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Audio System

About Me

Found 220 results

  1. Timbre definition from Wikipedia, “In music, timbre, also known as tone color, is the quality of a musical note or sound or tone that distinguishes different types of sound production, such as voices and musical instruments, string instruments, wind instruments, and percussion instruments. The physical characteristics of sound that determine the perception of timbre include spectrum and envelope. In psychoacoustics, timbre is also called tone quality and tone color.” I suggest reading the Wikipedia article as timbre contains both subjective and objective attributes, both of which are discussed in detail in this post. From a sound reproduction perspective, if ones goal is to reproduce music as faithfully as possible, then timbre (and all of its subjective and objective attributes) is a significant factor. I consider room acoustics the worst offender for destroying timbre (i.e. tone quality). If you are into the scientific research, there are a number of references in the Audio Engineering Society's library, here are a couple AES E-Library » Natural Timbre in Room Correction Systems (Part II) and AES E-Library » The Influence of the Room and of Loudspeaker Position on the Timbre of Reproduced Sound in Domestic Rooms Professional acousticians and audio engineers routinely take acoustic measurements as part of their everyday job. If you have been doing it full-time for a career, then you can read an acoustic measurement graph and hear the sound in your head. Same as how a musician can read notes off a music sheet and hum the tune (some with perfect pitch) in their head. While every acoustic space is unique, there are a couple of basic tenets that hold true for small room acoustics, which the majority of our listening rooms fall under this classification. These tenets are controlling room resonances and overall room decay times (i.e. RT60). This is based on a large body of knowledge specifically on small room acoustics. Here is a quick overview with a few reference links. Just like electronic engineers use circuit diagrams and part’s list (BOM) to communicate the designs and sonic signatures of audio amplifiers, acousticians use time, energy, and frequency information to communicate the sonic signature of an acoustic environment (i.e. both speakers and room). In this article, you will be able to correlate what you see with what you hear (literally) and vice versa. The Design Process I am going to analyze my listening room and come up with two designs to improve the timbre of the room acoustics, one passive, and the other active. Here are a couple of pics of my very live, untreated room. I try and make my analysis balanced between 50% what I hear and 50% what I measure. Based on that analysis, I will design and implement passive acoustic treatments. Then take another set of measurements and binaural recording of the speaker/room combo with the acoustic treatments in place. Next, Digital Room Correction (DRC). I will fine tune the frequency response using a “target” or “designed” frequency response to reproduce the best effort tonal balance and fine tune the impulse response, (i.e. timing) for the best possible timbre from the speaker/room combo. Meaning achieving the best possible tone quality (i.e. timbre) is limited by the physical dimensions of the listening room. Given that we can digitally manipulate all three dimensions of sound (amplitude, frequency, and time), we can create any sonic signature we want, with the limitation being the physical dimensions of the room itself. Technically, this is called a transfer function. A transfer function at this level encompasses everything that makes up the sonic signature of the speaker/room combo. Because of digital audio, we can design and implement our own transfer functions (i.e. sonic signatures) in software with distortion and noise levels far below what we can perceive and correction at a level of resolution far greater that our ears (read:brain) can discriminate. Historically it was thought that we could only discriminate to 1/3 of an octave (hence the 1/3 octave analog equalizer). Later research has determined that we can discriminate somewhat closer to 1/6 of an octave. So when viewing acoustical frequency response graphs, 1/6 octave smoothing is the preferred resolution to view the graphs as that is the most accurate representation of how our ear hears (or more technically correct, how the brain interprets the electrical signals). In the digital domain, we have digital filters that can have 65,535 “bands” (or more). Compared to a 31 band 1/3 octave analog equalizer... That's a revolution. I chose a linear phase filter (as opposed to minimum phase) as this produces the best phase coherence and time alignment. Not only is the sound “time aligned”, but some early reflections are reduced so that the phase coherence holds together long enough to hear the depth on the recording before the 3D image is destroyed by comb filtering effects of the room. Comb filtering is the root of all evil for an audiophile. Reducing early sound reflections, (and diffusing later reflections), is critical to the realistic reproduction of any stereo recording and achieving best possible timbre (i.e. tone quality). You want to hear enough of the recording long enough so that the phase coherence or sound stage is heard before the room takes over and interferes with comb filtering “location” cues that blurs the (depth of) image and colors the sound quality with the tone of the room. But first, this is what we need to listen for. It is a bit of science, hopefully presented in a fun and easy to hear manner as it is important to understand what is happening and especially what it sounds like. We all listen to it, but can we hear it? Pretty cool the Hass effect. “The Haas effect is a psychoacoustic effect, described in 1949 by Helmut Haas in his Ph.D. thesis. It is often equated with the underlying precedence effect (or law of the first wavefront).” “Haas found that humans localize sound sources in the direction of the first arriving sound despite the presence of a single reflection from a different direction. A single auditory event is perceived. A reflection arriving later than 1 millisecond after the direct sound increases the perceived level and spaciousness (more precisely the perceived width of the sound source). A single reflection arriving within 5 to 30 milliseconds can be up to 10 dB louder (My note: that’s twice as loud!) than the direct sound without being perceived as a secondary auditory event (echo). This time span varies with the reflection level. If the direct sound is coming from the same direction the listener is facing, the reflection's direction has no significant effect on the results. A reflection with attenuated higher frequencies expands the time span that echo suppression is active. Increased room reverberation time also expands the time span of echo suppression.” Key concept. It is amazing how a 5 millisecond delay can have that much width. The majority of rock and pop (and most mono multi-track) recordings use the Hass effect extensively, along with more digital delays, reverbs, stereo expanders, etc. If you listen to rock and pop, or any other mono recorded, multi-track recording, it is fake stereo. It's all an illusion and fools our brain every time (speaking as someone that spent over 10,000 hours in the recording/mixing chair doing exactly that). Personally, I don't care. When I crank up SRV's Tin Pan Alley (DR 15) on my rock and roll audiophile system and it feels like I am at Buddy Guy's Legends night club in Chicago, the illusion is complete for me. A bit more physics, as this is directly related to speaker location and listening position. Sound travels roughly 1 foot per 1 millisecond. The wavelength of a 20 KHz frequency is 0.68 of an inch. If my stereo's equilateral triangle is out even by an inch, I will already have destroyed some of the high frequency image (especially depth of field), because the equilateral triangle is misaligned and I am creating comb filtering at high frequencies. The learning from this is that time alignment of everything is critical, due to the Haas effect, and its role in reproducing proper timbre. The better aligned the equilateral triangle, the more phase coherent image can be reproduced, which is one of the key attributes of reproducing the most realistic timbre. Additionally, this is why early reflections need to be tamed, typically 15 dB below the direct signal, so we don’t get the Haas effect blurring the time alignment of the stereo image (especially depth). My design approach to modern room tuning techniques includes using passive acoustic treatment to minimize room resonances, early reflections, and over all room decay time (RT60). I also use state of the art DRC software to trim the frequency response for best effort tonal balance, time align the signal so that the waveform (all frequencies) arrives at the same time in the listening area, and minimize early reflections to enhance the depth and overall phase coherence of the stereo image before comb filtering destroys the recorded illusion. Acoustic Analysis and Design Fellow CA readers, I am the recipient of the 2nd worst possible sounding room award, only beaten by a room shaped like a cube. This is because the length of the room is almost twice the width. Additionally, my stereo is set up off center in the room. So how do I know it is the 2nd worst possible sounding room? I am using Bob Gold’s room mode calculator that will produce a nice graphic display of the room modes given the dimensions of my room. According to the calculator, my rooms Schroeder cutoff frequency is 92 Hz. This is my room’s fundamental transition frequency, below this frequency, the room behaves as a resonator, above, a diffuser/reflector. This transition point is far from smooth and resonates below the cutoff and rings (like a filter) above the cutoff. Just like blowing air across the mouth of a near empty coke bottle, every room resonates a tone that rides on top of all low frequency notes. Depending on how bad it is, like my room ratio for example, will produce what is sometimes called “one note” bass tone, meaning the rooms resonant frequencies are so dominant (i.e. too much amplitude) so all the bass notes (and sometimes drums) sound like just one note is playing. Also called “room boom”. You too can work out your rooms resonant frequencies using this calculator. Here is a frequency response measurement of my room to see if it correlates with the model. Many thanks to JohnM for his most excellent REW measurement software. This measurement correlates well with the model. Major peaks and valleys between 92 Hz to 300 Hz. That’s the ultimate challenge isn’t it, 2nd worst possible sounding room from an acoustic perspective. If I can make this room sound good… Note the blue horizontal line is mine to help delineate the problem areas. The circled mid-range area also represents a problem area. Initially looks like too much amplitude, but the real culprit for the raised amplitude is mid-range room reverb build up. We need to look at another view to see it. This brings up a story I feel is worth sharing so you can understand where I am coming from on this. As mentioned elsewhere on my blog, I had the good fortune to have been a live sound, recording/mixing engineer for 10 years. SQ was of major importance to me and I worked extra hours to ensure the artist/group got the best possible sound I could come up with. I worked in a several state of the art acoustic spaces, with this one below sounding so good that I gave up on my home system. The studio control room facilities I worked in were designed from the ground up acoustically to be state of the art. The rooms sounded incredible. Perfect neutral timbre. If you ever get a chance to visit a properly designed studio control room and listen to some music... I got so used to state of the art sound, that no matter what I did in my home stereo it paled in comparison to the sound of the state of the art control rooms. And I am not talking about the gear. The biggest difference between working in the control room and listening at home was the timbre (i.e. tone quality) of the rooms. The studio control room is designed so that the engineer sitting behind the console would hear the sound of the music picked up by the mic and room of the studio before the sound of the control room could be heard. Also known as a reflection free zone (RFZ). RFZ is control room design based on knowledge of the Haas effect. That meant obtaining a reflection free zone at the mix position and ensuring that any room timbre (i.e. tone quality and all of its subjective and objective attributes) was as neutral sounding as possible. I.e. no coke bottle resonance effects, no boxiness, etc. If you saw the blueprints for one of these control rooms, you would see no surface is parallel and are designed to ensure early reflections did not enter the RFZ and later reflections were thoroughly diffused so any room sound was perceived as a neutral sounding extension that made the room sound a bit bigger than it really was. A very neat psychoacoustic trick. As mentioned, the point was to hear the direct sound from the mic in the studio, plus the early reflections (i.e. tonal colorations) before you could hear the sound (i.e. timbre) of the control room. That way, when you were placing mics and eq'ing, you were not making decisions based on a hearing the tonal colorations of the control room, mixed in with the sound from the studio. When I compared the acoustics of my home listening space versus the state of the art control room I was working in +8 hours a day, the timbre gap was so great, I gave up on a traditional speaker setup at home. Mostly I listened to headphones. Sometimes, I invited the boys over to the studio when it wasn't busy and we would listen to tunes there. While looking at some programming sites, I came across a few Digital Signal Processing (DSP) articles. One of them was showing how you can use a well-known DSP technique, called convolution where you can digitally mix (i.e. real-time convolve) the “bit-perfect” music signal with a digital filter (both in the frequency and time domain) that was the inverse (well, they really are algorithms) of the measured room response. Convolution is a transfer function. JRiver MC has a state of the art convolution engine to host these designed digital filters. What can be done in software far exceeds what can be done in hardware and analog domain. Every modern consumer and pro A/D D/A is performing DSP on the audio signal with digital filters (in conjunction with analog filters) already. “The precision offered by Media Center's 64bit audio engine is billions of times greater than the best hardware can utilize. In other words, it is bit-perfect on all known hardware” A bit more searching and I found a few DRC software products that used this filter design for audio. One is called Audiolense. I downloaded the demo and ran it on my crappy Logitech G51 computer speakers. If it can make those sound good… As soon as I heard it, I knew that someone (Bernt!) had figured this out in the digital domain, which is a revolution compared to what we can do in hardware/analog audio. This is what I was waiting for. For me, it is a new ball game and gave me the opportunity to get back into listening to music the way I heard it in those acoustically (near) perfect rooms, or at least come a lot closer than ever before. Back to the passive acoustic filter design. The first thing I need are bass traps that have good absorption capabilities from 92 to 300 Hz. When I was in the pro audio industry, I used ASC Tube Traps (and RPG products) extensively with good success. Unfortunately, I don’t have budgets like that anymore, but I think I have found a reasonably priced bass trap that should do the job. It is a corner trap, and should go directly behind the speakers in the corners. Because of my room’s offset, the best I can do is directly behind the speakers in a sorta corner. The idea here is twofold; one is to dampen the low end sound coming off the back of the speaker cabinet so the refection off the wall and back to the listening position is minimized. This would correspond to about 4 or 5 milliseconds delay. Remember the Haas effect video on what 5 milliseconds delay sounded like? That’s roughly 5 feet of distance, and in this case, after the main sound wave arrives, a secondary wave arrives off the wall from behind the speakers and confuses my brain on location. In this case, destroys the image from front to back. Depth of field, due to early reflections (and comb filtering) is the first thing to go. It is the green circled portion in the graph below. With the bass traps in place, it should help dampen those resonances/ringing from 92 to 300 Hz, plus dampen the impact off the back of the speaker. This should result in a tighter (i.e. more transient) bass sound with minimal 5 millisecond later reflection so it does not blur the (depth of the) image. This is captured on the binaural recording. We can also measure this with an Energy Time Curve (ETC). Technically, we can measure the room’s early reflections with an ETC, typically from 0 to 40 or 50 milliseconds. That’s 40 to 50 feet of travel after the direct sound arrives at the listening position. That way we can inspect anywhere along the time curve and with the wavelength calculator, turn that into distance. This allows us to figure out where the early reflections are coming from and to either dampen or diffuse accordingly. Looking at the spikes on the graph and corresponding millisecond time reading, can be translated into feet using the wavelength calculator. Then measuring from the mic position to the point of reflection to identify where passive acoustic treatments should go. And it is mostly the same type of acoustical treatments, one to tame the room’s resonant/ringing frequencies with bass trapping in corners. Next is diffusion or absorption of the early reflections off the floor, ceiling, and side walls. Of course, the back wall and front wall (with the windows). The windows may benefit from heavy velour curtains. Ideally, the speakers would be mounted in soffits, like in recording studio control rooms, but it’s just my living room, so it’s a design tradeoff (ha ha). Pretty easy to correlate as one can take a tape measure, or string, or a laser distance measure, measuring from the mic, with a mirror to find the reflection points and correlate to the ETC by using the wavelength calculator. This is an ETC measurement of my untreated room. I can label the reflections based on translating to a physical measure in the room. As it stands it is not too bad as the rule of thumb is that all early reflections should be 15 dB or more down from the main signal amplitude. I am almost there. This is simply by virtue that my listening position is as far away from any reflecting surfaces as possible, given the constraints of my room. Check out this waterfall graph showing at which range of frequencies are producing the long decay times. This means my room is very lively as the carpet is indeed the only real absorbent material in a room that is otherwise all drywall, glass, tile, and hardwood (on top of being the 2nd worst room ratio). What you are seeing here is sound measured in 3 dimensions, vertical scale is level or amplitude in decibels, the horizontals scale is frequency in hertz and the z scale is time in milliseconds. In my case, the time scale is from 0 to 300 milliseconds, meaning the sound has travelled roughly 300 feet (10x the length of the room and 20x the width of the room) in the room when the microphone measured 300 milliseconds after the direct sound, so that we get the sound of the room and it’s decay and display in a visual 3D graph. I have circled the two problem areas. The one on the left is showing the room resonances with peaks and valleys, that I identified earlier. The one in the lower middle is showing the long midrange decays times, which build up more than other frequencies and caused me to incorrectly compensate by lowering the DRC "target" frequency response by -3 dB at 2 KHz. More on that later. Let’s look at shape of the decay over time. There are ITU, IEC, ISO, BBC, and other standards bodies specification of the reverb time (spec’d as RT60) or more properly, early decay time, for critical listening environments of a minimum volume of 2500 cubic feet. The specification or preferred range is from .4 to .6 seconds decay across the frequency band, with some rise in the bottom end allowed. That’s 400 to 600 milliseconds max. I am definitely over the .6 second mark in the midrange as circled in the graph (turns out to be .7 seconds). In this case, some broadband absorbers with good absorption in the midrange will be called for in this design. These should be mounted at the first reflection point on the ceiling and the rear wall to not only reduce early reflections, but further dampen the “brightness” and “boxiness” sound of the untreated room. If my room happened to be the opposite, i.e. dead sounding, then I would put diffuser panels on the ceiling and rear wall instead, with that .4 to .6 second decay as the target RT60. That’s the analysis of my room acoustics and some basic acoustic design, not only based on measurements, but extended hours listening for early reflections, room modes, and midrange comb filtering. My design is to dampen the back of that pounding 15” woofer and the room modes at the cutoff frequency and harmonic ringing. In addition, absorb broadband midrange due to bare walls, plus take care of the early reflections (floor has carpet, the ceiling gets the absorber) to get rid of that “boxy” sound. We will see if it is enough or not. As a last resort, I can hang heavy (velour) curtains over the front windows plus a good portion of the wall. Adding Passive Acoustical Room Treatments Every listening room has a fundamental resonant frequency (plus harmonics) that will need some taming. It is simply a function of the physical dimensions of the room. Depending on how “live” or “dead” sounding the room is will determine the number of diffusers and/or absorbers for any particular sound environment to achieve the recommended RT60 decay time. The ideal design is to have all sound at all frequencies decay at the same rate and meet halfway between the RT60 specification of .4 to .6 seconds. Every critical listening environment could benefit from this basic passive acoustic filter design pattern. A more encompassing design pattern looks may look like this: I have used this design pattern (and portions thereof) extensively and successfully when I was in the pro audio business Here is what I ended up installing in my room. 6 panels, 4 clipped onto the back wall and 2 on the ceiling to take care of the early reflections. 2 corner bass traps behind the speakers: Measurements Here are a few measurements to see how the passive acoustic treatments helped out the acoustics, even though I can hear the difference just standing in the room. These overlays are to compare before and after acoustic treatments. I have zoomed in the vertical scale to 2 dB per division to show detail, which exaggerates the "un-smoothness" of the frequency response. The acoustic treatments are able to significantly dampen the circled areas almost by 5 dB at 200 Hz and 3 dB through the midrange, which is reducing the room power by half. Said another way, the passive acoustic treatments reduce the room gain by half in the identified problem areas. That's significant. The early reflection in the 4 to 5 millisecond range has been reduced considerably as a result of the bass traps placed behind the speakers and reducing the reflection off the wall behind the speakers. This is key to the kick drum having definition and hearing all the bass notes from the bass guitar at equal loudness, both in the frequency and time domain. Compare the two 3D waterfall graphs above, the first one before treatments and the latter after. The mid-range decay times (the boxiness sound) have been reduced as circled. Also note, the 200 Hz peak and decay has also been reduced 5 dB. I was going to screen cast switching between the two graphs so you could get a real good sense of the passive acoustic filters at work as it is much more than just the circled points, the overall sound is further diffused. Because of the passive acoustic treatments, my room's RT60 is now within the .4 to .6 second specification across the frequency range. If I was to add any more absorbent material, I might add a couple more ceiling absorbers right over the listening position to reduce the comb filtering effects of the couch, or adding heavy velour drapes to the windows in the front of the room. Based on my listening tests, the bottom end and midrange are much tighter defined, as is the overall stereo 3D image. An overall improvement in frequency response smoothness, with tighter definition or imaging or timing. Sounds more focused. It is easier to hear the tone quality change towards the end. It seems I am right in there for the proper decay time. The sonic improvements that I hear line right up with what I measure and vice versa. So from a timbre perspective, I am pretty happy with the end result. Analysis and Design Part 2 After living with the acoustic treatments for a week and listening everyday, have made a major improvement in tone quality. Dampening the “one note” bass room mode and dampening the “boxiness” comb filtering in the mids. The decay time is within specification as evidenced by both the measurements and binaural recordings that you can hear the timbre (or tone quality) improvement yourself. What further improvements can I make to the speaker/room interface? How can I further improve the timbre? There seems to be more room to improve, especially given the frequency response still deviates quite a few (14) dB, when I should be in the +- 3 dB range across the frequency band. Even then, 1 dB either way is audible. How do I further smoothen the frequency response? Also, what about phase coherence and timing at the listening position? Can I improve that? I remember owning Thiel CS 3.6 time aligned speakers in the consumer world and when I was recording/mixing, I was using the Urei 813C time aligns. I can hear time alignment, and I can measure it. So how do I improve the time alignment (as my speakers don’t have that feature built-in – many don’t as it is hard to do - meaning expensive) plus how do I further minimize early reflections to get the best image possible at the listening position? Basically I need both frequency and time alignment capabilities. Just like every piece of audio gear has a sonic signature, the revolution that is digital audio, provides a facility to correct the sound in the digital domain at high resolution (64 bit data path) and low distortion. Given the computing power and sophisticated DSP software we have today, there is a classification of software that is called Digital Room Correction (DRC) software. Therefore, I can easily create any sonic signature I want since I have more control over the frequency and time domain than my ears can discriminate using software like this. With the software, you can use default digital filters, or using a Designer, create your own. This is designing the transfer function for the speaker/room combo. In this case, a linear phase FIR filter. Digital Room Correction How do we do this? We design the digital filter using a “target” frequency response, one that we design in software. If time domain correction is enabled, which it is in my case, then the impulse values change with target frequency response. The best impulse response can be achieved by matching the target's high frequency roll-off, with the natural roll-off of the tweeters filtered frequency response. For me, this tunes the filter to yield the best possible timbre for the speaker/room combo. When this is tuned properly, the timbre tunes in like a guitar string being brought into tune. I have guitars, mics, A/D converter, so I can compare live and recorded timbre of the guitars, plus shakers, tambourines, triangles, etc. Here is an example of a "designed target" frequency response using Audiolense. I draw or enter in the data points of the frequency response curve I want (red dots). Once I treated my room acoustically, I no longer needed to drop the target frequency response by - 3 dB at 2 KHz. That was a learning for me. Actually, a re-learning for me as I remember reading this in Don Davis excellent book on Sound System Engineering "You can't effectively (digital) eq a reverberant field". Here is another view of the target plus the uncorrected frequency response of my speakers in the main form view of Audiolense. Note how the targets frequency extremes match the speakers natural roll-offs at the extremes. I snuck in a little bottom end lift on the target, but given the Klipsch QB3 alignment of the ported bass bin, I can tuck in little more low end and still have the bass sound tight and not over tax the amplifiers. Now I can have Audiolense generate the digital FIR filter (which is almost an inverse of the uncorrected response, I say almost because there are other algorithms at play here): Here is the resultant corrected frequency response: The uncorrected frequency response is on top and the corrected frequency response is on the bottom (along with the target). In addition to the acoustical treatments, and short of building a state of the art critical listening room from the ground up, I know of no other way to achieve this level of timbre correction, given my awful room ratio. Let’s look at a few measurements. Frequency response. I have zoomed way in on the amplitude scale again to show detail. The DRC is able to correct for a 14 dB swing and reduce it to +- 2 dB deviation. The spectral response is similar to preferred spectral responses as described in B&K's paper (Figure 5) and Dr. Sean Olive's paper (slide 24). ETC looks to be in spec as almost all early reflections are – 15 dB below the main signal arrival. The early reflection of around 2 milliseconds is the first reflection off the floor to the listening position. Other than mounting the speakers in soffits, not much can be done there. The good news is on how diffuse the later reflections are. Which means the room adds little tone color to the reproduced music through the speaker/room combo. The blue waterfall graph is as good as it is going to get given my room ratio. I can play with the decay time of the 50 to 60 Hz wave by adjusting a parameter in the time domain window in Audiolense’s Correction Procedure Designer as a next step to tune this back a bit, but I don’t notice it too much in the sound. Conclusion I had a lot of fun doing this. The timbre changes between the untreated room, treated room, and with DRC, are definitely audible. If achieving the best possible timbre from your audiophile system is of interest to you, then this article and my previous article, Speaker to Room Calibration Walkthrough, may be of some use. Happy listening!
  2. Note: an updated version of this text is my main audio site (audioroot.net). Why the heck would anyone buy a DDDAC1794? This thing seems very much out of place in the arena USB and FireWire DACs out there, it costs a lot of money, and it requires an intimate relationshop with a soldering iron, too. However, I have been in the DIY world long enough to know that nothing beats a good DIY system. I had many and very different DACs in the past. My beloved Stokes DIY Tube DAC was restricted to S/PDIF and red-book 44/16 audio. I plunged into computer audio with a not-so-great Headroom USB DAC. Then I hot rodded an Apogee Mini FireWire DAC with a hefty DIY power supply. I use an Audiolab M-DAC in our living-room system (and it tends to break from time to time). My main system had a Linnenberg UDC1. And I've listened to the Weiss and many other cost-a-lot stuff. However, while some of those DACs sound pretty good compared to others, they all screw up the music in the same way. And I don't mean the painful "S" sounds and similar boorishness from crummy DACs. Even the «good» DACs take away the life, flesh and breath from the music, very much in contrast my trusted vinyl rig (Scheu platter and bearing, Teres motor, and Schröder arm). The Stokes Tube DAC and the Audiolab M-DAC both allow using different built-in digital filters with different characteristics. The different filters usually sound slightly different, and some sound «better» than others, but they never get rid of the artificial sound completely. But, maybe unfortunately, the filters cannot be turned off completely. When I «stumbled» over Doede Douma's description of his DDDAC1794 that does away with digital filters I knew I had to try a non-oversampling (NOS) DAC sooner or later. Why not just skip the digital oversampling/filter, if it affects the sound by inventing new sound data that never existed in the first place? Doedes technical description and documentation is very comprehensive and makes a lot of sense. My only hesitation was that I didn't want to start yet another DIY project that I'd never finish, because time is limited (there's a family, work, and too many other hobbies). But Doede sells completely assembled and tested DACs modules, power supplies, and USB interfaces. He even gave me a copy of the files needed to order a very nice custom-made chassis for the DDDAC1794 at Schaeffer AG. And when I asked him about the specifics of the additional bits and pieces needed to build a complete DAC, he simply included these in the package. For example, when I asked about which power switch would fit in the chassis, Doede just put the switch in the package (three switches, to be precise. Just in case I'd break the first and loose the second one). All this allowed me to build the DDDAC1794 in no time. The only gripe was when the Schaeffer chassis was a wee to too small to fit the power-supply heat sinks, but there was an easy fix (just a little side note to illustrate how smooth building the DDDAC1794 was: before I found the DDDAC1794 on the net, I asked the local Bryston distributor if I could borrow one of their DACs to give it a try. They keep promising I could have one once they receive one. In the meantime, I am playing my music using the DDDAC1794). How does it sound? Spectacular? Phantastic? Superb? Damn good? Fucking great? Yes, all of this. But that's all secondary with the DDDAC1794. The really important thing is that the DDDAC1794 doesen't sound like a DAC at all. It's a bit like a vinyl rig on steroids, but without the pops, clicks, and rumble (and I don't mean the old record player your dad had when he was a boy, but the freaky good 2013 stuff). The music and all the little details are just there in a very relaxed way. Ry Cooder is having a party in my house, Phil Collins' (yes!) drumsticks are flying in front of me, Sophie Hunger has moved to my house (was close anyway), Willy DeVille has risen from the dead, no more doubts about No Doubt, Timber Timbre is timber timbered, Mark Knopfler is in Dire Straits, Marianne Faithfull finally confessed her love for Bruce Springsteen, Jeff's Wine is as Lilac as it gets, Glen Hansard got a shave, Depeche Mode are Exit(er)ing, Lou Reed made me a Perfect Day, and Giant Sand and Marla Glen just called to be the next acts in my listening room. In short: I hear the music, not a DAC. In contrast to oversampling and digial filters, the NOS concept not only works, but also sounds good! As a final and very important comment, I'd like to congratulate Doede not only for designing the DDDAC1794, but also for documenting everything in full detail. The deep insight into how the DDDAC1794 works provided a lot of confidence that convinced me to try Doede's design and to buy his stuff. One can only guess why others don't do that. Update 11.3.2013: Doede sent me two Sowter 1298 transformers, which he designed as an alternative to the standard coupling capacitor in the analog out line. Apart from avoiding the coupling capacitor in the signal, the transformers also allow using the inverted output of the DAC chips, thus cancelling out even-order distortion. I immediately noticed the sound improvement with the transformers. The music sounded as if the musicians just got a pay increase! The transformers were expensive, but the money was well spent in my case.
  3. mitchco

    CA Articles

    This blog entry points to the list of articles I wrote for CA. I hope you enjoy them! https://www.computeraudiophile.com/profile/8172-mitchco/?do=content&type=cms_records2&change_section=1
  4. Edge.org, the website of the science and technology think thank Edge Foundation, has a great collection of answers from various experts to their "Edge Question 2011", "What Scientific Concept Would Improve Everybody's Cognitive Toolkit?" Among them is this great contribution from Richard Dawkins that I think applies extremely well to the audiophile world: The Double-Blind Control Experiment Not all concepts wielded by professional scientists would improve everybody's cognitive toolkit. We are here not looking for tools with which research scientists might benefit their science. We are looking for tools to help non-scientists understand science better, and equip them to make better judgments throughout their lives. Why do half of all Americans believe in ghosts, three quarters believe in angels, a third believe in astrology, three quarters believe in Hell? Why do a quarter of all Americans and believe that the President of the United States was born outside the country and is therefore ineligible to be President? Why do more than 40 percent of Americans think the universe began after the domestication of the dog? Let's not give the defeatist answer and blame it all on stupidity. That's probably part of the story, but let's be optimistic and concentrate on something remediable: lack of training in how to think critically, and how to discount personal opinion, prejudice and anecdote, in favour of evidence. I believe that the double-blind control experiment does double duty. It is more than just an excellent research tool. It also has educational, didactic value in teaching people how to think critically. My thesis is that you needn't actually do double-blind control experiments in order to experience an improvement in your cognitive toolkit. You only need to understand the principle, grasp why it is necessary, and revel in its elegance. If all schools taught their pupils how to do a double-blind control experiment, our cognitive toolkits would be improved in the following ways: 1. We would learn not to generalise from anecdotes. 2. We would learn how to assess the likelihood that an apparently important effect might have happened by chance alone. 3. We would learn how extremely difficult it is to eliminate subjective bias, and that subjective bias does not imply dishonesty or venality of any kind. This lesson goes deeper. It has the salutary effect of undermining respect for authority, and respect for personal opinion. 4. We would learn not to be seduced by homeopaths and other quacks and charlatans, who would consequently be put out of business. 5. We would learn critical and sceptical habits of thought more generally, which not only would improve our cognitive toolkit but might save the world.
  5. Digital Future of Audio This appeared in the thread: In a perfect world, where would the D to A conversion occur? The topic here is where is audio headed, for normal consumers and for audiophiles. I put his hypothesis to you: Audio will be digital all the way through. I challenge you to a battle of reason and arguments, I give you: The challengers: Mr. A Mr. A grew up under difficult curcumstaces; knowledge was scares and life was hard. Little by little he was nursed to life and he spread out through the world. Mr. A prospered and branched out, grew stronger and had a good life. His first tools were tube amplifiers and he did well and had no competitors to speak of. Mr. A used class amplification, that was what his tools were good at. Along came a new different tool, the transistor. In the beginning the transistor was small and not much of a threat, but then it grew stronger and cheaper. Mr. A's cousin Mr. AB liked the small size and cost of the transistor and started using the new tool. Mr. AB's constructions with transistors grew stronger and suddenly Mr. A was getting beaten by his cousin. Mr. A fought back for a while, but in the end he called in his troops and the withdrew to a few high grounds where they could still defend them selves with their tubes. Mr. AB Mr. AB was the man of the hour! He expanded out in areas where Mr. A had newer even thought of going. The transistors evolved into mosfet's and much later into super powerful IGBT's and they had a blast. Everybody needed them and the world lay at their feet. Mr. AB's cousin Mr. C liked the small size and cost of the transistor and said: I can build a computer with then. My earlier venture with tubes were a disaster, and relays were no better. Mr. C Mr. C's smart constructions with transistors grew slowly stronger. Suddenly Mr. C found that he could put many transistors in one chip known as an IC. Mr. C builds a computer that can count and calculate and all sorts of things. That changed the game - big time! Mr. C is the man of the hour! He expands out in areas where Mr. AB had newer even thought of going. Mr. C has found blue waters and created needs that no one before him had dreamed about. Mr. A and Mr. AB still had a little friendly competition, but Mr. AB was clearly the younger, more agile and prospering of the two. Mr. D Mr. C's son Mr. D comes along. He is a scholar, a philosopher, a leader and a patron of the arts. He's grown up with the the many possibilities created by everybody that came before him. Mr. D has no tools but his knowledge and his visions. In his view, everybody else is wasting their efforts alone in each their camp. He's vision is to form a conglomerate that combines his leadership with all the best tools in the world. That way every too can come to it's right an be used in the most efficient way. So Mr. D starts programming PLC's and industrial computers that constantly measures and optimizes machines and factories and businesses to be more efficient and competitive. That changed the game - big time! Mr. D is the man of the hour! He expands out in areas where Mr. AB had newer even thought of going. Mr. D has found blue waters and created oportunities that no one before him had dreamed about. The Arts So Mr. D comes home form a long day at work, too late to go out for one of his much loved concerts. He just wants to kick back with a glass of wine and some good music. He has audio systems on his shelves build by those that came before him, and while they are all fine, they all seem quite antiquated compared to his new increasingly efficient and predictable businesses. He grabs a beer to cool down while listening to his favorite music in the red hot glow of Mr. A's tube amplifier sitting on the shelf along with all the other gear. He thinks: Mr. A - built a thug The pioneer that built the class A tube amp. The amp is a thug with little sophistication. Big and wasteful like a SUV with a V12 engine. Mr. A did try with a class A solid state, but it is still expensive an a furnace int the room. Mr. AB - built a bully The contender that built the class AB solid state amp. Now that is a little more like it, if only the push - pull action is well adjusted and mosfeds operate linear and no clipping and no thermal glide and . . . It still gets things done by insisting, so no sophistication in communicating with the outside. Mr. C - built the technical boxer Now Mr. C built the switch mode amp saving space and less heat. His pile of boxes and interconnects is however just as high as everybody else. And he is still no more sophisticated in his communication with the speakers. He is just a tad better at insisting and gets off the mark faster. Mr. D - What will it be? Now Mr. D is thinking: how far can I take it? - Sensors - feedback - system under control - adaptive system - designed frequency response (eq) - designed derivatives: d(frequency response) / dt - designed voicing: XX-tube, legacy amp/speaker system - designed venue: Albert Hall, Opera in Sidney, wooden church - designed venue adapted to the music What do you think? Find my blog: “Confessions of a DigiPhile” at http://www.computeraudiophile.com/blogs/7638-DigiPete
  6. Just wanted to share various sources around on the net on the subject of acoustics and digital room equalization / correction (Digital Room-EQ ) that I have run into so far.... Disclaimers: • I am sure I touched only a small percentage of what is around, so please feel free to comment & add as you see fit. • I have not checked, read, verified or tested (let alone used) everything. Don’t shoot the messenger. • I have no interest other than share info... --------------- Papers / background / calculators / general Room mode theory crash course: Acoustics Crash Course 1 - Modes Room mode calculators: Room Mode / Standing Wave Calculator hunecke.de | Calculators Bob Golds Acoustics Frontiers paper (must read!): http://blog.acousticfrontiers.com/storage/acoustic_measurement_standards.pdf DRC Wiki: DRC HTS: Equalization | Calibration at Home Theater Forum and Systems - HomeTheaterShack.com --------------- Measurement software: DRC: DRC: Digital Room Correction REW: REW - Room EQ Wizard Home Page Audiolense: Juice HiFi ARC: ARC System 2 Acourate: AudioVero TrueRTA: True Audio: Audio Spectrum Analyzer and Loudspeaker Design Software FuzzMeasure: FuzzMeasure Pro 3 Ultimate Equalizer: Bodzio Software Various/other/overviews: Room Acoustics Measurement Tools For PC and Mac | Record, Mix & Master The Well-Tempered Computer --------------- Affordable measurement kit: Mic: Behringer: MEASUREMENT CONDENSER MICROPHONE ECM8000 (get it calibrated) --------------- Measurement kit + software / package solutions: Dayton Audio: Dayton Audio Dayton Audio OmniMic V2 Acoustic Frontiers: XTZ Room Analyzer II Standard Acoustic Measurement Software Package | Acoustic Frontiers --------------- Convolution engines (Convolvers; need to run ‘in’ or ‘under’ your player as a plug-in): Convolver (windows) : Convolver — a convolution plug-in JRIVER (windows): Native JRiver 64bit fp convolution engine for Room Correction (FIR filters)? BruteFIR (Linux): BruteFIR LAConvolver(Mac): Lernvall Audio Fink - Package Database - Package jconv (Convolution Engine for JACK) Mellowmuse IR1A - Zero Latency convolver RTAS, Audio Unit and VST plugin for OSX and Windows --------------- Parametric EQ (no time domain, just frequency/amplitude): apulSoft apQualizr : apulSoft - apQualizr (see comment Bob Stern) FabFilter : FabFilter - Quality Audio Plug-Ins for Mixing, Mastering and Recording - VST VST3 AU AAX RTAS AudioSuite (see comment Bob Stern) work in progress.... any comments welcome..
  7. CDal

    Nad M51

    I would need help since I am getting all kind of different advices. I have an iMac in the basement ( about 60' of cable length from my HiFi units) I would like to know what is the best and cheapest way to connect my iMac to a Nad M51 Direct Digital Dac. I would like to be able to play 24 bits/44.1, 48, 88.2, 96, and maybe 176.4, and 192 KHz. Thank you.
  8. how does one "load" 600+ into this system? i ripped these 600+ cds onto my pc harddrive 4+years ago using windows media center and player(windows media lossless codex) and onto mediamonkey(in flac) both my pc and my maxtor one touch external backup hard drive failed!! i have no desire or patience to do do that again,EVER! so, if i purchase this sooloos system-media core 200 who do i get to rip and load the cds on the harddrive and how much does that cost? thanks bobbmd/[email protected]
  9. I'm new into computer audiophile and almost new to everything music wireless. I have several HD (USB & DLNA) and an M-Audio Audiophile Pro, which I connect to a micro Hi-Fi Stereo though Aux (RCA), I'm looking to move forward to a new level. Where can I find an article with equipment recommendations and installation guides? I'm thinking about getting an Internet Radio (Cocktail Audio X10) with 2TB of HD, what do you think? What speakers should be decent for this? Thanks to all, Jeannette
  10. Starting Point A commonly stated "scientific" reason for alledged audible differences between (speaker) cables is the fact that electrical signals travel at different speeds for different frequencies. Background The characteristics of a cable are often simplified to its impedance (Z), its capicatance ©, and its inductance (L) where Z = sqrt(L/C). This is a simplification of the more complete formula Z = sqrt((R+j*2*pi*f*L)/(G+j*2*pi*f*C)) where R is the resistance and G the conductance. For high frequencies, i.e., for very large f, R and G are much smaller than 2*pi*f*L and 2*pi*f*C, respectively. Thus, the formula simplifies to Z= sqrt((j*2*pi*f*L)/(j*2*pi*f*C)) = sqrt(L/C). Unfortunately, audio frequencies are not very large by any measure (which is why skin effects and similar high frequency effects do not effectively matter for audio cables). Thus, in the range of 20 Hz to 20 KHz, the signal propagates at greatly varying speeds from 5 million m/s to aprox. 125 million m/s. That's a factor of 25 in speed! Luckily, audio cables for home use are very short. Let us consider a typical run of 10 metres. A signal of 20 Hz needs 10 m / (5*10^6 m/s) = 2*10^-6 s for this distance, i.e., 2 microseconds. A signal of 20 KHz needs 10 m / (1.25*10^8 m/s) = 0.08*10^-6 s for this distance, i.e., 0.08 microseconds. That is, the time difference for these two extreme audible frequences is 2-0.08 = 1.92 microseconds. A further factor is the dielectric constant of the insulation, which determines slowdown of the propagation (the so-called velocity factor). For typical insulations used, the velocity factor is around 0.66. Thus, the difference will be boosted to 2.86 microseconds. Conclusion Even for extreme audio frequencies (20 Hz vs 20.000 Hz), the "time smearing" is below the audible range (>= 5-10 microseconds). Any reasonably constructed (speaker) cable will not exhibit audible problems at runs up to 35 feet. References: Wikipedia: Transmission Line, available from http://en.wikipedia.org/wiki/Transmission_line Jim Brown: Transmission Lines at Audio Frequencies, and a Bit of History, available from http://audiosystemsgroup.com/TransLines-LowFreq.pdf Wikipedia: Speakerwire, available from http://en.wikipedia.org/wiki/Speaker_wire
  11. Am new here. Have been building a system with an Asus computer using J River media player into an Asus Xonar ST sound card. This feeds with an optical cable a Marantz sr7005 driving Klipsh speakers with a 7.1 configuration. My question is the weak point of my system may be the on board cd player that came with the computer. The system sounds great now (play largely classical and jazz) but I syspect an better cd playerthan the one that came with the computer may improve performance? Any comments would be appreciated. Jim
  12. Hello Computer Audiophile, sorry if my english is bad guys. Well here's the problem I'm facing right now, both my PC and Sony Vaoi laptop are Realtek HD Sound Card, my laptop have this HDMI port, sometimes I use the HDMI as a DAC, well it seems that I can't afford any DAC at the moment, well my HDMI audio device only output up to 16/48 only, doesn't reach to 24/192 as any HDMI said that it can output to 24/192 that far. Well, back to the problem, I'm using JRiver Media Center 17, foobar2000, Aimp3 and Winamp. Right now I've read CA's FAQs and I just read about Windows internal DACs is mostly will output worst audio signal, and upsampling downsampling problem. I've one set my Windows Sound to 24/96kHz from the Control Panel and just play 16/44.1 and 16/48 file without thinking bit transparency of an audio. Right now I'm setting back to 16/44.1kHz to play 16bit 44.1 audio. Can this be fixed? I don't own any USB DACs but really need one, the problem is that money is the 1st thing to buy these best USB DACs, maybe affordable USB DACs will be my choice around $200 below. Well for now, how can I get BitPerfect from these players itself, do I have to change the Output Format from the control panel when everytime I wanted to play the right sample and bit depth for a music. What is the best choice, WASAPI or ASIO? I noticed that when everytime when I play a song that is 16/44.1 from JRiverMC 17 using ASIO I get this output playing my audio in 32bit/44.1? But when using WASAPI I get the right output for the music, I'm very happy for that it always plays the music at the right output. I have 24/96 Dream Theater A Turn Of Dramatic Events, I love this band and their music is so good. Ok now, when I wanted to play these 24/96kHz do I have to go back to the control panel and change the output format back to 24bit, 96000Hz (Studio Quality) setting? I'm very confused, does every USB DACs will solved all these problem? I don't need to change anything when play a 24/96 or 16/44.1 when I owned a USB DAC? Thanks Guys and sorry again for a bad english!
  13. As some posters have doubted the benefit of resolutions and sample rates higher than 16 bit and 44.1 kHz, I set up a listening test. Robert von Bahr of BIS has very kindly arranged for the permission to use a track from BIS-SACD-1949, "endBeginning" by New York Polyphony (available from eClassical). I made available 8 versions of the track "Lamentationes Jeremiae - IV. Mem.". In addition to the original 24/96 track, there are versions that have been first downsampled and/or truncated to various permutations of 96, 48 and 44.1 kHz and 24 and 16 bits, and then upsampled back to original resolution, thus discarding varying amounts of information. I asked forum members to listen to the different tracks, and for each, write down a score between 1 and 10 for the sound quality, as well as a verbal assessment of each track, and send me the results in a private message (or by email). Along with the tracks the download page also contained a link to a document containing a list of what track is what resolution. I encrypted the file with gpg in order to provide the digital equivalent of a "closed envelope", and I will post the password to decrypt the file as soon as the test period is over. There is no "trick" here - this is a serious test, and there are differences between all the files in terms of resolution or processing. In all cases but one, the differences are measurable and ought to be audible. I also made it clear that this is a *listening* test, not a measuring test. Yes, it would be interesting to see if it is possible to determine the resolutions of the tracks by measurements, but that can be done after the listening test is completed. After user "goldsdad" sent me results that he got by measuring that showed that it was relatively easy to cheat by using analysis software, I decided to stop collecting "official" results, but still wait with revealing the content of the files, so that people who wanted to do the test for themselves could still do that - thus I will publish the file contents, and the results I had already collected, after end of day, Monday, March 19. Many thanks to Robert, The New York Polyphony, BIS and eClassical for making this possible!
  14. Updated with more info on Audio DiffMaker, plus ABX listening tests. Lots of discussion around this article: 24/192 Music Downloads...and why they make no sense http://people.xiph.org/~xiphmont/demo/neil-young.html I decided to run a science experiment using Audio DiffMaker to compare 16/44 to 24/192 format of the same master from Soundkeeper Recordings: http://soundkeeperrecordings.com/format.htm I have used Audio DiffMaker before to compare FLAC vs WAV and comparing two bit-perfect music players on my computer audio playback system. Here is the result of my 16/44 vs 24/192 experiment. First a refresher on how Audio DiffMaker works: There are also a two papers, http://www.libinst.com/AES%20Audio%20Differencing%20Paper.pdf and http://www.libinst.com/Detecting%20Differences%20(slides).pdf The help file that comes with the program is very well documented and goes into much more detail. Updated - I wanted to provide more with respect to how Audio DiffMaker works and why it is an important state of the art measurement tool in any Audiophiles arsenal. Audio DiffMaker’s Differencing Process Excerpt from the DiffMaker Help file on how the differencing process works: While it may not be possible to show whether alteration is having effects directly on the listener, it is possible to determine whether an audio signal has been changed. The existence of any changes to a digital recording of an audio signal can be detected by the simple process of subtraction, performed on a sample-by-sample basis. If each audio sample is the same, then subtracting one from the other leaves nothing (zero signal). A recorded copy of the original signal (called the "Reference") can be mathematically subtracted from a recorded copy of the possibly changed signal (called the "Compared" signal). This results in a "Difference" signal recording that can be evaluated by ear or other analysis. If the resulting Difference signal, when played as audio, is effectively silence or at least is not perceivable to a listener when played at levels in which it would occur when it was part of the "Compared" signal, then the investigator can with good confidence conclude that the change has made no audible difference. The problems and operational, perceptual, or psychological complications about listening for whether sound is being changed are greatly reduced by transforming the task into the much simpler issue of listening for anything significant at all. The evaluation of the result is done by ear, and the user doesn't need to question hearing ability to use the tool. Audio DiffMaker test, encourages you to still "trust your ears". Audio DiffMaker is a state of the art differencing tool that automates this workflow from 5 years ago: http://forum.audacityteam.org/viewtopic.php?f=28&t=3873#p15071 One of the reasons it is state of the art is because the software can differentiate time differences in decimal places in the parts per million (ppm): “The sample rates or speeds of player decks and soundcards are constantly drifting, if only by very small amounts. But even as little a change in sample rate as 0.01ppm (one hundredth of a part per million) can cause two otherwise identical files to leave difference sound after subtracting.” In order to compare the two formats, I had to up sample the 16/44 to 24/192. I used http://www.voxengo.com/product/r8brainpro/ to perform the sample rate conversion: I used the default settings. Then I used Audacity to edit the waveforms so I am just looking at the first 40 seconds of each waveform. Then it is a matter of loading the two waveforms into Audio DiffMaker and extracting the difference. According to DiffMaker, the difference file is -94 dB. I opened up the difference file in Audacity and here is what is left over: Something definitely there. Here is the frequency analysis: I have also included the difference file as an attachment to this post. Given that the majority of content is 20KHz and above, I can’t hear anything on the difference file. Note that this is one data point. I have used Audio DiffMaker for a while now and here is one tip that will help you get consistent results if you decide to try it out. This is the output status window from the DiffMaker progam as it is running. Note the arrow. It says that the sample rate error is low enough not to require adjustment. If the sample rate error is too high, there will be a notification as such on this line, then the program tries to automatically align the tracks. However, there seems to be a bug in the program, as noted in one of my other posts, so the track alignment does not seem to work or work very well. Therefore, I am unable to get consistent results. If you look in the status window and see that your comparison requires sample rate adjustment, then here is what you can do. Open up the waveforms in your favorite digital audio editor and ensure that the both waveforms “start” at exactly the same time. That’s the trick. This is why I sample the first 40 seconds of the waveform, because in most cases, you do not need to line the waveforms up. Such is the case with the Soundkeeper filesas they both start at exactly the same time. If you do need to line the waveforms up because you are recording the samples, then you can trim them later in your favorite digital audio editor. It is tedious as it may take a couple of passes before you get it lined up exactly. Edited to add this section. I ran another DiffMaker test, this time on Kote Moun Yo? samples from Equinox. I really enjoyed this recording as it definitely has ultrasonic information recorded (i.e. percussion instruments) and is crystal clear sound with very low noise floor. I would say state of the art recording. Great job Barry! http://soundkeeperrecordings.com/format.htm I followed the same process as above. Again, the point in this is to either confirm or deny Monty’s claim that 16/44 is already better than our ears can hear and our sound system can reproduce. 24/192 should contain much more audio information than 16/44, so by comparing 16/44 to 24/192 using DiffMaker will show exactly how much difference there is between the two. In order for me to digitally compare the 16/44 to 24/192, I up-sampled the 16/44 to 24/192. If the R8 Brain resampler I used is doing its job proper, there should be no waveform changes as there is no information being added (or lost!), simply a (lossless) file format change. Here is what Audio DiffMaker reports as being the difference. -100dB difference file. It is very similar to my first test above, showing I can repeat the results, even on a completely different song/master. Here is what the Difference waveform looks like. And frequency analysis. As you can see, the frequency plot shows ultrasonic energy, even though it is very low in overall level. Again, I have attached the difference file so you can listen to it. I cannot hear the ultrasonic information. Part 2 Listening Tests Given that the difference between 16/44 versus 24/192 is ultrasonic energy, it is important to verify that the gear used can actually reproduce ultrasonic energy. I used my Lynx L22 pro sound card that has a ruler flat frequency response out to at least 50KHz: http://i1217.photobucket.com/albums/dd381/mitchatola/lynxl22-1.jpg I used my Sennheiser headphones with a custom Class A headphone amp that I built from the Audio Amateur from years gone by: On the right is a toroid transformer feeding a regulated power supply and then my perf boards of the amp itself on the far left. I have measured the frequency response out to +200HKz. The headphone amp has enough clean power that you can place the headphones on the floor opened and crank it up like it was a boom box. Next step is to verify that my gear can play ultrasonic information properly. These intermodulation test files provided by Monty’s article should be played first on your system to ensure you hear nothing at all. If you do hear tones, pops or clicks, that means the system under test is producing intermodulation distortion. http://people.xiph.org/~xiphmont/demo/neil-young.html#toc_intermod With my particular computer system, Lynx L22 and Class A headphone amp, I did not hear any tones, clicks or pops. Ok onto step 2. ABX testing. For listening tests that provide any level of statistical probability, double blind is the only way to go. I used Foobar2000 http://www.foobar2000.org/ and the ABX plugin http://www.foobar2000.org/components/view/foo_abx I made sure that I clicked on the Hide Results checkbox before I started the tests. First up, 16/44 vs 24/192. Here was the problem with this test. I could just tell by a very small delay when my DAC was switching from 16/44 to 24/192. So I was able to “game” the test: So I resampled the 16/44 to 24/192 so I could not hear the DAC switch sample rates. Here are the results: Obviously I cannot hear the difference. This correlates with the DiffMaker results as well. The difference is so small that I was guessing, even though I was trying not to. Since I cannot (significantly) measure or hear the difference between 16/44 and 24/192, I tried one more experiment where there is a known difference – MP3. I took the 16/44 and converted it using the best MP3 codec (LAME) and encoded at 192Kbps bit rate. I used this bit rate as I listen to a lot of music on Zune and this is the default bit-rate when I download the music onto my disk for playing. As you may imagine, there is a reason that Microsoft chose this bit-rate and I will show why shortly. Now comparing the 16/44 to the MP3 version produces the following Difference file in Audio DiffMaker: And if I open up the waveform in Audacity: Frequency Analysis: I have included the Difference file again so you can hear the results. And it correlates very well with the other two other MP3 difference tests I performed here: http://www.computeraudiophile.com/blogs/FLAC-vs-WAV-Part-2-Final-Results#comment-131768 So the $64 million dollar question is, can I hear the difference in an ABX test for 16/44 and MP3? While I did better than the 16/44 vs 24/192, it is in the territory of guessing :-) Listening closely, I thought I could hear a loss of transients on the percussion, but just barely perceptible to my ears. Another way I can listen is to use Audio Diffmaker where I can reconstruct the comparison track by adding the difference back to the reference. By incrementally increasing the difference track level, I can easily hear the difference when the difference track is boosted by about +6dB. I would hazard a guess this is the reason why Microsoft (and others) choose 192Kbps with MP3 as it gives the best fidelity versus file size. And likely the reason why most people don’t complain about it as most people (including me) cannot hear a quality difference, even under ABX testing conditions. Conclusion Well, for me, my ears, on my equipment, my test and listening results confirms Monty’s article that 16/44 is enough for my ears. This is also qualified by the science and engineering in the Digital Audio field: http://www.computeraudiophile.com/blogs/1644-vs-24192-Experiment#comment-135987 In fact, it may be that even high bitrate MP3’s is enough resolution, but that’s another debate. Full disclosure, I am 53 years old and given the hearing loss versus age http://www.roger-russell.com/hearing/hearing.htm in the chart below, I may not be the best candidate for trying to hear ultrasonic audio information :-) A quick hearing test from: http://www.phys.unsw.edu.au/jw/hearing.html confirms that I can hear to at least 12KHz, but down at 16KHz. It is no suprise to me why I don't hear ultrasonic audio information: My perspective is this. If I was going to pick one cause to get behind in the world of music, it would not be over high resolution file formats. It would be the Loudness War. Almost 30 years ago, the pop band, The Police, created a very popular album called Synchronicity: http://en.wikipedia.org/wiki/Synchronicity_(The_Police_album) With an overall Dynamic Range of 15 http://dr.loudness-war.info/details.php?id=12040 and the final cut on the album, Murder By Numbers, with a DR of 18 is an excellent example of taking the full advantage of the Red Book standard. The disc sounds fantastic. What happened since then? The Loudness War in less than 2 minutes: [video=youtube_share;3Gmex_4hreQ] Given that this is CA, I would think everyone could correlate what they see in the visual representation of the waveform and what they hear. As I have discussed before, there is a direct correlation to what is measured with what is heard – it’s fundamental to the princples of audio. You can see and hear the difference, even over YouTube! Final thoughts: All of the software used to perform both measurements and listening tests is free. Therefore, if you are curious and want to verify or deny Monty’s (and as it turns out, me too) claim, you can perform the same tests yourself. Happy listening!<p><a href="/monthly_2012_05/58cd9bc1280a9_16441vs24192difference_zip.4e172f8cd059cc25e231d1cdde27b118" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28078" src="/monthly_2012_05/58cd9bc1280a9_16441vs24192difference_zip.4e172f8cd059cc25e231d1cdde27b118" class="ipsImage ipsImage_thumbnailed" alt=""></a></p><p><a href="/monthly_2012_05/58cd9bc12d67b_SR002KoteMounYo16441vs24192Difference_zip.0639a2fcdb04739c72b1b2340c337153" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28079" src="/monthly_2012_05/58cd9bc12d67b_SR002KoteMounYo16441vs24192Difference_zip.0639a2fcdb04739c72b1b2340c337153" class="ipsImage ipsImage_thumbnailed" alt=""></a></p><p><a href="/monthly_2012_05/58cd9bc133bc8_SR002KoteMounYo16441vsMP3Difference_zip.ad5a74a83f5e5795890afd0fded68864" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28080" src="/monthly_2012_05/58cd9bc133bc8_SR002KoteMounYo16441vsMP3Difference_zip.ad5a74a83f5e5795890afd0fded68864" class="ipsImage ipsImage_thumbnailed" alt=""></a></p><p><a href="/monthly_2012_05/58cd9bc956d27_16441vs24192difference_zip.59be35f20697d46dd7e5a4c52b7d071f" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28328" src="/monthly_2012_05/58cd9bc956d27_16441vs24192difference_zip.59be35f20697d46dd7e5a4c52b7d071f" class="ipsImage ipsImage_thumbnailed" alt=""></a></p><p><a href="/monthly_2012_05/58cd9bc95b9b7_SR002KoteMounYo16441vs24192Difference_zip.a6ccdafd25b35df086dc7aa4690793b8" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28329" src="/monthly_2012_05/58cd9bc95b9b7_SR002KoteMounYo16441vs24192Difference_zip.a6ccdafd25b35df086dc7aa4690793b8" class="ipsImage ipsImage_thumbnailed" alt=""></a></p><p><a href="/monthly_2012_05/58cd9bc9603f0_SR002KoteMounYo16441vsMP3Difference_zip.882cfc4894a8fe525818867b5cc2bc89" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28330" src="/monthly_2012_05/58cd9bc9603f0_SR002KoteMounYo16441vsMP3Difference_zip.882cfc4894a8fe525818867b5cc2bc89" class="ipsImage ipsImage_thumbnailed" alt=""></a></p>
  15. biberfan


    Source 2 Ghz Core 2 Duo Mac Mini (Mac OS X Snow Leopard) (5 GB RAM) SSD boot drive G-Drive External Drive (Firewire 800) iTunes Library Audirvana Plus BitPerfect Software - Apple Lossless, MP3, AAC (Baroque, classical, jazz) Remote Control iPad 2 with Apple Remote VNC DAC Bel Canto DAC 3 Cables Halide Bridge Acoustic Zen Wow! XLR (Balanced) Acoustic Zen Satori Shotgun (speakers) Shuntaya Power (amp) Amp Classé CA-2300 Speakers B&W 703 (biwired)
  16. Dears, Finally I've pulled the trigger and bought Burson HA-160D, OH-MY, This small little peace of Art from Burson is making my speakers (B&W 603 S3's) & Amp, sing like never before. currently I'm using it with my NAD C372 Integrated, and using MacBook 2Ghz/2GB via USB cable supplied by Burson. and my music preference is Female-Vocals, Classical. Musicality, & Jazz, -AND- My Quest has Began, hence I'm here to seek your expertise and help with the following: 1. What USB cable to buy? 2. How can I use my Mac book via optical out? what cable and Toslink converter? 2a which cable will give me Hi-Rez quality? 96? 384? etc. 3. after trying some of the iTunes alternatives I liked PM and Ammra but Ammra full version is insanely high for me so may be Ammra-Junior) or any other softeare to look for, which gives similar sound like the above 2? 4. I'm using HA-160D as a CD source to my integrated amp. and turned on the volume permanently to FULL on burson. and controlling the volume of NAD by Nad's remote control, now my question is -- will this in any way damage the burson, and is this a good idea to use it like this in a long run? using burson as a PRE is not an option for me for now with current amp. 5. Best ripping method for hi-res. (Preferably using Mac, but if any other solution is also fine) 6. Now, last but not the least. which headphones? I know there are many out there from cheap to expensive so I'm confused, and my budget is not that high in between $500 to 550 USD. keeping my music preferences mentioned above. 7. any one out there using the Burson HA-160D with Power Amp or Integtrated Amp? what is the best Integrated-Amp/Power-Amp that suits this if used HA-160D as a Pre. Cheers Pramod.
  17. In part 1, I used a null test technique to show that both FLAC and WAV (lossless) file formats are identical. In this post, I have expanded the null test to cover off playing the same FLAC and WAV files dynamically from JRiver and capturing the audio waveform after the Digital to Analog conversion and analog line output stage. Here is a high level block diagram of my test setup: For playback, I am using the exact same original FLAC and converted (by JRiver) WAV file I used in Part 1. It is Tom Petty and Heartbreakers Refugee at 24/96. JRiver is set up for bit perfect playback with no DSP, resampling, or anything else in the signal chain. I used the native Lynx ASIO driver to communicate between the sound card and JRiver. All sample rates for the tests are at 24/96. My Win7 64 Bit HTPC build is nothing special. No special power supply or SSD or interconnects. Side note, for Windows users, always invaluable to check your PC for latency with http://www.thesycon.de/deu/latency_check.shtml I have tested the frequency response of my Lynx L22 sound card using REW http://www.hometheatershack.com/roomeq/ and noise levels, distortion, etc., using RightMark Audio Analyzer http://audio.rightmark.org/index_new.shtml For capturing (i.e. recording) the audio waveforms, I used a Dell M4600 latptop and the onboard HD audio chip and driver. Here is the noise measurement of the on board sound chip. Not as good as my Lynx card above, but a check to see that everything is in working order. I used Audio Diffmaker http://www.libinst.com/Audio%20DiffMaker.htm for recording the waveforms that were coming off the analog outputs of my playback PC. Here is the process used by Audio DiffMaker: As an aside, I should point out that you can use this software to objectively measure anything in your audio playback chain that you have changed. Whether that be power supply, DAC, interconnects, music players, SSD, VST plugins, or whatever. Remember, if you are audibly hearing a difference when you change something in your audio system (ABX testing), the audio waveform must have changed, and if it has changed, it can be objectively measured. I find there is a direct correlation between what I hear and what I measure. For me, to form any valid opinion about audio reproduction, I want to correlate my subjective results with my objective results and vice versa. I want a balanced view. In the Audio DiffMaker help file, the software program is able to line up the waveforms if the program material is within 1 second of each other (protip). Here I am capturing the first 40 seconds of TP’s Refugee in Audio DiffMaker: I did this twice, once playing the FLAC and then the WAV, without making any changes on either computer. To test the DiffMaker software (and everything else) is working correctly, I took the FLAC recording and compared it to itself. Theoretically, it should null itself out completely. And it does. Ok so now let’s compare the two recordings, one FLAC and the other WAV: What the result is saying is that the difference signal is almost -90 dB. I repeated the test ten times and obtained the same results. You can listen to the difference track for yourself as it is attached to this post. PLEASE BE CAREFUL as you will need to turn up the volume (likely to max) to hear anything. I suggest doing this in volume level stages so you can verify there are no other artificats while listening. As you can hear for yourself, a faint ghost track of the music, that nulls itself out completely halfway through the track and slowly drifts back into being barely audible at the end. According to the DiffMaker documentation, this is sample rate drift and there is a checkbox in the settings to compensate for this drift: “Any test in which the signal rate (such as clock speed for a digital source, or tape speed or turntable speed for an analog source) is not constant can result in a large and audible residual level in the Difference track. This is usually heard as a weak version of the Reference track that is present over only a portion of the Difference track, normally dropping into silence midway through the track, then becoming perceptible again toward the end. When severe, it can sound like a "flanging" effect in the high frequencies over the length of the track. For this reason, it is best to allow DiffMaker to compensate for sample rate drift. The default setting is to allow this compensation, with an accuracy level of "4".” Of course this makes sense given that I used a different computer to record on versus the playback computer and I did not have the two sample rate clocks synched together. The DiffMaker software recommends this approach, but I have no way of synching the sample rate clock on the Dell to my Lynx card. So when this is not possible, the DiffMaker documentation indicates to use the sample rate compensation. However, when I tried the sample rate compensation, the DiffMaker program thru the following error: I sent an email to the software manufacture and will follow up once I hear back. Given that the signal is almost -90 dB from the reference and that the noise level of my Dell sound card is -86 dB, we are definitely nearing the limits of my gear. Also, given that the dynamic range of most music material we listen to is less than 20dB http://en.wikipedia.org/wiki/Dynamic_range#Audio it seems unlikely that I could hear the difference track, relative to the reference level – that’s a 90 dB difference. Subjective Listening Tests In JRiver, I played the FLAC and WAV (and vice versa) several times through headphones and speakers. I did this sighted and blind. I also played back the recorded reference and compare files in Audio DiffMaker using headphones. Finally, I played back the Reference + Difference track. In my subjective listening tests, I could not hear any differences between the FLAC and WAV files in any combination of the above. Not only from the playback machine but also the recorded tracks. They all sounded identical to me. There seems to be good correlation between objective and subjective results. As a side note, I have been into audio and music for over 40 years. For 8 of those years I was a recording/mixing engineer where I was trained and relied upon to note very small audible changes. http://www.thepikes.com/bio The reason I am saying this is because of psychoacoustic http://en.wikipedia.org/wiki/Psychoacoustics effects, our ears can be easily fooled http://en.wikipedia.org/wiki/Auditory_illusion or put in a positive way, our ears adapt to changes very quickly. In fact, most recording, mixing, and mastering engineers use these psychoacoustic effects on purpose. For example, the HAAS effect http://en.wikipedia.org/wiki/Haas_effect#Experiments_and_findings to make the sound more full, wider, sense of air, etc. All tricks played on our ears: http://www.algorithmix.com/en/kstereo.htm including some remastered material we download from HDTracks. So do we not trust our ears? I am not saying that. What I am doing is bringing a balance of both subjective and objective thoughts together so we can correlate what we hear with what we measure and vice versa. Again, when performing ABX listening tests, if you are hearing an audible difference, then the waveform must have changed. If the waveform has changed then we can measure the difference. Btw, all of the software used in these tests is free. I would encourage you to download the software’s and try this out for yourself as it does not require any special equipment. Further, you can objectively quantify any differences throughout the audio chain in your playback system. In conclusion, using my ears and measurement software, on my system, I cannot hear or (significantly) measure any difference between FLAC and WAV. Not only just file formats, but the rest of the audio playback chain as well. Happy Listening!<p><a href="/monthly_2012_05/58cd9bc10d5d3_TPRefugeeflacwavdifference_zip.7edacfe8fdec5923b37786c66c719085" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28073" src="/monthly_2012_05/58cd9bc10d5d3_TPRefugeeflacwavdifference_zip.7edacfe8fdec5923b37786c66c719085" class="ipsImage ipsImage_thumbnailed" alt=""></a></p><p><a href="/monthly_2012_05/58cd9bc112c71_FLACvsMP3AudioDiffMakertest_zip.1e0ae86bc8309e7c67b5e3bfa5859ea6" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28074" src="/monthly_2012_05/58cd9bc112c71_FLACvsMP3AudioDiffMakertest_zip.1e0ae86bc8309e7c67b5e3bfa5859ea6" class="ipsImage ipsImage_thumbnailed" alt=""></a></p><p><a href="/monthly_2012_05/58cd9bc117b12_FLACvsMP3FileNulltest_zip.75cfcaa945e30ee84f2f3496a30f3221" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28075" src="/monthly_2012_05/58cd9bc117b12_FLACvsMP3FileNulltest_zip.75cfcaa945e30ee84f2f3496a30f3221" class="ipsImage ipsImage_thumbnailed" alt=""></a></p><p><a href="/monthly_2012_05/58cd9bc93ec4e_TPRefugeeflacwavdifference_zip.da60e6c8f17aaf005c1def1aff6ab600" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28323" src="/monthly_2012_05/58cd9bc93ec4e_TPRefugeeflacwavdifference_zip.da60e6c8f17aaf005c1def1aff6ab600" class="ipsImage ipsImage_thumbnailed" alt=""></a></p><p><a href="/monthly_2012_05/58cd9bc944659_FLACvsMP3AudioDiffMakertest_zip.76f778e131f30391613959b5ca5c6cda" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28324" src="/monthly_2012_05/58cd9bc944659_FLACvsMP3AudioDiffMakertest_zip.76f778e131f30391613959b5ca5c6cda" class="ipsImage ipsImage_thumbnailed" alt=""></a></p><p><a href="/monthly_2012_05/58cd9bc948d99_FLACvsMP3FileNulltest_zip.10aa080a234c3daba1815c93edfa470c" class="ipsAttachLink ipsAttachLink_image"><img data-fileid="28325" src="/monthly_2012_05/58cd9bc948d99_FLACvsMP3FileNulltest_zip.10aa080a234c3daba1815c93edfa470c" class="ipsImage ipsImage_thumbnailed" alt=""></a></p>
  18. Lenrick

    NAD C390DD II

    I've got everything connected and it is up and running. First impression: sounds really good, but needs some fine tuning. I have to set the digital crossover between signal going to the main speakers and the subwoofers. The main speakers are the Focal Chorus 826 V with a frequency response of 45Hz - 28kHz, and the subwoofers are the Velodyne MiniVee 8" with a frequency response of 28-120Hz. So I guess it makes scene to have the crossover somewhere in the range 50-120Hz (the settings are discrete in steps of 10Hz, so 45Hz in not an option). Instinctively I would like to use the main speakers as low as possible, meaning a crossover at 50Hz but I just have to try and listen. A very nice feature that I like is that you can 'disable' the inputs you are not using, making the selection faster and simpler. I will use three inputs: the digital coax 1 for the Apple MacBook (with a M2Tech HiFace inbetween), the optical 1 for a Apple Airport Express for streaming, and the front USB for an occasional USB flash drive. The rest of the inputs are disabled so I don't have to browse through them every time I change source. Links: http://www.focal.com/en/home-audio-loudspeakers/hifi-speakers/floorstanding-speakers/chorus-826-v.php http://velodyne.com/subwoofers/shop-by-series/minivee-series/minivee.html
  19. Seems that one Bartholomaus Traubeck has devised a turntable that will play the rings of trees, rather than conventional vinyl, as reported by HuffPo in this report: http://www.huffingtonpost.com/2012/02/01/huffpost-arts-interviews-_n_1247581.html?ref=arts Years - 'scratching' from Bartholomäus Traubeck on Vimeo
  20. Seek and preserve maximum source information when transferring analog audio to digital formats. Don’t let anyone tell you that it doesn’t matter, that your albums are not in pristine condition, that your equipment isn’t good enough, or you are not a trained listener. http://bit.ly/xm3g5C
  21. So I've finally gotten around to locating the speakers and came up with some interesting results: 1) All locations had their deepest nulls at 100Hz, ranging from -7dB to -24dB. 2) All locations had their highest peak at 125Hz, ranging from +4dB to +8dB. 3) Secondary nulls were consistently at 160Hz and followed by secondary peaks at 200Hz. The location that I've settled on for now is 35* off axis and puts the tweeters 40" from the back of my chair and 37" from each other. They're set up to fire about a foot behind my head when seated and raked back on a pair of Auralex MoPads so that their driver's fire at my ears at a comfortable seating height. This position yields the second-best bass response +/-8dB from 50Hz to 400Hz versus the +7/-14dB results that I measured from where I had my speakers before I started this whole thing. I've decided to spend some time with this setup as I read more and look further into REW. My gut, or ears, are still telling my that the soundstage isn't quite to my liking and I'm still definitely hearing a good amount of the room. On that last point, I've been going round and round and round as to how to treat for bass and FRP in a DIY fashion. I'm limited to treating the front of the room where the speakers are as far as bass trapping goes so I'm pretty sure that I'm going to install some "SuperChunks" in the front two corners and then a wall-to-wall soffit trap of 16" x 16" x 10'. The superchunks would run from the floor to the bottom of the soffit trap. Then the plan would be to treat the wall behind the speakers along with the first reflection points on the sides with portable traps, mounts a cloud trap at that FRP and possible use another portable panel somewhere behind the listening position. With winter in full swing the project of building those may have to wait until spring but we'll see. Either way I'll be using the time doing more research, possibly more measurements, and getting to better "know" the current setup. Bill
  22. jemhayward

    My Journey

    I've been a keen audiophile for over thirty years, though, in that era, the word audiophile was not used. In many ways a much simpler world where all amplifiers sounded the same, turntables rumbled and wowed, unless you had lots of money to spend, tapes came on reels, and valve amplifiers were available in the local junk shop for a few pounds... In my impoverished teenage years I built amplifiers, speakers and turntables from kits and from articles in the press, and had many hours of pleasure and frustration extracting more and more from my growing vinyl collection. My working life began at a time when the hifi press realised that some things sounded better than others even when the measured parameters were similar. Subjectivism was the new rationalism. My TD160 made way for a Rega 3 (though I should have kept my obscure Audiocraft arm) and the Rega soon made way for a Linn LP12 and the Basik Arm was quickly replaced by an Ittok. I'd used Decca cartridges, but after a brief journey through Linn/AT MM carts, I discovered the moving coil, and the Naim Pre-Amp, but still kept faith with my various power amplifiers; a Leak Stereo 20, home made MOSFET monoblocks, and a lovely home built Linsley Hood class A amp. Whilst trying to improve on my Naim 62, which lead to my building a completely new preamp, I happened to discover that the Linn Isobarik that I had always coveted had become relatively affordable on the used market, and my obsession with active Isobariks began. I used Linn amps and crossovers, and then Naim, but would have really loved a combination of the two, but a change to a much smaller house made the Isobariks difficult to live with (they sounded fine in a tiny room, but were difficult to squeeze past!) so I thought I'd try my other fantasy speaker, the Quad ESL57. This proved my suspicion that the Isobariks lacked transparency and the holographic imagery that I said I didn't need, did make music more real. I remained faithful to vinyl for many years but eventually was forced to get into CD replay. A Meridian 200 transport looked, felt and sounded lovely, and the Arcam black box gave the CD playing experience some semblance of musicality, though never that final edge that LPs delivered. A few mods helped a bit, but an Audio Alchemy DDE helped a lot more. Then I heard the Naim CDS. So, about six years ago, my system was Linn LP12 Lingo, ARO, Lyra Lydian, Naim CDS/CDPS, Naim 82, 2xHICAP, Naim 250, Quad ESL57 and Linn Sizmik Sub. It sounded pretty good. A friend who likes a decent stereo, but is not a fanatic had a problem with a scratched CD, which was difficult to replace, and being a computer programmer, he decided that he should back up his CD collection to HDD just in case. He did a lot of painstaking research, and settled on EAC as a ripping system, and FLAC as his file format. I needed to get MP3s for my portable player, so I emulated his set up, and started collecting FLAC and MP3 rips of my favourite CDs. I was using my laptop as a signal generator to track down an annoying room resonance and so, when I thought I'd fixed it, I played a familiar track, but used the FLAC file so as to avoid the cable swap. It sounded remarkably good. It lacked the substance of the CD via the CDS, but in terms of detail its was at least as good, maybe better. I acquired a used mini Mac, with a non-apple PSU, but it worked ok, and was cheap, but it didn't sound any better than my laptop, though fitted in the rack better. I got a M-Audio USB to SPDIF interface, and got a Behringer DEQ2496 to try to fix the room/speaker interaction issues. The Behringer worked as a DAC, so I could play 24/96 high res, through my system. a Linn studio master was bought, and suddenly I had better sound quality than anything previously encountered. The CDS still had more "guts" and LPs still had more "soul" but in a slightly sanitised way the sound from the computer was more real. I was invited to my local dealer to hear the Linn DS range. I was interested, if only to turn a multi-box system into a one box solution, but the Linn, from Klimax all the way down to Sneaky was resolving things way beyond my Mac/M-Audio/Behringer setup. The Sneaky and Majik sounded identical to me, the Akurate was only slightly better and the Klimax, a bit better yet, but way beyond my budget, so I bought the Sneaky as it would also serve as my kitchen music system. The Mac was passed on to my son, the M-Audio went into the drawer. The Behringer went into retirement, as we'd gone mad and replaced the Quads with Martin Logan Summits, solving all the room issues, and expanding the system bandwidth and transparency by an order of magnitude. The CDS went to Hong Kong via eBay, and FLAC was now my primary digital source. The Linn DS was working well, and we finally got a Squeezebox touch, which we thought may be useful and I secretly hoped it would be as good as the Linn, so I could sell the Sneaky to provide funds for a new cartridge. One evening with the Touch/Behringer in place was enough to make my wife veto that idea, but we did discover that the Behringer DAC was better than the Linn Dac by a small but noticeable margin. So, best sound so far... Linn DS with Behringer, next Squeezebox Touch with Behringer, then Mini Mac M-Audio Behringer bringing up the rear. All though my Naim amps. It was all sounding good, but I felt I wasn't getting the last ounce of performance from my Summits and occasional over enthusiasm with the volume knob would cause the NAP250 to go into current limit shut down. I'd heard the Summits on the end of a NC552/NAP500 so I knew there was more to come. I spotted an EAR890 used, at a good price, and so decided to try the valve world again. Many years ago I had heard EAR549s and was stunned by the non-valve sound and arresting clarity, hoping the 890 would be similar, I took the plunge, sold the NAP250 (for a very good price!) and was really taken aback by how much clearer the EAR890 was than the NAP250, whilst losing none of the power and slam that the 250 does so well. Next job was improving my ageing NAC82 preamp - servicing it proved expensive, and actually made little or no difference to the sound, but whilst it was away, I was forced to use a passive pot between DAC and power amp, and when I slotted the NAC82 back in, I lost some transparency, fine details disappearing and an overall loss of realism. So, eBay it was for the NAC82 and I was in search of a DAC and better preamp. Naim DAC and NAC252 looked a viable combo, though the potential costs made my eyes water, even secondhand, but a demo at my dealer proved that the naim DAC was much, much better than the Behringer, so that went on the wish list and I started saving up. A chance email call from a friend about computer audio led him, on my recommendation, to try a Squeezebox Touch, and he tried this through the DAC of his Mark Levinson CD player, but this wouldn't allow 24/96 play that I recommended very highly, so he looked for a new CD player with the sort of budget most of us only dream of. He finally settled on the Audiolab 8200CD, as that gave him the best CD player he could audition, and high resolution DAC all in one box. The fact he'd gone for something so "cheap" (for him) rather amazed me... "just listen" was his response. I started lurking on PinkFish forums hoping to find a good preamp to go with the Naim DAC I was looking forward to, and started hearing things about the Audiolab stuff, particularly the MDAC, which appealed because it was small (fits on my rack I had built for NAIM PSUs) had volume control, and the design ideas seemed to be pretty sound. It was also relatively cheap, very hyped (so easy to sell on if no good) and also had balanced outputs so could drive my EAR890 via long cables, making room layout a bit easier. The MDAC eventually hit my system late last year. In terms of sound quality improvement, it is probably one of the biggest single step changes in quality I have experienced. It is better than the Naim DAC (with a moderately good Naim preamp) by quite a large margin. It may not better a naim DAC into NAC552, but it is only £600, that difference buys a lot of music. So, my system is now: Linn LP12/Lingo, Naim ARO, Zu DL-103, Naim Prefix, Naim HICAP, EMU 404 ADC, Linn Sneaky DS, Audiolab MDAC, EAR890, ML Summits. If you have read this far, I should reward you with some things I think I have learned. I think my conclusions about digital streaming sources are flawed. My assertion that the mini Mac is relatively poor may be as much to do with the interference that the PSU made with the Naim amps, than any judgement on the Mac itself. Naim amps are very sensitive to mains quality, and I've always found that switching off my Lingo PSU improved the sound of CDs, so I suspect the PSU was affecting the overall sound. Into the same Behringer DAC the Squeezebox Touch was notably worse than the Linn DS, whereas my friend with the Audiolab was happy, and used to some very high end kit... but I now know that the Touch has high jitter on its digital output. The MDAC copes well with jitter, so I suspect a re-match into the MDAC may give a less conclusive result. Thirty years ago I had a system where the turnatble cost twice as much as the amp, which cost twice as much as the speakers, that was the way on those days. I now have speaker that would cost twice as much as my power amp, a preamp/dac that cost not much more than a tenth of the cost of the power amp, and could be happy with a source that costs less than the price of ten CDs... The world has changed.
  23. Having a software pc repair is typical, Either you can get it through your local repair shop and let you be his or her regular customer and later on get a discount or some freebies. There are also ways like searching the net for answer for your specific problem and as we all know, The net is a one stop resource for everything and there are DYI or Do it yourself as we all know it kind of way to let alone fix the problem. But we also know that there are certain pc issues that not all can fix, only those few whom have devoted thier time and life in perfecting or mastering thier craft to be at your service. It is just like there are consumers and there are service or product seller, just a typical way of seeing things. Get the picture? We all can't take away all those doubts and fears, specially now that everyone can pretty much purchase everything online, with just a click of a mouse. There are frauds and there are scams, you just have to find those legitimate business online. They are establish to help your needs and what's good about this services is you don't have to leave your home carrying your heavy computer and wasting your time, most of this service provider are online 24/7, waiting and ready to help you. Specially this one service that i've tried in which i thought that my important files are history but still, they've manage to repair my pc and and give a free one month free scan for maintenance and what's more they are the cheapest in the online market when it comes to online remote pc repair service.
  24. There doesn't appear to be much information available on the BT100 blu-tooth device, so I asked them for more details. See below for CA's response: ------------------- The maximum audio resolution supported by the BT100 is 16/48. The apt-X protocol used is the standard apt-x bluetooth. However, your music will still be upsampled to 24/384 via the BT100.
  25. Some people regard the appearance of a "computer" in an audio system setting as inappropriate and ugly. My wife does. I don't end up seeming to care that much, particularly as I am frequently swapping the things around, but I can surely appreciate the fine touch of someone like Jeffrey Stephenson. See some of his stunning work in wood at http://slipperyskip.com.
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