Like the sound from the PA but not in the box

Wow that is lighter and fluffy. I am jealous.

I should break out my big daddy swiss army knife the 414..That has a dynamic range above 144. Try to take advantage of this 6i6.

Here is a jazzy rip off the DAW I added red 2 4 band EQ -4 -2 +4 +4, Red 3 compressor 4:1 taking 5bd off. Same mics as last night. The VSTs make it sound worse. So I put the dry no VSTS up too.

Can anybody Vouch for the Focusrite VST that come with thie product line?

HIgher target peaks to try and sound full and big. Watch out. It is starting to sound amplified.

MJB there is no distortion cause I was kinda problem solving. I want it clean so I hear if there is any kind of pinch or cut or breaknup. My ole' ADA MP-1 is plenty capable to handle the dirty side.
 

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Thinking about how I would like to integrate my rack stuff to the 6i6. I went to use a LDC and SM58 together guess what? The 6i6 has a ganged phantom power system. It applies the 48v to both sides. Man, problem after problem.

The Gen1 6i6 was $100 off reverb. Probably does not have the best preamps. I have invested in rack equipment.

Would a TB12 or Aphex 107 or RNP 8030 sound better? In the 6i6. Even the Mackie 16 channel mixer has preamps on each channel. Supposedly.

I am 45 been playing music almost 4 decades. Been in studios, I wanna do it myself.
 
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Thinking about how I would like to integrate my rack stuff to the 6i6. I went to use a LDC and SM58 together guess what? The 6i6 has a ganged phantom power system. It applies the 48v to both sides. Man, problem after problem.

Hi,
That's no problem. The sm58 will happily 'ignore' phantom power.
It's good practice to avoid plugging in/out whilst phantom is engaged but other than that just work away. :)
 
An update, I feel there is progress. You tell me. The sound has improved with more volume with the addition of phantom power. In earlier post Rich suggested adding a direct also. There are 2 which one is better? 2nd one has no compression and -6 .

This is +48 LDC cabinet and direct, with a VF-1 really bad fake spring reverb. There are some nice vibrations and reverberations I am finding moving the mic around. I want pure tone. The search continues..
 

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Latency of under 10mS is usually OK. Mines 8.9 and I can work happily with that and it doesn't bother me. It's the equivalent of having your amp 8 feet or so away from you.

Back to latency. In Reaper the upper corner states about 9 ms.

On a careful listen all the instruments are not in time. This is the 9 ms of latency. How do I record with 0 latency?

2 short clips to illustrate the problem. Same riff. Careful on playing I was tapping the mic for a percussion section. Should NOT be too loud.

The second clip is nudged 9 ms. Sounds to me to be more in time...
 

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Digital systems have latency - if it's below around ten,a few people will notice but pressing a key on a piano VSTi at 9ms is not perceived by most as a delay between key press and sound. get to 15ms and there starts to be a proper delay, and it's terrible to work with. Zero never happens because of how digital audio works, Complete data words get generated, moved around then decoded. That takes time. It is normal. Get it down to single figures and all is good.

Neither sound in time to me. Made worse by the inherent problems of banging a mic. You don't reallly have a percussive sound, so the start of each hit is blurred to a degree. If yiu use your nails, does it get better? Probably not good because of the unsuitability of using a mic as a drum.
 
Lets talk quantization. So I rap out some midi hits for beats. Load the sample from the mic. Tap it into reaper. Then I double click the work flow, and turn on quantization.

Its looped , and requantized, how can it not be in time..?
 
It doesn’t introduce extra latency because of the internal structure but all a to d and d to a takes time. The only question is can you hear it. As I said. If you walk across the room, can you hear that extra delay? I can’t. If you go to a gig, how far from the stage do you go until the drummers hands drift from your ears? If you hear latency then something needs fixing.
 
Is it a common practice to reamp? Like record at mic level in the DAW, then use a ReaInsert to externally reamp the track in different preamps to line level?
 
I guess you could do that type of reamping...but the more common use is for recording guitar tracks DI into the DAW, then running the track out to different guitar amps and dialing them in to taste. Mostly done to allow people the option of not committing to an amp sound during initial tracking...or you can also split the guitar signal, record through an amp and also DI...and then use the DI track to add-to the other or in case you don't like the original.

I think with mic preamps...the real "juice" comes from the interaction of the mic and the preamp...so feeding a line level signal to a preamp after the fact is not quite the same thing...plus, you already used a preamp to record the first mic track...which is different from the way guitars van be recorded DI.
IOW...the first mic is already adding something to the signal, but with a DI track, it's coming direct from the source.
Of course...the best thing is to try things out, you may like the results...or you may find that it's too much effort for minimal gain.
 
One thing for sure I have yet to figure it out.

I was thinking that it could be the line level dropping from +4 to -10 at the interface. How would I work around that? The PA is all +4 still and the speakers sounds as it should sound. I throw a cel phone in the center of the room and play guitar and voice. Raw through the samsung cel voice recorder it sounds more like me.

From the manual ,
INST the input configuration for the jack contacts at Inputs 1 .and 2 can be selected via software from Focusrite Control.The green LEDs illuminate when INST is selected .With INST selected, the gain range and input impedance are altered (relative to LINE), and the input is made unbalanced .This optimises it for the direct connection of instruments (usually via a 2-pole (TS) jack plug) .When INST is off, the inputs are suitable for the connection of line level signals. .Line level signals may be connected either in balanced form via a 3-pole (TRS) jack or unbalanced, via a 2-pole (TS) jack.

So does it matter if they are altered to line anyways? Are they telling me I need a XLR to TRS 3 pole cord to the interface?
 

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A quick purchase of 2 x XLR to TRS cords. NOW I AM +4! To the 6i6 too.

So, I found where all the volume went. The cords cut it down to -10 as suspected at the interface.
 

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Bad , news. It is the same weird sound.

Instru or line is the same sound, but different level. That is weird.

The most change I got was, not mic, not position, not player...it was selecting ASIO driver.

My EL-34 stack has no TRS inputs , its not a MESA with a selection for line or instru. However up until that point of amplification it is +4 will it even matter at the amp unbalanced? Or is that what is changing the sound. The PA is huge , but all +4.

So how with a Marshall EL-34 stack do I stay +4?
 
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Frustrated. How is it that I can throw my cel phone in the center of the room and capture a better sound than anything going through the interface?

Frustrated...and yeah , I mess up all the time. To err is human.
 

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You seem to be confusing a lot of different things that aren't necessarily related.

+4 dBu is the nominal signal level for professional audio recording and sound reinforcement systems. Within this category there are many different types of signal. This is the specification for line level, and it relates to voltage. Connections are generally XLR balanced, 1/4 inch TRS balanced, or 1/4 TS unbalanced.

-10 dBV is a different spec for line level audio used for consumer electronics. Home stereo equipment, MP3 players, computer speakers and stock computer sound cards are examples that typically will use a -10 signal. The voltage is much lower. The typical interface for this type of signal will connect with a 3.5mm (1/8) plug, or RCA connectors. Unbalanced.

Sometimes you can switch between the 2 levels depending on what equipment you have. If you want to run audio out from your computer to powered studio monitors or a PA system or something running at +4, you need some kind of interface that can do that. Set to +4. You can run the same audio out to cheap home entertainment computer speakers, but you need to set the level to -10.

A -10 singal running through a +4 system will make your speakers sound like soup cans. The output will be low. A PA system with a mixer channel that has stereo RCA connections will typically do the conversion for you, or allow you to switch. The output will be okay if it's converted properly to the right line operating level. A +4 signal running through a -10 system will sound like something is about to blow up.

For pro audio (+4 systems), you might be connecting sources that are either mic, line or instrument level signals. A preamp is a device that converts mic or instrument level signals (typically mic signals) to line level. A microphone converts sound waves to electricity. The impedance of the signal is much lower than line level, and the voltage is usually much lower than line level. The preamp will allow you to boost the low output level of the mic up to the correct voltage, and convert the impedance. Instrument level signals are generally higher than mic signals but lower than line level. You don't need as much boost. They're also typically much higher impedance than line level. If you connect an instrument source to a line input the impedance will be wrong. That's why the Focusrite has a switch.

As to what the issue is with your system, nobody can really guess unless you want to tell us what the entire chain is. What microphone? What preamp? What interface? What kind of computer? What operating system? What recording software? Did you download and install the correct drivers?
 
Thanks for that clarification. Are you hearing soup cans? I hear something that is not right and I want to get it squared away. It is so frustrating I get 1 measure in, hate the sound and call it quits.

The 6i6 is set to line in the software, as opposed to instru. They do not say mathematically or show a -10 +4 value.

I hear thinning, and a weird echo that get stronger with a more complex sound. The echo is in every room and direct. That is strange. I can effect the echo with the quality slider when I am using ASIO4ALL.

Hear in this clip I hear an echo, or ambient delay, however there is none. What would cause this? It is a complex sound to exacerbate the condition.
 

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As to what the issue is with your system, nobody can really guess unless you want to tell us what the entire chain is. What microphone? What preamp? What interface? What kind of computer? What operating system? What recording software? Did you download and install the correct drivers?

The Focusrite driver has its own version of ASIO. You shouldn't need ASIO4ALL.
 
Thanks for that clarification. Are you hearing soup cans? I hear something that is not right and I want to get it squared away. It is so frustrating I get 1 measure in, hate the sound and call it quits.

The 6i6 is set to line in the software, as opposed to instru. They do not say mathematically or show a -10 +4 value.

I hear thinning, and a weird echo that get stronger with a more complex sound. The echo is in every room. That is strange. I can effect the echo with the quality slider when I am using ASIO4ALL.

Hear in this clip I hear an echo, or delay, however there is none. What would cause this? It is a complex sound to exacerbate the condition.
We still do not know what you are doing, or even why.

If you have an amp and a guitar and a mic, and you say the amp sounds big/great, WHY are you not just recording that?

I haven't touched my electric in over a year probably, but attached me flailing for half a minute on my old Epi LP Special II, plugged into a MicroCube with an SM58 in front of it. There was some reverb and it was a "classic stack" setting, all the knobs (gain, volume tone, master) straight up. The MP3 was bounced out without any modification/normalization/etc. - recorded *way too hot for normal mixing*, with the gain on my old Saffire was set about 2:00, so lots of headroom using the SM58 - and mind you, this was a 2 watt amp.

(Slight noise at the end of me just talking to the back of the mic from about 6 feet away. Image in the MP3 is the setup.)
 

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