Skip to main content

Hardware set up for live discussion recorded for Podcast

I'm looking to buy a bunch of audio equipment for amplifying/recording live talks in front of an audience.

I want to be able to record each mic as it's own .wav file (into my laptop using Adobe Audition or similar) before sending it to the PA so I can separately adjust gain during the live performance without affecting the recording.

My plan for live sound : 4x RODE M2 mic > audio interface F https://www.amazon.co.uk/gp/product/B01E6T547Y/?tag=r06fa-20 Scarlett 18i8 main out into Yamaha MG10XU mixer > PA

For recording: audio interface > via USB into laptop

But how do I take four channels out of the audio interface so they go into four separate channels on a mixer ?

Or do I need a mic splitter as well ? There must be a better / cheaper way ?

I'm going round in circles so any help would be HUGELY appreciated !

Comments

pcrecord Tue, 08/15/2017 - 10:16
Actually you could plug your mics into the Scarlett (Which has a internal mixer controled by a software) and do the mix there while recording.
Then you could take the output of the scarlett directly to the PA. No need for an external mixer. BUT, it's not always working easy and you may need to EQ each mic differently depending on the speaker. Since the Scarlett only control gain, volume and pans, a mixer is then needed if you need more control.
To do so, you need an interface with the same amount of outputs as mic inputs because each mics should be directed to a seperate mixer input to be properly mixed into the PA.

The realtime mixer of the scarlett let's you make the routing of the input and output... you can mix 1 mic per mix and send 1 mix per output. (some may be stereo output so you'd need to pan the input to make 2 mono

DonnyThompson Tue, 08/15/2017 - 15:07
The Alto seems like overkill for the scenario you are describing. Being that your flow is to send your mix out to a house PA, there are a lot of features to the mixer that you won't need - or likely even use - in the scenario you e described.
Correct me if I'm mistaken, but you just want to be able to send 4 separate mic/line signals out to a house mixer, so that you can control the volume of those mics without effecting the input level to your DAW, right?
Your best bet would be to drop the concept of a mixer in between your DAW rig and the house PA, and step up to an audio interface - such as a Presonus or Focusrite - that has 4 discreet line outputs on the rear. These would be sent to 4 input channels of the house PA, where further processing could be dialed in (EQ, GR, Levels) separate from the input levels of the mics to your DAW. You would control the level of the signals being sent to the house PA using the accompanying software mixer that comes with those devices. Adjusting the line out gain this way to the house, wont effect the input gain to your DAW.

Kurt Foster Tue, 08/15/2017 - 15:14
agreed ..... which is why i made my recommendation for the KORG thingie. you can set the levels to record 4 tracks of audio with the pre amps and then mix those to the house mains thru the faders to the 2 buss. cheap too. the only drawback to it is 44.1 / 16 bit .... but since it's only speech it shouldn't be a problem.

kmetal Tue, 08/15/2017 - 15:46
I'd just plug into the Scarlett which has enough outs to pass each mic into the PA or mixer if needed, or just use the stereo main outs.

I'd then just use a low buffer setting in audition or reaper like 64-128, or 256 if the laptop can't handle it. Using the built in channel eq in audition (if it still has it) or some pluggins if it doesn't. Latency wouldnt be an issue at these settings, especially for voice. Even amp sims in realtime aren't an issue (for me @256).

It's is simple as pluggin in the mics to the interface, and the interface to the PA/laptop, and record enabling the tracks.

Now you've got the ability to add eq and compression in realtime with very little bit to the CPU (assuming something from the last 5-7 years), and you have total control over the levels.

If the PA doesn't have an eq, you can eq the master out for feedback supession and/or room compensation.

its a nice compact easy setup, with full recall, and excellent dsp processing within audition. Compression is important for podcasts since most are not professional broadcasters and wouldn't have the type of mic technique and voice control a pro would. Less hardware means less physical gain staging as well.

The more splitters and hardware you add the more money, and potential problems and troubleshooting comes into play. Live is all about reliability first, quality second.

I'd take that money otherwise spent on a mixer and get a second hard drive for backup, or an isolation transformer for nice clean power, or one of those rack cases that has a holder for a laptop on top so setup was crazy fast.

Kurt Foster Tue, 08/15/2017 - 16:14
kmetal, post: 452121, member: 37533 wrote: I'd just plug into the Scarlett which has enough outs to pass each mic into the PA or mixer if needed, or just use the stereo main outs.

I'd then just use a low buffer setting in audition or reaper like 64-128, or 256 if the laptop can't handle it. Using the built in channel eq in audition (if it still has it) or some pluggins if it doesn't. Latency wouldnt be an issue at these settings, especially for voice. Even amp sims in realtime aren't an issue (for me @256).

It's is simple as pluggin in the mics to the interface, and the interface to the PA/laptop, and record enabling the tracks.

Now you've got the ability to add eq and compression in realtime with very little bit to the CPU (assuming something from the last 5-7 years), and you have total control over the levels.

If the PA doesn't have an eq, you can eq the master out for feedback supession and/or room compensation.

its a nice compact easy setup, with full recall, and excellent dsp processing within audition. Compression is important for podcasts since most are not professional broadcasters and wouldn't have the type of mic technique and voice control a pro would. Less hardware means less physical gain staging as well.

The more splitters and hardware you add the more money, and potential problems and troubleshooting comes into play. Live is all about reliability first, quality second.

I'd take that money otherwise spent on a mixer and get a second hard drive for backup, or an isolation transformer for nice clean power, or one of those rack cases that has a holder for a laptop on top so setup was crazy fast.


it seems like using a sledge hammer to kill a gnat. the OP doesn't have a PA. he'd have to buy one. instead he can use the D888 as a mixer that records. then when it's finished, just plug the USB2 into a computer and it shows up as a hard drive on his desktop with out drivers, configuring bullsh*t etc.

kmetal Tue, 08/15/2017 - 16:54
Kurt Foster, post: 452124, member: 7836 wrote: it seems like using a sledge hammer to kill a gnat. the OP doesn't have a PA. he'd have to buy one. instead he can use the D888 as a mixer that records. then when it's finished, just plug the USB2 into a computer and it shows up as a hard drive on his desktop with out drivers, configuring bullsh*t etc.

True, I didn't realize the OP didn't have a PA. I was under the (mis)understanding he had the Scarlett, laptop, and PA, or was going to get them reguardless. My angle was to just use what they had to the fullest.

If that's not the case, i completely +1 your suggestion as it's just about perfect for the application and very reasonably priced.

Good call.

pcrecord Wed, 08/16/2017 - 10:34
curbahn, post: 452114, member: 50772 wrote: Someone suggested I buy the Alto Pro Live802 and do a loop in the insert cable to make it an output with a simple soldering job !
I have know doubt it could work.. but Alto products are far for high quality.
If you are going to buy your own PA. Look for a small Soundcraft or Allen&Heat mixer and amplified speakers. Some even has computer connectivity that allows you to mix live and record with a computer.
Unless you prefer a stand alone recorder like the D888 that has been suggested.

But 1 important question hasn't been answered : WHY ??
Are you gonna sell the audio or the audio/video ? What level of quality do you expect from the recording ? To be transcripted only or to sell it..?

curbahn Wed, 08/16/2017 - 11:09
pcrecord, post: 452131, member: 46460 wrote: I have know doubt it could work.. but Alto products are far for high quality.

do you mean far 'from' high quality ? uh oh .. I've bought it now

pcrecord, post: 452131, member: 46460 wrote: But 1 important question hasn't been answered : WHY ??

The audio will be edited and mixed down into a podcast

kmetal, post: 452121, member: 37533 wrote: It's is simple as pluggin in the mics to the interface, and the interface to the PA/laptop, and record enabling the tracks.

As i'm using Audio Technica 2010 condenser mics I want to be able to graphic eq the live sound and control if there is any feedback probs

Kurt Foster, post: 452124, member: 7836 wrote: the OP doesn't have a PA. he'd have to buy one.

the OP is a woman

thanks everyone for your help - the first event is on the 30th of Aug , I've now sunk £857 into kit so will let you know how I get on !

kmetal Wed, 08/16/2017 - 11:35
curbahn, post: 452134, member: 50772 wrote: do you mean far 'from' high quality ? uh oh .. I've bought it now



The audio will be edited and mixed down into a podcast



As i'm using Audio Technica 2010 condenser mics I want to be able to graphic eq the live sound and control if there is any feedback probs



the OP is a woman

thanks everyone for your help - the first event is on the 30th of Aug , I've now sunk £857 into kit so will let you know how I get on !


Using the method i described will work fine for your purposes. You can use a graphic EQ on the master bus in your daw and/or a hardware eq on the PA.

If you use the method i described it can save you significant time during mixdown since your settings will be set already. Since your Eqing for the room, you may have to tweak or even restart the settings from scratch but you'll have a reference, at least.

pcrecord Wed, 08/16/2017 - 12:02
curbahn, post: 452134, member: 50772 wrote: do you mean far 'from' high quality ? uh oh .. I've bought it now
For podcast you'll be fine ! don't worry.
Quality is relative of purpose. ;)
You get Pro and amateur members here. And any others that drop by from a google search.
I do voice overs with a mic that cost more than 1k into a 2k preamp into a 1k converter. So yeah there is better than the Alto.
Do you need better, probably not.

One thing tho the Live 802’s built-in USB port allows engineers, producers, and performers to convert two channels of 24-bit audio (Main Mix or Sub Mix 1/2) straight to a computer. So with only 2 channels you won't be able to record 4 mics seperately.

curbahn Wed, 08/16/2017 - 13:57
pcrecord, post: 452139, member: 46460 wrote: For podcast you'll be fine ! don't worry.
Quality is relative of purpose. ;)
You get Pro and amateur members here. And any others that drop by from a google search.
I do voice overs with a mic that cost more than 1k into a 2k preamp into a 1k converter. So yeah there is better than the Alto.
Do you need better, probably not.

One thing tho the Live 802’s built-in USB port allows engineers, producers, and performers to convert two channels of 24-bit audio (Main Mix or Sub Mix 1/2) straight to a computer. So with only 2 channels you won't be able to record 4 mics seperately.

I'll use the inserts as outs from the mixer into the audio interface _just learnt a trick to solder a loop in a 1/4 jack for this purpose :)

curbahn Wed, 08/16/2017 - 14:00
kmetal, post: 452137, member: 37533 wrote: Using the method i described will work fine for your purposes. You can use a graphic EQ on the master bus in your daw and/or a hardware eq on the PA.

If you use the method i described it can save you significant time during mixdown since your settings will be set already. Since your Eqing for the room, you may have to tweak or even restart the settings from scratch but you'll have a reference, at least.

Not sure if I'll have access to a desk for the PA - as far as I know I'm just plugging into it ... and I don't trust my computer to do all that live processing .. it's a mac book pro but prob about 5yrs old, I've put more ram in but still, sometimes editing multiple tracks adobe audition will crash on me

DonnyThompson Wed, 08/16/2017 - 14:57
curbahn, post: 452141, member: 50772 wrote: Not sure if I'll have access to a desk for the PA - as far as I know I'm just plugging into it ... and I don't trust my computer to do all that live processing .. it's a mac book pro but prob about 5yrs old, I've put more ram in but still, sometimes editing multiple tracks adobe audition will crash on me
That doesn't sound right to me (and I'm not even a Mac user (anymore... it's been years).
What CPU and how much RAM? The crashing you described shouldn't be happening by simply editing multiple tracks... unless we're at a semantics thing, with you using the term "editing" to perhaps mean something else. (?)
This could be a simple configuration setting, either in your computer settings or your DAW. Your MBP is five years old... But What you describe still shouldn't be happening working with basic commands.
I've done very dense processing work in the past on systems older (and less powerful) than your MBP, using what would now be considered to be totally obsolete by computer geeks...
Do you have more than one audio device installed on your system? If so, have you checked to make sure the one you use for your editing work is the default device? Have you visited the manufacturers site to make sure your drivers (and if applicable, firmware) is up to date? Are you running a virus protection app that is active while you are working in your DAW program?
Just tossing out some thoughts here....

kmetal Wed, 08/16/2017 - 19:05
All great thoughts from Donny.

I used run audition on a (2001) 566mhz celeron desktop, and later a (2006) core 2 duo laptop, and I used a soundblaster USB card, and later an m-audio FW1814. My average recordings were 8-24 tracks with some edits and punches and more pluggins than the tracks needed (lol first thing I was taught at the studio was less efx make a better mix in general, regiarless of CPU power).

Adobe audition not only sounds better than any other daw besides Samplitude, it's processing is easy on the computer. I'm talking I used to run multiple convolution reverbs.

So there's something not right as far as settings or hardware or configuration in general, because those systems are/were way less powerful than a MBP from even 10 years ago.

pcrecord Thu, 08/17/2017 - 05:10
curbahn, post: 452140, member: 50772 wrote: I'll use the inserts as outs from the mixer into the audio interface _just learnt a trick to solder a loop in a 1/4 jack for this purpose :)
Yes you have 4 inserts on the 802.
But, you might not want to do this if the preamps of the interface sounds better than those in your mixer.. You need to test this first !
The best audio path might be mic / interface / live mixer...

But WAIT, what I was talking about is to take your Alto mixer AS an audio interface (therefore having just this unit to bring to venues)
Didn't you think of that ? this was the center of the discussion we are having !

I suspect you should have researched a bit more before buying it. (doubled, since the 802 wasn't a recommendation made here)

curbahn Thu, 08/17/2017 - 05:19
Ok I have since updated my laptop to Sierra and BOUGHT an annual subscription to Adobe Audition and am confident the crashes will stop now . Thanks for alerting me to take action on that front ...

pcrecord, post: 452151, member: 46460 wrote:

But WAIT, what I was talking about is to take your Alto mixer AS an audio interface (therefore having just this unit to bring to venues)
Didn't you think of that ? this was the center of the discussion we are having !

As an audio interface the Alto only provides 2 outs, I want each mic to have it's own file so it's easier to edit for the podcast.

Ideally I'd have one unit to do it all but I felt that I would loose capabilities if I relied on one thing to do it all. Also, I want to be able to record podcasts in a studio with this kit - which I will be able to do by removing the mixer from the equation. I'm happy to have a mixer for live stuff and an interface for studio stuff.

curbahn Thu, 08/17/2017 - 05:23
pcrecord, post: 452151, member: 46460 wrote: Yes you have 4 inserts on the 802.
But, you might not want to do this if the preamps of the interface sounds better than those in your mixer.. You need to test this first !
The best audio path might be mic / interface / live mixer...

TRUE ! hadn't thought of that. the presonus audiobox has 4 outs AND L/R main outs - just not sure if I can configure them to correspond with the 4in...

pcrecord Thu, 08/17/2017 - 05:33
curbahn, post: 452154, member: 50772 wrote: TRUE ! hadn't thought of that. the presonus audiobox has 4 outs AND L/R main outs - just not sure if I can configure them to correspond with the 4in...
Yes the audiobox 44VSL has 4 in and 4 line out... it also has a realtime mixer that let's you choose what plays where. I have no doubt each mics could be assigned to different output.
The realtime mixer also have effects EQ, Comp, Gate. This unit could be used without an external mixer (Which is the second senario we discussed earlier) ;)


Attached files

DonnyThompson Thu, 08/17/2017 - 14:28
pcrecord, post: 452156, member: 46460 wrote: Yes the audiobox 44VSL has 4 in and 4 line out... it also has a realtime mixer that let's you choose what plays where. I have no doubt each mics could be assigned to different output.
The realtime mixer also have effects EQ, Comp, Gate. This unit could be used without an external mixer (Which is the second senario we discussed earlier) ;)


This is the option I presented early on in this thread. As long as the OP gets the Presonus model with the 4 line outs on the rear (in addition to the Stereo outs for monitoring), then the built-in audio box software will allow the user to route any of the mic inputs to the various line outs, and yes, HPF, EQ, GR and some FX can be applied to those mics in the software mixing section, pre or post.
This is the scenario to use if there is a house PA resident. If not, you'd have to have a PA of your own.
This interface/pre can be used in any studio environment with any decent dynamic, condenser or ribbon mics, and on any DAW platform.
The pres are the Presonus XMax, same as in the Presonus SL desks. The converters are of a decent quality and would be fine for what you want to be able to do.
-d.

curbahn Fri, 08/18/2017 - 13:30
Help ! you're going to love this. I bought the presonus, the mixing desk and did the soldering on a 1/4 balanced jack to make the insert on the desk a direct out.. plugged everything in .. was dancing about with excitement at the fact it was all working and then suddenly i notice a hiss.. coming from the presonus. I quickly unplugged the 1/4 jack and the hiss there - always - on the presonus . Did I blow the preamp on the interface ? or was the hiss always there and I only just had the headphones loud enough to notice ?

farrrkk. so annoyed.

The solder job was like this: 1/4 balanced jack to 1/4 balanced jack. On one end I left all the wires as they were and joined the ring to the tip, and left the other end as normal. Then I plugged the soldered end into the mixing desk. Correct ? or is that the equation to blow up a pre amp ??

DonnyThompson Fri, 08/18/2017 - 15:11
I've never encountered hiss, or any noise coming from my VSL1818. I'm not saying it's not possible that your pre has an issue - but to echo Boswell, I don't think the soldering you did would have blown a pre.
You didn't send anything powered to the Presonus channel, did you?
Take the Alto mixer out of the chain.
Plug in a mic, preferably one that doesnt require a huge amount of gain. Use a GOOD condenser (or dynamic) if you have one. Plug in your phones to the Presonus - start at a minimum input gain, low headphone volume and then gradually increase both the input gain and the headphones -and listen... is the noise still there?

curbahn Sun, 08/20/2017 - 06:16
DonnyThompson, post: 452176, member: 46114 wrote: I've never encountered hiss, or any noise coming from my VSL1818. I'm not saying it's not possible that your pre has an issue - but to echo Boswell, I don't think the soldering you did would have blown a pre.
You didn't send anything powered to the Presonus channel, did you?
Take the Alto mixer out of the chain.
Plug in a mic, preferably one that doesnt require a huge amount of gain. Use a GOOD condenser (or dynamic) if you have one. Plug in your phones to the Presonus - start at a minimum input gain, low headphone volume and then gradually increase both the input gain and the headphones -and listen... is the noise still there?
yep the hiss is there , anywhere past about 2 o'clock on the knob and i can hear hiss.

DonnyThompson Mon, 08/21/2017 - 01:37
are you hearing this hiss while monitoring through speakers as well? Or just through the headphones connected to the device? I'm wondering if it may be the headphone amp in the Presonus that is faulty (?)
How hot are you running the headphone volume?
I seriously doubt you damaged any of the preamps.
How difficult would it be to return the Presonus you have and swap it out for another one?

Boswell Mon, 08/21/2017 - 02:44
curbahn, post: 452172, member: 50772 wrote: The solder job was like this: 1/4 balanced jack to 1/4 balanced jack. On one end I left all the wires as they were and joined the ring to the tip, and left the other end as normal. Then I plugged the soldered end into the mixing desk.
I just re-read this bit of one of your previous posts. If this is actually what you did, then your cable was TRS (balanced) jack plugs on both ends but now with the two signal conductors connected together. This would be correct for the pick-off at the insert jack on the mixer, but does not work for a balanced input jack on the Presonus audio interface. This is because they are differential inputs, i.e. they amplify the difference between the two signal conductors, which, if you have them connected together at the mixer end, is zero. The difference equation is never perfect, so you would still be able hear some signal, but you would have to turn the gain up a long way to hear it, far enough to hear the noise of the interface itself.

What I suggest you do as a trial is to take one of your modified cables and make a further modification. Where you have joined the tip and ring contacts, unsolder the cable wire that goes to the ring contact and instead solder it to the sleeve contact, i.e. where the cable screen is connected. Check that the tip and ring are still joined together and to the cable wire that goes to the tip connection. This means that, at the other end of the cable, the tip contact still has the signal but the ring contact now has ground, so the differential input sees the full signal.

curbahn Mon, 08/21/2017 - 08:48
DonnyThompson, post: 452207, member: 46114 wrote: are you hearing this hiss while monitoring through speakers as well? Or just through the headphones connected to the device? I'm wondering if it may be the headphone amp in the Presonus that is faulty (?)
How hot are you running the headphone volume?
I seriously doubt you damaged any of the preamps.
How difficult would it be to return the Presonus you have and swap it out for another one?

I've packaged it up and sent it back to amazon - hopefully they give me a full refund . Ordered teh Scarlett18i8 instead. I don't think it was the headphone amp as the hiss was visible in the spectral frequency display .

curbahn Mon, 08/21/2017 - 08:51
Boswell, post: 452210, member: 29034 wrote: I just re-read this bit of one of your previous posts. If this is actually what you did, then your cable was TRS (balanced) jack plugs on both ends but now with the two signal conductors connected together. This would be correct for the pick-off at the insert jack on the mixer, but does not work for a balanced input jack on the Presonus audio interface. This is because they are differential inputs, i.e. they amplify the difference between the two signal conductors, which, if you have them connected together at the mixer end, is zero. The difference equation is never perfect, so you would still be able hear some signal, but you would have to turn the gain up a long way to hear it, far enough to hear the noise of the interface itself.

What I suggest you do as a trial is to take one of your modified cables and make a further modification. Where you have joined the tip and ring contacts, unsolder the cable wire that goes to the ring contact and instead solder it to the sleeve contact, i.e. where the cable screen is connected. Check that the tip and ring are still joined together and to the cable wire that goes to the tip connection. This means that, at the other end of the cable, the tip contact still has the signal but the ring contact now has ground, so the differential input sees the full signal.

interesting ! tbh I'm terrified to try anything now. I'm going to wait for a friend to come help with the soldering . but will tell him what you've said as most online info about the 'simple soldering loop' trick says to have a stereo 1/4 jack on one end and a mono on the other , would that be similar to your instructions ?

Boswell Mon, 08/21/2017 - 09:58
curbahn, post: 452217, member: 50772 wrote: interesting ! tbh I'm terrified to try anything now. I'm going to wait for a friend to come help with the soldering . but will tell him what you've said as most online info about the 'simple soldering loop' trick says to have a stereo 1/4 jack on one end and a mono on the other , would that be similar to your instructions ?
Yes, exactly. The usual route to make these cables is to start with a TS (unbalanced/mono) jack cable or loom and replace the TS plugs on one end only with TRS (balanced/stereo) plugs that have tip and ring soldered together.

In fact, taking your route and starting with a TRS lead actually results in a cable that rejects external interference better. This is because the + and - signal leads inside the screen are routed side-by-side from the source (mixer) to the destination (interface), so interference that manages to penetrate the screen and affect both signal leads gets reduced owing to the nature of the differential input that I mentioned earlier. This can't happen with a single signal conductor.

curbahn Mon, 08/21/2017 - 10:08
Boswell, post: 452219, member: 29034 wrote:
In fact, taking your route and starting with a TRS lead actually results in a cable that rejects external interference better. This is because the + and - signal leads inside the screen are routed side-by-side from the source (mixer) to the destination (interface), so interference that manages to penetrate the screen and affect both signal leads gets reduced owing to the nature of the differential input that I mentioned earlier. This can't happen with a single signal conductor.

sooOOOoo it's better to use TRS balanced/stereo on both ends ? and like you suggest, (to just one end) connect the tip and ring but disconnect the original wire to the ring and instead, solder it to the sleeve / shelf (why does everything have multiple different names !!??) ?

Also, is there a way to test the cable before plugging it in (and blowing things up ) ?

Boswell Mon, 08/21/2017 - 15:35
curbahn, post: 452220, member: 50772 wrote: sooOOOoo it's better to use TRS balanced/stereo on both ends ? and like you suggest, (to just one end) connect the tip and ring but disconnect the original wire to the ring and instead, solder it to the sleeve / shelf (why does everything have multiple different names !!??) ? Also, is there a way to test the cable before plugging it in (and blowing things up ) ?
I doubt very much whether the difference between starting with a TS or a TRS cable would have any noticeable effect in your proposed usage. But since you have the TRS cables, use one of them for the trial involving moving the ring wire. When trying out the extra-modified cable, I forgot to warn you to start with the interface gain control for that channel at minimum. It's unlikely you will need to move it from that position.

Testing of the loose cable is easy if you have a multimeter or some other continuity tester. With one probe on the tip contact at the interface end plug. you should get continuity (a buzz or beep) when you touch the other probe on both the ring and tip of the mixer end plug, but not the sleeve of either end plug. Move the interface end probe to the ring contact, and it should show continuity to the sleeves of both plugs, but not the tip contacts. Done.

Boswell Tue, 08/29/2017 - 02:49
curbahn, post: 452217, member: 50772 wrote: I'm going to wait for a friend to come help with the soldering . but will tell him what you've said as most online info about the 'simple soldering loop' trick says to have a stereo 1/4 jack on one end and a mono on the other , would that be similar to your instructions ?
Did you ever get your friend in to help with the soldering?

Tags

x

Register