Best bass rig ideas

  • Thread starter Thread starter LazerBeakShiek
  • Start date Start date
Falling out of time with what? There is no reference to time unless you're talking about tempo... if so, no real means to measure tempo unless I pull out a metronome... Which I ain't doing.

It all sounds fine to me.
 
The way the insert , doesn't keep up.

Is it important to keep the Processor in the FX loop of the interface at the same bit rate? Like can I use a USB Roland thing at 44.1k sample rate, on a DAW and interface set to 48k ? Will that screw up midi devices looking for a master BPM or timings?

Does matching a 44.1k sample rate with 48k have anything to do with midi timing?
 
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This is style dependent in my opinion. Metal and modern rock aren't going to be the same recording process as classic rock, pop, punk etc...

I've recorded bass rigs (some really nice ones), I've recorded with hardware, I've recorded direct, and I've recorded direct only to add a sim after the fact. They all have their respective place in my opinion.

The biggest factors in my opinion is the bass, the strings (way more so on bass than guitar), and the technique the player uses.
 
LazerBeakShiek said:
The way the insert , doesn't keep up.

Is it important to keep the Processor in the FX loop of the interface at the same bit rate? Like can I use a USB Roland thing at 44.1k sample rate, on a DAW and interface set to 48k ? Will that screw up midi devices looking for a master BPM or timings?

Does matching a 44.1k sample rate with 48k have anything to do with midi timing?
Click to expand...

This only matters if you are connecting it digitally via s/pdif, adat, or AES/EBU. If you are running it in an analog fx loop it doesn't matter.
 
YouTube
I no longer have the words to explain this.
YouTube
How do I make it so I don't chop and nudge?
 
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How are you monitoring the bass track as you are recording it? When you record the bass, are you listening to the bass track after it goes through the DAW? or are you listening to the bass through the UAD monitor mix in Console?

In the 2nd video, the Input source for the bass says HD Audio Stereo Input 1. What is that? Shouldn't it be the UAD?
 
Good questions.

I was using headphones when recording to monitor from the Apollo headphone jack.

It says HD stereo 1 because to get sound in the video I flipped the driver from Apollo USB ASIO to ASIO4All. It came out the laptop speakers for that playback only. The headphones would not be heard too faint on video.
 
Do you hear any latency while you're recording? Or is it only after you recorded and you're playing it back that the bass track doesn't line up?

Stupid questions, I know, but I want to get a better understanding.
 
Not stupid.

It was like that as I was recording. I was fighting the track because it seemed to be compensating for my corrections. Throwing it back out of time. If that is understandable.

Yeah , the whole time I was like what the hell is this? I was aware it was out of time, in real time.
 
Okay so, you can hear the latency as you're recording? You hit the string and half a second later, you hear the audio (or whatever the delay is)

So, what happens when you play the bass with the UAD and you do not have Reaper running? Like, completely shut down Reaper and just use the UAD and the Console software. You shouldn't hear any latency at all.
 
What is this in the menu? Medium?

Annotation 2020-03-24 151310.webp

With Reaper closed there is no latency from me plucking a string noticed.

Should I turn this off?
Annotation 2020-03-24 153430.webp
 
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No latency with just the Console program. That tells me you are listening to the recorded track in Reaper when you're recording the bass and not the direct monitoring from the UAD Apollo. I might be wrong, but that is the most common cause of latency type delays like you're seeing. I'm not familiar with Reaper, but be sure you are listening to the bass directly from the UAD and not a foldback track from Reaper. There has to be a way to turn the track monitor off in Reaper.

The input delay compensation is to offset the recorded track by the amount of latency of the slowest plug-in. I'm not sure of the details of how and when it is used. I have it turned off in my UAD Console settings.
 
Chili said:
No latency with just the Console program. That tells me you are listening to the recorded track in Reaper when you're recording the bass and not the direct monitoring from the UAD Apollo. I might be wrong, but that is the most common cause of latency type delays like you're seeing.
Click to expand...

Where or how should I be monitoring?

I have the monitors plugged in to the back of the Apollo. The headphones are plugged into the Apollo's front. There is more to it?
 
LazerBeakShiek said:
Where or how should I be monitoring?
Click to expand...

Probably someone who knows Reaper can help to show you where to look to see if you are direct monitoring or not.

Did you try recording with the input compensation delay off? I'm curious to see how that works. I'll give it a go myself later tonight.
 
Totally awesome. The UA input delay compensation was turned off, and the tracks layer well in synch with each other.

I didnt nudge anything. What song did you record? I did 'Stuck on You' by Lionel Richie.
Annotation 2020-03-25 044709.webp

Very cool.

Downside of shutting the IDC off is that now in the DAW corner there is 38/60 ms of latency delay showing, instead of 2.8/6. It is constant, so I can work with it.

Here is the mp3 of the short project. I stacked like 6 instruments to crude timing. They line up, the mix wasnt too important.
StuckOnYou by Mark C | Free Listening on SoundCloud
 
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Glad to hear that fixed it. Glad to know what that button does. I didn't get a chance to test it last night. We had our last neighborhood happy hour before lockdown last night. Priorities!!

You can set your buffer to a lower setting. Mine is at 256 or 512, or something. But honestly, that delay should not matter if you are direct monitoring while recording. The higher the number, the easier on your CPU and you shouldn't notice the difference. Ideally.
 
Should I take the sample rate down to match the hardware at 44.1k and 24 bit? Set the sample rate to 24?
 
If you're planning to put your tunes to video, then 48khz is a good idea. If you're just going to be making MP3s or CD's and keep it all in the audio realm, then 44.1khz is good. But be consistent across the board. I break my own rule and do 44.1khz for everything whether it goes to video or not. I really haven't seen a problem.

I do 24bit for everything.

I don't buy into the notion that higher bit depths or higher sample rates sound better. 44.1khz and 24bit sounds great all day long. The file size is manageable and the CPU workload is light.
 
1024 bit buffer seems excessive. My lowly Tascam is fine at 128 bits.

A higher sample rate should result in lower latency. Assuming the driver is not causing additional delay, the latency will be Buffer size /Sample rate. Theoretically, 256 buffer at 96kHz is the same latency as 128 buffer at 48kHz. Also, remember that you need to multiply that by 2 for I/O round trips. If your system and driver are fast enough to poll the USB ports before there is a buffer overflow, you can lower the rate. If your system can handle 64bit buffers, then you can really lower latency by lowering buffers and upping the bit rate.

With direct monitoring, there will be no I/O round trip latency. It is looped back from the input to the output directly.
 
Like Chili, I tend to do everything at 44.1/24. If there are times when I know something is intended for video, I will change to 48/24. However, I often forget to change back, and discover later that a whole bunch of audio stuff has been done at 48/24 as well. It has not been a problem, specially as everything gets rendered to what I want in the end anyway.

I have a buffer size of 256 (the default), which results in Reaper showing me a latency of 13/15. I have never found need to change.
 
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