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by steve1977
708 days ago
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In audio, you can usually tolerate a few milliseconds of latency (more when mixing, less when recording). This means you can buffer your audio stream, for example in blocks of 64 or 128 samples. As far as I know, these millisecond latencies are orders of magnitude higher than what would be tolerable in HFT. There the units are microseconds and nanoseconds. |
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