What causes latency in audio APIs?
Latency in audio APIs can be caused by several factors, including buffer size, processing time, and network delays.
How can I reduce latency in my audio application?
To reduce latency, you can try lowering the buffer size, optimizing your audio processing code, and using a faster network connection.
Are there specific audio APIs that are better for low latency?
Yes, some audio APIs like ASIO and WASAPI are designed for low-latency performance, especially in professional audio applications.
What is the ideal buffer size for real-time audio processing?
The ideal buffer size can vary, but typically, a size between 64 to 256 samples is recommended for real-time audio processing.
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