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  1. I have searched and found many similar questions, but not this one. I've experimented and can't come up with an answer.

    I have a Standalone "Digital TV Converter Box". This receives broadcast TV, and can act as a DVR. Mine is a "viewtv", but it's available from dozens of different vendors. It records combined video and audio (and optionally closed captions / subtitles) into a BDAV format file. I want to watch the program on other platforms, and sometimes edit the file and separate programs, and the BDAV format is inconvenient (to say the least). ffmpeg can easily perform the first part of converting the file into something more useable:

    ffmpeg -hide_banner -err_detect ignore_err -loglevel verbose -stats -i "channel09202022-0858.mts" -map 0 -c copy "channel-09202022-0858.m2ts"

    Here is where my frustration starts. There is usually a delay between the video and audio file, and it's different every time. MediaInfo reports something this:

    Audio ID : 4352 (0x1100)
    Menu ID : 1 (0x1)
    Format : AC-3
    Format/Info : Audio Coding 3
    Commercial name : Dolby Digital
    Codec ID : 129
    Duration : 3 h 7 min
    Bit rate mode :
    Constant Bit rate : 192 kb/s
    Channel(s) : 2 channels
    Channel layout : L R
    Sampling rate : 48.0 kHz
    Frame rate : 31.250 FPS (1536 SPF)
    Compression mode : Lossy
    Delay relative to video : -1 s 501 ms
    Stream size : 258 MiB (9%)
    Language : English
    Service kind : Complete Main


    This delay can confuse some editing programs. If I cut segments out with ffmpeg the audio and video don't quite start in the same place. I would really like to have ffmpeg re-synchronize the files, but have not been able to do it. I've tried -vsync with various options (1, 2, vfr) -async with various options including -async 1 which the documentation seems to indicate will synchronize the beginning, -fflags +genpts+igndts in various combinations, -af aresample=async=1 , in every combination I can think of. In every single case, ffmpeg reads and writes with no reported errors: but the new file still has a "Delay relative to video" carried through to the new file. I've spend quite a lot of time on this, and all I have to show for it is severe aggravation.

    The only thing I've found so far that comes close is to convert the file once, get the value of the delay, and then do this:

    ffmpeg -err_detect ignore_err -hide_banner -loglevel info -stats -async 1 -itsoffset -1.501 -i "Story-09202022-0858.m2ts" ^ -strict normal -map 0 -c:v copy -c:s mov_text -c:a aac -b:a 192k ^ -ar 48000 Story-09202022-0858.mkv

    This usually (but not always) re-encodes the audio in a way that does not show any significant delay between video and audio. However, it requires manually entering the values every time, and it must be exactly as I've shown it. Since -async is going to be "deprecated" (for no good reason I can see), I tried the so-called replacement of -af aresample=async=1 in this example and it did not work.

    Can someone please tell me if there is a way to get ffmpeg to look at the information in a file that says what the delay is and then "fix" it? It seems very odd to me that there isn't a way to fix this synchronization between video and audio when the value of the off is is already known and available within the file.

    If not, is there at least a way to obtain the value of the offset so I can write a command procedure (I normally use Windows for this, and have everything in BAT files) that will automate the procedure?

    As I've said, I've searched through many forums, and all of the examples I've found have to do with audio and video being different tempos, or different durations, or similar problems, none of which apply here. I've never seen an example of fixing the initial offset other than with -itsoffset.

    If it matters, the ultimate output will be AVC (MP4) video and AAC audio. Unfortunately, just copying the audio in AC3 isn't going to work as there are some systems (such as DLNA servers and Roku Media Player) that can't handle it properly.

    Thanks.
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    Originally Posted by BartZLederman View Post
    I have searched and found many similar questions, but not this one. I've experimented and can't come up with an answer.

    I have a Standalone "Digital TV Converter Box". This receives broadcast TV, and can act as a DVR. Mine is a "viewtv", but it's available from dozens of different vendors. It records combined video and audio (and optionally closed captions / subtitles) into a BDAV format file. I want to watch the program on other platforms, and sometimes edit the file and separate programs, and the BDAV format is inconvenient (to say the least). ffmpeg can easily perform the first part of converting the file into something more useable:

    ffmpeg -hide_banner -err_detect ignore_err -loglevel verbose -stats -i "channel09202022-0858.mts" -map 0 -c copy "channel-09202022-0858.m2ts"

    Here is where my frustration starts. There is usually a delay between the video and audio file, and it's different every time. MediaInfo reports something this:

    Audio ID : 4352 (0x1100)
    Menu ID : 1 (0x1)
    Format : AC-3
    Format/Info : Audio Coding 3
    Commercial name : Dolby Digital
    Codec ID : 129
    Duration : 3 h 7 min
    Bit rate mode :
    Constant Bit rate : 192 kb/s
    Channel(s) : 2 channels
    Channel layout : L R
    Sampling rate : 48.0 kHz
    Frame rate : 31.250 FPS (1536 SPF)
    Compression mode : Lossy
    Delay relative to video : -1 s 501 ms
    Stream size : 258 MiB (9%)
    Language : English
    Service kind : Complete Main


    This delay can confuse some editing programs. If I cut segments out with ffmpeg the audio and video don't quite start in the same place. I would really like to have ffmpeg re-synchronize the files, but have not been able to do it. I've tried -vsync with various options (1, 2, vfr) -async with various options including -async 1 which the documentation seems to indicate will synchronize the beginning, -fflags +genpts+igndts in various combinations, -af aresample=async=1 , in every combination I can think of. In every single case, ffmpeg reads and writes with no reported errors: but the new file still has a "Delay relative to video" carried through to the new file. I've spend quite a lot of time on this, and all I have to show for it is severe aggravation.

    The only thing I've found so far that comes close is to convert the file once, get the value of the delay, and then do this:

    ffmpeg -err_detect ignore_err -hide_banner -loglevel info -stats -async 1 -itsoffset -1.501 -i "Story-09202022-0858.m2ts" ^ -strict normal -map 0 -c:v copy -c mov_text -c:a aac -b:a 192k ^ -ar 48000 Story-09202022-0858.mkv

    This usually (but not always) re-encodes the audio in a way that does not show any significant delay between video and audio. However, it requires manually entering the values every time, and it must be exactly as I've shown it. Since -async is going to be "deprecated" (for no good reason I can see), I tried the so-called replacement of -af aresample=async=1 in this example and it did not work.

    Can someone please tell me if there is a way to get ffmpeg to look at the information in a file that says what the delay is and then "fix" it? It seems very odd to me that there isn't a way to fix this synchronization between video and audio when the value of the off is is already known and available within the file.

    If not, is there at least a way to obtain the value of the offset so I can write a command procedure (I normally use Windows for this, and have everything in BAT files) that will automate the procedure?

    As I've said, I've searched through many forums, and all of the examples I've found have to do with audio and video being different tempos, or different durations, or similar problems, none of which apply here. I've never seen an example of fixing the initial offset other than with -itsoffset.

    If it matters, the ultimate output will be AVC (MP4) video and AAC audio. Unfortunately, just copying the audio in AC3 isn't going to work as there are some systems (such as DLNA servers and Roku Media Player) that can't handle it properly.

    Thanks.
    try with $ ffmpeg -i file.mkv -itsoffset 3 -i file.mkv -c:a copy -c:v copy -map 0:v:0 -map 1:a:0 out.mkv
    more info here - https://www.geekyhacker.com/2020/05/17/synchronize-audio-and-video-with-ffmpeg/
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  3. Thank you for your response. However, it's basically what I've already done.

    That article gives an example of a known 3 second delay between video and audio, which is what I've done in the example I showed that (mostly) worked.

    I'm hoping for one that will pick up the delay from the file instead of my having to manually supply one.

    Bart.
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  4. Member
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    Originally Posted by BartZLederman View Post
    Thank you for your response. However, it's basically what I've already done.

    That article gives an example of a known 3 second delay between video and audio, which is what I've done in the example I showed that (mostly) worked.

    I'm hoping for one that will pick up the delay from the file instead of my having to manually supply one.

    Bart.
    i did find this "delay relative to video" - https://www.reddit.com/r/ffmpeg/comments/q9iz69/delay_relative_to_video_what_is_it/
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  5. To get the delay:

    ffprobe -v 0 -show_entries stream=start_time -select_streams a:0 -of compact=p=0:nk=1 "video.mkv"
    Last edited by Luke M; 20th Sep 2022 at 19:59.
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  6. For your example in post #1
    Encode audio to aac:
    Code:
    ffmpeg -itsoffset -1.501 -i "Story-09202022-0858.m2ts" -map 0 -c:v copy -c:s mov_text -c:a aac -b:a 192k -ar 48000 Story-09202022-0858.mkv
    keep ac3 audio:
    Code:
    ffmpeg -itsoffset -1.501 -i "Story-09202022-0858.m2ts" -map 0 -c:v copy -c:s mov_text -c:a copy Story-09202022-0858.mkv
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  7. Either I didn't explain properly, or you missed the point.

    I know how to get the two streams to synchronize by manually entering a value for itsoffset. I posted an example that (usually) works.

    What I'm looking for is a way to synchronize the two streams WITHOUT using itsoffset. I want a command line that will make ffmpeg synchronize the two streams automatically by picking up the information that is already in the input file.

    As a cross check, could someone tell me if there is something wrong with my explanation that is leading people to give me a "solution" that is identical to what I've already said I have done?
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  8. Originally Posted by BartZLederman View Post
    As a cross check, could someone tell me if there is something wrong with my explanation that is leading people to give me a "solution" that is identical to what I've already said I have done?
    It's not identical, but forget my post.
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  9. Member
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    Originally Posted by BartZLederman View Post
    Either I didn't explain properly, or you missed the point.

    I know how to get the two streams to synchronize by manually entering a value for itsoffset. I posted an example that (usually) works.

    What I'm looking for is a way to synchronize the two streams WITHOUT using itsoffset. I want a command line that will make ffmpeg synchronize the two streams automatically by picking up the information that is already in the input file.

    As a cross check, could someone tell me if there is something wrong with my explanation that is leading people to give me a "solution" that is identical to what I've already said I have done?
    see if this will work - https://www.anycodings.com/1questions/2312573/ffmpeg-automatically-synchronize-audio-and-video
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