Initial commit: RemSound v1.0
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namespace RemSound.Core;
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/// <summary>
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/// How often the sender cuts the audio stream into a packet for transmission. Smaller frames
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/// = more packets per second = lower send-side latency, but more network/CPU overhead per
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/// second.
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///
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/// Mapping per codec:
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/// * PCM: Standard = 5 ms (240 samples), Tight = 2.5 ms (120 samples).
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/// * Opus 20 ms: Standard = 20 ms, Tight = 10 ms.
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/// * Opus 10 ms: Standard = 10 ms, Tight = 5 ms.
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///
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/// "Tight" is documented as LAN-only because the smaller frame size means less time for the
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/// network to absorb jitter before the next packet arrives. On a stable LAN it cuts ~2.5 ms
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/// off the send-side accumulator latency without audible cost; over WAN with typical jitter
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/// it'll glitch.
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/// </summary>
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public enum SendRate
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{
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Standard = 0,
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Tight = 1,
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}
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