
  codec_tag_string=mp4a
  
  codec_tag=0x6134706d
  
  sample_fmt=s16p
  
  sample_rate=44100
  
  channels=1
  
  channel_layout=mono
  
  bits_per_sample=0
  
  id=N/A
  
  r_frame_rate=0/0
  
  avg_frame_rate=0/0
  
  time_base=1/44100
  
  start_pts=-1105
  
  start_time=-0.025057
  
  duration_ts=2321489
  
  duration=52.641474
  
  bit_rate=94949
  
  max_bit_rate=N/A
  
  bits_per_raw_sample=N/A
  
  nb_frames=2016
  
  nb_read_frames=N/A
  
  nb_read_packets=N/A
  
  DISPOSITION:default=1
  
  DISPOSITION:dub=0
  
  DISPOSITION:original=0
  
  DISPOSITION:comment=0
  
  DISPOSITION:lyrics=0
  
  DISPOSITION:karaoke=0
  
  DISPOSITION:forced=0
  
  DISPOSITION:hearing_impaired=0
  
  DISPOSITION:visual_impaired=0
  
  DISPOSITION:clean_effects=0
  
  DISPOSITION:attached_pic=0
  
  TAG:language=und
  
  TAG:handler_name=SoundHandler
  
  [/STREAM]

  I tried using AAC, but did not get good results with the native FFMPEG encoder

• FFmpeg hangs when using -vsync 1 for libvpx

  31 mars 2014, par user3466413

  I'm trying to transcode a .mov, ultimately generated by GoToMeeting, to a .webm. I've found to get the audio and video in sync for other formats that I need -vsync 1 and encoding at a constant framerate using -r:v 30. This causes FFmpeg to take the input .mov file, which is variable framerate, and transform it to constant framerate at 30 frames per second. In doing so it duplicates frames, but appears to just make many reference frames so the file size doesn't get massive.

  This approach works perfectly fine when encoding using libx264 and aac to a .mp4 container. This makes a file playable in all browsers that support MPEG-4. But, when I encode using using libvpx and libtheora to a .webm container for other browsers, particularly Chrome, FFmpeg just hangs or runs ridiculously slowly. It doesn't respond to pressing Q to abort it, either ; I have to kill the process.

  So, I get output that looks like this :

  user@virtualbox:~/Desktop$ ffmpeg -y -i original.mov -vcodec libvpx -acodec libvorbis -r:v 30 -vsync 1 omgwork.webm
  
  ffmpeg version N-61874-g57e939d Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Mar 27 2014 11:00:31 with gcc 4.8 (Ubuntu/Linaro 4.8.1-10ubuntu9)
  
    configuration: --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-x11grab
  
    libavutil      52. 69.100 / 52. 69.100
  
    libavcodec     55. 55.100 / 55. 55.100
  
    libavformat    55. 35.101 / 55. 35.101
  
    libavdevice    55. 11.100 / 55. 11.100
  
    libavfilter     4.  3.100 /  4.  3.100
  
    libswscale      2.  5.102 /  2.  5.102
  
    libswresample   0. 18.100 /  0. 18.100
  
    libpostproc    52.  3.100 / 52.  3.100
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x254b940] decoding for stream 0 failed
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'original.mov':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 537199360
  
      compatible_brands: qt  
  
      creation_time   : 2014-03-20 02:09:22
  
    Duration: 03:08:26.72, start: 0.000000, bitrate: 75 kb/s
  
      Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv), 1368x728 [SAR 1:1 DAR 171:91], 20 kb/s, 0.60 fps, 300 tbr, 600 tbn, 1200 tbc (default)
  
      Metadata:
  
        creation_time   : 2014-03-20 02:09:22
  
        handler_name    : Apple Alias Data Handler
  
      Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 16000 Hz, mono, fltp, 47 kb/s (default)
  
      Metadata:
  
        creation_time   : 2014-03-20 02:09:22
  
        handler_name    : Apple Alias Data Handler
  
      Stream #0:2(eng): Data: none (rtp  / 0x20707472), 6 kb/s
  
      Metadata:
  
        creation_time   : 2014-03-20 02:09:22
  
        handler_name    : Apple Alias Data Handler
  
      Stream #0:3(und): Data: none (rtp  / 0x20707472), 0 kb/s
  
      Metadata:
  
        creation_time   : 2014-03-20 02:09:22
  
        handler_name    : Apple Alias Data Handler
  
  [libvpx @ 0x255e460] v1.2.0
  
  Output #0, webm, to 'omgwork.webm':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 537199360
  
      compatible_brands: qt  
  
      encoder         : Lavf55.35.101
  
      Stream #0:0(und): Video: vp8 (libvpx), yuv420p, 1368x728 [SAR 1:1 DAR 171:91], q=-1--1, 200 kb/s, 1k tbn, 30 tbc (default)
  
      Metadata:
  
        creation_time   : 2014-03-20 02:09:22
  
        handler_name    : Apple Alias Data Handler
  
      Stream #0:1(eng): Audio: vorbis (libvorbis), 16000 Hz, mono, fltp (default)
  
      Metadata:
  
        creation_time   : 2014-03-20 02:09:22
  
        handler_name    : Apple Alias Data Handler
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (h264 -> libvpx)
  
    Stream #0:1 -> #0:1 (aac -> libvorbis)
  
  Press [q] to stop, [?] for help
  
  frame=  117 fps= 10 q=0.0 size=       4kB time=00:00:03.90 bitrate=   8.0kbits/s dup=111 drop=0

  Note how it says it's three minutes into transcoding...and it stays there for minutes on end. This video takes only about 10 minutes to completely transcode to MPEG-4 and there is no delay whatsoever from the Press [q] to stop, [?] for help message and to the first progress message. For WebM, the progress message only appears after a few minutes, and then just crawls along so slowly that it would take literally days to complete.

  When this is happening, FFmpeg maxes out all the CPU cores I give it (for example -threads 4 uses all four CPU cores). Aside from making my CPU burst into flames, it doesn't speed things up at all.

  I've tried :

  • Using Windows and Linux. Both do the same thing.
  • Using the latest available Windows binary and compiling FFmpeg from the Git repository. No improvement or change.
  • Using -vsync 0 and -vsync 2. The file does transcode when I do this, but the audio and video remain out of sync just as it did with other formats I tried. -vsync 1 seems absolutely essential for this to work,
  • Searching, and the closest I could find was bug #3208. But, I'm not specifying -fflags at all ; you can see the exact command I'm running.

  Ideas ?