Originally posted by datenwolf
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You're going to explain to all the developers of every pro-audio and music creation software who might consider Linux as a platform for their work that their samples will be converted twice, once from the floating point format that their code uses into fixed point and then again into integer format before it hits the audio hardware? And that when they share audio data between applications, which will both be using floating point format, it will be converted to and then from fixed point as an intermediate?
Could you at least spend several months hanging out with audio developers before you try to redesign the kernel subsystems that we rely on?
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