Increasing sample rate only increases the frequency that can be represented. The waveform for an A440 sampled at 1K is absolutely identical to an A440 sampled at 192K.
Having a higher frequency leaves you with more time slots to put sound to, therefore allowing you to push sound through your computer a little bit faster. Not only that, but thatâ€™s the point here.
Except, for practical purposes, I think you have to consider CPU load as well as any number of other factors, including the interface, driver quality etc. that may require you to increase your buffer size when using a high sample rate to avoid artifacts.
I know. At this point, letâ€™s say we are to use supercomputers for the purpose. DragonPianoPlayer explained the driver part. Iâ€™ll give an example for CPU, RAM and the rest at the end of this post.
In any case, for simply recording audio, 192K is way overkill, unless it's for a trio of porpoise, field mouse and fruit bat.
I somewhat agree. Depends on what you do and I wouldnâ€™t use 192kHz for recording. I am not considering latency jitter here, just the average keystroke-pleasure time, but it can reduce problems with latency jitter, too. Itâ€™s important for playing because we donâ€™t want to wait for responses after hitting keys. I like to hear sound response immediately (<5ms).
Also, the engineering of such converters requires a compromise between speed and accuracy, so unless the 192K converter is extremely expensive, it may very well be introducing some distortion just by itself that simply wouldn't occur at a lower rate.
I believe ASUS Xonar Essence STX can help a lot. Also, you may go with 96kHz, if you hear distortion while playing at 192kHz.
OK, let's be sure we are talking about the same thing. This thread concerns MIDI signals via a USB to Host connection. MIDI by itself has little to no latency. Latency appears in this scenario when you have to deal with (for example) a VSTi, audio driver, and the appropriate output buffers.
Yes. MIDI is limited by the rest of the system, and thatâ€™s USB in this case. No creature on Earth can percept these latencies.
I'm thinking that the VSTi and audio drivers are the biggest sources of latency in this scenario. I would agree that the latency due to the buffers scales inversely with the sample frequency. You can also minimize the latency of the audio driver by using an ASIO driver (on Windows, because this works directly with the hardware and does not need to program through multiple interface layers, if I understand the importance of ASIO drivers).
Yes, in case weâ€™re dealing with an old PC, but I guess whoever wants to do stuff weâ€™re discussing here will buy a good enough computer in a year or two. By good enough I mean capable of making latency produced by VST host and VSTi close to zero. And aving ASIO, weâ€™re basically left only with buffer latency.
Doesn't the DAW need to move or create twice the amount of data at 44k as at 22k? Aren't the sample sizes on the disks or in RAM twice as big? Or am I missing something here? I'm lookin at this from the viewpoint of someone with a degree in physics.
Right. Youâ€™re not missing anything. I slept well, so neither am I.
A friend of mine has a studio not far from my apartment. For this sort of games, he uses Phenom II 550, 4 GB DDR3 1333MHz. Iâ€™m not sure about HDD. Interface is M-Audio, both controller and audio processing unit. Again, not sure about models. I tried it and it works.