If you're really not bothered about file size, are happy with WMP then just use that to rip in. Wav format. I always used Sound forge but there is some free software that blatantly copied the soundforge interface called wavosaur, I never got on with either audacity or gold wave.
But in Linux you also need to bypass the audio system (probably Pulse Audio) in order to have optimum results right?
I use guayadeque to bypass pulse audio and use alsa directly.
Its interesting you say that because I have had the opposite experience. Perhaps its my lack of in depth knowledge with linux or the fact that I havent seriously used linux since about 2013, but I could never get the same quality out of it. Has it changed or am I just wrong about linux audio?
yes - ALSA usually allows only one sound source at a time, similar to windows kernel streaming.
Pulseaudio is designed for surround sound, bluetooth and other consumer grade uses, so unfair to expect it to perform well for professional or audiophile use
I use JACK and ALSA and audio quality still seems unchanged. That said, my current audio gear isn't as good as pro gear I used to have access to. My age and love of live music don't help either
2 Likes
Professionals record in 32-bit floating point and then downsample that for the market. That has nothing to do with hardware. It's all done in software. Red Book audio defines 44.1 KHz as the standard for audio. In movies, the audio is sampled at 48 KHz. As Freaksmacker said,
But there's really no excuse to stick with this standard when we can produce much higher sample rates. You do however need the equipment that supports and reproduces these high sample rates of 96 KHz and up. This would kick off a resurgence of home audio which probably saw its peak in the late 70s, just before the CD arrived. Should this ever happen, our descendants will be confounded by how we ever thought 128 kbps MP3s were good, because the shortcomings of lossy compression can be exposed by good equipment.
You are largely confusing frequency range and hearing range with sampling. The sample rate dictates how many samples are taken of a waveform per cycle. The higher the sample rate, the closer the digital waveform gets to the analog. Of course your could never truly reproduce the analog, but you can get pretty damn close. This has nothing to do with hearing ability. Yes, the maximum threshold of hearing in humans is 20 KHz. I know infrasound can be felt, but I'm not sure about ultrasound.
ya ? I agree. There has always been just one rule of thumb. Always try and acquire your audio in the highest possible quality. Sample rate,bit rate,bit depth and storage formats is often confused and mixed up.
OP was asking about ripping CDs in Windows, didn't want to go off on tangents into unrelated audio theory
Digital sampling at any quality is still theoretically inferior to an analogue recording, but the brain interpolates the digital samples to a continuous signal anyway and the distinction isn't audible.
No recording can capture the full dynamics of a live performance.
Even rock and pop music live is a very limited experience compared to hearing a full orchestra performing Bach or Beethoven in a philharmonic hall
Jimi Hendrix - the sound quality is often terrible, but it really doesn't matter when listening
1 Like
Agreed, you can always make a compressed copy for some specific need, but for archival purposes, they should be in some form of lossless.
flac is smaller and wav has a bit more size. wav is uncompressed. people do feel wav fares a bit better compared to flacs. mp3 is total crap unless itâs 192 or 256 (or above) and maybe thatâs why artists donât mind people downloading compressed files from streaming sites. window media player does rip em off but you can also check out others. itunes have alac (apple losless) and can rip other formats too.