giro

Might I also add a plausible reason as to why dipping at 250hz can warm, clear up or 'un-muddy' sound... On sound.westhost.com it was determined that amplifier power splits in half around this region. See under 1.3 - Power Distribution and SPL: http://sound.westhost.com/bi-amp.htm

@4est I've dedicated a fair amount of spare time familiarizing sampling and reconstruction - not so much the maths/boffinry involved, but have toyed with zoh resampler and other gui plugins for example. Funny someone should mention upsampling and acoustics. I'm posting a separate thread with good questions regarding these and I do intend to use REW and a nos R2R dac in future. While on subject of filtering and aware there is no ideal filter due to large multitude of sampling devices and methods used in the past, the following is just a general suggestion; On recording side, oversampling prevents foldback (aliasing) only and doesn't prevent distortion created by eventual low pass filter - Regardless if its linear or minimum phase, or positioned before or after capture stage (capture = 'sampling'), there is always steep decimation circa 20k at some point on all 44.1/48 media. With this in mind identify worse case scenario on A/D side and enter new filter (on playback) to eliminate worse case anomalies EVEN if at expense of reduced frequency response (sounds odd but I explain later on*). Now take distortion-less signal and somehow synthesize harmonics/ambient echoes that would otherwise be present naturally (electronic harmonics/studio ambiance) and impose this on material at playback, with particularly attention paid to nature of cymbals and/or harmonics and maybe my previous post regarding vinyl. This may require extension of audioband but thats easily done. I do know of a few plugins that have a 'step' function and an associated resonance effect (e.g. Sonalksis Stereo TBK) - not sure what STEP filtering is exactly. *Regarding under-cutting flat-to-20k response which is seen on everything today. My opinion is that this response is an annoyance and only prevails because marketing cannot seemingly identify between; Transparency (what producers/engineers want) and Musicality (what average joe wants, especially 'low res' stuff). Designers prefer to comply with what producers want to work with and pay little consideration to 'musicality/warmth' that the human ear prefers. I know warmth can depend on other factors i.e. cone material as mentioned, but that doesn't mean we can't add warmth with dsp (or cooling depending on viewpoint). Aware that some active monitors do employ further analog processing (e.g. room correction - but don't see how low passing below 20k could affect this). Anyway, because filtering was raised I thought I'd diverge slightly but original question regarding vinyl synthesis still stands aswell. Thanks for reading

Hi all, Firstly I didn't hope to invoke much discussion on SS equipment, it's merely a dsp question. I appreciate replies so far but I want to focus on LFO/vinyl/harmonics and now filtering - seeing as someone has raised it (see following post). As I wrote, Izotope vinyl plugin does have "warp" options but these are disabled because vst host (in my case foobar) proves incompatible. I can only imagine this option being a simple LFO, unless someone can suggest intuitive alternative? My guess is perhaps adding subtle LFO of ~0.75hz somewhere, e.g. 45 rotations per 60seconds (60 / 45 = 0.75hz ) if my maths serves well. I also plan on researching how exactly surface noise of vinyl otherwise affects audio frequencies (other than introducing just surface noise). I'm thinking perhaps harmonic 'distortions' (or structure as someone has put) but oscillating and at multiple frequencies. For example: what about super-imposing a 52khz tone (in mono on both channels), oscillating at 0.75hz? I read that a single ultrasonic tone on its own is inaudible but when paired with an audible tone, can change the perceived pitch of the audible one. From what I've learned I believe this is an electrical phenomenon, so supertweeters need not apply unless you are situated exactly in stereo sweet-spot in an anechoic chamber - which doesn't ever happen. 52hkz is just a ballpark figure - I'm positive a 'blank' vinyl exerts sound in ultrasonic band. While remembering material pressed to vinyl is often filtered above 16khz or so, what proves an interesting question is where effective stereo separation ends on a typical LP? Can anyone enlighten? @ tranz The magic frequencies link is interesting - I found a similar pages which sparked initial curiosity believe it or not. Particularly Isochronic tones: 1 Hz @ 27.5 Hz – Delta | Lethargic 2 Hz @ 55 Hz – Delta | Deep Sleep 3 Hz @ 110 Hz – Delta | Dreamless 4 Hz @ 220 Hz – Theta | Drowsy 6 Hz @ 440 Hz – Theta | Fantasy 8 Hz @ 880 Hz – Alpha | Relaxed 12 Hz @ 1,720 Hz – Alpha | Conscious 16 Hz @ 3,520 Hz – Beta | Focused 24 Hz @ 7,040 Hz – Beta | Active 32 Hz @ 14,080 Hz – Beta | Fear /!\ These are more fundamental 'natural' or bin-aural frequencies however, they must oscillate for the 'desired effect'. For more info see Ultimate Isochronic Brainwave Generator | Online & Free Brainwave and Environment Data Visualization | Hiro Noguchi (a good example is 2hz @ 55hz - I have seen this effect in 'ambient' music, it exerts much an "human ohm" effect - very therapeutic) @Dave you make a good point about harmonic structure. With digital, especially how it's composed (at 24/96) alot of the upper end translates badly when down rezzed to 44.1/48k. Theres nothing stopping bin-aural or ultrasonic tones being superimposed at playback, just to make things more natural. This is why I raise the nature of vinyl - perhaps this is why it sounds so pleasant even with associated distortions. Ideas on any of the above welcomed - may prove interesting. Thanks

Hi and thank you all for the replies. I apologize for horrendous spelling. sandyk, All is well my system consists of paper nomex and poly cones. I was focusing on what I could possible do with dsp to emulate vinyl, in a sense. I should have said I don't really wish to spend cash, just play about with various phase distortions (and/or ones that 'vary'/oscillate). low frequency oscillation. I'm thinking I can add more varied sound throughout track. 'riding the wave'. As you can probably tell I'm not scientific and i'm sure it's evident in this post. I have done enough research to know all playback devices are transparent, strictly linear and very little distortion. this and what an lfo is. Well I think all vinyl is warped to an extent. As decades progressed pressing improved, so it became less of an issue. Not sure if warp contributed to warmth, was just an idea, considering a turntable spins, I think to myself, the music must 'ride the warp', so there is more musical variation (no two vinyl press, nor playback session will sound the same). I was experimenting with izotopes 'vinyl' plug-in. It has a warp function but it is disabled for some reason. Something to do with vst host being incompatible (foobar2000) and nothing to do with the fact it's freeware. There is a 'wear' fader/control and this does seem to eliminate some harshness. I realise LFO can be added as 1-100hz. (seconds) But RPM is rotations per minute, so it's not the same.

Ok we all no dac's are designed to be as transparent as possible. But I don't like it and I'm many will agree. So are there any good ways (by means of adding dsps to playback chain) to add warmth to digital media being played back? An I idea I had was perhaps using a tape emulator plug-in used for mastering but using at playback. then I got more creative and thought about LFO's. For example is adding an lfo the equivelant to synthesizing vinyl warp? If not what would be the easiest way to do this? If I understand correctly, introducing low frequency oscilation will 'vary' the sound at a set rate, introducing phase distortion, which is all warmth is = less fatiguing (more varioation). Keep in mind the RPM rates of vinyl aswell, 33, 45, 48 rpm, I think to emulate WARP you just need to add lfo at same rate. There is also wow and flutter, this I've yet to read over. I'm positive that all these distortions makes music. Just a thought, can anyone comment?

Ok thanks for the answers, I did more research, found a good post on this topic here . Thanks again.

There is no evidence of this out of band anywere on the web that I can see, any examples? anyone?

Ok, thanks for the response. y

Regarding playback, I realise the first ever CD Players's (such as those featuring PCM53JP-V) did not feature "oversampling" at all, making analog filters after reconstruction prone to distortion. But why were analog filters introdued? What's the reason? What would be the net output if an analog filter (after any nos device) were removed entirely? any input appreciated

*Purely Experimental* - Back to basics - I'm after a music player that is strictly fixed point front-to-back in order to emulate the process Philips employed for getting 16bit performance out of a 14bit mulitbit. However instead of 14bit, I want to experiment with 8bit (which is not far from vinyl hifi performance regarding SNR). Essentially I want to emulate vinyl. I did research and discovered fixed point decoder MPEG audio decoder (MAD) CoolPlayer is an option but no re-sampling or dither plugins are available for it. I'm sure it wouldn't be too hard to impliment sox plugin for it eitherway. I also found MAD plugin for winamp, I'm sure to achieve "8bit", the plugin upsamples x64 and dithers down for clean 8bit out - though it would be nice to know what it's going on internally and change resampler to sox (also mad based). As I say, the mad plugin for winamp probably is my best bet but incorporating sox would top it off nicely - i'm contacting the developer for more info. Unsure if winamp is fixed point front to back though ~ ? Any thoughts? Cheers

Hello, Just signed up to CA. I am very intriqued in mulitbit. I started researching in my spare time when I wanted to know more about the digital to analog conversion process. Starting with history makes sense to me, is it not wise to stick/work with designs made by the people who invented digital in the first place? The reasons I favour mulitbit: I see bitstream as TDA1540 on roids. My opinion. From what I've read, it would appear noise shaping was formulated to overcome the problem phillips faced with the 14bitter after sony having declared 16bit as the standard (perhaps a marketing /move tactic on sony's part - contending). If it wasn't for this, we might not have bitstream today at all, right? The fact bitstream oversamples x256 aswell. I agree with a statement on another forum: "I think with NOS DACs what makes them sound analog is they use the lowest possible sample rate and that means the lowest possible glitching. ... - running oversampled increases the rate of glitch production by the oversampling ratio." Another point regarding the western economic model, which remains mostly unchanged and applies to everything produced for the mass consumer market / consumer culture, Inflation makes it so that quality is increasingly getting harder to afford, on average. Bitstream was a cheaper alternative to the multibit, perfect for the mass market. That says it all for me really. Also, our ears are non linear transducers - The main reason I think vinyl sounds the way it does, over modern digital is because of the integral linear distortion factor inherant with vinyl players. Ears respond to modulation, they enjoy fluctuation and variation - all modern dacs today are designed to be as transparent as possible (and rightly so) and most are 'inherently linear and monotonic over thier entire range', so it's down to who ever composes the music to insure a varied sound. I personally think (analog domain) non linearities associated with vinyl and mulitbit is what makes them sound the less digital and 'warmer'. I might add, if I wanted to define what warm sounds like to someone a good example was the main menu soundtrack on gran tourismo 4. I don't know how they do it but that's what I picture warm as. just my 2c ...