As an Amazon Associate I earn from qualifying purchases. This write-up is based on Universal Audio’s published specifications and documentation, not hands-on testing.

The Apollo x8 is Universal Audio’s 18×24 rack-mount Thunderbolt 3 audio interface with four Unison mic preamps, two Hi-Z inputs, eight line inputs, dual optical ports, S/PDIF, word clock and HEXA Core DSP. It is built for studios that use a lot of line-level gear and outboard processing and want UAD plug-ins in real time.

See the Universal Audio Apollo x8 on Amazon. The linked listing is the Apollo x8 Gen 2 with UAD Analog Classics, the current generation.

Apollo x8 key specs

I/O	18 x 24 over Thunderbolt 3
Inputs	4 Unison mic preamps, 2 Hi-Z instrument inputs, 8 ¼” line inputs
Digital	2 optical Toslink I/O (ADAT S/MUX), RCA coaxial S/PDIF, BNC word clock
DSP	HEXA Core
Conversion	D/A rated at 130 dB dynamic range, −127 dB THD (UA figures)
Monitoring	Speaker switching, talkback, Monitor Correction by Sonarworks, up to 7.1 surround

Source: Universal Audio’s Apollo x8 page.

Unison preamps on the x8

The four mic preamps are Unison preamps, the same system used on the desktop Apollos. Loading a preamp emulation in Console changes the hardware preamp’s impedance and gain behaviour to match the original unit, so a mic responds as it would through that hardware. The emulation is part of the recorded signal. Two Hi-Z inputs also work with Unison amp emulations, so guitar and bass can be tracked through an amp model with near-zero latency while a clean signal is kept for later if the routing allows.

Because the x8 has only four of these preamps, studios that track full drum kits usually add an eight-channel ADAT preamp on one of the optical ports.

Why so many line inputs

The x8 has only four mic preamps but eight line inputs. That suits studios with outboard preamps, synths, drum machines or a summing mixer. External preamps connect at line level, so a studio with four favourite hardware preamps can keep them permanently patched, while still using the four Unison preamps for other sources.

An example studio layout

A typical x8 room might look like this:

• The four Unison preamps take vocal and instrument mics in the live room.
• Line inputs carry two hardware preamps, a stereo synth and a drum machine, all patched permanently.
• Line outputs send to a hardware compressor and a spring reverb, returning on spare line inputs for hybrid mixing.
• An eight-channel ADAT preamp on one optical port adds drum mics when a full kit is recorded.
• The monitor outputs feed main monitors, with alternate outputs on a second pair of reference speakers.

This kind of setup is where the x8 makes more sense than a desktop unit: the value is in leaving everything connected so ideas can be captured without re-patching.

HEXA Core DSP

Six DSP chips give the most real-time UAD processing in the Apollo X line alongside the x16. Large tracking sessions with Unison preamps and compression on every input, and mixing sessions with many UAD plug-ins, both benefit. UA lists more than 200 UAD titles; a starter set is included.

Digital I/O and clocking

Two optical ports allow 16 channels of ADAT at standard sample rates; S/MUX lets ADAT carry higher sample rates at reduced channel counts. Word clock on BNC lets the x8 act as clock master or follow an external clock, which matters in studios with several digital devices.

Console features

UA highlights Auto-Gain, bass management, 7.1 surround monitoring and Plug-In Scenes in Console. These help when switching between projects or when monitoring in surround. For immersive formats beyond 7.1, UA points users to the x16.

Who should consider the Apollo x8

• Studios with outboard gear and synths that need many line inputs.
• Engineers who mix with lots of UAD plug-ins.
• Rooms that need surround monitoring up to 7.1.

Who should look elsewhere

• Home producers recording one or two sources at a time; a desktop Apollo such as the Apollo x4 is easier to use.
• Studios that need eight mic preamps in one box.
• Immersive mixers working in Dolby Atmos; consider the x16.

DAW workflow: hybrid sessions

1. Create hardware insert routing in your DAW for each piece of outboard gear connected to the line outputs and inputs.
2. Use your DAW’s latency measurement for those inserts so returns stay in time.
3. Save Console sessions with projects to recall preamps and monitor settings.
4. Print hardware processing to audio before archiving a project.

Alternatives: the Apollo x4 for a desktop unit, or the Apollo x16 for sixteen line channels and immersive monitoring.

How to set up the Apollo x8

The Apollo x8 is a rack unit with a lot of I/O. So setup goes best with a plan for cables and clocking.

1. Rack it. Leave space for airflow and cable access at the back.
2. Connect Thunderbolt. Use a Thunderbolt 3 or 4 cable to your Mac or Windows computer.
3. Install UA Connect. It installs the driver, Console and plug-in licenses. Register the unit and update firmware.
4. Wire inputs. Use the four Unison preamps for mics and the eight 1/4″ line inputs for synths, outboard or other preamps.
5. Wire outputs. Connect the two monitor outs. Use the eight line outs for ALT speakers, surround or hardware sends.
6. Set clocking. If you add ADAT or S/PDIF gear, pick one master clock. BNC word clock is available too.
7. Build Console mixes. Create headphone cues for the two headphone outputs and save a Plug-In Scene.

Then test every input with a quick recording. Label cables as you go, since an x8 rack fills up fast.

Apollo x8 vs other Apollo interfaces

This table compares the Apollo x8 with nearby models, based on UA’s published specs.

Feature	Apollo x8	Apollo x4	Apollo x16	Apollo Twin X
I/O	18 x 24	12 x 18	18 x 20	10 x 6
Unison preamps	4	4	None	2
Line inputs	8 x 1/4″	Not listed separately	16 (DB-25)	Not listed separately
DSP	HEXA Core	QUAD Core	HEXA Core	DUO or QUAD
D/A dynamic range	130 dB	129 dB	133 dB	129 dB
Headphone outs	2	2	Not listed	1
Surround monitoring	Up to 7.1	Not listed	Up to 9.1.6	Not listed
Form factor	Rack	Desktop	Rack	Desktop

Put simply, the x8 keeps the x4’s preamps but adds line I/O, HEXA Core DSP and 7.1 monitoring. The x16 drops preamps in favor of 16 line channels.

Expanding with two optical ports

UA lists two optical Toslink I/O ports with ADAT S/MUX. With two eight-channel ADAT preamps, you can add 16 more mic inputs at standard sample rates. That is enough for a full drum kit plus a band.

S/MUX lets ADAT carry audio at higher sample rates, but with fewer channels per port. So check your preamp’s manual before you plan a high-rate session. In addition, set one master clock. The x8 has Dual-Crystal Clocking, according to UA, so it often makes sense as the master.

Running 7.1 surround

The eight line outputs can feed a 7.1 surround system. Choose the surround mode in Console and route each output to the right speaker. Then use Bass Management to integrate a subwoofer.

Monitor Correction can help level and balance the speakers. UA notes it requires SoundID Reference and a measurement mic. For bigger immersive formats, look at the x16, which UA lists up to 9.1.6. Our guide to Dolby Atmos mixing and mastering explains the formats.

Who the Apollo x8 is for

• Project studios that record bands and need room to grow.
• Producers with synths, drum machines and outboard gear.
• Engineers who want HEXA Core DSP for heavy tracking chains.
• Rooms that need 7.1 monitoring.

Who should skip it

• Solo creators who record one or two sources. The Twin X fits better.
• Desktop-only setups without rack space. Consider the x4.
• Mix rooms needing 16 line channels or 9.1.6. Look at the x16.
• Computers without Thunderbolt.

Pros and cons

Pros	Cons
Four Unison preamps plus eight line inputs	Rack unit, not a desktop box
HEXA Core DSP	Thunderbolt only
Two optical ports, S/PDIF and word clock	Only four mic preamps without expansion
130 dB D/A and -127 dB THD (UA figures)	Monitor Correction needs extra software and a mic
Up to 7.1 monitoring, talkback and ALT	More I/O than small studios need

Using line inputs for hardware

Eight line inputs make the x8 a good hub for gear. For example, leave synths plugged in all the time. Then record them without repatching. Likewise, use line outs and inputs as hardware inserts for compressors or EQs.

To keep inserts in time, use your DAW’s hardware insert tool. It measures round-trip delay. Also, keep levels matched between sends and returns, so you hear the gear’s effect rather than a volume change.

Tracking a band with the x8

A typical band session uses the four Unison preamps for the most important mics, such as vocal, kick, snare and bass. An ADAT preamp then handles toms and overheads. Meanwhile, the two headphone outputs feed two cue mixes.

Build those cues in Console. For instance, give the drummer more click and bass, and give the singer more vocal. If more performers need mixes, route cues through line outputs to a headphone amp. Then save the whole setup as a Plug-In Scene for next time.

Managing HEXA Core DSP

HEXA Core gives the x8 plenty of DSP. Still, a dozen heavy chains can add up. Watch Console’s DSP meter as you build a session. In addition, use lighter plug-ins for headphone reverb, and move mixing plug-ins to native versions where possible.

For full specs, see UA’s official Apollo x8 page.

Choosing Unison preamps on the x8

Each of the four Unison preamps can load a different emulation. This lets you match the preamp to the source. However, adding color everywhere can crowd a mix, so be selective.

• Vocals: A tube-style preamp such as the UA 610 adds warmth.
• Kick and bass: A punchier preamp can add weight.
• Snare: Try a clean or slightly colored setting and watch the peaks.
• Guitar DI: Use a Hi-Z input with a UAD amp plug-in for monitoring.

Because Unison is printed, choose carefully. If in doubt, record cleaner and color the sound in the mix.

Gain staging with Auto-Gain

UA’s Console includes Auto-Gain. Each performer plays at full level, and Console sets the preamp gain. This speeds up band setups, where several inputs need levels at once.

Even so, check every channel by ear. Leave headroom, because players often hit harder during a real take. Then save the session as a Plug-In Scene so the next setup starts from a good place.

Is the Apollo x8 worth it for you?

The x8 makes sense when you need preamps and a lot of line I/O in one rack. As a result, it works well as the center of a growing studio. On the other hand, small setups may not use its inputs. Count the sources you record at once and the gear you want connected. If both numbers are high, the x8 is a strong match. UA also says up to four Thunderbolt Apollos can be linked, so it can grow with you.

Using talkback and ALT speakers

Talkback lets you talk to performers through their headphones. This is essential when the band is in another room. Press the button, give directions, and release it before the take starts.

ALT switching helps you check mixes. Connect a second speaker pair to the line outputs, then switch between pairs in Console. For example, compare large monitors with small speakers. Keep levels matched, because a louder pair tends to sound better and can mislead you. In addition, combine ALT checks with Monitor Correction for a more reliable picture of your mix.

Common mistakes

• Clock conflicts. Choose one clock master for all digital devices.
• Expecting full ADAT channels at high sample rates. S/MUX reduces channels as the rate rises.
• Treating line inputs as mic inputs. Line inputs need a preamp in front of a microphone.
• Not saving Console sessions. Save Console setups with your DAW projects.
• Double monitoring. Disable DAW input monitoring when monitoring through Console.
• Two clock masters. Set one master clock when using ADAT or S/PDIF gear.
• Ignoring S/MUX channel limits. ADAT channel counts drop at higher sample rates.
• Skipping insert delay compensation. Ping hardware inserts so they stay in time.
• Skipping registration. Included plug-ins stay locked until you register the unit.
• Monitoring twice. Turn off DAW input monitoring when you monitor through Console, or you will hear a doubled sound.

Setup checklist

1. Mount the unit and connect by Thunderbolt 3.
2. Install UA Connect and drivers, then register.
3. Decide clock master and set it in Console.
4. Patch line gear and outboard to line inputs and outputs, and label channels.
5. Set up monitor outputs, alternate speakers and talkback.
6. Calibrate with Monitor Correction if mixing on this system.

Related: Thunderbolt audio interface picks, Dolby Atmos mixing and mastering and the recording hub.

Check the current price of the Universal Audio Apollo x8 on Amazon.

FAQ

When your multitrack mixes are ready, our stem mastering can finish them from grouped stems. It keeps more control than a single stereo file.

As an Amazon Associate I earn from qualifying purchases. This write-up is based on Universal Audio’s published specifications and documentation, not hands-on testing.

The Apollo x4 is Universal Audio’s largest desktop Apollo: a 12×18 Thunderbolt audio interface with four Unison mic preamps, two Hi-Z instrument inputs, optical I/O and QUAD Core DSP for real-time UAD plug-ins. It sits between the two-preamp Twin X and the rack-mount x8, and suits producers who record up to four sources at once from the desktop.

See the Universal Audio Apollo x4 on Amazon. The linked listing is the Apollo x4 Gen 2 with UAD Analog Classics.

Apollo x4 key specs

I/O	12 x 18 over Thunderbolt
Inputs	4 Unison mic preamps, 2 Hi-Z instrument inputs, optical Toslink (ADAT or S/PDIF)
DSP	QUAD Core
Conversion	D/A rated at 129 dB dynamic range (UA figure)
Monitoring	Monitor controller with alternate speakers, talkback, Apollo Monitor Correction by Sonarworks
Connection	Thunderbolt 3 or Thunderbolt 4 controller, depending on unit

Source: Universal Audio’s Apollo x4 page.

Four preamps on the desktop

Four preamps cover a singer and guitarist with two mics each, a small drum setup (kick, snare and two overheads), or a two-host podcast with two spare inputs. All four are Unison preamps, so each can load a different preamp emulation: a Neve-style preamp on the vocal and an API-style one on the guitar amp, for example. Each emulation is part of the recorded input path.

Example sessions

• Singer-songwriter with a band feel: vocal mic on input 1, two mics on an acoustic guitar on inputs 2 and 3, and a DI from a bass or keyboard on a Hi-Z input. Everything goes down in one take with separate tracks.
• Small drum recording: kick, snare and a pair of overheads on the four preamps. Add an ADAT preamp on the optical input when you want toms and room mics.
• Producer desk: a vocal mic and a guitar amp mic on two preamps, a stereo synth on the other two, and Console handling headphone mixes for a guest artist.

The common thread is that the x4 lets a small room work like a larger one without moving to a rack. The monitor knob, talkback and speaker switching stay within reach, which matters when you are both engineer and performer.

QUAD Core DSP

Four DSP chips give room for a Unison preamp plus compression and EQ on several inputs at the same time, with near-zero latency monitoring through UA’s Console. They also run UAD plug-ins during mixing inside DAWs that support VST, AU or AAX. Heavier plug-ins use more DSP, so watch the meter in Console.

Thunderbolt 3 vs Thunderbolt 4 units

UA says that during a transition period the x4 Gen 2 ships with either a Thunderbolt 3 or a Thunderbolt 4 controller. Audio performance, latency and features are the same, but host requirements differ: units with Thunderbolt 4 need an Apple silicon Mac, an Intel Mac on a recent macOS, or a Windows PC with a Thunderbolt 4 port. Check your computer against UA’s current requirements before buying.

Monitor control and correction

The x4 includes speaker switching, talkback, and Apollo Monitor Correction by Sonarworks for monitors and headphones. Correction can make an untreated room more trustworthy, but acoustic treatment and monitor placement still matter.

Expansion

The optical port adds eight channels from an ADAT preamp, and up to four Thunderbolt Apollos can be linked, controlled from the x4 on the desk.

Who should consider the Apollo x4

• Producers recording up to four sources at once.
• Small studios that want desktop control rather than a rack unit.
• Users who mix with UAD plug-ins and want more DSP than the Twin X.

Who should look elsewhere

• Solo artists who never record more than two sources; see the Apollo Twin X or Apollo Solo.
• Studios needing eight preamps or more line I/O, which is the territory of rack-mount Apollos.
• Computers without a compatible Thunderbolt port.

DAW workflow: four-mic session

1. Name inputs in Console and save a Console session alongside the DAW project.
2. Monitor each player with their own headphone mix using Console’s cue sends.
3. Record dry and monitor through Console effects, or print a Unison preamp as part of the tone.
4. After tracking, raise the DAW buffer and use UAD plug-ins for mixing on the QUAD Core DSP.

Alternatives: the Apollo Twin X for two preamps, or a rack-mount Apollo x8 for more line I/O.

How to set up the Apollo x4

Getting the Apollo x4 running takes about the same steps as any Apollo. Follow them in order for a smooth first session.

1. Check your port. UA lists Thunderbolt 3 and 4 support on Mac, including Apple silicon and Intel, and on Windows.
2. Connect power and Thunderbolt. Plug in the power supply, then the Thunderbolt cable.
3. Install UA Connect. It installs the driver, Console and plug-in licenses.
4. Register and update. Register the unit so your bundle unlocks, then update firmware.
5. Select the Apollo in your DAW. Set it as the input and output device.
6. Build your Console mix. Assign Unison preamps, set gain and create headphone mixes.
7. Connect monitors. Use the two monitor outs. Add a second speaker pair on the ALT line outputs if you have one.

Then record a short test take on each input. That confirms routing before a real session starts.

Apollo x4 vs other Apollo interfaces

The table compares the Apollo x4 with nearby models, using UA’s published figures. It shows where the extra preamps and outputs fit.

Feature	Apollo x4	Apollo Twin X	Apollo x8	Apollo Solo
I/O	12 x 18	10 x 6	18 x 24	2 x 4
Unison preamps	4	2	4	2
Hi-Z inputs	2	1	2	1
DSP	QUAD Core	DUO or QUAD	HEXA Core	SOLO Core
Headphone outs	2	1	2	1
Optical	Toslink I/O	Input	2 x Toslink I/O	No
Form factor	Desktop	Desktop	Rack	Desktop

In short, the x4 doubles the Twin X’s preamps and headphone outputs. Meanwhile, the x8 adds HEXA Core DSP and more line I/O in a rack format.

Running two headphone mixes

Two headphone outputs make duets and lessons easier. Each performer can get a separate mix. Console’s cue sends let you build these mixes from inputs and playback.

1. Open the cue section in Console and pick a headphone output.
2. Raise each input’s cue send for that performer.
3. Add a little playback and reverb to taste.
4. Repeat for the second headphone output.

Ask each performer what they want to hear more of. Singers often want more of their voice, while guitarists may want more drums.

Using the optical port for more inputs

UA lists optical Toslink I/O that works as ADAT or S/PDIF. With an ADAT preamp, you can add eight more mic inputs at standard sample rates. That turns the x4 into a small drum-tracking rig.

First, connect optical cables both ways. Next, choose ADAT in Console. Then set one device as the clock master, usually the Apollo. Finally, check for clicks before recording, since they point to clock problems.

Who the Apollo x4 is for

• Home studios that record up to four mics at once.
• Duos who need two separate headphone mixes.
• Producers who want QUAD Core DSP on the desk.
• Mixers who want ALT speakers, talkback and Monitor Correction.

Who should skip it

• Solo artists who record one source. The Solo or Twin X costs less space.
• Studios that need rack mounting and lots of line I/O. Consider the x8.
• Computers without Thunderbolt.
• Anyone who never plans to use UAD plug-ins.

Pros and cons

Pros	Cons
Four Unison preamps on a desktop unit	Thunderbolt only
Two headphone outputs	Desktop size, not rack-mount
QUAD Core DSP	Heavy chains on four inputs can use up DSP
Optical I/O for expansion	Monitor Correction needs extra software and a mic
129 dB D/A (UA figure), talkback and ALT outs	Needs its own power supply

Auto-Gain and Plug-In Scenes

UA’s Console now includes Auto-Gain. Each performer plays at full level, and Console sets the preamp gain. That helps when you’re tracking several people at once.

Plug-In Scenes save Console setups. For instance, store one scene for vocals and another for a full band. As a result, you can switch session types quickly. Still, check gain by ear afterwards, because performers often get louder during real takes.

Setting up Bass Management

If you use a subwoofer, Bass Management helps integrate it. The feature handles the crossover between your speakers and the sub. Set it up in Console, then check the low end with music you know well. Also, combine it with Monitor Correction for a more even response in an untreated room.

Growing beyond the x4

UA says up to four Thunderbolt Apollos can be linked. Dante is also available through the Apollo x16D. So the x4 can stay on the desk while a rack unit adds inputs later. For full specs, see UA’s official Apollo x4 page.

Choosing Unison preamps for a four-mic session

With four Unison preamps, you can give each source its own character. However, too much color on every channel can make a mix muddy. Start simple and add color where it helps.

• Lead vocal: A tube-style preamp such as the UA 610 adds warmth.
• Acoustic guitar: A cleaner preamp keeps string detail.
• Kick or bass amp: A punchier preamp can add weight, but watch the gain.
• Room mic: Try a colored preamp here, since room tracks often benefit from character.

Remember that Unison changes are recorded. So, if you are unsure, keep a cleaner setting and add plug-ins in the mix.

Recording guitar through the Hi-Z inputs

The x4 has two Hi-Z inputs, so two guitarists can plug in directly. Alternatively, one player can record bass and guitar without repatching. Load a UAD amp plug-in in Console to monitor an amp tone with no audible latency.

Next, decide whether to print the amp sound. Many producers record the amp tone and a clean DI on separate tracks. That way, they can re-amp later if the tone doesn’t fit the mix.

Managing DSP across four inputs

QUAD Core DSP covers a lot, but four full chains add up. Console shows DSP use, so watch it while you build a session. If it gets high, use lighter plug-ins for headphone reverb.

In addition, move mixing plug-ins to native versions where you can. That leaves DSP free for tracking, which is where realtime processing matters most. Linking another Apollo also adds DSP for mixing.

Using talkback and ALT speakers

Talkback lets you speak to performers through their headphones. This matters when the performer is in another room. Press the talkback button, give directions, then release it before the take.

ALT speaker switching is just as useful. Connect a second pair of speakers to the ALT line outputs. Then switch between pairs in Console to hear how a mix translates. For example, compare full-range monitors against small speakers. Problems that hide on one pair often stand out on the other.

Also, keep both pairs at matched levels. Otherwise, the louder pair will usually sound better, and that can mislead you.

Is the Apollo x4 right for you?

The Apollo x4 sits in a useful middle spot. It gives you four preamps and two headphone mixes without a rack. As a result, it suits small studios that record more than one person. On the other hand, solo creators may not need the extra inputs, and big studios may want the x8. Think about how many sources you record at once, now and in the future. If the answer is three or four, the x4 is a strong fit. For more options, compare our best Thunderbolt audio interfaces roundup.

Common mistakes

• Not checking the Thunderbolt controller version. Host requirements differ between Thunderbolt 3 and 4 units.
• Using all four preamps at high gain with Unison saturation. Emulations react to level; too hot sounds distorted.
• Printing reverb. Monitor reverb through Console auxes rather than recording it.
• Double monitoring. Disable DAW input monitoring when Console is monitoring.
• Ignoring phase on multi-mic sources. Check polarity between mics on the same source.
• Monitoring twice. Leaving DAW input monitoring on while using Console causes a doubled sound.
• One headphone mix for two players. Use both outputs with separate cue mixes.
• Clock conflicts with ADAT gear. Set one master clock to avoid clicks.
• Skipping registration. Included plug-ins stay locked until you register.

Setup checklist

1. Confirm your computer meets UA’s Thunderbolt requirements.
2. Install UA Connect and drivers, register the unit.
3. Open Console and name each input.
4. Load Unison preamps and set gain per source.
5. Build headphone mixes and set up talkback.
6. Select the Apollo in your DAW and record a test of every input.

Further reading: Thunderbolt audio interface picks, what a mic preamp does, and the recording hub.

Check the current price of the Universal Audio Apollo x4 on Amazon.

FAQ

When your multitrack session is mixed, our stem mastering can finish it from grouped stems. That gives more control than a single stereo file.

As an Amazon Associate I earn from qualifying purchases. This write-up is based on Universal Audio’s published specifications and documentation, not hands-on testing.

The Apollo x16 is Universal Audio’s 18×20 rack-mount Thunderbolt 3 interface with sixteen channels of line-level analog I/O, HEXA Core DSP and its highest-resolution converters, built for mixing, mastering and immersive work rather than tracking microphones directly. It has no built-in mic preamps; it is designed to connect to consoles, outboard preamps and monitor systems up to 9.1.6.

See the Universal Audio Apollo x16 on Amazon. The linked listing is the Apollo x16 Gen 2 with UAD Analog Classics, the current generation.

Apollo x16 key specs

I/O	18 x 20 over Thunderbolt 3
Analog	16 line inputs and 16 line outputs
Mic preamps	None built in
DSP	HEXA Core
Conversion	D/A rated at 133 dB dynamic range, −129 dB THD (UA figures)
Immersive	Monitoring formats up to 9.1.6 for Dolby Atmos, Auro-3D, Sony 360 Reality Audio
Correction	Apollo Monitor Correction by Sonarworks, stereo and immersive

Source: Universal Audio’s Apollo x16 page.

No preamps: why that is the point

The x16 is a converter and monitoring hub. Studios with a recording console or a rack of outboard preamps already have their mic gain sorted, and want clean conversion for many channels. Sixteen line inputs take those preamps or console direct outs; sixteen outputs feed monitors, summing mixers or outboard processors. If you need to plug a microphone in directly, pair the x16 with an external preamp or choose an Apollo with Unison preamps.

Immersive monitoring

UA says the x16 supports mixing formats up to 9.1.6, which means sixteen speaker channels for formats such as Dolby Atmos. Sixteen analog outputs drive a full immersive speaker layout from one unit, and Apollo Monitor Correction can calibrate both stereo and immersive monitor setups. For the basics of the format, see what Dolby Atmos mixing and mastering involve.

Mixing and mastering use

UA rates the x16’s converters as its best, which matters most where small details are judged: mastering, critical mixing and analog summing. Hybrid engineers can send stems out to analog gear and back without adding another converter.

A hybrid mixing workflow

1. Group the mix into stems in the DAW: drums, bass, music, vocals and effects.
2. Route each stem to a pair of x16 outputs feeding an analog summing mixer or outboard processors.
3. Return the summed or processed signal to x16 inputs and record it back to a new stereo track.
4. Compare the analog print with an in-the-box version at matched loudness before deciding which to keep.

Because every output and return passes through the same high-grade converters, the round trip adds as little as possible beyond what the analog gear contributes. The DSP can still run UAD plug-ins on the DAW side, so hybrid and in-the-box processing can be mixed freely. Level matching matters: analog gear often has a sweet spot for input level, and driving it too hard or too softly changes the sound more than any converter difference.

HEXA Core DSP and expansion

Six DSP chips run UAD plug-ins in Console and in DAWs. UA says up to four x16 units can be linked for 64 channels, or an x16 can be combined with other Thunderbolt Apollos and controlled from a desktop Apollo. UA also offers an x16D variant with Dante networking.

Who should consider the Apollo x16

• Studios with a console or outboard preamps that need many channels of conversion.
• Immersive mixers building a speaker system up to 9.1.6.
• Mastering and hybrid mix engineers working with analog gear.

Who should look elsewhere

• Anyone who needs mic preamps in the interface; see the Apollo x8 or Apollo x4.
• Home studios without line-level gear to connect.
• Users without a Thunderbolt computer.

DAW workflow for immersive mixing

1. Choose the immersive monitor format in Console to match your speaker layout.
2. Route the DAW’s immersive renderer outputs to the matching x16 outputs.
3. Calibrate every speaker for level and delay with Monitor Correction.
4. Check the mix’s stereo and binaural versions as well as the full speaker version, since many listeners will hear a down-mix.

Alternatives

• The Apollo x8 if you need built-in mic preamps.
• The x16D variant if your studio runs on Dante networking.

Planning the room around an x16

Before buying, count the channels you need. A 7.1.4 Atmos layout needs twelve outputs; 9.1.6 needs sixteen. Add the stereo main monitors and any outboard sends, and you may need a second unit or a separate monitor controller. On the input side, list each preamp and line source you want permanently patched. Planning this on paper avoids buying a unit that is either too small or much larger than your room needs.

How to set up the Apollo x16

The Apollo x16 is a rack interface, so setup starts with cabling. Planning the DB-25 snakes first saves a lot of time later.

1. Rack the unit. Give it airflow and keep cables tidy behind the rack.
2. Connect DB-25 snakes. UA lists two DB-25 inputs for lines 1–16 and two DB-25 outputs. The outputs carry lines 1–16 or ALT and surround feeds.
3. Wire the monitors. The main monitor outputs are XLR.
4. Connect Thunderbolt. Use a Thunderbolt 3 or 4 cable to your Mac or Windows computer.
5. Install UA Connect. It installs the driver, Console and plug-in licenses. Register the unit and update firmware.
6. Set clocking. Use the BNC word clock or AES if you integrate other digital gear. Choose one master clock.
7. Pick your monitor mode. Choose stereo or an immersive format in Console. Then calibrate with Monitor Correction if you use it.

Finally, label every snake tail. With 16 channels each way, clear labels prevent routing mistakes.

Apollo x16 vs other Apollo interfaces

This table compares the Apollo x16 with its siblings, based on UA’s published specs. The main difference is preamps versus line-level I/O.

Feature	Apollo x16	Apollo x8	Apollo x4	Apollo Twin X
I/O	18 x 20	18 x 24	12 x 18	10 x 6
Mic preamps	None	4 Unison	4 Unison	2 Unison
DSP	HEXA Core	HEXA Core	QUAD Core	DUO or QUAD
D/A dynamic range	133 dB	130 dB	129 dB	129 dB
Analog connectors	DB-25	1/4″	1/4″	1/4″
Digital I/O	AES, word clock	2 optical, S/PDIF, word clock	Optical	Optical in
Surround monitoring	Up to 9.1.6	Up to 7.1	Not listed	Not listed

Put simply, choose the x16 for line-level mixing and immersive monitoring. Pick the x8 if you need preamps in the same rack unit.

Pairing the x16 with outboard preamps

Because the x16 has no mic preamps, you bring your own. This suits studios that already own favorite preamps. Run each preamp into a DB-25 input line and set the gain on the preamp itself.

Alternatively, link the x16 with an Apollo that has Unison preamps. UA says you can link up to four Thunderbolt Apollos. So an x8 can handle mics while the x16 handles line I/O and monitoring.

Keep in mind that Unison works only on Unison preamps. Outboard preamps sound like themselves, and you can still add UAD plug-ins in Console.

Using the x16 for analog summing and outboard

Sixteen line outputs make the x16 useful for hybrid mixing. You can send stems to an analog summing mixer and return the stereo bus. You can also patch hardware compressors and EQs as inserts.

To keep timing correct, use your DAW’s hardware insert tool. It measures round-trip delay and compensates. Meanwhile, keep levels consistent between outputs and returns so you don’t push gear too hard.

Who the Apollo x16 is for

• Mix and mastering engineers who work at line level.
• Immersive mixers who need monitoring up to 9.1.6.
• Studios with outboard preamps, summing mixers or hardware inserts.
• Rooms expanding an existing Apollo system.

Who should skip it

• Home producers who need built-in mic preamps. The x8 or x4 fits better.
• Anyone without DB-25 snakes or a patchbay plan.
• Laptop users who want a small desktop unit.
• Computers without Thunderbolt.

Pros and cons

Pros	Cons
133 dB D/A dynamic range (UA figure)	No mic preamps
16 line inputs and outputs	Needs DB-25 snakes
Immersive monitoring up to 9.1.6	Overkill for small stereo rooms
HEXA Core DSP	Thunderbolt only
AES and word clock I/O	Monitor Correction needs extra software and a mic

Setting up immersive monitoring

Immersive work needs careful speaker setup. First, place speakers according to the format’s layout guide. Next, choose the matching monitor mode in Console. Then route the DB-25 outputs to each speaker.

UA says Monitor Correction works on stereo and immersive monitor outputs. This helps balance levels and frequency response across many speakers. Also, Bass Management lets you integrate a subwoofer. For background, read our guide to Dolby Atmos mixing and mastering.

In addition, UA lists ALT monitoring in all modes. You get two ALT sets in stereo and one in immersive. That lets you compare a stereo pair against the full array.

Plug-in bundles and DSP use

UA lists the x16 with UAD Analog Classics or UAD Complete. In fact, UA says it can include up to 200+ plug-ins depending on the bundle. The HEXA Core DSP runs those plug-ins in Console or in your DAW through VST, AU and AAX 64.

For mixing, combine DSP and native processing. As a result, large sessions stay stable. Save DSP for plug-ins you want to hear through Console in real time.

For full specs, see UA’s official Apollo x16 page.

A mastering chain with the x16

Mastering engineers often mix hardware and software. The x16 supports that with clean conversion and many line outputs. Here is one way to organize a chain.

1. Play the mix out. Send the stereo mix from two line outputs to your analog EQ.
2. Chain the hardware. Route the EQ into a compressor, then back into two line inputs.
3. Add digital processing. Insert UAD or native limiters on the return track.
4. Compare carefully. Use level-matched A/B checks so loudness does not fool you.

Also, keep notes of every hardware setting. That way, you can recall a master when a client asks for a revision. For more detail, see our mastering guide.

Gain structure at line level

Line-level work still needs good gain staging. First, set your DAW output so peaks leave headroom before they hit the converters. Next, match the level your outboard expects. Then trim the return so it sits near the original level.

Consistent levels matter for two reasons. They keep hardware in its sweet spot. In addition, they make fair comparisons possible, because louder usually sounds better at first.

Is the Apollo x16 worth it for your room?

The answer depends on your work. If you mix, master or work in immersive formats, the x16’s line I/O and monitoring features fit well. If you record bands, you will still need preamps from elsewhere. Therefore, many studios pair it with an x8 or outboard preamps. Think about your next few years of projects before you choose.

Using Plug-In Scenes and Console for sessions

UA’s updated Console includes Plug-In Scenes. You can save a set of Console plug-ins and recall it later. This is handy when one room handles mixing, mastering and immersive work on different days.

For example, save one scene for stereo mastering with your monitor chain. Then save another for immersive mixing with Bass Management. As a result, switching jobs takes seconds rather than minutes of rebuilding.

Likewise, name scenes clearly and back them up. Moreover, keep a short checklist for each one, listing monitor mode, clock source and ALT speakers. Small habits like this keep a complex room predictable, especially when another engineer uses the space.

Common mistakes

• Buying it as a first interface. Without external preamps, you cannot record a mic.
• Skipping calibration in immersive rooms. Speaker levels and delays must be aligned before mixing.
• Clocking confusion. In multi-unit or Dante setups, decide on one clock master.
• Level mismatch with consoles. Match operating levels between the console and x16 to avoid clipping or noise.
• Ignoring Console sessions. Save routing and monitor setups with projects.
• Two clock masters. If the x16 and AES gear both act as master, you get clicks. Choose one.
• Unlabeled snakes. With 32 analog lines, labeling saves hours of troubleshooting.
• Skipping insert delay compensation. Hardware inserts drift out of time without a ping test.
• Expecting Unison on outboard preamps. Unison works only on Apollo Unison inputs.
• Skipping registration. Included plug-ins stay locked until you register the unit in UA Connect.

Setup checklist

1. Rack the unit, connect Thunderbolt 3 and install UA software.
2. Plan the routing: which inputs take preamps or console outs, which outputs feed speakers.
3. Set the monitor format in Console for stereo or immersive work.
4. Run Monitor Correction calibration.
5. Set clock source and test every channel.

Related: Dolby Atmos mastering, Thunderbolt audio interface picks and the mastering hub.

Check the current price of the Universal Audio Apollo x16 on Amazon.

FAQ

When your mixes are finished, our stem mastering can complete them from grouped stems. It is ideal for detailed, multi-source productions.

As an Amazon Associate I earn from qualifying purchases. This write-up is based on Universal Audio’s published specifications and documentation, not hands-on testing.

The Apollo Twin X is Universal Audio’s 10×6 Thunderbolt desktop audio interface with two Unison mic preamps, a Hi-Z instrument input, optical input, and onboard DUO or QUAD Core DSP for running UAD plug-ins in real time. Its main draw is tracking through emulations of classic preamps, compressors and amps with near-zero latency, plus a built-in monitor controller with talkback.

See the Universal Audio Apollo Twin X on Amazon. The linked listing is the current Apollo Twin X DUO Gen 2 bundle with UAD Analog Classics; a QUAD version is also sold.

Apollo Twin X key specs

I/O	10 x 6 over Thunderbolt
Inputs	2 Unison mic preamps, Hi-Z instrument input, optical Toslink input (ADAT or S/PDIF)
Outputs	Two ¼” monitor outs, two ¼” line outs (ALT), one headphone out
DSP	DUO or QUAD Core, depending on model
Conversion	D/A rated at 129 dB dynamic range (UA figure)
Monitoring	Alternate speaker switching, talkback, Apollo Monitor Correction powered by Sonarworks
Compatibility	macOS (including Apple silicon) and Windows via Thunderbolt

Source: Universal Audio’s Apollo Twin X page. Several editions are sold, including bundle editions such as Heritage Edition; the hardware is the same, the included plug-ins differ.

Unison preamps

Unison is UA’s term for preamp plug-ins that work together with the hardware. When you load a Unison preamp emulation, such as a Neve 1073 or API model, in the preamp slot, the physical preamp changes its input impedance and gain behaviour to match the original unit. The idea is that the mic responds as if plugged into that hardware, not just that a plug-in colours the signal after the fact.

Real-time UAD processing

The DSP chips inside the Apollo run UAD plug-ins. Through UA’s Console software you can record through a compressor, EQ, amp sim or reverb with near-zero latency, and choose whether to print the effect or only monitor through it. This is the main reason people buy Apollo over a similarly priced native interface. UA says more than 200 UAD plug-ins are available, though only a starter set is included.

DUO and QUAD refer to the number of DSP chips. QUAD runs roughly twice as many DSP plug-ins at once. For one or two singers recording through a preamp, compressor and reverb, DUO is often enough; heavier chains or mixing with UAD plug-ins favour QUAD.

Monitor controller and correction

The large knob controls monitor level, and the ALT outputs let you switch to a second pair of speakers. Talkback lets you speak to a performer in another room. Apollo Monitor Correction, powered by Sonarworks, calibrates monitor and headphone outputs, which helps when mixing in an untreated room. Check what is included versus requires setup on UA’s site.

Expansion

The optical input adds eight channels from an ADAT preamp. UA says up to four Thunderbolt Apollo interfaces can be linked, so a Twin X can grow into a larger rig, controlled from the desktop unit.

Who should consider the Apollo Twin X

• Vocalists and producers who want to track through classic preamp and compressor emulations.
• Guitarists who want amp simulations without latency.
• Small studios that want monitor control and talkback on the desktop.

Who should look elsewhere

• Computers without Thunderbolt. UA sells a USB version of the Twin X for Windows; check which one matches your machine.
• Users who do not plan to use UAD plug-ins; a native interface without DSP will do the same recording job.
• Anyone needing more than two mic preamps without adding ADAT.

DAW workflow with Console

1. Open Console, load a Unison preamp on input 1 and add a compressor and EQ for the singer to hear while recording.
2. Decide whether to record those effects (Rec) or monitor them only (Mon), so you know what is printed.
3. Add a reverb on an aux for headphone comfort, without recording it.
4. In the DAW, disable input monitoring for tracks being monitored through Console.
5. When mixing, use UAD plug-ins inside the DAW on the same DSP.

Alternatives: the Volt 476 for four inputs without DSP, or the Apollo x4 for four Unison preamps.

How to set up the Apollo Twin X

Setup takes a few steps, and doing them in order saves time. First, check that your computer has a Thunderbolt port. UA lists support for Thunderbolt 3 and 4 on Mac and Windows.

1. Connect power and Thunderbolt. The Twin X uses its own power supply. Plug in the Thunderbolt cable before you install software.
2. Install UA Connect. Download it from UA’s site. It installs the driver, Console and your plug-in licenses.
3. Register the unit. Sign in to your UA account so the included plug-in bundle unlocks.
4. Update firmware. UA Connect prompts you when an update is available. Let it finish before you unplug anything.
5. Pick the Apollo in your DAW. Choose it as the input and output device in the audio settings.
6. Open Console. Set up your input channels, Unison preamps and monitor mix here.
7. Calibrate, if you use correction. Monitor Correction needs SoundID Reference and a compatible measurement mic.

After that, run a short test recording. Check that the signal reaches the DAW and that monitoring sounds right.

Choosing DUO or QUAD for the Apollo Twin X

The Apollo Twin X comes with DUO or QUAD Core DSP. The difference is how many UAD plug-ins you can run at once on the unit. The I/O is the same on both.

DUO suits a singer-songwriter who tracks one or two sources through a preamp and a compressor. QUAD gives more headroom for heavier chains, such as a channel strip plus reverb on several inputs. It also helps if you mix with UAD plug-ins inside the DAW.

Remember that DSP is not the only processing you have. UA describes a hybrid workflow, where DSP handles tracking and your computer’s native plug-ins handle the rest. So choose based on how many plug-ins you want while recording, not the full mix.

Apollo Twin X vs other Universal Audio interfaces

The table below compares the Twin X with nearby models, using UA’s published specs. It helps you see where the Twin X sits in the lineup.

Feature	Apollo Twin X	Apollo Solo	Apollo x4	Volt 476
I/O	10 x 6	2 x 4	12 x 18	4 x 4
Mic preamps	2 Unison	2 Unison	4 Unison	2 (Vintage mode)
DSP	DUO or QUAD	SOLO	QUAD	None
Optical	Input	No	Input and output	No
Headphone outs	1	1	2	1
Talkback and ALT	Yes	No	Yes	No
Connection	Thunderbolt	Thunderbolt (USB version on Windows)	Thunderbolt	USB-C

In short, the Twin X adds DSP, optical expansion and monitor control over the Solo. The x4 adds more preamps and a second headphone output. The Volt 476 skips DSP entirely but offers MIDI and USB-C.

Expanding inputs with the optical port

The optical Toslink input accepts ADAT or S/PDIF. With an ADAT preamp, you can add up to eight extra inputs at standard sample rates. That is a simple way to record a drum kit later without replacing the Twin X.

To set it up, connect the preamp’s optical output to the Twin X. Next, choose ADAT or S/PDIF in Console settings. Then set the clock so both units stay in sync. Usually the Apollo acts as the clock source, and the preamp follows it.

Keep in mind that ADAT channel counts drop at higher sample rates. Check your preamp’s manual if you work at 88.2 kHz or above.

Who the Apollo Twin X is for

• Producers who want to track through UAD plug-ins with no audible latency.
• Vocalists and guitarists who record one or two sources at a time.
• Mixers who want accurate monitoring with Monitor Correction and ALT speakers.
• Anyone planning to grow through optical or by linking more Apollos.

Who should skip it

• Bands that track several mics at once from day one. Look at the x4 or Apollo x8 instead.
• Computers without Thunderbolt.
• Users who never plan to use UAD plug-ins. A non-DSP interface makes more sense.
• Anyone who needs two headphone outputs on the unit itself.

Pros and cons

Pros	Cons
Realtime UAD processing while tracking	Thunderbolt only
Two Unison preamps plus Hi-Z input	One headphone output
129 dB D/A dynamic range (UA figure)	DSP limits plug-in counts on DUO
Talkback, ALT outs and Monitor Correction	Correction needs extra software and a mic
Optical input and Apollo linking	Needs its own power supply

Gain staging with Auto-Gain

UA’s Console now includes Auto-Gain. You ask the performer to sing or play at full level, and Console sets the preamp gain for you. This is useful when you record yourself and can’t watch meters.

Even so, check the result. Leave some headroom, because a performer often gets louder during a real take. If peaks get close to clipping, lower the gain a little.

Also, remember that Unison settings change the tone at different gain levels. So set the gain first, and then judge the sound. For more vocal tips, see our guide to recording vocals at home.

Linking more Apollos later

UA says you can link up to four Thunderbolt Apollo interfaces, for up to 128 channels. As a result, the Twin X can stay on your desk as a controller while a bigger rack unit adds inputs. Dante networking is available through the Apollo x16D.

When you link units, their DSP combines for mixing. However, plan your tracking chains around the unit each input lives on. For the full spec list, check UA’s official Apollo Twin X page.

A simple vocal tracking chain

A basic chain helps you start fast. Then you can adjust it as you learn the plug-ins. Here is a starting point using plug-ins UA lists in the Analog Classics bundle.

1. Preamp: Load the UA 610-B in the Unison slot. Set gain so peaks sit with headroom.
2. Compressor: Add the Teletronix LA-2A. Aim for a few dB of gain reduction on loud phrases.
3. Reverb on an aux: Send a little reverb to the headphones only. That way the singer hears space, but the dry track is recorded.

Decide carefully what you print. Console lets you record the processed signal or monitor it while recording a dry track. If you are new to compression, print light settings or monitor only.

Mixing on headphones with the Twin X

Not everyone has a treated room. For that reason, UA lets Monitor Correction calibrate both the main monitor outputs and the headphone output. This needs SoundID Reference and, for speakers, a measurement mic.

Even with correction, check mixes on other systems. For example, try earbuds, a car or a small speaker. Each check reveals problems that one setup can hide. Also, take breaks, because tired ears make poor mix decisions.

Common mistakes

• Printing effects you are unsure about. Console can record dry while you monitor wet. Commit only when you are sure.
• Running out of DSP. Each plug-in uses DSP. Check the DSP meter in Console before adding more.
• Monitoring twice. If Console and your DAW both monitor the input, you hear a delayed double. Mute DAW input monitoring.
• Mismatched Thunderbolt versions. UA notes that some units ship with Thunderbolt 4 controllers that have specific host requirements. Check before buying.
• Ignoring gain staging in Unison. Emulations react to level like the hardware; too much gain can distort.
• Monitoring twice. If DAW input monitoring is on while you also hear Console, you get a doubled, phasey sound. Turn one off.
• Forgetting the clock with optical gear. Mismatched clocks cause clicks and pops. Set one clock source.
• Overloading DUO DSP. Heavy chains on every input can run out of DSP. Print simpler chains or use native plug-ins for mixing.
• Skipping registration. Included plug-ins stay locked until you register the unit.

Setup checklist

1. Install UA Connect and the drivers, then register the unit.
2. Connect by Thunderbolt and open Console.
3. Load a Unison preamp on input 1 and set gain while performing.
4. Add a compressor in the insert slots and decide whether to print or monitor only.
5. Select the Apollo in your DAW and turn off DAW input monitoring.
6. Set up Monitor Correction if you plan to mix on this system.

For more on choosing a Thunderbolt unit, see our Thunderbolt audio interface picks and the general guide on how to choose an audio interface. More tracking guides live in the recording hub.

Check the current price of the Universal Audio Apollo Twin X on Amazon.

FAQ

When your mix is done, our single track mastering can finish it for streaming and release. Send a stereo mix with peaks below 0 dBFS.

As an Amazon Associate I earn from qualifying purchases. This write-up is based on Universal Audio’s published specifications and documentation, not hands-on testing.

The Apollo Solo is Universal Audio’s smallest Apollo: a bus-powered 2×4 Thunderbolt 3 desktop interface with Unison mic preamps, 24-bit/192 kHz conversion and a UAD-2 SOLO Core DSP chip for recording through UAD plug-ins in real time. A USB-C version, Apollo Solo USB, offers the same idea for computers without Thunderbolt.

See the Universal Audio Apollo Solo on Amazon. The linked listing is the Apollo Solo USB Heritage Edition; a Thunderbolt version also exists.

Apollo Solo key specs

I/O	2 x 4, 24-bit/192 kHz
Preamps	Unison mic preamps, Hi-Z instrument input
DSP	UAD-2 SOLO Core (one SHARC chip)
Connection	Thunderbolt 3, bus-powered (cable sold separately); Apollo Solo USB uses USB-C
Software	Realtime Analog Classics bundle, LUNA compatibility

Source: Universal Audio’s Apollo Solo page.

What the SOLO Core can and cannot do

The SOLO Core is a single DSP chip. It handles a Unison preamp plus a modest chain, such as a compressor and an EQ, on a vocal while tracking. It is not designed to run large mix sessions full of UAD plug-ins; for that, a DUO or QUAD unit like the Apollo Twin X has more headroom. Many Solo owners record through UAD plug-ins and mix with native plug-ins on the computer.

Unison preamps

Unison links the preamp plug-in to the hardware. When you load an emulation of a classic preamp, the Solo’s physical preamp adjusts its impedance and gain behaviour to match. The result is printed into your recording as part of the input path. The Realtime Analog Classics bundle includes the UA 610-B preamp and EQ, Teletronix LA-2A and Fairchild compressors, and guitar amp emulations, according to UA.

A practical vocal chain on the Solo

A sensible starting point for vocals is a Unison preamp, a gentle compressor and a reverb used only for monitoring:

1. Preamp: load the included 610-B emulation in the Unison slot and set gain so the loudest phrases peak well below clipping.
2. Compressor: add an LA-2A-style compressor and aim for a few decibels of gain reduction on loud lines.
3. Reverb for the singer: add a reverb on an aux in Console so the singer hears some space, but keep it out of the recording.

This uses a small amount of DSP and gives a recorded vocal that already sounds finished enough to write and arrange around, while leaving the big decisions for the mix. If the DSP meter climbs too high, remove the compressor and add it later as a native plug-in.

The same approach works for guitar: a Hi-Z input into an amp emulation, monitored through Console, with a clean DI recorded at the same time if your setup allows, so you can change the amp sound later.

Thunderbolt or USB

The Thunderbolt version of the Apollo Solo connects by Thunderbolt 3 and is bus-powered from the computer, which makes it neat for laptops. The Thunderbolt cable is not included. UA also sells Apollo Solo USB with the same conversion, preamps and DSP over USB-C. Pick the version that matches the ports on your computer, and check UA’s compatibility list for your operating system.

Headphone output

UA describes the headphone amplifier as louder, lower-noise and flatter than others in its class. For anyone who tracks and mixes on headphones, a strong headphone output is useful; still, judge with your own headphones and check level matching when comparing.

Who should consider the Apollo Solo

• Singers and solo producers recording one source at a time.
• Laptop users who want a compact interface with UAD processing.
• Anyone curious about Unison and UAD workflow before buying a larger Apollo.

Who should look elsewhere

• Producers wanting to mix with many UAD plug-ins at once; more DSP is needed.
• Duos or bands needing several mic inputs at once.
• Users who do not intend to use UAD plug-ins; a simpler interface does the same job.

DAW workflow for a solo artist

A typical session uses one Unison preamp for the vocal and the Hi-Z input for guitar or bass. In Console, monitor with a light compressor and reverb, record dry, then open the same UAD plug-ins in the DAW to mix. Because SOLO Core DSP is limited, check the DSP meter before stacking processing. Alternatives: the Apollo Twin X for two preamps and more DSP, or a Volt 276 if you do not need DSP.

Apollo Solo vs Apollo Solo USB: pick the right one

This choice matters more than any other with the Apollo Solo. The two versions share conversion, Unison preamps and SOLO Core DSP. However, they differ in connection, power and platform.

Point	Apollo Solo (Thunderbolt 3)	Apollo Solo USB
Connection	Thunderbolt 3 (cable sold separately)	USB-C
Power	Bus powered over Thunderbolt	External power supply required
Platforms	Mac and Windows with Thunderbolt 3 or 4	Windows only
Older Macs	Backward compatible with Thunderbolt 1 and 2 on Mac, per UA	Not for Mac

So, Mac users should buy the Thunderbolt model. Windows users without Thunderbolt need the USB model. The Amazon listing above is the USB Heritage Edition, so check your computer before you order.

How to set up the Apollo Solo step by step

The Apollo Solo needs UA’s software before it will pass audio. Follow these steps in order for a smooth first session.

1. Install UA Connect. Download it from UA’s site. It installs the drivers, Console and your plug-ins.
2. Connect the interface. Use a Thunderbolt 3 cable for the Thunderbolt model. For the USB model, connect USB-C and the power supply.
3. Update firmware if prompted. Let the update finish before you unplug anything.
4. Register the unit. This unlocks the bundled plug-ins in your UA account.
5. Open Console. You should see two preamp channels plus the Hi-Z input on the front.
6. Select the Apollo in your DAW. It appears as a Core Audio device on Mac and an ASIO device on Windows.
7. Set your sample rate. 44.1 or 48 kHz suits most work. Higher rates use more DSP per plug-in.

Buffer size and latency on an Apollo

Apollo works differently from most interfaces. When you monitor through Console, UAD plug-ins run on the DSP chip. As a result, UA says tracking happens at near-zero latency regardless of the audio buffer size.

In practice, that means you can leave the DAW buffer at 256 or 512 samples while recording. Your computer stays stable, and the singer still hears no delay. The catch is that native DAW plug-ins and virtual instruments still follow the buffer. So, if you play a software synth live, drop the buffer to 64 or 128 samples.

Routing in Console: what gets recorded

Console gives each input an insert section. A simple rule keeps your recordings predictable.

• Unison preamp slot: always recorded, because it is part of the input path.
• Inserts set to Rec: recorded into your DAW.
• Inserts set to Mon: heard in headphones only, so the track stays dry.
• Aux sends: heard only, ideal for reverb and delay.

Write down which inserts you printed. Later, you will know whether a vocal already has compression on it. For broader context, see our guide on recording vocals at home.

Managing the SOLO Core DSP

The SOLO Core is one SHARC chip. UA says it can run several plug-ins at once while tracking. Even so, it fills up faster than DUO or QUAD units. These habits stretch it further.

• Use one Unison preamp and one compressor per live input.
• Put a single reverb on an aux instead of one per channel.
• Pick lighter “Legacy” versions of plug-ins when DSP runs low.
• Track at 44.1 or 48 kHz, since higher rates cost more DSP.
• Print or bounce UAD effects before you mix with more of them.

UA also notes you can add a UAD-2 Satellite or another Apollo over Thunderbolt for more mixing DSP. However, Satellite DSP does not add power for real-time tracking.

Apollo Solo vs similar interfaces

This table compares the Solo with close options using published specs from UA.

Feature	Apollo Solo	Apollo Twin X	Apollo x4	UA Volt 276
I/O	2 x 4	10 x 6	12 x 18	2 x 2
Mic preamps	2 Unison	2 Unison	4 Unison	2, with 610 mode
DSP	SOLO Core	DUO or QUAD	QUAD Core	None
Optical input	No	Yes	Yes (I/O)	No
Talkback and ALT speakers	No	Yes	Yes	No
Connection	Thunderbolt 3 or USB-C	Thunderbolt	Thunderbolt	USB-C

To sum up, the Solo is the entry ticket to Unison and UAD. The Twin X adds DSP, optical input and monitor control. Meanwhile, the Volt 276 gives analog character without DSP.

Apollo Solo pros and cons

Pros

• Unison preamps let you track through classic preamp emulations.
• Near-zero latency UAD processing while tracking.
• Bus powered on the Thunderbolt model, with an all-metal body.
• Strong headphone amp, according to UA.
• Bundled plug-ins cover preamps, compressors, EQ, amps and reverb.
• Can expand via Thunderbolt daisy-chaining.

Cons

• A single DSP chip limits big chains and UAD mixing.
• No optical input, talkback or second speaker output.
• Thunderbolt cable not included.
• The USB model is Windows only and needs a power supply.
• Large plug-in downloads take disk space.

Choosing a Unison preamp for each source

Unison preamps change how the input stage behaves, so the choice shapes your tone. Start with the UA 610-B, which UA lists in its Realtime Analog Classics bundle, then try others as you learn. These are sensible starting points, not rules.

• Vocals: A tube-style preamp such as the 610-B adds warmth. Keep the gain moderate, because pushing it adds saturation.
• Acoustic guitar: Aim for a cleaner setting so the strings stay detailed.
• Bass DI: The Hi-Z input with the Ampeg SVT-VR Classic (part of the Realtime Analog Classics bundle) gives an amp tone. Record a clean DI as well if your session allows.
• Electric guitar: Use the Hi-Z input with the Marshall Plexi Classic, if your bundle includes it, for monitoring. Then decide whether to print it.

Because Unison is printed, set it while listening on headphones. If you are unsure, choose the cleaner option and add color in the mix.

Setting up headphone monitoring

The Solo has one headphone output, so it suits a single performer. Console builds the headphone mix from your inputs, auxes and DAW playback.

1. Set DAW input monitoring off for armed tracks.
2. In Console, raise the input fader until the voice sits well against the track.
3. Send a little reverb from the aux for comfort.
4. Adjust the headphone level with the front knob.

UA describes the headphone amp as louder and lower in noise than others in its class. Even so, protect your ears and keep levels moderate during long sessions.

Quick decision guide

Choose the Apollo Solo if you record one or two sources and want to track through UAD plug-ins. Choose the Twin X if you need more DSP, an optical input or talkback. Choose a non-DSP interface if you never plan to use UAD plug-ins. For full details, see UA’s official Apollo Solo page.

Laptop and travel tips

The Thunderbolt Solo is popular for mobile recording, since it is bus powered. A few habits keep remote sessions reliable.

• Pack your own Thunderbolt 3 cable, because UA does not include one.
• Plug the laptop into mains power during long sessions when possible.
• Save often and avoid letting the laptop sleep mid-session.
• Install only the plug-ins you need on a small laptop drive.
• Open Console and test inputs before the performer arrives.

With these steps, the Solo makes a compact travel rig for vocals and DI guitar.

Common mistakes

• Assuming the cable is in the box. UA notes the Thunderbolt 3 cable is sold separately.
• Overloading the single DSP chip. Watch the DSP meter in Console. If it fills up, print the effect or move processing to native plug-ins.
• Buying the wrong connection type. Thunderbolt and USB versions are separate products.
• Double monitoring. Monitor through Console and disable input monitoring in the DAW.
• Printing heavy effects. Record through a preamp and light compression; keep reverb for monitoring only.
• Buying the USB model for a Mac. UA lists Apollo Solo USB as Windows only.
• Dropping the buffer for no reason. Console tracking is near-zero latency at any buffer, so keep it higher for stability.
• Skipping registration. Bundled plug-ins stay locked until you register the unit in your UA account.

Setup checklist

1. Install UA Connect and drivers, then register the unit to unlock bundled plug-ins.
2. Connect with a Thunderbolt 3 cable (or USB-C for the USB model).
3. Open Console, load a Unison preamp and set gain.
4. Add a compressor; choose whether to record it or monitor only.
5. Select the Apollo in your DAW and turn off DAW input monitoring.
6. Record and listen back.

Related reading: Thunderbolt audio interface picks, how to record vocals at home and the recording hub.

Check the current price of the Universal Audio Apollo Solo on Amazon.

FAQ

Once your song is mixed, our single track mastering can finish it for release. Send the final mix without a limiter on the master.

As an Amazon Associate I earn from qualifying purchases. This write-up is based on PreSonus’s published specifications and documentation, not hands-on testing.

The PreSonus AudioBox USB 96 is a bus-powered 2×2 USB 2.0 audio interface with two combo mic/instrument inputs, 48V phantom power, 24-bit/96 kHz conversion, balanced outputs, a headphone output and 5-pin MIDI in and out. It is a simple, low-cost interface for beginners, singer-songwriters and podcasters who record one or two sources at a time. The current version is sold as the 25th Anniversary edition.

See the PreSonus AudioBox USB 96 on Amazon. The linked listing is the AudioBox 96 25th Anniversary edition.

PreSonus AudioBox USB 96 key specs

I/O	2 in / 2 out, USB 2.0, class-compliant
Inputs	2 combo XLR / ¼” mic and instrument inputs
Mic preamp gain	0 to 35 dB (PreSonus 25th Anniversary spec sheet)
Instrument gain	+5 to 50 dB
Phantom power	+48V
Conversion	24-bit, 44.1–96 kHz, 105 dB dynamic range (A-weighted)
Outputs	2 balanced ¼” TRS, 1 headphone output (60 mW/ch at 60 Ω)
MIDI	Dual 5-pin DIN in and out
Power	USB bus-powered

Source: PreSonus’s AudioBox USB 96 25th Anniversary page. Older owner’s manuals list a different gain range, so check the sheet for the edition you buy.

Preamp gain and microphone choice

The 25th Anniversary specifications list 35 dB of mic preamp gain. That is fine for condenser microphones, which have high output, and for loud sources like guitar amps or drums. Low-output dynamic microphones on quiet speech may need more gain than this, which is why some podcasters pair such mics with an inline booster or choose an interface with more gain. If you plan to use a condenser mic, the 48V phantom power has you covered.

Getting good results from a two-input interface

At this level, the interface is rarely what holds a recording back. A few habits matter more:

• Treat the space you record in. A vocal recorded in a closet full of clothes often sounds better than one recorded in an empty room with a more expensive interface.
• Get close to the source. Moving the mic closer raises level without adding preamp noise.
• Record at 24-bit, 44.1 or 48 kHz. Higher sample rates increase file size and CPU load with little benefit for most home projects.
• Set a sensible buffer. Use a small buffer while recording and a larger one while mixing.

Mix control for latency-free monitoring

The Mix knob blends what you are recording with what the computer is playing back. Turn it towards the inputs to hear yourself directly with no delay, and towards playback to hear the track. This is the simplest way to avoid the distracting delay you hear when monitoring through the DAW with a large buffer.

MIDI I/O

Few interfaces at this level include 5-pin MIDI. It lets you connect an older keyboard, a drum machine or a synth directly, without a separate MIDI interface. Modern USB MIDI controllers connect to the computer directly and do not need it.

Build and compatibility

PreSonus describes a steel chassis built for travel. The interface is class-compliant, so no driver install is needed on macOS, and PreSonus lists compatibility with Apple silicon Macs and iOS/iPadOS devices. Windows users can install PreSonus’s driver for lower latency.

Included software

Registration includes a six-month trial of Fender Studio Pro recording software. A separate AudioBox 96 Studio bundle adds headphones and a microphone and lists a Fender Studio Pro license. Check what your bundle includes.

Who should consider the AudioBox USB 96

• Beginners who want a simple two-input interface.
• Singer-songwriters recording voice and guitar.
• Musicians with 5-pin MIDI gear.

Who should look elsewhere

• Podcasters using low-output dynamic mics who want more gain.
• Anyone wanting higher sample rates, loopback or more inputs; see the PreSonus Studio 24c for a USB-C step up.
• Users who need DSP effects while tracking.

DAW workflow: first recordings

1. Install your DAW and select the AudioBox as input and output device.
2. Create two mono tracks, one for input 1 and one for input 2.
3. Set the project to 48 kHz, 24-bit and a buffer of 128 samples while recording.
4. Record a few takes, then raise the buffer for mixing.

Alternatives: the Volt 276 for built-in compression, or a Focusrite Vocaster model for podcasting.

Recording at 24-bit

The AudioBox records at 24-bit, which gives plenty of room below digital full scale. You do not need to push levels close to the top; leave headroom and normalise later if needed.

How to set up the PreSonus AudioBox USB 96

The PreSonus AudioBox USB 96 is one of the easiest interfaces to get running. Even so, the right order of steps avoids most beginner problems. Follow this setup before your first session.

1. Connect it by USB. The interface is bus powered over USB 2.0, so one cable carries both audio and power.
2. Handle drivers. It is class compliant, so macOS needs no driver. On Windows, install PreSonus’s driver and Universal Control from the product page for lower latency.
3. Choose the device in your DAW. Set the AudioBox as both input and output. On Windows, select its ASIO driver rather than a generic one.
4. Set sample rate and bit depth. PreSonus lists 44.1, 48, 88.2 and 96 kHz at 24-bit. For most music, 44.1 or 48 kHz at 24-bit is ideal.
5. Pick a tracking buffer. Start at 128 samples. If you hear clicks, raise it one step.
6. Lower all volumes first. Turn down the main and headphone levels before powering your speakers.

Next, plug in a mic or guitar and check that your DAW meter moves. If not, confirm the track’s input matches the jack you used.

Connecting mics, guitars and keyboards

Each front combo jack takes two kinds of plug. According to PreSonus, the XLR part is a balanced mic input. Meanwhile, the 1/4-inch part is an unbalanced TS instrument input. Here is how to use them well.

• Microphones: Use an XLR cable. Switch on 48V phantom power only for condensers.
• Electric guitar or bass: Use a standard TS instrument cable into the 1/4-inch socket. PreSonus lists +5 to 50 dB of instrument gain.
• Keyboards and synths: Connect left and right outputs to inputs 1 and 2. Start with the gain low, because keyboards are much hotter than guitars.

Note that the 48V switch powers both mic inputs at once on many small interfaces. So, check the manual before pairing a condenser with a ribbon mic.

Setting gain the right way

Good gain staging matters more than any interface spec. First, have the performer play or sing at their loudest. Then raise the gain until your DAW meter peaks near -12 to -6 dBFS. Because the unit records at 24-bit, there is plenty of room below full scale. As a result, you lose nothing by leaving headroom.

The mic preamps offer 0 to 35 dB of gain, per PreSonus. If you max out the knob and the signal is still weak, move closer to the mic. Alternatively, use a more sensitive condenser or an inline booster. Our guide on choosing an audio interface explains how gain needs change by mic type.

Using the Mix knob for zero-latency monitoring

The Mix knob is your main monitoring tool. It blends the live inputs with computer playback, with no latency on the input side.

1. Arm a track in your DAW, but switch off software input monitoring.
2. Put on headphones and start the backing track.
3. Turn the Mix knob toward the inputs until you hear yourself clearly.
4. Then nudge it back toward playback until the balance feels natural.

Singers often want a little more of their own voice than feels “right” in a mix. That is normal, and it helps with pitch.

A first-session workflow for singer-songwriters

A common use is voice plus acoustic guitar. Here is a simple, repeatable way to record both.

• Place a condenser on input 1 for vocals, about a hand’s width from the mouth with a pop filter.
• Put a second mic on input 2, aimed near where the guitar neck meets the body.
• Create two mono tracks in your DAW, one per input.
• Record a click or guide track first, then perform both parts together.
• If spill between mics is a problem, record guitar first and add the vocal later.

Recording parts separately gives cleaner tracks and more mixing control. Our recording hub covers mic placement in more depth.

PreSonus AudioBox USB 96 vs similar interfaces

This comparison uses each maker’s published specs. It shows where the USB 96 fits among simple two-input boxes.

Feature	AudioBox USB 96	AudioBox iTwo	PreSonus Studio 24c	UA Volt 276
Inputs	2 combo mic/instrument	2 combo, line/instrument switch	2 combo mic/line/instrument	2 mic/line/instrument
Mic gain	0 to 35 dB	0 to 52 dB	See PreSonus specs	See UA specs
Max sample rate	96 kHz	96 kHz	192 kHz	192 kHz
Connector	USB 2.0	USB 2.0 and iPad	USB-C	USB-C
Monitoring	Variable Mix knob	Fixed 50/50 Direct button	Hardware monitoring	Direct Monitoring
MIDI	5-pin in/out	5-pin in/out	5-pin in/out	In/out
Extras	Steel chassis	iPad recording app	LED ladder meters	76 compressor, 610 mode

In short, the USB 96 is the plainest option. The iTwo adds preamp gain and iPad focus, while the 24c and Volt 276 move to USB-C and higher sample rates.

PreSonus AudioBox USB 96 pros and cons

Pros

• Class compliant, so it works on Mac and iPad without drivers.
• Variable Mix knob for zero-latency monitoring.
• 5-pin MIDI in and out, rare at this level.
• Steel chassis weighing 0.59 kg, per PreSonus.
• 105 dB converter dynamic range on input and output.

Cons

• Only 35 dB of mic gain, low for quiet dynamic mics.
• Tops out at 96 kHz.
• USB 2.0 connector, not USB-C.
• No loopback, metering screen or DSP effects.

Headphones and studio monitors

PreSonus rates the headphone output at 60 mW per channel into 60 ohms. That is enough for most closed-back studio headphones. However, very high-impedance models may not get as loud, so check their rating before you buy.

For speakers, use the two balanced 1/4-inch TRS main outputs. Balanced TRS-to-TRS or TRS-to-XLR cables reduce hum, especially over longer runs. Place the monitors at ear height, forming an equal triangle with your head. Then set their own volume controls so the main knob sits around halfway at a normal level.

Recording on an iPad or iPhone

PreSonus lists native compatibility with iOS and iPadOS devices. Because the unit is bus powered, though, it draws power from the connected device. Therefore, use a suitable adapter for your device’s port, and check PreSonus’s notes on powering it from a mobile device.

Once connected, open your recording app and allow microphone access in Settings. Then arm a track and check that input 1 or 2 shows signal.

Troubleshooting common problems

• No sound in headphones: Check the headphone level and the Mix knob position. Then confirm your DAW output is set to the AudioBox.
• Clicks and pops: Raise the buffer size. Also close other audio apps and use a direct USB port.
• Hum or buzz: Use balanced cables to your monitors. Move the interface away from power bricks.
• Very quiet vocal: Move closer to the mic, or switch to a condenser with phantom power.
• Guitar sounds distorted: Lower the instrument gain, especially with active pickups.

For full specs and downloads, see the official PreSonus AudioBox USB 96 page.

Quick decision guide

Pick the AudioBox USB 96 if you want a plain two-input box for vocals, guitar and MIDI gear. It suits beginners who value a simple layout over extra features. On the other hand, skip it if you podcast with quiet dynamic mics, since 35 dB of gain is limiting. Likewise, streamers who need loopback should look elsewhere. Finally, anyone planning to record a drum kit or full band will outgrow two inputs quickly, so a four- or eight-input interface is a better long-term choice.

Common mistakes

• Leaving phantom power on with sensitive gear. Turn it off before connecting or disconnecting mics that do not need it.
• Turning the gain fully up. If you need all the gain, move closer to the mic or consider a booster.
• Hearing yourself twice. If the Mix knob is on inputs and DAW monitoring is on, you will hear a delayed copy.
• Using an underpowered hub. Bus-powered devices need a reliable USB port.
• Recording too hot. Keep peaks well below the clip light.
• Plugging a keyboard in at high gain. Line-level sources are hot, so start the knob low.
• Skipping the Windows driver. The generic driver adds latency. Install PreSonus’s driver instead.

Setup checklist

1. Connect to your computer by USB and select the AudioBox in your DAW.
2. Plug in a mic or instrument and enable 48V for condenser mics.
3. Set gain while performing at your loudest.
4. Set the Mix knob for comfortable direct monitoring.
5. Connect monitors to the balanced outputs and record a test.

For more guidance, see how to choose an audio interface, do you need an audio interface, and our recording hub.

Check the current price of the PreSonus AudioBox USB 96 on Amazon.

FAQ

Once your song is mixed, our single track mastering can prepare it for release. Send the final stereo mix with some headroom.

As an Amazon Associate I earn from qualifying purchases. This write-up is based on PreSonus’s published specifications and documentation, not hands-on testing.

The PreSonus AudioBox iTwo is a bus-powered 2×2 USB audio and MIDI interface for macOS, Windows and iPadOS with two Class A mic preamps, two switchable line/instrument inputs, 48V phantom power, 24-bit/96 kHz conversion and 5-pin MIDI I/O. Its focus is mobile recording: it works with PreSonus’s Capture Duo app on iPad and can send sessions to the desktop for mixing.

See the PreSonus AudioBox iTwo on Amazon. The linked listing is the AudioBox iTwo Studio bundle, which adds headphones, a microphone and software to the interface; bundle contents may differ from what is described here.

PreSonus AudioBox iTwo key specs

I/O	2 x 2 USB 2.0 and iPad
Inputs	2 combo inputs with Class A mic preamps; switchable line/instrument
Mic gain	0 to 52 dB (owner’s manual)
Phantom power	+48V
Conversion	24-bit / 96 kHz
Outputs	Balanced ¼” main outputs, headphone output with its own level
MIDI	5-pin in and out
Weight	1.35 lb (0.62 kg), metal chassis

Sources: PreSonus’s AudioBox iTwo page and the iOne/iTwo owner’s manual.

Recording on iPad

The iTwo is MFi (Made for iOS) compliant. On iPad, PreSonus’s free Capture Duo app records two tracks at up to 24-bit/96 kHz, with level and pan controls and basic editing. Sessions can be sent wirelessly to PreSonus’s desktop software for editing and mixing. That makes the iTwo useful for capturing rehearsals, demos or interviews without a laptop.

One known setup point: iPadOS must allow the recording app to use the microphone, even when the audio comes from the interface. If an app shows no input, check that Microphone access is enabled for it in iPad Settings.

Preamps and gain

The owner’s manual lists 0 to 52 dB of mic gain from Class A preamps, which covers condensers and most dynamic mics on singing or instruments. Each input switches between line and instrument level, so a keyboard or a guitar can be connected directly.

Zero-latency monitoring

An internal analog mixer lets you blend input and playback without delay, and the headphone output has its own level control. Monitor through the hardware while recording and turn off software monitoring to avoid hearing a delayed double.

Included software

PreSonus now bundles Fender Studio Pro recording software for macOS and Windows, alongside the Capture apps for iPad. Check the current bundle on PreSonus’s page.

A mobile demo workflow

1. Set up the iTwo with a vocal mic on input 1 and a guitar or keyboard on input 2.
2. Record the idea into Capture Duo on iPad, keeping levels conservative.
3. Send the session to your computer and open it in your DAW.
4. Replace parts, add arrangement and mix, using the same interface at your desk.

Because the same interface is used in both places, the input sound stays consistent between the demo and the studio session, which makes it easier to keep parts from the first take.

How the iTwo compares

• Versus the AudioBox USB 96: both are 2×2 with MIDI, but the iTwo is aimed at iPad recording and its manual lists more mic gain.
• Versus the Studio 24c: the 24c uses USB-C, records up to 192 kHz and has LED meters, but focuses on computer use.

Who should consider the AudioBox iTwo

• Musicians who want to record on iPad as well as a computer.
• Songwriters capturing demos and rehearsals on the go.
• Users with 5-pin MIDI gear.

Who should look elsewhere

• Computer-only users who would prefer USB-C and higher sample rates.
• Anyone needing more than two inputs.
• Podcasters wanting loopback and phone integration.

Using the iTwo on a computer

On macOS and Windows the iTwo works as a standard two-input interface. Select it as the audio device in your DAW, set 48 kHz and a small buffer while recording, and use its analog blend for latency-free monitoring. The 5-pin MIDI ports appear as a MIDI device, so an older keyboard or drum machine can be recorded as MIDI.

Alternatives

• The AudioBox USB 96 if you only record on a computer.
• The Volt 276 for USB-C and a built-in compressor.
• A Focusrite Vocaster Two if your main use is a two-person podcast.

Who the iTwo suits best

The iTwo is a good fit for songwriters who capture ideas away from the desk, teachers recording lessons on iPad, and bands who want simple two-track rehearsal recordings. It is less suited to people who need many inputs, loopback for streaming, or onboard processing.

How to set up the PreSonus AudioBox iTwo on a computer

Setting up the PreSonus AudioBox iTwo on a Mac or PC is simple. Still, a few choices early on decide how clean and responsive your recordings feel. Work through these steps in order.

1. Connect the USB cable. The iTwo is bus powered over USB 2.0, so it needs no wall adapter on a computer.
2. Install drivers if needed. On a Mac it works as a standard audio device. On Windows, install the driver and Universal Control from your PreSonus account for the lowest latency.
3. Pick the device in your DAW. Choose the AudioBox iTwo for input and output in your audio preferences. On Windows, select its ASIO driver.
4. Set the sample rate. PreSonus lists 44.1, 48, 88.2 and 96 kHz. Use 44.1 kHz for music or 48 kHz for video.
5. Set a tracking buffer. Try 128 samples when recording. Raise it to 512 or more for mixing.
6. Start with the main volume down. Then bring up your speakers and headphones slowly.

How to set up the iTwo with an iPad

The iPad workflow is the main reason to choose this interface. Because it is MFi compliant, it works with Capture Duo and other iPad recording apps.

1. Connect the iTwo to your iPad with the correct cable or adapter for your iPad’s port.
2. Check power. PreSonus notes that the iTwo works with standard iOS power accessories, so check their notes for your model.
3. Install Capture Duo or your preferred recording app.
4. In iPad Settings, allow microphone access for that app. Without it, the app sees no input.
5. Arm two tracks, one per input, and record a short test.

After the test, play it back on headphones. Listen for clipping, hum or a thin guitar tone before the real session.

Setting gain on the iTwo

The iTwo has signal-present and clip LEDs rather than full meters. Therefore, use your software meters as the main guide.

First, set the input switch to line or instrument where needed. Next, have the performer play at their loudest. Raise the gain until your DAW meter peaks around -12 to -6 dBFS. If the clip LED lights, turn the gain down.

PreSonus rates the mic preamps at 0 to 52 dB of gain. That suits condensers and most dynamics on vocals. However, very quiet dynamic mics on soft speech may need more gain than that. In those cases, an inline mic booster or a higher-gain interface helps.

Zero-latency monitoring explained

The internal analog mixer lets you hear inputs with no delay. According to PreSonus, the Direct button gives a fixed 50/50 mix of input and playback. In other words, it is an on/off blend, not a variable knob.

If the 50/50 balance does not suit you, adjust playback level in your DAW instead. Meanwhile, turn off software monitoring on the armed track. Otherwise, you hear the direct signal and a delayed copy together.

For more on avoiding delay, our recording hub links to monitoring and tracking guides.

Recording MIDI gear through the iTwo

The 5-pin MIDI in and out ports let older synths and drum machines talk to your computer or iPad. Connect your keyboard’s MIDI out to the iTwo’s MIDI in. Then select the iTwo as a MIDI input in your DAW. For audio, run the synth’s outputs into the two line inputs at the same time. As a result, you capture both the notes and the sound in one pass.

PreSonus AudioBox iTwo vs similar interfaces

The table below compares the iTwo with close options, using published manufacturer specs.

Feature	AudioBox iTwo	AudioBox USB 96	UA Volt 276	Vocaster Two
Inputs	2 combo, Class A preamps	2 combo	2 mic/line/instrument	2 XLR mic
Mic gain	0 to 52 dB	0 to 35 dB	See UA specs	High gain for dynamic mics
Max sample rate	96 kHz	96 kHz	192 kHz	See Focusrite specs
Connection	USB 2.0 and iPad	USB 2.0	USB-C, iPad and iPhone	USB-C
MIDI	5-pin in/out	5-pin in/out	In/out	None listed
Headphone outputs	1	1	1	2
Standout extra	iPad recording with Capture Duo	Simple, low-cost 2×2	76 compressor, 610 preamp mode	Loopback, phone input, Show Mix

To sum up, the iTwo wins for iPad recording and preamp gain over the USB 96. The Volt 276 adds onboard tone tools. Meanwhile, the Vocaster Two is the podcast specialist.

PreSonus AudioBox iTwo pros and cons

Pros

• Works on macOS, Windows and iPadOS.
• Class A preamps with up to 52 dB of gain, per PreSonus.
• Zero-latency analog monitoring via the Direct button.
• 5-pin MIDI in and out for older gear.
• Light metal chassis at 1.35 lb, with a Kensington lock hole.

Cons

• Sample rate tops out at 96 kHz.
• Direct monitoring is a fixed 50/50 blend, not a variable knob.
• Only clip and signal LEDs, not full meters.
• USB 2.0 connector rather than USB-C.
• No loopback for streaming.

Choosing microphones for the iTwo

Because the preamps reach 52 dB of gain, mic choice matters. Here is how common types pair with the iTwo.

• Condenser mics. These are sensitive, so they need less gain. Switch on +48V phantom power first. They suit vocals, acoustic guitar and room sound.
• Standard dynamic mics. Handheld stage mics work well for loud singing, guitar amps and drums. For quiet speech, you may run near the top of the gain range.
• Low-output dynamic and ribbon mics. Broadcast-style dynamics and ribbons can need more than 52 dB for soft sources. Plan for an inline booster in that case.

Also, never switch phantom power on or off with your speakers turned up. The pop can be loud. Ribbon mics in particular should be checked against the maker’s advice before using phantom power.

Headphones and speakers

PreSonus rates the headphone output at 60 mW per channel into 60 ohms. That drives most studio headphones at comfortable levels. Very high-impedance models may sound quieter, though, so check your headphone’s specs.

For speakers, connect powered monitors to the two balanced 1/4-inch TRS main outputs. Use balanced TRS cables where possible to reduce hum over longer runs. Then set the large main volume knob low before turning the speakers on.

Tips for rehearsal and demo recordings

The iTwo is often used to capture a band or a songwriter quickly. A few habits make those quick takes far more usable.

• Place one mic for the room and one close to the singer. Then pan them apart in the mix for width and clarity.
• Record a 10-second level check before every song, since volume often rises once players warm up.
• Name and save each take right away, so files transfer cleanly to your computer.
• Keep the interface away from amps and power strips to reduce hum.
• Back up iPad sessions to your computer the same day.

Finally, keep peaks conservative. A slightly quiet take can be raised later, but a clipped take usually cannot be fixed. For broader guidance, see our guide on choosing an audio interface.

Quick decision guide

Still unsure? Choose the iTwo if you record on an iPad and a computer and want MIDI. Choose a USB-C model with loopback if you stream. Choose a four-input interface if you track drums or a full band. The official PreSonus AudioBox iTwo page lists the complete specifications.

Common mistakes

• Not granting microphone access on iPad. Without it, recording apps cannot see the interface.
• Using a weak power source with iPad. Check PreSonus’s notes on power accessories for your iPad model and connector.
• Wrong input switch setting. Set line or instrument correctly; a guitar on line level will sound thin and quiet.
• Double monitoring. Use the hardware blend and disable monitoring in the app.
• Recording too hot. Leave headroom for loud moments.
• Expecting a variable monitor blend. The Direct button is a fixed 50/50 mix, so set playback in your DAW.
• Using a generic Windows driver. Install PreSonus’s driver for lower latency.

Setup checklist

1. On a computer: connect by USB and select the AudioBox iTwo in your DAW.
2. On iPad: connect with the appropriate cable or adapter, install Capture Duo, and allow microphone access.
3. Set each input to the right type and set gain.
4. Enable 48V for condenser mics only.
5. Record a short test, then transfer it to your computer to confirm the workflow.

More: DAW software for iPad Pro, how to choose an audio interface and the recording hub.

Check the current price of the PreSonus AudioBox iTwo on Amazon.

FAQ

When your demos are ready for release, our single track mastering can finish them. Send a final stereo mix with headroom.

As an Amazon Associate I earn from qualifying purchases. This write-up is based on MOTU’s published specifications and documentation, not hands-on testing.

The MOTU M4 is a 4-in, 4-out USB-C audio interface that adds two balanced line inputs, two extra line outputs and an input monitor mix knob to the two combo mic/guitar preamps, 120 dB output dynamic range and color LCD metering of its smaller sibling. Those extras make it a better fit for producers who connect a keyboard or synth alongside mics, or who need a second pair of outputs for another set of speakers or a headphone amp.

See the MOTU M4 on Amazon.

MOTU M4 key specs

Mic/guitar inputs	2 XLR/TRS combo jacks with independent gain, 48V and MON switch
Line inputs	2 rear 1/4″ TRS (balanced or unbalanced), 115 dB dynamic range, +18 dBu max
Line outputs	4 balanced 1/4″ TRS (DC coupled), 120 dB dynamic range, +16 dBu max
RCA outputs	4 unbalanced, mirroring the TRS outs
Headphones	1 stereo 1/4″ output with separate level
Front controls	Monitor and phones volume, input monitor mix knob, 3-4 switch
Mic EIN	-129 dBu
MIDI	In and out
Sample rates	44.1 to 192 kHz
Display	Full-color LCD (160 x 128) for input and output levels
Power and bus	USB-C, USB 2.0 class compliant, bus powered
Dimensions	20.955 x 10.8 x 4.5 cm, 0.7 kg

Source: MOTU M4 tech specs and the M4 product overview.

What the M4 adds over the M2

Both models share the same preamps, converter technology, metering screen and bus-powered design. The differences are all about connections and monitoring control:

• Two dedicated line inputs on the rear, so a stereo keyboard or drum machine can stay plugged in while both front preamps remain free for a mic and guitar. All four inputs can be recorded at once.
• Outputs 3-4, a second balanced pair (plus mirrored RCA). The front-panel 3-4 switch relates to how these outputs are used for monitoring; check MOTU’s documentation for exact behavior.
• An input monitor mix knob that blends live inputs against computer playback, which MOTU describes as controlling the balance between the two.
• A slightly wider and heavier metal case to fit the extra jacks.

Read our MOTU M2 write-up if two inputs and two outputs are enough for you.

Why a monitor blend knob matters

When you track vocals or guitar with direct monitoring, you need to hear yourself and the backing track at a comfortable ratio. Without a blend control, you adjust the playback level in software and hope the balance works. The M4’s knob lets the performer change that ratio instantly from the front panel, which helps when you are recording yourself and cannot easily reach the mouse.

Using four outputs

Four line outputs open up several setups that a two-output interface cannot handle as neatly:

• Two pairs of studio monitors, such as main speakers and a small reference speaker, to check mixes on different systems.
• Sending a separate cue mix to an external headphone amplifier.
• Routing to outboard effects and back through the line inputs, using the extra outputs as hardware sends.
• Feeding a DJ mixer or PA via the RCA outputs.

Measured performance claims

MOTU’s published numbers put the line outputs at 120 dB dynamic range with -110 dB THD+N, the line inputs at 115 dB with -106 dB THD+N, and the mic preamps at -129 dBu EIN. MOTU attributes the output figures to ESS Sabre32 Ultra DAC technology, which also drives the headphone amp. Spec sheets cannot tell you how an interface sounds to your ears, but these figures put the M4 among the quieter small interfaces on paper. Our guide to mic preamps explains what EIN measures.

MOTU also quotes 2.5 ms round-trip latency at 96 kHz with a 32-sample buffer. Your own results will depend on your computer and session.

Loopback, drivers and platforms

As with the M2, the M4 driver includes loopback channels for mixing computer audio with live inputs, a common need for streaming. The interface is class compliant on Mac and iOS (iOS requires an adapter and, per MOTU, a powered USB hub), and uses a MOTU driver on Windows. See our guide on how to use an audio interface for setup basics.

Bundled software

MOTU includes Performer Lite, Ableton Live Lite and a library of loops and one-shots. Bundles can change, so confirm on MOTU’s site.

Who should buy the MOTU M4

• Producers with a keyboard or synth who also record vocals or guitar.
• Home studios with two sets of monitors.
• Performers who want a hardware blend between input and playback.
• Streamers who need loopback plus extra line inputs for other sources.

Who might choose something else

• Recording more than two mics at once needs an interface with more preamps.
• Two separate headphone outputs for two performers are not on the M4.
• No ADAT or S/PDIF, so there is no digital route for expansion.

For a rack-sized alternative with more preamps, compare our PreSonus Studio 68c write-up. Our guide on how to choose an audio interface helps weigh inputs, outputs and monitoring.

How to set up the MOTU M4 step by step

Getting clean, low-latency audio from the MOTU M4 takes about ten minutes. The order matters, because driver and buffer choices affect everything after them. Follow these steps before you record your first take.

1. Install the right driver. On Windows, install MOTU’s driver before you plug in the interface. On a Mac, the M4 works without a driver, but MOTU offers an optional one. Download both from MOTU’s support pages.
2. Connect with the supplied cable. Plug the M4 into a USB-C or USB Type A port. It is bus powered, so no wall adapter is needed. Avoid unpowered hubs, since they can starve the unit of power.
3. Select the M4 in your DAW. Open your audio preferences and choose the M4 for both input and output. On Windows, pick the MOTU ASIO driver rather than a generic one.
4. Set the sample rate. 44.1 or 48 kHz suits most music and video work. Higher rates up to 192 kHz are supported, but they use more processing power and disk space.
5. Choose a buffer size. Start at 64 or 128 samples while tracking. Raise it to 512 or 1024 when mixing with many plug-ins.
6. Turn the monitor knobs down. Lower the monitor and phones levels before you power your speakers. Then raise them slowly.

Once these basics are set, test each input with the LCD meters before you press record.

Setting gain with the color LCD

The full-color screen is one of the M4’s most useful tools. It shows levels for every input and output at once. That means you can set gain without staring at your DAW.

First, ask the performer to play or sing at their loudest. Next, turn the gain knob until the meter peaks well below the top. A safe target is roughly -12 to -6 dBFS on the loudest parts. That leaves headroom for surprise peaks. If a meter reaches the top, the signal is clipping, so back the gain off.

Line inputs 3-4 have no gain knob. Instead, set the level at the source, such as the volume on your keyboard. Our guide to using an audio interface covers gain staging in more detail.

Monitoring: hardware vs software

The M4 gives you two ways to hear yourself. Each has a clear use, so pick one per session.

Hardware (direct) monitoring sends the input straight to your headphones. Press the MON switch on the input you want to hear. Then use the input monitor mix knob to balance it against playback. Because the signal never passes through the computer, there is no latency to notice. However, you will not hear plug-ins such as reverb or amp sims on the input.

Software monitoring routes the input through your DAW. As a result, you can hear effects while you record. MOTU quotes 2.5 ms round-trip latency at 96 kHz with a 32-sample buffer. Real results depend on your computer and session size, though.

Never use both at once. If MON is on and your DAW track is also armed for monitoring, you will hear a doubled, phasey signal.

A sample session: synth plus vocal

Here is a practical setup that shows why the M4’s extra inputs matter. Picture a producer recording a stereo synth and a vocal in one pass.

• Plug the condenser mic into input 1 and switch on 48V phantom power.
• Connect the synth’s left and right outputs to rear line inputs 3 and 4.
• Leave input 2 free for a guitar or a second mic later.
• Create a mono track for input 1 and a stereo track for inputs 3-4 in your DAW.
• Turn on MON for input 1, then set the blend knob so the vocal sits just above the track.

With this layout, nothing needs re-patching between takes. That speed is the main reason to choose the M4 over a two-input box.

Using loopback for streaming and podcasts

Loopback sends your computer’s own audio back in as a recordable input. For instance, a streamer can mix game audio, a browser call and a live mic in one program. To use it, select the M4’s loopback channels as inputs in your streaming or recording software. Then mute any track that would feed loopback back into itself. Otherwise, you can create a feedback loop.

MOTU M4 vs similar interfaces

The M4 sits in a crowded class. This table compares it with close alternatives, using each maker’s published specs.

Feature	MOTU M4	MOTU M2	Focusrite Scarlett 4i4	PreSonus Studio 68c
Mic preamps	2	2	2	4
Extra line inputs	2 rear TRS	None	2 rear line	None (combo inputs)
Line outputs	4 TRS + 4 RCA mirrored	2 TRS + 2 RCA mirrored	4 balanced	Main + outputs 3/4
Headphone outputs	1	1	1	1 (Cue A/B)
Digital I/O	None	None	None	S/PDIF in/out
MIDI	In/out	In/out	In/out	In/out (breakout)
Power	USB bus	USB bus	USB bus	12V adapter
Metering	Color LCD	Color LCD	Gain halos	Front-panel meters

In short, the M4 is the pick for line inputs plus clear metering. The 68c suits anyone who needs four mic preamps and S/PDIF instead.

MOTU M4 pros and cons

Pros

• Four simultaneous inputs, with two kept for line sources.
• 120 dB dynamic range on the main outputs, per MOTU.
• Color LCD shows every input and output level at a glance.
• Hardware monitor blend knob for latency-free tracking.
• Bus powered and housed in a metal case, so it travels well.
• Loopback built into the driver for streaming.

Cons

• Only two mic preamps, so drum or band recording is limited.
• One headphone output, which makes two-person sessions awkward.
• No ADAT or S/PDIF for adding preamps later.
• No built-in DSP effects for monitoring.

Routing outputs 3-4 in your DAW

Outputs 3-4 do nothing until you send audio to them. Most DAWs let you assign any track, bus or the master to a chosen output pair. Here are three common setups.

• Second speaker pair. Duplicate the master to outputs 3-4. Then mute one pair at a time to compare speakers. Some DAWs and monitor controllers make this easier with a speaker switch.
• Headphone cue mix. Create a send bus routed to outputs 3-4 and connect a headphone amp. Now a second performer gets their own balance.
• Hardware insert. Route a track out of 3-4 into a pedal or compressor. Next, return it to line inputs 3-4 and record the result.

Label the outputs in your DAW’s I/O setup. Clear names save time and stop you sending a mix to the wrong place.

Buffer size cheat sheet

Buffer size trades latency for stability. Smaller buffers feel tighter but strain the computer. Larger ones are stable but add delay. Use this table as a starting point, then adjust for your machine.

Task	Suggested buffer	Why
Tracking vocals with software effects	32-128 samples	Keeps delay low enough to sing in time
Playing virtual instruments	64-128 samples	Notes respond quickly to your keyboard
Tracking with MON direct monitoring	256 samples	You hear the input directly, so delay matters less
Mixing large sessions	512-1024 samples	Frees processing power for plug-ins

If you hear clicks or dropouts, raise the buffer one step. Also close other apps that use audio, such as browsers or video calls.

For more context on connections, see MOTU’s official M4 page, which lists the full I/O and system requirements.

Common mistakes

• Plugging a keyboard into the front preamps by default. On the M4 the rear line inputs keep the preamps free for mics.
• Leaving the monitor mix knob fully toward input. You will not hear the backing track and may think playback is broken.
• Assuming outputs 3-4 carry the main mix automatically. Check routing in your DAW and MOTU’s documentation.
• Running iOS without enough power. MOTU recommends a powered USB hub for iOS.
• Using a generic Windows driver. Install MOTU’s ASIO driver, or latency will be much higher.
• Tracking at a huge buffer. A buffer of 1024 samples makes software monitoring feel laggy. Drop it while recording.
• Doubling the monitoring. Use MON or DAW input monitoring, not both.

Check the current price of the MOTU M4 on Amazon.

FAQ

When the tracks you record on your interface are mixed, our stem mastering service gives the final master extra control over each group. Upload your stems and we will return a polished, release-ready master.

As an Amazon Associate I earn from qualifying purchases. This write-up is based on MOTU’s published specifications and documentation, not hands-on testing.

The MOTU M2 is a bus-powered 2-in, 2-out USB-C audio interface with two combo mic/line/guitar inputs, ESS Sabre32 Ultra DAC technology rated at 120 dB dynamic range on the main outputs, a full-color LCD level meter and MIDI in and out. It is aimed at home recording, songwriting, streaming and podcasting, for anyone who needs two inputs and wants clear metering on the front panel.

See the MOTU M2 on Amazon.

MOTU M2 key specs

Inputs	2 XLR/TRS combo jacks for mic, line or Hi-Z guitar
Preamp features	Independent gain, 48V phantom power and monitor (MON) switch per input
Outputs	2 balanced 1/4″ TRS line outs (DC coupled), 2 RCA outs (mirrored)
Headphones	1 x 1/4″ TRS stereo with its own volume control
MIDI	5-pin MIDI in and out
Sample rates	44.1 to 192 kHz
Line output	120 dB dynamic range (A-weighted), -110 dB THD+N, +16 dBu max
Mic input	115 dB dynamic range, -129 dBu EIN, +10 dBu max at minimum gain
Headphone output	115 dB dynamic range, -110 dB THD+N, +12.5 dBu max
Metering	160 x 128 pixel full-color LCD with input and output meters
Connection and power	USB-C (USB 2.0 audio class compliant, works with USB Type A), bus powered
Size and weight	19.05 x 10.8 x 4.5 cm, 0.61 kg

All figures are from the MOTU M2 specifications page.

Audio quality on paper

MOTU highlights the M2’s ESS Sabre32 Ultra DAC technology, which it says drives both the main outputs and the headphone output. The published 120 dB dynamic range on the line outputs is a strong figure for a two-channel interface. For monitoring and mix decisions, a clean, low-noise output is arguably more important than the inputs, since you hear everything through it.

On the input side, MOTU lists -129 dBu EIN for the mic preamps. Lower EIN means less self-noise at high gain, which matters most with quiet sources or low-output dynamic mics. Our explainer on mic preamps covers what EIN and gain range mean in practice.

Metering on the front panel

The M2’s most distinctive feature for its class is the full-color LCD showing input and output levels. Most small interfaces use simple LED rings or a few segment LEDs. A proper meter makes it easier to set recording levels with headroom, and it shows you output level without looking at the screen. That suits recording vocals or guitar while you are away from the computer.

Monitoring and latency

Each input has a MON switch that sends the input directly to the outputs, so you can hear yourself without the delay of routing through software. MOTU also quotes a round-trip latency of 2.5 ms at 96 kHz with a 32-sample buffer for monitoring through the computer. Actual latency depends on your computer, buffer setting and plugins.

Unlike the M4, the M2 has no knob to blend input and playback. Direct monitoring is either on or off per input.

Loopback for streaming and podcasts

MOTU’s driver provides loopback channels, which route computer audio back into the computer as an input. That lets you mix sound from a browser, game or call with your microphone in streaming or recording software. It is useful for streamers and remote podcast hosts who want one interface to handle everything.

Connectivity and compatibility

The M2 connects via USB-C and is USB audio class compliant, so it works on Mac without a driver (an optional driver is available) and on iOS devices with a suitable adapter. Windows uses MOTU’s driver. It is powered from the USB bus, so there is no power supply to carry. If you are new to connecting and configuring an interface, our guide on how to use an audio interface covers drivers, buffer size and routing.

The rear panel has balanced TRS outputs for studio monitors and mirrored RCA outputs for consumer speakers or a DJ mixer. MIDI in and out lets you connect a hardware synth or older keyboard controller without a separate MIDI interface.

Included software

MOTU bundles its Performer Lite DAW, Ableton Live Lite and a collection of loops and sounds from third-party providers. Check MOTU’s page for the current bundle, as included software can change.

Who the MOTU M2 suits

• Singer-songwriters recording a voice and a guitar at the same time.
• Producers who want clean monitoring outputs for mixing at home.
• Streamers and podcasters who need loopback.
• Anyone who wants proper metering instead of basic LEDs.

Who should look elsewhere

• If you need more than two simultaneous inputs, for example to record a drum kit, look at larger interfaces.
• If you want a blend control for direct monitoring or separate line inputs, the MOTU M4 adds both.
• If you need digital expansion such as ADAT, the M2 does not offer it.

For comparison, see our write-ups of the Focusrite Scarlett 4i4 and the PreSonus Studio 26c, two other compact USB interfaces. Our guide on how to choose an audio interface covers what to prioritize.

Common mistakes

• Leaving MON on while also monitoring through software. You hear the signal twice, which can sound like phasing or echo.
• Turning on 48V for every mic. Only condenser mics and some active mics need phantom power.
• Recording too hot. Use the LCD meter and leave headroom below the top.
• Using the RCA and TRS outputs at once without thinking. They are mirrored, so both carry the same signal.

How to set up the MOTU M2 step by step

The M2 is simple to connect, but the driver choice matters. MOTU’s M2 specifications page lists the full technical data.

1. Install the right driver

On Windows, install MOTU’s driver for low latency and loopback. On a Mac, the M2 is class compliant. However, the optional MOTU driver is needed for loopback and MOTU’s lowest published latency figures.

2. Connect and power on

The M2 is bus-powered over USB-C and has its own power switch. Connect it directly to the computer, then switch it on.

3. Set rate and buffer

Choose 44.1 or 48 kHz in your DAW. Use a small buffer for tracking. MOTU quotes 2.5 ms round-trip latency at 32 samples and 96 kHz with its driver, though your computer sets the practical limit.

4. Set gain with the LCD meters

Ask for the loudest part of the performance. Raise gain until the colour meters show healthy level without reaching the top. In the DAW, aim for peaks around −12 dBFS.

5. Use the monitor switches

Each input has a monitor switch for direct monitoring. Press it to hear that input with no computer delay. Then turn off software monitoring on the armed track.

6. Connect speakers

Use the balanced TRS outputs for studio monitors. The RCA outputs mirror them, which is handy for a consumer amplifier or DJ mixer.

A vocal and guitar recording workflow

Put a vocal mic on input 1 and a guitar on input 2. Turn on 48V only for condenser mics. Plug a guitar straight into the combo jack, and check MOTU’s manual for input settings.

Next, set gain at full performance level and enable direct monitoring on both inputs. Record a test and listen on monitors. Adjust mic position before changing gain.

When tracking ends, raise the buffer for mixing. Use the bundled Performer Lite or Live Lite to edit and mix. Finally, export and check the mix on headphones and small speakers.

MOTU M2 vs similar interfaces

The table compares published figures from each maker.

Spec	MOTU M2	Scarlett 4i4 (4th Gen)	UA Volt 2	Clarett+ 2Pre
Mic preamps	2	2	2	2
Mic EIN	−129 dBu	−127 dBu (A-wtd)	−127 dBu	−129 dBu
Line output dynamic range	120 dB	120 dB	110 dB	124 dB
Max line output	+16 dBu	+16 dBu	15 dBu	+18 dBu
Line outputs	2 TRS plus mirrored RCA	4	2	4
MIDI	In/out	In/out	In/out	In/out
Auto Gain	No	Yes	No	No

The M2’s strengths are low noise, its colour LCD meters and mirrored RCA outputs. The Scarlett 4i4 adds Auto Gain and more outputs. The Volt 2 adds Vintage mode. The Clarett+ 2Pre adds ADAT expansion and higher output level.

Pros and cons

Pros

• Low mic noise at −129 dBu EIN.
• Colour LCD meters for inputs and outputs.
• ESS Sabre32 DAC with 120 dB output dynamic range.
• Mirrored RCA outputs.
• MIDI in and out.
• Bus-powered with a power switch.

Cons

• Loopback on Mac needs the optional driver.
• One headphone output.
• No Auto Gain.
• No expansion port.

Reading the LCD meters

The colour LCD shows input and output levels at a glance. That is the M2’s most visible feature. Many small interfaces only offer a single LED per channel. Full meters make gain setting quicker and more accurate.

Watch the meters while the performer plays loudly. Keep peaks below the top of the scale. If the meter hits the top often, lower the gain. Then check the DAW’s meter too, since it shows exactly what is being recorded.

Using loopback for streaming and podcasts

Loopback sends computer audio back into your recording or streaming app. On Windows it comes with MOTU’s driver. On Mac it needs the optional MOTU driver.

Use it to record a video call guest, add music to a stream or capture a tutorial. Record the loopback on its own track when possible. Mute playback of that track to avoid feedback.

Choosing microphones for the M2

Condenser mics pair easily with the M2. Turn on 48V phantom power on the input you use. For dynamic mics on loud singing, the M2 also works well.

Quiet speech on a low-output dynamic mic may need more gain. In that case, test your mic before committing, or add an inline booster. The mic input accepts up to +10 dBu before clipping, so keep gain low on loud sources such as guitar amps.

Using the RCA outputs

The RCA outputs mirror the main TRS outputs. That makes it easy to connect a hi-fi amplifier, powered speakers with RCA inputs, or a DJ mixer. Use TRS for studio monitors where possible, since balanced connections reject more noise over long cables.

Who should buy it and who should skip it

Buy the M2 if you record one or two sources and want clear metering and low noise. It suits singer-songwriters, home producers and streamers on Windows. Mac users who need loopback should install the optional driver.

Skip it if you need four outputs for a second speaker pair, where the four-output MOTU M4 or Scarlett 4i4 suits better. Skip it as well if you want Auto Gain or room to expand with ADAT.

Common setup mistakes

• Skipping the Windows driver. Without it, latency is higher and loopback is missing.
• Expecting loopback on Mac without the driver. Install MOTU’s optional driver.
• Double monitoring. Use the monitor switch or software monitoring, not both.
• Leaving the power switch off. The M2 has its own switch, so check it first when nothing appears.
• Recording loopback into a loop. Mute the receiving track.

Buffer size and latency in practice

Buffer size controls how much audio the computer processes at once. Smaller buffers mean less delay but more CPU strain. Larger buffers are safer but add delay.

For tracking, start at 64 samples. If you hear clicks, move up to 128. For mixing, use 512 or 1024 samples so plugins have room. MOTU’s published 2.5 ms figure assumes a 32-sample buffer at 96 kHz with its driver. Real sessions with many plugins usually need more.

Direct monitoring sidesteps the problem while recording. The monitor switches let the performer hear the input with no computer delay at any buffer size.

Gain staging and headroom

Record at 24-bit and leave headroom. Average levels around −18 dBFS with peaks near −10 dBFS are a safe target. That keeps the low noise floor well below the signal, without risking clipping.

On the output side, the TRS outputs reach +16 dBu. Use the monitor knob for volume and keep the DAW master near unity. If your speakers are too loud at low knob settings, turn down their input trim instead.

Using the M2 with MIDI gear

The 5-pin MIDI in and out ports connect keyboards, drum machines and older synths. Your DAW sees them once the M2 is connected. That removes the need for a separate MIDI interface on a small desk.

A simple producer setup uses a keyboard on MIDI in, a synth on MIDI out, and the synth’s audio into the two combo inputs. You can then sequence and record the synth with one compact box.

Loopback driver notes

Yes, with MOTU’s optional Mac driver installed. Without it, the M2 still works as a class-compliant interface, but loopback is not available. On Windows, loopback comes with MOTU’s standard driver.

Final checks before a session

Before recording, confirm the M2 is switched on and selected in your DAW. Check the sample rate. Then record ten seconds from each input and play it back. This catches cable faults and routing errors before they cost you a take.

Headphones and monitoring levels

The headphone output offers 115 dB dynamic range and up to +12.5 dBu. Start with the headphone knob low, then raise it slowly. Loud levels tire your ears and make mix decisions less reliable.

Closed-back headphones suit recording, because they leak less sound into the mic. Open-back headphones are often preferred for mixing. Check your mix on both, plus speakers, before you call it finished.

Check the current price of the MOTU M2 on Amazon.

FAQ

Once your recordings and mix are done, our single track mastering service can prepare them for release. Send your mix and we will return a master ready for streaming.

As an Amazon Associate I earn from qualifying purchases. This write-up is based on Focusrite’s published specifications and documentation, not hands-on testing.

The Focusrite Vocaster Two is a two-microphone USB-C podcast interface with Auto Gain, Enhance voice presets, a mute button per host, two headphone outputs, Bluetooth and wired phone connections, a camera output and computer loopback. It is designed for shows with a host and a guest in the same room, plus remote callers, and it trades music-studio flexibility for speed of setup.

See the Focusrite Vocaster Two on Amazon.

Focusrite Vocaster Two key features

Mic inputs	2 XLR, with enough gain for low-output dynamic mics (Focusrite: “no booster needed”)
Headphones	2 outputs, one per presenter, each with its own Show Mix level
Phone	Bluetooth, or a single TRRS cable
Camera out	TRS output to send the show audio to a camera
Loopback	Capture computer audio such as music beds or call audio
Software	Vocaster Hub for levels, Enhance, routing and Show Mix

Details are on Focusrite’s Vocaster Two page.

Auto Gain and Enhance

Auto Gain is a one-button level set: press it, talk at your normal show volume, and the interface sets the mic gain for you. It solves the most common podcast problem, which is a host who is too quiet or clipping. Enhance applies one of several presets designed for spoken voice. It is a fast starting point; if you prefer to shape the sound yourself later, record without Enhance and treat the voice in your editor.

Two hosts, two headphone mixes

Each presenter gets a headphone output and their own level. That sounds minor, but in a two-person show it stops one host from having to listen too loud or too quiet. Each host also has a mute button, so a cough or sneeze can be cut on the fly without touching the computer.

Phone guests and computer audio

Bluetooth lets you bring a phone call or music from a phone into the show. A TRRS cable does the same with a wire, which some people prefer for reliability. Loopback captures audio playing on your computer, such as an online call, an intro sting or a music bed, so it can be recorded and streamed together with the mics.

Setting up a two-host room

The interface is only part of the sound. Two mics in one room pick up each other’s voice, so the further apart the hosts sit and the closer each speaks to their own mic, the cleaner each track will be. Dynamic microphones a few inches from the mouth, pointed away from the other host, reduce spill. Soft furnishings, a rug and curtains reduce the echo that makes home podcasts sound like they were recorded in a kitchen.

Keep both hosts at a similar distance from their mics so their levels match. Auto Gain helps set levels, but it cannot fix a host who leans back halfway through the show.

A simple remote-guest workflow

1. Connect the guest by phone (Bluetooth or TRRS) or through a call app on the computer using loopback.
2. Check that the guest hears both hosts but not their own voice delayed back to them.
3. Record the in-room mics on their own tracks, and the guest on a separate track if your software allows.
4. Ask the guest to record their own audio locally as a backup when quality matters.

Camera output for video podcasts

The camera output sends the show mix to a camera through a single TRS cable. This means the camera file carries the good audio, which can save syncing in the edit for simple video shows. For multi-camera shoots you may still prefer to record the audio separately in your DAW.

Included software

Focusrite lists six months of Hindenburg PRO and six months of Acast Influencer with registration. Bundles change over time, so check what is offered when you buy.

Who should consider the Vocaster Two

• Two-person podcasts recorded in one room.
• Shows that bring in phone or online guests.
• Streamers and video podcasters who want camera audio without a separate mixer.

Who should look elsewhere

• Solo hosts. The Vocaster One has the same idea for one mic.
• Musicians who need instrument inputs, MIDI or line inputs for synths; a music interface fits better.
• Shows with three or more in-room hosts, which need more mic inputs.

Common mistakes with a podcast interface

• Running Auto Gain while speaking quietly. Talk at the volume you will use during the show, including laughs, or levels will be set too high.
• Not testing the phone route. Make a test call before the show to check levels and that the caller cannot hear an echo of themselves.
• Using condenser mics in a noisy room. Dynamic mics close to the mouth pick up less of the room.
• Recording only the stereo mix. If your software allows, record each host to its own track so you can fix one voice without affecting the other.
• Forgetting the mute buttons are on. Check mute status before every take.

Setup checklist

1. Register the unit and install Vocaster Hub.
2. Connect both mics and headphones, then press Auto Gain for each host.
3. Choose an Enhance preset or leave it off.
4. Pair a phone by Bluetooth or connect it with TRRS, and test a call.
5. Set your recording software to the Vocaster Two and record a test.

For recording and editing tips, read how to record and edit a voice-over or podcast and our overview of audio interfaces for podcasting. More guides are in the recording hub.

How to set up the Focusrite Vocaster Two step by step

The checklist below covers the basics. These steps add detail for a two-host show.

1. Choose a powered USB port

Focusrite states that Vocaster Two needs a USB port that supplies 900 mA. Connect it straight to the computer instead of an unpowered hub.

2. Install Vocaster Hub

Vocaster Hub controls Enhance, the Show Mix, loopback and Bluetooth. Install it, connect the interface and check for firmware updates.

3. Position both mics

Seat the hosts facing each other or side by side, with mics pointed away from each other. Keep each mic close to its speaker. That reduces bleed between the two channels.

4. Run Auto Gain for each host

Each host has separate Auto Gain. Ask each person to talk at show volume during the process. Then check the meters while both talk and laugh together.

5. Pick Enhance presets

Choose a preset for each host on its own. Voices differ, so the same preset may not suit both. Record short tests and compare.

6. Record to separate tracks

Record each host on their own track in your software. Add a track for loopback or phone audio too. Separate tracks make editing and level balancing much easier.

A two-host episode workflow

Start each session with a quick sound check. Then agree on a hand signal for pauses. Each host can use their own Mute button for coughs.

During recording, avoid talking over each other too often. Crosstalk is hard to edit later. Afterwards, edit gaps, balance the two voices and set loudness to your platform’s target.

Vocaster Two vs similar interfaces

The table compares published features. Focusrite’s Vocaster Two product page lists full specs.

Feature	Vocaster Two	Vocaster One	UA Volt 2	Scarlett 4i4 (4th Gen)
Mic inputs	2	1	2	2
Headphone outputs	2	1	1	1
Auto Gain	Yes, per host	Yes	No	Yes
Mute buttons	Separate host and guest	One	No	No
Bluetooth	Yes (5.0)	No	No	No
Phone connection	Stereo TRRS and Bluetooth	Mono TRRS	No	No
Camera output	Yes	Yes	No	No
MIDI	No	No	Yes	Yes

Vocaster Two is the only one here with two headphone outputs and Bluetooth. Volt 2 and the Scarlett 4i4 also record two mics, but they target music rather than speech.

Pros and cons

Pros

• Two mic inputs with separate Auto Gain and Mute.
• Two headphone outputs.
• Bluetooth and wired phone connections.
• Camera output for video podcasts.
• Loopback for remote guests.
• Bus-powered from a 900 mA USB port.

Cons

• No instrument inputs or MIDI.
• Only two mic inputs, so a third guest needs another solution.
• Enhance cannot fix a very echoey room.

Using Bluetooth for phone guests

Bluetooth is the quickest way to bring a phone caller into the show. Pair your phone with Vocaster Two in the phone’s Bluetooth settings. Then make the call as normal. The caller’s audio enters the mix, and they hear both hosts.

Run a test before every interview. Check that the caller hears you clearly and that their level matches the hosts. Keep the phone on charge for long calls, and turn off notifications so alerts do not interrupt the recording.

A wired TRRS connection is another option. It avoids pairing and wireless issues. Use whichever is more reliable in your space.

Choosing microphones for two hosts

Use the same mic model for both hosts if possible. Matching mics make the two voices sit together naturally. Broadcast-style dynamic mics are a common pick because they reject room sound and bleed.

Condenser mics capture more detail. However, in a shared room, each condenser also picks up the other host. That bleed can sound hollow. If you use condensers, treat the room and keep the mics close.

Room setup for two people

Hard walls cause echo, which makes speech sound distant. Record in a room with carpet, curtains or soft furniture. Then position the hosts so that each mic points away from the other person.

Keep at least an arm’s length between the two mics where possible. Ask both hosts to stay at a consistent distance from their mic. A boom arm for each helps.

Video podcasts with the camera output

The camera output sends the mix to a camera’s mic input. That puts usable audio straight on the video file. Set the camera input gain low and adjust from Vocaster Hub.

Still, record a separate audio file on the computer. It is a backup and gives cleaner edits. A clap at the start helps you sync audio and video later.

Who should buy it and who should skip it

Buy Vocaster Two if two people share a room for your show, or if you take phone guests often. It suits interview shows, co-hosted podcasts and streams with a partner.

Skip it if you host alone, because Vocaster One is smaller and does the job. Skip it also if you record music, since a music interface like the Scarlett 4i4 has instrument inputs and MIDI. If you need three or more mics, look at a larger interface or a podcast mixer.

Editing two voices

Edit each host’s track separately. First, remove noise and gaps. Then balance the two voices so neither dominates. Gentle compression on each track evens out levels. Finally, set overall loudness to your platform’s target and check the episode on earbuds.

Using the Show Mix and loopback

The Show Mix combines both hosts, the phone guest and computer audio into one stereo feed. That feed is ready for streaming or quick-turnaround shows. Meanwhile, you can still record each source on its own track for detailed editing.

For a video call guest, use the loopback. It brings the call audio into your recording software. Record it on its own track, so you can lower the guest or fix their level later.

Watch out for echo. If the remote guest hears their own voice back, check that your call app is not receiving the loopback. Check your routing so the guest’s own audio is not sent back to them.

More mistakes with Vocaster Two

• Using one preset for both hosts. Pick presets per voice.
• Running Auto Gain with only one person talking. Each host needs their own check.
• Recording to one stereo file only. Separate tracks give you control in the edit.
• Forgetting phone notifications. Silence them before a Bluetooth call.
• Mics facing each other. Point them away to reduce bleed.
• Using a low-power hub. Connect to a 900 mA port.

A repeatable show routine

Consistency helps listeners. Set up the mics in the same place each time and keep the same presets. Save a template in your recording software with tracks for each host, the guest and loopback.

Before each episode, run Auto Gain for both hosts and do a short test recording. After the show, back up the raw files before editing. This routine makes each episode sound like the last.

Headphone mixes for host and guest

Two headphone outputs mean each host has their own level. That matters, because people prefer different volumes. A host with louder headphones may speak more quietly, which changes the recording.

Start with both headphone controls low, then raise them slowly. Ask each host whether they can hear themselves and the other person clearly. If a host cannot hear their own voice, they tend to talk louder or move away from the mic.

For long sessions, keep headphone levels moderate. Lower levels reduce ear fatigue and help everyone stay focused through the episode.

Final checks before recording

Run a quick checklist before every Vocaster Two session. Confirm both mics show signal. Check both headphone outputs. Test the phone or call audio. Then record thirty seconds of both hosts talking and play it back. Listen for noise, echo and level differences before starting the real episode.

Keep a short written note of each host’s preset and seating position. It saves time and keeps the sound consistent between episodes.

Check the current price of the Focusrite Vocaster Two on Amazon.

FAQ

When your episode is edited, our single track mastering can bring it to a consistent, polished loudness. Send the final mixed file.