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Sample rate question


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There have been double blind tests to see if either self-proclaimed audiophiles, pro engineers or laypeople could really tell the difference between 44.1khz and 96khz playback. The results were that they couldn't. They COULD tell the difference between a piece of music *recorded* at 96khz and one recorded at 44.1khz, but this doesn't conclusively mean 96khz sounds better - the authors of the study said that the discrepancy could be due to the fact that engineers who record at 96khz may pay closer attention to the mix and generally put more time into it.

Anyway, keep your playback at 96. No harm in recording there except your HD space as IBBIAZ said.

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Anyway, keep your playback at 96. No harm in recording there except your HD space as IBBIAZ said.

Just a note, it's not JUST the space - it's also the amount of disk I/O involved in recording & playback. Meaning, regardless of the size of your disk, 24/96 means it will be reading/writing more data at a time, so if it's slower (<7200RPM) or fragmented or what not, 24/96 will exacerbate the issue and be more likely to cause dropouts, etc.

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You didn't mention what bit depth you're currently using. First don't confuse sample rate and bit depth. Sample rate (for example 44,100 Hz) is the number of times per second that your device records a "piece" of audio. Bit rate (for example 24 bits) tells you how accurately your computer or audio device can playback/record each of those thousands of pieces of audio.

44,100 is a LOT of pieces of audio per second, and without getting into details, that's about the limit of what the human ear can hear. Once you get into higher sample rates (for example 96,000 Hz) it's physically more than your brain can process. So the audio benefits of super high recording rates are questionable. One potential advantage is in downsampling or time stretching audio. If you record at a high sample rate, the programs that downsample (for instance if you want to record to a CD which is at 44,100 Hz) and the programs that timestretch audio will have a lot more info to work with and can produce better results (especially with time stretching).

Bit depth is a little more subtle in how it affects your audio, but unlike going from 44,100 Hz to 96,000 Hz where it's physically questionable whether humans can hear the difference, there is certainly a difference between 16bit audio and 24bit audio. The most important benefit of 24bit audio imo is the increased recording headroom it gives you. More bits means you can record more subtle changes in the audio, and at 16bits you're gonna miss some stuff. "Missing stuff" here would result in digital clipping and that's bad. Also, the greater ability to record subtle changes in audio means that your device can better tell the difference between actual audio and random noise. Layman's terms: you get a lower noise floor and more headroom with higher bit depths.

So bottom line, I usually record at 24bit/44,100Hz if it's a general project. If I'm feeling special I record at 24bit/48,000Hz to eek the last little bit of humanly perceptible audio out of my source.

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Layman's terms: you get a lower noise floor and more headroom with higher bit depths.

Maybe this is my problem.

I still feel like I don't get enough headroom and as a result my loudness suffers still. I've asked about it on the forums before and I've tried a number of ways to fix this issue but I still don't think I'm getting as loud as I can possibly get.

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"loudness" as you put it, is a matter of mastering. you can get more than enough headroom with 16bit/44.1KHz. and finally, as cliche as it is mentioning the loudness wars and stuff, I'll keep it short and simple: loud does not equal good.

personally, I just record with that setting, because it, like the people mentioned before, saves space and CPU while recording, and I don't have to convert it back to 16/44.1 for distribution. also, I don't have half the skills of professional engineers, so chances are, even if I recorded with higher settings, I couldn't get it to sound any better than what I can now.

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"loudness" as you put it, is a matter of mastering. you can get more than enough headroom with 16bit/44.1KHz. and finally, as cliche as it is mentioning the loudness wars and stuff, I'll keep it short and simple: loud does not equal good.

personally, I just record with that setting, because it, like the people mentioned before, saves space and CPU while recording, and I don't have to convert it back to 16/44.1 for distribution. also, I don't have half the skills of professional engineers, so chances are, even if I recorded with higher settings, I couldn't get it to sound any better than what I can now.

Here you go: http://www.korg.com/product.aspx?pd=289

Record at 1 bit-depth, better than analog.

Cheers,

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Yeah loudness really has nothing to do with recording or working at 24/96. If your stuff isn't loud compared to your reference material, it's your mixing/mastering ability that is the problem, possibly influenced by some of your gear and source material.

It still baffles me. Even with my education I wouldn't know where to begin in mixing to be louder.

Also, I do understand that loud doesn't equal good. But it's a matter of gaining a skill for me rather than trying to make it sound better by being louder. Other people's rock/metal music is louder than me. I wish to obtain that level that they have.

And in rock music, loud DOES equal good. :)

To quote the bumper sticker my uncle had on his truck, "If it's too loud, you're too old!"

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Mmm... for the original post, another significant difference between 44.1K and 48/96/whatever K is that many, many common applications of sound (CD use, DVD soundtracks, etc.) are not compatible with the higher grade stuff, so it'll all get reduced to 44.1K, in the end. I can hear the difference between 44.1 & 48 (since Reason records in 48, but outputs in 44.1... ouch), but that's about as far as I can go.

As for 'getting louder'... yeah, it's all in the mixing. My suggestion is to work backwards - first, make the source as loud as you can before clipping/crud occurs, then follow the line to the next amp, make that as loud as you can, etc., until you get to the final mixer. That generally gets you the maximum sound you can achieve without tricks such as compression.

Hard to explain, but once you understand that there's an order to where sound goes when it's produced it becomes easier.

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