Product Review: Positive Grid Spark Mini

Silly me, I thought the Positive Grid Spark Mini was a fairly new product, but my resident guitar repair guru and guy-who-will-try-to-fix-anything about town, Brian Stewart (Tree Strings Music), has already repaired one in his Red Wing shop. I haven’t yet heard what the fault was in that unit. I ordered a white one from Amazon, thinking it might be a fun practice and outdoor jamming amp. I’ve had it about a week and, sadly, the fun is wearing off fast. The good and bad news is that almost everything about this amp is driven by a phone/tablet app, iPhone or Android. The good is that it has hidden power if you’re willing to climb the usual steep software learning curve. The bad is, like most apps, it’s glitchy, unpredictable and often counter-intuitive, almost completely inflexible, and very dumbed-down while pretending to be a product for the sophisticated, discriminating guitarist (the ultimate oxymoron?). A lot of the positive reviews you will find for this amp begin with something like “I’m new to guitar and have only been playing about a few months . . .” It’s easy to like or even love something if you don’t have anything to compare it to. In my case, it’s hard for me to look at any product with the eyes of a newbie. So prepare to be disappointed if you’re hoping for that kind of bubbly, happy-talk review. At 74 and after 50 years in various areas of pro audio and music, there is nothing new about me except for the crap that keeps popping up every time I have a doctor’s appointment. Having spent 20-some years in test and reliability engineering I tend to find more things wrong with software than right.

 

You can’t beat the Mini’s physical controls for simplicity. On the top of the amp chassis, you get 4-position Preset switch (Rhythm, Lead, Solo, and Custom), a Guitar volume, a Music volume control (Bluetooth or Aux In signals), and a guitar input. The back of the chassis has 3.175mm (aka 1/8”) Line Out and Aux Input jacks, a USB-C port for charging the battery and (sometime in the future) a functioning digital audio interface), a Bluetooth “Pair” switch (the Pair switch also fires up a rudimentary guitar tuner), and a power switch. The amp comes with a cute leatherette strap and a pair of buttons to attach the strap on the side. The amp is a 10W Class-D unit that, supposedly produces 90dBSPL at 1m. The cabinet has two 2” speakers and a bottom-facing passive radiator. The 3Ah battery supposedly provides power for 8 hours (on mid-to-low power output) and charges from empty to full in 3 hours. The firmware contains “33 Amp Models, 43 Effects, (Noise Gate, Compressor, Distortion, Modulation/EQ, Delay, Reverb – fixed in that order) and the USB interface is a 44kHz/16 bit A/D. You also get a a free download of PreSonus Studio One Prime recording software with your original purchase. Registering for your software is the closest thing to registering for warranty with Positive Grid. You can buy (for $110) a Spark Control footswitch to either control the presets, turn on and off various virtual pedals, or a combination of those functions. The amp is 146.5 x 123 x 165 mm (5.76 x 4.84 x 6.49 in) and weighs 1.5 kg (3.3 lb).

As usual, the included paper “Quick Start Guide” is close to useless. Not so typically, Positive Grid hasn’t provided much in the way of useful information on their website, YouTube, or anywhere else. Figuring out the app and the various features of the amp that are only accessed through the app is up to the buyer.

For a beginning guitarist who doesn’t know any other musicians, some of the Mini’s app features are probably fun-to-useful. This “screenshot” is really a compilation of three different screens as typically displayed on a phone. The middle one is an example of a dumbed-down imitation of a fairly common DAW guitar pedal screen; like the one in Logic Pro. A big difference between the DAW pedal boards and the Spark is that you can’t reshuffle the order of the pedals to suit your purposes.

After spending considerable time playing with the various pedals I can say “they work.” The compressors in the Comp/Wah section aren’t up to DAW standards, but they are probably as good as most hardware pedals. The “Wah” function, also included in this group, is “Temporarily Disabled.” As usual, I don’t like the distortion (Drive) pedals much, but I rarely do. About half of the Drive pedals are red-flagged, which means you’ll have to spend $20 or more to enable those pedals on your device. So it goes for the Amp models, too. Most of the red-flagged amp models are variations on the mediocre Marshall models. The Mod/EQ models are predictable and not bad. The Delays are ok, except for the absence of a multi-tap delay. The Reverbs are typically pretty good, since digital reverb plug-ins have been fairly well staked-out territory for at least 20 years. I didn’t find a favorite from the verbs, but I didn’t find anything I hated either.

Irritatingly, with my Samsung tablet and the Samsung Music player, anytime I open the Spark app the music player starts playing something from my current playlist through the Spark Mini. Before you start babbling about some “play on Bluetooth connection” toggle in the player, get a grip on yourself. No other Bluetooth device that I own has this behavior: from consumer buds to Shure in-ears to three different Bluetooth speaker systems. It is a glitch in the Spark app and that has been logged by Positive Grid’s customer service and I wasn’t the first to make the complaint. If everything else was excellent this wouldn’t be a deal-breaker, just unpredictably irritating. (If it does this when I first open up the app, will it spontaneously do the same during a gig?) Yes, I could turn off the Music volume, but if I am using it as a backing track at the time it sort of defeats the purpose of that function.

With that out of the way, my impression of the guitar amp is somewhat positive. I’m not fond of electric guitar distortion in the usual buzz-box fashion, but some of the amp models deliver decent slightly over-driven sounds with the kind of amp EQ and tone you’d expect from what I’m guessing are the amps being modeled. Some of the setups both by other users and Positive Grid are fair-to-decent. I had some high hopes for Pat Metheny style sounds, but the lack of multi-tap delays squashed that. You could just add a pedal delay up front but that would defeat my purpose. I have an old MacBook Pro with MainStage that will do everything this unit does with a ton more effects including my multi-tap delay that I’d rather use with a small wired power speaker than add a pedal that is almost as big as this amp.

And speaking of power, I wouldn’t be surprised if the Mini an produce 10W, but the distortion at that output would be objectionable. That goes for the spec’d 90dBSPL@1m acoustic output, too. At any volume over a moderately loud voice or a strummed full-size acoustic guitar, the bottom end of this amp clips indecently. It is not a pleasant distortion, either. It is the usual splatting sound of digital clipping. That was the straw that broke the back of my interest in the Positive Grid Spark Mini. There were moments when I thought I was about to find the sweet spot for several of the Presets but “almost there” was as close as I got to something useful. When the amp sounded good, it was too quiet to compete with a couple of acoustic guitars. When it was loud enough to cut through a small instrument crowd, it sounded awful.

For a beginning practice amp the Spark Mini isn’t bad. Most beginners, however, will have a terrible time with the mediocre application software that is an absolute necessity for using the amp. Advanced users will be frustrated with the user-hostile programming of the app and disappointed with the little amp’s small performance.

NOTE: This article was sent to me by the author, Mark Amundson, several years ago to use in my Audio Engineering classes. The article had been on my website as a link in the original Microsoft Word format for a decade. Unfortunately, Mark’s article seems to have vanished from FOH Magazine’s article history, which is a huge loss for audio engineers who actually care about optimizing distortion and noise characteristics in their signal path. A version of this discussion is included in his book; Live Sound Practice and Theory.

Manuscript: Madness to the Method – Gain Structure
Magazine: FOH
Manuscript Type: Technical Article
Issue Assignment: May 2003
Word Doc No: GainStructureMadness.doc
Figures: -
Photos: One
Editor: Bill Evans
Revision: -
Date: 05/12/03
Word Count: 1516

Madness to the Method – Gain Structure

By Mark Amundson

In this dissertation I am going to do a little Q&A, ala the old Audio Cyclopedias about questions you may have wondered about, but just did what you were told. I am going to throw down a generous helping of electronics history as a way of answering the question, and to remind us all were all this technology came from.

Q: WHY IS 0dBu THE REFERENCE SIGNAL LEVEL?

The question could easy re-phrased why is 0.773 volts RMS (0dBu) the standard and not some other convenient number like 0.1, 1, or 10 volts?

The answer goes way back to Alexander Graham Bell’s era when no such thing as radio or broadcasting was thought of. As the “Bell System” and “American Telephone and Telegraph” (AT&T) became the monopoly in the phone service industry, Western Electric Company was formed as a subsidiary of the Bell System to design and produce telephone gear for the whole country. After much trial and error, a standard two-wire pair transmission line was developed with 600-ohm source and load impedances to maximally send carbon microphone signals down the wires. With the right construction materials, voice signals (about –20dBu) could transit 5 miles with a passable loss of signal amplitude.

When Lee DeForest invented the Vacuum Tube Triode for signal amplification, his “killer app” was re-boosting feeble telephone signals, thus creating long-distance phone service in the second decade of the last century. Western Electric still had a lock on the electronics industry in the 1920’s as broadcast radio was just emerging, so naturally it had the highest technology suitable to fulfill civilian and military requests for standard “Public Address” apparatus. By the early 1930’s Western Electric had the first quality dynamic microphone (requiring no DC power unlike carbon mics) and combined vacuum tube amplification connected to the first efficient “loud-speaking apparatus” that we now know as horn loaded drivers.

As broadcast radio became widespread, and specialized companies like Electro-Voice, MagnaVox, and Shure Brothers came to supply (with Western Electric) the needs of public address and broadcast gear, the 600-ohm line cabling still held as the lowest loss method of distributing and processing audio signals. From that era, a one-milliwatt reference level into 600-ohms became the reference level, or 0dBm (zero deci-Bels referenced to one milliwatt). 0dBm is exactly 0.773 volts RMS, but as technology marched on audio electronics moved from power matching to “bridging” impedance matching, the 0.773 volts without any specified load impedance was now described as 0dBu (zero deci-Bels unreferenced).

To answer the lingering question of what became of Western Electric, government anti-monopoly policies in the 1930’s forced the breakup of AT&T (the first time) into RCA for broadcast, Bell Labs for telephony, and All-Technical Products (Altec) for public address. Altec slowly became Altec-Lansing, then split back to Altec and James B Lansing Inc., then on to JBL.

Q: WHY GAIN (TRIM) TO LINE LEVELS AND THEN MIX AFTERWARDS?

This question is more math than history, but we still thank the early broadcast pioneers of the 1930’s for the first work on defining signal-to-noise and noise source definition. This question could also be formed as what is the best method to minimize hiss in the mixing console?

The answer comes from the invention of the radio, and techniques used to maximize signal-to-noise ratio; and thus transmission distance. As a signal is created, processed, and sent to its final destination; there is a signal-to-noise ratio (SNR) degradation. As each stage, or processing block passes on the signal, the noise eventually encroaches on the signal level. The number of dB drop of SNR per stage is defined as its noise figure (or noise factor for you dB challenged). A noise factor of 6dB or less per amplification (gain) stage is considered a low-noise design for a preamp.

To better visualize this idea, lets put some example numbers to work. If a typical dynamic mic and voice put out –50dBu signal peaks, and the console’s referred input noise is –128dBu, you have a 78dB SNR which is respectable in live sound applications. As the signal proceeds through the channel mic preamp, eq section, channel fader or VCA, summing amps, master fader, and balanced line driver, there is a noise figure penalty to be paid. The good news is if two gain stages are cascaded together, the noise figure of the first stage dominates with the second gain stage noise figure effectively divided by the gain of the first. What this means is that cheaper electronics can be used after the mic preamp, with a high gain preamp covering for the sins of the rest the console’s electronics.

On other item to be shared is that attenuation circuits (eq filters, faders, pots, VCA’s, etc.) can generally be assumed to be direct losses in SNR, with every dB in attenuation a corresponding dB increase in noise figure. So the theoretical perfect (low noise) mixing console setup would be faders maxed, eq flat, and amplifier gains a perfect match between mic level and power amp full power sensitivity.

But no realistic scenario exists on a mixer without faders to “mix” with. So the next best answer is take your desired loudest channel in the mix, set its preamp gain control (gain, trim, etc.) for about 0dBu average level, and keep the channel, group, master faders reasonably high, but preserve some headroom for the occasional “louder” demand. This minor compromise yields the best SNR while still giving mix flexibility. This practice also applies to the gain of all the other signal source channels, but with the obvious idea that their faders would be more attenuated.

Q: WHY IS +22dBu THE COMMON MAXIMUM LEVEL

This answer also comes from electronics history, but only a half-century back. The dawn of the first mass-produced transistors had a typical maximum voltage level of 30 to 40 volts. Of these early transistors, many were targeted for industrial controls and “analog” computers for military and aerospace usage. The most common analog computer section was the “operational amplifier” or “op-amp”. Because these op-amp sections were designed to be near perfect mathematical gain stages with both positive and negative voltage swing capabilities. By taking the limitations of the transistors plus the need for a bi-polar (plus and minus) power supply, the standard of +/-15 volt supply levels was instituted, and is still used today.

As transistors got grouped on one silicon die, integrated circuits (ICs) were born with the first standard products becoming IC op-amps. As IC prices dropped in the late 1960’s and early 1970’s, more IC op-amps started finding there way into audio equipment, still requiring their +/-15 volt power supplies. Today’s pro-audio signal processing and mixing gear is largely composed of IC op-amps and a few application-specific ICs plus just a few necessary un-integrated transistors. The common legacy of supplying them with +/-15 volt levels still exists, with op-amps capable of near +/-14 volt audio signal swings. This level translates to about 10 volts RMS, or +22dBu at which the circuits would exhibit clipping of the signals. Some math trickery maybe also in maximum output specifications as you can gain another 6dB in level by stating the output as balanced; in which each balanced output contact swings in opposite polarity to double the levels.

With the above explanations that we should set our levels close to 0dBu and keep away from the clip levels around +22dBu, there leaves plenty of headroom for classifying what peak signals can be and what is required to get the drive channels (eqs, crossovers, and power amps) to full output. Most power amp manufacturers set their sensitivity values to around 0 to +9dBu for full unclipped speaker drive. By keeping the post pre-amp levels at or below the power amp sensitivity values, mostly assures a clipping free production. Of course that relies on your keeping the power amp input attenuator controls full up. You do not want to be “low-noise” all the way to the amp, and then throw away all that SNR at the last attenuator do you?

FINAL WORDS

Our latest generation of audio production personnel deserve to be educated on how we use our gain structure procedures, and why these methods came about. Some may argue about exact levels and forming up mixes, but I am coming at this from an electrical engineering view and attempting to shine light on what the design engineers consider optimum use, rather than operator tactics “that seem to work for me”. We need to appreciate that live sound borrowed heavily from the telephone and radio broadcast pioneering work, plus how electronics achievements impacted our practices. The twentieth century could be termed the “electronics” century, and it looks like the twenty-first century will be the “photonics” century; with fiber-optics promising near unlimited bandwidth for passing analog or digital signals from baseband to radio-frequency to optical-frequency signals.

MDA