Audio Settings
Audio Settings is where you connect AnalogDAW to your hardware and tune the balance between latency and stability. Open it from ☰ ▸ Audio Settings….
These are app/hardware settings (which device, how big the buffer). The recording quality (sample rate / bit depth) of each project is set separately in Session Settings.
Core Audio (engine on/off)
The Core Audio toggle turns the audio engine on or off. It's normally on (green, "Enabled"). On macOS, enabling it confirms microphone permission so inputs are accessible. On iPhone and iPad, AnalogDAW enables it automatically after creating, opening, recovering, or relinking a session, so the built-in mic and active network/audio plug-ins are ready without another visit to Settings. Turn it off only if you want to release the audio device.
Core Audio being enabled does not by itself open the microphone. A No Input track with a receiver plug-in before Record Point uses the output/render path only; the input path is activated only for hardware input monitoring or recording.
On macOS the input path is held open for a few seconds after you finish with it, so that switching Input Monitoring off and back on is instant instead of restarting the audio engine each time. Your microphone-in-use indicator can therefore stay lit briefly after you turn monitoring off or stop a take. Nothing is captured during that window — the live input goes silent the moment you switch monitoring off; only the device stays reserved.
The input is released once that window passes with nothing using it and the transport parked. It is also released immediately if the input device is unplugged, when you open another session, and whenever you turn Core Audio off. If the transport is rolling or an export is running when the window elapses, the release simply waits — stopping the engine mid-playback would cut reverb and delay tails, and mid-export would spoil the render.
Devices
macOS
- Output Device — where playback goes (interface, speakers, headphones).
- Input Device — where recording comes from. AnalogDAW uses one active input device at a time (Logic-style). To combine several interfaces, create an Aggregate Device in macOS Audio MIDI Setup and select it here.
AnalogDAW remembers the devices you choose and re-selects them at the next launch, identifying them by the interface itself rather than by the slot macOS happens to give it — so an interface that is already plugged in when you start the app comes up selected, and one that enumerates a moment late is picked up as soon as it appears.
After updating from a version that saved temporary Core Audio device numbers, reselect your input and output once in Audio Settings. Older project input assignments may also show as unavailable until you reselect them. AnalogDAW does not guess from those old numbers because macOS can reuse one for a different interface, which could otherwise send playback to—or record from—the wrong device.
Use Refresh Devices if you plug in hardware and it doesn't appear, then Apply.
iPhone / iPad
The current system input and output (e.g. Built-in Microphone / Built-in Speaker, or a connected interface/headset) are shown for reference; iOS manages routing through the system.
I/O Buffer Size
The buffer size is the core trade-off in any DAW. AnalogDAW offers the sizes your selected interface reports as usable from this set: 64, 128, 256, 512, 1024, or 2048 samples.
- Smaller buffer → lower latency (live input feels immediate) but more CPU load and a higher risk of audio glitches.
- Larger buffer → rock-solid playback with little CPU strain, but more delay when monitoring live input.
Rule of thumb: use a small buffer (64–256) while recording so performers feel in time, and a larger buffer (512–1024) while mixing so you can run lots of plug-ins without dropouts.
Multicore mixing
AnalogDAW can spread busy mixes across your Mac or iPad's performance cores automatically — there's nothing to configure. Light sessions stay on a lower- overhead serial path; when a callback approaches its deadline, the worker pool engages for eligible track, bus, and aux DSP. This also applies to a session with only a few tracks when an aux effect, bus/master chain, or plug-in makes the callback expensive; light callbacks still avoid multicore synchronization overhead when it would cost more than it saves.
Parallel work is not limited to tracks: dependency-independent track channels, group buses, and aux/effect returns can share the pool. The master chain and order-sensitive summing stay serial. In internal synthetic testing a heavy 48-track render ran about 2.8× faster across the pool than on a single core; real-session results depend on routing and plug-in cost.
When the audio device exposes a Core Audio workgroup, AnalogDAW's render workers join it as real-time audio peers. Consecutive buffers reuse one short active worker burst instead of repeating setup hundreds of times per second; helpers leave and return to their lower-overhead parked state when that burst goes idle. If measured parallel rendering remains slower than the serial callback that requested help, AnalogDAW automatically retreats and cools down before retrying. When the workgroup is unavailable, playback uses the same deterministic mixer with normal system scheduling.
Background anticipative read-ahead is not enabled in the current build. Its first implementation could not reliably stay ahead of 32/64-sample callbacks, so prerecorded tracks currently use the synchronous multicore path too. This avoids silence/dropouts from a starved read-ahead ring while that scheduler is being redesigned and device-validated.
This only changes how fast the mix is computed, never what it sounds like: the output is bit-for-bit identical no matter how many cores are used, and an exported file is byte-for-byte the same every time — safe for mastering. It pairs naturally with a larger buffer for the most headroom while mixing.
Resulting Latency
A live readout of the round-trip latency (and the output portion) for your current buffer and device, in milliseconds. The transport's tracking safety pill shows a related monitoring-latency estimate — roughly the buffer time, plus any master Maximizer look-ahead you have enabled — rated with a green/orange/red light. Lower is better for recording.
Recording Delay
A fine compensation control (±2048 samples) that nudges recorded audio earlier or later to line up perfectly with the rest of the session. Most setups need 0, but some interfaces report their latency slightly off; if your recordings land a hair early or late against the grid, adjust this and re-test. Use the reset button to return to 0.
Sample Rate (current)
Shows the sample rate the hardware is currently running at. AnalogDAW drives your interface to the project sample rate, which you set in Session Settings.
The project rate wins. AnalogDAW actively changes your interface's rate to match the session — it does not resample the session to fit the interface. That happens whenever you:
- open a session whose rate differs from what the interface is running,
- change the project sample rate in Session Settings, or
- select or hot-plug a different input or output device.
You may notice the audio drop out briefly while the interface re-clocks; some interfaces take a second or two to settle, and a few will change rate for every app using them. AnalogDAW never does this while you are recording or exporting — a rate change requested then is held back and applied the next time the transport is safely stopped, so a take can't be interrupted mid-render.
If the interface can't run the project rate, this row turns amber with a warning triangle: the engine falls back to a rate the device supports, and recording and hardware input monitoring are disabled until you bring the two into agreement (open Session Settings and pick a supported rate — the warning offers a one-click switch). Any selected Record Arm and Input Monitoring buttons on affected tracks switch off when they become disabled; the track's input assignment, clips, takes, and comp stay unchanged.
The readout and affected track controls also update after rate changes made by another app or the interface's control panel; you do not need to reselect the same device to refresh them.
Processing Threads
Informational — shows how many CPU cores are available for audio processing.
Bluetooth headphones
AirPods and other wireless headphones play at the session's full rate.
A Bluetooth headset can run one of two profiles, and the choice covers the whole connection rather than one direction:
- High-quality playback — listening only, at the session's full rate (48 kHz on AirPods Max).
- Hands-free — opens the headset's microphone and, in exchange, drops the link to a telephony rate (24 kHz on current hardware, half of 48) for everything you hear.
Anything that opens a wireless headset's microphone switches the connection to the second one. So AnalogDAW does not: it never picks a Bluetooth headset as your input device on its own while any wired input exists — an interface, or the built-in microphone. That covers a headset macOS has made the system default input, and a headset that arrives mid-session, which is offered as an output only rather than as both halves of a Switch. Connect AirPods, press play, and you get full quality.
(On a Mac with no wired input at all, AnalogDAW does fall back to the headset — being unable to record from anything is worse than recording at a voice rate.)
The trade is that a headset's own microphone is not something AnalogDAW reaches for. Track through an interface, the built-in microphone, or a wired headset, and keep the wireless ones for listening — which is what they are good at.
If you pick one anyway. On a Mac you can still choose a Bluetooth headset in the Input Device list; some people want it for a scratch vocal or a voice memo. AnalogDAW allows it and warns you underneath the picker, because the cost is not obvious: while that headset is also your output, everything you hear drops to half rate. Switch the input back to an interface or the built-in microphone to restore full quality. On iPhone and iPad the headset microphone is simply not offered, for the same reason.
Hot-plug and disconnect handling
If you connect a new audio device mid-session, AnalogDAW offers to Switch to it or Keep Current. If a device you were using disconnects, you're notified so you can pick a replacement in Audio Settings. Tracks assigned to the missing interface keep that assignment visible for repair. A brief device-discovery refresh does not discard your Record Arm or Input Monitoring choice if the interface immediately reappears. Once the disconnect is confirmed, those buttons switch off and remain disabled. Reconnecting after a confirmed disconnect does not silently re-arm a track; confirm the input and arm it again before recording.
Control Surface
The Control Surface ▸ Configure… row opens the setup for external MIDI: hardware mixing/transport controllers (Mackie/MCU surfaces or generic MIDI controllers) and MIDI Clock sync for sequencers, arpeggiators, virtual MIDI ports, and hosted instruments. See Control Surfaces.
Next: Exporting & Sharing →