Using 44.1kHz LPCM track on Blu-ray?

[Onti]

Hi! I’m not sure but I think 48 kHz is required for Blu-ray. However, I tend to see that you share audio files taken from Laserdisc as 16-bit, 44.1 kHz (FLAC). I imagine I must convert the 44.1KHz (FLAC) to 48 KHz LPCM with eac3to. If I create Blu-ray, burn it and pop it into a Blu-ray player. What would happen to that new 48 kHz LPCM track? Any errors? The audio will play too fast?

If one audio is at 44.1 kHz on a Laserdisc and could be recorded at 48 KHz, why don’t you do that? There must be some important reason. Would some errors take place?

[deleted user]

The laserdisc tracks are digital and their native digital sampling rate is 44100 Hz, thats why they are not "recorded" at 48 kHz, rather they are captured bit-perfectly as they are, at least usually. You can resample to 48000 Hz, ideally with a good resampler like Izotope 64 Bit SRC. Idk if the BR standard allows 44100 Hz.

[zoidberg]

Blu ray specs require 48kHz or greater sampling rate. Some media players will play files with 44.1kHz audio, others will not

[spoRv]

DVD specs allow for audio track these frequencies: 48khz, 96khz; Blu-ray specs allow 48khz, 96khz, 192khz.

https://en.wikipedia.org/wiki/DVD-Video
https://en.wikipedia.org/wiki/Blu-ray#Audio

[Onti]

Thank you very much, guys!

The laserdisc tracks are digital and their native digital sampling rate is 44100 Hz, thats why they are not "recorded" at 48 kHz, rather they are captured bit-perfectly as they are, at least usually. You can resample to 48000 Hz, ideally with a good resampler like Izotope 64 Bit SRC. Idk if the BR standard allows 44100 Hz.

Ok, this is what I have done with the FLAC file…

1.Sound Forge Pro - Process - Izotope 64 Bit SRC

2.New Sample Rate - I choose 48.000

[Image: f4f7865f265ca0ad0eb80e65856e21c7o.jpg]

And now, should I select the “Use simplified quality setting” check box? If I do so, I would only have to drag the Quality slider to adjust the plug-in’s controls automatically. When the “Use simplified quality setting” check box is cleared, I can adjust the plug-in’s controls manually.

I don’t understand what I have marked in bold. I can't see that check box in my window (Mac version):

Steepness

The plug in uses a low-pass filter to discard frequencies that cannot be represented or are undesirable in your audio output.

This setting establishes the steepness of the transition band of the low-pass filter. Higher settings will reject unwanted frequencies, but can cause more ringing in the time domain and a higher CPU load.

Max filter length

Sets the maximum length of the filters used for resampling. The default setting will work well for most applications, but you can increase the setting if very high-quality output is desired for uncommon source or destination sampling rates.

Cutoff scaling

Allows you to scale the cutoff frequency of the plug-in’s low-pass filter from the Nyquist frequency.

Typical values are near 1. Higher values will offer a flatter pass-band, and lower values will offer better aliasing suppression.

Alias suppression

Sets the amount of suppression in the low-pass filter’s stop-band. Frequencies in the stop-band that are not fully attenuated will result in aliasing. Higher settings will result in better quality, and lower settings can minimize CPU load.

Prering

Low-pass filters are characterized by the amount of ringing they introduce into their output. Higher Steepness settings produce increased ringing. A setting of 100% produces a linear phase filter with equal pre and post ringing. A setting of 0% produces a minimum phase filter that offers no preringing but has nonlinear phase distortion. Intermediate settings allow a tradeoff between preringing and  postringing and allows you to linearize phase in the pass-band.

Set the sample rate only (do not resample)

Select this check box to change the playback rate without resampling the data. This means that the original pitch of the file is not preserved.

In short, I just want to change the sample rate but at the same time make sure I keep the quality of the Laserdisc track intact. What should I select?

[stwd4nder2]

Just use ffmpeg

Code:

ffmpeg -i input.flac -acodec pcm_s16le -f s16le -ar 48000 output.pcm

That'll convert your flac to 48000hz PCM without any quality loss.

[spoRv]

That'll convert your flac to 48000hz PCM without any quality loss.

Well, if you convert from 44100hz to 48000hz there WILL BE some quality loss, even if small - and I'm sure inaudible to anyone but dolphins, whales, bats and superheroes, perhaps...

[stwd4nder2]

That'll convert your flac to 48000hz PCM without any quality loss.

Well, if you convert from 44100hz to 48000hz there WILL BE some quality loss, even if small - and I'm sure inaudible to anyone but dolphins, whales, bats and superheroes, perhaps...

Going from 44.1kHZ to 48kHZ shouldn't give any quality loss, right? It wouldn't be as good as something natively at 48000HZ, but that shouldn't be an issue in this case. Unless I'm misunderstanding then you'd only be losing anything if you went from 48 down to 44.1

[zoidberg]

Unless you absolutely bork the conversion you won't notice. If you're in any doubt just use a setup that allows you to listen to the 44.1kHz natively ie media player, HTPC etc

[spoRv]

If you are not a:
- dolphin
- whales
- bat
- superheroe (with great hearing like Superman)
you should not worry!