It Looked Fine Until It Wasn't

We build a lot of audio software. Karaoke apps, DAWs, voice changers, you name it. Our customers often use cross-platform frameworks like Electron and React Native so they can share as much code as possible across mobile and desktop platforms. That makes sense. The temptation to use WebAudio sneaks in during early development, especially when teams are moving fast.

Unfortunately, as the product grows and demands become more sophisticated, the limitations start to stack up. We hit a hard wall any time we try to move between managed and native territory with timing-sensitive code. For example, we want to run real-time AI inference on microphone input, things like source separation, noise suppression, speech enhancement, or even translation. This should be routine work in 2025. But it's not routine in the browser.

The browser stack gives us no way to guarantee the real-time performance we need. You can't control buffers precisely. You can't prioritize threads. You can't coordinate work with native layers without paying a penalty. That lack of determinism makes anything advanced a gamble. If you're trying to build a serious audio app on WebAudio, you end up wrestling against the system instead of working with it.

What WebAudio Actually Gave Us (and What It Didn't)

WebAudio covered the basics. We were able to build graphs with GainNodes and filters. We had microphone access through getUserMedia. We could see the routing paths in devtools. It worked well enough for prototyping and simple flows.

But when we tried to do more, it fell short. We needed control over buffer sizes. We needed device-level synchronization. We needed to manage multiple input and output channels with confidence. None of that was available. Latency was inconsistent. Underruns appeared without warning. Device behavior varied from machine to machine. It felt like trying to build a production studio on playground equipment.

Audio is Systematically Underestimated

Audio problems are subtle. It's different from other subsystems. UI failures are obvious. Network requests time out or throw clear errors. Audio degrades slowly. Distortion creeps in. Sample drift builds. Users don't have the vocabulary; they tell us “it's choppy” or “I couldn't hear anything.” Our customers often think “it's just audio, we'll add it at the end.”

We understand, finally, why every serious audio application avoids WebAudio. Whether it's a DAW, a DSP engine, or a voice processor, they all rely on native stacks like CoreAudio, ALSA, or ASIO. They aren't doing that for fun. It's because those are the only layers that give real control.

What We Learned About WASM the Hard Way

We weren't naive. We figured we could use WebAssembly to bridge the gap. We took a stable C++ audio engine, compiled it, and dropped it into the browser. It ran. Sort of.

Threading was the bottleneck. Shared memory worked in theory, but coordination was inconsistent. Thread startup was sluggish. Priorities were not reliable. We couldn't guarantee timely buffer handling. Inference models choked under unpredictable scheduling. Trying to hold a real-time pipeline together across serialized or message-passed layers was exhausting and ultimately fragile.

We tested the same model in Electron. Threading was slightly better but still unreliable for real-time work. React Native was worse: the JS thread simply couldn't keep up. WASM wasn't saving us from the core limitations of these platforms.

So We're Building It

We are building a real audio layer. It's native and portable. It runs on macOS, Windows, Linux, iOS, and Android. It gives you full access to define and deploy your audio graph in high level code, and run in the native layer. It's stable under load. It behaves the same across environments. It lets us build the features we wanted without wrestling with the runtime.

It integrates with Electron. It integrates with React Native. It doesn't ask you to leave your stack behind, but it also doesn't pretend the browser can do more than it can.