First of all I appreciate audio doesn't have an FPS, but I've come across a problem and wondered if anyone has a solution?
I've got two blu rays of the same movie, the first is 23.976 FPS, 1080p video and DTS MA 5.1 audio. The second is 25 fps, 1080i video and DTS MA 7.1 audio. What I want to achieve is 1080p video with DTS MA 7.1 audio, however due to the FPS difference the audio and video are different lengths.
I really want to keep the video 23.976 FPS so it is compatible with most North American Blu Ray players, plus I want the maximum quality without downconverting anything. I know of a few programs that can stretch/shrink audio, for example MeGui can do it and adjust the pitch correctly for 25 FPS > 23.976 FPS, but it converts the DTS MA to AC3 which I don't want. Does anyone know how I can do this and retain the DTS MA 7.1 audio?
I don't really know much about DTS but I've read about splitting a file into wav's (in this scenario 8 separate files) - could I do that, adjust the length of each WAV and then rebuild the DTS file maybe? Or is there a program I'm not familiar with that can easily do this? Or should I just give up and stick with the DTS MA 5.1 Audio?
My end goal is to make a new blu ray, as above with maximum quality on both the video and audio.
Many thanks in advance.
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Last edited by luciofulci; 28th Apr 2019 at 06:21.
Generally when audio is sped up for PAL, it's not pitch corrected, probably because it's not easy to do without messing with the phase relationship between channels for multichannel audio, so you'd generally want to apply an equivalent lack of pitch correction when slowing it down again. There's exceptions, but it should be easy enough to listen to both to see if there's a pitch difference.
I don't think there's any free DTS MA encoders, so the next best option might be a wave file. Bluray supports 8 channel LPCM up to 96kHz.
You could slow the audio down while converting to flac with MeGUI (as it's lossless), then convert the flac file to wave with something else. I don't know why MeGUI doesn't have a PCM option, although...
eac3to does. You might be able to open the audio with the HD Streams Extractor under MeGUI's Tools menu and use it to convert to wave. It can apply PAL to NTSC slowdown.
If you're not overly familiar with the HD Streams Extractor, the conversion options are disabled by default, for some odd reason, and the menu item for enabling them doesn't go out of it's way to look like a menu item.
Thanks for the info hello_hello.
I have now used MeGui to split the file into 8 wavs, initially I couldn't find the option but if you select 'all demux options' from that settings menu you screen-shotted it gives you the option. One thing I noticed is it refers to the 7.1 audio as strange setup (see below) - is that something to be worried about? I wonder if that blu ray company have somehow taken the 5.1 from the other disc and bodged it to make fake 7.1?
[Attachment 48919 - Click to enlarge]
Once split to wavs these are the files I have:
[Attachment 48920 - Click to enlarge]
Does this look normal? I don't have DTS Master Audio Suite (too expensive) but I found a manual for it and it shows that 7.1 DTS should have L, R, C, LFE, Lss, Rss, Lsr and Rsr. I don't have the last four, instead I've got SL, SR, BL and BR?
Also I tried loading one of the wav files into MeGUI, adjusting the FPS and exporting to FLAC. I will then import the FLAC into audacity and export as a WAV, which I understand will result in no loss of quality as both formats are lossless.
Upon further inspection the video I am syncing to is showing as 24 FPS not 23.976 FPS, it shows this in mediainfo and TSmuxer. The thing is when I changed the FPS of the wav file in MeGUI to 24 FPS the file length was about 6 seconds short, so I tried changing it to 23.976 FPS and the length is pretty much exact (it's a second off). That doesn't make sense to me, unless for some reason my video is being reported as 24 FPS but is actually 23.976 FPS?
Sorry for all the questions! Has anyone come across this before?
There's no surround channels in PC-land, so the audio needs to be decoded to a standard wave-file layout. That's normal and has to happen even when you simply play the audio on a PC. http://www.cs.bath.ac.uk/~jpff/NOS-DREAM/researchdev/wave-ex/wave_ex.html#POSITIONS
I think it was originally based on the official cinema layout that nobody uses, except cinemas.
To make it more interesting, codecs don't always use the same channel order internally.
None of that should matter as the audio should be re-mapped to the wave file layout when it's decoded, and encoders expect a wave file layout as the input and remap the channels as required for encoding. Well... there's some possible ambiguity when it comes to 7.1ch audio, as Microsoft chose the theatre layout as the default Windows 7.1ch layout originally (L R C LFE BL BR FCL FCR) instead of the layout for home cinema (L R C LFE BL BR SL SR). I think that changed with Vista, or maybe Win7. The theatre layout is also the default for AAC. As a result, there was some hackery involved in the early days of AAC 7.1ch encoding. You can read about it here, but the upshot is, if you ever encode 7.1ch audio as AAC, use the QAAC encoder.
If you combine the channels into a single wave file again, be sure to use the wave file channel order. (L R C LFE BL BR SL SR)
Anyway.... the channels are labelled correctly.
Side L/R = Lss/Rss (Lss/Rss might mean surround side).
Back L/R = Lsr/Rsr (surround rear, I assume)
For the record, 5.1ch audio is sometimes decoded with the surround channels as Side and sometimes as Back. Side is preferable given the 7.1ch layout, but all lossy encoders will accept either Side or Back as the surround channels for 5.1ch audio.
Why are you splitting the audio into single channel files though? You can apply slowdown without splitting the channels, you can do it while the audio is being extracted/decoded, and you don't have to worry about channel ordering. If you have an issue playing huge wave files (the maximum wave file size is 4GB, although some programs have a 2GB limit) and w64 isn't an option, decode directly to flac.
w64 is a 64 bit wave file, to support file sizes over 4GB, and most programs should be able to open them these days.
Wave to flac is lossless as long as the bit depth isn't decreased. ie a 24 bit wave file converted to 24 bit flac is lossless. 24 bit wav to 16 bit flac is technically lossy, but not MP3/AC3/AAC type lossy.
Don't worry about the file length. It's not uncommon for the video and audio to be different lengths. The audio sync is what matters. There could be 6 seconds more studio promos or some other garbage at the beginning, but there's no guarantee audio from a 25fps source and 24fps video will sync anyway. Sometimes they can be edited a little differently. You can only try.
Last edited by hello_hello; 30th Apr 2019 at 14:06.
Thanks very much for your help, I'm happy to report that adjusting the file to 24 FPS results in perfect sync - I imagine it's not always that easy but it worked out great.
Very much appreciate the advice.
Further to the above I'm just trying to decide how much of a benefit this will be. Here are the two audio streams:
5.1 DTS HD 1536 kbps
7.1 DTS HD 768 kbps
Which is likely to give the better experience? The higher core bit rate or two extra channels? Am I perhaps better sticking with the 5.1?
Core bitrate has no influence on the quality of the lossless DTS-HD MA extension (XLL). It's only relevant for older players that can play just the core.
Thanks for confirming Sneaker - much appreciated. DTS-HD is a whole new world to me and I don't understand a lot of it.
Last edited by luciofulci; 6th May 2019 at 02:27.
One final (hopefully!) question. I've got a third audio source I want to include from a DVD which is 2.0 AC3. The DVD is 29.97 fps and I need to add the audio to a 24 fps video. How would I achieve this? If I get the audio in sync at the beginning it's slightly out at the end. MeGUI and Eac3to don't seem to have any option for adjusting 29.97 to 24. Is the only option to use a program like audacity to alter the length?