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Post by skandinav on Jan 29, 2024 16:17:41 GMT
Re 3. All daws are adequate for pro studio sound. I ve recently figured out the secret about daw summing quality. For that I have been blocked at 2 crappy sites: warmplace.ru and oldschooldaw.com moderated by bastards. They banned me and removed the info. I will explain the secret now: so the secret is that the bad summing engines do not compensate for phase differences between tracks ie differences between all left channels or between all right channels in all tracks i.e.not differences between l and r. Those differences are not detected by correlometers! They are of the same type as pre-membrane comb filtering between direct sound and room reverb happening with one monomic. So if same frequencies for example in kick and bass guitar are of opposite polarity they will null during mixdown. In a room such crap never happens when you play a signal and polarity reversed signal from l and r speakersbut in digital domain that crap happens. That's why people get muddy mixes. Only 2 daws currently provide remedies for such phase decorrelation,they are old emagic logic and digital performer. Enjoy! How those 2 daws solve phase crappiness? They convert sound to fft form where there are no negative numbers for polarity/voltage, so numbers are always positive and summed as positive which provides even cleaner mixing than on analog consoles or inside a premise.
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Post by skandinav on Jan 29, 2024 23:12:17 GMT
Re 3. Addition: same crappy nulling happens in samplers like ni kontakt when playing 2 or more notes (poliphony). The sampler without this nulling is evidently motu machfive3. Poor kontakt users... Free shortcircuit was even of higher quality. There are some claims that last good version of kontakt was 2. But due to limited number of libs for it it's useless except for loading old sf2 which is pointless as they can be loaded to better samplers as shortcircuit. The peculiar trait of phase decorrelation fix is rigidness in sound which is opposed to grainy soft sound of daws/samplers without such fix.
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Post by skandinav on Jan 29, 2024 23:13:03 GMT
Re 3. Addition: same crappy nulling happens in samplers like ni kontakt when playing 2 or more notes (poliphony). The sampler without this nulling is evidently motu machfive3. Poor kontakt users... Free shortcircuit was even of higher quality. There are some claims that last good version of kontakt was 2. But due to limited number of libs for it it's useless except for loading old sf2 which is pointless as they can be loaded to better samplers as shortcircuit. The peculiar trait of phase decorrelation fix is rigidness in sound which is opposed to grainy soft sound of daws/samplers without such fix.
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Post by skandinav on Jan 30, 2024 8:14:46 GMT
Re #2. Impotant point is if cheap chinese mics are incapable to capture high frequencies then neither boosting highs nor lowering mids is of help since there is nothing to boost or reveal except noise. Any type of synthesizers of highs like steinberg spectralizer, prosoniq mixciter do not produce believable results. They sound harsh or unnatural. Exciters/saturators are even worse as they produce noise instead of harmonics. Waveshapers seem to produce only series of odd harmonics by squary flattenning waveforms with fake result as well. They act like overdrive. Rode nt1a seems to capture high harmonics as per online video reviews but it's not cheap. I bet all neumans in 90s could capture high harmonics. Even cheap pc plastic electrets like genius could do it. The paradigm has changed to the worse.
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Post by skandinav on Jan 31, 2024 11:43:53 GMT
#7. As per japanese guy of freeverb3 convolution reverbs are of higher quality than algorithmic ones. I suspect it's bullshet from start of the new millenium when altiverb issued first windows version. In fact irs always sound more static and foggy than good algorithmic ones. I will disclose the secret behind best commercial reverbs like adaptiverb and vss4 which explains superiority of algorithhmic reverbs. It all started in prosoniq products who had discovered the way to split sound into harmonics and noise in a satisfactury way in early 90s. The info was published by me at gearspace.com and of course removed by bastards with the corresponding ban. So what does that tech allow? It allows to add sustain to harmonical content without smashing noise in time, also it allows to reverberate harmonics dualmonolly but noise stereophonically so only noise will have opposite polarity and will disappear at mono summing which underlies vss4's so called mono compatibility. Those top algos may have also other improvements but corner stone principle is always the same. History of the method: roomulator-rayverb-adaptiverb-vss4. Prosoniq's audio editor could do mythical things as per rumours beating mediocre ozon rx alltogether but was never issued for windows. Most high quality products like cedar are not known/promoted to general public. All of them of course handle phase as correctly as digital performer mentioned earlier here.
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Post by skandinav on Apr 22, 2024 20:50:44 GMT
#4. Have been recently astonished to switch between monoized and stereo version of a song with my ~115° loudspeaker angle which i had somehow missed. I was shocked by the fact that mono sounded fuller with the same width but more detailed and pronounced center while in stereo there was kind of a hole in center though the song was mixed to have phase difference no worse than +0.5. So wide placement of speakers as i have written before creates own powerful stereo effect and mono sounds totally like stereo though stereostage is missing but it's nothing like listening to mono in headphones - it's more like real performance in concert hall far from stage where reverb dominates over direct sound. Some benefits of incorrect placement of speakers really exist. While listening at 45°-60 angle sound was too artificial as it was too narrow.
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Post by skandinav on Apr 22, 2024 21:54:11 GMT
#4. It's the question of what matters more: frequency spectrum or position of instruments in stereofield? For example, in foobar2000 on android which is ultimate quality soft player to my mind it's impossible to use mathaudio correction curve to fix headphones frequency response therefore i had to eq by ear. The procedure of equing in fact is almost the same everywhere: supressing lowmids and tilt eq boosting highs with peak boosting lows or not touching lows {for loudspeakers). The result can be miraculous beating results from scientifically produced curves from roomeq. If even well mixed music from hi end hw can be further improved by some own manipulations why avoid those? But badly/recorded mixed music indeed needs this equing. I have not found a case where such equing would spoil any music. How increased clarity and transparency can spoil music if the result is less masking of high harmonics by low harmonics? But if we return to the beginning of the post the conclusion will be like that: if incorrect unscientific angle of speaker placement gives more pleasant frequency response to audio material it can be more preferrable than more precise location of instruments in stereo field. Hifi has the right to exist by one major reason: however well music is mixed it's ultimately hw's reproduction of highs which defines the clarity of music and that reproduction can be improved by hardcore equing matched for hw in question. I suspect that equing pair of 80kg (each) hifi heco anniversary speakers may render better results than buying some pro monitors of the same cost but much smaller size. I suspect this topic is the unexplored land and still requires tests and comparisons. For me it was a suprise that so called pro headphones by akg, sennheiser turned out to be crap as to their frequency response and no better than hifi philips headphones... all of them had to undergo calibration procedure with roomeq and obtained hardcore correction curves which have made them suitable for mixing. By the way, stay away from using correction curves from internet. All of them are fake!
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