is 44.1KHz a sufficient sampling rate?
So there is a question why to downsample anything to 44 kHz when it is likely to get upsampled upon playback...
[blockquote]A. If a low pass filter has been identified with exceptionally good performance as a prefilter for a conversion to 44.1KHz, is it available as a plug-in, or as standalone software? [Or is the digital filter in Audition 3 about as good as it gets?]
B. If capturing analogue audio at 44.1Khz, are analogue pre-filters available that outperform the best digital filters that might be used in a digital to digital downsampling as referred to in question A?[/blockquote]
B. If capturing analogue audio at 44.1Khz, are analogue pre-filters available that outperform the best digital filters that might be used in a downsampling as referred to in question A?
[No one samples at 44.1kHz. They sample at many times this, then filter and downsample digitally - precisely because analogue filters are so pitiful (in this application) compared to digital. All but the worst ADCs and DACs are oversampled.
A. If a low pass filter has been identified with exceptionally good performance as a prefilter for a conversion to 44.1KHz, is it available as a plug-in, or as standalone software? [Or is the digital filter in Audition 3 about as good as it gets?]
This really must be my last post for now!My AVR is a Yamaha HTR-5750. Would its DAC perform ok at 44.1KHz (fed from a pc running Vista) for the purpose of comparing the three files?Alternatively if I burned a CD I could play it on a CD player. Could a CD player be expected to perform sufficently well to compare the three files [e.g. Denon DCM-370]? Presumably I'd dither down to 16 bits.
MLXXX, could to try to ABX the unfiltered file versus the lowpassed ones? The lowpassed ones have a suffix that goes from l1 to l4.
It sounds like you are confusing resampling with oversamling. Soundcards do not resample to 96 kHz, or any other high frequency. Resampling involves recalculating all sample values. DACS oversample, usually by much larger amounts (e.g. 128x to 5.6 MHZ) which means interpolating additional values between the original samples, without changing the original values, shifting the alias images to a very high frequency range.
[You probably cannot provide fair enough conditions for such test.First - you need a high-quality downsampling algorithm (which uses a steep filter with linear phase/constant group delay), are you sure you can get this? If you cannot, do not bother.
Indeed, if I cannot find a high quality downsampling algorithm for converting the 96/24 recording to 44.1Khz, without introducing audible diminishment or artifacts as a result of the digital filtering settings, I cannot proceed further.
MLXXX, I think there is something wrong going on with your audio setup. The first lowpassed sample, triangleMod_2_2496_l1.flac, has been lowpassed just over 45 KHz with a high quality filter (sox filter with a 256 sample window: sox filter 0-46000 256). The files are different only over 45 KHz (they go up to 48 KHz). You can test this by substracting the original and the filtered files.There is also some pre-ringing before the triangle hit due to the filtering, but it is around 40 samples long (0.4 ms) has a frequency over 45 KHz and a level of around -65 dB, not audible by any means.Also, I don't think your speakers or headphones can go that high. So the difference is most likely due to something else. Maybe your card is resampling or it is clipping, or there is some strange kind of intermodulation going on. As for intermodulation, I find strange that filtering such a small frequency range, where there is not any exceptionally high energy, causes an audible intermodulation difference into the audible range.The right channel peaks at -0.47 dB before filtering. After filtering at 45 KHz, it peaks to -0.68 dB. Maybe the first one is clipping and the second one is not, but this is only speculation.
Something very odd happens with this triangle file.
It seems that this triangle sample suffers from an "Intersample Overload". Although the sample values are below maximum, the reconstructed signal is clipping...This could be considered as a non legitimate signal since it might cause clipping in DA and sample-rate converters.I'd like to suggest to reduce the level of the triangle sample by 3 (or even 6) dB and redo the tests.
My Terratec DMX6 fire 24/96 sound card does not resample and I used ASIO and Kernel streaming output modes for bypassing Windows Kernel resampling.For excluding possible software/device problems at certain sample rates I resampled the file to 44.1 and then back to 96 kHz. These two versions sound identical to me (couldn't ABX). For this test I used foobar's SSRC in the Ultra mode.Something very odd happens with this triangle file.