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Using 44.1kHz LPCM track on Blu-ray?
#11
The native ffmpeg resampler is pretty awful: https://src.infinitewave.ca/

The "soxr" one seems to be good, but I don't see any reason not to simply use the best one in existence, which to my knowledge is izotope 64 bit SRC. After all we're concerned about preservation. Why unnecessarily introduce artifacts into the audio?

The reason the resampler makes a difference is because PCM does not actually contain the entire waveform at every point in time, only enough to be able to reconstruct it (near) perfectly according to the Nyquist theorem. So what a resampler does, simplifiedly speaking, is, it reconstructs the full waveform and then it measures new samples at the correct places of the new samplerate. If the reconstruction is not precise enough mathematically, you introduce errors/distortions that are then baked into the resampled audio.

Whether it's audible or not shouldn't even matter in my opinion because there's no reason to not just do it right when it's easy to do. Why accept a degradation of the audio if it's not necessary? But I have read accounts of people in the audio industry who say the resampler differences are audible. Someday I'll have to do an A/B test myself.

About the settings ... I personally just use the simplified one with "Highest Quality" since I'm not exactly an expert. I should note that I use it in Sound Forge, which it used to ship with.
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Thanks given by: bronan , Onti
#12
The 44.1kHz captures are the preservations as far as I'm concerned, what happens after that is up to the end user. After all who knows what audio advances will come in the near future
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#13
Wow, according to that site eac3to definitely seems to be the best free option. It even beats Izotope in some tests like Passband, Transition and Phase.
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#14
(2021-07-19, 10:01 PM)bronan Wrote: Wow, according to that site eac3to definitely seems to be the best free option. It even beats Izotope in some tests like Passband, Transition and Phase.

Interesting!

Poor me, that I always used Foobar2000 PPHS or SSRC or Avisynth again with SSRC... Sad
Sadly my projects are lost due to an HDD crash... Sad
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#15
A lot of Blu-ray players will play back mkv files with 44.1 PCM, so that's what I usually do.
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#16
Thanks again to all of you!
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#17
(2021-07-20, 06:12 AM)BusterD Wrote: A lot of Blu-ray players will play back mkv files with 44.1 PCM, so that's what I usually do.

I do this too. Here's the rub though - I have all devices going HDMI into my TV, then an optical cable from there out to my amp. Well, I found out after recording the output from my TV that it resamples to 48kHz before outputting (LG OLED). Now I wonder how good the conversion actually is, I might be better off just converting it with the highest quality app haha
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#18
I have a Panasonic UHD player with an audio-only HDMI port, but when I use it with mkv files, my receiver often thinks that 2.0 PCM or 1.0 PCM is 7.1 for some reason, so I can't use Dolby Pro-logic or anything. But thankfully the player also has a coax audio port, which works just fine with the receiver.

I really hate HDMI sometimes...
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#19
Btw I should mention that test is the opposite of what we're doing. It's taking 96kHz and reducing to 44.1kHz. There are ultra sonic frequencies above our hearing range that can muddy the lower frequencies when converted down, so you have to employ a low-pass filter to properly remove them before doing so. The ones that look really bad aren't doing this properly.

Anyhow, we are talking about going from 44->48, which means no frequencies are being lost, we're just adding more sample points to what's already there. If you know position X is 0.1 and position X + 1 is 0.3 you can pretty safely interpret that a new point between them would be 0.2. In theory you shouldn't hear any difference between that. Going the other way gets trickier since your wave is going out of bounds and must be contained to the new frequency limit of the sample rate.
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