Do these 3 things before closing this tab:
1Fix the driver behind crashes, sound loss and screen glitches2Repair Windows errors before they cause bigger problems3Scan for outdated or missing drivers - takes under a minuteA crackle, a delayed note, a click at the loop point and a late first attack can sound like the same problem—but they have different causes. Crackle often points to a missed real-time audio deadline; lag is a latency trade-off; a seam click can occur in otherwise clean playback; and a clipped first transient may begin before recording is ready. Diagnose each separately before changing buffer settings or adding fades.
What kind of audio bug are you hearing?
| Symptom | Likely area to investigate |
|---|---|
| Random crackles, gaps or dropouts during playback | Real-time callback workload, buffer depth, scheduling, or the audio path |
| Notes feel delayed while playing or monitoring | Round-trip latency, buffer depth, sample-rate conversion, or the active route |
| A click exactly where a loop wraps | Waveform discontinuity or non-zero amplitude at the seam |
| The first note sounds clipped or starts late | Capture timing, controller latency, or recording-start behavior |
| Playback stops or behaves differently after a call or route change | Interruption and audio-session lifecycle handling |
These are starting points, not guarantees: more than one issue can coexist. In particular, a click at a loop boundary does not prove that the audio stream underruns.
Why does a mobile looper crackle?
Real-time audio has a deadline: the app must deliver each block of audio before the system needs it. If processing misses that deadline, the result may be an underrun or another interruption in the audio flow. Google’s Oboe guidance describes glitches as commonly associated with underruns, but buffer size is only one possible cause. Blocking work in the audio thread, excess DSP, scheduler preemption, workload-dependent clock changes, content underflows and audio-path defects can also contribute. See Oboe’s glitch mechanisms and its troubleshooting guide.
Keep the callback predictable
Android’s low-latency audio guidance warns that callbacks can arrive only a few milliseconds apart. Keep callback work bounded and avoid:
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- Allocating or freeing memory
- File or network input/output
- Waiting on mutexes or other locks
- Sleeping
- Heavy one-time computations
Move non-real-time work elsewhere and design a deliberate way to pass data into the audio thread. In a looper, that includes avoiding unpredictable work when a loop is armed, a layer changes, or a recording is committed.
Use symptoms to narrow the cause
Track underrun or xrun counts where the platform exposes them, but do not treat a clean count as proof that every route is glitch-free: a device-path defect may not increment the counter. Oboe’s troubleshooting note offers useful clues: if very large buffers do not help and the problem is worse on slower devices, the workload may exceed available real-time compute. If glitches are intermittent, coincide with system activity, or ease when the buffer grows, scheduler preemption is one possibility—not a confirmed diagnosis.
How should you tune the buffer?
A deeper buffer gives the system more time to deliver audio and can reduce deadline misses, but it also delays what the player hears. For a looper, that trade-off directly affects monitoring and the feel of playing along. There is no single safe setting that guarantees both low latency and stable playback across phones and routes.
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Start from the opened stream, then measure
Android’s current low-latency guide recommends tuning after opening the stream. It describes Oboe’s default usable buffer as two bursts and demonstrates setting AAudio to two burst sizes as a starting point. If a buffer that is too small causes underruns, increase it and check AAudioStream_getXRunCount(). Treat two bursts as a starting point for measurement, not a universal target.
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| Configuration in the Android Developers example | Reported round-trip latency |
|---|---|
| Following the listed recommendations | 20 ms |
| Performance mode not set to low latency | 205 ms |
| AAudio at 44.1 kHz | 160 ms |
| Oboe sample-rate conversion at 44.1 kHz | 23 ms |
Android explicitly cautions that results can vary greatly between devices. These configuration-specific figures are not predictions for a particular looper, phone, or Bluetooth route, and they are not population averages. Measure round-trip latency and glitch behavior on the devices and input/output routes you intend to support.
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Why might Android ignore the audio path you requested?
A request for low-latency playback, a sample rate, or a particular stream mode does not establish that the device actually opened that path. Android recommends Oboe for the lowest possible latency; Oboe uses AAudio on Android 8.1 (API level 27) and later, and OpenSL ES on earlier Android versions. Android also recommends low-latency performance mode and exclusive sharing mode, while noting that actual stream capabilities depend on the device. Verify the opened stream and active route rather than assuming the request was honored.
Check sample rate, callback use and format
Android’s guide recommends using the device’s natural sample rate where possible—commonly 48 kHz—or using Oboe for sample-rate conversion when the app needs another rate. Framework resampling can add latency and may prevent the expected low-latency path. The Oboe FAQ also lists missing callbacks, unsupported channel counts and other stream-availability constraints among possible causes of an unexpected path.
Account for Bluetooth and stream limits
The Oboe FAQ specifically identifies unsupported Bluetooth low-latency streams and exhaustion of available low-latency streams as possible reasons a request is not honored. Test the routes your users will actually use, including Bluetooth where relevant; do not infer wired-route behavior from a Bluetooth test or vice versa. If low-latency stream availability is a constraint, one stream with in-app mixing may be more practical than opening a separate playback stream for every loop layer.
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What should a looper do when iOS audio is interrupted or rerouted?
On Apple platforms, interruptions and route changes are part of normal audio-session lifecycle handling, not exceptional states to ignore. An incoming call, for example, can take audio focus and pause playback. Apple’s interruption guidance says to observe audio-session notifications and update the app’s response when interruptions begin and end.
Resume only when the app’s prior state calls for it
Apple’s archived Audio Guidelines By App Type says: “When an interruption ends, don’t start playing or recording audio again unless the app was doing so prior to the interruption.” For a looper, preserve enough transport and recording state to make that decision deliberately; do not automatically restart a recording that the user had not started or that the app had not been running before the interruption.
Recheck audio properties after a route change
A route change can alter sample rate, buffer duration and latency. Query relevant audio properties again after the change and adapt the audio engine as needed. The cited Apple guide is archived, so check the current SDK documentation for the exact APIs and platform requirements in the version you ship.
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Handle recording permission explicitly
Request recording permission explicitly and give the user a clear recording state when access is unavailable or an interruption occurs. The archived Apple guidance recommends updating the UI during interruptions; a visible state helps distinguish a paused capture from a loop that is still recording.
Why does a looper click at the loop point?
A loop can click even when its stream has no underruns. If the waveform jumps abruptly from the loop’s end back to its start—or either edge has a non-zero amplitude—the discontinuity can be audible. Loopy Pro’s troubleshooting article on transients describes these boundary issues and product-specific fade and crossfade controls.
Make the wrap continuous without smearing the attack
Inspect the waveform at both edges. A short fade or crossfade can soften a discontinuity, but too much can blur a transient that should remain crisp. Apple’s AVFAudio sequencer example avoids clicks from a non-zero-crossing clip start by setting volume to zero and ramping it up. The example reports an approximately 5 ms render-buffer smoothing interval for that implementation; it is not a guaranteed duration on every device or audio engine.
Separate a seam click from a clipped first attack
If the problem is at the start of a newly captured loop rather than at every wrap, investigate capture and control timing before editing the seam. Loopy Pro discusses early playing and controller latency as possible causes, along with compensation and threshold-recording options in its own product context. Those are not universal mobile-platform features.
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How do you stop repeated loops drifting out of time?
Do not assume that replaying a floating-point duration forever will keep the musical grid exact. Apple’s sequencer example manually schedules each iteration at the loop boundary rather than relying on its built-in looping option, to avoid timing drift accumulating from floating-point error. For a looper, anchor transport events to stable frame or host-time positions and test many consecutive wraps. A first cycle that sounds aligned does not establish that later cycles will stay aligned.
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