Which DAW Is Best For Audio Recording?

[Nabeel Ansari]

Gunna agree with this^

did you try recording in Edison?

Since I record to songs and not to make excerpts, I record straight to the playlist. Seems much easier that way.

[Frederic Petitpas]

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[SonicThHedgog]

Since I record to songs and not to make excerpts, I record straight to the playlist. Seems much easier that way.

I can only record straight to the playlist when im lucky o.O how do you do that/

[Nabeel Ansari]

I set my buffer to the lowest and never have a problem, latency is 3 ms.

When I set it higher, I get 6 to 8 ms of lag... How does the buffer affect the sound ? My sounds seems to record top notch, when I playback it always sounds as recorded.

When I set my interface to record at 192khz however, when I playback the program says the computer is too slow. I have a 6600 + with Windows 7 Ultimate 64

edit: and 4Gb of hyperx 800MHz and I keep the FW interface at 96kHz

Why are you recording at 192khz? You know that makes it WAY MORE work on the CPU right? I don't know much about AMD processors, but that one sounds ancient...

Just record at 44.1 or 48 khz, I guarantee it will pretty much sound the same...

Anyways, buffer is like how much you can process. If you don't use virtual instruments, a better example would be how many effects you can use before the playback starts to stutter. The higher the buffer, the more latency (not ideal for recording at high buffers), the more effects/virtual instruments like synthesizers you can process. The lower the buffer, the lower your latency (ideal situation for recording).

Your buffer doesn't affect the sound quality of your recordings, it affects the latency of when your playing will reach the computer. That's why you feel there's a little lag between you playing and hearing it on the computer, it's a high buffer (high latency).

If you find your audio in general (not your recordings themselves) starts stuttering and the FL Studio CPU meter starts to freak out, you need to raise your buffer. If you raise it too much where it becomes an uncomfortable amount of latency, then you either bounce some other tracks into audio clips or you get a new computer. I don't record much anymore because it's too much of a hassle to deal with my audio getting all stuttery because my computer can't handle it. I'm getting a new system with a great processor, so my processor will be able to handle low latency buffers even with a lot of other stuff in the projects. (so I can record comfortably with other stuff in the background, like other instruments and effects)

I can only record straight to the playlist when im lucky o.O how do you do that/

I click the record button and hit record to playlist.

[SonicThHedgog]

I set my buffer to the lowest and never have a problem, latency is 3 ms.

When I set it higher, I get 6 to 8 ms of lag... How does the buffer affect the sound ? My sounds seems to record top notch, when I playback it always sounds as recorded.

When I set my interface to record at 192khz however, when I playback the program says the computer is too slow. I have a 6600 + with Windows 7 Ultimate 64

edit: and 4Gb of hyperx 800MHz and I keep the FW interface at 96kHz

The only thing i would record at higher then 96khz is clean eletric, acoustic instuments and pianos, an orchestra, but other then that , recording that high is overkill.

there should not even be an diffrence in quality from 48-192 from people hearing unless its very loud I think.

[Nabeel Ansari]

The only thing i would record at higher then 96khz is clean eletric, acoustic instuments and pianos, an orchestra, but other then that , recording that high is overkill.

there should not even be an diffrence in quality from 48-192 from people hearing unless its very loud I think.

Basically this ^

[Frederic Petitpas]

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[dannthr]

Basic recording theory:

Your sample rate should be twice your top recorded frequency (look up Nyquist Frequency) + maybe 10-20% for good measure.

Every bit added to your bit-depth adds about 6.02dB of dynamic range. 16-bit vs 24-bit is a matter of increasing your dynamic range (your signal to noise ratio) by about 48dB.

[AngelCityOutlaw]

I don't understand all the FL Studio recording issues. :/

My first time ever recording, my latency/volume was just fine.

RAM (amount, not Mhz speed or timings) doesn't decide your speed/max workload, the processor does. Going from 2GB RAM to 16GB RAM if you have buffer underruns won't change a damn thing if you're running a Pentium 4/Athlon 64. That's a misconception that amount of RAM influences the speed of your computer directly. It doesn't increase speed, it increases the maximum amount of stuff you can have open/loaded. Latency has absolutely nothing to do with RAM.

In other words, get a better processor, not more RAM. Your processor is most likely the cause of your audio playback clipping/stuttering. Also, go into the Audio settings and when you're recording, set the buffer rate to as high as possible where it's stuff comfortable to record. That's where it's possible to set it to low latency. If the audio stutters, it means your computer can't handle processing your work at a low buffer rate.

Cool

10chars

[SonicThHedgog]

Thanks for the infos guys

Anyway, at the end of the day, if the music ends up on a CD it's gonna be 44.1kHz

So... Record mode = Low Buffer & Mixing mode = High buffer ! I finally get it .. I was afraid of touching it when I got 3 ms (which I find perfect/don't feel the lag)

Remeber to dither when you go from anything higher then 16bit to keep dynamic range in your track. I usually use Pow R in ableton.......wait I dont think fl gives you a choice.

[Frederic Petitpas]

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[Fishy]

You should dither whenever you resample. If you record and work at 24 bit, you should use dither when you export your 16-bit wav. Likewise if you record at 48kHz, you should dither for your 44.1 export.

Dither basically adds noise equivalent to the level of the least significant bit available. While this adds a tinsy tiny bit more noise, it helps to decorrelate the quantization error of the resample from your signal by randomizing the error.

End result: you get a pretty much inaudible increase in your noise floor, but the resampling process sounds less like distortion and more like harmless noise.

"I punch those numbers into my calculator it makes a happy face." - Cave Johnson

[Nabeel Ansari]

"When life gives you lemons, DON'T MAKE LEMONADE! Make life give those lemons back!" - Cave Johnson

[Frederic Petitpas]

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[Nabeel Ansari]

You guys don't know what Cave means here rofl

Someone who's "cave"

http://half-life.wikia.com/wiki/Cave_Johnson

[LuIzA]

Likewise if you record at 48kHz, you should dither for your 44.1 export.

technically, dither only has to do with the noise floor, and dynamic range. Going down on sample rate can cause aliasing, to prevent that, some processors use anti-aliasing filters, which are basically LPF around half the sampling rate (22.05 KHz for 44.1KHz). check out for more info: http://en.wikipedia.org/wiki/Aliasing

Other than that, for recording like most of us do, there's no real point in working at a higher sample rate your work is going to end up on (44.1 KHz for CDs and mp3s). You won't get a huge difference, or maybe even a noticeable one in most cases.

[Fishy]

technically, dither only has to do with the noise floor, and dynamic range.

those aren't the point of dither, it's side effect that only really comes into play at very low bit depths (8-bit etc).

The point of dither is decorrelating quantization errors from your signal when you sample or resample audio. Granted it's more way important for bit depth changes then it is for sample rate changes (where aliasing is the main gremlin) but it still has a positive effect on any resampling.

The only effect is has on your dynamic range is you can potentially hear signals below the least significant bit of a shallower bit depth. By randomising all values a little dither might push some very low level signals just above the least significant bit where they can be heard, but thats right on the noise floor and already barely there at all. It makes a huge different for 8-bit signals, but almost no effect for 16-bit (in terms of dynamic range).

Other than that, for recording like most of us do, there's no real point in working at a higher sample rate your work is going to end up on (44.1 KHz for CDs and mp3s). You won't get a huge difference, or maybe even a noticeable one in most cases.

This^. 44.1kHz all the way.

If you wanna release at 88.2kHz, then go for it. No one will be able to appreciate it properly though. You need a badass setup and trained ears to really hear it.