Right, but one response to this ...

(2019-04-15, 06:34 PM)BusterD Wrote: So to sync this audio, I’m first going to set the sample rate at 44056Hz (without actually resampling to this rate) and then resample to 48kHz instead of 44.1, since I’ll need to get 48Hz for BD compatibility anyway, and I don’t want to resample the audio twice.  This will mean no 44.1kHz for this audio, but oh well.

... might be be to perform a pulldown on the audio.  As AviSynth reads the output accurately, MKVToolNix muxes the output, and the resulting MKV seems to play back without incident, avoiding resampling becomes feasible again.  I couldn't test this without SoundForge, because I don't have it, so I went the ffmpeg route.  Given that we keep the original sampling rate to preserve quality and to remux with MKV rather than BD in mind, it seems that this remains possible in this case, provided that SoundForge can output audio at any sampling rate.

Oh I see. Yeah that definitely works, I've also tried it in the past. The obvious problems are that the compatibility with hardware players and with the Blu Ray standard is very questionable.

Now, I always say I don't care about compatibility and I don't, but there is another aspect to this that is actually more important, and that is the native sample rate of the hardware you are using to play the track back. Let's take a normal computer sound card. Usually these run at 44.1 khz or 48 kHz. Any finer setting usually isn't possible. Even the expensive semi-professional Firewire audio interface I worked with only allows to set the sample rate to a few defaults.

So what this means is that upon playback, a resampling will occur anyway to match the sampling rate of the hardware output. And that resampling very likely won't be of very high quality, just the rudimentary default algorithms that the system already has. So while you don't lose any quality on a pure data level (keeping the bit-perfect representation), actually playing it back will result in quality loss from the resampling done by the playback hardware. Therefore, for best quality, it's arguably preferrable to use a high quality resampler and deliver in 48kHz and thus guarantee best possible playback quality.

So my approach, for example with Cinema DTS, is typically:
1) Perform sync with bit-perfect capture at native sample rate, keeping basically all of the quality.
2) Keep/share this sync for archival purposes.
3) Use high quality resampler (izotope 64 bit SRC) to resample to 48kHz for delivery with projects.

Thoughts?

(2019-04-16, 10:39 PM)TomArrow Wrote: The obvious problems are that the compatibility with hardware players and with the Blu Ray standard is very questionable.

There's no question that BD compatibility is out the window without resampling to 48kHz, which is why the established procedure is already to make two tracks as you do with Cinema DTS.  In respect of how well sound cards handle original/irregular sampling rates, it's up to the one receiving the track to judge and is therefore out of the hands of the one providing it.

Going back to the OP, can the blu ray 1080i be IVTC'd back to 23.976? If so then upping the framerate to 24.000 would allow you to keep the LD audio at 44.1kHz.

I don't think NTSC LD players were capable of outputting anything other than 59.94Hz. All the variations within the standard affect the chroma subcarrier (and in the case of Japan's NTSC the IRE level and white point).

Thanks again to all of you for the help.

(2019-04-16, 04:56 PM)TomArrow Wrote: @OP
What kind of capture do you have of the LD? Is it a bit-perfect capture or simply a capture of the Laserdisc player's output at 44100? If it's bit-perfect, I'm gonna argue that whatever recording software you used simply adapted itself to the speed the data came in, and thus you ended up with 44.1 kHz audio (because that's what the stream itself is coded as) even though it may have played slower on the Laserdisc player.

Yeah, it's bit-perfect. I use Sound Forge for recording, and when recording from digital, I have to set my sound card to use the external audio signal as sync (M-Audio Delta Control Panel > hardware > sync source > external - spdif). So you are at least correct in that if I record a 44.1kHz signal with Sound Forge set to 48kHz, the resulting file is just the 44.1 audio with the samples sped up to 48kHz. If I then use Sound Forge to set the sampling rate to 44.1 without resampling, the file will play back correctly. So I assume the something similar would happen if the LD audio was actually 44056Hz.

(2019-04-16, 04:56 PM)TomArrow Wrote: To verify this theory, you could do an analogue capture of the audio from the laserdisc player and see if it syncs better, which would be the case if the native 44.1 kHz audio is played back at ~44,056 Hz. You would then possibly also be able to see a bit of a frequency cutoff on the spectrum of the audio file.

Tried capturing an episode's worth of LD audio as a test through the analog inputs (where my PC's internal clock is used for sync timing rather than the external digital input) and still got sync drift compared to the BD. So I guess that's just how the LD was recorded.

(2019-04-16, 10:25 PM)Chewtobacca Wrote: Right, but one response to this ...

(2019-04-15, 06:34 PM)BusterD Wrote: So to sync this audio, I’m first going to set the sample rate at 44056Hz (without actually resampling to this rate) and then resample to 48kHz instead of 44.1, since I’ll need to get 48Hz for BD compatibility anyway, and I don’t want to resample the audio twice.  This will mean no 44.1kHz for this audio, but oh well.

... might be be to perform a pulldown on the audio.  As AviSynth reads the output accurately, MKVToolNix muxes the output, and the resulting MKV seems to play back without incident, avoiding resampling becomes feasible again.  I couldn't test this without SoundForge, because I don't have it, so I went the ffmpeg route.  Given that we keep the original sampling rate to preserve quality and to remux with MKV rather than BD in mind, it seems that this remains possible in this case, provided that SoundForge can output audio at any sampling rate.

Hmm, Sound Forge refuses to play a 44056Hz file but it will allow me to edit one, and MPEG Video Wizard DVD will play it, albeit somewhat distorted and slowed down. I use the latter program for spot checking a synched audio file since it allows playback of 2 files simultaneously using the same cursor, anyone know another program that does this? Tried old versions of Adobe Premier and Audition (the ones that Adobe was giving away free keys for) a long time ago, but wasn't able to get either of them to play 2 files simultaneously, not sure if they've improved in the meantime.

So I guess I can make edits on the original 44056 file, then check the sync in MPEG Video Wizard DVD, and I can always briefly set the sampling rate back to 44.1kHz if I need to play a section of the file in Sound Forge, then set it to 44056Hz once more. Will make editing a bit more trouble, but probably not too much.

Haven't tried muxing 44056Hz audio yet, but MPC-HC will definitely play just the wav file. Not sure if it's playing at the correct speed though, as my receiver still says 44.1kHz.

(2019-04-17, 05:43 PM)zoidberg Wrote: Going back to the OP, can the blu ray 1080i be IVTC'd back to 23.976? If so then upping the framerate to 24.000 would allow you to keep the LD audio at 44.1kHz.

I'm sure it can, but I have close to zero experience with working with AviSynth or anything like that. Plus re-encoding the video opens a whole nother can of worms, as there's a bunch of problems with the video that could use fixing, as I mentioned in the OP.

(2019-04-17, 05:43 PM)zoidberg Wrote: I don't think NTSC LD players were capable of outputting anything other than 59.94Hz. All the variations within the standard affect the chroma subcarrier (and in the case of Japan's NTSC the IRE level and white point).

Yeah, I've pretty much come to the same conclusion. I assume that there's frame skips in the LD video signal, but haven't looked at the video closely yet. Just captured the audio.

(2019-04-17, 05:57 PM)BusterD Wrote: Yeah, I've pretty much come to the same conclusion.  I assume that there's frame skips in the LD video signal, but haven't looked at the video closely yet. Just captured the audio.

Given it's anime and mostly likely shot on 2's (or greater) you may never see any skips/blending.

Using eac3to's -slowdown should bring your LD capture to the correct speed

Thanks, but I'm not going to be preserving the LD video (just the audio), and I don't need it for synching to the BD audio.

(2019-04-17, 05:52 PM)BusterD Wrote:

(2019-04-16, 10:25 PM)Chewtobacca Wrote: Right, but one response to this ...

(2019-04-15, 06:34 PM)BusterD Wrote: So to sync this audio, I’m first going to set the sample rate at 44056Hz (without actually resampling to this rate) and then resample to 48kHz instead of 44.1, since I’ll need to get 48Hz for BD compatibility anyway, and I don’t want to resample the audio twice.  This will mean no 44.1kHz for this audio, but oh well.

... might be be to perform a pulldown on the audio.  As AviSynth reads the output accurately, MKVToolNix muxes the output, and the resulting MKV seems to play back without incident, avoiding resampling becomes feasible again.  I couldn't test this without SoundForge, because I don't have it, so I went the ffmpeg route.  Given that we keep the original sampling rate to preserve quality and to remux with MKV rather than BD in mind, it seems that this remains possible in this case, provided that SoundForge can output audio at any sampling rate.

Hmm, Sound Forge refuses to play a 44056Hz file but it will allow me to edit one, and MPEG Video Wizard DVD will play it, albeit somewhat distorted and slowed down.  I use the latter program for spot checking a synched audio file since it allows playback of 2 files simultaneously using the same cursor, anyone know another program that does this?  Tried old versions of Adobe Premier and Audition (the ones that Adobe was giving away free keys for) a long time ago, but wasn't able to get either of them to play 2 files simultaneously, not sure if they've improved in the meantime.

So I guess I can make edits on the original 44056 file, then check the sync in MPEG Video Wizard DVD, and I can always briefly set the sampling rate back to 44.1kHz if I need to play a section of the file in Sound Forge, then set it to 44056Hz once more.  Will make editing a bit more trouble, but probably not too much.

Haven't tried muxing 44056Hz audio yet, but MPC-HC will definitely play just the wav file.  Not sure if it's playing at the correct speed though, as my receiver still says 44.1kHz.

There's another way of doing what you intend to do. Just leave the LD audio at 44.1 kHz and instead retime the Blu Ray audio to which you're trying to sync. This is how I typically do my Cinema DTS syncs, as I also want to keep the bit perfect representation. Practically this means, for example, that I take the Blu Ray audio and speed it up to resemble the Cinema DTS speed. So basically the opposite of what you've been doing to the LD audio.

Let's say Blu Ray audio is 48kHz and we consider these 48,000 Hz to be the slowed down speed and we intend to figure out the sped up speed. So then we get the equation x * 1000/1001  = 48,000 Hz.

Resolve for x: x = 48,000 / (1000/1001) = 48,000 * 1001 / 1000 = 48,048 Hz

So, interpret the Blu Ray audio as 48,048 Hz, then resample it to 44,1 kHz and then you can do your sync with the LD audio at 44,1 kHz, including audio preview and all the bells and whistles.

When your sync is done, all you need to do is what you already know how to do: Change to 44056 Hz and you already have your working sync. Archive that one, and for any project you can resample to 48kHz.

I typically do the syncing itself in Cubase, which also doesn't like weird sampling rates afair, but allows me to play simultaneously with multiple tracks for the syncing, so that's always been my workaround.