I localized the problem to my old analog mixer that I have used between the sound card and the powered speakers.I tested this with a difference signal of the reference and the #4 (invert mix paste). When the mixer is in the chain the difference signal sounds like a rather loud "pop".
triangleMod_lp1.flac - This is the triangleMod sample with a lowpass filter nominally at 22kHz - the transition band is quite wide, but the filter has very high stopband attenuation.triangleMod_lp2.flac - This is the triangleMod sample with a lowpass filter nominally at 20kHz - the transition band is quite wide, but the filter has very high stopband attenuation.
For your ABXing pleasure - versions reduced by 3dB and 6dB.
Cabbagerat, thanks for the first set of uploads. [The file wiith the added noise is very noisy indeed when played back on my systems. The noise gives the impression, on my equipment, of being well within the audible range.]
Edit: the applied lowpasses are:triangleMod_2_2496_l1.flac : 46 KHz lowpasstriangleMod_2_2496_l2.flac : 30 KHz lowpasstriangleMod_2_2496_l3.flac : 26 KHz lowpasstriangleMod_2_2496_l4.flac : 21 KHz lowpassThe lowpass takes 2.5 KHz approx. For example, in case of the 21 KHz lowpass, it is -0.1 dB at 19950 Hz and -90 dB at 22600 Hz.
would that be a basis for concluding that if |3 were downconverted from 96KHz to 44.1KHz, its perceptible sound quality would change to a similar degree?
Quote from: MLXXX on 07 May, 2008, 11:55:24 AMA. 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?]A comparison of some SRCs (96 -> 44.1) is available on http://src.infinitewave.ca/. But there's only Audition 2, not 3. Anyway, "Adobe Audition 2 Pre/Post Filter" graphs look good enough.
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?]
2. KikeG's version_4, which from its spectrum looks like a 22KHz lowpass filter does appear to sound very slightly duller than his version_1, but I will try to get my Vista HTPC SPDIF operating at 96KHz, before making a more specific comment.
I have a disquiet about 44.1KHz, but I have not so far been able to prove that, for my own middle-aged hearing, it is an inadequate sample rate.
Your concern should not be with the sample rate, it should be with the playback circuit architecture.
I think what he meant was there is no comparison data for Audition 3 on the SRC site.
krabapple, I was leaving that to last.At this stage (by comparing the |4 and |1 samples), I am trying to establish whether digital filters exist that can be placed in a 96KHz digital path so as to reduce the response in the digital domain at 22050Hz to a negligible level, but not affect the perceived played back sound using a 96KHz rate DAC.By keeping the playback at 96KHz I am removing one set of variables from the analysis; though it appears creating another complexity in the analysis; as per my next paragraph.What I have not read up on, is how current sample rate converters handle the pre-filtering necessary to avoid aliases. They appear to sidestep (or postpone) the issue somewhat by oversampling in the first instance such that pre-filtering requirements would be much laxer. But still as part of the processing, any part of the source signal that was in the zone just under 22050Hz must be strongly attennuated if not eliminated. Is there reason to suppose that the postponed pre-filtering would be more effective at not disturbing source frequencies in the zone below 22050Hz, than the digital filtering used for upload |4, provided to us by KikeG?I come to this forum with a dated general knowledge of electronics, and a curiousity to understand why 96/24 is being so strongly promoted if 44.1/24 (or even 44.1/16) are really quite adequate, even in extreme circumstances such as a cowbell concerto!
My results were:DAC of old audio video receiver: nothing audible unless volume above a particular setting (presumably non-linearity commenced in the amplifier at that point)
As far as I know, static nonlinearity (found in amplifiers) spawns only higher harmonics (integral multiples of base frequency), not frequencies lower than the base one. Perhaps the device has poorly designed power supply and you hear some interference from power grid or air.
MLXXX, can you record this 'click' with, say, soundcard of your HTPC? It would be interesting to look at its spectrum.
Quote from: MLXXX on 08 May, 2008, 10:11:41 AMIndeed, 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.According to infinitewave, SSRC which is freely available as a plugin for Foobar2000 seems pretty much perfect to me. In most graphs it comes pretty close to the white 'ideal' line. I use it to convert 5.1 channel 24bit/96kHz DVD-A rips. Why not give it a try?
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.
This illustrates the tension in digital filtering between quantity (a wide passband bandwith) and quality (minimal spurious responses).[/color]
SoleBastard,Sorry I overlooked your post until now. SSRC indeed has an enviably steep cut-off. However, I see that in the the infinitewave sweep graph for it, there are some artifacts.R8brain free is probably at the other extreme: wonderfully free of artifacts, but with a mild rolloff (commencing at just over 18KHz).This illustrates the tension in digital filtering between quantity (a wide passband bandwith) and quality (minimal spurious responses).