Is there any software recommend to handle TS in the following way?

[ihmcjacky]

To scan through the TS file and chop the dropout frames to produce a clean no bad frame TS file?

[poisondeathray]

None that works very well - you usually end up cutting out (around) the bad parts

projectx works ok for mpeg2 ts streams

otherwise , you can try tsdoctor or tspe

[ihmcjacky]

Thanks for your kind reply, actually my main problem is the video/audio out of sync problem

When the TS file contains dropouts, it always appears a situation that my encoded mp4 file is out of sync (TS with no dropouts works perfect)

And the weird thing is that the "out of sync" situation is not equal, e.g. in the 1st 10 minutes, it is ok, but at 10-20 minutes, audio runs faster than the scene by 500ms,

then 20-28 minutes, the audio even runs faster (1s out of sync), so I wonder if there is any tools to add the silence to the m4a audio in different parts so as to make it sync again in different parts.

I have tried tmpgenc, but it only allows time shifting as a whole, but not part of the audio (as the out of sync situation is not equally distributed to the whole file)

Or is there any method that can fix the aac audio extracted from the transport stream? (The aac audio plays noramally after extracting from the transport stream but I wonder why when I try to mux the cutted audio and the video file, they become out of sync )

[poisondeathray]

no, the dropouts 99% of the time require lots of work (cut up in segments, cut out bad sections, manually syncing each segment, then joining) . There is no 1 click software to fix it. It's usually easier to do in an editor NLE like premiere, vegas - where you can see the timeline for video & audio waveform to manually sync

(Some mpeg2 streams can be partially fixed by projectx or videoredo quickstream fix, but if it's a h.264 stream, you're out of luck)

Or is there any method that can fix the aac audio extracted from the transport stream? (The aac audio plays noramally after extracting from the transport stream but I wonder why when I try to mux the cutted audio and the video file, they become out of sync )

It's usually better to leave it in container (e.g. cut using avisynth script with video & audio in container, don't demux) - higher success rate

[ihmcjacky]

Thanks for the quick reply, I usually import my transport stream through megui and let megui to help me writh the script as follows:

LoadPlugin("C:\Program Files (x86)\megui\tools\dgindex\DGDecode.dll")
DGDecode_mpeg2source("J:\encode_temp\movie1.d2v", cpu=4)
Load_Stdcall_Plugin("C:\Program Files (x86)\megui\tools\yadif\yadif.dll")
Yadif(mode=1, order=1)
#crop
LanczosResize(1440,1080) # Lanczos (Sharp)
#denoise

__film = last
__t0 = __film.trim(1886, 80888)
__t1 = __film.trim(88082, 119317)
__t2 = __film.trim(124714, 148744)
__t3 = __film.trim(155936, 187523)
__t4 = __film.trim(192923, 240757)
__t5 = __film.trim(243456, 264557)
__t6 = __film.trim(268156, 269053)
__t7 = __film.trim(272652, 313170)
__t8 = __film.trim(320364, 380360)
__t9 = __film.trim(387554, 395107)
__t0 ++ __t1 ++ __t2 ++ __t3 ++ __t4 ++ __t5 ++ __t6 ++ __t7 ++ __t8 ++ __t9

encode the X264 mp4 and mux with the audio.

What you mean is to use avisynth to read the TS directly for encoding?

[poisondeathray]

If you are using dgindex (dgdecode.dll), this is a mpeg2 stream, not h.264 stream. Then videoredo or projectx or pvastrumento can often fix them

What you mean is to use avisynth to read the TS directly for encoding?

Include audio in the script using audiodub(), so you can use trim() so it cuts both video & audio at the same time. So you can use that script directly to encode audio.

If it's aac audio, you can use ffaudiosource() or bassaudiosource() to load it. Use delayaudio() to enter the delay value written in the name in seconds. There are full instructions that come with dgindex.

If it's in sync with that script (preview in a media player), then cuts should stay in sync. Otherwise use DirectShowSource() with the video in the container (instead of dgindex)

If the sync issues were from dropouts, you have fix them first, or cut them out, otherwise nothing will work

[ihmcjacky]

I have inputted the 6 channel audio track into premere and adjust the audio, now in the preview window, they are sync.
But when I try to output the adjusted audio, premere gives me error message by saying that I am outputting channels
larger than the original, but I only choose to ouput 5.1 channel, that is equal to the original one, then why is there still
an error?

[poisondeathray]

In your sequence settings, did you select 5.1 for master audio track ?

Why are your channels broken out ? Was the original 5.1 (1 track), or 6 mono tracks ?

If all you are doing is manual sync (not individual channel work), 5.1 can appear on 1 track and you can export it as 5.1 single track

eg.

[ihmcjacky]

I don't know whether it is exactly 5.1 channels or not, this is the properties of the audio file

[poisondeathray]

yes it's (5.1) is the same thing as 6 channel aac - but it should import as 1 track , not 6 mono

did you setup the sequence settings with 5.1 audio ?

not sure what's going on. If you export 6 mono wavs you can use eac3to or another program to encode

[ihmcjacky]

Now I can import as 5.1 aac file but still premier pro does not let me to export with the same reason as before

[poisondeathray]

What were your export settings?

You probably forgot to set the sequence settings for the audio master at 5.1 (it's 2.0 by default) - I got the same error when I left it at 2.0