LazerBeakShiek
Rad Racing Team
Hell with it.
Like the sound from the PA but not in the box
- Thread starter LazerBeakShiek
- Start date
TalismanRich
Well-known member
Did you call the number or just do online support? It looks like you got a reply from the moderator of the UA forum, but the forum is NOT UA, its more like HR.com but concentrating on UA products.
LazerBeakShiek
Rad Racing Team
They do not answer their phones.
On the website they need to change..HAVE THE PRODUCT REGISTERED BY SERIAL NUMBER. Let the owner do everything himself and have complete control. Makes sense for me.
Edit- by Thursday the product was authorized. Busy for the holidays.
On the website they need to change..HAVE THE PRODUCT REGISTERED BY SERIAL NUMBER. Let the owner do everything himself and have complete control. Makes sense for me.
Edit- by Thursday the product was authorized. Busy for the holidays.
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LazerBeakShiek
Rad Racing Team
Going on. Possibly getting better. Neve Sim, and the acoustic.
VG aa , and the VG aaa via the Apollo w/ a Preamp sim.
VG aa , and the VG aaa via the Apollo w/ a Preamp sim.
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LazerBeakShiek
Rad Racing Team
What do I need to know about equipment that outputs at 48k vs 96k? For instance..
The connection to the USB interface from the rack device is XLR. The Reaper DAW screen continues to say 96k rate. However I confirm that the device is only 48k output from the DA menu section.
What is this going to effect? Like my Fantom is 48k. I do not see the point in recording at 48 so leave the Apollo at the default 96k sampling rate with 512 blocks.
Also the Apollo has a lower latency than the Focusrite. The 6i6 was 9ms latency, the Apollo USB is 6-7ms or less. On boot up once 5ms depending on whats is running in the background. I figured that was worth mentioning. Nice impressive work UA.
The connection to the USB interface from the rack device is XLR. The Reaper DAW screen continues to say 96k rate. However I confirm that the device is only 48k output from the DA menu section.
What is this going to effect? Like my Fantom is 48k. I do not see the point in recording at 48 so leave the Apollo at the default 96k sampling rate with 512 blocks.
Also the Apollo has a lower latency than the Focusrite. The 6i6 was 9ms latency, the Apollo USB is 6-7ms or less. On boot up once 5ms depending on whats is running in the background. I figured that was worth mentioning. Nice impressive work UA.
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gecko zzed
Grumpy Mod
I leave interface and Reaper both set to 44.1
From Reaper (clicking on top right hand corner of the screen or going through preferences) you can control the ASIO settings on the interface.
Sometimes I change to 48, but only when I'm involved in video work.
Any latency below 10 is fine.
From Reaper (clicking on top right hand corner of the screen or going through preferences) you can control the ASIO settings on the interface.
Sometimes I change to 48, but only when I'm involved in video work.
Any latency below 10 is fine.
TalismanRich
Well-known member
I am running at 88kHz and 24 bit. I think I'm set at 128 bit buffers. It helps the latency a bit, and it's plenty of headroom for anything I might do. I don't worry about file sizes these days. With terrabytes of storage so cheap, why bother? Final mixdowns go to 44.1K
LazerBeakShiek
Rad Racing Team
Um, I mean . Does that information need to match? I would assume 96k device recorded in a 48k environment would lose something.
However the 48k device into a 96k environment seemed , OK. Are my ears just not good enough to hear the diff?
However the 48k device into a 96k environment seemed , OK. Are my ears just not good enough to hear the diff?
TalismanRich
Well-known member
I don't know if you can record a 96K device into a 48K environment. If the rates don't match, something won't work. I have Reaper set the sample rate and bit size of the interface.
That said, I've built projects where I imported files at 44.1k from my Zoom R24 and added tracks at 96k. Reaper seemed to handle it just fine.
The reason for the higher sampling rates is just to give a more accurate sample of the waveform, and to avoid any possible aliasing of any high frequencies when converting. My thought would be to use the highest rate through the process to maintain as much leeway as possible until you do the final conversion. I chose to use 88K as its a multiple of 44K. I honestly don't know if it would make a difference going from 96k to 44k. I don't have a clue as to the math that is used in the conversion process.
I certainly can't hear a difference between 44k and 96k produced file, as I've lost the top end of the hearing spectrum over the years. If my tinnitus flares up due to sinus issues, its even worse. I don't hear the sparkle of the cymbals like I used to!
That said, I've built projects where I imported files at 44.1k from my Zoom R24 and added tracks at 96k. Reaper seemed to handle it just fine.
The reason for the higher sampling rates is just to give a more accurate sample of the waveform, and to avoid any possible aliasing of any high frequencies when converting. My thought would be to use the highest rate through the process to maintain as much leeway as possible until you do the final conversion. I chose to use 88K as its a multiple of 44K. I honestly don't know if it would make a difference going from 96k to 44k. I don't have a clue as to the math that is used in the conversion process.
I certainly can't hear a difference between 44k and 96k produced file, as I've lost the top end of the hearing spectrum over the years. If my tinnitus flares up due to sinus issues, its even worse. I don't hear the sparkle of the cymbals like I used to!
I would certainly not profess ANY knowledge of digtal sampling mathematics but am reliably informed that a sample rate beyond 44.1kHz does NOT result in "more accurate sampling" That is assuming a cut off frequency of 20kHz. 44.1Khz can reconstruct a perfect 20Hz sine. (the hardware might not be 'perfect' but upping the sample rate will not fix that)
There is I understand a very rare possiblity of problems at 44.1 if recordng high frequency content at high levels, i.e. close to 0dB fs but I doubt that wouild worry most home recording jockeys.
Of course, as stated, disc are now so cheap and so colossal that if you want to record at 96kHz, no reason not to.
Dave.
There is I understand a very rare possiblity of problems at 44.1 if recordng high frequency content at high levels, i.e. close to 0dB fs but I doubt that wouild worry most home recording jockeys.
Of course, as stated, disc are now so cheap and so colossal that if you want to record at 96kHz, no reason not to.
Dave.
rob aylestone
Moderator
I record and output at the same rate to avoid up and down sampling. In my video world - video and stills, the errors always seem to creep in when creating intra-frames from two others, or throwing data away when going down in size. I cannot believe in audio it works for you. When I've been in very top end studios, you'll see them take time on mic placement rigging, then listening and making small adjustments. I see this as the key thing you can do to get good recordings. Perfect mic placement at 48K, or 44.1K (my two output sample rates I get asked for) rather than 96K or higher and simply being unable to hear it - while moving a mic a few inches I can hear. In our less than premium recording spaces, placement and capture surely must be the sensible place to spend time rather than recording this less than perfect space in higher resolution than is needed?
snow lizard
Dedicated Slacker
Given 2 files of the same length in time and same bit depth, eg. 24/48 and 24/96, the one sampled at 96k will be twice the size of the one at 48k. Computer horsepower keeps improving, but once you mix a session with a lot of tracks and a lot of plugin processing going on, the 96k file has twice as many samples. It chews resources faster. Some of the lower end converters might run and sound better at one sample rate or another. Or not. Might be interesting to check.
At the end of the day, different people have different ways they like to work. The most popular sample rates are 44.1 for audio and 48 for video. People often choose the target rate of the format they will end up in to avoid Sample Rate Conversion, which is an extra processing step. SRC works, but it can cause complications. A lot of folks like to avoid it.
Recording source files at 24 bits rather than 16 is much more popular than running high sample rates. 24 bit files are much better at handling post processing. High sample rates don't necessarily make your SM57 or square wave synth sound any better. In theory, 96k should allow us to capture stuff beyond the range of human hearing. But why would we need to? Coded messages for the dog?
192 kHz sampling is a marketing idea that doesn't really work all that well unless you're trying to record the echolocation calls of bats or something. Even then, most of the stuff you'd find in a recording studio was never meant to handle frequencies that high. You'd need specialized tools to make it work.
At the end of the day, different people have different ways they like to work. The most popular sample rates are 44.1 for audio and 48 for video. People often choose the target rate of the format they will end up in to avoid Sample Rate Conversion, which is an extra processing step. SRC works, but it can cause complications. A lot of folks like to avoid it.
Recording source files at 24 bits rather than 16 is much more popular than running high sample rates. 24 bit files are much better at handling post processing. High sample rates don't necessarily make your SM57 or square wave synth sound any better. In theory, 96k should allow us to capture stuff beyond the range of human hearing. But why would we need to? Coded messages for the dog?
192 kHz sampling is a marketing idea that doesn't really work all that well unless you're trying to record the echolocation calls of bats or something. Even then, most of the stuff you'd find in a recording studio was never meant to handle frequencies that high. You'd need specialized tools to make it work.
LazerBeakShiek
Rad Racing Team
Really the 44.1 is adequate, ok. I will keep the sample rates conservative to help resource use.
A quick question on low cuts. There are low cut options at every point in the chain. How many times does the low cut need to be enabled ? If I turn the low cut on at the microphone, does the preamp need the low cut button pressed as well? Will rumble be created from a continuity mismatch?
A quick question on low cuts. There are low cut options at every point in the chain. How many times does the low cut need to be enabled ? If I turn the low cut on at the microphone, does the preamp need the low cut button pressed as well? Will rumble be created from a continuity mismatch?
rob aylestone
Moderator
Low cut? If you need it, then the last point in the signal chain is the best place, anywhere else means you might have to add more later if you overcooked it. Where is your rumble coming from? I use low cut almost automatically on live recordings because the bottom end is often a mess, and I use it live. In my studio projects I rarely ever use it, unless there's a problem to solve.
LazerBeakShiek
Rad Racing Team
After setting the sampling to 48k and 128bit chunks the latency dropped to 3-3.5. Nice one. Still nice and clear.
On 'newdec' I was recording some measures of voice, bass, and guitar. Keeping the mic about 12" back as a reference. Levels are -18db. This seems really quiet. When I turn it up it does not distort. The mains are near 0. The tracks are -19 guitar, -23 bass, -16 Drumsoftware, vocal is -18 or so on the slider as well.
On TestDD I was turning up the mains to peak at -6 . This sounds more appropriate.
Does it sound ok, because I hear no static? Then there are now a bunch of UA preamps, and eq's , and the pluggins I need to sort through. See what sounds like what.
Any recommendations based on what you hear?
Location
On 'newdec' I was recording some measures of voice, bass, and guitar. Keeping the mic about 12" back as a reference. Levels are -18db. This seems really quiet. When I turn it up it does not distort. The mains are near 0. The tracks are -19 guitar, -23 bass, -16 Drumsoftware, vocal is -18 or so on the slider as well.
On TestDD I was turning up the mains to peak at -6 . This sounds more appropriate.
Does it sound ok, because I hear no static? Then there are now a bunch of UA preamps, and eq's , and the pluggins I need to sort through. See what sounds like what.
Any recommendations based on what you hear?
Location
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TalismanRich
Well-known member
I threw both files into Audacity, just to see how much they could be boosted to hit max 0dB peaks. Newdec, was -19dB below peak, the TestDD was -6dB. I think a good point to shoot for would be about 1/2 way between them, as you have plenty of headroom, but won't be getting close to the noise floor. You'll probably find that the peaks are snare and bass drum hits, and handclaps.
I didn't hear any distortion on either one.
I've routinely normalized some tracks to about -2 or 3dB when they were recorded significantly lower than other tracks in a project.
Soundwise, newdec seemed to have a lot of sibilance. Some of it might the Sony 7506 headphones I'm using right now. I find the top end to be pretty aggressive on them.
I didn't hear any distortion on either one.
I've routinely normalized some tracks to about -2 or 3dB when they were recorded significantly lower than other tracks in a project.
Soundwise, newdec seemed to have a lot of sibilance. Some of it might the Sony 7506 headphones I'm using right now. I find the top end to be pretty aggressive on them.
LazerBeakShiek
Rad Racing Team
Sibilance what are you suggesting for that? Dropping the highs with an EQ.
When I was looking for an answer for the low cut switch and which one to use. I found this in the P420 manual. AKG is suggesting to NOT SING DIRECTLY INTO THE MICROPHONE. Are they serious? So I need to be at an angle? Always?
When I was looking for an answer for the low cut switch and which one to use. I found this in the P420 manual. AKG is suggesting to NOT SING DIRECTLY INTO THE MICROPHONE. Are they serious? So I need to be at an angle? Always?
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TalismanRich
Well-known member
Sibilance can be a combination of several things, aggressive high end of a mic, a bright voice with lots of "s" type sounds, adding EQ in the 5-8K range. When I'm trying to get a mix, it makes a big difference what I'm listening on. If the speakers are kind of dark, or lack extended frequency response, you might boost frequencies to make up for what appears to be missing. If the speakers are bright, you add bottom and cut highs, which then makes it dark on other system.
As for AKG's recommendation, if you don't have an effective pop filter, then you need to avoid going straight at the mic, especially if you are close. The plosives will pop out at you. If you look at a many announcers, they will often have the mic about nose level and pointing down at the mouth. That's because the natural flow of air is usually down and outward, not up. You get the proximity boost that announcers like, without the pops.
A lttle trick, if you hang a few shreds of tissue paper at the front of the mic and watch if they move, you'll know if you have an issue with air hitting the mic. Good pop filters will alleviate a lot of the problem.
As for AKG's recommendation, if you don't have an effective pop filter, then you need to avoid going straight at the mic, especially if you are close. The plosives will pop out at you. If you look at a many announcers, they will often have the mic about nose level and pointing down at the mouth. That's because the natural flow of air is usually down and outward, not up. You get the proximity boost that announcers like, without the pops.
A lttle trick, if you hang a few shreds of tissue paper at the front of the mic and watch if they move, you'll know if you have an issue with air hitting the mic. Good pop filters will alleviate a lot of the problem.
LazerBeakShiek
Rad Racing Team
So how come CD's sound great from device to device, from homes to cars? How do they make it seem so easy?? Is there a pluggin for that?
I admitt I like the mic preamps coloration. The added color, depth like a funnel, and even quack is amazing. I would love to have a real life preamp collection. The Apollo twin is fantastic like that. Couldn't be happier with the recommendation. Kudos to you and all at United Audio.
LazerBeakShiek
Rad Racing Team
Is this Baking? Trying to understand the terms. Snow was mentioning this too.