Cubase 5 @ 96Khz problematic...suggestions?

symbolofmylife

Hi Guys,

MY pc setup at home is as follows: i5 3 GHZ, , 8 GB DDR3 Ram, Seagate 1TB HDD 7200rpm 6GB/s 32MB (partitioned), Windows 7 64x with a Focusrite Scarlett 2i2 USB interface.

Ok, whatever that means...

So, I am running Cubase 5 and initially set my project up to run/record at 96KhZ, 32 bit float. From there down to 48Khz, 32 bit float, and then finally 44.1KhZ, 32 bit float.

It appears that I can only get the machine to run (playback & record) without glitches when the settings are at 44.1KhZ, 32 bit float. At higher bitrates the ASIO time overloads and playback/recording stops. At 44.1 the ASIO time meter runs up about 20%.

I am also running about 4 VST's, and using the ASIO Full Duplex Driver (I tried the Generic Low Latency one also, but it does not appear to work well. I also tried ASIO4ALL but I do not get sound when I use it...channel strips do light up but no sound is played)

I'm not too clued about pc specs...so here come the noob questions:

1. Does playback/recording at a higher sample rate add loads more strain on the PC? (I'm guessing, yes... ??? But to the point where the system wont record as the load is too much?Even when going up as little as 48 from 44.1?)
2. Should this machine not be able to carry this load?
3. I have read online to invest in a separate harddrive to record to. At present I am using one partitioned drive. Will a separate HDD sort this out (or at least really perform better)?
4. Are there better generic drivers out there?

Manfred-Klose

what audio interface/ soundcard are you using ?

symbolofmylife

Manfred Klose wrote what audio interface/ soundcard are you using ?

The Focusrite Scarlett 2i2 (for sound out and recording)

ezietsman

The bitrate doesn't put more load on your PC. It works like this:

Your software receives a chunk of sound data from the soundcard. If it is running at 96kHz, that means you will have 96000 data points (amplitude versus time) per second. Once the chunk is read, lets say there is 256 data points per chunk, your computer does stuff to it. However, it only has 1/96000 s to do what it must do before the next chunk needs to be read.

Imagine this is your sound coming in:

. . . . . . . . . . . . . . . . . . . . . . . .

your computer reads it in chunks:

|. . . . |. . . . |. . . . |. . . . |. . . . |. . . . |

Between successive chunks you have 1/bitrate (1/96000 in your case) seconds to run VSTs and run all the other things. This takes time. If you try and do too much the software will miss the next chunk and that chunk will be dropped, the software will catch the next one instead.

Another thing: Computers can deal with integer numbers MUCH quicker than floats. Have you tried using integers instead of floating values? Your computer should be fast enough to deal with this though, <WINDOZE USERS> can you tell your computer to NOT put your CPU in power saving mode? Linux puts CPUs in a lower clockspeed mode when they don't work very hard, and this messes with realtime/low latency audio, but you can tell it to stay put at 3GHz or whatever.

symbolofmylife

ez wrote The bitrate doesn't put more load on your PC. It works like this:

Your software receives a chunk of sound data from the soundcard. If it is running at 96kHz, that means you will have 96000 data points (amplitude versus time) per second. Once the chunk is read, lets say there is 256 data points per chunk, your computer does stuff to it. However, it only has 1/96000 s to do what it must do before the next chunk needs to be read.

Imagine this is your sound coming in:

. . . . . . . . . . . . . . . . . . . . . . . .

your computer reads it in chunks:

|. . . . |. . . . |. . . . |. . . . |. . . . |. . . . |

Between successive chunks you have 1/bitrate (1/96000 in your case) seconds to run VSTs and run all the other things. This takes time. If you try and do too much the software will miss the next chunk and that chunk will be dropped, the software will catch the next one instead.

Another thing: Computers can deal with integer numbers MUCH quicker than floats. Have you tried using integers instead of floating values? Your computer should be fast enough to deal with this though, <WINDOZE USERS> can you tell your computer to NOT put your CPU in power saving mode? Linux puts CPUs in a lower clockspeed mode when they don't work very hard, and this messes with realtime/low latency audio, but you can tell it to stay put at 3GHz or whatever.

Ok...that makes sense, thanks...but still, then I do not understand why this only occurs in 48 or 96KhZ settings and how to get around it (if you say the pc should be capable...but it's struggling)? I've tried muting the VST channels (to attempt to free up some resources) to only run recording/recorded audio but that does not help either.

ezietsman

Are you running the ASIO drivers and the Cubase project at the same bitrate? I assume you have installed the USB 2.0 drivers for the 2i2? Else anything over 44.1 won't work well or at all.

Also, have you disabled the onboard sound device? I have little experience with recording on windows, sorry.

Malkav

Have you tried freezing the tracks running the vsts?

symbolofmylife

ez wrote Are you running the ASIO drivers and the Cubase project at the same bitrate? I assume you have installed the USB 2.0 drivers for the 2i2? Else anything over 44.1 won't work well or at all.

Also, have you disabled the onboard sound device? I have little experience with recording on windows, sorry.

Yes, from what I can see when you change the project settings the driver bitrate also changes accordingly (or am i wrong?)

The 2i2's drivers are installed yes and visible from the Driver Setting section, but selecting it also leads to no sound....even if I set it's buffer way up....

The only sound device the PC has is the 2i2. There's no onboard sound card, and I have set windows theme sounds to "no sound".

Chad Adam Browne wrote Have you tried freezing the tracks running the vsts?

Not yet no. Will that still enable me to edit them (the midi events)?

Malkav

Chad Adam Browne wrote Have you tried freezing the tracks running the vsts?

Not yet no. Will that still enable me to edit them (the midi events)?
[/quote]

Nope, you gotta unfreeze the one you wanna work on in order to edit it again, a little time consuming but it can free up enough resources to get things running well again.

Also certain VSTs kill processing power, for instance sometimes running 1 instance of Omnisphere can completely own a PC, depending on how complex the synth is...

greetings

1. Make sure your power setting is set to high performance.

2. Don't record at 32 bit. Record at 96 kHz, 24 bit. Your computer should easily handle it. I don't even think the 2i2 supports 32 bit floating. 96 kHz, 24 bit is the professional audio standard, I don't think you need more than that. I do know that many mastering studios still don't support 32 bit floating yet.

3. A separate harddrive would help, but you shouldn't have these problems without it.

symbolofmylife

Frodo Baggins wrote 1. Make sure your power setting is set to high performance.

2. Don't record at 32 bit. Record at 96 kHz, 24 bit. Your computer should easily handle it. I don't even think the 2i2 supports 32 bit floating. 96 kHz 24 bit is the professional audio standard, I don't think you need more than that. I do know that many mastering studios still don't support 32 bit floating yet.

3. A separate harddrive would help, but you shouldn't have these problem without it.

Thanks! On the Focusrite, you're right....silly me....it only supports 24....so I'll change that tonight and see if it makes a difference. You think it will? (I also read on some forum that 32 float and 24 are basically the same apparently, so that's why I ask)

As far as I remember I did set the pc to maximize performance, but will also double check.

Malkav

Frodo Baggins wrote 2. Don't record at 32 bit. Record at 96 kHz, 24 bit. Your computer should easily handle it. I don't even think the 2i2 supports 32 bit floating. 96 kHz, 24 bit is the professional audio standard, I don't think you need more than that. I do know that many mastering studios still don't support 32 bit floating yet.

Very valid points here ?

I know it may seem like a weird thing to bring up but with guitars I normally tend to record at 46kHz just cause if you bounce down to 44.1kHz from 96kHz you lose a bit of headroom which makes anticipating how the guitar will sound in a mix a little difficult ? If this isn't for a cd release then don't worry about that, I just like to make sure things sound as similar in my DAW as they will through a stereo ?

symbolofmylife

Chad Adam Browne wrote
Frodo Baggins wrote 2. Don't record at 32 bit. Record at 96 kHz, 24 bit. Your computer should easily handle it. I don't even think the 2i2 supports 32 bit floating. 96 kHz, 24 bit is the professional audio standard, I don't think you need more than that. I do know that many mastering studios still don't support 32 bit floating yet.
Very valid points here ?

I know it may seem like a weird thing to bring up but with guitars I normally tend to record at 46kHz just cause if you bounce down to 44.1kHz from 96kHz you lose a bit of headroom which makes anticipating how the guitar will sound in a mix a little difficult ? If this isn't for a cd release then don't worry about that, I just like to make sure things sound as similar in my DAW as they will through a stereo ?

So actually, save for 46khz I may as well just stick with 44.1 than actually?

nick

ez wrote
<WINDOZE USERS> can you tell your computer to NOT put your CPU in power saving mode? Linux puts CPUs in a lower clockspeed mode when they don't work very hard, and this messes with realtime/low latency audio, but you can tell it to stay put at 3GHz or whatever.

If its an Intel CPU, disable C1E in the BIOS, this is the the feature that lowers the CPU clockrate to save power.

Malkav

symbolofmylife wrote So actually, save for 46khz I may as well just stick with 44.1 than actually?

Well it really depends on the scenario, I just think that in the case of some guitar tones recording them at a higher samplerate becomes redundant and makes your mix unpredictable ?

greetings

Don't you mean 48kHz? (not 46kHz)

32 bit and 24 bit definitely aren't the same. I wouldn't bother with 32 just yet, it's not supported widely enough and plenty of great records were even made at 16 bit. That does lower dynamic range though. 16 bit range = 96dB and 24 bit range = 144dB.

We had a workshop with Young Guru (Jay-Z's recording engineer) two weeks ago, and he still records at 44.1kHz, and 24 bit. With a higher sample rate like 96 kHz your latency will be slightly lower, though. There is a reason that audio moved to 96kHz/24 bit. I think it has something to do with making sure that some sort of digital noise floor was completely eradicated from the sampling process (if I remember correctly, it was something to do with eradicating noise).

I would go 96/24. It's quite simply higher quality (more samples per second). But, funnily enough, the guy who makes Lavry converters says sample rates above 96kHz are just marketing hype and don't make any noticeable difference to the sound. He actually said that sampling above 96kHz could be detrimental to the sound, so he sticks to 96kHz. I can't tell you all the scientific reasons as I don't know enough about it.

greetings

Oh, and using 32 bit though it isn't supported probably was the reason, as the hardware doesn't support it.

majestikc

Stick with 24 bits and 44.1khz sample rate and you'll be more than fine.

The sample rate is the thing that craps on your cpu, so doubling it to 96khz is effectively halving your cpu/putting twice as much strain on it (for no real noticeable gains)

The sample rate can be thought of as how often or how much the sound is described.

CD quality audio has 44,100 of these measurements a second. That’s called 44.1 kilohertz (khz).

The bit rate can be thought of as how well the sound is described.

CD quality audio has 65,536 volumes to choose from for every sample that’s measured. That’s called 16-bit audio (because 2 to the 16th power is 65,536).

greetings

majestikc wrote Stick with 24 bits and 44.1khz sample rate and you'll be more than fine.

The sample rate is the thing that craps on your cpu, so doubling it to 96khz is effectively halving your cpu/putting twice as much strain on it (for no real noticeable gains)

The sample rate can be thought of as how often or how much the sound is described.

CD quality audio has 44,100 of these measurements a second. That’s called 44.1 kilohertz (khz).

The bit rate can be thought of as how well the sound is described.

CD quality audio has 65,536 volumes to choose from for every sample that’s measured. That’s called 16-bit audio (because 2 to the 16th power is 65,536).

He has an i5 quad core and 8gb RAM. It should be no problem to run at 96/24.

The gains are noticeable. Well, I can notice them.

majestikc

I haven't tested it for a long time, I dropped to 44.1 a while back (2yrs maybe) BECAUSE of cpu issues and never tried it since, I'll test it out tonight.

(when mixing you can always push the sample rate up to 1024, always had problems with Sonar skipping midi notes if it went any higher than that)