i’m a bit confused because to my knowledge capturing the pcm audio of a laserdisc through optical (toslink) is bit-perfect. i mean what else can one do? increasing the capture from 16bit to 24bit wont do anything, except a higher filesize. increasing the samplerate from 44.1khz to 48khz on capture wont do anything either, in terms of quality.
i mean, the laserdisc pcm is 16bit @ 44.1khz - what more can you do?
so what do people refer to when they talk about bit-perfect and saying that a normal digital capture wont be enough?
please enlighten me 😃
Forgive the lateness of this post, I thought it may be of use to anyone still looking into this.
My understanding is Laserdisc PCM is 16bit/44.1kHz - no more. So whatever capture you do, it would be advised to keep it 44.1kHz as record at 48kHz (or higher) will create sync issues with the video.
Yes, you could set the DAW at 24bit/44.1kHz, but the only advantage of that would be if you need to use plug-in to alter the sound.
I’ve never tried to capture from toslink to DAW as my LD Player doesn’t have an optical out for digital stereo tracks. If it’s just PCM, then I would assume any DAW interface with a toslink input should receive the digital stereo signal.
DTS 5.1 audio, while coming from the same toslink output, needs a decoder to turn it form code into 5.1 channels of audio. This is normally handled by an AV receiver amp. Same deal as the RF AC3 tracks that need a demodulation to turn it into a 5.1 digital signal that would go to an AV receiver amp via toslink or coaxial. So other than using a line level convertor to convert your AV receivers speaker outputs into line level to input to an analogue 6 channel input - I couldn’t tell you how to capture an un-decoded AC3 or DTS signal to 6 channel digital PCM audio while maintaining digital all the way down the chain.