Is it just a problem of the arrangement, or a lousy mix? Well, maybe, but I think there is a more fundamental problem too: not enough dynamic range.

Yes, this is another post about the Loudness Wars!

(Looking to hate on Death Magnetic? Find jaw-dropping comparison audio files right over here in Metallica Wins the Loudness Wars — and by the way, for those wondering “if Metallica ‘won,’ then who lost,” the answer is: “everybody with ears.”)

You can see when a mix has this problem

Thanks to digital editors, audio has become a visual medium. Check out this comparison of the original RUSH track and the remake. If you know anything about dynamic range, you’ll spot immediately which one is which.

If you don’t know anything about dynamic range, first of all, thanks for sticking with me, and second of all, the nutshell explanation is that the remixed track is at damn near 100% maximum volume for the whole song. See the top waveform, in blue.

If even the quiet instruments and passages are at maximum volume, there’s no way for the loud stuff to stand out or have any impact. The difference between the loudest thing and the softest thing is called “dynamic range,” and without it, music sounds like crap.

… and you can hear it, too

Now let’s listen. Here is a comparison of the two tracks at equal RMS gain. Play this back at whatever level is comfortable, and notice which one sounds better.

 Original RUSH recording

 Remake/tribute recording

The mixes are different, yes. But the remake mix is much less effective, I think, due to the heavy-handed use of compression. All the life has been squeezed out of it.

“Sounding better” is a subjective thing. If you love loud guitars and keyboards, you might even prefer the remake mix, which is !#&$!@& drowning in both. However, one of the ways great arrangements convey emotion is by manipulating intensity… and if you’re at 100% for the whole song, you have nowhere left to go. This is what makes the remake mix less effective.

Dynamic Range, by the numbers

I used the “TT Dynamic Range Meter” plug-in* for Pro Tools to monitor the dynamic range of these two mixes. The remake clip above hovers steadily around 7-8dB of dynamic range. You can see the red bars in the image at right — an indication that, at least according to the people behind the plug-in, the remake mix has been overcompressed.

The original averages 12dB of dynamic range, but bounces quite a bit between 11–13dB. Even the fact that its level isn’t constant is a good sign; it means the track has room to breathe. There is space in the mix for the drums to have impact. The mix isn’t completely saturated.

To be fair, this remake of “Subdivisions” really isn’t that great an example of the overuse of compression in modern music. The mix is not nearly as crushed as some others, like the Death Magnetic CD (DR = 2.5dB).

There is a world of difference between 2.5dB and 7.5dB of dynamic range — and I’d take the 7.5dB any day. The smaller that number gets, the worse the music sounds, to the point where compression introduces distortion that literally wears out your ears as you listen.

But despite the other sins committed within this remake mix, I think it would have sounded better with more dynamic range. Because if your goal is preserving the DR while you mix and master, it means you’re no longer compromising the song’s emotional impact just to make it as loud as possible.

Learn More

Learn more about Dynamic Range Day. Huge kudos to Ian Shepherd for his work on this issue! Be sure to check out Ian’s wonderful post on How the Loudness War is crushing Dave Grohl’s analogue ambitions, as it has more great examples of how modern music doesn’t sound as good as it ought to.

See also Ian’s post on the TT Dynamic Range Meter, and how to get your own copy.

The MCA SP1 is a $50 condenser mic based on MXL’s version of the Schoeps transformerless circuit. The circuit design is capable of greatness; it is very well understood, and easy to improve. Numerous technicians have offered commercial modifications to this mic over the years, and today you can buy a mod kit for the SP-1 — a set of handpicked replacement components that transforms the mic into an exceptional studio tool. [Full disclosure: the kit was my idea.]

But the days of modding new SP1s are over. Marshall Electronics has apparently swapped the venerable “V57″ circuit board set for a single surface-mount PCB. While it is technically possible to modify an SMT board, it is not worth the effort and risk of drilling out all the components you’d want to swap. For example, the mod kit referenced above changes 21 different components.

For years, I have been recommending that would-be DIYers buy the ~$50 SP1 for modding. But now there is a distinct risk of getting an SMT mic that you’d have to return. I can’t recommend buying SP1s new any longer. Instead, pick up an older SP1 on Ebay. Consider the following mics equally desirable, for they all use the V57 PCB set: MXL 2006, MXL V63M, MXL V250. But ask the seller to send a photo of the circuit board first, just to be sure.

See the original V57 PCBs at right. This is what you want to see in your SP1 if your intention is to modify it.

If the Ebay route doesn’t work, I recommend buying an MXL 990 instead. The MXL 990 uses the same circuit and capsule as the SP1, and is actually easier to modify. (And yes, there is a mod kit for this one too.)

Update! It turns out the MXL 990 has gone SMT too, and likely the entire product line. We’ve heard reports of SMT in the 770 and 990S as well. The only bright light in this dark development is that it has pushed several independent DIY providers to supply retrofit PCB kits for these microphones; watch for an announcement on Microphone-Parts.com soon regarding an enhanced MXL 990 PCB.

[SMT SP1 PCB photo credit: David Hamon]

Audio-Technica ATR2100-USB

Several readers asked for a direct comparison of these two mics, which differ in price despite sharing specs, capsules, and features.

So, we’ve tested these mics head-to-head, recording both simultaneously. Also we’ve compared the analog and digital outputs to see how well the onboard amplifier and ADC compares to outboard gear.

Session 1: ATR2100 USB, Analog vs Digital

My first task was to get both of the ATR2100-USB’s outputs to show up in Pro Tools simultaneously. I did this via Pro Tools’ “Aggregate I/O” feature.

I couldn’t find a way to aggregate my Digi 002 Rack along with the ATR2100’s USB output. But I was able to aggregate my Mackie Onyx Blackjack (a USB mic pre) with the ATR2100. See the setup at right.

In Pro Tools, I set the “Playback Engine” to this aggregate I/O virtual device. The Blackjack’s 2 inputs showed up as the first two inputs, while the ATR2100 showed up in #3.

I put the Blackjack’s analog gain control at about 5:00, just shy of maxed out. Then I used the Sound pref pane to set the USB input’s gain level so that both channels were peaking in the same place in Pro Tools. I got them within 0.5dB of one another.

Note that the USB output is a 16-bit device, whereas the Blackjack is a 24-bit device. You can expect to hear a difference in the noise floor of these two channels.

Sound Samples: ATR2100, Analog vs Digital

 ATR2100, via 16-bit USB

 ATR2100, via analog output

Listening Notes

The sound quality of these two tracks is very similar. There is more noise in the USB track, as expected. My RTA shows about 5dB more noise above 5kHz, in the USB track. Both tracks are picking up some room noise (my computer fan, mostly), but the USB track has additional hiss above 1kHz.

In short, the sound quality of the USB output is decent, and in my opinion the mic in USB mode would work well for podcasting or video voiceover (for screencasts and so on). The actual tone and response of the mic in USB mode is very impressive for the price.

With an analog preamp (and external ADC/audio interface), this mic sounds even better; the high-frequency hiss of the mic’s onboard 16-bit USB ADC goes away. (The room noise is somewhat distracting, but is representative of recording with a desktop computer in a room with a hard floor. I prefer recording to a laptop whose fan is suppressed, but my laptop was unavailable for this session.)

Surprising anomaly – polarity inversion of digital output

Weirdly, the digital output has inverted polarity as compared to the analog output. The mic’s analog output appears to have the same polarity as my other mics, suggesting that the digital output is “out of phase.”

(I inverted one of the tracks prior to exporting the audio samples linked above.)

Session 2: AT2100-USB vs AT2005USB

For this test, I plugged in both Audio-Technica dynamic USB mics via their included USB cables, and created an aggregate audio device that included both. I recorded to a 24-bit Pro Tools session (although the output of both mics was 16 bit).

Both microphones are sensitive to plosives at my working distance of ~4 inches. I tend not to criticize microphones for plosive performance, for two reasons:

1. P-pops and other plosives can be prevented via mic position, speaking technique, and the use of external blast filters.
2. P-pops and other plosives can be removed from an audio track fairly seamlessly by applying a steep low-frequency rolloff, e.g. 12–18dB/octave below 90–150Hz (depending on the frequency range of the voice).

… nonetheless, I mention these mics’ plosive performance here because if you are considering buying one of these you might consider buying a pop filter too. (See the Conclusion section for compatible foam windscreens from Audio-Technica.)

For this particular comparison session, I opted to mount an external pop filter (a dual layer mesh unit from Sabra Som), rather than reposition the mics or clean up plosives after tracking.

My working distance was about 4 inches, and I tracked both mics simultaneously, which means you’re hearing exactly the same “performance” in every pair of tracks below.

(By the way, anyone recording voice — their own or someone else’s — should go read Randy Coppinger’s 10 Voice Recording Essentials right now.)

Sound Samples: AT2100-USB vs AT2005USB

Narration:
 AT2005USB, via 16-bit USB

 ATR2100-USB, via 16-bit USB

Proximity:
 AT2005USB, via 16-bit USB

 ATR2100-USB, via 16-bit USB

Listening Notes

The differences between these two mics are subtle. You could punch in one for the other and nobody would notice.

I occasionally hear a slightly fuller low-frequency response from the ATR2100-USB. But perhaps I’d swayed a half-inch closer to that mic while speaking, causing a boost in proximity effect? I’ll say it again: I think the differences are so subtle as to be negligible. If you are trying to decide between these two microphones, shop based on features and price, not for sonic distinctions.

If you’re wondering why the noise level is so high in the first part of the two proximity test tracks, it is because the signal level (that is, the volume of my voice) is much lower when the I stand farther away from the mic. The three distances in each of those tracks are presented at equal RMS gain, which means the distant tracks had to be turned up much louder… resulting in amplification of the noise floor, too.

More Comparisons

So long as we’re here, I recorded a few more head-to-head comparisons so you can hear the ATR2100-USB against a couple other microphones you should be considering.

ATR2100-USB vs Shure SM57

 Shure SM57

 ATR2100-USB, via analog output

These two mics sound pretty similar on my voice. The ATR2100-USB is fuller in the low mids. Both mics need external pop filters (which is no fault of the SM57; to be fair, I should have used an SM58). As mentioned previously, I think the built-in windscreen of the ATR2100-USB is inadequate.

ATR2100-USB vs Studio Projects LSM

 Studio Projects LSM, via USB

 ATR2100-USB, via USB

This is a really interesting contrast. The LSM is much more restrained on the high end, but sounds a bit eshy. I wouldn’t have expected the dynamic mic to sound brighter than the condenser, but here it does. And I think the ATR2100-USB is giving a more accurate representation of my voice.

That’s not to say the LSM doesn’t sound good. It is warmer than the dynamic mic, and while arguably not as accurate on top, is likely to be easier to listen to over long periods.

Be aware the LSM is a condenser mic. In general, its high-sensitivity capsule is more likely to pick up mouth sounds and ambient noises from your room than would most dynamic mics. If you’re recording in a noisy space, you’ll be better served by a dynamic mic. But the LSM remains a personal favorite; it won a blind test of inexpensive condensers for podcasting last year, and works well on many acoustic and electric sources.

These two tracks were recorded through a two-layer external pop filter, because both of these mics are sensitive to plosives.

ATR2100-USB vs Rode Podcaster

 RODE Podcaster, via USB

 ATR2100-USB, via USB

This is another very revealing comparison: listen to the difference in low-frequency response! The Podcaster sounds high-passed. But now I understand why: it sounds more like a radio voice, pre-EQ’d.

This comparison is a bit unfair to the Podcaster, because I stood 4–5 inches from the mic. I know from previous tests that the Podcaster sounds more balanced at closer working distances.

Which of these you prefer is entirely subjective. The Podcaster sounds more polished, or arguably thin and scooped. The ATR2100-USB sounds fuller and warmer, or arguably a bit wooly. For low-pitched voices, the ATR2100-USB might want a bit more distance, whereas the Podcaster wants less.

Note too that the Podcaster’s street price is $229, or about 4.5 times more than the ATR2100-USB.

Conclusions

I stand by my earlier finding, that the ATR2100-USB and AT2005USB are great-sounding mics for the price. If you’re shopping for an inexpensive podcasting solution, both of these mics should be on your shortlist. Thanks to their dual-output design, both mics will retain their utility when you expand your podcasting rig to include an external preamp/interface.

Both these mics need better blast filtering. They don’t come with foam windscreens, but they should. The following Audio-Technica windscreens are compatible:

• For both AT2005USB and ATR2100-USB: the AT8112 windscreen is compatible
• For the ATR2100-USB: the AT8114 is also compatible

Kudos to Audio-Technica for introducing useful audio tools at these remarkable price points!

Credits and Disclosures

The Rode Podcaster and both Audio-Technica microphones were loaned to me by their respective manufacturers for the purposes of this review. The LSM, SM57, Onyx Blackjack, and other gear mentioned herein are personal property. We received no compensation from Mackie, Shure, Studio Projects, Rode, or Audio-Technica for this review.

Maybe it’s just because of old studio photos of Frank Sinatra.

The explanation I’ve heard, probably from some totally reliable source like one of the audio forums, is that microphone capsules will last longer if they’re not continually baked by the heat rising from the tube. Hang the mic capsule-end-down, and the capsule won’t be in the in the proverbial line of fire. It’s an easy thing to do, and if it was good enough for Sinatra’s U47, it ought to be good enough for all of the cheap tube mics I own that “sound just like the U47.”

The problem is, this explanation doesn’t really make any sense. The capsule deck in every tube mic I’ve seen is sealed off from the chamber where the tube is. Plus, the headbasket is vented to the outside air. Does the capsule really get any warmer due to tube heat?

I don’t believe so, but as they used to say in Chicago, that and a dollar will get you on the “L.”

There’s one other factor worth considering: the electrolytic capacitors found in most audio gear have a useful life span that is inversely proportional to operating temperature. The hotter they run, the sooner they die. So, maybe we shouldn’t be asking so much about the capsule temperature as the PCB temperature.

(Who am I kidding? Of course we’re going to ask about both. But jump to the TL;DR (summary) if you’re not up for a nerdy elaboration on microphone component temperature.)

Tube Mic Temperature Test

I hung a temperature probe in an Apex 460. The sensor was laying alongside the PCB. I let the mic warm up for several hours in each position.

PCB Temperature – with mic inverted (tube above PCB)

Ambient temperature: 72.5° F
PCB temperature:     90.0° F
Delta:               17.5° F

PCB Temperature – with capsule at top (tube below PCB)

Ambient temperature: 70.9° F
PCB temperature:     96.1° F
Delta:               25.2° F

This answers one question: Inverting the Apex 460 keeps the PCB ~8° F cooler. Does it matter? More on that shortly.

What about the capsule? I didn’t expect much of a delta, for the reasons stated previously. The capsule deck has small holes drilled into it to allow passage of the capsule leads to the PCB. Otherwise, the deck seems like a pretty effective firewall. Nonetheless, let’s test and see:

Capsule Temperature – with mic inverted

Ambient temperature: 68° F
Capsule temperature: 78.4° F
Delta:               10.4° F

Capsule Temperature – with capsule at top

Ambient temperature: 67.3° F
Capsule temperature: 82.4° F
Delta:               15.1° F

Inverting the Apex 460 keeps the capsule ~5°F cooler.

Will these results be consistent for every tube mic? No, of course not — it depends on the heat output of the tube, the presence of ventilation ports, and the interior structure of the mic’s circuitry and chassis.

Regarding the last point: some mics put the tube between the capsule and the PCB. With such mics you have to choose whether to cook the circuit or the capsule.

The experts weigh in

I asked some microphone designers for their thoughts on whether tube mics should be used upside down.

David Royer

With the Mojave microphones, I don’t favor one position over the other… the 5840 tube runs relatively cool and because of the way it is installed in the microphone, heat related problems are reduced.

Dave went on to analyze the power output of various tube models — and to suggest a much more practical explanation for why Frank’s U47 is usually pictured upside-down:

With the Mojave microphones, the power being dissipated inside the microphone is just under a watt.

With a microphone using a 12AX7 or similar dual triode, with both halves of it in operation, the power dissipation would be just under two watts.

As for the Neumann U 47, the VF 14’s heater and its dropping resistor … dissipated 4 watts between them. With the U-47, it made sense to have them upside down.

But I think that an even more important reason for U-47’s being used upside down was to keep them out of the way, and well out of range of a singer turning the pages of a score.

I also asked Dr. Charles Chen, the lead microphone designer at Lauten Audio.

Charles Chen

The tube will heat the air; as the hot air rises, it will increase the temperature of the air [surrounding] the capsule if the capsule is above the tube. Theoretically this will change the tension of the diaphragm a little bit, and therefore its frequency response. But the slight frequency change due to the temperature change might not be detected, and even if it is noticeable, we cannot say whether it is good or bad.

Dr. Chen noted that inverting the mic will change the recorded signal due to changes in reflection and refraction of the incoming audio waves. But, as with the diaphragm tension, we can’t predict whether this would generally be perceived as a beneficial or detrimental change.

As if reading my mind, he also mentioned the issue of capacitor temperature:

The life of electrolytic capacitors, compared to the other components on the PCB, is most sensitive to high temperatures.

But how sensitive?

The lifespan of electrolytic capacitors

We’ve come this far. We might as well get our geek fully on. We already know that the lifespan of electrolytic capacitors is determined significantly by operating temperature. Arrhenius’ Law of Chemical Activity suggests that the lifespan of a capacitor doubles for every 10 degree Celsius decrease in temperature.

For the Apex 460 specifically: the PCB measures 8°F cooler when the mic is inverted. According to the capacitor lifespan calculator linked above, there is a nontrivial extension of usable life due to this 8°F drop when the mic is inverted — something like 20,000 hours more, given a 4°C drop in operating temperature.

That said, even at a higher temperature, the capacitor for which I ran the equation will live 60,000 hours. That gives you more than 20 years of daily 8-hour sessions before you have to find a replacement capacitor.

TL;DR

Should you worry about the orientation of your tube mic?

As in so many cases, the answer is: it depends. But in general, probably not.

If your mic uses a subminiature tube, as found in the Mojave mics, orientation probably makes no difference to lifespan, because the tube isn’t putting out enough heat to matter.

If your mic uses a hotter tube, and the tube is not isolated from the PCB or capsule, then orientation might help the mic last longer. But you won’t live long enough to notice.

Unless you keep your mics powered up 24×7, I think the mic’s orientation makes no practical difference to lifespan.

Regarding the capsule, I think it is even less likely to be affected by tube heat. The test results show a ~5°F delta depending on the mic’s orientation. I can’t prove it, but I suspect one “spitty” singer does more damage to your mic’s capsule than a 5°F temperature change.

In summary: although your mic’s components might theoretically live longer if you orient them so the tube heat goes in the other direction, those same components will probably already outlive your recording career. Therefore, you should mount the microphone so that the artist is unimpeded in delivering an inspired, energetic performance. This outweighs technical considerations.

Broadcast dynamic mics aren’t necessarily friendly to beginners, though. They tend to have low sensitivity levels, which means they require preamps with a lot of clean gain. That problem was precisely the reason we tested all the leading USB mic pre’s with the Shure SM7B in our Budget USB audio interface shootout. We reasoned that most podcasters can’t afford a high-end analog preamp and separate analog-digital converter, and would benefit from knowing that there are affordable USB audio interfaces that have enough gain to drive a low-output dynamic. (For the record, we made a couple great recommendations in that piece, so if you’re shopping for an inexpensive signal chain for your broadcast dynamic mic, click that link to find our picks.)

Recently, we’ve seen a couple intriguing new podcast mics come on the market: dynamic mics with onboard amplifiers, ADCs, and USB output. This means podcasters can plug these mics directly into their computers and get all the benefits of a dynamic voice mic with none of the hassle of needing a separate preamp and converter.

How do they compare to the industry standards? Let’s find out.

The Contenders

We tested three moving-coil USB dynamics, plus a hybrid USB/analog condenser that shined in our shootout of $200 condensers for podcasting, the Studio Projects LSM.

To that mix, we added two high-end broadcast dynamics. These are not USB mics, so we used the Mackie Onyx Blackjack, an affordable 24-bit mic pre and audio interface that performed well in our Budget USB audio interface shootout. Both these analog dynamics were used with the Cloud Microphones “Cloudlifter” (see my review of the Cloudlifter CL-1 mic activator.)

Model	Podcaster	ATR2100-USB	AT2005USB	LSM	SM7B	M-99
Type	USB Dynamic	USB/Analog Dynamic	USB/Analog Dynamic	USB/Analog Condenser	Analog Dynamic	Analog Dynamic
Street	$229	$49	$89	$149	$340	$459
Bit Depth	18	16	16	16	n/a	n/a
Output	USB	USB/XLR	USB/XLR	USB/XLR	XLR	XLR
Headphone Jack	Yes	Yes	Yes	No	No	No
Shockmount	No	No	No	No	No	Yes
Desk Stand	No	Yes	Yes	Integral	No	No

This audio test is completely unfair: the samples below include USB mics ranging in price from $50-$230, against a 24-bit signal chain worth over $750. Do you think you’ll hear a difference?

Optimizing input gain for USB mics

It is worth noting that the input level for all these USB mics is adjustable via software, at least under OSX. If you set the input level too high, you’ll clip the mic’s signal. Too low, and you’ll have a very noisy track when you apply digital gain or compression in your DAW. For best results, set up the mic in the position you’ll use it, and speak into the mic as you will when tracking. Set the input gain slider (found in System Preferences, Sound pref pane on OSX) so the input level peaks at the second-last bar. If the rightmost bar lights up, you risk clipping. If the mic has 16-bit output, you’ll want as hot a level as you can get without clipping, to maximize the signal to noise ratio.

This step will make an audible difference in the noise floor of a 16-bit USB audio track. I’ve tested it, and confirmed the results.

Audio Samples

All mics were recorded directly into Pro Tools, then RMS gain-matched, exported at 24/44.1, fine-tuned and converted to 320 kbps MP3 via Sample Manager.

Listen blind to the samples below, then click the button at the bottom to reveal which mic is which.

(Find the 24-bit WAV archive here [ZIP, 11MB].)

In all cases, I recorded at a distance that seemed best for the mic. For example, the LSM sounded muddy and dull at 2 inches, so I backed off to about 5 inches. Conversely, the Podcaster sounded thin and strident if I wasn’t right on top of the mic, so I recorded the Podcaster track at a distance of about an inch.

The SM7B’s tone switches were all set flat. On the M-99, I’d enabled the “presence enhance” EQ circuit, which boosts the highs to compensate for proximity when worked close.

Proximity Test

As noted above, each of these mics prefers its own working distance. A mic’s “proximity effect” is the amount of bass boost it produces as you move the mic closer to the sound source. Some mic designs, like the SM7B, keep the capsule far enough back from the grille that proximity effect is reduced. Other mics, like the Podcaster and M-99, change tonality significantly as you move closer or farther.

Proximity effect is a tool, not a fault. Voice artists and broadcasters develop their technique to use proximity effect to their advantage. Amateur speakers and podcasters need to proceed with caution, though; until you have developed your technique, the tone of your speech track will change if you move around the microphone during the podcast… unless you pick a mic with minimal proximity effect, like the SM7B or Electro-Voice RE20.

Compare the proximity effect for these microphones

Audio-Technica AT2005USB

Audio-Technica ATR2100-USB

beyerdynamic M 99

RØDE Podcaster

Shure SM7B

Studio Projects LSM

Listening Notes

I conducted my own blind listening test with the 24-bit WAV audio tracks. (To be clear, none of the USB mics has 24-bit output, but I recorded them all into a 24-bit Pro Tools session, so my raw audio had been upsampled to 24 bit.) Several of the USB mics stood out immediately because their level of self-noise is higher. Specifically, I heard more hiss from the two Audio-Technica mics and the LSM. I didn’t find the noise level on these tracks to be offensive, though.

There is a slight 100Hz hum audible in a few of these tracks. This was environmental noise, not a fault of any of the microphones. See my recent story on how to fix USB microphone hum for the explanation.

Blind listening results

In a blind test, I picked my favorite VO mic, the beyerdynamic M-99 — and not simply because I recognized it. (Truth be told, I thought at the time it was the SM7B.) It sounded the most natural of all of these, with good articulation and no hint of sibilance. And I love the low noise floor (which, granted, is due in part to the Cloudlifter and Onyx Blackjack). To my ear, this track conveys authority.

The SM7B sounds good here; it is my second-favorite of the bunch. It has a flatter midrange than the M-99, and is more accurate for it if perhaps not as flattering (IMO) as the scooped sound of the M-99.

The Podcaster sounds pushed in the upper mids. My voice sounds hard and nasal through this mic. The highs have good detail, but the track lacks warmth. When I compare it to the other mics, the Podcaster track sounds bandpassed, or aggressively EQ’d in a way that is unflattering to my voice (or, perhaps, to my ear).

That said, I must point out that the Podcaster has the second-lowest self-noise of all the mics in this test. It doesn’t achieve the deep, black silence of the M-99, but it comes close. It has lower self-noise than the SM7B, in fact. Kudos to Rode for building a very low-noise amplifier and ADC circuit into this mic.

The LSM sounded really nice here. I can hear, again, why I picked it as a favorite in our Best $200 Podcasting Condenser Mic Shootout. It’s not a neutral sound, but its color seems to work on my voice. It isn’t as smooth in the mids as the M-99, and owing to the sensitivity of the condenser capsule, the LSM picks up a bit more mouth noise than is ideal — which of course is one of the reasons I prefer dynamics for this application. Nonetheless, the LSM turned in a very usable track.

The two Audio-Technica mics impressed me. For the price, both sound great. I think neither A-T mic provides as high fidelity a track as the pricier analog mics here. In comparison to the M-99, both A-T mics sound a bit veiled, a bit less detailed. But I think they’d have sounded better — less veiled, more open — if I’d backed off the mic slightly. But even so, as I say, for the price these mics seem difficult to beat.

Conclusion

Spend more, get more

Let’s get the obvious but arguably painful conclusion out of the way first: my $460 analog mic, through a $300 24-bit signal chain, sounds better than anything else here. But it should, at 3-16x the price of the USB mics in this test. If your budget allows for more than the cost of these USB mics, head over to our Editor’s Choice Podcasting Gear listing to find the best sound for your budget.

Versatility

Consider too what you’ll use the mic for. If the only thing you’ll ever record is your own voice, then you should focus on a mic that shines for that application. If you’ll record instruments too, or if you’ll use the mic on stage, then you should consider these factors.

The LSM, AT2005, and ATR2100 all have analog as well as USB outputs. This means you can use them on stage or with an outboard preamp/interface. I think the LSM is especially versatile because (as we heard in the Best $200 Condenser series) it sounds really good on a lot of sources.

All three USB dynamics (but not the LSM) have headphone jacks and gain controls, enabling you to monitor your own voice as you record. The other mics in this test would need to be monitored through your DAW.

Accessories

All these USB mics come with USB cables; the A-T mics and the LSM include analog XLR cables too. The A-T mics also have tripod desk stands in the box. None of the USB mics come with shockmounts. In the case of the Rode Podcaster, the PSM1 shockmount is available separately for about $40, and I’d recommend it given my experience with the mic’s sensitivity to mechanical vibration.

Recommendations

I am very pleasantly surprised by the sound quality of the Audio-Technica mics in this test. At street prices of $50–$90, these are unbeatable for a low-cost podcasting solution.

What’s the difference between these mics? Their specs are identical. Their accessories are identical. Their cartridges appear to be identical. One lists for $149, with a 1-year warranty, while the other lists for $80, with a lifetime warranty. (Which, yes, seems backwards.) The pricier mic has a nicer finish, especially on the grille; the ATR2100 USB has a gray-painted grille that to me looks less professional than the AT2005.

When I cut back and forth between these two tracks, I found moments in each that I preferred over the other. These are performance differences. I think the sound of these two mics is essentially interchangeable.

Update! I tested these two Audio-Technica mics head-to-head, and posted comparison audio: mic vs mic, analog vs USB outputs, and additional ATR2100-USB comparisons: SM57, Rode Podcaster, Studio Projects LSM. See ATR2100-USB and AT2005USB, A Closer Look.

And that’s a fairly amazing piece of news. Because it means I can recommend the ~$50 (street price) ATR2100USB as the best entry-level podcasting mic on the market. The mic’s sound quality is fantastic for this price point. And I think new podcasters will sound better through this mic than through any of the condenser mics being pitched to the podcasting market.

The Studio Projects LSM, while not a dynamic, is a contender for a workhorse mic role in any studio. It’s a really solid little mic, with a usefully compact form factor. I continue to recommend this mic for just about any home studio application.

The Rode Podcaster impressed me for its high sensitivity and very low noise circuitry. The sound of the mic didn’t flatter my voice, but I can imagine voices on which the mic would work better.

The Podcaster does have a pronounced EQ curve, which I can hear even in the RODE Soundbooth application, but it doesn’t sound bad there. If your budget has room for the Podcaster, I recommend trying it on your own voice to see if it’s a fit.

Don’t forget proximity

Remember, whatever mic you end up with will favor a particular working distance for your voice. Try the mic at a range of distances, from right on it to eight or ten inches back. Find the distance that sounds best to you.

Credits and Disclosures

The narration track is an excerpt from Mixerman’s wonderful book, Zen and the Art of Mixing.

The Rode Podcaster and both Audio-Technica microphones were loaned to me by their respective manufacturers for the purposes of this review. The LSM, SM7B, and M-99 are personal property, as are the Cloudlifter and Onyx Blackjack. We received no compensation from beyerdynamic, Mackie, Cloud, Shure, Studio Projects, Rode, or Audio-Technica for this review.

Comments?

I’d love to hear your experiences with podcasting mics, especially if you’ve used more than one. Do you prefer condensers or dynamics? How many mics did you try before finding one you love?

The problem might not be the microphone. The hum might exist in your room. The mic is just making it obvious.

I recently had a bad hum problem with a particular USB mic. I tried everything I could think of to make the noise go away:

• changed USB cables
• added a ferrite choke to the USB cable
• changed USB ports on the computer
• changed from desktop to laptop
• turned off all the fluorescent lights in the house
• unplugged and reset every connection in the audio chain
• turned off the HVAC and every other appliance that might be causing line noise

None of these made any difference at all.

I tried a different USB mic. The hum was still there! It was less audible, but my spectrum analyzer showed a pronounced spike at 100Hz. Then I tried an analog mic through an external ADC. The 100Hz spike was still there. Clearly the hum was real — not a defective mic, not a bad cable, not some sort of weird USB crosstalk within the computer.

The problem was obvious, once I found it. For every test, I had mounted the mic to a boom arm attached to my desktop. (See my shootout and review of podcast/broadcast microphone boom arms.) Sitting on the same surface, three feet away, was a Seagate external USB disk drive, which runs 24×7 as a dedicated backup drive (running the OSX “Time Machine” utility). The vibrations from the spinning disk traveled across the desk, up the boom arm, and into the microphone. The mics without shockmounts, whether external or internal, captured more of the 100Hz hum.

I ran three tests at a fixed input gain level, using the USB mic most sensitive to hum. First up is the worst-case scenario, with the mic on its hard mount — essentially making it mechanically coupled to the disk drive’s motor. The 100Hz hum peaks at -44dB. (Click any of these images to zoom in.)

I replaced the mic’s hard mount with an elastic shockmount. The shockmount bought me 20dB of isolation; the hum dropped to -64dB.

Then I un-mounted the external Seagate drive from the desktop, and let its motor spin down. Predictably, the hum disappeared into the noise floor of the mic’s USB circuitry, around -100dB.

How to fix podcasting microphone hum

The lesson is that if your mic is picking up unwanted noise, the problem might not be some sort of esoteric EMI or RFI, or a bad mic or cable, but actual mechanical noise in your environment. Try this: take the mic off its desktop stand (USB mics nearly always come with desktop stands) and hold it it your hand. If that kills the hum, then there’s something on your desk that’s making the noise your mic is hearing. The culprit is probably the computer itself, or a disk drive, or a stereo component with a fan.

If you can’t get the noise source off the desk, you could try isolating it with Sorbothane feet or some other decoupling solution. I tested this idea by laying the Seagate external disk drive sideways on three Sorbothane feet; the 100Hz noise heard by the mic dropped by 30dB, well below the threshold of audibility. In other words, decoupling the disk drive from the desktop worked better at suppressing hum than putting the mic into a shockmount.

Another option is to mount the mic on a heavy floor stand. And I always recommend using a shockmount for the mic as well.

Bonus tip – freeware RTA

By the way, the RTA screenshots above are from BlueCat Software’s freeware FreqAnalyst spectrum analyzer plug-in. It is a great tool, and the price is right.

RØDE SmartLav

To be clear, this is not a studio microphone. Rather, this is an inexpensive, totally convenient portable voice recording solution that’s great for recording one person speaking. If you’re giving a presentation and you want to record yourself, this is the best bargain I’ve seen for an unobtrusive, decent-sounding audio track. Or if you’re recording a panel discussion on video, give each speaker one of these. Sure, each would already have a mic on a stand in front of them for the PA, but what about when they turn their heads or walk around? They can’t walk away from a lav mic!

The SmartLav is fully compatible with Rode’s new “Rode Rec” app for iPhone (also compatible with iPad and iPod Touch). There are two versions, LE and HD; the latter costs $5.99 and adds editing and export options, compression, EQ, and iZotope presets for optimizing a variety of recording types (some of which are likely more useful for other external mics like Rode’s RØDE’s iXY). But the free LE version is more than adequate for recording and exporting high-quality audio.

Introducing the SmartLav

In Use

Using the SmartLav is extraordinarily easy: plug it into your iDevice’s headphone/mic jack, and clip the mic to your shirt or jacket, about 6 inches from your mouth.

The mic will hear you, but will not pick up much ambient noise from the room. And because it is an omnidirectional mic, it won’t hear much of a level change if you turn your head away from the microphone while you’re speaking. That’s one of the reasons lav mics work well for voice capture.

Audio Samples

This test tries to answer the question, “why not just lay your iPhone on the podium or conference table in front of you, and record with the built-in mic?” I plugged the SmartLav into an iPad, and simultaneously recorded myself with an iPhone5.

This test is a bit unfair to the SmartLav, because I did the recordings in my home studio, which is a relatively well-treated space. In a hall or conference room, the reflected sound — the size of the room — is much more apparent. Nonetheless:

I matched gain levels during tracking to within 1dB, using the analog level control within the Rode Rec app. Then I emailed both tracks to myself from their respective devices, imported both to Pro Tools for gain matching, compression, and limiting — as you’d do for most podcast and speech tracks.

The difference is clearly audible. The iPhone built-in mic hears an awful lot of room sound, even in my treated room. There is a hiss in the Lav track, an inescapable consequence of using a miniature capsule, but the noise in the iPhone’s track has a wider frequency range — broadband noise that is harder to filter out, and to my ear, more distracting. The SmartLav track is significantly better sounding.

In a bigger room, say a church or conference room, the room sound would be even worse.

Alternatives

Your iPhone’s built-in mic is not a reasonable alternative; it is designed for close-range pickup, which only happens if you can hold the phone right in front of your mouth. That’s not an option during a sermon, presentation, or wedding ceremony.

How about a portable digital recorder, like the Zoom H2 or H4N? Those are great for capturing the sound of a room, which is precisely what the SmartLav is designed not to do.

How about the Apple earbud mic? This certainly approaches the lavalier model, and probably sounds fine, but you can’t exactly wear earbuds in any of the environments where the SmartLav was intended to be used.

How about a lav mic from another vendor? These work well, so long as you have a portable recorder to plug them into. I’m not sure any of those recorders offer Dropbox, email, and Soundcloud filesharing, though, so the combination of the SmartLav and RodeRec app seem to provide a superior solution. I imagine you could find an adapter cable to plug a 3rd-party lav mic into the iPhone’s TRRS jack, if you want to hunt one down. On the other hand, the SmartLav just works.

RØDE Rec

I bought the HD version of the app for $5.99, in part because I know that the guys at Audiofile Engineering had their hands on it. (See my recent review of the company’s SampleManager application, which has become a key part of my audio file workflow.) I have not yet played with the various filters and editing features, but I found the app easy to use for basic field recording and file-sharing.

Conclusion

The SmartLav is an inexpensive, portable, unobtrusive, high-quality speech capture solution. If you need a lav mic and can leverage iDevices for recording, storage, and filesharing, the SmartLav and RØDE Rec app provide a unique feature set that puts them at the top of your shopping list.

I started my professional recording career in the early 1990’s, when 2-inch tape was the main recording format for all but the few who could afford a Sony 3348 digital machine. We recorded to either a Studer, Sony/MCI, Ampex or Otari tape machine and mixed through our analog boards to a half-inch tape machine, usually a Studer or Ampex.

Each tape machine had its advantages and disadvantages: the JH-24 (Sony/MCI) could punch like no other and sounded great, but alignment was a pain and the remote was little more than functional. The Ampex machines sounded fantastic, but forget about punching at all. The Studers had the best of both worlds, with good punching and great sound — at a very high price. In contrast, the Otari machines and Sony APRs had modern features, like auto-alignment and built-in synchronization, but were not considered great-sounding machines. You pays your money and takes your choice. [JH-24 photo from Flickr user Ronan_C]

Fortunately for me, I started right on the cusp of affordable digital multitrack recorders, in the forms of the Alesis ADAT and Tascam DA-88s. Pro Tools (then called Sound Tools) was in its infancy, but worked very well for two-track editing. A few esoteric digital workstations, like the NED Synclavier, AMS Audiofile and Fairlight showed up in post-houses and top-level composer’s studios, but were way out of reach for the average person.

For a little perspective, a decent Sony/MCI JH 24 machine could be purchased used in the early 90’s for around $15,000–not including wiring or required maintenance (e.g., several thousand dollars for a new headstack). At that time a reel of 2-inch tape cost almost $200 and gave you 14 minutes of 24 tracks, so a typical 48 track project (one to three songs) would set you back at least $400 in tape costs.

ADATS and DA-88s brought the cost of 24 track digital audio down to $10,000–15,000, and you could buy just one 8-track machine to start with for around $3000. Digital tape was cheap, with ADATs relying on S-VHS tapes and DA-88s relying on Hi-8 tape cassettes. Each one of those formats, and their various spin-offs, had sonic and functional differences, like synchronizers, remotes, sample rates, etc. These modular digital machines revolutionized the home recording scene before Digidesign (now Avid), EMagic (now Apple), Opcode (now gone), Steinberg and MOTU took the lead with computer based solutions.

So here we are today, with me feeling old and telling stories of yore, with everyone all a-flutter with renewed interest in getting back to the “good old days” of recording to analog tape and pressing vinyl. The changeover from all analog recording to mostly digital recording during the late 90’s brought with it lots of bias, superstition and just plain ignorance. It is true that the very first generation of digital recorders, like the Sony F1 and early DAT machines, didn’t sound as good as the state-of-the-art analog machines. However, the low cost and ease-of-use of the new digital machines guaranteed their success. Luckily, pro audio and audiophile users pushed manufacturers to create better sounding converters and better tools to process the sound (now known as plugins).

It is my belief that much of the pain of switching over to digital recording was due to the tools that engineers had mastered for analog recording. For instance, applying EQ and compression (or no compression) to tape to make up for the color that the tape added didn’t sound so great when recording to digital. Bright FET microphones and harsh transistor preamp tones became rounded off in a pleasing way on tape, and by the 100th mix pass, the high-end was rolled off and the transients smeared so much that the final mix sounded phat, warm and fuzzy. It took experienced engineers a minute (or years) to gather their thoughts, re-examine their tools and learn how to take advantage of the clarity, quiet, and unforgiving purity of digital recording. At that point recordists moved towards super-fast, ultra-clean and high-gain preamps and transparent compression. Low cost digital processors stopped using transformers and tubes, which lowered costs and also lowered THD, while widening frequency bandwidth specs from DC to light. We had finally found it: perfect, clean, sterile audio!

Fast forward to now. Today’s great equipment designers came up studying the fundamentals of analog design, and learned digital design and processing during the evolution of digital. This has led to a hybrid approach of using colorful transformers and tubes alongside super-clean analog and digital processing for the best of both worlds. Recording artists like Lenny Kravitz led the way to digital, and then reverted back to analog, showing us the extremes of each medium. I think almost everyone is now satisfied that by using good sounding analog gear and even affordable converters, digital recording is not only acceptable, but can sound downright amazing.

Humans just can’t let things rest, though. Now we want to re-integrate production techniques and sonics that can most easily be found or created using our old friend(?) analog tape, alongside our digital gear. Fortunately, tape machines have not been out of use for very long, and the techs and engineers at top studios still remember how to align and maintain analog machines. Unfortunately, the companies that made 2-inch tape have all closed their doors. Some new companies have done a good job picking up the pieces, and have been manufacturing analog tape for a few years. While the quality and sonics of the new tape is not exactly the same as the best tape of late 90’s, the stock is very good and totally useable. It is, however, expensive, with a reel of half-inch running about $100, and a new reel of 2” tape from ATR or RMGI fetching just over $300.

I, for one, was happy to see 2” tape slowly fade away, and early on I was toting my Pro Tools rig to mix sessions where the studio begrudgingly set up their SSL synchronizer to talk to my Pro Tools rig. Nowadays the Pro Tools rig has taken the place of the tape machines in almost every studio, and a premium has to be paid to use the analog machines, if they are even still available.

So, how do we know when the analog machine will add just the right amount of punch, warmth, fuzziness or just plain magic to our drums, vocals, guitar, bass? Recently, I worked on a project that was recorded at a few different world-class studios, using both analog and digital recorders and used the experience to document some notable differences in the sonics of each recording format. I will summarize each tracking session and give audio examples, where possible for each instrument and format.

Drums

Here’s the basic scenario: a Latin pop record with some great players, mostly recorded using acoustic instruments (or amplifiers) in great studios. The drums, played by Mario Calire, were recorded at Henson (formerly A&M) in studio A and were engineered by Elliot Scheiner. If you don’t know who Elliot is, look him up — he has recorded, mixed and produced most of your favorite records, from Steely Dan to the Foo Fighters.

The drums were only recorded to 2” tape, so no direct comparison can be given here, but I’ll include a sample that you can compare against your drum sounds. Keep in mind that Elliot’s drum sound will probably kill yours regardless of analog or digital, but it is a great benchmark to work towards.

Session Setup

The rest of the album was recorded at Ocean Way in Hollywood, using an Ampex 2” machine and their fantastic Focusrite console. The 2” tape we used was NOS 3M 996 tape stock that Henson Studios sold to us, and we recorded at a level of +5/185 at 30 ips, which is a quiet, but not overly hot level. For reference, most of the R&B stuff I recorded in the 90s was at +9/185, while a lot of rock stuff was recorded at +3/185. Basically, the higher the level, the lower the noise floor and the punchier the sounds. So, drum machines, synths and smooth vocals sound huge with very little tape hiss at a high level and a tape speed of 30ips, while distorted guitars, loud, live drums and screaming vocals sound huge, with a higher, but acceptable, noise floor at a lower recording level and lower tape speed (15 ips). No noise reduction was used on this project.

For the remainder of the instruments, all tracks were recorded buss out from the console and multed (split) at the patchbay to both the tape machine and the Pro Tools interface. The output of the tape machine’s repro head fed a Prism Dream converter and then went to Pro Tools digitally. After each recording pass, the tape tracks were nudged forward in Pro Tools by a bit over 9000 samples (about 10ms at 96khz), which was the delay between the direct signal and the output of the repro head of the tape machine. We recorded into Pro Tools at 96khz, 24bit. Vocals and percussion were recorded at our private studios, so there was no tape used, but the few synth tracks we used were recorded through tape. I didn’t print tape and Pro Tools for the synth tracks because I could spend time playing back midi instruments and adjusting the record level to get the desired tape saturation vs tone for the synths. For the live players, I didn’t have the luxury of time to experiment with tape levels, so I went with my experience for best practice.

WAV Audio Archive

[Download the 16-bit, 44.1 kHz WAV audio files here.]

Bass (acoustic and electric)

WIth drums out of the way, we were onto bass. First we recorded acoustic bass, played by Carlitos del-Puerto. I used a FET 47 on the body and a KM54 on the neck, with both mics about 8 inches from the instrument. Both signals were mixed at the console and bussed to one output.

Acoustic bass tracks are the sum of the Fet 47 and KM54:
Acoustic Bass, 24-bit digital
Acoustic Bass, analog tape

Electric bass was then recorded using Ocean Way’s custom Jensen transformer DI box into the console and then through their respective tape and Pro Tools paths.

Electric bass (direct box):
Electric Bass, 24-bit digital
Electric Bass, analog tape

In both pairs of samples, listen for the difference in the low end response, apparent level (thanks to tape compression) and clarity of each path. During mixing, I used only the bass recorded onto tape for all the songs.

Guitars (acoustic and electric)

Guitars were next and the player was Grecco Buratto on both acoustic and electric. Acoustic guitar was miked with a Neumann KM 54 around the 14th fret. After a few minutes, I decided the analog tape was not the way to go with acoustic guitar, so we recorded only direct to Pro Tools. Electric, however, was another story. Grecco came to the session armed to the teeth with amps, cabinets and pedals. We set up four or five different pairs of cabinets in the live room and his pedalboard, amps and switcher were in the control room with us. Each cabinet was miked with three mics (isn’t a big studio wonderful!): an SM57, an MD 421 and a FET47. I usually buss the 57 and 421 to one track and record the more distant 47 as a room mic onto a separate track. For the audio examples, there are a variety of styles and one example also includes the mix processing to show how different the guitar wound up sounding during the mix. For the most part, electric guitars on tape were used for the final mix.

All guitar tracks are the sum of an SM57 and MD421 close mics:
Funk Electric Guitar, 24-bit digital
Funk Electric Guitar, analog tape

Electric Guitar, 24-bit digital
Electric Guitar, analog tape

Electric Guitar, 24-bit digital
Electric Guitar, analog tape

Electric Guitar, analog tape, EQ'd for mix

Piano

Piano was interesting. We had two pianos to choose from and we went with the smaller, 7-foot Steinway B, over the 9-foot D. Piano was played by co-producer Carlos Rodgarman. I miked the piano with two Royer R-121s in a sort of XY pair, right above the hammers and focused about halfway down the strings. The setup looks like a “V” with the point toward the strings. The backside of the mics point over the players shoulders. The live room in studio A is huge–over 2000 square feet with 23′ ceilings. With this mic setup, the piano sounded live, but not bathed in reverb. For this type of pop music, I like a bright, punchy piano with good stereo spread and the Royers provide that very well–and still fold towards mono without sound phasey. The ribbons also take EQ well, and I provided a sample of tape, Pro Tools and the processed version used for the mix.

The processed version wound up heavily EQ’d, but in this dense funk mix it sounded great. For some songs, the tape felt a bit thinner than the digital recordings, while for some songs, the mellower tape sound was just the ticket. Each version had its benefits. With the right compression and EQ, I generally find that even a poorly recorded piano can be made to sit well in most mixes–as long as the piano doesn’t have to be featured.

Conclusion

All-in-all it was a good experiment and well worth the extra mile. For the drums and bass, the tape certainly added a punchiness that would otherwise take a bit of processing to achieve. The electric guitars benefited from the rounding off that the tape provided on certain tracks and on others I like the clarity of digital signal.

Acoustic guitar didn’t really work for me on tape, and piano was a mixed bag.

For day-to-day recording I don’t have the luxury of tape, but for the important projects or the special case where a certain color is desired, tape can certainly add its sonic signature. As for printing my stereo mixes to tape vs digital… don’t get me started…

matthew mcglynn

Many thanks to Adam Kagan for conceiving and delivering this piece! Follow Adam online via http://about.me/adamkagan.

If you enjoy challenging long-held assumptions about whether vintage gear always sounds better, you’ll probably enjoy our most famous mic shootout, the U87 vs. the U87Ai.

The Mic Database has nearly 1400 microphones in it, with at least one good photograph of each. These days, most vendors publish decent product photos, which we use whenever we can. But there are hundreds of interesting older mics for which no such photos exist. We need photos of all those mics. If you have mics in your collection that aren’t already on this website, please shoot photos and send ‘em in!

But first, read this so your photos won’t suck.

Quick guide to home “macro” photography

1. Find a neutral-colored wall. White is best. A corner is better still.
2. If possible, create a backdrop from a large sheet of paper or poster board. (See photo at right.)
3. Use as much ambient light as possible.
4. Turn off the camera flash, or find a way to bounce/diffuse it.
5. Stand the mic up on its butt end, with no shockmount.
6. Turn the camera sideways (portrait mode).
7. Use a tripod, or brace the camera on a table, for any exposure slower than 1/60th of a second.

The photo shows my rig from a year ago, but the real point is the staging, not the camera gear.

The curved posterboard provides a smooth (“infinite”) backdrop. The white sidewall helps bounce light onto the subject. The room has multiple overhead track lights (note the shadows), helping to provide even lighting from multiple angles. Further, two flashes are used, with diffusors, to create more light and softer/fewer shadows. The on-camera flash is bounced off the ceiling. The side flash lights the subject directly, but also bounces off the side wall.

The problem with most photos

It’s all about the light. Your on-camera flash provides a single point of extraordinarily bright light, which causes unwanted specular highlights as well as extreme shadows. Neither is helping your mic photo.

Professional studio photographers have multiple light sources, all of which are diffused. The cheapest way to reproduce this effect is to shoot in a brightly-lit room. Either turn off your camera flash, or find a way to bounce or diffuse it.

For example, bounce the flash off the ceiling (if it can be articulated). Or, if the flash can’t be pointed elsewhere, hold a piece of thin white paper or waxed paper in front of it when you take the photo — instant DIY diffusor!

The photo at right shows the difference. The left half of the image used direct on-camera flash. For the second image, I increased the ambient light in the room, then held a piece of printer paper in front of the on-camera flash. No additional gear was used — well, except for a piece of 8.5×11” paper. Notice how much more evenly lit the subject is, and how soft the shadow is.

Shiny microphones are notoriously difficult to photograph, so while neither of these images is great, the second one is much more usable, with less editing required. (It’s a DIYAC RM5 ribbon, by the way.)

Better cameras take better photos

I hesitate to point this out, in part because it’s obvious but also because I’d rather have you send me nicely staged and lit iPhone photos than no photos at all.

Pricier cameras tend to have better glass in the lenses, and larger sensor chips. Both contribute to image quality. See sample at right, which compares the output of a typical pocket digital camera (in this case, a ~$300 Canon from SD450 from about five years ago) to an EOS Rebel XTi with a nice lens. The larger camera reveals much better detail. The image from the SLR suffers from much less distortion, and fewer compression artifacts. The image from the pocket camera looks smeared.

Still, for the purposes of showing what the mic looks like, the photo from the smaller camera is adequate.

Post-processing images

Even your neutral, “infinite” backdrop will probably show some shadows. With the right gear and technique, you’ll have multiple, light and diffuse shadows rather than one sharp-edged dark shadow. Whatever shadow you have, the photo will be improved if you remove it.

Here’s the wrong way to do it: Photoshop’s “magic wand” tool. Yes, you can click and magically select most of the shadow. But the edges are ragged. The result is that the mic looks like it has scabs.

The far better way is to outline the mic with the Paths tool, then convert the path to a selection, and fill with white. You’ll get perfect, crisp edges.

By the way, this video is the most entertaining way to learn about the Paths tool, courtesy “You Suck at Photoshop.”

How to take a terrible mic photo

Lay it on your desk in front of a 1986-era PC keyboard that has enough grime and junk stuffed among the keys that, if you dug it all out, you could use to reassemble an entire hamburger. And maybe a cat. Use a wide-angle lens and hold the camera very close; this makes the mic bulge, and maximizes flashburn. Then wiggle while you shoot.

Not even Photoshop can help that picture.

Photoshop is truly like Pro Tools for images, but the same rules apply. There is only so much studio horror that can be “fixed in the mix.”

Respect for Curves

While we’re on the subject of Photoshop, I’ll give a shoutout to Curves. It’s the best color and contrast correction tool I’ve found. Check out the example at right; it shows what my camera’s automatic white balance did with the goofy lighting at the Anaheim Convention Center during NAMM. A couple curve corrections later, and we have a black microphone again. Nice, eh?

What about a Macro Box?

You can DIY a macro studio. I used to have one of these, and got pretty decent results with it. But this giant box was always in the way; it couldn’t be flattened for storage. Also, I have gotten better results without the box. Still, this is a valid low-cost way to shoot some nice photos.

In a nutshell, you’d put a posterboard backdrop into a cardboard box, and cut tissue-paper windows in the top and sides. You can use it outside, too, sometimes without additional lighting. See the canonical howto in Strobist’s DIY $10 Macro Photo Studio.

Macro box photo courtesy Flickr user FutureFashion

Help Me Help You

I’m on a mission to document every microphone ever made. But I need a photo of each of those microphones first. If you have mics in your locker that are not already present in the mic database on this website, please follow the steps above and shoot photos for me. Contact me here.

You get bonus point for shooting photos of the circuit board and capsule too.

A new contender in this category is the sE Electronics X1. Like the NT1A, it also features a fixed Cardioid polar pattern at a competitive price. In this review, I will compare it to the established Rode NT1A.

One of the things that I’ve always noticed about the NT1A is how light it is. It always feels lighter than I expect a mic of its size to be. So when I first picked up the X1, I experienced the opposite sensation. It’s a lot heavier than I thought it would be. It feels substantial. Often the weight of a product psychologically contributes to one’s perception of quality, and in the case of the X1, the mic’s weight instills confidence that it is built solidly and is a quality product. Whether that perception is justified by studio performance, of course, remains to be seen.

sE X1 Accessories

The X1 ships in a foam-lined box containing the mic, a clip and an instruction manual with a frequency response chart. When I first opened the box, I was greeted with a picture of one of sE Electronics’ high-end mics with a shock mount, so I was a bit disappointed to find no shock mount bundled with the X1.

Besides the weight of the mic, another thing that struck me was the mic’s finish. The entire mic is covered with rubberized textured finish, even the grille! I think this is a brilliant idea, because the coating dampens the mic body and grille, preventing mechanically-induced resonances from affecting the audio. I have noticed this phenomena with many mics, especially those with undampened brass bodies. These mic bodies (and grilles) can produce a distinct sustaining tone when tapped. Because the components, including the pickup element, are not internally isolated from the body, mechanical vibrations can introduce resonance peaks into the audio signal. The downside to the X1’s rubberized finish is that it seems to scuff easily, and attract dust.

I ran into a problem with the mic clip included with the X1. While I did manage to wrestle the darn thing onto the mic once, it was entirely too difficult and frustrating, and was impossible to do without thoroughly blemishing the finish around the bottom of the mic where the clip attaches. The clips’ inner diameter may be roughly the right size to fit onto the mic, but the clip itself is too rigid. Mounting the clip on the mic is entirely too laborious. And once you get it on, it’s so tight that it is difficult to rotate the mic around in the clip. I take great care in fine-positioning my mics in a session, but this clip left me without the ability to do so. Attempts at small position changes would only result in sudden, big moves. For the rest of my sessions, I used an Audix D-Clip, which worked the way a simple mic clip should. Unless you plan to use an X1 for some kind of permanent installation, I’d say save yourself the frustration and use a different mic clip.

sE X1 Features

Unlike the no-frills NT1A, the X1 sports a -10db pad and a high pass filter, which cuts 4db/octave starting at 500hz. The switches don’t feel as solid as the rest of the mic. It’s hard to tell just by looking at them if they’re engaged or not, due to the lack of clear indicator markings. In fact, it’s hard to tell if you’ve engaged them or not, because they wiggle side to side a fair amount.

sE X1 in use: the sessions

I hold a part-time faculty position at Clackamas Community College’s music technology department. We have a thriving music program and lots of talent, so I enlisted the help of two promising student vocalists for my first shootout. For this I opted to set up both mics positioned side by side as closely as possible so that they would capture the same performance. The artist’s movement during a performance could cause slight volume and tonal fluctuations between the two mics, but I feel the resulting differences are negligible as compared to the potential differences between separate performances.

The signal chain for these sessions was as follows:
sE X1/Rode NT1A →
  True Systems Precision 8 Mic Preamp →
   Digidesign 192 I/O →
    ProTools 8 HD

For vocal recordings, an external pop filter was used.

Male Vocals

First I recorded Joe Collins, who is a baritone. He performed an excerpt of a tune called “It Hurts Me Hurting You” by Intervision–a fantastic local Portland band.

At the first listen, both mics sounded very similar. No glaring differences between them stood out to me. The X1 has a little more high end going on which brings out more sibilance. I really dig how the Rode handles Joe’s siblance, and I would likely not put a de-esser on that track. With the X1, I would likely reach for a de-esser.

To my ear, the X1 was leaner on the low-end, but only very slightly so–nothing that a decrease in proximity effect on the NT1-A couldn’t replicate. Overall, there isn’t a huge difference between the mics.

Female Vocals

Next I recorded Brooke Braaten, who is a mezzo-soprano, but has a wide range. She performed an acapella cover of “Settle Down” by the wonderful New Zealand artist Kimbra.

I’m glad she sang such a dynamic piece because it threw a lot of information at these two mics. During several passages, she caught me by surprise and sang more powerfully than during the mic check and than I had anticipated. While neither mic peaked out in ProTools, you can audibly hear the signals being pushed in those spots. I could have re-recorded, but even when you leave headroom it can still happen during sessions. I hear the X1’s signal falling apart slightly during the peaks, whereas the Rode passes a sound that would still be very usable in a mix.

The difference in the high end and the resulting sibilance is more apparent here than it was with Joe, and I again prefer the NT1A. The track recorded through the X1 was pretty uncomfortable and fatiguing to listen to during “s’s”, “ch’s” and some “t’s”. And again the NT1A exhibited more low-mids — perhaps a little more than I’d like for Brooke’s voice, but when it comes to dealing with proximity effect/low mids on vocals I always prefer having to use EQ subtractively than additively. For high-end on vocals it’s the opposite. There’s nothing that I dislike more than having to attenuate the high-end on a vocal track after the fact. No matter which EQ I use, to me it always ends up giving the track the sensation of having a towel thrown over the voice. I’d much rather start with the warm sound of the NT1A and add a high shelf at 12k (if I need more cut) than having to deal with the excessive sibilance from the X1 track.

Guitars

A student called Roberto was kind enough to play one of his own songs for me. I positioned both mics as closely as possible together, about 8 inches in front of the 13th fret (where the neck of the guitar meets the body). This is a proven sweet spot for mono-miking an acoustic guitar.

Before tracking, Roberto warned me (and lamented the fact) that the strings on his Fender DG-series guitar were old and worn. This is a good example of a real-world scenario that I and many other engineers commonly have to deal with.

For this instrument, the extra articulation offered by the X1 was welcome, to my ears. I suspect the Rode track would need the help of some additive EQ to help cut through a full mix, and also for my taste. To me the X1 adds just the right kind and amount of brightness to the guitar. The track does not sound brittle or thin, but rather pleasantly crisp. I also like how the X1 controls the low mids. In a mix with other instruments, I don’t want too much of those from the acoustic guitar, so I usually cut around 200–400hz. I would definitely do so with the NT1A track, but probably not with the X1; the balance of the X1 on this guitar is excellent, and is faithful to the sound of the instrument.

Joe Collins, the male singer we heard earlier, also plays the guitar, quite well I must add. He performed an original composition on his Epiphone Dreadnaught. In contrast to Roberto’s strumming, Joe employed a finger-picking style.

Joe’s guitar was the quietest track I recorded with these mics, so this was a good test to see if the mics are suitable for quiet acoustic sources. A mic’s self-noise comes into play here. The NT1A, with its very impressive 5dBA rating, shines here. The noise floor is entirely inaudible, even through headphones. The sE X1, according to its tech specs, produces 16dBA self-noise. That is a middle-of-the-road value for microphones in this price range [see a listing of 9 competitive models in our Best $200 Podcasting Condenser Mic Shootout].

The noise in the X1 track is clearly audible.

I used the same mic array here as with the previous guitar recording, but positioned a little further back from the instrument. Nonetheless, the NT1A still gave me more bass response than I want. I thought the lows were well-controlled in the X1 track.

Yet for this production, which would be either solo instrumental or acoustic with vocals, I preferred the more neutral-sounding high-end of the NT1A than the slightly boosted highs of the X1.

Listening Key

In all the above audio tracks, Mic #1 is the sE Electronics X1. Mic #2 is the Rode NT1-A.

Conclusion

The X1 definitely has some features that stood out to me. I applaud sE for the interesting finish on the mic; I think other manufacturers should take note. The -10db pad would allow me to use the X1 for loud sources (such as close drum overheads), which I know from experience the NT1A can not handle.

I’ve done enough critical listening to comfortably say I hear no audible differences between the larger midrange section of the frequency spectrum of these two microphones. Their capture of the timbre of each source was the same, even for the variety of sources I recorded. But the X1 and NT1A are clearly different in the upper mids, and in the “air” section, and also in the low end.

For all-around studio usage, I would give the edge to the Rode NT1-A. My preference is based in part on the accessories, because the NT1-A includes a shockmount and a sturdy zip bag. In contrast, the clip included with the X1 was a disaster. To someone on a budget–the most likely kind of person to buy the X1–it will not seem reasonable to spend more money on a third-party mic clip. Nor to transport the X1 in a cardboard box.

I wish the switches for the pad and the HPF on the X1 matched the mics’ otherwise very sturdy construction.

In terms of the sound of these two microphones, I preferred the warmer and more faithful Rode won me over, verses the brighter and leaner X1.