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• Converting a spesific audio sream and copy the rest

  15 mai 2022, par Arete

  Is it possible to convert one of the audio tracks in a video file to a different format while copying/remuxing the other audio tracks in the file with FFmpeg ?

  


  More specifically, is it possible to do this without explicitly specify all the audio tracks you want to keep ?

  


  I have tried using the -map function,

  


  ffmpeg -i INPUT -c:v copy -c:a libfdk_aac -map 1:a:2 OUTPUT.mkv


  


  but this only encodes the first audio track and excludes the rest. I know I can probably explicitly tell FFmpeg to mux all the rest of the audio tracks, but this is really tedious. It means I will have to count every audio track and write e considerably longer command.

  


  Is there not a way to tell FFmpeg "convert audio track X and copy the rest" ?

  


  Here is the MediaInfo of the file. Note that there are over 10 audio tracks but I have excluded them in this example to keep it simple.

  


  Video
ID                                       : 4113 (0x1011)
Menu ID                                  : 1 (0x1)
Format                                   : HEVC
Format/Info                              : High Efficiency Video Coding
Format profile                           : Main 10@L5.1@High
HDR format                               : SMPTE ST 2086, HDR10 compatible
Codec ID                                 : 36
Duration                                 : 2 h 30 min
Width                                    : 3 840 pixels
Height                                   : 2 160 pixels
Display aspect ratio                     : 16:9
Frame rate                               : 23.976 (24000/1001) FPS
Color space                              : YUV
Chroma subsampling                       : 4:2:0 (Type 2)
Bit depth                                : 10 bits
Color range                              : Limited
Color primaries                          : BT.2020
Transfer characteristics                 : PQ
Matrix coefficients                      : BT.2020 non-constant
Mastering display color primaries        : Display P3
Mastering display luminance              : min: 0.0050 cd/m2, max: 4000 cd/m2
Maximum Content Light Level              : 349 cd/m2
Maximum Frame-Average Light Level        : 86 cd/m2

Audio #1
ID                                       : 4352 (0x1100)
Menu ID                                  : 1 (0x1)
Format                                   : DTS XLL
Format/Info                              : Digital Theater Systems
Commercial name                          : DTS-HD Master Audio
Muxing mode                              : Stream extension
Codec ID                                 : 134
Duration                                 : 2 h 30 min
Bit rate mode                            : Variable
Channel(s)                               : 6 channels
Channel layout                           : C L R Ls Rs LFE
Sampling rate                            : 48.0 kHz
Frame rate                               : 93.750 FPS (512 SPF)
Bit depth                                : 24 bits
Compression mode                         : Lossless

Audio #2
ID                                       : 4353 (0x1101)
Menu ID                                  : 1 (0x1)
Format                                   : AC-3
Format/Info                              : Audio Coding 3
Commercial name                          : Dolby Digital
Codec ID                                 : 129
Duration                                 : 2 h 30 min
Bit rate mode                            : Constant
Bit rate                                 : 640 kb/s
Channel(s)                               : 6 channels
Channel layout                           : L R C LFE Ls Rs
Sampling rate                            : 48.0 kHz
Frame rate                               : 31.250 FPS (1536 SPF)
Bit depth                                : 16 bits
Compression mode                         : Lossy
Delay relative to video                  : -9 ms
Stream size                              : 687 MiB (1%)
Service kind                             : Complete Main

Audio #3
ID                                       : 4354 (0x1102)
Menu ID                                  : 1 (0x1)
Format                                   : AC-3
Format/Info                              : Audio Coding 3
Commercial name                          : Dolby Digital
Codec ID                                 : 129
Duration                                 : 2 h 30 min
Bit rate mode                            : Constant
Bit rate                                 : 448 kb/s
Channel(s)                               : 6 channels
Channel layout                           : L R C LFE Ls Rs
Sampling rate                            : 48.0 kHz
Frame rate                               : 31.250 FPS (1536 SPF)
Bit depth                                : 16 bits
Compression mode                         : Lossy
Delay relative to video                  : -9 ms
Stream size                              : 481 MiB (1%)
Service kind                             : Complete Main


  


• ffmpeg recording audio / video sync issues [closed]

  1er novembre 2022, par sling jones

  I'm using ffmpeg to record NTSC analog video on Linux Fedora 36 using a Blackmagic Intensity Pro 4K for video and a Scarlett 2i2 for audio. I'm using a TBC to avoid dropped frames and to ensure a constant S-Video Y/C framerate on the analog end.

  


  The problem I'm running into is that on playback the audio will start out relatively in sync with the video at the beginning of the captured file but will eventually run ahead of the video eventually becoming many seconds off.

  


  Nothing I do seems to change this or change the degree to which it happens. The audio and video stay in sync throughout the entire video as I'm monitoring the source so I don't understand how they can diverge so much once encoded into a digital file ?

  


  Here is the command I am using :

  


  ffmpeg -hwaccel cuda -fflags +igndts -format_code ntsc -f decklink -raw_format auto -vsync passthrough -rtbufsize 1500M -thread_queue_size 512 -i 'Intensity Pro 4K' -f pulse -rtbufsize 500M -thread_queue_size 512 -i 'Scarlett 2i2 Camera Analog Stereo' -c copy -map 0:1 -map 1:0 "/tmp/ffmpeg-raw/file-raw.avi"


  


  here's the ffprobe output from one of my files :

  


  Input #0, avi, from 'test2-raw.avi':
  Metadata:
    software        : Lavf59.33.100
  Duration: 00:11:42.87, start: 0.000000, bitrate: 169341 kb/s
  Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x486, 167801 kb/s, 59.94 fps, 59.94 tbr, 59.94 tbn
  Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s



  


  As you can see in my code snippet, I've been throwing things at the wall for a bit. I've tried different rtbufsizes, adding -copyts, and going through the different -vsync options. I've tried it with and without hardware acceleration ( I do have a NVIDIA card), +igdts did get rid of a warning but did not help with the sync, as did changing the thread queue sizes.

  


  OBS can do this, why can't I ?

  


• Extracting a bit broken AC3 audio from a video file

  3 janvier 2023, par WYSIWYG

  I Googled for about 2 hours for this problem and I can't seem to find the answer anywhere.

  


  So I have a 1080p video file which has about 2 seconds within it that's broken (at 3 minutes and 53 seconds). I found a way to extract the non-broken parts of it, also I got a replacement for the broken frames from another version of the same video (in 720p), upscaled them to 1080p and replaced them in the 1080p file (reencoding the whole file in the process). Here comes the problematic part - at the same time (3 min 53 sec), the audio is also broken and it's in Dolby Digital 5.1 AC3 format. When I try to extract the audio track from the original 1080p file and place the result in the new fixed version, which contains the upscaled video frames, the audio obviously is cut at the problematic 2 seconds and it desyncs (starts to play earlier) for the rest of the file.

  


  What I want to do and I'm unable to understand how to do it is to extract the AC3 stream in such a way that the 2 broken seconds are filled with silence and the total length of the audio stay the same as the original, so it won't desync. Is this possible and if it is, what is the correct FFMPEG command for it ?

  


  Thank you in advance.

  


  I've tried :

  


  ffmpeg -i broken.mkv -c:a copy audio.ac3

  


  which presented an output AC3 file, which is missing the 2 seconds from 03:53 to 03:55

  


  so when I combine the resulting AC3 file with the repaired video, by using :

  


  ffmpeg -i fixed.mkv -i audio.ac3 -c copy fixed-with-audio.mkv

  


  it is desynced, so at video time 03:53 I hear the audio that should play at 03:55