Real-Time Audio Diagnostics

Web Audio & Stream Latency Benchmark

Benchmark client hardware audio latency directly in your browser. Inspect hardware input sample rates, audio context buffer delay, and FFT frequency bands required for real-time speech-to-text.

Ready

-- kHz

Hardware Sample Rate

-- ms

Audio Context Latency

Streaming Telephony Architecture Standard: Voice AI engines (Deepgram, Cartesia, OpenAI Realtime) require 16-bit linear PCM audio sampled at 16,000Hz (16kHz) or 8,000Hz (8kHz for G.711 telephony). Client Web Audio contexts must downsample high-fidelity 48kHz audio buffers without inducing garbage collection pauses.

AudioWorklet Downsampler (48kHz to 16kHz Mono)

// Linear PCM Downsampler (AudioWorkletProcessor)
class PCMDownsamplerProcessor extends AudioWorkletProcessor {
  constructor() {
    super();
    this.buffer = [];
  }

  process(inputs, outputs, parameters) {
    const input = inputs[0];
    if (input.length > 0) {
      const channelData = input[0];
      for (let i = 0; i < channelData.length; i += 3) {
        const s = Math.max(-1, Math.min(1, channelData[i]));
        const val = s < 0 ? s * 0x8000 : s * 0x7FFF;
        this.port.postMessage(val);
      }
    }
    return true;
  }
}
registerProcessor('pcm-downsampler', PCMDownsamplerProcessor);

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