![]() So the final verdict would be to resample all my samples to 44.1Khz in Sound Forge before loading them into my song in Renoise? (as I noticed there were a few others at 22KHz too)Īnd if sampling rate is independent from the number of channels, why all the 96000KHz and 192000KHz sampling rates you hear about as used in studios and 5. And It-Alien, I think you’re right about it using a filter of some sort, it was just that I was doing it the wrong way first and it didn’t actually tell me it was using an interpolation or something. So this 22KHz sample I have is theoretically only able to contain frequencies from 0-11024Hz, as the EQ plugin graph showed. So, when playing a 96000Hz file with variable rate setting, 96000Hz audio data is fed to Android and then sample rate converted to either 44100 or 48000Hz. ![]() ![]() And kmkrbes, I do think I was mixing the two up. When using the variable rate setting, UAPP will feed the sample data as is up to the sample rate that the Android device supports internally (before SRC). I think I’m beginning to understand this a lot more. So is it safe to just set the sample rate, and equalise the sample as if it contained those frequencies? (Of course when changing the sample rate, the sample becomes twice as fast, but I then play the note back an octave lower in Renoise - not sure if this preserves maximum quality, though.)Īnd can someone please enlighten me why some mono samples are 44.1KHz? Isn’t 22KHz all you need if it is mono? (provided you are working at CD-quality)Īny help on any of this is really appreciated. I do want to have a boost at 15000Hz, however, with a Q of 1.0 and a gain of 0.4dB. So I’m not really sure what is going on here. When I set the sample rate to 44.1KHz only (without resampling) in Sound Forge, and use a peak EQ at 15000Hz, I still get an audible boost. However, when I input that 22KHz sample, the range of the graph is only 0-11024. When the input is a 44.1KHz wave, the plugin window shows a graph from 0-22049 Hz, allowing you to enter values in that frequency range. The thing is that I have a 22KHz mono wave file (I still don’t get this, as I thought a 44.1KHz stereo wave has a 22KHz wave per channel) that I want to parametrically equalise using a VST plugin. In Sound Forge I can apply an anti-alias filter while resampling, but I’m not even sure if I need this or not. However, it seems the resampling algorithm is not very good the sound can get rather gritty if resampling between, for example, 44 100 and. Is it ultimately better to resample a 22KHz sample to 44.1KHz using Renoise’s wave renderer (outputting as 44.1KHz 16-bit stereo will of course resample all lower quality samples to this), or using Sound Forge? There is the 'Real time resampling' option in the audio configuration, which resamples the sample to the audio device output's sample rate in real time during playback to get correct review playback rate.
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