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Apogee Jam with a Mac... Why?

Nov 3, 2003
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Long story but I'll keep it brief...

I have a Mac Mini which has an 1/8" inch (3.5mm) audio input jack next to the headphone out jack. (I understand some of the newer Macs don't have this.)

To record my bass with GarageBand, all I need to do is is plug straight into my computer from my bass, no interface is necessary, just the right cable or adapter which costs about $6. The one minor thing is that in order to get my signal to an ideal level, I have to max out the gain slider in the audio preferences pane of my Mac. No big deal, but I thought the Apogee Jam would be an improvement over this method. It's also supposed to have a superior preamp and A/D converters, so I figured I'd give it a shot.

I bought an Apogee last week, and not only is it not an improvement, it's actually not as good. With the Jam, I have no control over the input level from the Mac, I can only control the gain on the Jam itself, and when setting that correctly, (to where the light just starts to flicker red and then backing off) my input signal is nowhere near as high as what I get using the other method.

I'm going to return the Jam, but before I do, can someone tell me why they use it and why I shouldn't return it? I'm wondering if I'm missing something here. I can understand the need for it if you want to plug into an iPhone or iPad, but for a desktop or laptop Mac it seems useless. Any thoughts?
 
You're going from a consumer grade product to an audio production tool. It implies spending a bit of time on it and mostly, getting used to the software.
The beauty of digital recording is infinite gain. All DAW programs, even the simplest ones have gain correction at all stages.
You want to set the JAM so it NEVER distorts and work from there to get comfortable levels with your software.
By all means, if you prefer what you get from the internal soundcard keep using it.
 
Here's an update: I contacted Apogee and they told me that if the signal isn't hot enough, I should turn up the gain on the Jam until the red light flickers. Because of it's built-in soft limiter, it won't clip unless I play with the light giving me a solid red light. I tried this and it improves things dramatically. I can now get a signal that is comparable in strength to the audi line-in method, though it's still not optimal. It is a bit strange though that this seems to go against the instructions written for the Jam.

As for the sound comparison between the two, upon testing them directly against each other, I find that the line-in method holds its own so long as the track remains dry. However, when adding compression, or especially amp simulation, the line-in track gets noisy very fast, while the Jam track stays clean. I've decided to keep the Jam for this reason.

That's how interfaces work.

As to the rest of your post: How are you monitoring yourself for a comparison between what the Jam is providing and what going into your soundcard is providing?

I don't think that's how all interfaces work. I've seen "how to" videos where the guy sets the input level on the interface, then sets the input level on the mac, and that translates to the right input level in GB--getting close to clipping without clipping. The reason I mention that you can't control input level is that, it seems to me, if using the Jam takes away your ability to set the level in the computer, then the ideal level on the Jam should translate to the ideal level on the computer. Otherwise, you're compromising the sound.

I'm using an old set of Sennheiser headphones for monitoring. I don't have computer monitors or speakers, but my headphones sound very good and I don't consider them a weakness in the chain.

You're going from a consumer grade product to an audio production tool. It implies spending a bit of time on it and mostly, getting used to the software.
The beauty of digital recording is infinite gain. All DAW programs, even the simplest ones have gain correction at all stages.
You want to set the JAM so it NEVER distorts and work from there to get comfortable levels with your software.
By all means, if you prefer what you get from the internal soundcard keep using it.

That's the thing, using the Jam doesn't allow me to adjust the recording level in GarageBand. I can adjust the track volume in relation to the master, but not the level of actual recording, so I imagine the Jam should be designed so that setting the level properly on the Jam gives me a proper recording level in GB. I guess I'm missing something.
 
That is how interfaces work. They take over the role of your soundcard. Your onboard controls are for your onboard soundcard, not an external interface. There might be some with tricky software that gives the impression of your onboard controls still affecting things, but that would just be in appearance. You can't have two sets of conversion happening at the same time. Apogee has built their software around the Maesteo platform for years.
 
That is how interfaces work. They take over the role of your soundcard. Your onboard controls are for your onboard soundcard, not an external interface. There might be some with tricky software that gives the impression of your onboard controls still affecting things, but that would just be in appearance. You can't have two sets of conversion happening at the same time. Apogee has built their software around the Maesteo platform for years.

Silky, I don't even know what a sound card is. I'm just telling you what I've seen. Here's an example:

Do you think my question isn't a valid one? Let's say we were were talking about outboard gear. I have a recording device that redlines at +4 db but has no way of adjusting gain. I buy a brand new preamp designed to work specifically with that recording device. Wouldn't you expect that preamp to redline at +4 db? I think that's a valid question.
 
You're confusing the issue. Your computer has a built in soundcard that handles AD and DA conversion. When you use an interface, you are no longer using the built in conversion, but rather that supplied by the interface. That's why you have to control things on the input side with the Apogee unit, which will also be the case with virtually any interface you use.

As to the video, which won't load for some reason; Anything you do in the box has absolutely nothing to do with the input signal level as it passes through the converters.
 
Silky, I don't even know what a sound card is. I'm just telling you what I've seen. Here's an example:

Do you think my question isn't a valid one? Let's say we were were talking about outboard gear. I have a recording device that redlines at +4 db but has no way of adjusting gain. I buy a brand new preamp designed to work specifically with that recording device. Wouldn't you expect that preamp to redline at +4 db? I think that's a valid question.

It is a valid question, and one that hints at one of the fundamental differences between analog and digital recording.

+4 dB on ANY analog device is going to be the same +4 db on EVERY analog device. It correlates to a real physical SPL that really exists and can only be measured one way.

With digital audio, where a given SPL registers on a given device's dB meter depends entirely on the properties of the AD/DA converters the manufacturers of the physical components have used, and the algorithms the programmers of the software for that device have used, which in turn depend on the algorithms the programmers of the operating system's drivers have used, etc. Decibels in digital audio are only a theoretical construct to make dealing with 0s and 1s easier for humans.

In practice, these levels are usually fairly close, with digital dB generally being a little bit quieter than "real" dB, IIRC (don't quote me on that - it might be the other way around). However, cheaper devices and software are less predictable in terms of where their dB scale lies in relation to the "real" dB scale.

This is why mastering is still an important part of digital recording.
 
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You're confusing the issue. Your computer has a built in soundcard that handles AD and DA conversion. When you use an interface, you are no longer using the built in conversion, but rather that supplied by the interface. That's why you have to control things on the input side with the Apogee unit, which will also be the case with virtually any interface you use.

As to the video, which won't load for some reason; Anything you do in the box has absolutely nothing to do with the input signal level as it passes through the converters.

I think you're misunderstanding the issue. I understand that the Jam is doing 2 things: #1 raising my gain to a certain level and #2 converting this signal from analog to digital. Those are two different things, yes? In a certain way, I don't really control the level with the Jam because the Jam is telling me what level to set it at. It tells me to raise the gain until the light turns red and then back off so the light stays green. That means Apogee has already determined a certain level that it wants me to use, (-10 db, 0 db, +4 db, whatever...) This is fixed. I can't tell the Jam to go higher without the light turning red.

After this conversion, once the signal is already in my computer, I should be able to control the level of the signal that is being recorded by my DAW, because that signal might be too low or too hot, right?. Why not? In this case, that signal is too low. Whatever level Apogee decided to make the Jam redline at is not hot enough to get an optimal signal in GarageBand. The manual for GarageBand states that with some interfaces, I cannot control the recording level with GarageBand. Apparently the Jam is one of these. It makes no sense.
 
I think you're misunderstanding the issue.

No, you're misunderstanding level management, specifically where it takes place. There's what happens before the signal passes through a converter, and what happens after it passes through a converter. Your Jam takes the place of your computer's soundcard, which is why you can no longer use the soundcard's (i.e. onboard) controls to handle the INPUT gain. At this stage it's recommended that you track with peaks hitting whatever the manufacturer recommends, but in general, prosumer level devices like to see peaks in the -12dbfs to -20dbfs range. This allows sufficient headroom to avoid overs while still providing a sufficient level to stay above the noise floor, which, in the age of digital recording, is VERY VERY VERY low.

Now, once that signal has passed through the converter and is living in the digital domain (frequently referred to as "in the box" or ITB), you can do whatever you want with it. All your Jam will now be handling the DA conversion to get all of the 1s and 0s (i.e. digital) into an analog format that can pump out of your speakers/headphones and into your ears. If you want to take that signal and amplify it, normalize it, lower the gain, adjust the dynamic range, expand it, compress it, etc. you can do so without the interface coming into the picture other than during the DA conversion where you want to hear what you're doing to the signal in the box.
 
It's been a while since I used garageband.

It doesn't allow you to control the individual track volume? (They might call it "instrument" volume). It doesn't have a gain plugin available on the track? I know you can control the master volume of the project.

Your headphones are plugged into the jam as well, right? Doesn't it have a headphone level control?
 
It doesn't allow you to control the individual track volume? (They might call it "instrument" volume). It doesn't have a gain plugin available on the track? I know you can control the master volume of the project.

It's been a while for me as well (to be honest, I've never really used it outside of one time on my wife's laptop for some VO work). There are individual track volume controls. To my knowledge, it does not come with a gain plugin unless it's built into one of the presets that seem to be the cornerstone of the program. I might be wrong about that. You can control the master volume, but I don't think it has dedicated controls for the master (i.e. system) versus the stereo output buss.
 
No, you're misunderstanding level management, specifically where it takes place. There's what happens before the signal passes through a converter, and what happens after it passes through a converter. Your Jam takes the place of your computer's soundcard, which is why you can no longer use the soundcard's (i.e. onboard) controls to handle the INPUT gain. At this stage it's recommended that you track with peaks hitting whatever the manufacturer recommends, but in general, prosumer level devices like to see peaks in the -12dbfs to -20dbfs range. This allows sufficient headroom to avoid overs while still providing a sufficient level to stay above the noise floor, which, in the age of digital recording, is VERY VERY VERY low.

Now, once that signal has passed through the converter and is living in the digital domain (frequently referred to as "in the box" or ITB), you can do whatever you want with it. All your Jam will now be handling the DA conversion to get all of the 1s and 0s (i.e. digital) into an analog format that can pump out of your speakers/headphones and into your ears. If you want to take that signal and amplify it, normalize it, lower the gain, adjust the dynamic range, expand it, compress it, etc. you can do so without the interface coming into the picture other than during the DA conversion where you want to hear what you're doing to the signal in the box.

I understand what you're saying, but you're essentially telling me that the Jam is not designed to give me an optimal recording level, even though it is designed for equipment I'm using. You're saying that once it's in the digital domain I can do whatever I want with it, but that's not the case. Once it is recorded, I can do what I want with it, yes, but prior to recording, in my computer, already in the digital domain, I can't adjust the recording level. I think I should be able to do that since I can adjust the master volume in GB and the volume for the headphones.

All of this would be a moot point if it sounded really good, but the weak input level makes the sound a bit thin compared to using the audio line-in jack. The only advantage so far is less hiss.

It's been a while since I used garageband.

It doesn't allow you to control the individual track volume? (They might call it "instrument" volume). It doesn't have a gain plugin available on the track? I know you can control the master volume of the project.

Your headphones are plugged into the jam as well, right? Doesn't it have a headphone level control?

With the Jam, GB allows me to control the individual track volume only AFTER the track has been recorded. So I don't have control over the track input level when recording other than how I set the gain on the Jam (and the Jam limits how high I should set the gain). When using the audio line-in method, instead of the Jam, I have full control over the recording level as well as the track volume after the fact. The Jam does not have a headphone out, but the problem I'm having isn't with being able to hear myself, it's with the Jam sending in a weak signal to be recorded.

It's been a while for me as well (to be honest, I've never really used it outside of one time on my wife's laptop for some VO work). There are individual track volume controls. To my knowledge, it does not come with a gain plugin unless it's built into one of the presets that seem to be the cornerstone of the program. I might be wrong about that. You can control the master volume, but I don't think it has dedicated controls for the master (i.e. system) versus the stereo output buss.
 
I think you're confused about what "optimal" recording level is.

Go ahead and try some experiments - make a recording that doesn't redlight your jam, make one with the gain on it cranked as high as it'll go, and make one using your audio in jack.

Now open all 3 in Audacity (Audacity download | SourceForge.net) Compare what their wave shapes look like and what the noise floor is. Run the Normalize function on all 3 and do the comparison again, noting not just how the 3 compare to each other, but how each one changes relative to itself before and after being normalized.

(In case you're curious, normalizing refers to amplifying the signal to the maximum possible level before digital clipping occurs, i.e. the loudest a computer can ever make that sound without compression.)
 
I understand what you're saying, but you're essentially telling me that the Jam is not designed to give me an optimal recording level, even though it is designed for equipment I'm using.
I'm not saying that at all. I'm just explaining how interfaces work and how varying levels of gain are applied to a signal both inside the box and outside of the box.

You're saying that once it's in the digital domain I can do whatever I want with it, but that's not the case.
Yes it is. Any inability to do so is simply because you haven't learned how at this point, or don't have software capable of performing the tasks you want.

Once it is recorded, I can do what I want with it, yes, but prior to recording, in my computer, already in the digital domain
Full stop. Prior to recording you are in the analog domain, not the digital one.

I can't adjust the recording level.
What software is paired with the Jam? Is it not Apogee's standard Maestro platform?

I think I should be able to do that since I can adjust the master volume in GB and the volume for the headphones.
The master fader controls the level of the output being applied to the DA conversion, not the AD. They are two completely separate functions. The master fader has exactly nothing to do with input gain. The same applies to the volume faders or gain plugins used on any of the individual track volumes or busses.

All of this would be a moot point if it sounded really good, but the weak input level makes the sound a bit thin compared to using the audio line-in jack. The only advantage so far is less hiss.
You keep saying "low volume" buy haven't indicated just how low. Is it somewhere in the ballpark of -18dbfs? If so it's not too low, it's at the right level for proper mixing technique. Don't be fooled by the appeal of trying to get everything as close to 0dbfs as possible. It's a rookie mistake that we're all, myself included, guilty of at one point in our learning process.
 
I think you're confused about what "optimal" recording level is.

Go ahead and try some experiments - make a recording that doesn't redlight your jam, make one with the gain on it cranked as high as it'll go, and make one using your audio in jack.

Now open all 3 in Audacity (Audacity download | SourceForge.net) Compare what their wave shapes look like and what the noise floor is. Run the Normalize function on all 3 and do the comparison again, noting not just how the 3 compare to each other, but how each one changes relative to itself before and after being normalized.

(In case you're curious, normalizing refers to amplifying the signal to the maximum possible level before digital clipping occurs, i.e. the loudest a computer can ever make that sound without compression.)

I'm going by the Garageband manual which says that, when recording, adjust the recording level to as high as possible without clipping. I'm not a recording engineer and I don't know how to use other software besides GB. I'm just going by what the manual says. Regardless of what the wave shapes look like, I prefer the tone from the audio-in jack, so I gave the Apogee to my friend today. It's his birthday and he's wanted one for a while.

Yes it is. Any inability to do so is simply because you haven't learned how at this point, or don't have software capable of performing the tasks you want.
My question was specifically about GB. GB doesn't let me do it.


Full stop. Prior to recording you are in the analog domain, not the digital one.

If my signal is registering in GB and I'm able to see the signal level when I play a note, isn't it already in the digital domain, even if I haven't pressed "record" yet?

What software is paired with the Jam? Is it not Apogee's standard Maestro platform?

No software comes with it. It's supposed to be plug and play with GarageBand and Logic. No drivers or anything to install.

The master fader controls the level of the output being applied to the DA conversion, not the AD. They are two completely separate functions. The master fader has exactly nothing to do with input gain. The same applies to the volume faders or gain plugins used on any of the individual track volumes or busses.

I get that. I suppose my question is, if you have a digital signal coming into and being registered by GB, why can't GB then further adjust that signal to taste? And if it just can't for whatever reason, why not make the Jam run hotter so that I can simply attenuate the signal to taste? There's another video where the guy sets the level on his interface but his guitar clips in GB, so he simply lowers the gain on his interface to a lower level. Makes sense, but you can't do that with the Jam.

You keep saying "low volume" buy haven't indicated just how low. Is it somewhere in the ballpark of -18dbfs? If so it's not too low, it's at the right level for proper mixing technique. Don't be fooled by the appeal of trying to get everything as close to 0dbfs as possible. It's a rookie mistake that we're all, myself included, guilty of at one point in our learning process.

GB doesn't give me a specific number, but the signal barely fills about 10% of the track info pane, whereas I can fill it at least halfway with the line-in jack, and videos I've seen state that halfway is 'about right'. The difference in volume and tone is substantial. As far as trying to get everything as close to 0 dbfs, I'm just using my ears for what sounds good and following the GB manual which says, "Adjust the recording level for the track so that the level as high as possible without causing clipping or distortion."

I'm giving the Jam to my buddy who can use it with his iPad and newer Macbook. The new ones don't have a dedicated audio-in jack so you need an interface in order to use headphones while recording. This thing isn't for me.
 
if you have a digital signal coming into and being registered by GB, why can't GB then further adjust that signal to taste?
Yes it can, this is what the individual track level, track plugins, and master volume are for.
And if it just can't for whatever reason, why not make the Jam run hotter so that I can simply attenuate the signal to taste?
You can do that too, just turn your gain up past the redlight point.

However, once the signal is clipped in the interface, there's no way to recover an unclipped signal - lowering the volume on the recording will not eliminate the artifacts. However, just like with amplifiers, whether or not a clipped signal sounds good to you is completely dependent on the characteristics of the preamp, and, also just like with amplifiers, digital clipping sounds like garbage. This is why you were instructed to keep the interface from redlighting.

From what I can gather, the only reason you like the sound of your audio in jack more than the sound of your interface is because it overdrives more pleasantly - this is no different than preferring an overdriven tube amp to an overdriven solid state amp. The difference is a tube amp can take the abuse. Keep overdriving that tiny 1/8th inch jack and eventually it will blow, and generally there's no way to replace them on mac products - take it from someone who's blown them up on 3 different laptops before I finally broke down and decided to learn what I was doing.
This thing isn't for me.
If you don't want to learn, then there's no point in trying to teach you.

This problem isn't going to go away with a different interface.

I strongly suggest you try the experiments I outlined so that you can visually see for yourself what's happening and correlate it to what your ears are hearing.
 
@callofcthulhu nailed it with his responses. A couple things I would add:

I'm going by the Garageband manual which says that, when recording, adjust the recording level to as high as possible without clipping.
Does the GarageBand manual really state that, or are you going by what you've heard online? I don't mean that in a snarky way. The whole "record as hot as possible without clipping" thing has been around forever, and it makes no sense in the modern world of digital recording. Shoot for -18 to -20dbfs on the peaks (some folks go up to -12 to -14dbfs, which is probably fine as well provided you don't have a super dynamic input signal).

My question was specifically about GB. GB doesn't let me do it.
A quick Google search provided me with over a full page of articles explaining how to increase waveform gain, so yes, GB does let you do it. Just like anything else software related, you have to take the time to learn how to do everything you want. Google is your friend in this regard. There's an insane amount of information out there for people in your position. Take advantage of it.

If my signal is registering in GB and I'm able to see the signal level when I play a note, isn't it already in the digital domain, even if I haven't pressed "record" yet?
Yes, but that's not the point you were making. You were talking about input signal levels which is a function of things happening in the analog domain, not the digital one. Just because the signal is passing the converter doesn't mean that adjustment made in the box will have an affect on things happening outside the box.

I get that. I suppose my question is, if you have a digital signal coming into and being registered by GB, why can't GB then further adjust that signal to taste?
You can, which is what I stated above and in my previous post. Those adjustments have nothing to do with the input signal and are used to resolve different issues, not the ones you're having. Moreover, once a signal is clipped, it's clipped. Lowering the volume/gain in the box will just make the clipped signal quieter. When you exceed 0dbfs, all of the 1s and 0s become 1s and you get harsh, nasty, unpleasant digital distortion. There's no reason to track that hot in the digital domain. You can pleasantly clip a preamp in the analog domain and then use something to attenuate it prior to crossing the converter, but that's a very different thing.

why not make the Jam run hotter so that I can simply attenuate the signal to taste?
That works fine provided you're not clipping. That said, if you're hitting -18 to -20dbfs, why bother? You're already at the optimal level for mixing.

There's another video where the guy sets the level on his interface but his guitar clips in GB, so he simply lowers the gain on his interface to a lower level. Makes sense, but you can't do that with the Jam.
Why can't you do that with the Jam? There's an gain control on the side of the unit. Reduce the level and track the part again. You shouldn't be tracking so hot that you're exceeding 0dbfs (i.e. clipping) anyway, but I understand that accidents do happen.

GB doesn't give me a specific number, but the signal barely fills about 10% of the track info pane, whereas I can fill it at least halfway with the line-in jack, and videos I've seen state that halfway is 'about right'
Don't get caught up in what the waveforms look like. The notion of halfway being "about right" is silly. The noise floor in the digital world is so low that it's a complete non-issue. Trust me; Record at -18 to -20dbfs and you'll have a much better time.

The difference in volume and tone is substantial.
Volume, yes. Tone, no. It's digital. Changing volume only changes volume, it does not have ANY affect on tone outside of things like Fletcher-Munsen curves.
 
Yes it can, this is what the individual track level, track plugins, and master volume are for.

You can do that too, just turn your gain up past the redlight point.

However, once the signal is clipped in the interface, there's no way to recover an unclipped signal - lowering the volume on the recording will not eliminate the artifacts. However, just like with amplifiers, whether or not a clipped signal sounds good to you is completely dependent on the characteristics of the preamp, and, also just like with amplifiers, digital clipping sounds like garbage. This is why you were instructed to keep the interface from redlighting.

From what I can gather, the only reason you like the sound of your audio in jack more than the sound of your interface is because it overdrives more pleasantly - this is no different than preferring an overdriven tube amp to an overdriven solid state amp. The difference is a tube amp can take the abuse. Keep overdriving that tiny 1/8th inch jack and eventually it will blow, and generally there's no way to replace them on mac products - take it from someone who's blown them up on 3 different laptops before I finally broke down and decided to learn what I was doing.

If you don't want to learn, then there's no point in trying to teach you.

This problem isn't going to go away with a different interface.

I strongly suggest you try the experiments I outlined so that you can visually see for yourself what's happening and correlate it to what your ears are hearing.

Again, the track level slider is deactivated when I use the Jam. I can use it when going into the audio line-in jack, but not when recording with the the Jam. Adjusting with the track plugins or master volume only effects the track volume after recording. They do not effect the actual recording level. I've said this 3 or 4 times now.

I never said anything about overdriving either the Jam or with the other method. I am not clipping anything anywhere and I don't understand why you bring that into the discussion or how you got that impression.