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To make a browser control change sound on the next beat, schedule the audio event in advance on the Web Audio API’s audio timeline. Do not start sound from a JavaScript timer or animation callback when the beat arrives: those callbacks can jitter. Keep the audio event’s target time separate from the visual update that shows the beat.
Why JavaScript timers should not start the sound
A browser sequencer has two timing jobs: decide when audio should play, and update the interface so it reflects that event. JavaScript timers are useful for waking a scheduler periodically, but their callbacks are not a reliable audio clock. If a callback runs late and only then starts a sound, the beat can slip.
Web Audio provides a separate timeline. An AudioContext exposes currentTime in seconds on that timeline, and scheduled source-node events use times on the same timeline. The Web Audio specification defines scheduled times relative to the context’s currentTime (W3C Web Audio API 1.1 Working Draft; MDN: BaseAudioContext.currentTime; MDN: AudioScheduledSourceNode.start()).
Build a look-ahead scheduler
Rather than asking JavaScript to start a sound exactly when a beat is due, periodically check which beats are approaching and schedule them before their target times. This is the look-ahead pattern in MDN’s step-sequencer tutorial: a timer checks whether notes fall within a scheduling window, then queues those notes on the audio timeline.
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- Use one context as the timing authority. Read the context’s
currentTimefor audio scheduling. Do not substituteDate.now(), which follows the system wall clock. - Calculate the beat period. For tempo in beats per minute, seconds per beat is
60 / tempo. Keep the target time of the next beat, rather than deriving each new target from when a timer happens to fire. - Wake the scheduler periodically. On each wake, compare the next beat’s target time with
audioContext.currentTime + scheduleAheadTime. Schedule every event whose target is inside that window. - Advance from the planned event time. After scheduling a beat, add one beat period to its target time. This prevents timer delays from becoming the new timing baseline.
- Keep event metadata. Store the beat number and target audio time for each scheduled event so the audio and visual layers can refer to the same event.
MDN’s tutorial uses a 25 ms check interval and a 0.1-second scheduling window as example values. They illustrate the technique, not a universal prescription. Choose and validate values for the application and devices you support.
Separate the audio event from the visual beat indicator
When the scheduler finds a beat inside its look-ahead window, it should start or otherwise control the relevant audio node at that beat’s future target time. For example, a source node can be scheduled with start(targetTime). The event is therefore already queued when the beat becomes due; it does not depend on the next UI callback.
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The interface should use the event queue to show which beat is due, comparing each event’s target time with the audio context’s current time. MDN’s sequencer tutorial follows this separation by queuing note times for audio and consulting those times while drawing the interface. An animation callback can update the highlight, but it should not be the clock that starts the sound.
Account for tempo changes and context suspension
Tempo changes
At a steady tempo, advance the next target by the current beat period. If the user changes tempo, define how that change affects already queued beats: preserve their scheduled times and apply the new period to later beats, or cancel and rebuild the relevant future schedule. Whichever behavior fits the instrument, keep target times on the audio timeline rather than recalculating them from timer wake-up times.
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Suspended audio
An AudioContext’s currentTime stops advancing while the context is suspended. The wall clock may continue, so assuming both clocks stayed aligned can leave the beat counter or queued targets inconsistent. Pause or reconcile the beat state during suspension; after audio resumes, establish a consistent next event from the resumed audio timeline. See MDN’s currentTime reference.
Choose latency behavior for the interaction
The latencyHint option can express the kind of audio behavior an application prefers, but it does not guarantee a fixed end-to-end delay. MDN describes the default interactive hint as suitable when audio responds to user actions or should coincide with visual cues. The browser chooses a low latency it can manage without glitches, potentially using more power. The playback hint trades latency for playback time and lower power use (MDN: AudioContext()).
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Choose based on the interaction’s needs and test on target devices: prioritize responsiveness when immediate control matters, and consider the playback trade-off when sustained playback and power use matter more. Neither hint turns browser timing into a guarantee of identical perceived timing across devices.
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For a small sequencer, direct Web Audio makes the scheduling model explicit and gives the team control over its transport and event queue. The trade-off is that the application must implement and maintain those behaviors itself. For more complex interactive audio, MDN recommends considering Tone.js, which provides broader music-application facilities.
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| Choice | Fits when | Trade-off |
|---|---|---|
| Direct Web Audio | The sequencer is small, and the team wants to shape scheduling and transport behavior itself. | More timing and music-application behavior must be built and maintained in the application. |
| Tone.js | The application is more complex or would benefit from broader interactive-music features. | The application relies on a framework rather than implementing all behavior directly with the browser API. |
The choice is about complexity, desired scheduling control, and the value of additional music features—not a promise that one option eliminates timing constraints.
Know what the timing model cannot guarantee
currentTime advances in audio-rendering-block increments while a context runs, and browser privacy settings can reduce its exposed precision. Browser and configuration differences mean you should not promise identical timing behavior everywhere. Validate the scheduling window, visual update behavior, and latency choices with the devices and browsers your application needs to support (MDN: BaseAudioContext.currentTime).
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