What makes a high end audio amp "better" then a low end unit? - Page 1
Forum section: electronics / beginners

author: Beamin
author_rank: Super Contributor
timestamp: October 31, 2017, 12:14:15 pm
content: I have always wanted to make my own amp that was compareable to a macintosh amp for a fraction of the price but what exactly makes it better? Lets ignore the warm harmonics the tube produce to make a better sound for simplicity so not get into the tube/solid state which is better debate.I know amps can be graded by their THD to give you a rough idea: For instance a shit sony or equivalent will have a THD of 0.8% while a harmon kardon in that power range will be around 0.08% and a Marantz will be around 0.03%. Not that this is the answer to good sound you can hear the difference in a sony and a better amp.So to get the best sound: Is it just a matter of hand selecting all the parts so the values are as close to design spec as possible? This would make sense why a cheap amp would sound worse because they can just buy 5% components and use them all where as higher end starts with a lower tolerance and then will hand pick them. Also would matching parts: LR channel transistors that are very close also play a big part?So if my understanding is correct you could take an amp that has a good circuit design (are almost all of them mathematically the best we can do since its really a simple circuit that's been improved over 100 years) and then hand selected the best/matched parts;IE buying 100 resistors and discarding 99% of them could you turn the cheap sony amp into a good one? Perhaps also over sizing filter caps/ over specing parts to run at less load etc.

author: Zero999
author_rank: Super Contributor
timestamp: October 31, 2017, 12:26:53 pm
content: Quote from: Beamin on October 31, 2017, 12:14:15 pmThe audio amplifier is a problem which was solved over 40 years ago. Unless it's poorly designed or badly overdriven, an audio amplifier will contribute a negligible amount of distortion, compared to the speakers and room acoustics. Lots of it is marketing bunk.Quote from: Beamin on October 31, 2017, 12:14:15 pmThat depends on the music, the nature of the distortion, the listener, as well as the speakers and acoustics. I doubt most people will be able to tell the difference between 0.03% THD and 0.8%THD.

author: Johnny10
author_rank: Frequent Contributor
timestamp: October 31, 2017, 12:46:27 pm
content: I collected 60's and 70's Mcintosh solid state amps and tube preamps.Well designed, solid construction, great transformers.And Hero999 is correct " audio amplifier is a problem which was solved over 40 years ago."Take a look at you tube videos from El Paso TubeAmps and you can learn the about cloning mcintosh amps and what makes each type of HI-FI tube amp unique.Seems the transformers are what makes the difference.ElPaso Tube Amps would say, the McIntosh transformer is something not easily recreated !

author: tszaboo
author_rank: Super Contributor
timestamp: October 31, 2017, 01:01:49 pm
content: Nowadays they make Class-D amplifiers with a single chip (digial input) 100+W output and <0.01% THD+N. I dont even understand why there are amplifiers anymore. Seems so pointless. I believe that good quality active speakers are the way to go.

author: Jr460
author_rank: Regular Contributor
timestamp: October 31, 2017, 01:07:07 pm
content: Based on the OP, he wants a home amp. Things are slightly different if you deal commercial amps for driving large systems. These are run much closer to max power. They need to is some cases support loads down to 2 Ohms and also have very good protection. Some people would test amps by shorting the output and then driving it at the max. Come back 24 hours latter and see if has not failed, and would put on a normal load and see if it works.Also is the mechanical side of things. Being able to withstand the road abuse, and being is direct hot sun or sometimes mist.

author: macboy
author_rank: Super Contributor
timestamp: October 31, 2017, 01:35:10 pm
content: Quote from: Beamin on October 31, 2017, 12:14:15 pmFirst of all, there are no legal standards against which audio amps are measured. So, every manufacturer make up its own specifications and limits. The "shit sony" gives a THD rating of 0.8% not because it actually produces that much distortion most of the time (it definitely does not), but because that THD rating allows higher power ratings. Most amps produce least distortion at a couple watts output, and increase from there. If that sony was rated at 0.08%, then maybe it could only manage 50 W before hitting that mark. But move the limit to 0.8% and it can push out 80 W at that limit. So do they market it as a 50W @ 0.08% or 80W @ 0.8% ? Sony chooses one way, others like HK go the opposite way. Most people buy based on power numbers, not THD numbers. The fact is that even a "shit sony" produces much less than 0.05% most of the time. (I'm not just waving my hands around, I have measured plenty of them).I would love if the manufacturers gave THD vs power and THD vs frequency graphs. IMD specifications. Noise floor graph (distribution of low vs high frequency noise). Step response. But they don't. So you can't compare amps by the numbers, unless you implicitly trust the test methodologies in certain HiFi magazines (tip: don't).

author: oldway
author_rank: Super Contributor
timestamp: October 31, 2017, 02:20:28 pm
content: I repair many vintage (years 60 and 70) high end amplifiers of different brands, both tubes and transistors and I also have the opportunity to listen to them.I am not a music expert and my age no longer allows me to judge a sound because my hearing is that of a person over 65 years, that's means very poor.I am sometimes surprised about some customers tell me .... the sound more or less perfect of some amplifiers compared to others .... And yet, on the test bench, they have similar caracteristics, no technical reason to give a different sound.I especially judge the amplifiers on the technical plan, the quality of manufacture, the diagram, the quality of the components, the estetics, ....

author: oldway
author_rank: Super Contributor
timestamp: October 31, 2017, 02:39:37 pm
content: For me, what really makes the difference is not the amplifier but the speakers.An experienced music lover can only discern distortions greater than 3 or 5%.A speaker produces a distortion of at least 1%, often 2%.So, do not tell me that a 0.1% distortion of an amplifier makes any difference.Moreover, paradoxically, it is the tube amplifiers that have the highest rate of distortion that are considered to have the best sound.A comprehensive and very interesting study of the speakers was made by G. A. Briggs, one of the founders of the British firm Wharfedale, in his book loudspeakers.A big technical difference between tube and transistor amplifiers is the damping factor ... that may partly explain the difference in sound of a tube amp.My personnal preference: I have a Marantz 2285B receiver and a Sansui AU-5500 amplifier. (I think the AU-5500 and AU-6600 are the best Sansui)

author: coppice
author_rank: Super Contributor
timestamp: October 31, 2017, 03:56:19 pm
content: Quote from: Hero999 on October 31, 2017, 12:26:53 pmActually 40 years ago (mid 70s) is when a lot of audio amplifier problems were being created. That was the era of 1MHz bandwidth audio amps, with masses of negative feedback, and bizarre transient performance. It was the mid to late 80s before the industry settled down to more sane designs, with proper testing of transient performance.More recently the move to switch mode amps, to allow things like compact and powerful 7.1 designs, created its own performance issues. Again, most of this was due to a focus on static testing, and too little transient testing. This is only now settling down.

author: Seekonk
author_rank: Super Contributor
timestamp: October 31, 2017, 04:08:43 pm
content: Just tell me what you can build "for a fraction of the price" that is better than what you can buy.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: October 31, 2017, 04:58:18 pm
content: Well, I've designed many audio amplifiers, some of these apparently quite good (from what reviewers wrote). The answer to the topic question is simple - a better sound quality. "High end" doesn't need to be expensive, some of my best amps were sold at about £300 retail and (IMHO) still could qualify as "high end" from the sound quality perspective. How to make an amplifier that sounds good - that is a much better question to ask. However there is no simple answer to that question. Earlier replies in this thread are from people who never made a good sounding amplifier, I supposeCheersAlex

author: coppice
author_rank: Super Contributor
timestamp: October 31, 2017, 05:13:41 pm
content: Quote from: Alex Nikitin on October 31, 2017, 04:58:18 pmWhat you say is accurate. There are good amps for £300. Good speakers, on the other hand, don't come cheap. Most people would get a better result by spending less on their amp, and using the extra cash for better speakers.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: October 31, 2017, 05:45:34 pm
content: Quote from: coppice on October 31, 2017, 05:13:41 pmI beg to differ. There are some good speakers which don't cost the earth (I'm quite happy with a pair I've bought for £250 second hand, including stands), but from my experience the quality of amplification is just as important as the quality of speakers in a system. Moreover, correct matching of these two parts is essential .CheersAlex

author: RandallMcRee
author_rank: Frequent Contributor
timestamp: October 31, 2017, 06:00:38 pm
content: Many of the key variables that differentiate one amplifier from another are not published in the audio industry, at least not widely. THD is, of course, just one small component of the overall performance of a complete system. Its not that important.Two key things I found in building a preamp are channel separation and channel matching.It was surprisingly hard, for example, to get crosstalk low enough so that one channel, at high volume, did not bleed through to the adjacent, silent, channel. Simply running two signal wires in close proximity can cause noticeable problems.For the same reason, grounding can be a big deal.Proper channel matching and separation are what create that 3-d presentation most folks find so wonderful. Well, the whole system contributes and, yes, the speakers are the key component.

author: b_force
author_rank: Super Contributor
timestamp: October 31, 2017, 06:05:41 pm
content: I have been professionally in the field for many years.The simple and only (scientific) correct answer is marketing and placebo.It doesn't cost much at all to make a perfect good performing amplifier.Although, you have to realise that it all depends on the context. Pro audio customers have different needs than tiny boom box producers, as well as guitar amplifiers.So that means the approuch and compromises will be (very) different as well.

author: nfmax
author_rank: Super Contributor
timestamp: October 31, 2017, 06:06:09 pm
content: Quote from: coppice on October 31, 2017, 03:56:19 pmThe QUAD 303, from50years ago, was one of the first all-transistor audio power amplifiers to be essentially 'perfect' (within its bandwidth, voltage swing, and load impedance limits, of course). Any more recent design that isn't 'perfect' has been done wrong!

author: oldway
author_rank: Super Contributor
timestamp: October 31, 2017, 06:07:41 pm
content: Quote from: coppice on October 31, 2017, 03:56:19 pmNot as simple of that.....Here are different schematics of power stage of amplifiers of the years mid 70:- Marantz 140dc: 1976-77- Marantz 2285B : 1977-80- Sansui AU-5500 : 1974-76There are very sofisticated (Marantz 140dc) and very simple power stages (Sansui AU-5500) in the same time.

author: nfmax
author_rank: Super Contributor
timestamp: October 31, 2017, 06:10:42 pm
content: Quote from: RandallMcRee on October 31, 2017, 06:00:38 pmI was working at QUAD when the 34 control unit was being developed. Roger Hill, the lab technician, tweaked the PCB layout by adding bits of copper tape as extra ground plane, while monitoring the crosstalk over the whole audio range using a frequency response analyser (a Wayne Kerr of some sort I think - I can't remember now). -60dB was the aim.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: October 31, 2017, 06:21:39 pm
content: Quote from: b_force on October 31, 2017, 06:05:41 pmSimple- yes, scientific- no. Only pseudo scientific, and as such, incorrect.CheersAlex

author: nfmax
author_rank: Super Contributor
timestamp: October 31, 2017, 06:30:46 pm
content: Quote from: Alex Nikitin on October 31, 2017, 06:21:39 pmThere is a great deal of fashion about the high-end audio business. Which is neither good nor bad, it depends if you like fashion. Treat it as a fashion business and it makes sense.That said, there are real audible effects which seldom get mentioned. My personal hot button is mechanical noise, usually mains hum from transformers. What good is it for the output audio to be completely noise and hum free if the amplifier itself is buzzing like a microwave oven warming up a Chicken Tikka!Talking of which...

author: oldway
author_rank: Super Contributor
timestamp: October 31, 2017, 06:44:28 pm
content: What is a good sounding amplifier?How to know if the sound you hear is the sound equivalent to the original sound?The appreciation of a sound is totally arbitrary and subjective.Do you have enough hearing?Have you tested your hearing?How old are you ? Me, for example, I have more than 65a and I do not allow myself to judge the acoustic quality of an amplifier because at my age, we do not hear beyond 12 Khz.Do you go to concerts?What kind of concerts?Because at certain types of concerts, the sound level is so high that it reaches the pain threshold and it can no longer be used as a comparison.I sometimes have customers who are real music lovers, in their thirties and who go to concerts (opera, classical music) ...These are references and I think their opinions are credible ....But there a lot of people who give their opinion about sound quality and who are not qualified to do it.

author: b_force
author_rank: Super Contributor
timestamp: October 31, 2017, 06:48:33 pm
content: Quote from: Alex Nikitin on October 31, 2017, 06:21:39 pmPseudo science?Don't think so, there are a billion papers about noise and distortion thresholds, and what other issues have an effect how we percieve things.(hearing, tasting, seeing)The fact that many manufactures try to tell that people need even less distortion to be even better (fill in subjective wine tasting like words), is just prove how strong markteing (and thus placebo effects) is.A normal human being is never able to detect those differences.Funny enough is that most comparisons are never done the proper way.But like I said it does depend on the context. In a line array with 4-6 high sensitive compression drivers, you have to be sure to make your SNR is as good as possible. Otherwise it WILL be noticable.

author: oldway
author_rank: Super Contributor
timestamp: October 31, 2017, 07:16:29 pm
content: There is yet another important factor in the reproduction of the sound that no one has spoken of, and yet it is essential, the acoustics of the local.It's funny to see that people are focusing their attention specifically on amplifiers when there are many other more important factors.

author: Kleinstein
author_rank: Super Contributor
timestamp: October 31, 2017, 07:43:10 pm
content: One should not be able to hear the difference between any two reasonably good amps. They should be all so close to perfect that one can not hear the difference.Modern class D amps are also good enough. At lower power there is no much sense in building your own class AB amplifier like in the 1970s/1980s, as integrated amps are good too.The more problematic parts are the speakers and if used passive cross over networks. From the technical perspective it absolutely makes sense to not have active speakers, as this one way for a possible improvement on the speakers.While building an class AB amp is rather straight forward and simple, the acoustic part for speakers is not that simple and might need more experimentation.

author: Zero999
author_rank: Super Contributor
timestamp: October 31, 2017, 08:01:24 pm
content: Quote from: coppice on October 31, 2017, 03:56:19 pmWhat do you mean? The NE5532 and TL072 were around back then and are more than good enough for audio and discrete transistor technology was more than good enough for audio power amplfieirs.

author: Cliff Matthews
author_rank: Unknown
timestamp: October 31, 2017, 08:03:11 pm
content: And then there's Nelson Pass... He must be one of the oldest amp designers on the planet.https://www.youtube.com/results?search_query=Nelson+Pass+lecturing+at+the+Burning+Amp

author: grifftech
author_rank: Frequent Contributor
timestamp: October 31, 2017, 09:00:35 pm
content: less mains hum and noise

author: b_force
author_rank: Super Contributor
timestamp: October 31, 2017, 09:56:14 pm
content: Quote from: oldway on October 31, 2017, 07:16:29 pmWell yes, that's actually the whole point, bit I didn't wanna go all that way, but it was essentially what I meant.If you put it all in perspective the amplifier is one of the very last things you need to put effort in.

author: Circlotron
author_rank: Super Contributor
timestamp: October 31, 2017, 09:57:14 pm
content: Quote from: oldway on October 31, 2017, 07:16:29 pmThey’re just staying on topic, addressing the original question.

author: Zero999
author_rank: Super Contributor
timestamp: October 31, 2017, 10:22:52 pm
content: Quote from: grifftech on October 31, 2017, 09:00:35 pmIt's possible to design a good, low cost, audio amplifier, with virtually imperceptible levels of noise.

author: oldway
author_rank: Super Contributor
timestamp: October 31, 2017, 11:09:58 pm
content: Quote from: Circlotron on October 31, 2017, 09:57:14 pmWhen we say "What makes a high end audio amp" better "then a low end unit?", We do not ask a question, we say that a high end audio amp is " better "then a low end unit....Is it true ?It's an assertion disguised as a question....

author: oldway
author_rank: Super Contributor
timestamp: October 31, 2017, 11:08:08 pm
content: Quote from: Circlotron on October 31, 2017, 09:57:14 pmWhen we say "What makes a high end audio amp" better "then a low end unit?", We do not ask a question, we say that a high end audio amp is " better "then a low end unit....Is it true ?It's an assertion disguised as a question....

author: coppice
author_rank: Super Contributor
timestamp: October 31, 2017, 11:21:22 pm
content: Quote from: nfmax on October 31, 2017, 06:06:09 pmThe QUAD 303 was an excellent amp, if a bit overpriced. There were quite a few good transistor amps 50 years ago, from Europe, Japan and the US. Then the industry went crazy, and most manufacturers, especially the high volume ones, produced terrible amps for some years.

author: wraper
author_rank: Unknown
timestamp: October 31, 2017, 11:23:57 pm
content: Quote from: Cliff Matthews on October 31, 2017, 08:03:11 pmHis products are mostly marketing BS IMO. All those super expensive amplifiers with inferior specs for a reason of beingfeedback free. And those were not some tube amps where this would be understandable, but amplifiers consisting from transistors.

author: coppice
author_rank: Super Contributor
timestamp: October 31, 2017, 11:33:54 pm
content: Quote from: wraper on October 31, 2017, 11:23:57 pmPoorly thought out feedback scheme were the cause of most of the horrible transient performance issues in amps 40 years ago, so its an obviously sound marketing ploy to go to the opposite extreme and say all feedback schemes must be bad. Its dumb, but most potential customers aren't really in a good position to make sense of more subtle arguments. Its like trying to get the public to buy into the idea of a balanced diet, rather than the latest extreme diet fad. The rational argument isn't very profitable.

author: vk6zgo
author_rank: Super Contributor
timestamp: October 31, 2017, 11:52:51 pm
content: Quote from: coppice on October 31, 2017, 03:56:19 pmI think the Idea of very wide bandwidth audio amplifiers had its origin in the tube era.An amplifier with a fairly large amount of negative feedback needed to extend the point at which the feedback changed to positive to well above the normal audio spectrum.The Achilles heel of tube amps was always the output transformer.A lousy transformer was more likely to bring the point ofphaserotation too close to the required audio frequency range.It seems the iPad thought I was talking about Football."Offside" indeed! Aaaaarrrrrgggghhhhh!

author: Audioguru
author_rank: Super Contributor
timestamp: November 01, 2017, 12:56:40 am
content: Quote from: oldway on October 31, 2017, 06:44:28 pmUnlike you, I am a music lover. When I became 69 years old I thought that people mumbled and radio and TV engineers were not aware of their terrible sound. I went to a free hearing test and it showed that my severe high frequencies hearing loss was normal for my age so they programmed a hearing aid for me to try. I couldn't stand it! Later I went to another free hearing test and it was the same results as the other test. They set up hearing aids from a different manufacturer with 20% high frequencies boost and I liked it. 2 weeks later they boosted it some more and after 2 more weeks they gave me a completely flat frequency response so that my hearing with the hearing aids is the same as when I was young. Sure they were expensive but they changed my life and are worth it. They also have features that normal hearing cannot do like a setting (each hearing aid has a push button) for extra sensitivity and I can hear people talking far way. They have an audio compressor so that nothing is too loud. They have a muting setting and have a setting that I use in noisy malls and restaurants that has noise reduction. They are directional: front, back, left and right and any combination of them.I do not understand why old people go around with poor hearing.Also vision. I became blind with cataracts so I had the bad lenses in my eyes removed and replaced with custom synthetic lenses. Now my vision is better than when I was young and I do not need glasses anymore. I do not understand why people go around .....Also my heart. I had a heart attack, nearly died so they fixed my arteries. Now I am strong, exercise a lot and feel like I am young again but I look like my age of 72. I do not understand why people .....I can hear when a sound system has distortion more than 0.1%, has frequency response problems and has noise. Some cheap systems are much worse than Sony and some expensive systems look expensive but sound bad. Some systems that are not cheap but not expensive sound fantastic. Some expensive speakers sound bad and some reasonably priced speakers sound good. Cheap speakers always sound cheap.

author: tszaboo
author_rank: Super Contributor
timestamp: November 01, 2017, 12:58:22 am
content: Quote from: wraper on October 31, 2017, 11:23:57 pmI got the impression, that he has no idea what he is doing. It is more an art, than real design. Like "look, i've connected a transistor the wrong way, and it still acgs like an amplifier. Lets connect two dozen of this in parallel, and sell it.I mean look at this:You know why it works? Because he puts both of them on the same heatsink, right around when the entire house is 32 degrees warm, they self bias themselves, then, maybe we get some output. Variance from circuit to circuit is probably triple digit percentage.

author: Circlotron
author_rank: Super Contributor
timestamp: November 01, 2017, 01:56:28 am
content: Quote from: Audioguru on November 01, 2017, 12:56:40 amI wonder if anyone has ever reviewed hearing aids as to which is the better for hifi listening?

author: BrianHG
author_rank: Super Contributor
timestamp: November 01, 2017, 03:29:42 am
content: In amp, and speaker choice, as for listening and evaluating, unless you have medical hearing loss, many of us can tell the difference, it's just that we don't know how to do so, so, we end up buying what pleases us most.Now, for that truly very few (avoiding audio foolery garbage), since you don't know the what original instruments supposed to sound like, here is a good method to judge, though you require both the amp and speaker in the equation:Pick a properly uncompressed studio recorded song with natural vocals without reverb or echo added. If the vocalist doesn't sound like the are there in front of you,not a vocalist on a mic and speaker, I mean there, a physical human being, without a microphone, no speakers, a presence/essence you can feel in the room with you. At all volume settings. Be honest with yourself here, you should always be comparing what you hear to real life, NEVER compare hardware A to hardware B. Do this right and you picked winning hardware. Affordable solutions to this refinement do exist, however, it takes someone experienced in the business, with access and tests of many hardware and they are not out to rip you off to help you find such quality equipment.Personally, It was recommended to me to purchase a Belles OCM 500 amp. I happy and will never will sell it. Though rated 0.1hz to 20Khz, I measured mine flat to 100KHz at full power. 250w/channel into 8 ohm, safe driving full load into 4 ohm. Some come equipped as mono-blocks with 1kw output peak. It is built like a tank and I don't expect it to ever die.A used one exists here:http://www.canuckaudiomart.com/details/649201738-belles-ocm-500-soloist-amplifier/A FN steal at 600$ cdn.Some finer specs...http://www.diyaudio.com/forums/swap-meet/205903-fs-ocm-500-david-belles-200wpc-amp-beast.htmlDarn, if it weren't for the shipping and that I'm lacking a pre-amp with balanced outputs, (it weighs like a concrete cinder block), I would buy this one and use it with mine as 2x 1Kw mono blocks...

author: Beamin
author_rank: Super Contributor
timestamp: November 01, 2017, 07:47:01 am
content: Quote from: NANDBlog on October 31, 2017, 01:01:49 pmYou need WAY more then just a chip to make good sound. Try driving some 100 watt speakers off a chip. The chip might be able to handle ear buds but not any sized speakers. They keep making amplifiers because you NEED them. How could a single chip handle 100 watts of clean power and still be under its limits to avoid clipping or blowing up for that matter?Unless you are being sarcastic.The reason tubes are still used (in preamps) is because they introduce an overtone that creates a warm sound. So they are actually creating distortion but wanted distortion.I can hear the difference in 0.8THD and 0.03% with decent speakers. I don't think most people could though. I stay away from anything over 0.1% THD because you are not getting the full benefit of your speakers.Now I'm going to finish reading this thread. Just had to comment when I read this.

author: Beamin
author_rank: Super Contributor
timestamp: November 01, 2017, 07:56:27 am
content: Quote from: oldway on October 31, 2017, 02:39:37 pmI have a Marantz amp I use from the 1990's that is 0.03% THD that is ratted at 50 watts. Hook that up to your best speakers then hook up a Sony 100 watt amp you bought at best buy. I can guarantee you there is a difference to be heard. I myself can't tell anything much better then that.What interesting is I herd a super audio CD being played over a 2000$ amp through some $3000 speakers (The ones with a single woofer and that strange grid material for the mid and tweeters at a) HIFI shop. The SACD actually sounded worse and tinny because it reveals the limitations of things. I was very surprised at the result.Do this test:

author: Beamin
author_rank: Super Contributor
timestamp: November 01, 2017, 08:25:13 am
content: Also:I didn't mention speakers because they are not part of my question. And personally I already have good speakers although to get the bass out of them you need a large amp and to mess with the EQ's.I still don't have any real answers as to what makes the amp "better" as in when it is paired with very good speakers what technically makes a Marantz sound so much better then a "Sherwood" or "Jensen" or insert shitty brand Walmart amp here?So I am still leaning towards components with both precise and accurate values as well as an amp design where the signal is well below the "clipping " range with large filter caps.Here is a very easy way to tell if an amp will sound good or not when I worked in an electronics store as a kid: Pick up the amp; is one side or corner of it really heavy? If yes that because it has a great big transformer in it and all the cheap amps feel light. Transformers are full of copper which costs a lot of money. Probably the most expensive part of a cheap amp and probably tied with the cost of transistor pairs with the build cost of an expensive amp.One last question : For the cable that connects my dvd player to the amp through the 47KHz digital connection should I go with the 300$ Kevlar insulation or the 400$ Teflon insulated cable?That was when I first discovered monster cable was full of shit: a 50$ "digital signal low loss high quality "cable for the PCM link. You can literally use two pieces of old telephone wire and get the EXACT same sound as their 50$ cable. A low frequency digital signal makes no difference for a short length or even long length of cable.

author: BrianHG
author_rank: Super Contributor
timestamp: November 01, 2017, 08:32:45 am
content: Price of components does not equate proper sound as noted in the above post.Total THD amp ratings usually are specked out with a dummy 8 ohm resistive load, not an inductive speaker with unpredictable cable length. It is easy possible to get a better sound amp with 0.5% distortion than one with 0.01% distortion. That 0.5% error may just be a little linearity at the extreme voltage swing, but, may be that amp keeps a perfect predictive curve with any mix of source signal frequency and a speaker who's impedance isn't linear over the spectrum with some nasty kick back current spikes from the woofer at a few undesirable l.ow frequencies while your so called 0.01% distortion amp, which give you that figure into an 8 ohm resistor wired right at the rear of the amp with controlled source signals would otherwise oscillate and give a tinny like sound, or damped out low frequencies under real world random source and speaker circumstances.A better designed amp, even with a lower quality distortion rating may better deal with the real world unpredictable circumstances. Don't be fooled by those who say they can hear the difference between a 0.1% distortion amp and a 0.01% distortion amp, it's not the difference in distortion they are hearing. It's all the other characteristic differences between the 2 amp, not a fine linearity signal error.

author: BrianHG
author_rank: Super Contributor
timestamp: November 01, 2017, 08:45:50 am
content: Don't fall for speaker cable BS. Unless you are willing to pay for 99% pure silver cables for the lower impedance, just buy oxygen free 16 gauge household solid core wire & make sure your left and right speaker wire length and impedance match.The only additional + is if you have 2 feeds coming from your amp on each channel to a speaker which has separate inputs for the woofer and mid/tweeter.If you want quality and cheap readily available alternative, use CAT5/6 network cable. For up to 75 watts RMS, at each end tie all the whites together at the - and all of the colored wires together as the +. For up to 200 watts RMS, use 2 network cables, 1 cable, all conductors merged as a + and a second as a -. Twist the 2 network cables around each other generating 1 gigantic twisted pair. You will get excellent results.

author: Zero999
author_rank: Super Contributor
timestamp: November 01, 2017, 09:14:44 am
content: Quote from: BrianHG on November 01, 2017, 08:45:50 amImpedance match? Not necessary for audio cables and differences in length of cable between the channels won't matter.Also I believe oxygen free cable is marketing BS, just use ordinary cable.Quote from: Beamin on November 01, 2017, 07:47:01 amThat is BS. Any amplifier will be able to handle 100W speakers, providing it is designed to drive the impedance of the speakers. For example, if the speakers are 8 Ohm, then a 10W amplifier will be able to drive them, with no problems.You're right about needing more than just a chip to sound good. You need decent speakers, which are much more important than the amplifier.QuoteEasily. A class D amplifier can handle more power than a class AB amplifier, given the same amount of silicon and heatsinking, because it's much more efficient, thus produces less heat. Clipping is dependant on the supply voltage and load impedance. The good thing about class D is the voltage loss in the output stage is lower than class AB, so for the same supply voltage, the output level can be higher, without distortion.QuoteI very much doubt that. With a decent pair of headphones, you stand a chance, but not much, especially at your age.QuoteThat might have been true for older amplifier designs, which still use a big lump of iron to convert the mains voltage to DC, but not in modern designs, which use a switched mode power supply.

author: hamster_nz
author_rank: Super Contributor
timestamp: November 01, 2017, 09:21:48 am
content: Quote from: Beamin on November 01, 2017, 07:47:01 amI know it isn't a single chip but:http://www.altronics.com.au/p/k5181-classic-250w-class-d-audio-amplifier-kit/PCB is about the size of a paperback. (117 x 167mm).From what I can remember about reading the magazine article a few years ago the specs were pretty good - as good or better than a similar rated high quality discrete amp. Somebody here will most likely have the magazine article and can provide numbers.... e.g. 10Hz - 20kHz +/-1dB, THD 0.01%

author: oldway
author_rank: Super Contributor
timestamp: November 01, 2017, 09:47:40 am
content: Most speakers give around 3% distortionmeasured 468-weighted 'distortion residue' reducing slightly at low levels. Electrostatic speakers can have lower harmonic distortion, but suffer higher intermodulation distortion. 3% distortion residue corresponds to 1 or 2% Total harmonic distortion. Professional monitors may maintain modest distortion up to around 110 dB SPL at 1 m, but almost all domestic speaker systems distort badly above 100 dB SPL.https://en.wikipedia.org/wiki/Loudspeaker_measurement#Distortion_measurement

author: BrianHG
author_rank: Super Contributor
timestamp: November 01, 2017, 09:52:01 am
content: Quote from: Hero999 on November 01, 2017, 09:14:44 amHun?Non matched left and right impedance speaker cables. Say you total Left wire is 0.3 Ohm and your Right's total is 0.6 Ohm, with 4 ohm speaker load on each, wont the left be a little louder than the right?I don't buy that.

author: RobK_NL
author_rank: Frequent Contributor
timestamp: November 01, 2017, 10:06:19 am
content: Quote from: BrianHG on November 01, 2017, 09:52:01 amThat 0.3 Ohm difference equates to 10 meters of 1.5mm2cable. That's more than what most people would have in total.

author: Zero999
author_rank: Super Contributor
timestamp: November 01, 2017, 10:11:27 am
content: Quote from: BrianHG on November 01, 2017, 09:52:01 amI doubt you'll be able to tell the difference between a loss of 7% and 13%, on the left vs right channels, although I'd aim for lower losses than that.Quote from: RobK_NL on November 01, 2017, 10:06:19 amTrue. If it's an issue. The answer is thicker cable.

author: BrianHG
author_rank: Super Contributor
timestamp: November 01, 2017, 10:53:13 am
content: Quote from: Hero999 on November 01, 2017, 10:11:27 amWith ease. Even without a blindfold. Maybe even with 1 ear covered.As for aiming for lower loss, yes, 7% & 13% is what should be an impossible extreme.

author: oldway
author_rank: Super Contributor
timestamp: November 01, 2017, 11:14:13 am
content: And the balance setting, what is it for?

author: rdl
author_rank: Super Contributor
timestamp: November 01, 2017, 11:36:05 am
content: I you go look at high end audio equipment, everything - not just amps, has a lot of money and effort put in to making it look good. I wouldn't be surprised if more is spent on looks than on parts.

author: Zero999
author_rank: Super Contributor
timestamp: November 01, 2017, 11:37:24 am
content: Quote from: BrianHG on November 01, 2017, 10:53:13 amI doubt it. The ratios between the signal seen at the amplifier and speaker are: RCABLE= 0.6R = 4/4.6 = 0.8696, RCABLE= 0.3R 4/4.3 = 0.93, which may seem significant, but human hearing is logarithmic, so it makes more sense to look at it in decibels: log10(0.8696) = -0.061dB, log10(0.93) = -0.0315dB, a difference of just 0.03dB. The difference in, audio levels on each track, the sensitivity between the left and right ear, the position of the listener in the room and acoustics will easily exceed a 0.03dB difference in attenuation between channels, caused by the cable.

author: oldway
author_rank: Super Contributor
timestamp: November 01, 2017, 11:59:02 am
content: An audio amplifier is neither more neither less than equivalent to an operational power amplifierNo secrets, nor magic here ....If a power amp sounds better than another, there must be a parameter of the amplifier that causes it.What is this parameter?If there is a difference between two audio power amplifiers, this difference must be measurable because the measuring instruments are more accurate, more sensitive and more reliable than the hearing.I agree that there are certain differences with regard to preamplifiers: tonal corrections for example ... (bass + trebble or bass + midrange + trebble, correction Db, inflection points, signal / noise ratio , ...)This topic is full of non technical audiofoolery and of arbitrary and subjective considerations.

author: Seekonk
author_rank: Super Contributor
timestamp: November 01, 2017, 12:41:28 pm
content: I have a pair of Magnaplaner speakers. When I still had good hearing, there was no noticeable difference between a really cheap amp and an expensive one using them. The Magnaplaner speakers are almost pure resistance. Few amps handle inductance well.

author: oldway
author_rank: Super Contributor
timestamp: November 01, 2017, 01:20:46 pm
content: Few amps handle inductance well.Speakers have a low value inductance (low value because there is a gap between the coil and the magnet) in serie with a large value resistance. They are mostly resistive.

author: Seekonk
author_rank: Super Contributor
timestamp: November 01, 2017, 01:37:40 pm
content: Perhaps I should have said motor, same thing. Anyway, speakers are not resistors.

author: oldway
author_rank: Super Contributor
timestamp: November 01, 2017, 01:46:32 pm
content: Few amps handle inductance well.What do you means with this ? Can't "handle" a complex load like a speaker ? An audio amplifier is designed to "handle" a load like a speaker....A transistor amplifier has a voltage feedback, so the kind of load does not matter.What's the problems ? Instability ? Oscillations ? Never measure anything like this ?

author: coppice
author_rank: Super Contributor
timestamp: November 01, 2017, 04:25:31 pm
content: Quote from: oldway on November 01, 2017, 01:46:32 pmInstability is a common issue with power amps and complex loads. It doesn't just affect speaker amps. Even little headphones amps frequently have problems. Plug one pair of headphones in and they sound great. Plug a different model of headphones in and it sounds like someone blowing raspberries. Plug those headphones into a different model of amp and they sound great. This issue is very common with notebook computers. A notebook might not be the first thing you'd go to for super hi-fi results, but the sound some notebook + headphone combinations produce is really awful.

author: IanMacdonald
author_rank: Frequent Contributor
timestamp: November 01, 2017, 04:56:46 pm
content: There is no question that amplifiers do sound different. One of the most offputting kinds of sound artifact is sibilance on vocals, and whilst you might think it results from too much snake oil being put in, it actually results from a tendency to parasitic oscillations when excited at a critical frequency. It is hard to measure such things on the test bench, and they don't show up in specifications.It is often described as the sound being 'too bright' and is a reason for preferring valves over solid state.I don't think this is a failing of transistors themselves, rather it is a side effect of specmanship -Designing amps that have the best possible paper specs at the expense of poor HF stability.Quad equipment is noted for its smooth sound, and I think one reason is that they don't try to achieve silly HF responses. (22kHz is the rolloff IIRC)

author: oldway
author_rank: Super Contributor
timestamp: November 01, 2017, 04:58:16 pm
content: Quote from: coppice on November 01, 2017, 04:25:31 pmGive me some examples of commercial audio amplifiers with transistors that are unstable when connected to speakers .... I'm curious, I've never met .... tube amplifiers, okay, they are inherently unstable because of the phase shift of the transformer, but the amplifiers with transistors, no.

author: oldway
author_rank: Super Contributor
timestamp: November 01, 2017, 05:14:37 pm
content: Quote from: IanMacdonald on November 01, 2017, 04:56:46 pmThis is another subject: the difference between a tube amplifier and a transistor amplifier.A transistor amplifier behaves like a voltage source. Because of its high feedback rate, it maintains its output voltage equal to that of the input signal.It does not control the current that passes through the speakers, it imposes the voltage on their terminals.We all know that if we apply for example a square wave to a partially inductive load, the current will not be a square wave.But the displacement of the membrane tends to follow the current's value.There is therefore a weakness of the transistor amplifier on this point.The tube amplifier behaves more like a current source, in particular because of its low rate of feedback and its high internal impedance. (low damping factor)It is an essential difference between these two technologies that explains the difference in sound between the two and it is perfectly measurable, by measuring the internal resistance of the amplifier and its damping factor.

author: Kleinstein
author_rank: Super Contributor
timestamp: November 01, 2017, 05:25:38 pm
content: Usually audio amps, like other low output impedance amps have no problem with inductive load. They can have a problem with a capacitive load. This especially true for amps build with super high feedback to get super low distortion up to frequencies in the 100 kHz range. Modern amps should handle all load swell, there is no real magic in getting this. A usual element to make it happen is an RL filter at the output.An inductive load might become a problem when the amplifier is going into clipping or current / SOA limiting. At that level sound is terrible and usually way to loud anyway. I poor designs this might lead to damage to the amplifier.With conventional speakers it is the cross over circuit that can make the impedance look odd, inductive and capacitive depending on the frequency. In a few nasty cases the load might also be nonlinear from inductors going close to saturation - but that would a fault of the speakers.The one point where amps can be slightly different is output impedance - usually it should be low, as a low impedance source can dampen resonances of he speakers and tolerate a nonlinear load better. So a good amplifier should be at low impedance end.However with passive crossover, the crossover circuit will usually dominate the impedance seem by the speaker drivers. To get the low impedance really to the separate speakers it takes active crossover - at least it is much easier that way. Some of the speakers / cross over networks (especially older ones) may be made to work better with a higher impedance amps. So a high output impedance and thus not really good amplifier might sound better.This is especially true when comparing with tube amps - they are generally high output impedance and thus are made for different sets of speakers. So the speakers should be separated in those made for tube amps and those for transistor amp - the wrong combination is not a good idea.

author: oldway
author_rank: Super Contributor
timestamp: November 01, 2017, 06:23:31 pm
content: I agree, I would add that the more the cross over is simple, the better it is.What I like most is the 2-way system, a 12-inch dual cone speaker for bass and midrange and one or more tweeters for trebles.The crossover is a simple unpolarized capacitor to protect the tweeter (s) against low frequency.

author: Fungus
author_rank: Super Contributor
timestamp: November 01, 2017, 06:41:06 pm
content: Quote from: NANDBlog on October 31, 2017, 01:01:49 pmYep. Class D has come on in leaps and bounds over the last decade. All the old class A and AB stuff is pretty much obsolete.Modern MOSFETs mean class D amps can be amazingly efficient. Tiny sub-$10 chips can power big powerful systems with almost zero THD.eg.http://www.ti.com/product/TPA3255/datasheetSo yeah, amplifier problem is solved. Look at speakers and room acoustics instead.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 01, 2017, 06:47:41 pm
content: After reading this thread I would like to ask - how many of "Oh it is so easy, Class D makes old technology obsolete, everything was solved 50 years ago" commentators did actually design a decent hi-fi amplifier for production?CheersAlex

author: Zero999
author_rank: Super Contributor
timestamp: November 01, 2017, 06:50:58 pm
content: Quote from: oldway on November 01, 2017, 05:14:37 pmAre you sure that the position of the speaker's cone is proportional to the current, rather than the voltage?I thought, it's better to use a voltage source to drive the speakers, rather than a current source, because it damps the resonance. It's easy to make a solid state amplifier with a constant current output. I've never made one and tested it on real speakers though.

author: oldway
author_rank: Super Contributor
timestamp: November 01, 2017, 07:15:45 pm
content: I dare not venture into the subject "speakers" because I am not a specialist in this matter.But I find that many of my customers prefer the sound of a tube amplifier rather than transistors.I think our ears may not prefer the most perfect sound ....Looking to reproduce a perfect sound is perhaps not the best solution.A transistor amplifier actually tends to force the speaker diaphragm to follow the signal.But perhaps that makes the sound of a transistor amplifier drier, colder than that of a tube amp.With a tube amp, the membrane is freer, it can vibrate according to its own resonances ...Where is the right solution in all this? I do not know but I find that many of my customers prefer the sound "Tubes".NB: see the Philips Motional Feedback Speakers.....http://hifipig.com/philips-motional-feedback-speakers/Also modifications of transistor amplifiers for current drive of speakers.https://www.current-drive.info/9QuoteToday, all commercially available audio amplifier and loudspeaker equipment works on voltage drive principle without significant exceptions. This means that the amplifier acts as a voltage source and therefore exhibits low output impedance. Thus, the amplifier as though forces the voltage across the load terminals to follow the applied signal without any regard to what the current through the load will be.However, both technical aspects and listening experiences plainly indicate thatvoltage drive is the wrong choiceif sound quality is of any importance. The fundamental reason is that the vague electromotive forces (EMF), that are generated by both the motion of the voice coil and its inductance, seriously impair the critical voltage-to-current conversion, that in the voltage drive principle is left as the job of the loudspeaker.https://www.current-drive.info/6

author: Fungus
author_rank: Super Contributor
timestamp: November 01, 2017, 07:40:15 pm
content: Quote from: Alex Nikitin on November 01, 2017, 06:47:41 pmI never said it was "easy", I said it was "solved".Whatever your feelings are about class D, this isn't the 1970s. Awesome parts are cheap and readily available. Reference designs abound. There's no need to "design" anything.

author: Johnny10
author_rank: Frequent Contributor
timestamp: November 01, 2017, 07:58:37 pm
content: I think it all goes back to the OP Beamin...He is an instigator !

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 01, 2017, 08:41:37 pm
content: Quote from: Fungus on November 01, 2017, 07:40:15 pmWell, if you are OK with a crap sound, there is indeed no need. Fortunately for manufacturers of that crap the majority of consumers don't give a #**! about the sound quality. Especially as it is "scientifically proven" that it does not matter. And so it goes. I am quite happy to be in a minority who cares about quality sound.CheersAlex

author: Circlotron
author_rank: Super Contributor
timestamp: November 01, 2017, 09:26:58 pm
content: Quote from: Hero999 on November 01, 2017, 06:50:58 pmAfter experimenting with this a bit on and off over the years I came to the conclusion that the voice coil *velocity* is proportional to the instantaneous signal voltage, and the voice coil *force* is proportional to the current. Voice coil instantaneous position is indeterminate because of suspension compliance, acoustic loading etc. My opinion is that the ideal setup would be current drive to the voice coil and motional feedback. Kind of similar to current mode control in a SMPS - it completely takes an unknown voice coil position out of the loop.

author: Circlotron
author_rank: Super Contributor
timestamp: November 01, 2017, 09:32:10 pm
content: And what complicates this further is if a dynamic mic was used in the original recording then peak signal voltage occurs at the mechanical zero crossing of the mic diaphragm whereas if a condenser mic was used then peak signal voltage occurs at max diaphragm excursion.

author: AndyC_772
author_rank: Super Contributor
timestamp: November 01, 2017, 09:50:01 pm
content: Quote from: BrianHG on November 01, 2017, 03:29:42 amI don't understand that whole way of thinking when it comes to hi-fi.My audio equipment exists for one reason, and one reason alone: to make sound that pleases me.It's not a scientific instrument. I've no requirement for the sound in my room to be objectively, quantifiably, as similar to some hypothetical "original" sound as possible. I just want to enjoy listening to it.The "original" sound is a total fallacy anyway. Unless the recorded content happens to be a live performance, it's probably a construction fabricated in a studio with the objective of sounding pleasing to the ear when reproduced through (usually fairly crappy) equipment.QuoteI don'tcare. Big difference!Last year I treated myself to a new system. I figured I'd reached the time in my life when I'm old enough to be able to afford the hi-fi I've always wanted, but not so old that my ears are worn out and can't appreciate it.I spent ages in various hi-fi shops, with a variety of familiar music, and came away with strong opinions of most of the kit I heard. The differences were anything but subtle; speakers in particular each have their own character, and surprisingly few had me wanting to go back for another listen.I don't doubt that they'd all measure well, but that's not what I'm spending good money for. I want to enjoy the music. Nothing else!In the end I bought a system from Arcam and Spendor, plus an exceptionally capable subwoofer from a small Norwegian outfit called Arendal Sound. The amp contains a DSP which can analyse the room characteristics and pre-process the signal to correct both the frequency and impulse responses at the listening position. For movies it's great, though I prefer the sound with it off for music.

author: Buriedcode
author_rank: Super Contributor
timestamp: November 01, 2017, 10:12:24 pm
content: I do love these threads. I'll admit, I have never designed a commercial amplifier, nor do I own any expensive amps, speakers or systems, I'm quite happy with my phone, my PC, or a raspberry-pi-based-portable player.But I follow these threads to see a whole range of views and nuggets of info about analogue design, only to see it (almost...) descend into a pissing match about who has the better ears, who has the best gear. What makes a 'high end' amp better than a 'low end' ? Depends on how you define "high end" and "low end". Depends on Power, features, reliability, cost, who you're trying to impress, how justified you feel in spending the cost of the amp, the general volume one listens to music... the brand name, warranty. In fact actual 'sound' must be way down the list..

author: Audioguru
author_rank: Super Contributor
timestamp: November 01, 2017, 11:43:05 pm
content: Goodamplifiershave a very low output impedance for a good damping factor. Then the resistance of the cable between the amp and the speaker also must be low. Then the speaker sounds "tight" and follows the voltage output from the amplifier.A poor damping factor results in the speaker resonating something like a one note bongo drum that sounds bad.Nobody mentioned the power supply? A cheap power supply has resistance that allows an amplifier to produce high momentary low distortion power called "music power" before its voltage and power output sag. When the voltage and power sag then the amplifier clips high output peaks that sounds bad.

author: BrianHG
author_rank: Super Contributor
timestamp: November 02, 2017, 12:12:33 am
content: Quote from: oldway on November 01, 2017, 11:14:13 amI don't have a balance setting. It's a stupid idea unless you are deliberately trying to correct for junky setup where some of the cheapest equipment today should be beyond the need of such correction.

author: Cliff Matthews
author_rank: Unknown
timestamp: November 02, 2017, 01:08:20 am
content: Quote from: BrianHG on November 02, 2017, 12:12:33 amMy NAD 3020 has no balance either. I've had it ~35 years and re-capped it once. Just some 2N3055's and MJ2955's designed by some Swedish guy and fabbed in Korea in believe.. FWIW, I mentioned Nelson Pass because a high-end shop nearby talks of him but sells Mark Levinson products looking more like military tanks than amps.. funny world this.

author: DrGeoff
author_rank: Frequent Contributor
timestamp: November 02, 2017, 01:18:49 am
content: Linearity, low noise and a stable power supply.

author: innkeeper
author_rank: Unknown
timestamp: November 02, 2017, 01:33:47 am
content: everything is great on paper until you throw a reactive load in the mix with varying rectance at various frequencies and varying phase shifts and room responses.how an amp reacts is as varied as the number of amp and speaker combinations you can put together.Gotta pitty people who is chasing that unreachable goal of the ultimate realistic sound. and even if they find it, it will be the ultimate sound only to him/her.it is a huge rabbit hole. and there is no bottom. and why there will always be room for one more power amplifierone thing i've seen to moving in a direction for a somewhat modern approach is individually amplified drivers that are compensated for via a dsp and room monitoring.shit gets crazy!

author: Fungus
author_rank: Super Contributor
timestamp: November 02, 2017, 07:04:21 am
content: Quote from: innkeeper on November 02, 2017, 01:33:47 amYep. This is more likely to be the future (IMHO) than spending a lot of money on amplifiers.Quote from: innkeeper on November 02, 2017, 01:33:47 amFor this reason.Quote from: Alex Nikitin on November 01, 2017, 08:41:37 pmI'm sure there's a point in there somewhere but no actual facts were presented, it's all Ad Hominem.Facts:Is there a massive sound difference between a high-end 1970's amp and a high-end 2017 amp? No, not really.Are all electronic parts better/cheaper/smaller/easier-to-get today than in the 1970s? Yes.Massivelyso.Nobody's saying you can build an amp for $10 (which is what will be in a consumer grade HiFi) but the days of exotic $10,000+ amplifiers are long gone.

author: IanMacdonald
author_rank: Frequent Contributor
timestamp: November 02, 2017, 09:13:05 am
content: Quote from: oldway on November 01, 2017, 05:14:37 pmEither type could be made to give a constant current output. The problem with doing so is what happens at the speaker's resonant frequency. At that point both efficiency and impedance increase drastically. With constant voltage drive those effects cancel out to a large extent. With constant current drive the delivered power would actually increase as well as the efficiency, leading to a massive 'bass boom' problem.

author: oldway
author_rank: Super Contributor
timestamp: November 02, 2017, 11:21:01 am
content: It is exactly on this principle (massive bass boom) that loudspeakers type bass reflex are based.They are tuned to the resonance frequency to boost these bass and recover them to add to the sound from the front of the speaker membrane.C. A. Briggs, in his book "The Loudspeakers" recommends the plane baffles rather than the bass reflex.Ideally, the plane baffle should have an infinite area, which is of course impossible .... It recommends to mount the speaker (s) in a wall, it is what looks the most like an infinite plan baffle.If a closed loudspeaker is used, it must have the largest possible volume.Regarding the resonance frequency of the speaker, it is attenuated in a closed speaker by the absorbent material that is inside .... In addition, there are many electronic means to suppress resonance (filters, graphic equalizer), and the sensitivity of the ear is reduced in the very low frequencies ....There are no reasons to avoid current drive of the speakers.....

author: Fungus
author_rank: Super Contributor
timestamp: November 02, 2017, 01:56:17 pm
content: Quote from: oldway on November 02, 2017, 11:21:01 amThey're also usually made of chipboard/mdf instead of fancy wood.It's not the manufacturers being cheapskates, those "woods" resonate less than the real thing.

author: David Hess
author_rank: Super Contributor
timestamp: November 02, 2017, 02:11:04 pm
content: Quote from: Beamin on October 31, 2017, 12:14:15 pmUltimately I think amplifier quality is in most cases just another case of marketing triumphing over engineering. But there are lots of engineering factors which affect quality.1. How much was it derated for operating at its worst case output power? This impacts reliability and operating life. Commercial designs tend to be much better than consumer designs in this respect which is expected.2. Is it a simple or complex class-AB design? At high frequencies the former might produce 0.1% THD while the later produces 0.001% or better. I am always struck by how simple class-AB designs literally duplicate the simplified schematic for an early 3 stage operational amplifier.3. How does it perform (or fail) under adverse operating conditions like a 2 ohms load or output short?4. How well does it recover from clipping? Tube amplifiers have a natural advantage here do to their softer saturation characteristics but some solid state designs are truly horrible in this regard for reasons having nothing to do with saturation.5. Does it protect the speakers against amplifier failure?Bob Cordell's book Designing Audio Power Amplifiers gives a great overview of simple versus complex designs and the attention to detail necessary for a high quality design.The thing which puzzles me is why poor designs are produced when most of the cost is in the output transistors and power supply that could be the same in a low or high quality design. This might simply be do to diminishing returns, poor source material, and low customer expectations.Quote from: macboy on October 31, 2017, 01:35:10 pmIn the US there has been since 1974:https://www.ftc.gov/enforcement/rules/rulemaking-regulatory-reform-proceedings/amplifier-ruleQuote from: NANDBlog on October 31, 2017, 01:01:49 pmThat level of distortion is typical for a good but simple class-AB design.Quote from: Fungus on November 01, 2017, 06:41:06 pmI notice that TI leaves intermodulation out of their datasheet and how does it respond to clipping? The other specifications do not strike me as particularly outstanding compared to a complex class-AB design. The efficiency is nice of course.I agree that the amplifier problem is solved but only in the sense that the speaker problem is solved. In both cases most consumer examples are mediocre but with theloudness wars, that is good enough.Quote from: oldway on November 02, 2017, 11:21:01 amIf they are tuned properly, then the result is not "massive bass boom" but extended low frequency response, a second order low frequency roll off, and lower impedance at resonance. The last is important because minimizing the cone movement minimizes intermodulation distortion which otherwise makes deep base sound boomy and indistinct. Some people of course like that sound but I place them into the same class of people who like the loud hiss from harmonic distortion, indistinct vocals, and excessive compression from the loudness wars.In a practical setting, the lower intermodulation distortion means being able to turn the volume up to shake the room and contents without sounding loud.Quote from: Fungus on November 02, 2017, 01:56:17 pmBut they are rarely heavy enough. On the other hand, most consumers would prefer speakers they can lift easily. I am somewhat amused when I find old bookshelf speakers which weight more than modern but cheap floor speakers.

author: Beamin
author_rank: Super Contributor
timestamp: November 03, 2017, 05:45:48 am
content: Quote from: Hero999 on November 01, 2017, 09:14:44 am

author: Zero999
author_rank: Super Contributor
timestamp: November 03, 2017, 09:44:41 am
content: Quote from: Beamin on November 03, 2017, 05:45:48 amI don't know your exact age but, going from your other posts, I know you're not young, therefore you will have some age-related hearing loss, especially at high frequencies.Did you subject yourself to a double blind listing test? I highly doubt it. Cognitive bias, aesthetics, size, cost, can easily explain differences in perceived sound quality. No one is immune to this. The only way to be sure one can tell the difference is a double blind test, repeated enough times.Headphones generally sound better than speakers, even cheap headphones vs high end speakers, because the ear is in the near field region, rather than the far field so it's easier to get a flat frequency response and reflections and absorption due to room acoustics are no longer factors.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 03, 2017, 09:51:26 am
content: Quote from: Hero999 on November 03, 2017, 09:44:41 amIt should read:The only way to be sureno one can tell the differenceis a double blind test, repeated enough times. Blind tests is the biggest con in audio, forget all these poor snake oil cable purveyors .CheersAlex

author: Zero999
author_rank: Super Contributor
timestamp: November 03, 2017, 10:03:40 am
content: Quote from: Alex Nikitin on November 03, 2017, 09:51:26 amWhat are you saying?Double blind tests are how it's done in medicine and other fields. They are the only way to prove there is an actual detectable difference, in something, other than cognitive bias and placebo!Oh sarcasm doesn't always get through the interweb very well. Yes, mustn't let science get in the way of these poor people.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 03, 2017, 11:23:00 am
content: Quote from: Hero999 on November 03, 2017, 10:03:40 amIf you would like my opinion on blind tests in audio, have a look at my messages inthat thread. I don't like repeating what I've already expressed elsewhere.CheersAlex

author: Beamin
author_rank: Super Contributor
timestamp: November 03, 2017, 12:18:33 pm
content: Quote from: Hero999 on November 03, 2017, 09:44:41 amYes next time I go to the speaker store I will hire scientists do set up a double blind placebo test with psychologists and statisticians to make sure the scientists have ruled out any hidden variables. This way it will ensure that my subjective taste in a 100% subjective matter is 100% sterilized and devoid of any emotional bias. I might end up getting a set that sound horrible to me but at least they were scientifically proven to be better. I will also have them build an anechoic chamber that is nothing like the room I will be using them in. Of course before I do any of this I will have an audiologist test my hearing to find out what I'm sensitive to when analyze the data from the acoustic scientists also to rule out hidden bias.Do you ever think "Hey what I just wrote was completely ridiculous and would make no sense to the average ( a technically minded but practical) person reading it and is also TOTALLY impractical and would never happen".Double blind placebo tests to go speaker shopping? Really? REALLY?Totally ridiculous is an understatement and you are kind of embarrassing yourself especially if you start making replies doubling down on why your post is not ridiculous..You buy what sounds good to you even if you are victim of audio phoolerly they still are what sounds good and you enjoy it.

author: coppice
author_rank: Super Contributor
timestamp: November 03, 2017, 12:22:04 pm
content: Quote from: Alex Nikitin on November 03, 2017, 11:23:00 amIf you want to argue that many double blind tests are unreasonable, because they are not well constructed to highlight the things they are supposed to be looking for, I'm sure you will get a lot of agreement from others here. If you are arguing that some dodgy double blind tests mean double blind tests are not a good thing (which is what I take from what you have written), then you aren't going to get much positive feedback from a science/engineering literate audience.If what you actually mean is much better double blind tests are needed, then highlight the weaknesses in current methods, and how things could be improved.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 03, 2017, 12:43:23 pm
content: Quote from: coppice on November 03, 2017, 12:22:04 pmFor all practical purposes in Hi-Fi audio double blind tests are a complete waste of time (and that is what science should be telling us). Somewhere, at the end of a rainbow, a useful audio DBT (very expensive and extremely difficult to set up, no doubt) may exist, however I am not wasting my time looking for that unicorn either. I wish I could have tests and methods which would be useful for a valid definition of sound quality, but so far I've discovered none which could give better results than a simple subjective listening test with all its drawbacks (as much as I hate to admit it, as an engineer). Yes, there is always a good chance to fool yourself, but at least there is also a good chance to do your job properly and make audio equipment that is enjoyable to listen to.CheersAlex

author: xani
author_rank: Frequent Contributor
timestamp: November 03, 2017, 01:00:20 pm
content: Quote from: Hero999 on November 03, 2017, 10:03:40 amHe's our resident audiophool, just ignore him.

author: Gyro
author_rank: Super Contributor
timestamp: November 03, 2017, 01:07:55 pm
content: On the contrary, he is a experienced engineer and respected contributor to this forum.

author: Zero999
author_rank: Super Contributor
timestamp: November 03, 2017, 01:48:37 pm
content: Quote from: Beamin on November 03, 2017, 12:18:33 pmThere is no need for that attitude.None of what you've said is scientific. It's all highly subjective. You can't be sure that you can actually tell the difference between 0.8% and 0.03% THD, because you've not proven you can.I thought we were talking about amplifiers not speakers? Suppose two amplifiers, set to exactly the same volume, playing the same music, were connected to a pair of speakers, via a relay, to switch between the two amplifiers, controlled randomly by a computer. If you can tell which amplifier is driving the speakers, significantly beyond 50% of the time, then you've proven that the two amplifiers actually sound different to you. Whether or not you prefer the one with the better figures, is up to you.Obviously you can't perform the test mentioned above in a Hi-Fi store and I never suggested it was, which was the point I was trying to make. You have many other factors, other than sound waves from the speakers, which influence your judgement, whether you like it or not. Some of these factors are important, when considering a purchase: aesthetics, ergonomics, build quality etc. but if you think you're basing your decision on sound quality, then you're kidding yourself. Someone else posted a very good video on the subject, but I can't find it.Fortunately, modern, well-designed solid state amplifiers, are audibly transparent, i.e. no one will be able to tell the difference between them, in the above scenario, as long as they're not overdriven.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 03, 2017, 02:03:04 pm
content: Quote from: Hero999 on November 03, 2017, 01:48:37 pm"you're kidding yourself".CheersAlex

author: nfmax
author_rank: Super Contributor
timestamp: November 03, 2017, 02:12:11 pm
content: Back in 1978...http://www.keith-snook.info/wireless-world-magazine/Wireless-World-1978/Valves%20versus%20Transistors%20DCD.pdfNone of which negates the audiophile'ssubjectivejudgment that the music sounds better when played through a system of which he (and for some reason, it almost always is a he) approves. Whether it's the class A, the no feedback, the polished mahogany case, the oxygen free copper or the warm feeling that comes from being able to afford it in the first place, itreally doessound better! High-end audio is a fashion business: provided some minimum standard of technical performance is met (and sometimes not even then) the numbers don't really count.And the enjoyment of music is, after all, a subjective thing.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 03, 2017, 02:21:21 pm
content: Quote from: nfmax on November 03, 2017, 02:12:11 pmWell, a perfect example of a wasted effort. The whole idea that the test like this is valid is not scientific. Unless we validate the test with a statistically correct positive result (i.e. the difference would be established which can be reliably perceived by the audience in the particular conditions of this test), the negative result means nothing and can not be used as an argument.CheersAlex

author: nfmax
author_rank: Super Contributor
timestamp: November 03, 2017, 02:45:36 pm
content: Quote from: Alex Nikitin on November 03, 2017, 02:21:21 pmAre you really saying that unless we can prove the positive result we cannot prove the negative result? Or do you mean that unless we can introduce some deliberate modification into one of the amplifiers under test thatisdetectable in a double-blind test, we have not established the validity of the double-blind testas a test?In that case, it's easy: just increase the signal levels to the point where one of the amplifiers starts clipping, or going into current overload. Then you will hear differences a-plenty!Disclaimer: A long time ago, I used to work at QUAD, as the junior lab engineer. My one claim to fame is developing the revised current limit circuit that turned the 405 power amplifier into the 405-2.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 03, 2017, 02:52:49 pm
content: Quote from: nfmax on November 03, 2017, 02:45:36 pmThat is the very reason why it is never done - it would show how insensitive the test of this kind is. If we are trying to spot the differences near the limit of our perception but the test conditions make it possible to spot only really gross differences, the test is not good for the intended use. Same if you try to measure a difference between two 9V batteries with a needle meter at 1000V range. That is why I only can regard this kind of DBT in audio as pseudo scientific. And so many people actually believe that crap and accept it as a "scientific proof"!CheersAlex

author: Zero999
author_rank: Super Contributor
timestamp: November 03, 2017, 03:30:38 pm
content: Quote from: nfmax on November 03, 2017, 02:45:36 pmThe test seems to be perfectly valid. I don't see how anyone from a scientific background can say otherwise. Arguments against this kind of test seem to be very similar to advocates of spiritual healing vs tried and tested medical treatments.Providing the amplifiers in the test were operated within their ratings, they were indistinguishable. No doubt one would get a similar result with modern audio amplifiers. Of course how the amplifier responds to overloads is also a valid reason for why some prefer value/tube amplifiers over solid state amplifiers, but that wasn't the purpose of the test, which was to determine if solid state amplifiers could be made to sound like valve amplifiers.Perhaps another test could be done, where distortion is deliberately introduced up till the point, where it becomes noticeable beyond chance?Another interesting test would be a digital processing system, designed to emulate the distortion of a valve amplifier, vs the real valve amplifier.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 03, 2017, 04:12:44 pm
content: Quote from: Hero999 on November 03, 2017, 03:30:38 pmOh, a strictly necessary requirement to establish a base line of a test sensitivity (otherwise how we know what we testing) now suddenly becomes something mystical?!CheersAlex

author: Zero999
author_rank: Super Contributor
timestamp: November 03, 2017, 04:34:19 pm
content: Quote from: Alex Nikitin on November 03, 2017, 04:12:44 pmWhy is that necessary?The test proves that human ear is insensitive to any differences between the amplifiers used in the test. Therefore if one is deciding which amplifier they want to use, they should focus on other things than perceived sound quality, such as: cost, reliability, availability of spare parts, response to overload etc.I agree, it would be interesting to know the point at which distortion becomes noticeable and whether or not it's pleasing to the listener, but that was beyond the scope of the test.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 03, 2017, 05:04:41 pm
content: Quote from: Hero999 on November 03, 2017, 04:34:19 pmNo, it proves nothing of the sort, sorry. That is a delusion, which is convenient to believe, that is all.In medical trials there are objective criteria which can be used to validate the results - various tests. DBTs in medicine are important and valid because the objective measurements and tests can be influenced by the placebo (i.e. psycho physiological) effects. But it is theobjective resultwhich matters. In audio DBT that concept is turned on its head - we are trying to use our conscioussubjectiveperception to validate the differences which may influence us mostly on a subconscious level. And that is one (but not the only one) reason why audio DBT are extremely insensitive for what they try to achieve. And obviously it is very convenient for someone who wants a particular result to use a test which gives that result no matter what.CheersAlex

author: oldway
author_rank: Super Contributor
timestamp: November 03, 2017, 07:06:29 pm
content: Nothing is more subjective and arbitrary than the appreciation of our hearing.I would like to know the results of a test (which are the amplifiers that sound best ... give a rating of 0 to 10) made by a hundred of confirmed audiophiles (a hundred for the results to be statistically valid) who would give their opinion on a dozen different amplifiers they would only know by a number.This test would take place in the same room, with the same sound source and with the same speakers. (and, of course, at the same power level)The amplifiers would be in the "flat" position of the tone controls, no activated filters, and no loudness.Would the opinions of these "specialists" be he same ?In my opinion, they will certainly make a distinction between tube amplifiers (current driven loudspeakers) and transistor amplifiers (voltage driven loudspeakers), but that's all, no difference between high end transistor amplifiers and low end transistor amplifiers...

author: Fungus
author_rank: Super Contributor
timestamp: November 03, 2017, 07:51:50 pm
content: Quote from: Alex Nikitin on November 03, 2017, 04:12:44 pmWhat on earth are you on about?According to you,allHiFi systems sound the same, there's no possible way to classify them as "good" or "bad".This is obviously, facepalmingly false.Or, if you accept that not all HiFis sound the same then AB testsmustbe possible and you're contradicting yourself.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 03, 2017, 08:54:32 pm
content: Quote from: Fungus on November 03, 2017, 07:51:50 pmNowhere I've said anything of that kind... .CheersAlex

author: Beamin
author_rank: Super Contributor
timestamp: November 03, 2017, 09:20:13 pm
content: I kind of enjoy it when these threads get derailed I started the thread purposely with as little chance of this as possible but I am no closer to getting ideas on this question as when I asked it. Does anyone "hear" really know? Because if they don't the back and fourth is not valid; based on the fact that no one knows the answer to the technical difference that would make the sound difference. There is no "sound" argument to be made.If someone did say "this" part/design makes "this" better then started saying why amps do/don't make a difference they would have a base to make their argument. At this point I see a lot of factless opinions and arguments based on really nothing. I guess this is why audio phoolerly is such a $$$ business. Even the technical people fall victim to it.

author: Beamin
author_rank: Super Contributor
timestamp: November 03, 2017, 09:21:19 pm
content: Quote from: Hero999 on November 03, 2017, 01:48:37 pmApologies if I sounded crass. I get sensitized to the disconnect that people have to the real world on tech forums and erroneously grouped you in with them.But seriously get a pair of decent speakers (not best buy brands mass produced Chinese import crap) and compare a Sony amp from best buy to a "good quality" amp from a hifi shop. In my case I used a Marantz I own and a 199$ Sony I borrowed from a friends living room. Maybe it wasn't the THD that I noticed; being that number is so small it could have been many other things but on the contrary that number is an order of magnitude difference between amps so it could be a contributor. I know audiophoolery is alive and well I am biased in that I consider myself educated in the issue and would be upset to learn I paid a lot more for something that was the same.Points I considered that "Biased" my judgement:Opening the amps up and seeing physically larger value parts such as caps and transformers. These parts are not cheap and add to the build cost substantially. If manufactures know their target audience will be overly critical of the sound quality they have to look out for this. Also when these factors are deep within the case and not advertised there is a reason why they put the extra money in other wise they are putting money into something that's not going to make a difference; which is a bad idea from a business point of view.I hate getting "phooled" it would take a bit to prove to me I paid extra for the same performance but unlike most people I am very flexible and with sufficient evidence I do change my point of view. I rarely double down on things when there is evidence to the contrary. This is rather a reason to question your views.Working in the business of audio when I was younger I found myself getting caught up in the BS they program into your head to make you sell more. Why teach a person to sell when you can sell them on the product? Then their recommendations come natural when they are selling to the customer and the worst sales person becomes an effective tool. Work for corporate America and you will notice the managers are always selling the employees on why their product is better if not best, even when it is shit. "We sell on value not price." Is code for we are ripping off the customer and charging more then our competition.So I think there is a difference. I can hear it. And if I can hear it, it exists to me. And that's all that matters at the end of the day.Also I don't like when people make generalizations about my age or other personal factors that I haven't openly revealed on the internet. So yes that triggered me into an overly aggressive response that wasn't warranted. Arrogantly I consider myself a bit of an enigma after hearing people say this to me. I rarely fit into a pattern or box and if you have me figured out you don't. I don't have me figured out. I have done so many 180's in life tht even my family realizes that anything is possible with me because I am so open to change. just my 0.02$

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 03, 2017, 09:59:38 pm
content: Quote from: Beamin on November 03, 2017, 09:20:13 pmWhat kind of an answer are you looking for? I did design many good amps, had my share of awards and reviews, took part in numerous Hi-Fi shows. I made all my designs so they sound right to my ear, hoping that at least some people will share that understanding of a good sound. At home I use a production amplifier that I've designed in 1996 (it was sold for about £300 retail then) without any changes and still happy with the sound it produces. I won't change it for any high-end amp. As one well known reviewer said in 1998"The fact is, had a bought {this amplifier} 15 years ago, I might never have become an audiophile. Instead of thinking,Something's got to sound realer than this, I would have settled back in my listening chair and thought,Sounds about right. Now, were's that copy of Don Carlos?"There is no simple answer on how to make good amps. If I would try to share here some of my knowledge in that area, I will be ridiculed, I suppose (I've tried in that thread I've referenced earlier and got accused in making an art out of a simple engineering task).CheersAlex

author: oldway
author_rank: Super Contributor
timestamp: November 03, 2017, 10:01:10 pm
content: If you think one amplifier sound better than another, but you don't have any technical explanation why it is so and you can't measure nothing to proove it, as Dave use to say, it smell bullshit.Reviews, Hi Fi shows and others are 99% marketing and commercial interests.Some people are even trying to convince you to buy $300.000 speakers....

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 03, 2017, 10:20:02 pm
content: Quote from: oldway on November 03, 2017, 10:01:10 pmThere is that 1%.CheersAlex

author: Loboscope
author_rank: Regular Contributor
timestamp: November 03, 2017, 10:29:44 pm
content: If you canheara difference of two amps (or any other technical gear in audio), you willmeasurethe difference. If you cannot measure a difference, then there will be no difference and all your hearing wasillusion.Lots of scientific studies have been made in the field of hearing. And a healthy ear will indeed distinguish very tiny differences of delay time between the two ears, tone pitches and generally our ears are very sensitive with a great dynamic range between extremely quiet sounds and very loud noise.But there are not only the ears, who are the receptors of the audible world around, but there is also a brain and its hearing areas. And the brain will always make its owninterpretationof the neuronal information the ear brings to him. This is probably the greatest part of the work our brain does continuously: tosortand tointerpretall the information our sensory organs bring to him.And it is not difficult to consternate our brain and to let it misinterpret sensual informations. Probably the most perturbing factor are ourexpectations! We will hear and we will understand, smell and so on, what we expect to hear, to understand, to smell. There are also lots of investigations about this item.So if you want to do some objective tests, you musteliminate all expectationsas good as possible. Any scientific test design must have been done so. To test tone pitch differences or time delay differences it will be easy, because there is simply no expectation if the next tone presented will be lower of higher or will be heard coming from a certain direction.But if you will see the brand name of the amp or/and its appearance will be impressive, you will expect it will sound better. Immediately and unconscious.So only a double-blind-test will guarantee a certain level of objectiveness. DBT of cables always could reveal, that there will be absolutely no difference between ultra cheap and ultra expensive cinch-cables and microphone-cables for example.I would say, that if an amp exceeds some certain thresholds in THD, Noise, IM, channel separation, flatness in frequency, frequency range and so on, it will sound very good. For example, if its noise-level will be around -90 dB, you will hear no noise when you hear music at normal sound level. If the noise level will be better, -100 or -110 dB, the amp will be better from its specs, but at normal sound level you will hear no difference in noise.So if you will have to look for the quality of an amp, you will look for its specs and you must know, witch level the construction will exceed in all its details as minimum that all sorts of distortions will remain unaudible (at normal hearíng sound levels). Any sort of distortion beyond ca. -90 dB will not more audible for all I know.Another criteria could be its robustness, capability of uninterrupted service, especially if the amp should be used for PA.

author: trys
author_rank: Regular Contributor
timestamp: November 03, 2017, 10:31:56 pm
content: Quote from: BrianHG on November 02, 2017, 12:12:33 amThere goes a person who wears out the floor / carpet / acoustic chamber for having to sit in absolutely precisely and exactly the correct spot to hear the music for himself only.

author: DrGeoff
author_rank: Frequent Contributor
timestamp: November 03, 2017, 10:44:02 pm
content: Quote from: trys on November 03, 2017, 10:31:56 pmThe preamp probably has no EQ either.I remember those preamps from the 80's that had a single volume knob and input selector. Assuming that everyone was listening in a perfect room which does not exist.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 03, 2017, 10:50:14 pm
content: Quote from: DrGeoff on November 03, 2017, 10:44:02 pmIf you need EQ your amp and speakers aren't good enough.CheersAlex

author: DrGeoff
author_rank: Frequent Contributor
timestamp: November 03, 2017, 11:02:41 pm
content: Quote from: Alex Nikitin on November 03, 2017, 10:50:14 pmThat is complete and utter bollocks.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 03, 2017, 11:06:05 pm
content: Quote from: DrGeoff on November 03, 2017, 11:02:41 pmI will politely disagree. One of my criteria of a good system that it should sound tonally balanced at any level. If you have a problem with the room you treat the room, EQ won't help. EQ is a band-aid for poor quality systems IMHO (OK, sometimes for poor quality recordings too).CheersAlex

author: Zero999
author_rank: Super Contributor
timestamp: November 03, 2017, 11:09:49 pm
content: Quote from: Alex Nikitin on November 03, 2017, 05:04:41 pmNo, amplifier test was designed to obtain an objective result: whether people can differentiate between the sound produced by three different amplifier designs, a new (at the time) solid state, an older solid state and a valve design. This is clearly stated at the start of the paper. Granted, asking them their preferences was subjective, but that was not the primary goal. The results of the tests, are objective, rather than subjective. They clearly show the three audio amplifiers are transparent to the human ear, which is what one would expect from a good amplifier design.If one is to test amplifiers on sound alone, then one needs to eliminate any other differences, which would influence the audience's judgement, otherwise the test is pointless.

author: BrianHG
author_rank: Super Contributor
timestamp: November 03, 2017, 11:13:19 pm
content: Quote from: trys on November 03, 2017, 10:31:56 pmIf I could afford the house and room to do so, just maybe.... For the times when I need such an experience, I just use headphones.But seriously, I have basic quality speakers and it doesn't matter where you are in the room, they don't need balance correction. When playing a mono sound, the have a central sound filling the entire room.

author: trys
author_rank: Regular Contributor
timestamp: November 03, 2017, 11:24:51 pm
content: Mono.

author: cdev
author_rank: Super Contributor
timestamp: November 03, 2017, 11:32:55 pm
content: Pros perform equalization of a sound setup when its installed in a space. The placement of speakers makes a dramatic difference in how they sound. So there is no such thing as a flat response.

author: retrolefty
author_rank: Super Contributor
timestamp: November 04, 2017, 12:00:32 am
content: QuoteIf you can show measurable improvement specifications for your amps Vs other reference amps, I would love to have you share your experience.If however you are of the "you can't measure what I hear" world then I'm not interested in what you have accomplished.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 04, 2017, 12:20:34 am
content: Quote from: retrolefty on November 04, 2017, 12:00:32 amWell,hereare the measurements of one of my designs. As you may see, nothing to be ashamed of in a production amplifier. However no measurement would show you if an amplifier will sound good. It can show that it may sound bad in particular circumstances but there is no direct correlation between the sound quality and measurements from my experience. I always measured all my amps extensively however often a change in an amplifier would be barely noticeable on the measurements and quite noticeable in the sound quality. I've spent considerable time about 20 years ago trying to find the way to measure amplifiers so it would give me a reasonable level of confidence in relation of the sound it produces. Essentially, I've drawn blank.CheersAlex

author: Circlotron
author_rank: Super Contributor
timestamp: November 04, 2017, 02:11:42 am
content: Quote from: retrolefty on November 04, 2017, 12:00:32 am"I often say that when you can measure what you are speaking about, and express it in numbers, you know something about it; but when you cannot measure it, when you cannot express it in numbers, your knowledge is of a meagre and unsatisfactory kind; it may be the beginning of knowledge, but you have scarcely, in your thoughts, advanced to the stage of science, whatever the matter may be."Wiliam Thompson aka Lord Kelvin. Lecture on "Electrical Units of Measurement" (3 May 1883), published in Popular Lectures Vol. I, p. 73

author: Electro Fan
author_rank: Super Contributor
timestamp: November 04, 2017, 03:18:11 am
content: Quote from: Circlotron on November 04, 2017, 02:11:42 amWhat about people, including scientists 500 or so years ago, who thought the Earth wasn’t the the center of the Universe but who couldn’t fully prove it with measured numbers - maybe they had little of value to offer also? Or maybe not everything that can be perceived or at least hypothesized can be fully measured? It’s ok to hold out for measurements but to dogmatically do so might be to disregard the fact that some future knowledge made possible by superior measuremts and analysis could render current knowledge inaccurate or incomplete.

author: oldway
author_rank: Super Contributor
timestamp: November 04, 2017, 03:32:26 am
content: You can't compare the complexity of universe with a simple audio amplifier.Quote100% bullshit....With actual knowledge in analog electronics and modern instruments whe have, if you can't measure it, it does not exist...There is no magic at all in an audio amplifier.....

author: not1xor1
author_rank: Frequent Contributor
timestamp: November 04, 2017, 05:56:43 am
content: Quoteit is OT, but I could not resistancient scientists (they were called philosophers then) did know that earth was a planet floating in to the space and made quite good measurements (taking into account what they could use) ofthe size of the earthand theearth-moon distanceit is just the fault of religion (and ideology) which pretends to deform the reality to conform to its own sick prejudices that made the catholic church persecute Galileoregarding the topic, scientists have to keep an open mind so it would be wrong to dismiss the hypothesis of something still unknown about audio equipment quality evaluation and measurement, but on the other side we have plenty of proofs of the power of cognitive biases so the above mentioned hypothesis is high unlikely to be real

author: _Wim_
author_rank: Super Contributor
timestamp: November 04, 2017, 08:21:45 am
content: The more “high end” you go in amps, the more “difference” there can be. But these differences are intentional by design (sometimes also buy pure incompetence), because they want to sound different from the competition (and hopefully be perceived as “better”).I have come across/read about amps with intentional high second harmonic distortion (tube like sound), solid state amps with very high output impedance (low damping factor),amps with a rising response towards high frequencies, rising low frequencies, amps with a pronounced mid range, etc…Sometimes you also see very high power amps with very low THD specs, but they only quote THD at nearly max power. During normal listening barely 1 watt is used, and sometimes such amps still suffer from quite a bit of cross-over distortion making THD specs at 100mW a lot worse and potentially hearable! So watch out for those also.So I must say I agree a bit with both parties here. The well-designed amp problem was indeed solved quite long ago, but finding one is not always as straight forward if you are in the ” audiophile world”. Typically low end mass produced gear is much better is this, because they just use design from the application note and that is it. But sometimes they do indeed suffer from too much price cuts, so they also are not good.So, in my opinion, buying the more expensive (250€) mass produced low end amp from the typical companies like Marantz, Denon, Sansui, Yamaha, etc should get you an excellent amp. Another approach is to carefully look for detailed published specs. I like to see THD vs power, THD vs freq, output impedance vs frequency and IMD spectrum. Typically when this data is available the designer want to show they have a well-designed product. Unfortunately pushing detailed specs is something that is not common, and the typical argument is that they “do not mean anything”, while the contrary is true off course… DIY is for sure also possible, but unless you can use salvaged parts, is a lot more expensive that a mass produced amp (but also lots more fun!)As to speakers and acoustics, this problem is not solved at all for the moment, mainly because to the major impact of the directional behavior of the speaker, and its corresponding interaction with the room.Maybe with active speakers and lots of DSP power, we will be able to steer directivity of the sound better (with still a reasonable sized speaker), allowing to have a more predictable sound quality in the future.So, if looking for a technical challenge, this would be were a invest my time in. I have built an automated rotary table to measure the directional behavior of my loudspeakers, and the more you measure this, the more you appreciate the shear complexity and trade-offs in trying to get “good” directional behavior. “Good” means a good match with the room they will be used in, so this makes the challenge even more difficult.

author: _Wim_
author_rank: Super Contributor
timestamp: November 04, 2017, 08:34:19 am
content: Quote from: Alex Nikitin on November 04, 2017, 12:20:34 amVery true, looks like a well designed amplifier.Quote from: Alex Nikitin on November 04, 2017, 12:20:34 amThis I cannot agree with. You should always be able to measure the difference between amps when you can hear them.

author: Ing. Max Koschuh
author_rank: Newbie
timestamp: November 04, 2017, 09:04:01 am
content: You can measure 'some' details of a schematic, like 'Voltage' on the output, or frequency response. This does not mean you can measure ALL variables that produce what humans would call a 'good sound'.You can measure the size and weight of a Stradivari and a low budget violin, along with their sound levels and frequency responses. However, two violins of same size, with absolutely identical 'measurements', may sound completely different.I just want to say: Yes, in theory you should not hear any differences in two amps having the same measurement results. But in practice, they are very likely to sound different.This is even harder to understand if you don't have 'audiophile ears'.I personally don't think 'audiophile ears' (like I have) are great. Because you always want to improve the sound experience of your stereo. And in normal life, you always hear and identify noise sources at night, while others say it's completely quiet. Like a cat seeing things in the dark. I can tell you, this is very annoying. If I could choose between great ears and normal ears, I would choose the normal ones.If YOU can't hear any differences in amps having the same measurements, that's fine. This will save you a lot of time and effort in finding the right equipment. And lot's of money too. You just choose the exteriour design you like, or pick an amp with the best figures on paper, done.Enjoy music! That's the only important thing.

author: _Wim_
author_rank: Super Contributor
timestamp: November 04, 2017, 09:15:13 am
content: Quote from: Ing. Max Koschuh on November 04, 2017, 09:04:01 am2 different violin that measure the same do not exist, and if they did, they would sound the same. You need to measure the violin in ALL directions in 3D, and in ALL directions it should have the same frequency responses and resonance decay behavior. If all of those are identical, you will not hear a difference also.But if you would measure two violins this way (or even 2 different loudspeakers), I think you would be very surprised on how big the differences actually are. We are talking about differences as high as +30db and more (with loudspeakers at least, but I expect similar peaks and dips with violins), we are not talking about 0.01% or something like that!

author: Ing. Max Koschuh
author_rank: Newbie
timestamp: November 04, 2017, 09:35:39 am
content: Yes, that's true.In fact, scientists still don't know why the Stradivari sounds that different to other makes. And if they could find an answer, the manufacturers would use it to compensate the gap.Someone should come up with a 'sound measuring device' for hifi systems, which is not just based on the variables that we are normally used to measure.

author: Loboscope
author_rank: Regular Contributor
timestamp: November 04, 2017, 09:57:26 am
content: It is a myth, that the old Stradivari violins sound better than modern violins and today´s violin makers will not reach the level of Stradivais. They do and the "gap" has been closed since a long time. There have been made double-blind-tests (sic!) with good violin players playing some Stardivaris and some (very good) modern violins. The professional colleagues - they could not see, witch violin was played - could not distinguish the Stradivaris from the new instruments.As I mentioned already above, it is humans "expectation" what rules his perception.

author: oldway
author_rank: Super Contributor
timestamp: November 04, 2017, 10:06:17 am
content: Quote from: Ing. Max Koschuh on November 04, 2017, 09:04:01 amMillion-dollar Strads fall to modern violins in blind ‘sound check’http://www.sciencemag.org/news/2017/05/million-dollar-strads-fall-modern-violins-blind-sound-checkAudiophile ears don't exist only audiophile brain....Blind test proove it....To compare amplifiers, you must test them in the same conditions with the same speakers.....

author: Beamin
author_rank: Super Contributor
timestamp: November 04, 2017, 11:02:10 am
content: Quote from: Alex Nikitin on November 03, 2017, 09:59:38 pmSo if I want to make a high end amp with very low distortion how important are specs of the components to design specs? Could I take a low end amp with a decent power supply; change out the parts that are very specific to the value they are supposed to be (take all parts that are say 10% tolerance and hand select them to 1% or less) and see an improvement? I guess my question is what makes the high end amps different then the crappy ones? Big filter caps? Big transformers? Well matched transistors (I'm guessing most amps are push/pull designs)?

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 04, 2017, 11:07:13 am
content: Quote from: _Wim_ on November 04, 2017, 08:34:19 amQuote from: oldway on November 04, 2017, 03:32:26 amIf there is, I've managed to put it on a production line. Here is what I wrote in another thread:QuoteCheersAlex

author: b_force
author_rank: Super Contributor
timestamp: November 04, 2017, 11:21:27 am
content: Quote from: Loboscope on November 04, 2017, 09:57:26 amExactly this (I have even seen reports with just medium priced violins, with the same outcome)In fact there are even tests and papers were people were blindfolded to taste food.They can easily fool you with fish that looks like steak and vice versa. No one can tell the difference. Sometimes even when people KNOW they are being tricked.There are really billions of well described and repeated experiments how our senses fool us on a major scale.Yet, people still don't want to believe it. Perfect way for the maqrketing departement to use these thing of course.To mu surprise engineers even fall for it most of the time. It should be a fundamental part of the knowledge of an engineer to have a good sense about these concepts.Yet most engineers call it even "bogus and fake" science.With all the best respect, but I think you should reframe your thoughts about science than, and go back to school to relearn a few things. Because it's a fundamental part how we humans even judge certain experiments.Even being supported by sales; products that people think (= sound) better, are technically nnot always better. There is absolutely no correlation between that "good specs".On the other there is (proven) a very good correlation about what people believe is better.This is also why I don't go into much debate about it anymore. The data is all there, proven so many times over and over again. One single little subjective dubious personal experiment is not going to change that. Debating about the whole concept can be fun and interesting. People still saying, "yes, but my experience is difference and this is my poor proof", is not.Finally what I don't understand is why it is so extremely difficult for most people to simply admit they choice certain things just because of subjective reasons. Almost like it's something bad, a heavy sin or something?

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 04, 2017, 11:24:13 am
content: Quote from: Beamin on November 04, 2017, 11:02:10 amNo and no. There is no magic way to make a good sound from component parameters or measurements alone. Better quality components usually help but do not make a high end out of a low end unit. There are some special cases when component quality is critical, but unless the rest of the design is up to scratch, just changing some or even all components for "better" is not necessarily helpful. Sometimes - yes, if the original component is of a really poor quality. The amplifier I've designed in 1997 (the model I still use at home) has no special components, nothing was selected (and that would be impossible at $500 retail price point). Nevertheless the same reviewer I've quoted earlier did call it a "high end" amplifier in 1998, from the sound alone (there were no looks to speak of, frankly).CheersAlex

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 04, 2017, 11:40:44 am
content: Quote from: b_force on November 04, 2017, 11:21:27 amIf I would believe in all that pseudo scientific crap, I would never manage to design a good sounding amplifier. IMHO the DBT method is fundamentally flawed in testing the subjective experience, and your numerous examples are just confirming it. Thousands of lemmings are happily using a wrong instrument to "prove" their concepts and then call it science.CheersAlex

author: oldway
author_rank: Super Contributor
timestamp: November 04, 2017, 11:43:45 am
content: QuoteA subjective test of the type "I think it sound better" done by only one person has absolutely no statistical value.These are obviously biased comments you make on your own project.This is not modesty that suffocates you !!!

author: Fungus
author_rank: Super Contributor
timestamp: November 04, 2017, 11:51:32 am
content: Quote from: Loboscope on November 03, 2017, 10:29:44 pmYep. The human ear is far inferior to the tools available.If some "golden ears" can hear a difference then that difference can be measured, analyzed, quantified, and used in future designs.For amplifiers it's even easier because they output something that can be measured directly by basic measuring instruments (electrons). Characterizing an amplifier, plotting curves vs. loads, etc., is easy to do.Of course you need a decent power supply, etc., but with today's components there's no reason you can't build an "audiophile" amplifier for under $500.Quote from: Alex Nikitin on November 03, 2017, 09:59:38 pmEven Mr. Naysayer agrees on that point.

author: b_force
author_rank: Super Contributor
timestamp: November 04, 2017, 11:52:16 am
content: Quote from: Alex Nikitin on November 04, 2017, 11:40:44 amBelieve what you wanna believe, I base my opinion on scientific papers and data. Or are we just going to ignore everything?I don't see any reason why a double blind test would be flawed? It's maybe not perfect, but it's at least a much better method to compare things objectivily.But what I find mostly interesting in your argumentation, is that you don't believe in things were scientists objectivily try to find methods and correlations between things. But you do believe in people who are extremely heavily biased by so many things.The fact that millions of people use something, is not prove it's technically a better product.That can also be prove that marketing of that company is just extremely well done for example.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 04, 2017, 11:54:57 am
content: Quote from: oldway on November 04, 2017, 11:43:45 am"There are three kinds of lies: lies, damned lies, and statistics."CheersAlex

author: Fungus
author_rank: Super Contributor
timestamp: November 04, 2017, 12:05:48 pm
content: Quote from: oldway on November 01, 2017, 11:14:13 amTo compensate for the position of the speakers in the room! eg. one might be further away from your comfy chair than the other.Quote from: DrGeoff on November 03, 2017, 11:02:41 pmYep. Not everybody lives in a anechoic chamber.Quote from: Alex Nikitin on November 03, 2017, 11:06:05 pmSome people have spouses who don't allow them to stick big, misshaped pieces of foam to the living room ceiling.Also: Some people have 'parties' on Saturday nights where they want to unbalance the sound output and recreate the ambience of a discotheque.

author: Fungus
author_rank: Super Contributor
timestamp: November 04, 2017, 12:07:50 pm
content: Quote from: cdev on November 03, 2017, 11:32:55 pmNooo, Alex can set up your HiFi for youbeforeshipping!

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 04, 2017, 12:10:32 pm
content: Quote from: b_force on November 04, 2017, 11:52:16 amI see many reasons. The fact that this is the only method available for a stab at objectivity in these fields doesn't make it better than it is.Quote from: b_force on November 04, 2017, 11:52:16 amI am one of these people who did try to find these methods and correlations and spent considerable amount of time and effort on that subject. I did not come to my current position lightly. However, as I've said, I had to change my views if I wanted to be productive in making good sounding equipment. I've learned a lot about what you can measure in an amplifier, for example, and the published measurements of my designs confirm it. Few amps measure better in most important respects at the price point. But I can easily change a component in an amplifier to a different value (with a measurable difference if you know what to measure, but nothing changes in measurements commonly used to qualify an amp's performance) and it won't be nice to listen too.CheersAlex

author: Fungus
author_rank: Super Contributor
timestamp: November 04, 2017, 12:14:16 pm
content: Quote from: Alex Nikitin on November 04, 2017, 12:20:34 amRubbish. An amplifier can be completely characterized electronically. It's resistance, capacitance and inductance can be measured to ten decimal places at all frequencies.We can take the best "audiophile" amplifier in the world and measure its characteristics and compare them against a "bad" amplifier. I bet there will be no surprises.Quote from: Alex Nikitin on November 04, 2017, 12:10:32 pmEarlier you said "no measurement would show you if an amplifier will sound good". What's it to be?Are you the only person in the world who knows this secret measurement? Does it need a special instrument that nobody else has?

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 04, 2017, 12:45:25 pm
content: Quote from: Fungus on November 04, 2017, 12:14:16 pmOh, there will be surprises. You see, a Hi-Fi amplifier is not used with steady signals, or even with repeated bursts which you would normally use to test an amp. If you want something to be stable (OK, not even to ten decimal places, that's physically impossible, but to, say, five decimal places) you need time for things to settle. It is not possible with music, especially with any appreciable amount of power output involved. This continuously changing power output leaves a signature on the circuit and the reaction of an amp to that signature is important. You can go the expensive route to make the amp bigger and extremely linear (and so extremely expensive as well if you want a low output impedance and an unconditional stability into a difficult load), or you can tune the reaction to be "invisible" or at least benign to our music perception.Quote from: Fungus on November 04, 2017, 12:14:16 pmDon't be silly. If you know what you change you would know what to measure. It is just that measurement will show the change (and so can be used to control that parameter, for example), but will be completely useless on it's own, as a measure of quality. And it may be different for a different amplifier design.CheersAlex

author: b_force
author_rank: Super Contributor
timestamp: November 04, 2017, 12:52:13 pm
content: Quote from: Alex Nikitin on November 04, 2017, 12:10:32 pmSo why don't you name them?I can't think of a better method to test equipment without listeners seeing the actual products.Double blind simply means that neither the participants nor the researchers know what is going on.The hole goal is to avoid biasing as much as possible. I don't see how you can be 'against' that??That's basically saying that you're against objective research?It's used in many many MANY others fields as well.On top of that there is the abx test. Which means the participants first hears fragment a, than b, than x.They need to decide if x was a or b. A proper setup also does a simple a - b comparison on top of this.

author: oldway
author_rank: Super Contributor
timestamp: November 04, 2017, 12:52:21 pm
content: Quote from: Alex Nikitin on November 04, 2017, 11:54:57 amAbsolutely stupid answer.....You deny scientific, technical and statistic rules and principles....only to say: I am an expert, my amplifier is a low cost high end project....I am probably the best...The audiofools theories found in this topic are based on the same principles of misuse of the truth used by proponents of the flat earth theory: they deny all technical and scientific evidence, they make false comparisons between subjects totally different, they turn the subject around by converting a comparison between amplifiers into a comparison of loudspeakers and finally, they systematically use bad faith.

author: b_force
author_rank: Super Contributor
timestamp: November 04, 2017, 12:58:38 pm
content: If people say that statistics are worse than lies, they clearly don't understand 1% of scientific research at all.Science is by definition based on statistics, simply by the fact that you can only can take a sample out of a limited group or amount.That by default means there is some kind of uncertainty, because it's absolutely impossible to check all possible variations.If people don't understand or know these basic principles and definitions, a discussion is worthless anyway.Than it's on the same level about debating if one religion is better than the other, or if Pepsi is better than Coca Cola.Nothing wrong with that, but a different and not an open and objective discussion.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 04, 2017, 01:04:09 pm
content: Quote from: oldway on November 04, 2017, 12:52:21 pmOh, that was a reaction to your absolutely stupid comment, nothing more. On a serious note, an attempt to use statistics with a fundamentally flawed method is stupid. A single listening test certainly has no statistical value, however it may have a useful value if you know the limits of such a test.The problem with statistical approach to a subjective test in an attempt to make it somewhat objective (already sounds somewhat stupid, doesn't it?) is that you need to build a significant number of samples and our perception is a very bad performer when dealing with a repeated stimulus.CheersAlex

author: _Wim_
author_rank: Super Contributor
timestamp: November 04, 2017, 01:16:25 pm
content: Quote from: Alex Nikitin on November 04, 2017, 11:40:44 amDBT to find correlations between measurable differences and subjective sound quality are indeed difficult (I would not go so far an inherently flawed).But if audiophile claim they hear “major differences” between amps that measure identical, but yet, cannot distinguish them in a DBT, then it is not the test that has failed…

author: _Wim_
author_rank: Super Contributor
timestamp: November 04, 2017, 01:18:48 pm
content: Quote from: Loboscope on November 04, 2017, 09:57:26 amI can very well imagine that is true. Yet, I would still expect them to have measurable differences when measuring them in all directions...

author: oldway
author_rank: Super Contributor
timestamp: November 04, 2017, 01:27:14 pm
content: What?? My absolutely stupid comment ?It is the height...!!!!!You are a talker but a nullity on the engineering level.You continue to mislead the truth and deny mathematical principles such as statistics and technical and scientific principles.We are on a technical forum to discuss techniques between people who know or want to learn, not with quacks like you.This discussion is useless and insane, I stop participating because I do not want to waste my time with fools anymore.

author: Fungus
author_rank: Super Contributor
timestamp: November 04, 2017, 01:32:58 pm
content: Quote from: Alex Nikitin on November 04, 2017, 12:45:25 pmOh, there will be surprises. You see, a Hi-Fi amplifier is not used with steady signals, or even with repeated bursts which you would normally use to test an amp. If you want something to be stable (OK, not even to ten decimal places, that's physically impossible, but to, say, five decimal places) you need time for things to settle.A load can be designed to step between whatever capacitive/inductive/resistive state you think causes problems with amplifiers. That change of state can be measured and optimized. It will be no more difficult than measuring the step response of an oscilloscope, ie. any competent R&D department can solve it in a morning or two (and probably many people here as well).But anyway...I'm done here. All you have is insinuation. No facts, no figures, not even the name of your 'secret' measurement.You don't pass the basic sniff test.

author: ludzinc
author_rank: Unknown
timestamp: November 04, 2017, 01:36:40 pm
content: *than

author: Fungus
author_rank: Super Contributor
timestamp: November 04, 2017, 01:44:21 pm
content: Quote from: rdl on November 01, 2017, 11:36:05 amYep.Quote from: Beamin on November 03, 2017, 09:21:19 pmNobody ever opens them up and looks inside? Really?There's never any ventilation slots on the sides that let you catch glimpses of the expensive-looking components within?Even if consumers never open them up, repair people will and they'll see "The Difference" and tell other people about it. It all helps keep the high priced manufacturers in business.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 04, 2017, 01:45:05 pm
content: Quote from: oldway on November 04, 2017, 01:27:14 pmI did show my amplifier designs and can show much more. Did you show anything? No. So who is a nullity?Quote from: oldway on November 04, 2017, 01:27:14 pmI did not. All I do is to point some serious flaws in a traditional approach which claims to be scientific and in reality isn't. Again, you did not provide any arguments against mine except the word "statistics" as if it explains everything.Quote from: oldway on November 04, 2017, 01:27:14 pmAs I've said, I can provide a proof in many thousands of amplifiers sold and used in many homes for many years. And these simple looking amps do hold their value very well even now, 20 years later.CheersAlex

author: Fungus
author_rank: Super Contributor
timestamp: November 04, 2017, 01:47:48 pm
content: Quote from: Alex Nikitin on November 04, 2017, 01:45:05 pmAnd you managed toretailthose for 300 quid.Thus proving everybody's point!

author: BNElecEng
author_rank: Regular Contributor
timestamp: November 04, 2017, 02:06:09 pm
content: Dipping into this conversation, does anyone know if Sennheiser have an amp integrated into their headphones? I'd take mine apart but it's a pain in the proverbial.In my personal experience, I've found the sound from them to be lovely, regardless of what device they have been plugged into. So I'm wondering if there's something active inside responsible for this or if it just down to the speakers and construction.

author: Fungus
author_rank: Super Contributor
timestamp: November 04, 2017, 02:10:40 pm
content: Quote from: BNElecEng on November 04, 2017, 02:06:09 pmConclusion: Maybe there's not much real difference between devices, despite what headphone amplifier sellers tell you.Quote from: BNElecEng on November 04, 2017, 02:06:09 pmYes.

author: BNElecEng
author_rank: Regular Contributor
timestamp: November 04, 2017, 02:15:40 pm
content: Not sure I can make the same conclusion since I don't know what is going on inside the headphones. Guess the only way to know for sure is to take them apart.I'll keep you posted on what I find.

author: glarsson
author_rank: Frequent Contributor
timestamp: November 04, 2017, 02:20:19 pm
content: Quote from: BNElecEng on November 04, 2017, 02:06:09 pmDo they have batteries or connect to an external power supply? No extra energy -> no amplification of the input signal is possible.

author: A Hellene
author_rank: Frequent Contributor
timestamp: November 04, 2017, 02:49:19 pm
content: Ah! Come on, guys!What about (lots of) people invoking a graphic (or a parametric) equaliser that, though it will raise the THD from the advertised 0.05% (which would be a flat mostly response aural reproduction of the audio source) to the eccophantic amount of more than 60% THD perhaps, that might be claimed to be 'sounding better' to their ears? And, that with no intention of correcting the speakers' or the room's acoustics...Quote from: A Hellene on August 18, 2017, 10:02:13 pmEmphasis is mine.-George

author: Beamin
author_rank: Super Contributor
timestamp: November 04, 2017, 02:50:59 pm
content: Quote from: Alex Nikitin on November 04, 2017, 11:24:13 amOK so lets assume no BS or audiophoolery. Whats makes my Marantz amp sound so much cleaner then the sony amp? The "100 watt" sony amp can't keep up with the 50 watt Marantz. I consider myself to have good ears but I can't tell the difference in any thing better (more $) then the Marantz amp (tube amps do sound a tad warmer like record vs CD) but there seems to be a huge sound difference in cheap vs mid range. I also compared a harmon kardin to my Marantz and sounded the same and then when looking at specs the HK also has 10X lower THD then the sony. When I did this test I had never heard of harmon kardon or THD so really just going off sound.As I said earlier the Marantz is physically a lot heavier and also lopsided. I assume that amp design in the early 2000's was not a limitation when I bought mine. By this time I'm sure any patents had expired and all speakers are modernized (not those low wattage high efficiency light weight magnet ones of the 1960's). That would probably mean there isn't part of your purchase money going to R&D anymore. So what are you paying for? (I limiting myself to two channel stereo amps no fancy 7.1 THX stickers or BS just stereo music amps)

author: Fungus
author_rank: Super Contributor
timestamp: November 04, 2017, 03:15:45 pm
content: Quote from: Beamin on November 04, 2017, 02:50:59 pmMarketing. I've long suspected that SONY engineers usually mess with the sound (mostly to give it more bass for their consumers).

author: Cliff Matthews
author_rank: Unknown
timestamp: November 04, 2017, 03:52:53 pm
content: Question to pose: Since audio PWM tech is here to stay, for analog buff's can it be improved?I submit this from attached PDF (Re: Model No53 from Levinson corp): (Are they hitting on any good audio future here?)With conventional switching amplifier designs (Class D and other variants), the output stage is theoretically limited to a single interleaving topology, which limits the audio passband frequency range, and results in high order distortion products that must be filtered out with brickwall filter types, which can introduce phase errors and other in-band artifacts. Care must be taken to ensure timing accuracy, as dead time (brief moments where neither the positive or negative stages are conducting current between pulses) and overlap (brief moments where both positive and negative output stages aresimultaneously conducting current, which is a catastrophic condition that can cause amplifier failure) must be thoroughly addressed.With our Interleaved Power Technology, there are eight half bridges (N=8) working synchronously to achieve a much higher PWM switching frequency (4 MHz). The modulation circuitry also features propriety refinements to ensure that there is absolutely zero dead time between pulses, as well as completely preventing the possibility of overlap.The charts below detail the tremendous improvements in bandwidth and low distortion provided by our Interleaved Power Technology (N=8), compared to conventional Class D (N=1) techniques.They give 2 charts:1) Sample Standard Class D output (N=1) 100 kHz, sidebands and odd harmonics are all present2) Sample IPT (N=8) with extremely marginal residuals around 800 kHzWith Interleaved Power Technology (N=8), the comparably much smaller byproducts of the PWM switching process are easily filtered out using a simple high frequency notch filter, which has no deleterious effect on sound quality, compared to brickwall filter types.And, the broad frequency response provided by our Interleaved Power Technology system is ideal for the latest wideband high resolution audio formats, with extended response all the way out to 95 kHz (-3dB, 8 ohms).Once the 5 or 10 year patents expire, could we expect anything better from PWM?

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 04, 2017, 03:58:38 pm
content: Quote from: Fungus on November 04, 2017, 03:15:45 pmI agree. I've seen cases (on cassette decks, but it certainly possible for amplifiers too) where Sony did deliberately decrease the sonic performance (without an obvious drop in measured parameters) on cheaper units from the same range.Also Marantz is one of few Japanese Hi-Fi manufacturers who actually listen to their equipmentand even cheaper units perform reasonably well from my experience.CheersAlex

author: xani
author_rank: Frequent Contributor
timestamp: November 04, 2017, 04:15:50 pm
content: Well if we really wanted to measure "difference between amplifiers" it would have to be whole chain as in "measure difference between movement of speaker membrane caused by defined signal" (after all speakers are not just pure RLC components). I still doubt there would be a meaningful difference between "audiophile" and "just a well designed and well specced" amplifier. And even if there are, clearly they do not matter enough to make a difference in double blind test.So all it is left for "audiophile" hardware to do is to... "distort" signal to sound better. Which is fine, if advertised of such, but it will never be.Quote from: b_force on November 04, 2017, 12:58:38 pm"People say" statistics are worse than lies because it isveryeasy to bend them (either by just taking data in "wrong" way or misinterpret results), either on purpose or by mistake. Same thing marketing does with various measured parameters, like advertising.And honestly I'm not surprised as it can be very hard to avoid various kinds of errors and biasses even if it "looks simple"Quote from: Gyro on November 03, 2017, 01:07:55 pmYou can still be technically competent and believe in silly stuff or push bullshit to people. See any tech-made-marketing. Or most people that go to church.

author: Fungus
author_rank: Super Contributor
timestamp: November 04, 2017, 04:31:22 pm
content: Quote from: Alex Nikitin on November 04, 2017, 03:58:38 pmI've noticed a definite (ie. non-subtle) bass boost on some Sony CD players.Source components like CD players are supposed to be the most neutral things in the audio chain but nobody's told the Sony marketing department that.nb. I don't know if other companies do this but Sony definitely seems to do it as a matter of policy.

author: Electro Fan
author_rank: Super Contributor
timestamp: November 04, 2017, 05:43:53 pm
content: Quote from: Beamin on October 31, 2017, 12:14:15 pmHi Beamin,Congrats on starting a stimulating thread.I can't answer your questions definitively but here are some comments and suggestions.First, your point about solid state vs. tube electronics is insightful - this topic alone is likely to stimulate some debate - but if you are really in search of what makes a Macintosh amplifier sound "better" (your word, not mine), perhaps you should keep your inquiry open to examining both solid state and tube amplifiers.Next, with respect to a "$#*!" Sony, I would say that it's a bit of a harsh comment on Sony. While Sony has over the years produced many relatively entry level (inexpensive) amplifiers, receivers, and other stereo equipment Sony was for a number of years pretty competitive in stereo equipment design. In the 1970s they had a model called the STR-7065 that rivaled the Marantz 2270 (generally a very well regarded receiver). The 7065 held it's own in parts, specs, and sound against the 2270 in the power amplifier section as well as in the pre-amplifier and tuner sections.Marantz 227070 watts per channel RMS 20-20kHz into 8 OhmsHarmonic Distortion 0.3%IM Distortion 0.3%Sony 706560 watts per channel RMS 20-20kHz into 8 OhmsHarmonic Distortion 0.2%IM Distortion 0.2%Regardless, let's take your question about replacing the parts on a "$#*!" amplifier. Maybe as an experiment you could select whatever you consider to be a "$#*!" amplifier and replace the parts with better quality parts, ie, parts that perform similar functions but with better values, ie, better individual measurable performance at the component level. The idea would be to retain the same basic circuit but see if the circuit once enhanced with better parts delivers better performance. Now we are down to the question of “What is better performance?” Is better performance something that can be measured (Harmonic Distortion, etc.)? Or is better performance something that can be heard? Or both? Is there really a measurement for everything we can perceive with our hearing? Who knows?Maybe you could give it a try and let us know what you determine with your measurements and listening.Next we have the notion that the amplifier is just a "really a simple circuit that's been improved over 100 years". 100 years is a fairly long time (one human lifetime or more in most cases) and the last 100 years have been subject to considerable developments in knowledge and technology, with technology accelerating our understanding of what causes what. The point here is that now, after 100 years, perhaps there is more than one simple circuit design for amplifiers. Toward this end, a second experiment might include studying the designs (including the parts and their specs, and the overall circuits and circuit performance) for some of the popular and/or more well regarded amplifiers over say the last 50 years - perhaps since the 1970s or so. You could look at the solid state designs from Phase Linear, Crown, Marantz, and others to see if you notice any differences in the designs. Likewise, you could look at tube amplifiers from Macintosh, Audio Research, VTL, and others to see if you notice any difference in the designs. These would be more expensive to obtain and certainly more expensive to surgically adjust with different parts so this part of the experimental process might require more study than hands-on work, but I bet if you look across the spectrum of amplifiers you will find that in both solid state and tube amplifiers there are actually quite a few different designs. Perhaps they all came from the same amplifier tree 100 years ago and perhaps they all have a common goal of amplifying a relatively small signal from a pre-amplifier and forwarding the larger amplified signal to speakers but I think you will find considerable differences in design. Even the most ardent, “science is science and not audiofoolery” members of this forum would have to recognize that there are quite a few different designs in the world of amplifiers. Class A, Class B, Class AB, Class D and more – especially when you consider both solid state and tube, not to mention analog vs. digital.If you listened, could you hear any differences between the various designs? This has become a key subject of the firestorm of a debate your post has kicked off. I'm not ready to go there just yet, but here are some other considerations.When we say "so to get to the best sound" (again your words, not mine) are we talking about some type of test tone or tones fed into the amplifier with an objective of determining the best sound exiting the amplifier? Or are we talking about the best sound after real music has left the amplifier and passed through the speakers into a room in which the speakers reside? How much consideration are we giving to the complexities of harmonics and any other interactions among all the sounds that occur in real music? And are we giving any consideration to these signals as they entered the amplifier? From what preamplifier was the music delivered? And perhaps even more importantly, from what source was the signal delivered to the preamplifier? For example, if it started as a signal embedded in a record (ie, as complex music etched into vinyl), how were these signals captured (recorded) before the record was cut? Setting aside the microphone or microphones that were used to record the signal(s), or the acoustics of the performance that was recorded, or the performance of the tape recorder, or the skill of the sound technician that did the recording used to make the record, was the stylus that was used to retrieve the mono or stereo signals from the record part of a ceramic cartridge, or a magnetic cartridge, or a moving coil cartridge? And what tone arm with what mass, what anti-skate, and what alignment geometry on what turntable with what suspension was used to hold the cartridge and stylus with the proper force so that the signal or signals could be retrieved from the record?I know, none of that is what you asked about. But at the end of the day, the amplifier is just one part of a larger chain - and it's somewhere in the middle to the back half of the chain. Most people here would say a power amplifier has relatively little impact on the overall sound - at least compared to the impact of the quality of the recording and the playback components which required acoustic and mechanical vibrations to be converted to electrical energy and compared to the impact of the quality of the back end components (the speakers) that convert electrical to mechanical and acoustic energy. The job of the amplifier is to amplify the signal, not add anything and not subtract or change anything. While in some important respects a power amplifier has among the easiest of the jobs in the chain it is important to remember that amplifying test tones might be somewhat easier than amplifying complex music filled with subtleties.Next, speaking of the speakers - this is where amplifier design can and should vary, ie, with consideration given to the speakers that will be driven by the amplifier. For example, if the amplifier will be driving very efficient horn speakers (think Klipschorn), the amplifier might need relatively little power output - perhaps a couple or a few watts from perhaps a 45 or 2A3 Single Ended Triode tube amplifier. On the other hand, some inefficient speakers might need a couple or a few hundred watts from a pentode tube amplifier or a solid state amplifier. To design an amplifier to sound "good" we need to have some idea of the load - which means we need to have some idea of the speaker design.The selection of the speakers can't be done in a vacuum (no pun intended) either because the speakers need to be placed properly with respect to room acoustics, ie, room shape, room size (how much air needs to be moved to achieve what Sound Pressure Level), and room surface treatments. There is a reason why symphony halls have the shapes and materials they have.Now, back to the original question, what makes an amplifier sound the way it does? It comes down to the design including the execution of the design (ie, including the parts and the craftsmanship/quality control of the manufacturing process). Will the differences in the design and execution be measurable? With a part change here or there perhaps. Will the differences between fundamentally different designs be measurable? Probably, but it might depend on the measurement equipment and skill being applied. Will any of these differences be audible? Perhaps, but it might depend to some extent on all the other variables in the system from signal recording and retrieval (cartridge and tone arm) to especially the particular speakers and especially the particular room that acoustically mates with the speakers.Net, net: Changing some parts in a given amplifier design and then measuring and listening might be a worthwhile experiment if this line of inquiry intrigues you. Likewise, if you have the opportunity, you might measure and listen to disparate amplifier designs and see what you determine, especially as those designs are embedded in various systems and rooms. My guess is that changing a few parts for similar but better values will be time consuming and tedious with minimal differences in overall amplifier output measurements and the differences if any will be very difficult to hear since by the time you take the amplifier apart and put it back together with different parts your mind will not be able to distinctly remember what it previously heard. Most A-B comparisons need to be instantaneous at the same volume levels in order to have a decent chance of revealing differences. On the other hand, putting fundamentally different amplifier designs into otherwise same systems and rooms will very possibly reveal some different overall sounds as well as some different amplifier measurements. Likewise, moving the same amplifier into different systems and rooms might also reveal some different overall sounds.What causes what in sound systems is a complex issue and developing a vocabulary that accurately and effectively conveys all the possible combinations of design, measuring, and listening can add a layer of complexity and confusion that is more likely to stimulate debates than agreement unless people develop some common goals, in which case the debates can sometimes be methodically narrowed down so as to isolate some areas of agreement about what causes what.Let us know what you determine if you make such measurement and listening tests.

author: hamster_nz
author_rank: Super Contributor
timestamp: November 04, 2017, 06:18:22 pm
content: The HiFi purest's dream dream of "truly lifelike" sound is an interesting one... put your headphones on (or semi-decent speakers) and have listen to an acoustic guitar microphone shootout:Even on no-so-hi-fi YouTube the different mics sound very different. Everything has already been quite significantly 'colored' at the first step of the recording process....

author: Electro Fan
author_rank: Super Contributor
timestamp: November 04, 2017, 08:15:42 pm
content: Quote from: hamster_nz on November 04, 2017, 06:18:22 pmHere is an interesting Q&A regarding the guitar post from the comments sections of the post:OnSugarHillTo the creator of this video... We can all hear these great mics and make an opinion on which sounds best. However, you are the only one who knows how that guitar sounded in that room while you were recording. Which one do you feel most accurately represents the sound in the room??ReplyAugminished StudioI didn't want to put my opinion in the video as I did not want any form of bias. But I was partial to the MBH 603. The 121 would be cool depending on the track. The U87 was the most "true".This gets at the questions of what is "better" sound or what is "the best" sound.Perhaps some people in this forum will say that none of the sounds from any of those microphones sounded different than any of the others. These people will say that the sounds heard were all just electronically recorded and played back and any differences heard would have to be considered audiofoolery, unless the differences could be pointed out consistently in double blind tests.Next we would have some people who would say they could hear differences but there might be some disagreement about which differences were better or best. Or perhaps someone might point out that we are all probably listening with different amplifiers and different speakers in different rooms. And no doubt we would have disagreement about whether the amplifier in our playback systems had any impact on the overall sound.In my opinion, which is worth what everyone is paying for it (that would of course be zero), a good objective for a hifi system is to play back music as close to what it sounded like when it was created by the original musicians. This is in fact the definition of "hifi", ie to retain a high degree of fidelity to the original. But even this is difficult to know unless we were present to measure and/or listen to the original when it was recorded.In the case of the youtube video creator he apparently thought one of the mics best captured the "true" sound of the guitar as it originally sounded, but that of course was based on the system (including the amplifier and speakers) he listened to when he played the recording (possibly in the same room as was used for the live guitar music - so the same room acoustics). Most likely his preference was based on his subjective opinion unless he "measured" the fidelity of the mic'd recording by comparing the measured recording to a measurement of the original music. (Not sure what single measurement or few measurements would capture all the subtleties that the ear can hear.) Further, even though he thought one mic was more "true" than the others, he had a preference for yet another mic.There are LOTS of variables at work: 1) some that are objective, easy to isolate and easy to measure; 2) others that are objective but harder to control and measure; and 3) some that are subjective. All of which can be somewhat hard to convey with words as words often have different meanings to different people. Which is probably why some people find supreme comfort in numbersMy overall opinions are that 1) different electronics will likely measure different and are sometimes audibly discernibly different, 2) not everything that is audibly discernibly different is readily measureable in the electronics, 3) we all vary in our abilities to hear (this can be measured in various manners), and 4) we vary in our preferences regarding what is good, better, and the best sound.

author: Zero999
author_rank: Super Contributor
timestamp: November 04, 2017, 09:26:39 pm
content: Quote from: Beamin on October 31, 2017, 12:14:15 pmSo what do you want from your amplifier?How much power does it need to output, into what load impedance? That will determine the supply voltage, current, along with the components used. VPEAK= (PRMS×2RL)0.5To avoid distortion, the supply voltage should always be a higher than the voltage drop of the output stage, plus a bit of headroom.What do you mean by sound good?If you mean not distort, or colour the sound, in any way detectable to the human ear, i.e. be audibly transparent, then that's fairly easy to achieve, at a fairly low cost, with modern discrete components and amplifier ICs. The most important thing is not to allow the output voltage to go anywhere near the point where clipping occurs. This is normally fairly easy to detect by listening to it, but basic VU meter which flashes a red light, when the output is clipping and another light, when the output is near-clipping, is a very good idea. The next thing is to ensure the IC or transistor output stage, is kept cool enough to avoid overheating or any thermal protection circuitry from kicking in and distorting the sound. That's relatively easy to do for modest power ratings (under a few hundred Watts), given that the average power dissipation is usually a fraction of the peak power output.If you want some level of distortion, then that becomes more tricky, because you need to know what sort of distortion you like. There's the world of valve amplifiers and digital processing, which attempts to emulate different valve amplifiers. I admit I can't be much help with that, as I like amplifiers which don't introduce any distortion. The main market for distorting amplifiers are musicians, who want their instruments and often vocals to be coloured by the valve sound of the 50s and 60s.Anything you build will work out more expensive than modern, massed produced equipment, but it will still be cheaper than high end, possibly audiophool equipment. Other than the fun and educational element, the main advantage of building your own, is you know how it will perform, especially if you use a decent, genuine audio amplifier IC, with a proper data sheet and figures to back it up, or a tried and tested design, again with all the test data to back it up.

author: fonograph
author_rank: Frequent Contributor
timestamp: November 04, 2017, 10:00:59 pm
content: Double blind test is to audiophool what sunshine is to mold.

author: Fungus
author_rank: Super Contributor
timestamp: November 04, 2017, 10:11:26 pm
content: Quote from: Electro Fan on November 04, 2017, 08:15:42 pmNope.I bet there's not a single one.It's obvious that anything with mechanical parts can influence sound, that some will be better than others.

author: Electro Fan
author_rank: Super Contributor
timestamp: November 04, 2017, 10:54:10 pm
content: Quote from: Fungus on November 04, 2017, 10:11:26 pmCool - if you are right then we’re all in agreement that different transducer designs with mechanical components in audio systems can provide discernible audible differences.

author: Zero999
author_rank: Super Contributor
timestamp: November 04, 2017, 11:04:20 pm
content: Quote from: Fungus on November 04, 2017, 10:11:26 pmThat's true and the same goes for amplifiers. The question is whether the differences are detectable by the human ear? Double blind testing is required to determine that, but it's more difficult for microphones, than for amplifiers. Unless the microphone is being used to record music played through a speaker, the musician needs to be made unaware of which microphone is being used to record them, otherwise it might influence how they play.

author: coppice
author_rank: Super Contributor
timestamp: November 04, 2017, 11:14:23 pm
content: Quote from: hamster_nz on November 04, 2017, 06:18:22 pmOf course microphones sound different, and often blatantly obviously different. Although microphones generally fall into patterns of response, like cardioid, no two models of mike have exactly the same spatial response pattern. That means the mix of direct and indirect (reflected) sound being picked up by the mics differs. Resonances, the difficulty of placing two mics in the same place (as you don't know exactly where its effective centre is), and other mechanical factors also change what is picked up.

author: Zero999
author_rank: Super Contributor
timestamp: November 04, 2017, 11:38:25 pm
content: Quote from: coppice on November 04, 2017, 11:14:23 pmYes that's true. I think I missed the point before. The differences between good microphone designs is greater than amplifiers and this intentional.

author: retrolefty
author_rank: Super Contributor
timestamp: November 05, 2017, 01:11:16 am
content: Quote from: Electro Fan on November 04, 2017, 10:54:10 pmElectro-mechanical devices (mics, speakers, turntable cartridges) may indeed be discernible depending on the specific devices being compared. If so it can also be measured, and if not, can be proven so via well designed double blind testing. Anything this else is just human faith based imagination.

author: Electro Fan
author_rank: Super Contributor
timestamp: November 05, 2017, 02:49:07 am
content: Quote from: retrolefty on November 05, 2017, 01:11:16 amSeems like a fine summary.

author: hamster_nz
author_rank: Super Contributor
timestamp: November 05, 2017, 05:06:53 am
content: Quote from: retrolefty on November 05, 2017, 01:11:16 amNo, I think it is worse than that.You can't achieve "transparent" HiFi (HiFi that is like being there)... Even for purely acoustic recordings the recording has already been shaped by the recording process - the venue, choice of mics, the placement of mics, the mixing of mics, the recording desk used, the mastering process and so on... even before the listener gets their hands on it.To jump analogies, Highend HiFi is like arguing which camera gives the most realistic image of a "landscape" when you take a photo of an oil painting. A crappy camera is easy for all to see, but between reasonably good cameras it is subjective - "Fuji has deeper greens making the grass look better", "Cannon has more muted blues, making the sky look better", "but Nikon has beast flesh tones, making the people more realistic (even though there are no flesh tones in a landscape)" - all this when looking at photos of a painting, which is already an artistic impression of what the landscape looked like.

author: SeanB
author_rank: Super Contributor
timestamp: November 05, 2017, 05:33:14 am
content: HIFi is a subjective thing, generally to me anything over 320k MP3 is as good as the CD, as I cannot hear the difference at 256k MP3, using the equipment I have. All else is merely just you believing things are better, and past a certain point all will behave pretty much the same. All that we can agree to is that a small speaker out of some random low cost manufacturer, where QC is the guy next door, will sound bad when paired with an amplifier made from whatever parts were cheap that day.In any case with speakers the most important part is the magnet, basket and coil and core, as that is what determines how good it is. Expensive speakers often use quite cheap drivers, and generally any amplifier will drive a speaker perfectly well at low levels, which is where they will spend most of the time being used.

author: BrianHG
author_rank: Super Contributor
timestamp: November 05, 2017, 06:43:12 am
content: After all this talk about bandwidth and THD, how come all amp specs ignore full power bandwidth, IE, a 20khz spec amp which has a max power of 100watts may do well at that frequency at 10 watts but begins to distort above 50 watts at 20Khz while it can do 5KHz fine at full 100 watts.Even more useful would be frequency phase shift at the higher frequencies at higher power levels. I know here, with opamps at low power, 20KHz may be nothing, but, with an amp designed to drive a speaker with high voltage and current through an unpredictable cable length may already exhibit phase behavior 's at 20KHz like a good quality audio op-amp would exhibit at 1MHz domain.

author: ruairi
author_rank: Frequent Contributor
timestamp: November 05, 2017, 06:57:41 am
content: Quote from: BrianHG on November 05, 2017, 06:43:12 amThat spec is ignored because music does not contain anywhere near that much power at higher frequencies, it's just not how we use power amps.Re your second point why will the phase shift at hi frequency change under normal use higher power levels?

author: BrianHG
author_rank: Super Contributor
timestamp: November 05, 2017, 07:11:56 am
content: Quote from: ruairi on November 05, 2017, 06:57:41 amI know higher frequencies in music have lower power levels. I just don't an alteration in phase as I change the volume level & change the music type.Op-amp example: With an TLO81 opamp, I can actually get an ok 1Mhz performance with a minimal gain setting and a 1vp-p signal out, but, going up to 10vp-p, it cant quite make it there. This may have partially a connection to the slew rate of the op-amp's output. In a audio power amp, slew-rate can play a similar role in distorting phase at higher signal power levels at it's peak frequency output.Remember, we are talking about the differences of what makes a good VS a bad audio amp & I think this little tid-bit is ignored. I know the Amp I personally chose for myself is a basic, all discrete MOSFet output design which can deliver the full 100KHz at 250 watts, it can also deliver a 1MHz signal up to around 10 watts. But, my really old equipment did much worse.Refinement in the higher frequencies steadily deteriorated with louder volumes in my older amp. Cymbals lost their distinct multitude of rings and were steadily changing into noise at loud volumes. My new super fast audio amp does not exhibit this problem.

author: _Wim_
author_rank: Super Contributor
timestamp: November 05, 2017, 07:24:28 am
content: Quote from: BrianHG on November 05, 2017, 07:11:56 amWhy would you think this would not show up in a THD plot vs frequency. If you measure THD vs Frequency at high power you will also clearly see this effect. This is why companies like Hypex for example plot THD vs frequency at very low power, mid power and high power. In this one plot you can already see quite a lot of information.

author: BrianHG
author_rank: Super Contributor
timestamp: November 05, 2017, 07:36:21 am
content: Quote from: _Wim_ on November 05, 2017, 07:24:28 amCareful, not all audio amplifier manufacturers care to spec everything at all power levels. We are in the realm of what makes a good VS bad amp here. A better response would be to only select from amp manufacturers who properly plot the THD at all power levels and frequencies. The problem here is that there may be good quality affordable designs out there which haven't gone through the trouble to do such plots while the majority of others may just be hiding nasty truths about the limitations of their design.

author: Fungus
author_rank: Super Contributor
timestamp: November 05, 2017, 07:39:04 am
content: Quote from: Hero999 on November 04, 2017, 11:04:20 pmNope.All speakers and microphoneshaveto be compromises. There's no way around it.eg. The ideal speaker would be a tiny point source of sound, but that's impossible to manufacture in practice because a speaker needs to move large amounts of air around. Real life speakers are therefore a compromise between stereo imagery and bass response. You can never have both.This isNOTtrue of amplifiers. Amplifiers aren't constrained by physical properties like air movement.

author: _Wim_
author_rank: Super Contributor
timestamp: November 05, 2017, 07:58:27 am
content: Quote from: BrianHG on November 05, 2017, 07:36:21 amYes, I know detailed specs are a problem. I just wanted to respond to your quote "After all this talk about bandwidth and THD, how come all amp specs ignore full power bandwidth", becasue in my opinion this does not bring any additional values if you have THD vs power vs frequency info.Quote from: BrianHG on November 05, 2017, 07:36:21 amI would not say it is "additional trouble", as during the design of the amp they should have measured this anyway (how would they design for good specs otherwise?). I think the main problem with the audio business is they do not want objective criteria to compare products, because that would mean that the many small companies in this branche would not be able to survive, because it hard to explain a mass produced device costing 10 times less has the same of better specsIn one of the previous post here was written that the fact that a high quality amp could be brought to the market by a smaller company for 300£ proves that the problem was solved long ago, and I could not agree more with this statement.

author: Electro Fan
author_rank: Super Contributor
timestamp: November 05, 2017, 08:47:59 am
content: Quote from: Fungus on November 05, 2017, 07:39:04 amFungus, please tell us:1. Why is the ideal speaker a tiny point source of sound?2. Are there any other attributes or specifications of "the ideal speaker" other than it being a "tiny point source of sound?"3. How much air is "large amounts of air"? Does the "ideal speaker" have a specification for moving "large amounts of air""4. What is your definition of "stereo imagery" and how much (or what specifications) would be ideal stereo imagery?5. What is your definition of "bass response" and how much (or what specifications) would be ideal bass response?6. Why can no speaker ever have both ideal bass response and stereo imagery?7. What are the best speakers you have heard in terms of bass response?8. What are the best speakers you have heard in terms of stereo imagery?9. What are the best speakers you have heard that make the best compromise between bass response and stereo imagery?10. (Bonus question): Is your favorite word "Nope"?Thanks, EF

author: Raj
author_rank: Frequent Contributor
timestamp: November 05, 2017, 08:53:15 am
content: I'd say the ability to recreate the waveform of input in output, but amplified, is the key.Does it output square, sine, saw tooth, of the same shape as input throughout and wider than the frequencies of hearing range or does it rounds off the edges or straightens out the curves.

author: _Wim_
author_rank: Super Contributor
timestamp: November 05, 2017, 09:17:42 am
content: Quote from: Raj on November 05, 2017, 08:53:15 amWaveforms with sharp corners (square and triangle) dictate infinite frequency response if NO rounding is required. Such waveforms can be described as a sum of sinusoidal waveforms (and will "sound" like that also). The more high frequency sinusoidal waveforms you add to the sum, the sharper the corners will be.Looking at a square waveform can be interesting (to see ringing for example) and easy to do (main advantage of square wave testing), but THD+IMD together with frequency plots will show the same information also in a way that makes it easier to compare performance ("good" square wave performance is a more subjective criteria)

author: Fungus
author_rank: Super Contributor
timestamp: November 05, 2017, 09:23:20 am
content: Quote from: Electro Fan on November 05, 2017, 08:47:59 amSimple version: A big round thing moving in and out doesn't produce as directional a sound as a smaller round thing moving in and out.Quote from: Electro Fan on November 05, 2017, 08:47:59 amHeadphones.Quote from: Electro Fan on November 05, 2017, 08:47:59 amNope.

author: BrianHG
author_rank: Super Contributor
timestamp: November 05, 2017, 10:00:49 am
content: Quote from: Fungus on November 05, 2017, 09:23:20 amMy Clements RT7 tower speakers outperforms GradoLabs SR1 headphones with stereo imagery even down to their astonishing flat 25Hz response. In fact, with properly down-converted to stereo DTS bluray movies, I've had sub-sonic effects start out in front of you, move right through you, then felt behind you as imaged by the location of the effects in the movie. Try doing that with a mono sub-woffer.

author: Psi
author_rank: Super Contributor
timestamp: November 05, 2017, 10:13:34 am
content: i'm a Klipsch person myself. Just love how clear they sound.yep, Klipsch reference series for speakers and a good ($250+) pair of sennheiser's for headphones.

author: hamster_nz
author_rank: Super Contributor
timestamp: November 05, 2017, 11:01:17 am
content: Quote from: BrianHG on November 05, 2017, 10:00:49 amAre you able to explain how we avoid constructive/destructive interference effects that occur when you have two (or more) speakers, vs when you have a single source (like an real musical instrument)?An A6 (1760 HZ) is about 20cm long, so with a sound on the center all it takes is a a speaker 10cm closer than the other (about half a head) to have destructive interference on the fundamental, leaving you only with reverb from the room...

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 05, 2017, 11:10:43 am
content: This thread makes an interesting reading in some respects. I would like to make a brief summary:1) There is apparently a religion - worshipping DBTs. Any objections or doubts rejected out of hand without a discussion by its adepts. Nobody said that he/she did participate in at least one properly constructed DBT, so this looks like a pure leap of faith.2) No replies to my simple question - if any of participants did design a Hi-Fi production amplifier. So no direct experience with the subject is admitted.3) Suddenly, there is an agreement that speakers and microphones are not as easy as amplifiers, being mysterious mechanical contraptions beyond the simple electronics world limits.4) The simple fact that an audio power amplifier has to deal quite intimately with that strange speaker creature somewhat escapes attention.As the result, the level of discussion (with few exceptions) is at a kindergarten stage. Which is appropriate for this Beginners section. Perhaps one day we may have a reasonably serious discussion about power amplifiers design. There are some interesting nuances there. Like how a non-linearity of the output impedance of the amplifier may affect the sonic performance (and why). Or how non-linear and delayed signals generated by a speaker and injected back into the amp's output affect the sonics depending on the amp's ability to deal with these. Or what are the effects of a power supply size and energy storage (and why, for example, that patented "Class I" Levinson amplifier linked here earlier uses a very conventional power supply technology with a huge and heavy toroidal transformer plus lots of large capacitors).CheersAlex

author: Zero999
author_rank: Super Contributor
timestamp: November 05, 2017, 11:48:18 am
content: Quote from: Alex Nikitin on November 05, 2017, 11:10:43 amBecause no, logical, well-thought out doubts to DBTs have been raised. They are used by the scientific community in other fields. The fact is, if no one can tell the difference between how amplifier A and B sound, then there are other factors which need to be considered in deciding which is better.QuoteNot as such but fairly recently, I've been involved in designing a new intercom for a noisy environment (both acoustically and electrically) at work. What's amazed me is how good cheap amplifier ICs are.QuoteThere's nothing mysterious about the fact that amplifiers introduce less distortion and audibly transparent compared to the transducers connected to either ends of them. It's backed up by solid science. Audio amplifiers don't need to be any "better" than they are today. They're already better than the human ears, speakers and microphones, so improving them further is a wasted effort.QuoteSpeakers are fairly benign, compared to what other power supplies and amplifiers have to put up with, yet not the same level of nonsense surrounds them as audio amplifiers. It just defies logic.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 05, 2017, 12:40:04 pm
content: Quote from: Hero999 on November 05, 2017, 11:48:18 amPlease note: you are using logic and common sense, which is fine, however I speak from many years of experience in the field. Logic is only good if the input is accurate and sufficient. Otherwise, logic is a very good instrument for fooling yourself.CheersAlex

author: _Wim_
author_rank: Super Contributor
timestamp: November 05, 2017, 12:44:35 pm
content: Quote from: Alex Nikitin on November 05, 2017, 11:10:43 am1) I have done several tests in the past to try and hear the effects of introduced all kinds of distortion. Tools used:- ABX plugin for foobar:http://www.foobar2000.org/components/view/foo_abx. This plugin allows you to compare 2 sound tracks, were you have modified one with distortion, and the other not. Then you can try to distinguish between the two with the software randomly playing one of both- Distorter VST plugin (can seem to find it online any more). Other VST plugin to generate distortion should work equally well.- Good headphones (Sony MDR-V6)=> my conclusion: it is not as easy to hear it as I thought it would have been. It for example also interesting to compare lossless with varies MP3 formats2) I have built amplifiers (but mainly based on designs by others), measured quite a few, even sold a few, but it was not a viable business for me (unfortunately, would love to be in the audio business, but it is difficult)3) problems with speakers and microphones is they radiate/record in 3D, resulting in an infinite amount of "signals". An amplifier on the other hand is only a single point in this space. If a speaker would radiate equally in all directions (point source), equalization would be very easy with DSP. Just record the impulse, calculate the reverse, download it to a DSP-convolver, et voila, a perfectly flat response…4) If you mean the amplifier becomes unstable under certain load conditions, I would not classify this amp as "well designed. When measuring the amplifier parameters, I agree one should use a realistic load (no problem using even a real speaker, only for measuring at max power, this might be “slightly annoying”)By making statement the discussion is at kindergarten stage and beginners, you do indeed ensure this is true...“Like how a non-linearity of the output impedance of the amplifier may affect the sonic performance (and why).”Output impedance is mainly determined by the amount of feedback. If non linearity’s exist, these will show up in the IMD distortion plots. For this it might also be necessary to test with burst signal (transient intermodulation distortion). From what level of distortion this is audible is open for debate, but please feel free to state what your minimum distortion level is for good sound…In my opinion levels below 0.03% (-70dB) are readily achievable and more than sufficient (try it with the app above to see if you can easily hear this!). At low power levels this can even be decreased to 0.1% because it will be buried down deep into the noise.“Or how non-linear and delayed signals generated by a speaker and injected back into the amp's output affect the sonics depending on the amp's ability to deal with these”An amp with very low output impedance act as a true voltage source, back emf is handled by the negative feedback. If not it will show up as measurable distortion.Somewhere in this thread you stated that some audible things cannot be measured, please tell me one design choice that make a difference that will not show up in measurement. From the examples you have given so far, this is not the caseP.S: you still did not reply to my remark on DBT that it should work perfectly to check for audible differences (same as with the foobar app above)

author: Fungus
author_rank: Super Contributor
timestamp: November 05, 2017, 01:25:29 pm
content: Quote from: Alex Nikitin on November 05, 2017, 11:10:43 amHuh? Your only "argument" is that DBTs don't work because nobody can provide a link to one.I can provide links to CEOs of audiophool companies running away screaming from DBTs, even when offered a million dollars toparticipate.Quote from: Alex Nikitin on November 05, 2017, 11:10:43 amNone is necessary.Quote from: Alex Nikitin on November 05, 2017, 11:10:43 ama) Nothing sudden about it, andb) The two aren't related. The "best" amplifier in the world can be fully characterized without connecting a single speaker to it. We can use it to know what the desirable properties of an amplifier are.Quote from: Alex Nikitin on November 05, 2017, 11:10:43 amJeez. If only Electronics Engineers could know what those things are. And how to measure them.You yourself claim to have designed an audiophile-grade amplifier that retailed for 300 pounds. You're actually disproving your own point every time you mention it, ie. you're demonstrating that amplifiers are a solved problem.

author: Fungus
author_rank: Super Contributor
timestamp: November 05, 2017, 01:36:51 pm
content: Quote from: Alex Nikitin on November 05, 2017, 12:40:04 pmPlease use logic to explain why a DBT won't work.We can start with easy stuff like speaker cables.(which have had DBT done to them, withpredictable results)

author: Fungus
author_rank: Super Contributor
timestamp: November 05, 2017, 01:40:25 pm
content: Quote from: Alex Nikitin on November 04, 2017, 12:10:32 pmYou still haven't told us what this secret measurement is that nobody else knows about.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 05, 2017, 02:19:17 pm
content: Quote from: _Wim_ on November 05, 2017, 12:44:35 pmAn interesting observation - people try this kind of experiment and happily accept that there is no difference you can hear if you can not sense it under these conditions. No one seems to question the validity of the test itself. As I've said earlier, our perception is very bad when dealing with a repeated stimulus. Thus when you try to build up a number of samples required for a proper statistical analysis you reduce the sensitivity almost to zero for the small differences we are trying to sense.Quote from: _Wim_ on November 05, 2017, 12:44:35 pmWe are not talking about bad designs. However, I should note that a very low output impedance over a wide frequency range and an ability at the same time to work on any real-world reactive load are not easy to achieve in a typical amplifier relying on a global NFB for linearity.Quote from: _Wim_ on November 05, 2017, 12:44:35 pm"Very low output impedance" and "true voltage source" are relative terms. And same as with distortion levels it is open to interpretation on what is acceptable. From my experience there is no simple and fast rule on any of these. No amplifier is perfect in all respects, even these designs which try to push linearity as far as possible (i.e. Halcro).Quote from: _Wim_ on November 05, 2017, 12:44:35 pmOn DBT see above. There are too many problems with these tests, essentially DBTs are not sensitive enough for the stuff we are trying to deal with. The easy way is to dismiss sonic variations not detectable in a DBT as non-existent, however for an amplifier designer it is a counter-productive approach, you are just fooling yourself and eventually your customers.On measurements - a great deal can be measured but to establish a definite link between these measurements and listening experience is difficult to impossible at the levels of quality we are trying to achieve. On what to measure - a simple example. You replace the mains transformer in an amplifier to a somewhat more powerful one, providing the same voltage under load so there is no change in the measured output power, distortion, frequency response, transient response etc. Would it affect the sound amplifier makes - more than likely. So what do you need to measure to see the difference? It is certainly a measurable difference. What could you do to correct the sound to accommodate a different transformer?CheersAlex

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 05, 2017, 02:30:28 pm
content: Quote from: Fungus on November 05, 2017, 01:25:29 pmIt is up to some degree a solved problemfor me. Could you do the same?CheersAlex

author: Zero999
author_rank: Super Contributor
timestamp: November 05, 2017, 04:20:31 pm
content: Quote from: Cliff Matthews on November 04, 2017, 03:52:53 pmI doubt it will objectively sound any different to any other modern class D amplifier. It sounds like more marketing, than anything else. A -3dB bandwidth of 95kHz for an audio amplifier? That seems like a liability, rather than an advantage!The harmonics on the plot are mostly well above the audio band (even the second, 10kHz, is 55dB down on the fundamental, which shouldn't be noticeable) so removing them will make no difference to the perceived sound. It's interesting they don't show spectrograms with other input frequencies than 5kHz, probably because that's the worst case scenario.This set up might be theoretically better for RFI and efficiency, but the results don't seem to indicate this and no comparison is made to a standard class D design.

author: Electro Fan
author_rank: Super Contributor
timestamp: November 05, 2017, 06:19:11 pm
content: Alex,Don't lose heart on this. It's a discussion and whether everyone agrees or not with a particular viewpoint I think we might be making some progress on what is a tough and often debated subject. I'm not even sure I can fully state what the debate is about but somewhere near the center of the debate is a question regarding hifi equipment that deals with what can be heard and what can be measured.Unfortunately, the issue sometimes turns a little contentious as some people dig in and then once they are certain of their position rather than playing nicely some folks resort to personal attacks. It's a fine line from a bit of humor to making someone feel bad. In my opinion the term "audiofool" (or however it's spelled officially) is kind of derogatory, but since I'm probably in that camp to some extent I'll suck it up and deal with it without getting too concerned by it.I think like you / as you have expressed - that people can get to a point in a discussion where logic strongly defends a position but if the logic is based on faulty input, or incomplete input, the output can be faulty. I tried to make a case that said 500 years ago many people were very sure that the Earth was the center of the Universe, in part because we didn't have the knowledge and science to mathematically prove otherwise with a sufficiently demonstrable line of reasoning. For sure, when it comes to witch hunts we can burn witches at the stake over and over thereby proving that witches burn.Lightening up the subject a bit, here is where I think we (some of us) left off. It is from retrolefty's summary:Electro-mechanical devices (mics, speakers, turntable cartridges) may indeed be discernible depending on the specific devices being compared. If so it can also be measured, and if not, can be proven so via well designed double blind testing. Anything else is just human faith based imagination.While I believe, like you (I think), that this forum has a near religion-like belief in the infallibility of DBTs I think it was refreshing (to at least a degree) that we got to retrolefty's summary (which I think is a fine summary) because it at least leaves open the idea that different transducer designs (speakers, mics, cartridges, etc.) with mechanical components in audio systems can provide discernible audible differences. Up to that point I was starting to think that the prevailing thought here is that little or nothing of value is discernible by human hearing. I was starting to think that the alternative to an audiofool was an engineer who sat in his hifi room but who never actually listened to his hifi; rather his use of the hifi was strictly as a DUT that provided a reason to measure things. In this stereo-(pun intended)-typical view, the anti-audiofool derives 100% of his enjoyment from watching numerical, graphical, and other readouts of measurements and 0% of his enjoyment is derived from actually listening to music. Or if the anti-audiofool is not fully science oriented in their hifi endeavors they might occasionally listen to their hifi but have no sense what sounded "good" because there would be no reason to listen critically as long as the spec sheets and measurements show good numbers. All of this is of course a bit sarcastic but hopefully not too derogatory coming back the other direction.My more sincere view is that the science oriented audio enthusiast relies extensively if not exclusively on specifications and measurements to help select, build, and install good sounding equipment at reasonable prices so as to in turn be able to enjoy listening to music.Where we are in this thread is that we seem to have reached a consensus (but probably not unanimity) that people can hear the differences in transducers and that those differences have measurable explanations. We even went a step further (at least in retrolefty's summary) and said that if the differences couldn't be measured but they could be shown to exist with DBTs then perhaps some differences could be accepted as humanly discernible even if we have not yet come up with a way to explain the differences with test equipment of some sort. To me, this is progress.Next we get to your subject: the amplifier. Fwiw, I'm with you. I think the current consensus in this forum is a bit dogmatic when we say that everything that needs to be solved with amplifiers has been solved. I'm not saying some amplifiers haven't reached a tremendous state of performance (ie, to a very high degree they amplify the signal faithfully without adding or subtracting anything) but I don't believe this is as widely true of amplifiers as some folks might think or want to believe. Having said that, I do think that making an amplifier go reasonably flat (say +/-0.25dB) from 1Hz to 100kHz is probably "easier" than making speakers that go from say 20Hz to 20kHz +/-3dB. And I think it is the relatively simple frequency response measurement (combined with perhaps a few other relatively simple measurements) that cause most people to hold firmly to the notion that speakers (and perhaps other transducers) are relatively "hard" to design and build, and amplifiers are relatively "easy" to design and build.Before we go further on the seemingly contentious subject of amplifiers, I think it might be good to look at the “half-full” side of the discussion, ie the notion that at least some hifi performance can be determined by human hearing. Again, I am heartened by retrolefty’s summary as it acknowledges more belief in human hearing than I thought might prevail around here.One of the biggest breakthroughs in this thread (for me) was the seeming acknowledgement that “imaging” is a real concept. Even our forum buddy Fungus seems not only ready to get behind the concept of imaging but he seemed to put it forward as one of the key determinants of an “ideal” system (along with bass response). Prior to this thread I would have bet that the scientifically correct hifi enthusiasts here would have dismissed imaging as an audiofool’s errand. So I was delighted to see that imaging is an accepted attribute.What is imaging? Well, I couldn’t get Fungus to define it but here’s my attempt at describing it with an example. I think imaging is the notion that when you go to see (and hopefully listen to and not just measure) a jazz band in a night club you might have the piano in one place on the stage, the bass in another location, and the drums in yet another location (since they can’t all occupy the exact same physical location). Back in the early 1960s these types of bands were popular and they sometimes played in night clubs. In the night clubs people in the audience sat at tables having drinks as they listened (but probably didn’t measure). During these times a really good band with a record deal might have recorded some of their performances “live”. In the process a sound technician might have deployed two microphones (one relatively left and one relatively right) which enabled making a two track (stereo) tape recording and ultimately a LP stereo record. The microphones would have picked up the sounds of all the instruments and also other sounds – such as the people in the audience speaking, clapping, or maybe occasionally clinking their drinking glasses.After the performance the recording would have been “mixed” by someone in a studio (perhaps in a team effort by sound technicians and maybe/probably with input from the band members). Here the band team would be relying on memories, impressions, preferences, and other subjective human input to determine how much of what was actually recorded on the tape during the live performance should be retained exactly as it was captured on the tape, and how much might be subject to editing. Perhaps the bass player wanted to hear more of the bass on the final version of the upcoming record, or perhaps maybe the band wanted to turn up the clink of a glass to help listeners get the sense of being in a night club.The point is that with just two tracks (stereo) lots of sounds from various directions – not only left and right but also front and back were captured and transferred from the microphones to the tape to the record. All of this results in the capture of sonic location “imaging” information – right to left and front to back. When you listen to a good pressing of a Bill Evans record listen for the sonic image, ie the relative locations of the instruments and other sounds. Depending on your hifi setup you might be pleasantly surprised at what you hear. If you are really into it, try moving your speakers a few feet or inches to see if the location of any of the sounds move as a result. (In general, but not always, try making the distance between your speakers roughly the same as the distance from the plane of the speakers to your listening position.)Now, how does the information on the record get played back by speakers in a room? Setting aside the “easy” amplifier, the speakers have to somehow convert electrical signals to mechanical motions and create acoustic energy that somehow conveys the “image”. In order for this to happen with “fidelity” (faithfulness to the original – as it was captured live and then mixed with “artistic” as well as engineering input), the speakers must be able to move air in a way that creates not only the frequencies (lows, mids, highs) but also the spatial imaging (left, right, front, and back) of the individual sounds, all while retaining the subtle details of the “original” sound which means that not only do the frequencies need to sound like they did, and in their respective locations, but with the same relative strengths (amplitudes) so that some sounds aren’t emphasized more than what was originally intended and so that some sounds some aren’t deemphasized more than intended. Why would some sound be emphasized or deemphasized? Well, consider our speakers working hard to be “flat”. If the speakers, for example, produce a 3dB hump at 100Hz, the sounds at 100Hz will come across significantly louder than intended by the original. Now do the speakers do all this by themselves? No, they do this depending on their own performance plus how they couple to acoustics of the room. Depending on the size, shape, and surface treatments of the room, the room is very likely to either emphasize (reflect) or deemphasize (absorb) some frequencies at the expense of others. So unless the room is designed (probably with lots of calculations when it was constructed) and then measured and treated (after it was constructed) with a knowledge of what speakers would be placed where in the room (and where relative to the listener), all of these acoustic variables could result in almost any sonic end result. This is why sometimes speakers (and overall systems) sound one way in a showroom and then sound noticeably different when installed at home in a different room.So, back to Fungus’ post about bass response, one of the things we find is that given the wavelength of a low note, some bass notes might be longer than the length of a person’s home listening room. (A low note in “natural” music, for example, might be created by a 32Hz organ pedal). If the speaker is even able to go down to 32Hz on it’s own (sans room) at 0db (ie, “flat) – and most speakers can’t – then when we put that speaker in a room that is say only 18 feet long (ie, deep front to back) the 32Hz bass note is going to hit the back wall before fully completing a wavelength and then reflect back toward the front wall (never mind anything it hits in between), and so on and so forth. (I think a 32Hz audio signal has a full wavelength of approximately 35 feet). Meanwhile all the other frequencies are also going to be reflected or absorbed based on the size, shape, and treatments of the room surfaces. (These timing differences can be further exacerbated by any delay in speaker crossovers and/or the placement of individual drivers relative to the plane of the speaker.) These interactions are going to have a significant effect on the overall sound.Now, back to imaging. Imaging is the is concept that if things were well recorded in the first place – with attention given to details – and if the details could be accurately retrieved from the recording (we have only touched upon the role of a stylus, cartridge, tone arm, and turntable) and if those details were transduced from the mechanical realm of the record to the magnetic and electrical realms of the cartridge while being mechanically supported properly by the tonearm and turntable and if the phono section of the preamplifier and the rest of the preamplifier faithfully amplified the phono signal before passing it to the power amplifier, which in turn faithfully amplified the signal into the speakers (we haven’t really gone very deep into speaker loads), and if the speakers were designed and built properly, and if the speakers were installed in the right locations in a properly designed room – well then, we might just have faithfully reproduced the spatial image as it was intended.When we say we passed the signal from here (say the preamp) to there (say the power amp), we are just using brevity. In reality, the “signal” is a composition of all the many signals (as represented by the frequencies, amplitudes, and phases) that were captured way back at the night club.So, what should we know about imaging? It’s a real thing and it’s filled with subtleties that once you experience it and know what you are listening for, it can make a big impact on your ears and your mind. Most of us know imaging left to right (as a stereo image) and many of us know imaging front to back, although I think most people think of imaging in a fairly gross (large) manner – perhaps as we know it from the THX signature sound as it blares and rumbles through a commercial or home theater. While the sound of a jet plane or a train can be very exhilarating, the same concepts of left to right and front to back can also be exciting in a more delicate and subtle way with the clink of a glass on a Bill Evans live night club recording.So, what does this have to do with say an “easy” amplifier? (Btw, this is where I get called out as an audiofool.) I would like to know which measurement on an amplifier spec sheet is the measurement that tells just how accurately the “easy” amplifier reproduces the sonic image. I have yet to see this spec on any spec sheet. Now before someone jumps in and says “well, but if we measured frequency response, and if we did this phase test, and we applied these rigorous test procedures, etc, etc, etc. we could show just how well an amplifier handles imaging”. To which I would say, maybe so, but a) I don’t see such tests being done and reported, and b) I bet things like the transformer(s), capacitors, and a variety of other components and circuits will have an impact on just how well aligned the amplifier output - with all the micro signals that make the composite signal input - retains the fidelity of all those details. Further, while imaging is an attribute that took several paragraphs to describe, I’m betting there other some other sonic “attributes” that could, with further description and discussion, also be surfaced as potentially present and discernible, albeit subtle - and that often go unmeasured.Next, we get to: what can people hear? My guess is that people as a species have similar hearing capabilities but within the species we as individuals have a fairly wide distribution of hearing capabilities. What we can hear is of course in many respects measurable (frequency response, etc.). But what we can perceive and recognize is perhaps a function of our learned experiences. Some people can listen to music and say “that was a C note” while others have no clue, or at least they have no way to accurately express what they heard. The point here is that just because some people can discern this, that, or the next thing doesn’t mean another person has the same capabilities. Hearing is a function of anatomy, but also training/learning. Likewise, the ability to express what is heard is also a skill.In summary, I’m happy to “hear” (pun intended) that we could at least get to a consensus on retrolefty’s axiom. While it might leave some folks thinking that the axiom should have been extended to strictly electrical components such as amplifiers, at least it seems to have opened our thinking about what causes what and how the causes and effects might be expressed in both what we can measure and what we can hear (and in turn, attempt to describe).PS, I’m really not trying to create a battle here – just trying to express some thoughts that lead to constructive discussion and possibly some learning for everyone, me included.Peace. EF

author: Bicurico
author_rank: Super Contributor
timestamp: November 05, 2017, 06:50:38 pm
content: Hi,I normally avoid posting on audio/HIFI/audiophool threads.I am fairly literate in electronics, I do appreciate music and I have a wide range of musical taste. Yet, I consider myself a normal person, a bit on the skeptical and realist side.While I do agree that there is a lot of audiophooling in the market, from special cables to power filters, etc., I challenge anyone to actually spend a few hours listining to a propper high end audiophile class setup.I have had the pleasure to do so and actually have a friend who designs and manufactures such audiophile amplifiers, mostly valve based.Never had I listened to music like that. The sound can be described as totally transparent, like when you look through a fogged glass (bad audio system) and then suddenly this glass is replaced by a crystal clear and clean glass. Also, it is amazing to close your eyes and feel the musicians located around you. It's like they are playing right in front of you, you can accustically locate them, feel them.All this is a totally subjective feeling and it definitly does not happen with:- cheap/bad quality setup (amplifier, speakers and source)- bad arranged speakers- room with bad accoustics- source with bad recording/mastering- ...Regarding "cheap setup", mind you, the best cheap amplifiers can be bought on eBay: look for premium amplifiers from the early 80ies! This is what I have now...I dare to say, that since the 70/80ies, people have lost the drive to hear HIFI quality music - they prefer to listen to their favorite songs through convinient means (phone, mp3, internet streaming), at the cost of sound quality. It's an option.So, before engineers start to claim EVERYTHING as audiophooling, please do ask some friend or store to give you a propper listening session. Take your favorite CD's with you and just listen to them. Be open minded to listen to their suggested CD's, which have been selected based on the quality they were produced. One great CD for this is the Dark Side of The Moon by Pink Floyd. Another popular CD to evaluate a HIFI setup is "Arne Domnerus Group - Jazz at the Pawnshop".I am pretty sure, you cannot just scientifically measure the quality of an amplifier, when it comes to reproducing music.For example, I had a comparitive test, where the same amplifier was driving three different pairs of speaker. All of them premium audiophile ones. And I definitly like the sound of one particular set of speakers more than the other two. But the host prefered a different speaker and he was far more literate than I am. Conclusion: in the end, it is the individual taste.After such audiophile listening session, I went back to my regular stereo set and noticed how bad it actually sounds...After some months, I don't notice that anymore, as I am used to it again.Chers,Vitor

author: Cliff Matthews
author_rank: Unknown
timestamp: November 05, 2017, 07:19:13 pm
content: Quote from: Hero999 on November 05, 2017, 04:20:31 pmThanks, so nothing remarkable for the future of PWM then. Yeah, I thought it was strange that Levinson compared their stuff to a signal coming from a 100khz system.. According to Elliot Sound Products, that's the lower-end of where PWM starts, so yeah, there's a few more harmonics down there.http://sound.whsites.net/articles/pwm.htm

author: Fungus
author_rank: Super Contributor
timestamp: November 05, 2017, 08:27:13 pm
content: Quote from: Alex Nikitin on November 05, 2017, 02:30:28 pmWhat's your point, exactly?

author: Fungus
author_rank: Super Contributor
timestamp: November 05, 2017, 08:35:52 pm
content: Quote from: Electro Fan on November 05, 2017, 06:19:11 pmLet's put it another way: Do you think amplifiers are going to get much better in the future?Quote from: Electro Fan on November 05, 2017, 06:19:11 pmRight.Because current "spec sheets" are written by marketers and the numbers you see were chosen in the 1960s. Marketers have no interest whatsoever in providing accurate information or allowing people to easily compare their amplifiers to other brands.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 05, 2017, 08:37:09 pm
content: Quote from: Electro Fan on November 05, 2017, 06:19:11 pmThank you. I am on Internet forums from about 1994 (ever heard of Canopus?).CheersAlex

author: Fungus
author_rank: Super Contributor
timestamp: November 05, 2017, 08:40:45 pm
content: Quote from: Bicurico on November 05, 2017, 06:50:38 pmHuh? To reach that conclusion wouldn't you have to listen to three different amplifiers driving the same speakers?Quote from: Bicurico on November 05, 2017, 06:50:38 pmA lot of it is also practice. You can learn what to listen for. A lot of systems might sound impressive initially but if you know what to listen for you can hear that the trumpeter at the back is a bit muted compared to other systems.

author: Fungus
author_rank: Super Contributor
timestamp: November 05, 2017, 08:50:49 pm
content: Quote from: Bicurico on November 05, 2017, 06:50:38 pmYes, but some of us have been doing that since the 1980s. The sound honestly hasn't got an order of magnitude better since then, and most of the advances are in speakers AFAICT.Quote from: Bicurico on November 05, 2017, 06:50:38 pmThere you go. You're proving my point.Modern amplifiers are smaller and more integrated but they're not particularly better.

author: hamster_nz
author_rank: Super Contributor
timestamp: November 05, 2017, 08:55:27 pm
content: I dropped out of HiFi when life changed in my mid/late 20s, and then went back in with DIY projects like home made subs and home made 300W power amps, and designed and built bi-amped active speakers using good kitset amp modules. These all worked fine (not great, just fine), but eventually I moved on to other things.For imaging speaker placement & room layout was everything. Fire the sound across the short side of the room (helping to avoid early reflections from the walls), any my rear ported speakers need to be away from the wall by ~40cm. However, this is now completely impractical now unless I am home alone - the main living area is ~5m x 12m, with lots of hard flooring and windows & doors down one of the long side - you can clap and hear the reverb time, and the room just 'eats up' sound. So I've abandoned all hope of HiFi - I'm happy just to have some speakers back in the house after languishing in the garage at our old house for 10 years due to lack of room.

author: _Wim_
author_rank: Super Contributor
timestamp: November 05, 2017, 09:27:42 pm
content: Quote from: Alex Nikitin on November 05, 2017, 02:19:17 pmBut if we cannot hear the differences with exaggerated distortion in this ABX test, why would I then here the difference when just listening to music?Would in your opinion a better test then be to use the same VST-plugin in the audio chain just listening to music, and one day listening with 0.1% of added distortion, and the next day without? Do you think the sensitivity would be increased then? As my music source is pc based and has a convolver plugin, I could easily do that also.?Quote from: Alex Nikitin on November 05, 2017, 02:19:17 pmI would measure both power rails. If they behave the same (no big sagging), the sound will be the same also. If not, I would expect to see a difference in distortion using burst signals.

author: Zero999
author_rank: Super Contributor
timestamp: November 05, 2017, 09:35:05 pm
content: I don't think anyone here has referred to imaging as audiophoolery. It's perfectly rational and logical. I've listened to the same track but mixed differently and have preffered one mix, over the other. No I didn't do a double blind test on myself, so I am open to the possibility that I may not be able to distinguish between the two, although the differences seemed clear, at the time.The point made by many people here is that a well designed amplifier has very little impact on the sound, compared to the recording, mixing and playback, from the speakers to the room acoustics. The same could be said for a video amplifier, in an old analogue TV: the camera used to record the content, the stage lighting, interference picked up by the signal as it was transmitted through the air, the quality of the CRT and room lighting had more of an impact on the quality of the image, than the video amplifier.

author: fonograph
author_rank: Frequent Contributor
timestamp: November 05, 2017, 10:44:37 pm
content: Placebo is one hell of a drug.

author: Electro Fan
author_rank: Super Contributor
timestamp: November 06, 2017, 12:31:32 am
content: Quote from: Fungus on November 05, 2017, 08:35:52 pmThat's a good question.There are reasons to think amplifiers will get better, possibly much better in some respects, the further we look out into the future. Certainly, one would think there aren't a lot reasons that amplifiers should get worse. In general, we not only have advancing technologies (including digital and ever increasing software defined and software controlled capabilities) which could conceivably improve amplifier performance and possibly lower prices for a given amount of performance. Further, also due to digital technology, we have the Internet which probably makes 80% or more of previously known information about anything of value available to anyone with a web browser. So one would think that the various lessons of amplifier design knowledge accumulated throughout history aren’t going to be lost and if anything will become more widely available to anyone working on advancing amplifier performance.There are also some reasons to think that amplifiers won't get much better. In fact, if someone is of the school that says already today amplifier performance exceeds the capabilities of human hearing, then there would be little reason to expect improvements in amplifiers, much less reason to expect amplifiers to get “much better”.This question regarding future developments however is somewhat different than the question of “can anyone hear any differences between amplifiers?” I tend to think the answer is yes but I realize this is probably a minority position on this forum and possibly elsewhere. Having said that, I will be first to say that most differences in amplifier performance are subtle and in general tend to have a lesser impact on overall system sound than speakers. I’ve listened to enough integrated amps, power amps (solid state and tube), and receivers over decades to be pretty confident that not all amplifiers sound the same. Certainly, there can be differences depending on output power and depending on the how the amplifier works with the particular speakers being driven. But even with comparably powered amps and the same set of speakers two amplifiers can sound perceptibly different. Just my opinion, of course.One of the reasons I’m not so bullish on amplifiers improving much is that my experience indicates that hifi in many respects (overall, not just amplifiers) peaked around the 1970s. So when I say that I think I can hear differences between amplifiers it’s in part because some of my favorite amplifiers were made in the 1970s and I’m not convinced that many or even most of the amplifiers made since then do sound better, or even as good – although many are much more expensive today (even after accounting for inflation).My theory is that sometimes there is confluence of events that focuses society, business, and sometimes the military on a particular area of innovation and we get eras with relatively high performance, and in some cases peak performance that endures for a period of time. I’m pretty sure the Romans with their aqueducts and columns would be impressed with how far we have advanced cement technology, although while they would marvel at our interstate clover leafs I’m not sure they would think our clover leaf roads would be prettier than their columns. The point is that their cement technology (and other technologies) represented a peak performance that endured for many years (several hundred through the Dark Ages, and beyond in some cases).Likewise during the Renaissance history showed some innovation and peak performance in fields such as the arts and political thinking as well as some good improvements in physics.While hifi probably will get at most a page or a few paragraphs in history books compared to the chapters and books earned by the Romans and the Renaissance, I think the 1970s were a peak performance era and perhaps the “golden era” of hifi. Of course the 70s were built on developments in the 40s, 50s, and 60s. The 40s brought the end of WWII and freed up time and funds for new endeavors including early consumer electronics, and especially hifi (in addition to TV, automobiles, and home construction). The 50s brought the development of stereo vs. mono, and thereby brought a big improvement in "high fidelity" made possible by the spatial imaging attributes as described in our “imaging” discussions, not to mention stereo gave the industry a reason and a way to sell twice as many speakers.The 50s and especially the 60s brought huge advances in transistor design and commercialization while continuing to make many markets for tube electronics. By the time we reached the 70s we had large numbers of companies of all sizes from the U.S. and Japan competing for hifi market share. Additionally, by the 1970s the 1960s rock and roll music had been written and distributed widely on records giving an abundance of music to drive hifi consumption and usage. In parallel, the development of the electric guitar and guitar amplifiers drove incremental demand for similar audio technologies including a demand for ever bigger amps and speakers.By the 1970s a lot of intersecting forces created a crescendo for hifi. In my opinion the competition for audio amplifiers hit a peak in the 1970s as tubes were still popular and transistor designs had not yet become highly evolved. The concept of negative feedback was just starting to make it’s way into the consciousness of hifi enthusiasts and high powered amps (from Phase Linear and others) were just emerging. At the same time tubes were perhaps at their best, least expensive, and most abundant. One of my favorite power tubes, the GE6550 was manufactured by General Electric in a version called the JAN 6550. JAN stood for Joint Army Navy. The demand for tubes was still strong in the military and history has shown that military investments often lead to or enhance commercial products and markets.Of all these reasons for the 70s becoming a peak in hfii I would not under estimate the impact of rock and roll. The 60s gave us the Beatles, the Rolling Stones, Led Zeppelin, Jimi Hendrix, and a very large number of other bands that remain classics today. The demand to feed “hardware” with “software” was significant.So, what happened to hifi starting in the 80s? Computers and digital music among other things. Computers peeled off increasing amounts of people’s time; more time spent with computers and less time spent with hifi. Digital nearly killed vinyl but it took awhile to realize that just because music could be digitized it might not necessarily sound the same as or even as good as analog. Much but perhaps not all of the gap has been closed in this area. But mostly, overall, the addressable market for hifi manufacturers became smaller and smaller as new markets took mindshare from former hifi enthusiasts. And we shouldn’t forget to mention that many of us who grew up during this timeframe moved on to careers and families. If some things drove hifi adoption upward, the WAF wasn’t one of them.All of that is to say – somewhat to your point and the point of others here – amplifier technology has not been on a huge slope of upward improvement since the 1970s, and if you look at price performance it’s possibly declined. Having said all of that, I’m still not personally convinced that amplifiers contribute nothing to overall hifi system performance. I believe that while terms such as definition, air, transparency, imaging, and more while exceedingly difficult to quantify and perhaps impossible to measure are discernible attributes and it’s my experience (empirical evidence?) and my sense of history that indicate some of the best amplifiers of the 1970s if A-B’d side by side with many of today’s amplifiers might in fact produce surprisingly good and in some cases superior results, even with DBTs. I can’t prove the differences would be discernible and they probably wouldn’t be discernible for many people. So - I can’t prove these things, but heck some people were pretty sure the Sun wasn’t the center of the Universe in 1500 and even though they couldn’t prove it, their thinking turned out to have some merit.Just my impressions. YMMV, EF

author: David Hess
author_rank: Super Contributor
timestamp: November 06, 2017, 05:58:08 am
content: Quote from: Beamin on November 04, 2017, 11:02:10 amWith some exceptions, the component tolerances and variations are not important.A good example of an exception is aluminum electrolytic coupling capacitors if used; a bad one can increase distortion by orders of magnitude over a good one and good ones are expensive which makes direct coupled amplifier designs popular.QuoteAny improvement would be marginal. Good designs are tolerant of component variation.QuoteLarger capacitors and transformers are expensive. Good output transistors (ring emitter, perforated emitter, MOSFETs) are more expensive than poor ones. Using more output transistors in parallel to reduce Ft and hfe droop is more expensive. Using matched transistors is more expensive.Design is critically important. There is lots of extra complexity like cascodes and active loads which can be used to improve distortion performance and other things. Cheap amplifiers leave out protection circuits which under normal operating conditions do essentially nothing but you will sure notice if they are needed and not present. Or cheap amplifier include protection circuits which activate under normal operating conditions distorting the output when they should not because the design was not robust enough to start with. Layout is critical to prevent ground loops and even magnetic coupling between circuits.Quote from: Fungus on November 05, 2017, 08:35:52 pmBetter for whom?They are not going to get better as far as performance compared to the best amplifiers made except in efficiency. And they do not need to because as other have pointed out, amplifier performance long ago reached the point of diminishing returns as a contributor to system performance.I would settle for an amplifier which does not contribute significantly to distortion of the system as a whole, operates correctly under adverse conditions, is reliable, and is user friendly. That leaves out a majority of consumer level designs most of which cannot be evaluated anyway.

author: BrianHG
author_rank: Super Contributor
timestamp: November 06, 2017, 06:54:49 am
content: Quote from: hamster_nz on November 05, 2017, 11:01:17 amYes, this is how your ears judge position of the original mono source. Having your speakers in an awkward position does alter the imaged location of the instruments depending on their source stereo recording as well, however, I have my speakers facing forward at one end of my room and I don't seem to have any issues of where I sit, though, if I am in a different room, I don't hear any stereo imaging.Also, in many loudspeaker designs, the woofer crossover, being an LC filter usually, doesn't throw thing out of wack at different signal levels with it's physical interplay of it's case being physical cavity tuned object. My Clements RT7 got around this by using a simple RC filter instead, though, only being less than 50% efficient at transferring the amp's voltage to the woofer being a great loss, the 0 error delay which otherwise would have been there with a series inductor in place of the resistor was a trade-off to get a full headphone like imagery at all frequency ranges which would otherwise be lost, or difficult to engineer away elsewhere in the mechanics. (I did not know this until after purchasing the speakers, a fluke damaged resistor in the crossover which led to repair showed me this...) As for the woofer being a small 7 inch driver, with a super strong solid magnet, and a case with a huge tunnel tube with VLF whistle type of structure gives a flat 25Hz, or -3db at 20Hz response which will knock you out of your chair even with the loss of power. Too bad the speakers are no longer made since full range single woofer driver speakers today usually skip out on the mechanics needed for full range in trade for a dedicated sup-woofer which usually is mono and doesn't do good quality recordings any justice unless it is a really expensive tuned setup.

author: BrianHG
author_rank: Super Contributor
timestamp: November 06, 2017, 07:58:29 am
content: Quote from: Bicurico on November 05, 2017, 06:50:38 pmThis is a difficult point to illustrate. Quality and transparent definition as well as other factors means different things to different people, so, I will share a little story from my past (20 years ago).A friend in North Carolina, (I'm from Montreal, Quebec), was having a big birthday and he heard my audio equipment which he told his wife about. She wanted to get him a small high end sound system for his office/man-room and since I was in town, we visited the only closest local HiFi audiophile grade store. In the budget of 1500$ for a good CD player, amp, and speakers & cable, nothing the store had sounded good, even vacuum tube amps and speakers sounded right to me and I recommended not to buy anything from there. Just because you have an 'Audiophile' store, if any still exist in your area, doesn't mean they necessarily have anything good to hear. The store owner may just be just as fooled by Audiophile Audiophoolery to have chosen the cherries from the overpriced phony stuff, or, the store is in the business of profiting on said overpriced stuff.Anyways, we ended ordering a setup from Montreal from a friend's store and got a good entry level Jolida vacuum tube amp setup which really enjoyable and pleasing quality sound which outperformed what we heard with the way overpriced setup in North Carolina.From my own personal experience, I got close to this quality sound with my own Class B amp BJT amp design back in the late 80's. But, to do so, I had it biased so heavily that it's idle power consumption was over 150 watts and it's heat sinks make a nice space heater with I drove the full 100wattsx2 out. It is possible to make a good BJT of MOSFet amp which can deliver most of the transparency of a vacuum tube amp except for 2 aspects unless you engineer them in.1. Handling of signal levels outside the power rails of your amp. A tube amp softly distorts as you approach the rails instead of a harsh clip and perhaps a nasty inversion spike.2. The air feeling of some vacuum tube designs. This where minuscule parts of your audio source get a little extra signal gain even in the presence of the major large signal tones when driving a dynamic load like a speaker. An effect some tube designs are purposely designed to enhance.#2 can be an additional + as it may help compensate for the shortcomings of a physical speaker trying to generate all the signals it is being given.Now, designing an amp to replicate #1 means you will be limiting the maximum potential power of your amp to allow for that soft distortion curve at the edge of your power supply, so, why dot it unless it is your goal to correct for someone who turns up the volume too loud on a pre-amp which has a +x db gain over-driving your amp and you want to hide that problem. (Or, just get a remastered CD version which has the volume boosted with soft clipping in the source...)Designing a circuit to replicate #2 is much more tricky. To do it right, you need to respond to a speaker load in real-time making a mess of how negative feedback supposed to operate, and in attempting to do so will just mess something else up in the process. The vacuum tube is such a simple direct device that exhibit this behavior that it is probably the best way to achieve it. It's just not worth doing it any other way, if someone wants that sound, just buy a tube amp.Now, what I've said here is not Audio Phoolery like unidirectional speaker cable, special audio grade network cables, special audio cable burn in kits. There is a price and quality range which does exist. Some of the better and more expensive (not super expensive) equipment tend to be simpler in design. No AMP all in 1 ICs, in fact, the all round best I've experienced only have a few transistor/mosfets in the audio path with many paralleled output transistors, large heatsinks and oversized toroid with huge power filter caps.When it comes to peoples tastes, some people just want loud solid boom and noise as long as it doesn't distort. Most sturdy negative feedback designs will make them happy with an EQ on occasion. They also tend to prefer those cheap remastered versions of original music which has just the volume pumped up and at times some cheap anti-hiss filter with occasional treble and bass enhancement.Those who look for that transparency, who listen to recordings with that quality of mastering, will tend closer to Tube or Class A amps, where the output isn't as linear as a negative feedback design, but, the fine breath of air still makes it through even with some massive fundamental tones included with the sound.Those that want the best of both worlds, well, you can find affordable equipment to truly perform this feat, but, the louder you want to go, say above 100 watts per channel, the price does sky-rocket. If you are lucky to have a friend with an audio store who has great ears, gone through the gamut of amps and speakers equipment out there, and is willing to direct you in the right direction for your needs and sell you everything at cost, you are lucky indeed. I know I paid more than what many here would consider for 2 (stereo) speakers and an amp back in 1992, but what I have is still with me and cremates anything you can buy today at 1/4 the price. You know when one of my SciFi buddy friends finally came over to watch the new StarTrek movie on BluRay, and halfway through the movie, after being blasted into his chair, gets up an touches the case of the speakers and says these 7inch woofer things case just don't vibrate at all as the house shakes, and everyone sounds like they are actually there right in front of you, not like my cheap surround center channel speaker. With speakers and amp at over 20 years old, unless I'm willing to fork over 15K$US today, nothing will come close to replicating this experience, especially if you forget the Hollywood movie recorded junk and go for the Telarc CD recordings, even the first from the Telarc classical recordings from the 80s (their technical quality and equipment used is on a completely different level, their original normal CDs surpass most modern hidef recordings), this is where I get immersed into the music.

author: Fungus
author_rank: Super Contributor
timestamp: November 06, 2017, 08:51:29 am
content: Quote from: BrianHG on November 06, 2017, 07:58:29 am"Wind and piss", as Paul used to call it.(Paul was a HiFi salesman I used to know. The "wind and piss" crowd were a large chunk of his customer base)

author: Fungus
author_rank: Super Contributor
timestamp: November 06, 2017, 09:00:17 am
content: Quote from: Electro Fan on November 06, 2017, 12:31:32 amYep. 1970s was full of small companies trying to make good HiFi. 2017 is full of large companies trying to maximize profits.Amplifiers peaked around 1980, IMHO.Quote from: Electro Fan on November 06, 2017, 12:31:32 amI don't think anybody here is arguing that they contribute nothing at all. It's obvious that bad amplifiers can be built.

author: Bicurico
author_rank: Super Contributor
timestamp: November 06, 2017, 09:08:36 am
content: Hi,Returning a little to the original topic: can you build a high end audio amp for little money and why are the audiophile ones so expensive?From the experience I whitnessed with my friend, there are some factors:1) A good transformer is expensive, especially if custom made for the choosen design2) Good valves don't come cheap and may hard to source (especially older russian ones)3) The case is probably one of the most expensive items, as it has to look good, which means it needs to be produced professionally - not cheap at all.4) PCB needs to be sourced.Of course you could just copy an existing good design, get yourself a good deal on transformer and valves and then just build everything up in a crude case. Still, you are probably spending more than a cheap mass-produced amplifier.The basic principle of manufacturing any product is to provide the customer with the worst product he is willing to pay for, as any improvement will not see a higher return.Because most people are happy with loud bass, as long as there is not a too big distortion and you can disturb your neighbours. Such a one chip amplifier can be sold really cheap and is basically crap. Interestingly, it is not that much of a crap anymore, as the designs and transistors evolved.Now let's look at an audiophile valve amplifier:1) The store selling it to you has a price marging of at least 50%. No joke!2) How many of such amplifiers does a manufacturer sell? Not many, for sure. Still he has to produce, source the parts, give support, etc.Also, if someone is paying a few US$1000 for an amplifier it really has to look nice: anodized aluminium, wood, granith, whatever. Gold plated connectors and so on.This does not come cheap and has nothing to do with the amplifier performance, well except for the design of the casing, as it acts as a shield and will have impact on the sound - this requires some expereimenting, which costs money.Finally, for a commercial product, there is some basic certification that has to be carried out, too.Also, in case of valves, there is the issue with valve matching. No two valves are equal in performance, so you need a trained ear to select pairs from your batch of valves.Regards,Vitor

author: Beamin
author_rank: Super Contributor
timestamp: November 06, 2017, 09:43:12 am
content: Quote from: Fungus on November 04, 2017, 03:15:45 pmThey why spend lots more money on parts that the person will never see and the advertisers will never advertise? Its just bad business practice to add to cost putting something in your product and not telling anyone about. Look at how CD players have stickers for "1BIT ADC!!!" of course it has an ADC it wouldn't work without one. They advertise stuff that's nt even "stuff" like bragging that it runs on "BOTH voltage AND current" The other brand doesn't have sticker for that therefore it must not be as good.If your statement is true then EVERY single person that ever paid more then $79.99 and bought an amp other than the junk "Sherwood 100 watt stereo amp" or other low/bottom end brand at every electronics store is getting ripped off. And every manufacturer is needlessly adding costs to their product to see no return on their efforts. That sounds like phoolery.Go out and buy the shitiest low end 4 channel radio shack brand amp for your car turn the volume up and tell me it sounds the same as the big heavy ones that cost more. First thing is your speakers will sound muddy then clip then it will blow a fuse. But the exact same wattage amp that costs more just won't do that. No.

author: BrianHG
author_rank: Super Contributor
timestamp: November 06, 2017, 09:46:55 am
content: Quote from: Bicurico on November 06, 2017, 09:08:36 amThings are not what they used to be, especially here in North America, but when it comes to the higher end specialist equipment, yes there still is a larger margin compared to selling equipment like Sony, Pioneer, Panasonic, Samsung, JVC...If you want some of the better stuff, it pay's to have a good friend who owns such a store who would give you a good price.

author: Bicurico
author_rank: Super Contributor
timestamp: November 06, 2017, 09:53:48 am
content: On the premium audiophile equipment, the discount is often made at the cost of the manufacturer.The manufacturer:a) Provides i.e. 1-2 amplifiers for the "show room"b) The store tries to sell whatever amplifier the customer likesc) The discounted sales price is split between store and manufacturerd) The manufacturer gets paid when the amplifier is finally soldThe problem of the manufacturer is that nobody buys an audiophile amplifier without listening to it first. So there are four options:a) Work with audio stores, that rip of the manufacturerb) Be that exquisite, that the potential customers fly in to the manufacturer and listen in his show roomc) Be a manufacturer that ships sample amplifiers to potential customers or, better, fly to the customer to set up a listening session at the customer's home -> this can be quite expensive, especially if the sales ration is lowd) Be famous enough to have potential customers listen at their friends home, who already own that particular equipmentFrom this difficulty in selling high end audio devices, it is clear that the marketing/commercial price is probably as high or higher than the manufacturing costs.Cheers,Vitor

author: Zero999
author_rank: Super Contributor
timestamp: November 06, 2017, 09:55:24 am
content: Just want to point out, that I don't think any one here meant to say that all amplifiers sound the same, full stop. The truth is, that when an amplifier is poorly designed or operated out of specification, i.e. clipping, then distortion and noticeable differences between designs, will occur.The point is, the difference between well-designed amplifiers, operated within their specifications (no clipping), will be inaudible.As far as amplifiers peaking in the 1970s or early 80s is concerned, well that's not true at all. Improvements have been made since then in efficiency, cost reduction and size. All with no loss of perceivable audio quality. This is enabled longer battery life in portable speakers and mobile phones, which I know don't sound very good, but that's due to the physical size of the speakers, rather than the amplifier.I think what has happened since the early 80s is, lots of cheap, poorly designed audio crap has flooded the market and most consumers are unable to tell which is good and bad. Competition has been tough and audio amplifier companies have realised that marketing will generate more profit, than good build quality and design alone, hence the rise of all this audiophool nonsense.Want to build a reasonably priced, audibly transparent amplifier?DO NOT USE VALVES/TUBES!They're expensive and have high distortion.AVOID PASSING THE AUDIO SIGNAL THROUGH A TRANSFORMER!Audio transformers are expensive and bugger up the signal more than any decent amplifier.The easiest way to build an audibly transparent amplifier, is with a decent IC with a complete specification. If you just want 60W per channel, then the LM3886 will be more than adequate: much lower distortion than any valve design and most discrete designs. Use a good PCB design, with low inductance power traces and plenty of decoupling. Add a basic clipping indicator, so the volume can be reduced, before noticeable distortion occurs. That's it. You now have an amplifier which outperforms all that BS audiophool stuff, at a fraction of the price, but will still cost much more than a mass produced unit. At least you have the full specification though.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 06, 2017, 10:03:30 am
content: Quote from: Bicurico on November 06, 2017, 09:08:36 amIn my time it was closer to 100% margin. £300 retail price amp was sold to trade at about £150 mark. And it was completely made in the UK ( including the casework, pcb and transformer). And we've managed to make a profit on it.CheersAlex

author: BrianHG
author_rank: Super Contributor
timestamp: November 06, 2017, 10:12:36 am
content: LM3886 = 50watts peak into 8 ohm according to the data sheet, or, only around a true 35 watts RMS.You can run the IC right to it's absolute recommended limits of +/- 42v, but you might be playing with fire if your power supply isn't regulated, just a transformer with diodes and cap.Within 35 watts, yes, SNR of 92db + 0.03% THD (THD RATING ONLY AT 30 WATTS PEAK INTO 8 OHMS, OR AROUND 21 WATTS RMS according to data sheet), Yes, the first 20 watts on this chip should sound really good, probably excellent.Wait a minute, this chip doesn't actually sound too good at all... It's kinda weak crap, according to the data sheet.Actually, using 2 of these in a bridge configuration might expand this 21 watt sweet spot to 80 watts, but now you need 4 chips for a stereo amp and an additional input differential amp.I need an order of magnitude more power with 0.03% THD and with that power, 200watts, I now need a SNR of at least 112db so that when at low volume the first 20 watts is as good as the full volume & while at full volume, even the fine soft sounds below a watt still make it through with the higher power low frequencies mixed in. Making above 50 watts sound good under all circumstances begins to get difficult. The higher the power, the more difficult it is to maintain good fine control for the quieter moments yet not begin to distort as the audio power goes above 40% of the output rating.Don't get me wrong, the LM3886 has it's place and sadly, there are those who will use it claiming over 50 watts (like Chinese single board Amp manufacturers) super clear when the data in the sheet shows it begins it's breakdown after 21 watts RMS. But if you want to fill and shake a room with controlled refined sound, this chip is definitelynotthe way to go.Quote from: Hero999 on November 06, 2017, 09:55:24 amExcept for your word 'outperforms', the rest I cant argue here. If I want an excellent 20 watt per channel stereo amp, with an extra functionally OK 15 watts extra headroom, but still good quality, and I now know all the specs, I would use the LM3886 and not waste my time designing my own circuit.As for the price, It would still cost me a little bit for a good chassis for heatsink, toroid, decoupling caps needed to keep this chip fed with a power supply which wont sag too much under a potential momentary peak total 100 watt load. Not counting the 2 chips, PCB, speaker posts and RCA Jacks and machining the case, AC plug, fuse and all my time, this would still cost me around 150$. This would be a 200$-300$ project for a nice finished usable super clean 20 watt per channel stereo amp. I'm still not counting my assembly time, PCB cad time and debugging potential problems.

author: Zero999
author_rank: Super Contributor
timestamp: November 06, 2017, 10:54:34 am
content: Quote from: BrianHG on November 06, 2017, 10:12:36 amIt's not that bad. At +/-28V, 60W is specified in to 4R with a THD of 0.03% which will be undetectable. 50W should be possible, in to 8R, with similar THD figures but +/-35V is required.QuoteMy post wasn't a response to you personally. Up until now, no one has specified the power output or any specific designs/components, so I thought I'd throw on in there. My assumption was the amplifier is to be used at moderate volume levels, for home listening, not at a party (less critical anyway, since the amount of people talking and their bodies messing up the acoustics, mask distortion by the speakers/amplifier to some extent) or night club, and 20W or so should be adequate.Another thing to bear in mind is that, even with a perfect amplifier and speakers, the air is inherently non-linear and distortion will increase, at higher power levels, due to physics, irrespective of how good the amplifier, speakers and room acoustics are.https://en.wikipedia.org/wiki/Nonlinear_acousticsQuoteI agree that the LM3886 is no good in designs above 50W or 60W (unless external pass transistors are added to the output stage, which will be tricky to stabilise) but refute the claim it will noticeably distort at 21W, unless the power supply voltage or load impedance is too low. The graphs on page 11 the data sheet clearly show, the THD improving, until the power exceeds the clipping level, causing it to deteriorate, from then onwards. The clipping thresholds are clearly 70W into 4R, VS= +/-28V and 60W into 8R, VS= +/-35V. Unlike cheaper chips, the figures given are RMS, not peak power, which can be confirmed by calculations, using the saturation voltage and applying Ohm's law.http://www.ti.com/lit/ds/symlink/lm3886.pdf

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 06, 2017, 11:18:00 am
content: Hmm, component cost-wise my power amplifier circuit from 1997 (both channels, omitting the power supply parts) was cheaper than two LM3886 chips with a similar power output into 8 Ohm (40W RMS). I've used N-channel Logic Level MOSFETs for the output in a very linear Class AB follower configuration, and a simple (7 BJT) voltage amplifier to drive it.CheersAlex

author: BrianHG
author_rank: Super Contributor
timestamp: November 06, 2017, 11:18:28 am
content: Quote from: Hero999 on November 06, 2017, 10:54:34 amAt 20Hz only. Even I can achieve that with a small op-amp feeding the base of a darlington transistor in emitter follower setup.Even on page 1, under description, paragraph 1, the claim is 0.1% from 20Hz to 20Khz only at 38 watts into 8 ohms. This does not jive with the graph on page 11 where they are only showing you the distortion at the bottom 20hz which shows 0.005% THD at 38 watts. We are not being told the whole story here by TI and you are ignoring something fundamentally important. (I'll wait and see if you can figure this one out for yourself (it's a doo-seee)...)Ok, I know how you feel about audio phoolery stuff as there is a vast cloud of such BS crap, but, there does exist quality 50watt amps you can buy, where when they say 50watts 0.01%THD, they mean that at all frequencies and loads. Most of the time, these amps are basic and only have a single volume control. But, such devices are not in demand and they do cost $$$.But like I said with this chip, and TI's pdf on page 4 states:30W, RL = 8?, 0.03 %20 Hz ? f ? 20 kHzAV = 26 dBWith this spec, I take as their sweet spot for this IC, and looking at the built in current limiter charts and how they may react to a speaker's inductive load, I will still say that you should power this IC for 50 watts peak, expect a super clean 21 watts RMS. And my assessment of building a nice amp chassis with heatsink, good toroid and nice decoupling caps to support 2 channels with 50 watts peak, PCBs, and all other parts, this is still a good 200$-300$ project to create something respectable enough and safe to put in my living room.

author: paulca
author_rank: Super Contributor
timestamp: November 06, 2017, 12:00:15 pm
content: Quote from: macboy on October 31, 2017, 01:35:10 pmI bought a Sony HiFi way back in the 90s. It was in the era when Awia were pumping out HiFi's claiming things like 300W RMS. Utter tosh. Sony countered with an HiFi that could actually meet it's rating of 80W RMS / 160W peak a side. They also matched it with speakers that could take that (and probably more). I don't have the THD figures handy, but having owned an Awia(sp?) it distorted long before it go anywhere near it's half way mark.Word on the market was it was very much under priced just to piss Awia off.I still have that amp. I can't find any reason to buy a different amp. It will have my neighbours at the door when the pictures start rattling on the walls and it still has room to go.I have tortured it. I once connected a really cheap karaoke mic to it to provide sound re-enforcement for a presentation. Due to the low gain of the mic on the podium and the distance from the speaker I had all the gains at MAX. Some bright spark clown grapped the mic and shouted "OWW!" into it. CLICK. The amp just went into protect for 2 seconds and came back on without a hint of being upset.I used it at a large party to provide music in the kitchen/dining room while the main party speakers (an 800W professional hired rig) did the front room. I simply left the volume at max and let the DJ run it however hard he wanted from the mixer (I know... one does not do this!) Not only did it sound amazing, survived, but most people said the sound was better in the kitchen than in the main room.Still not getting rid of it. The CD unit died about 10 years ago, the tape decks (yes it had those) were removed and binned about 8 years ago, so I just have the top half of the HiFi remaining. The rotatry encoded volume knob and EQ adjust don't work anymore, but the remote does.It sits left on 100% of the time in my living room set to high levels of 24 of 32 on the knob. You simply can't here it, not a single hiss, buzz or humm, but drop a movie Dolby intro and things start vibrating at some serious power.Yes, there are probably better amps, but I kill amps pretty regularly. This one seems to survive me so I'm keeping it.

author: paulca
author_rank: Super Contributor
timestamp: November 06, 2017, 12:14:46 pm
content: On modern amps what peeves me is the "Nanny state" limiters. Particularly headphone amps.I used to use a TRIO 1980s amp to drive headphones. If I so choose I could turn the volume up enough to melt the drivers in seconds. Believe me, I did. I think that amp ate about 4 pairs of panasonic and techniq headphones in the 5 or 6 years I had it. Until one day it just stopped working.These days you'll find it hard to get that. Most headphone amps are limited to "sensible and health listening levels". What is the point in listening to rock music at sensible levels? Seriously.If want to destroy my ears then I should be allowed to. If I want to turn it up far enough to set my headphones on fire, then I should be allowed to.The other thing that makes me chortle is PC Speaker systems. Like these:https://www.ebay.co.uk/itm/Genius-SW-F2-1-500-speaker-system-for-PC-Laptop-MP3-12W-R-M-S-500W-P-M-P-O/282616179220?hash=item41cd3caa14:g:d0MAAOSwhGlZl0H412W RMS... possibly. 500W PMPO, LOL I have seen speaker sets advertised as 1500W but the power pack supplied delivers 10W. They simply make stuff up!As to what makes a good amp. I think it comes down to quality of the components used, not just the amps and transformers but the support components too. 99% of domestic goods HiFis will use the cheapest white label components they can get. The bean counters will strip anything they can back. Then there are the support features, such as sensible limiters, multi band limiters, sensibly multi-staging with sensible gain structures, proper protect, rugged-ness, thermal protection, clean PSUs, RF shielding, gold plated connectors etc. etc. etc.The price of high end professional amps isn't really due to the overall price of their components and design, it's more about what audiophiles will or expect to pay for it. This is not limited to Audio gear. Take any market and search ebay for a price ranges and you will find a wide range of prices for the exact same item. People will log in and say, "I want to buy a thingie", but..., "I expect to get a good quality one it will cost at least £100". So they set a minimum price of £100. Chances are the very same seller is selling an item for £10 and £100 and maybe £200 and people will pay that, because it's what they expect to pay. They believe they always get what they pay for.High end gear also suffers from low demand. However if you need a guaranteed quality amp for a studio you will have to pay for it. So they can charge, sometimes, insane amounts of money to get something that is 2% better across the spec range than an amp costing a 1/10th as much.

author: BrianHG
author_rank: Super Contributor
timestamp: November 06, 2017, 12:27:27 pm
content: Quote from: paulca on November 06, 2017, 12:14:46 pmSeparate from this topic, start a new headphone amp project thread. You may find an affordable solution here among us which might be affordable and please you......

author: cdev
author_rank: Super Contributor
timestamp: November 06, 2017, 12:29:51 pm
content: What are the best amp modules for use with external line level audio, like from a computer?I would like to upgrade my computer's sound with a clean powerful quiet audio module that can use my power supply and drive good external speakers.. like the small Dayton Audio ones without distortion.

author: paulca
author_rank: Super Contributor
timestamp: November 06, 2017, 12:31:32 pm
content: Quote from: BrianHG on November 06, 2017, 12:27:27 pmThanks. I got a digital Projekt Box headphone amp which I love, massive range, quiet, clear, the only downside is, I wish it would go just a little bit hotter beyond it's nanny state limiter. I'm not entirely convinced I could do better with my electronics skills.

author: BrianHG
author_rank: Super Contributor
timestamp: November 06, 2017, 12:41:36 pm
content: Quote from: cdev on November 06, 2017, 12:29:51 pmNeed more info... Are you talking about just using your PC's 12v? If you want a total of 50 watts out, is it good for you PC's +12v rail's load to jump around 2-52watts with the sound?What kind of speakers? 8 Ohm speakers need a respectable voltage to give you a reasonable power. Like +/- 24v for a unbalanced output amp, or +/-12v for a bridged balanced output amp.A bridged DC 12v class D amp module from China should give you around 15 watts peak per channel into an 8 ohm speaker, or, 35 watts peak per channel into a 4 ohm speaker. (This is a respectable power output for PC speakers which will fill a small room really good)

author: alexxa
author_rank: Newbie
timestamp: November 06, 2017, 01:03:52 pm
content: Quote from: b_force on October 31, 2017, 06:05:41 pmThe QUAD 303, from 50 years ago, was one of the first all-transistor audio power amplifiers to be essentially 'perfect' (within its bandwidth, voltage swing, and load impedance limits, of course). Any more recent design that isn't 'perfect' has been done wrong!

author: Zero999
author_rank: Super Contributor
timestamp: November 06, 2017, 01:29:48 pm
content: Quote from: BrianHG on November 06, 2017, 11:18:28 amThe distortion does rise at higher frequencies, but I don't see what all the fuss is about? TI don't seem to have manipulated the figures at all!I admit I missed the fact the figures I quoted were only for 20Hz, but plenty of data is provided for higher frequencies. Refer to figure 18: +/-35V, 50W into an 8 Ohm load, THD+N <0.01% <10kHz, rising to slightly above 0.013% at 20kHz, but no one will notice the harmonic distortion above 10kHz because it will be ultrasonic. The situation (figure 16) is worse with a 4R load but still <0.03% THD+N to 20kHz. Graphs are also given at 20kHz, see figures 25 to 27, which is helpful, but beyond the audio band.If anything TI have made the figures look worse than any one can hear. The THD+N tests were conducted with a bandwidth of 80kHz, which is excessive, because it will show a lot of ultrasonic harmonics, which no one will hear. If the bandwidth were lowered to actual audio range of 20kHz, it will more accurately reflect what can be heard, rather than what can be easily measured.

author: paulca
author_rank: Super Contributor
timestamp: November 06, 2017, 01:40:45 pm
content: Quote from: cdev on November 06, 2017, 12:29:51 pmI would recommend an external digital USB amp (ideally optical rather than USB, but you will be lucky if your Mobo supports it and the drivers do, so might need a PCI sound card). From £50 to £5000 it just depends on what you want to spend. I wouldn't recommend trying to use the PC power supply. They are incredibly noisy devices, lots of HF RF around, this is why internal sound cards and mobo built in audio chips are so noisy. I expect a lot of that noise will end up on the high power 12V rails too. I wouldn't put the amplifier inside the PC case either. I'd put it as far away from the PC as is practically possible.For speakers I have something similar to these for gaming and desktop use. Cheap, cheerful, great sounding, but they are designed for close range listening, they lack bottom end and roll off completely low down (not suprising for 4-5 inch speakers) . They were a present.https://www.djstore.com/numark-nwave360?gclid=Cj0KCQiArYDQBRDoARIsAMR8s_SwIrg5LckfvhsgCEBmLrFdxkW5nxiHAU-Da_n7Ki78HgjDfTxdmjoaAlzhEALw_wcBIf I had of bought them for myself I would looked at similar but with a USB digital or optical input and a 2.1 configuration. Nurmark make "okay" hobby/semi-professional DJ equipment. Denon make higher end stuff, either will probably do a digital input set of active speakers. If you've got the money there are much higher end stuff, but what they really give you over the £100-£300 range, I can't say.

author: BrianHG
author_rank: Super Contributor
timestamp: November 06, 2017, 01:47:37 pm
content: Quote from: Hero999 on November 06, 2017, 01:29:48 pmOk, I did not mean that TI messed up the figures. I'm saying that combine everything together, and the first paragraph printed on line page 1 description paragraph #1 says:QuoteThis is not a lie. It's the first thing written right there! In the .pdf data sheet.Your distortion charts are under controlled circumstances with a controlled single signal.You still haven't figured out why the distortion is going down instead of up with power at the very low frequencies. You are missing something you actually know about which is important and you just need to flex that grey matter to figure out why. (I provided a hint in my first post pointing out what's going on...)Once you do so, you will know the consequences on the sound output driving a speaker load and I'll show you easily how this error makes an audible effect.Actually now that I think about it, you are missing 2 things that are important... (I'll be back when I wake up & see what you or anybody else have come up with...)

author: Electro Fan
author_rank: Super Contributor
timestamp: November 06, 2017, 04:03:03 pm
content: Quote from: Hero999 on November 06, 2017, 09:55:24 amRoger that, and AgreedQuoteMight depend on the definition of a "well designed amplifier" but in any event you are certainly entitled to your opinion.QuoteNot sure what the definition of "reasonably priced" is, but DO NOT USE TUBES and AVOID OUTPUT TRANSFORMERS? Really?

author: Zero999
author_rank: Super Contributor
timestamp: November 06, 2017, 06:38:08 pm
content: Quote from: Electro Fan on November 06, 2017, 04:03:03 pmI agree, reasonably priced is a subjective statement, but I stand by my comment about tubes and audio transformers being incompatible with a low distortion amplifier. Transformers are expensive, big and bulky things which introduce plenty of distortion and getting one to work down to 20Hz, requires it to be three times as big as a 60Hz mains transformer, of the same power rating as well as compromising on high frequency response. Solid state devices avoid the need for an output transformer and achieve less distortion.Quote from: BrianHG on November 06, 2017, 01:47:37 pmAt lower power levels there will be more crossover distortion and noise, relative to the signal. Is that what you're referring to? Yes I did miss that, but still isn't bad. Extrapolating the the worst graph, figure 25, shows only 0.06% THD+N at 10mW (hardly audible at any frequency through an inefficient, high quality speaker, let alone, at the ultrasonic frequency of 20kHz).That needs to be put into perspective. What sort of dynamic range does the source have? Suppose it's a CD (vinyl will be much worse) and you've got the volume set to a reasonable level, yet the output is low because a quiet part of the music is playing, there'll be more distortion due to the quantisation error of the DAC, than the amplifier, at those low power levels.

author: Fungus
author_rank: Super Contributor
timestamp: November 06, 2017, 06:48:11 pm
content: Quote from: cdev on November 06, 2017, 12:29:51 pmMe? I'd get a USB sound output gadget and some active speakers.USB sound is easy:https://www.ebay.com/itm/291614747261(Randomly chosen seller, search for [img=https://www.ebay.com/sch/i.html?_nkw=pcm2704]http://PCM2704 to find more like it)I own three PCM2704 devices and the sound from them is impeccable.For under $10 you should have plenty left over for some nice speakers.

author: Buriedcode
author_rank: Super Contributor
timestamp: November 07, 2017, 02:49:45 am
content: Just wanted to second the USB soundcard option. Whilst it'll most likely have more jitter than an on-board/built-in soundcard (since the sample rate comes off a PLL driven by USB timing) it isn't noticeable, but the fact its 'not in the PC' reduces noise. This doesn't' seem to be much of a problem these days, but there is a definite difference between internal and external when recording.Even the $1 CM108 USB soundcard dongles sound pretty good.

author: BrianHG
author_rank: Super Contributor
timestamp: November 07, 2017, 09:23:49 am
content: Quote from: Hero999 on November 06, 2017, 06:38:08 pmCrossover distortion, you figured it out #1. Now, you seem to be missing it's importance and consequences with regard to music, how we hear, and signal power at different frequencies which pertains to #2. So, I went through the trouble of setting up this little illustration & audio clip. I deliberately chosen cheap pop-rock recording, normal CD quality, originally taken from a cheap .mp3. No fancy soft and loud classical music with wide dynamic range.See illustration and optional listen to 2 second clip.As you can see, at the higher frequencies, the audio power is a fraction that of the low frequency bass drum, yet, it still sounds as loud to our ears. In fact, looking at that full power 50 watt signal where the high hat cymbal drums mixed in, which you can hear in the audio, is recorded at full volume. No tricks, yet, the audio power of the above 8Khz signal, well within .mp3 and 16 bit 44.1k audio, has only peaked at 0.7watts and still has definition at 0.06watts. According to TI's specs, lower power and higher frequencies is this IC's weak spot. (Hint: When testing aligning audio systems, Pink noise is used instead of white noise... Now what's special about the 'Pink Noise')What would happen if I want to lower my volume at 12 watts today, not full blast 50. The first cymbal strike would go from 0.7w down to 0.17w and the second would be down at 0.015w. (I can still clearly hear both High Hat cymbals in this song on my equipment at 12w, 50w and full 250w, though I don't own any audio equipment which shows any cross-over distortion on my scope...)Remember, crossover distortion is not a slight smooth angular bend or deformation along the full signal voltage in the output signal vs the input signal. It is 2 small 'voids' where neither the NPN or PNP output transistors are driving the output with anything for a short time making a flat linear void where these small wattage signals can sit inside. For the most part, depending on how bad the crossover distortion is and how temperature of the IC amp usually shifts that about makes this IC great for box radios, some TVs, ghetto-blasters, compact all in 1 systems, but I find this un-acceptable for high fidelity solutions. Especially that enough Class B and Class AB designs exist which can do better, except that the ones without such distortion tend to get hot and need a big heatsink. And if you want peak minimal distortion at low power, and a bit of linearity loss at full power, but still no sign of any small signal being chewed up by the big bass drum strikes, Class A amps are the way to go.This lastly brings me to Class D amps. In theory, a properly designed Class D amp shouldn't have this problem. In an attempt to visualize the working mechanics of a class D amp, The crossover problems should be gone, but, what seems to happen is as the output power goes up, refinement transfer of the source signal does get sloppy, or worse. I do have a few Ideas on how to design a clean musical class D amp at all power levels, but, it means using expensive GAN fets for speed and low rds figures and FPGA components and building everything from scratch the amp would only have digital in.

author: BrianHG
author_rank: Super Contributor
timestamp: November 07, 2017, 09:35:21 am
content: I wonder If I tried my classical music CDs as an example. One good symphony bass drum strike going down to a small quiet section with flutes and chimes. The power difference would be 50 watts to something like 0.1w in the quiet section's higher frequency tones... I choose not to because I realized that would not be fair for mainstream normal pop music recorded mainly all at one volume and it wouldn't even show on the screen capture. You would only get a single pixel wiggling up and down during the quite passage.

author: alexanderbrevig
author_rank: Frequent Contributor
timestamp: November 07, 2017, 09:37:23 am
content: What makes an amp better?This is my list:Appropriate wattage and headroom for its application.Sane thermal shutdown.Can handle abuse even if handled by ignorant voluntary 'stage hands' or visitors.THD an order of magnitude less than the speakers it's driving.Appropriate frequency response for it's intended load.Bridgeable for double power, easily configurable.Phase linear DSP EQ.A good power supply (yes, most of them has more than sufficient PSU but I've encountered cheapo Behringers and Peavy, they got noisy when sharing grid with incandescent dimmers), for home use it should reject noise imposed by things like ethernet over power grid (again, not all do for some reason).Do amps sound different? Absolutely.Do I hear a difference between a properly powerful Crown vs a properly powerful Crest or QSC (assuming no DSP and driving the same speakers)? Absolutely not.

author: paulca
author_rank: Super Contributor
timestamp: November 07, 2017, 10:19:20 am
content: Quote from: BrianHG on November 07, 2017, 09:23:49 amNot sure if it effects your results, but mp3 is, of course, lossful. One of the first things it does it divide the frequency spectrum into 64 (I think) bands and throws the odd ones away, dropping the bandwidth requirements by 2. This is called physco-acoustics where the ear will generally ignore frequencies either side of each other. It then quantises the remaining waveform using DCT. This is further quantized in the time domain. The result, at high compression levels, results in quantization noise artifacts. Thus, not only does it remove a lot of the signal at encoding time, but adds other stuff back in at decode.An interesting experiment is to run an MP3 encode, decode, encode in a loop, so you encode, decode and re-encode a sample multiple times. Surprisingly it will eventually reach a stable state where it cannot be quantized further and subsequent re-encodes lose nothing more.

author: BrianHG
author_rank: Super Contributor
timestamp: November 07, 2017, 10:31:28 am
content: Quote from: paulca on November 07, 2017, 10:19:20 amJust wanted to show that even a .mp3 can have a wide dynamic response, with soft cymbals miked in with a powerful bass drum kick, well, in the 8Khz region anyways and still can be used with a good amp. I know .mp3 would drop out above 12khz for low bitrates, 15 khz for high bitrates when there is a good bass drum and snare drum strike where a .flac or cd original doesn't exhibit this flaw in the transcoding.

author: Fungus
author_rank: Super Contributor
timestamp: November 07, 2017, 11:40:38 am
content: Quote from: Buriedcode on November 07, 2017, 02:49:45 amA PLL might not be exactly the desired frequency but they're reasonably stable, they don't cause jitter per se.Those TI PCM2704 chips have a "unique system that recovers the audio clock from USB packet data. On-chip analog PLLs with SpAct(tm) enable playback with low clock jitter".Presumably that means they look at the rate the data is arriving over USB and re-tune the PLL accordingly, over time.Whatever they're doing, it seems to work. I own three different variants of those things now, that blue one is a good choice for connecting to active speakers (and also has optical/coax, etc. if you need them).

author: Zero999
author_rank: Super Contributor
timestamp: November 07, 2017, 01:02:42 pm
content: Quote from: BrianHG on November 07, 2017, 09:23:49 amAlso there's no mention in the data sheet of how much of it is crossover distortion, how much is due to general non-linearity and lastly noise, which will also become greater, at lower power levels, but might be more subtle.How noticeable will it be though? And how much will the amplifier contribute to the distortion, compared to what's already there in the audio source?Let's look at the figures, assuming 0.1% THD+N, which is worse than any of the figures given on the LM3886 graphs:10mWPout, 0.1% THD:VP@10mWPin to 8R:VP= (P*R)0.5= (0.01*8)0.5= 0.2828VPVP(ERROR), 0.2828VP, 0.1% THD:VP(ERROR)= VP*0.1% = 0.2828*0.001 = 282.8*10-6VIn the example with 50WPVP= (P*R)0.5= (50*8)0.5= 20VPLet's look at the quantisation voltage of the DAC, using a 16-bit per sample, which in this case will be used to represent the full +/-20V swing, of the amplifier.VPP= 2*VP= 2*20 = 40VPPVERROR= VPP/(2BITS) = 40/216= 40/65536 = 610.35*10-6VThis is over double the 282.8µV error of the amplifier and how many 16-bit ADCs and DACs have an error and linearity as good as 1-bit? Other factors are: the recording is unlikely to use the entire 16-bit dynamic range and the distortion in the recording equipment. The real error in the source will be worse than 610.35µV error on the output of the amplifier!One could argue that listening at low levels, will cause the amplifier's error to dominate, but the real THD of the LM3886 is much lower than 0.1%, which was an extreme example and the quieter the sound is, the chances are greater that the errors will fall below the threshold of hearing.I agree that class D amplifiers are not free from distortion but if it's better than what can be recorded and heard, then they're transparent and contribute no additional distortion to the sound.Conclusion: the sound is only as good as its source and one's hearing, no matter how good the final power amplifier is and aiming to be significantly better than that, results in diminishing returns, until it's pointless.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 07, 2017, 01:30:12 pm
content: Quote from: Hero999 on November 07, 2017, 01:02:42 pmThat is presumably correct if you are listening only to continuous sine waves, used in all quoted measurements. If that is what you do, I have no objections.CheersAlex

author: macboy
author_rank: Super Contributor
timestamp: November 07, 2017, 02:52:20 pm
content: Quote from: David Hess on November 02, 2017, 02:11:04 pmCertainly the FTC has set forthrules. And it even reviews and updates those rules over time. But ever since 1974 and still today, those rules are so loose and broad that there is no chance whatsoever of direct comparison between amplifiers based on the numbers produced. The FTC rules allow the THD level and the bandwidth to be at the discretion of the tester (manufacturer). This means that the el-cheapo will be tested at 1% THD and 1kHz and might have a chance of producing decently high power numbers within those limits. The respectable manufacturers will set their limits to say 0.05% and 20 Hz - 20 kHz, and a different (always lower) power number will fall out of the testing due to the more restrictive specs. Both tests meet the rules, because the arbitrarily chosen THD level and bandwidth are given alongside the power rating.The FTC "Rules" are not "standards".

author: Zero999
author_rank: Super Contributor
timestamp: November 07, 2017, 03:22:23 pm
content: The more I look into this, the more pointless aiming for ridiculously low levels of distortion and noise becomes.How noisy is a 500R dynamic microphone?VN= (4kTBR)0.5k = 1.3803*10-23T = 20oC + 273.16 = 293.16oKB = 20kHzR = 500OhmVN= (4kTBR)0.5= (4*1.3803*10-23*293.16*20*103*500)0.5= 402.32*10-9VTaken from the data sheet of a random dynamic microphone, the sensitivity is -72dB/Pa (94dB SPL).V = 10(-72/20)= 251*10-6Vhttps://www.bogen.com/products/pdfs/microphonespdfs/HDO100m.pdfSuppose we want 10V out of the amplifier, full range, when the peak sound into the microphone is 94dB.AV= 10/251*10-6= 39.81*103The noise, due to the 500R microphone on the amplifier's output now is:VN= 39.81*103*402.32*10-9= 15.61mVMuch higher than the figures discussed above.

author: David Hess
author_rank: Super Contributor
timestamp: November 08, 2017, 04:21:06 am
content: Quote from: BrianHG on November 07, 2017, 09:23:49 amThe idle current for a class-AB design is insignificant compared to the output power. The large heat sink is required regardless.Good designs are closer to class-B than class-AB because of a problem different than a "void" in gain at the crossover point. When both output transistors are conducting, gain is contributed by both so the gain (transconductance) actually increases producing a hump at zero output current. It is not the lack of gain at crossover but the variation in gain at crossover and on either side which produce something like a flattened "W" in the transconductance graph.A further problem is that the bias conditions change with temperature so after a period of high output power, the transistors are hotter and the bias voltage should track to maintain the same idle current. Bias circuits are designed to track temperature but it is the temperature of the transistor dies which matters and there is a considerable time lag between them and the heat sink or even transistor cases. Better designs mount the temperature sensor as close as possible to the transistor dies to minimize lag in the temperature measurement. I usually epoxy the temperature sensing diode directly to the transistor's metal tabs.In a good design, the output stage contributes a majority of the distortion because of the above difficulties and there are all kinds of clever ways to handle them. On semiconductor and others produce transistors which have a temperature measurement diode mounted adjacent to the transistor die to minimize thermal lag. MOSFETs have less temperature dependence to start with although this comes at the expense of lower transconductance and higher cost. Local feedback around the output stage can reduce the magnitude of gain variation at the cost of complexity.Simple class-AB designs can achieve 0.1% distortion worst case. Complex ones can achieve 0.001% or better. Considering how much distortion large signal audio transducers (speakers) produce, it is very easy to reach the point of diminishing returns as far as distortion and it might be better to spend more on better source material, a better environment, better speakers, and handling clipping or at least not making it worse. That last thing is difficult to convey in technical specifications however and marketing is too untrustworthy to believe.

author: BrianHG
author_rank: Super Contributor
timestamp: November 08, 2017, 08:20:39 am
content: Quote from: Hero999 on November 07, 2017, 01:02:42 pmOk, we are getting closer to the same page, though we don't need such a great overall THD or linearity, since this is not how our ears work, lets take a look at a real LM3886's measured crossover distortion and it's actual consequences. You are going overboard with respect to super low THD and super linearity into the realm of I dare say audiophoolery. What I am showing here is that even with a great 0.01% overall THD, even in the bounds of my above audio example at both my volume choices, an audio effect is being produced which has an effect on the sound because of where it sits and how it is shaped.Here is a 1Khz sound driven though a LM3886 driving a 6 ohm speaker:(Scope shot taken fromhttp://www.diyaudio.com/forums/chip-amps/312970-lm3886-crossover-hf-noise-output.html)This distortion bend at +0.5v and -0.5v has a visible angle edge to it. Even at full volume, during the the 'Releasing open the high hats cymbals part open', the middle 25% of that amplitude in my audio sample would play at a lower volume getting a boost once it passed that 0.5v threshold. In fact it actually worse. Inside, that 'cross-over' zone, since there the driving transistors aren't fully conduction, the response at higher frequencies like 8KHz may only spike through during the loudest peaks while the rest being muted even more than the example 1KHz tone. If I lower the volume of my example volume to 12watts, 50% of the amplitude of the 'Releasing open the high hats cymbals part open' section of the music would sit in that distortion zone creating a sort of gated noise filter audio effect. (IE, cut down the volume, especially treble, when there is too little sound to remove hiss, kick up the volume once you pass a threshold. Though the effect may make your source audio appear to have less background noise in it than it actually has...)Now as to your comment about achieving that unnecessary impossible THD linearity with your source dac and an amp too perfect, what if I were to tell you that I can make an amp with only 2 active components, whose only 'sin' is that the overall THD is 0.3%, much worse than the LM3886 because the output looks like this:In this hand drawing approximation the blue represents the 0% error THD desired output, a full 50 watt signal. The red represents the output of my lousy 0.3% THD amp. With this cheap 2 active component amp, the distortion is curved and smoothed across the full 50 watt output. But, now, I can guarantee that no matter circuit temperature, it is impossible to mess up the medium-small signals like in the 1KHz LM3886 scope-shot. In fact, the middle 12 watts, even the middle 25 watts would have a THD comparable to the LM3886 and the error at the high powered edges wouldn't even be noticed by the ears since our ears have difficulty hearing fine true linear definition at high power levels, especially if the transformation is a smooth gradual transition over a wide span. The key feature is that middle unpredictable gap in the LM3886 is impossible in my design.

author: BrianHG
author_rank: Super Contributor
timestamp: November 08, 2017, 08:46:20 am
content: @Hero999, as for your DAC THD accuracy. If my audio sample is at full range, 16bits, =+/-20v, the high hats I measured at +/-2v will have a resolution of 13 bits. Are you saying that a 16bit dac can't produce any reasonably definable audible sound with 13 bits. Even the tiny 0.7v high hat cymbals release has over 11 bits of definition in it. In fact, using my cheap Amiga 8 bit audio sound, the high hats would still contain 3 bits of definition, with a little background hiss from my cheap 8 bit sampler, it would still generate an recognized audible cymbal sound. A 12 bit dac on a 1$ MCU today would roast that definition. Your don't need to micromanage every single last 1 bit from an audio DAC like it is instrumentation. The original uncompressed 16 bit CD audio if 'Phenomenal' and way beyond most of our hearing capabilities, though, it is nice to find the penultimate reference if you really want that perfect linearity over the full voltage and frequency range. A slight linearity error from -1v through +1v leading to a lowered THD would never be noticed by the ear.However, that 'cross-over distortion hole' in the sound, our ear can fill in some of the missing sound and you can get used to it, but, get rid of it completely, now your audio amp will have a transparent open window/air quality of sound, with real refined image at any volume with any type of reasonable audio source starting from even around 1/4 the quality of a CD and up...

author: Gyro
author_rank: Super Contributor
timestamp: November 08, 2017, 09:38:38 am
content: Quote from: David Hess on November 08, 2017, 04:21:06 amThis is one area where integrated devices like the LM3886 actually benefit, as they have direct temperature monitoring of the output device junctions. I remember Sanken (I think) producing audio output transistors with on-die monitoring too, unfortunately heavily faked.Already mentioned in another thread, but there's a reasonably detailed spice analysis of the LM3886 and the effect of circuit parasitics here...https://www.neurochrome.com/taming-the-lm3886-chip-amplifier/stability/Interesting to see the effect of fairly small layout inductances on phase margin.

author: Zero999
author_rank: Super Contributor
timestamp: November 08, 2017, 09:50:49 am
content: Quote from: BrianHG on November 08, 2017, 08:46:20 amI agree with you about the number of bits required for audio transparency. In many cases 8-bit per sample is enough, especially for lots of pop music which is heavily compressed to start with.QuoteI see your point about cross-over distortion.Regarding the output waveform from the LM3886 posted in your previous post: they look pretty bad, but we can't be sure genuine parts were used. The person conducting the test ordered some LM3886s off a random ebay seller, so could be fake. I'll reserve judgement until I see an example using genuine parts.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 08, 2017, 10:20:56 am
content: Quote from: Hero999 on November 08, 2017, 09:50:49 amIf you follow the link you'll find that these distortion were caused by an insufficient current drive on the MUTE pin. A nice picture to illustrate a concept of crossover distortion but it does not show real distortion of a properly operated LM3886.CheersAlexP.S. - just FWIW, I do not consider "CD quality" (uncompressed 16bit 44.1kHz) as sufficient for Hi-Fi audio. As a fact I am rarely (maybe once a year) listen to this kind of quality. My CD player is off almost all the time, I listen to vinyl and tape, occasionally to hi-res digital files (24bit 96kHz).

author: BrianHG
author_rank: Super Contributor
timestamp: November 08, 2017, 11:04:23 am
content: Quote from: Alex Nikitin on November 08, 2017, 10:20:56 amGood catch. The super tiny print of the middle paragraph explaining the solution led to me ignoring that point. I guess the LM3886 is better that what was shown. Now I need to find a proper setup driven to it's 50watts peak capability to see how it performs under load, not just at 1w like shown in the scopeshot.QuoteThe high end CDs I have, like from Telarc Records, do appear to be well mastered. Though I would not call them the penultimate HiFi, I cant argue that 24/96 hires audio has obviously 256x the dynamic amplitude while for most of us with 20Khz to hearing bandwidth, the extra 2x oversampling, or better 4x of 24/192 audio does produce much better defined captures of things like that initial striking hit of a drum stick on the Ride and Clash cymbals, and you can much more clearly make out the positioning and imaging of the set of springs rattling under the snare drum, which only just barely make it through on better quality CD recordings. On regular cheap modern pop recordings, the springs are more akin to a simple white or pink noise.

author: paulca
author_rank: Super Contributor
timestamp: November 08, 2017, 11:32:26 am
content: Quote from: Alex Nikitin on November 08, 2017, 10:20:56 amI think it depends on what you want to do with the recording. If you want to listen to it, then 44.1k 16bit is absolutely fine. In fact in many, many tests it has been proved that HiFi aficionados do NOT correctly identify higher rate recordings with any statistically significant margin.To argue against it argues against physical limitations of our ears and physco-acoustics. Remember what you think you hear bares a lot less resemblance to what the sound actually is than you think. It also argues against sampling theorem. Our ears are evolved for specific purposes, hunting, defense and language, not listening to music, so they suck at the later.The dynamic range on tape... depends on the tape, domestic tape is useless compared to high speed, wide, studio mastering tape for example. The quality of vinyl is orders of magnitude lower than CD in particular due to noise and groove width. To get the same range as CD you would have to make the groves very wide and well spaced and thus not get much on the record. What vinyl users often refer to as "better sound quality and warmth" is actually just noise and imperfections not originally recorded on the medium, but added by the archaic technology. Digital CDs are much cleaner, so if the recording is sterile and cold so will the play back be. This means you hear it as the artist recorded it.Vinyl is akin to TV pictures which are deliberately blurred and persisted compared to a reference monitor, the later makes normal TV and movies look cold, sterile and ugly. Try watching a DVD on a good high end PC monitor and you'll see what I mean. Then watch it on a good TV. Reverse the experiment and view your PC on your TV with the same settings. It will look blurred and softened etc.

author: Zero999
author_rank: Super Contributor
timestamp: November 08, 2017, 11:48:11 am
content: Quote from: Alex Nikitin on November 08, 2017, 10:20:56 amAh that makes sense. Thanks for reading the whole thread.QuoteWhat's wrong with CD quality? It's much lower distortion and noise than vinyl and tape: hiss, hiss, flutter, flutter, wow, wow. If you like that sort of thing then good for you, but steer clear of any digital format, unless it's ripped from vinyl or tape.Another interesting fact is, plenty of vinyl and tape recordings will have already been through a 16-bit ADC and DAC, before you get them. Many early digital recordings were actually initially sampled at just above CD quality, 16-bit 50kHz, then transferred to tape and vinyl for release. Tape was considered to be too noisy and distorted for the master/initial recording.https://en.wikipedia.org/wiki/Hard_disk_recorder#History24-bit 96kHz? That's just a waste of bandwidth! Good for recording and signal processing, but not worth it for playback.

author: BrianHG
author_rank: Super Contributor
timestamp: November 08, 2017, 11:52:56 am
content: Quote from: paulca on November 08, 2017, 11:32:26 amError: Blurred, Warmth??? What are you talking about? Vinyl has a 70KHz bandwidth. Blurred? My 180 gram vinyl copy of Supertramp's Crime of the Century has cymbals strikes (I HD sampled and verified that signals past 50KHz exist) are so sharp and crystal clear compared to the cheap 22 Khz CD version, at full volume on my audio system, which is flat to 100KHz, the playback rips my ear drums to shreds with the most strikingly razor sound I've experienced except for 192KHz sampled audio... Don't confuse a mastering choice when the vinyl version, or those writeups in audiophoolery magazines specifically cherry picking smoothly mastered/muffled junk as examples, with a properly full bandwidth pressed vinyl is created, the sound except for the needle traveling groove noise, is astounding for a physical medium. That being said, 24/192k digital recordings free of all noise still cremates it.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 08, 2017, 12:08:14 pm
content: Quote from: Hero999 on November 08, 2017, 11:48:11 amLet's say my ears tell me otherwise. And yes, I am usually unhappy with vinyl or tape recorded from low resolution digital sources , so I'm trying to avoid these.CheersAlex

author: Zero999
author_rank: Super Contributor
timestamp: November 08, 2017, 12:12:36 pm
content: Quote from: BrianHG on November 08, 2017, 11:52:56 amI'm sure you're right about the bandwidth of vinyl, but it won't make any difference, when it exceeds the bandwidth of anyone's hearing, by more than an octave!The distortion and noise of vinyl are much greater than CD, which is theoretically lower noise, than the microphones used to make the recording.

author: coppice
author_rank: Super Contributor
timestamp: November 08, 2017, 12:37:50 pm
content: Quote from: BrianHG on November 08, 2017, 11:52:56 amYou only get that 70kHz bandwidth when using a Shibata or other complex stylus, originally developed to recover the sub-carrier signal on CD4 records. In the hey day of vinyl records only round and elliptical stylii were used in systems limited to stereo playback, so the bandwidth of replay systems was much lower.

author: BrianHG
author_rank: Super Contributor
timestamp: November 08, 2017, 12:49:55 pm
content: Quote from: Hero999 on November 08, 2017, 12:12:36 pmFor hearing within first 20KHz, trying to reproduce the audio by sampling at 44.1k, it's the same effect of having a digital scope with a sampling rate of only 2x the max bandwidth. With a repetitive signal, like most instruments, a quality up-sampling will do most of the time. When it comes to metallic instruments which struck or hit, the initial contact which might need to be a nasty squareish pulse only 1 to 4 sine cycles long will be messed up and it's location imaged by it's phase between the left and right channel might be missing the correct amplitude on one channel VS the other as you go beyond 15KHz. This problem doesn't exist in analog formats or 96KHz recordings. Once again, we are dealing with a complete audio train, (source quality, sampling quality, playback hardware, amp, speakers...) expected to perform a reproduction which sounds like being there in person. Less than 0.1% of the music recorded today comes close to the attention and expertise and $$$ in mastering equipment required to create such mastered recordings.In the past, because of the loss involved in all analog recordings, for a time, there was a real conscious effort in learning how to record to the best you can with whats available. In the beginning of the digital age, this was still there when everything was new, a little hiss was still there and good practices were employed. Now, the cheapest studio junk may be close to noise free & everyone records to maximum over-volume since this sound best on cheap consumer junk of 10-20 years ago which still had hiss on soft recordings when you had to turn the volume up, and in the car on FM radio where most of the music is first heard. (Loudness War) Today, even cheap consumer junk no longer has this hiss, but if new releases had proper volume recording levels, these songs would appear weak next to adjacent music. So for now, we are stuck with maxed out clipped recordings though I hope this trend will begin to soften up a bit. A lot of music I like has been forever trashed because every time the bass drum hits, or is a good vocalist stretches out their vocal chords, either the signal hits the flat line limits of +/-32767, or, a quick momentary volume lowering is heard as automatic sound compression kicks in. And now, even though the recording equipment of the last 10 years could have easily handle the dynamic range easily without any hiss or need for compression tricks, we are left with a unnecessary permanent distorted loss of what the performance truly was.

author: paulca
author_rank: Super Contributor
timestamp: November 08, 2017, 01:05:55 pm
content: Quote from: BrianHG on November 08, 2017, 11:52:56 amI'm sorry but 70Khz-100KHz theoretical bandwidth is one thing, however playing that with a physical mechanical stylus and expecting true representation is another. Kinematics, momentum, resonance, secondary sympathetics, feedback and just a few of the issues with vinyl. Claiming you can hear frequencies over 20Khz is another thing all together. If you are anywhere older than 30 you are unlikely to hear anything even close to 20Khz. Try downloading a test signal app and testing your ears, but, absolutely critically, have someone else push the button, including non-signal controls and record your responses. You can look it up on charts that will guess you age from the results. It will surprise you just how little you actually hear. The test MUST be blind and controlled though to remove cognitive bias.I don't doubt there are frequencies over 22Khz on your high rate sample of vinyl, what I would doubt is if those frequencies are actually coming from the record and if they were there because they were part of the recording. Cymbals are basically pink noise, there would be very, very little way to determine if the frequencies you see are from the cymbal or are artifacts. Even if you modeled the cymbal in the most complicated kinematics and material simulator on a super computer it would still probably be in-deterministic that those frequencies are from the cymbal and not secondaries or just foreign noise.One test you could do is to record the same section from the vinyl multiple times and subtract the waveforms (or invert and add). Do this with a CD and you might find the odd stray noise frequency, but do it with vinyl and you will get a lot of random non-sense in there.

author: coppice
author_rank: Super Contributor
timestamp: November 08, 2017, 01:11:04 pm
content: Quote from: paulca on November 08, 2017, 01:05:55 pmThere is a huge spread in the ages at which hearing bandwidth shrinks. I was still being driven crazy by a room full of EGA monitors whistling at 22kHz when I was over 40.

author: paulca
author_rank: Super Contributor
timestamp: November 08, 2017, 01:16:50 pm
content: Quote from: coppice on November 08, 2017, 01:11:04 pmSubjective and prone to bias. Also possibly harmonics in lower frequencies. However I do know what you mean. I can remember way back in my 20s when I stopped being able to tell if the TV was on or not when I walked into a room, even if it was in hot standby with a black screen I could originally here it's whine... then I couldn't anymore and never have again.You can wind up parents using a dog whistle. A lot of them are audible to young kids, but not you or their parents. The kids will ask what the noise is when their parents think they are mad as they can't hear it.

author: BrianHG
author_rank: Super Contributor
timestamp: November 08, 2017, 01:28:05 pm
content: Quote from: paulca on November 08, 2017, 01:05:55 pmJust so we are clear, my fundamental hearing limit is currently at least 18Khz. (Recently checked...)As for the 70KHz claim of the vinyl. Yes, I am wiling to do a double check and post the results here. Note that I have already sampled the Supertramp album at 24/96 and will verify if I need to investigate further. I know that as you go higher in frequency on the vinyl, the maximum signal level attainable does shrink. 70k was the highest I've seen on Audacity's spectrum analyzer. (we are leaving the arena of what makes a good amp and entering what are the capabilities of vinyl...)When I was young, I could not go into shopping malls because the ultrasonic motion detectors at 25KHz was deafening and even with my fingers stuck in my ears, it drove me mad. My parents thought I was just loony and hated shopping. It's a pain when no one else in the family has any clue about technology, sound, power, or anything more technical than accounting, law and some simple medical info.(Thanks to this, I really beginning to think that truly, outside of medical hearing loss, there are those of us who just hear so much more than others and we are lacking the correct vocabulary to describe that difference since it's natural to each of us to hear what we always been hearing all of our life.... Sort of like a whole new set of colors which are invisible to 50% of everybody else, so, names were never given to them since they are meaningless.)

author: coppice
author_rank: Super Contributor
timestamp: November 08, 2017, 01:48:19 pm
content: Quote from: BrianHG on November 08, 2017, 01:28:05 pmA few years ago I looked to see if I could find any studies into the variation of audio bandwidth across the population. I didn't find any. There are a number of studies into the rate at which that bandwidth shrinks with age, but they don't seem to document anything about the maxima they found, typically not mentioning what happens above 20kHz.It puzzles most people in their early 20s when a 40 year old can tell them when the last 22kHz monitor in the room has finally been turned off.

author: paulca
author_rank: Super Contributor
timestamp: November 08, 2017, 01:57:24 pm
content: Quote from: BrianHG on November 08, 2017, 12:49:55 pmDigital clipping (unfiltered).

author: BrianHG
author_rank: Super Contributor
timestamp: November 08, 2017, 01:58:26 pm
content: To think I'm 50 and still can hear 18KHz. Even recently been to a families house who still has a CRT TV and hearing that FN 15.7KHz wine. Even in the early 90s, I got myself a super expensive 35inch multiscan Monitor with VGA in and made my own 31Khz scan de-interlacer just to be able to enjoy TV without that bloody sound, or interlace flicker. I wonder if all the old extra crappy TV's destroyed anyone's hearing out there...Thanks to mixing consoles with small CRT monitors, or, Atari ST computers for MIDI with their monitor involved in the sound studio's, and with coil based microphones, you will find that bloody 15.7KHz residue in many music recordings. If it is not a .mp3, if you go through a number of uncompressed CD recordings, put them into Audition, set the spectrum analyzer for at least 4k point, or 16k point and look. In pop music, at least 30% of released recordings will have weak solid tone at 15.768KHz. The occasional rare release has it loud enough that I can heard it.

author: Alex Nikitin
author_rank: Super Contributor
timestamp: November 08, 2017, 02:08:38 pm
content: Quote from: BrianHG on November 08, 2017, 01:58:26 pmI am 58 and can hear (checked recently using headphones) 16kHz OK and 17kHz if I push the level up. When I was 18 I could hear up to ~21kHz.CheersAlex

author: paulca
author_rank: Super Contributor
timestamp: November 08, 2017, 02:12:23 pm
content: I haven't tested myself in a while. It really is hard to test yourself though. You would be surprised how many things you hear that aren't there and I sure you "hear" things that you really didn't hear because you know they are there.There are apps to self test that basically blind test you, asking you to hit a button when you hear a sound and they randomly play different frequencies with frequent random periods of silence.

author: BrianHG
author_rank: Super Contributor
timestamp: November 08, 2017, 02:51:11 pm
content: Quote from: paulca on November 08, 2017, 02:12:23 pmSuch apps are bs. Just load Audition, or Audacity, make a new project at at 96k sample rate and generate a 12khz, then 14khz, then 15,16,17,18,19,20khz sine wave and listen. Just be honest with yourself, if your head and ears are being driven into the ground, you can hear it.Note that this type of test doesn't determine your sensitivity, just if you can hear that fundamental sine wave. Any test software which tries to determine you sensitivity isn't worth much unless you have properly verified tuned headphones, proper amp and playback card. Otherwise, you wont be certain of anything. Like a youtube video I saw with a high frequency tone test. I sampled the sound playing back through my PC and everything above 12KHz was missing, as seen in the comments that no one could hear anything above the 12k test though the presenter guaranteed that there was a sound playing there.

author: BrianHG
author_rank: Super Contributor
timestamp: November 08, 2017, 03:06:04 pm
content: Enjoy my sound test example which has 2 seconds of 14khz,15khz,16khz,17khz,18khz,19khz,20khz.Make sure your sound card is set to at least 96KHz sample rate, otherwise, down-sampling may filter out the higher frequencies.(Yup, 18k is my upper limit, 17k is still easily heard. This file doesn't sound on my laptop, but plays perfectly on my main PC with a ASUS motherboard 192khz sound...)

author: paulca
author_rank: Super Contributor
timestamp: November 08, 2017, 03:06:11 pm
content: Quote from: BrianHG on November 08, 2017, 02:51:11 pmThis is the problem. This is why in actual hearing test assessments you are kept completely separated from the tester so you can't even see them pushing buttons. The equipment they use costs 1000s as they can't even have a click when they initiate a sound.If you know when the sound starts you might very well believe you can hear it when in fact you can't.You must do it blind and you cannot blind test yourself.

author: coppice
author_rank: Super Contributor
timestamp: November 08, 2017, 03:13:33 pm
content: Quote from: paulca on November 08, 2017, 03:06:11 pmIts interesting how easily people are fooled into thinking they hear a tone, when all they heard is an initial wideband click. I think the professional hearing test systems actually soft start (gently ramp up) and soft stop the tones to completely avoid any broadband effects at the transitions.

author: BrianHG
author_rank: Super Contributor
timestamp: November 08, 2017, 03:16:52 pm
content: Now you are talking about some sort of placebo effect, or, trying to imagine you hear something. Like trying to guess the letters on a eye test chart 1 line below where you can actually read.The flac files I made have a separator beep and click. Trust me, starting at 14KHz, I ca clearly hear it and each frequency up, it gets fainter. I hear nothing at all on the 19 and 20 Khz. For 18Khz, I need to crank up the volume to the max on my headphones. So, I can still detect 18k, but I'm sure 30 years ago, I could have done the same at 20k. Over 40 years ago, even 25k, but, it is well know that quite a few children under the age of 10 can hear this high up as those bloody ultrasonic motion detector alarms used to drive me crazy at that age.

author: paulca
author_rank: Super Contributor
timestamp: November 08, 2017, 04:09:47 pm
content: As coppice hinted to you are arguing against many decades of psychology. Also you seem to actually believe our senses are empirical. The truth is hugely different.You can study up on the senses and perception and find things that will deeply scare you as to just how inaccurate our perception is and just how much if it is made up. Ball park figures are something like: of the 100% your physical sense devices take in 90% is filtered out. Of the 100% of stimulus youperceive80% is generated by your brain. So your brain takes only 10% of the data from the world around then fills in 80% of what it sends to your conscious.As a random example. Look at yourself in a mirror, now switch eyes, left to right, look at your left eye, look at your right eye. Obviously your eyes move to do so. Can you see your eye ball move? No. You see it in one position and then suddenly it's in it's new position. The actual motion might take 30-40ms but you never see this motion because it's filtered out completely. Even looking at the screen in front of you, switch from one corner to the other. Obviously your eye needs to move through every point along the way but you won't see it. It's filtered out. I can't remember the name of this, but your eyes has two modes, something like point and track. The above is an example of the "point" mechanism post-filtering. The intermediate points are imperceptible to you.Here is a question that might freak you out... where did the time go for the 30-40ms and why don't you perceive it passing?

author: Buriedcode
author_rank: Super Contributor
timestamp: November 08, 2017, 05:38:25 pm
content: I knew this thread would get to the core of perception. There's still plenty of electronics posts, which is something!But I didn't think it would get to folks saying "well I can hear up to xxxkhz!" It seems people assume that human hearing cuts off at a certain frequency, it of course doesn't, its just attenuated, like any other filter. You can sense 20kHz if its loud enough regardless of age, you can even sense 40kHz, again if its loud enough, but as to whether one would consider that "hearing"? that is a different matter, after all we can sense 15Hz, but is it the ears sensing the stimulus?. Also, being able to hear >20kHz doesn't seem to offer any advantage, and in fact must be quite irritating (like having perfect pitch).Some people should also let go of the idea that their senses are consistent. It doesn't matter how 'good' or clever, you think you are, or even being aware of ones own biases (which apparently makes us *less* accurate) you cannot escape this. As paulca alluded to, our minds do a hell of a lot of filtering to the raw signals our senses pick up, even to the point of pretty much creating 'experience' itself, using the real world stimulus as a guide to update its model. This means there are several layers between what sound enters your ears, and what you 'hear', and its mostly dynamic. It's the reason behind optical and auditory illusions, it adjusts 'EQ' which is why we become accustomed to the sound of certain headphones, or sound systems (including the speakers and the comb filter effect of the room).But referring to the topic.. I think we can all agree that the "sound" of an amplifier is just one of many many parameters that affect the sound we hear. Even if we (try to) exclude cognitive bias, given the same speakers but a different room, will sound different. Also.. it will sound different depending on what you have heard during the course of the day, loud bass-heavy noise will make you less sensitive to bass frequencies, likewise with higher frequencies - the mind naturally tries to flatten your "EQ" as constant stimulus isn't novel and so, filtered out. The effect isn't particularly dramatic, but its definitely there, and not considered cognitive bias, perhaps its what"listening fatigue" is?All this means doing A/B tests, even double blind controlled studies can only go so far. It can highlight certain limits (perception of THD, max frequency, min frequency etc..) but these should not be considered overly accurate, so any conclusions drawn should be taken with a pinch of salt. Just because a study is 'double blind, placebo controlled' doesn't make it absolutely accurate, just that compared to other methods, it is the most accurate way to access things.

author: onesixright
author_rank: Frequent Contributor
timestamp: November 08, 2017, 05:43:00 pm
content: What about all the plane crashes? Pilots that rather trust there senses and not the instrumentsOnce you have been in true IFR conditions -btw simulator will do fine- you very, very quickly find out how not to trust your senses. Because whats up is actually down and vice versa.Sorry, IMO the this topic kind of de-railed ;-) couldn't resist.

author: Zero999
author_rank: Super Contributor
timestamp: November 08, 2017, 06:08:07 pm
content: Quote from: BrianHG on November 08, 2017, 12:49:55 pmIt's questionable whether the difference in phase at higher frequencies is perceptible, but even if is, analogue recording equipment can mess it up completely, especially vinyl recordings which have to be passed through a filter first, to compensate for the dreadful non-linear frequency response. The phase of the lower frequency signals (much more important) can be altered, more than a CD quality recording.QuoteI can't ever remember noticing any hiss on digital recordings, only old analogue tape and vinyl, which I won't touch. I agree with you about the poor mixing of some recordings and the loudness war ruining some music.

author: David Hess
author_rank: Super Contributor
timestamp: November 08, 2017, 06:37:07 pm
content: Quote from: Gyro on November 08, 2017, 09:38:38 amAn integrated design provides fast thermal feedback to control the output stage bias however this also creates thermal feedback between the output and input stages. Thermal feedback is a if not the major factor limiting performance of precision integrated circuits. In precision operational amplifiers, it limits open loop gain and for this reason, precision designs make an effort to keep output loading low. I suspect a external error amplifier would improve the performance of the LM3886 but it is already quite good.Thermal feedback issues affecting output stage bias can be avoided entirely. Douglas Self wrote about various "non-switching" amplifiers (1) which used various automatic bias control schemes developed and produced during the 1970s and 1980 but they did not catch on. The Linear TechnologyLT1166 - Power Output Stage Automatic Bias Systemis a modern and current example of this idea.He also reported that the performance of these systems while good was not exceptional. Automatic bias avoids the thermal issue but transconductance variation between the output devices still exists. I suspect the benefit was understated because standard audio tests do not take into account this sort of distortion and nobody bothered to combine this with local feedback.Bob Cordell published a design taking into account both of these aspects of thermal feedback but not local feedback as shown below. Note that while the LT1166 was intended to be used with MOSFETs, it can also be used with bipolar transistors.(1) "Non-switching" in this case means that neither output transistor is allowed to fall into cutoff.

author: coppice
author_rank: Super Contributor
timestamp: November 08, 2017, 07:33:46 pm
content: Quote from: Hero999 on November 08, 2017, 06:08:07 pmPhase perception is not really open to question these days, as its an extensively researched topic. Its one of the areas where lossy music compression codecs achieve a lot of their bit rate gains. We are sensitive to phase at low frequencies, but above a couple of kilohertz nobody has demonstrated an ability to detect phase. However, there is another component to this. Some filters mess with the phase so much you can hear pre-echo effects. Its not the actual phase, but the serious temporal smearing which you can detect. This is largely a digital issue, as the kinds of filters which would cause that much temporal smearing aren't really used in analogue designs.

author: Gyro
author_rank: Super Contributor
timestamp: November 08, 2017, 08:02:45 pm
content: Quote from: David Hess on November 08, 2017, 06:37:07 pmThanks David, I hadn't come across the LT1166.Yes you're right of course, it does also make the input stage very vunerable to thermal feedback, especially on a die that can quick peak at 160'C. Hopefully they've made the geometry small and balanced and avoided serious gradients across it.It would be interesting to see a die photo - there is a pdf available for the bare die version but unfortunately it only has a diagram showing bond-out pad coordinates.Chris.

author: David Hess
author_rank: Super Contributor
timestamp: November 08, 2017, 08:52:37 pm
content: Quote from: Gyro on November 08, 2017, 08:02:45 pmI just noticed that TI has discontinued the LME49830 and all similar drivers in favor of their class-D audio products but I do not really consider that a loss since a discrete implementation is straightforward. The LT1166 has been around for quite some time now and there is no indication that it will be discontinued although I am not aware of it being used in commercial designs.

author: Fungus
author_rank: Super Contributor
timestamp: November 09, 2017, 10:47:54 am
content: Quote from: BrianHG on November 08, 2017, 03:16:52 pmSo, about average then.Quote from: BrianHG on November 08, 2017, 03:16:52 pmIt's well below the -3dB point, though, right?

author: BrianHG
author_rank: Super Contributor
timestamp: November 09, 2017, 11:02:05 am
content: Quote from: Fungus on November 09, 2017, 10:47:54 amYup. I'm guessing even below -10db...I figure within 10 years, my limit will be down to 16Khz, then, things will acceleratingly get worse...As for me at 50 being average, this I'm not so sure. Non of my friends in the past even at the age of 40 were still able to hear a CRT's 15.7KHz which I can still hear. You do know about the fly 16KHz buzzing cell phone ring-tone which teenagers use in class since their teacher's even as young as 30 are not able to hear their phone ringing or receiving a msg...

author: Fungus
author_rank: Super Contributor
timestamp: November 09, 2017, 11:19:35 am
content: Quote from: BrianHG on November 08, 2017, 12:49:55 pmOh, dear.Somebody hasn't watched this enough times:

author: Fungus
author_rank: Super Contributor
timestamp: November 09, 2017, 11:32:34 am
content: Quote from: BrianHG on November 09, 2017, 11:02:05 amI'm over 50 and I can still hear something similar to what you claim, ie. 18kHz is definitely there when I turn the volume up, even if it's hard to say it's a "tone".In musical terms though, I doubt frequencies that high are adding much to the performance, let alone the 70kHz you claim is on your vinyl (which I doubt is coming from any instrument, it will be an artifact of the recording).Quote from: BrianHG on November 09, 2017, 11:02:05 amI heard about that. Even if we assume their phones can reproduce such a tone I still think it's a urban myth, along with ultrasonic pest repellents.

author: paulca
author_rank: Super Contributor
timestamp: November 09, 2017, 03:22:22 pm
content: I have a query which relates to this frequency discussion, though I am coming from the digital side.Sample Theorem and Nyquist Theorem suggests that to capture a signal you must have 2*f bandwidth. Now this makes sense in one level, in that for each wave you need at least a high and a low.However, a 10Khz sine wave cannot be represented accurately within a 20khz bandwidth. At 20Khz the 10KHz sine wave will turn into a 10Khz saw tooth wave, and that's assuming it's correctly phase aligned. To fairly sample a sine wave of frequency f you would need much more than 2*f bandwidth. I realise that filters will smooth the saw wave, but that's not a perfect reflection of the original.Maybe I'm thinking far too digital sample theorem here and in analogue signals it doesn't matter as much.I noticed oscilloscopes recommend 5*f minimum. is this so that you at least get a 5 sample reflection of the original waveform shape?

author: Fungus
author_rank: Super Contributor
timestamp: November 09, 2017, 03:55:06 pm
content: Quote from: paulca on November 09, 2017, 03:22:22 pmYou need to watch the video I posted above a few times.Signal reconstruction is not "join the dots with straight lines".Quote from: paulca on November 09, 2017, 03:22:22 pmYou're half right. You need more than 2*f, yes, but not very much more (in theory only atinyfraction more, in practice a bit more than that... up to 2.5*f is common on things like oscilloscopes).

author: AndyC_772
author_rank: Super Contributor
timestamp: November 09, 2017, 04:26:22 pm
content: With a 10 kHz sine wave sampled at 20 kHz, you could be unlucky and sample precisely at each zero crossing, so the output would be all zeros. The thing to note here is simply that you need a sample rate which is more than twice the highest signal frequency, not just exactly twice the highest frequency.What is often misunderstood is that there is only one correct way to reconstruct an analogue signal from digital samples, and it's not joining the dots with straight lines.Mathematically, there is only one solution to the problem of fitting a curve through the sampled points which does not include frequency components above 0.5x the sampling rate. Two common incorrect methods - linearly interpolating between samples with straight lines, and holding a fixed output between one sample time and the next - both introduce higher frequency components which do not exist in the original data.It's interesting to note also that the mathematically correct solution yields precisely the same waveform regardless of the relative timing between the original signal and the sampled points. A common misconception is that if a portion of a signal is sampled near its peaks, then the reconstructed portion will have greater amplitude than if it's sampled away from them. This is just plain wrong, unless the discussion is specifically about the practicalities of different reconstruction filters.Faster sampling may in many cases may yield a better result than sampling at the bare minimum required to meet the Nyquist criterion, but the reason is to do with practical limitations of real filters.

author: Fungus
author_rank: Super Contributor
timestamp: November 09, 2017, 05:22:17 pm
content: Quote from: AndyC_772 on November 09, 2017, 04:26:22 pmMonty's video (posted above) has the best/clearest explanation I know of.(It can take several viewings to fully grok though ... at least in my case)

author: paulca
author_rank: Super Contributor
timestamp: November 09, 2017, 08:58:05 pm
content: Quote from: AndyC_772 on November 09, 2017, 04:26:22 pmI did this all at Uni in Digital Communications, but it's all leaked out, probably into beer cans at some point.I do remember there was something about an infinite number of solutions to the sample caused by octave harmonics and some such. Digital samples have aggressive cut off filters to remove these harmonics. For example if you sample a 20Khz signal at 10Khz you would appear to have a 10Khz waveform at the end, when of course you don't, you have just sampled every other wave. Apparently it goes the other way too.Anyway, something like that, as I said most of it has leaked out from between my ears as I work purely in software these days I don't really have to deal with ADC DACs.I've added the video above to my list to watch later.

author: coppice
author_rank: Super Contributor
timestamp: November 09, 2017, 09:41:26 pm
content: Quote from: Fungus on November 09, 2017, 03:55:06 pmTo cover both the common misconceptions about sample reconstruction, reconstruction is not:Joining the dots with straight lines, orDrawing steps between the samples, so the waveform resembles a histogramThe samples are a series of instantaneous points on the original waveform. There is only one waveform which is band limited to half the sample rate which will precisely join all of these points. So, you apply the stream of instantaneous samples to a suitable band limiting filter, and out comes precisely the original waveform.Once you understand what sampling and reconstruction is really about, the join the dots idea just looks wacky. However, the stepped waveform idea has important merit in the real world. Its approximately what you get out of most DACs. Look up "zero order hold" and you will find a full analysis of this kind of system. Once you band limit the stepped waveform to half the sample rate you end up with something similar to the original signal, but with its high frequency response rolled off in a very predictable way. To precisely recreate the original signal you need to compensate for this roll-off by either boosting the highest frequencies in the band with a digital filter before the DAC, or use a similarly non-flat analogue filter, in addition to the band limiting filter, after the DAC.One caveat in this talk of signal filtering is you need to be careful about the phase response of any filters used, if you want to recover the original waveform precisely.

author: Fungus
author_rank: Super Contributor
timestamp: November 10, 2017, 12:19:01 am
content: Quote from: paulca on November 09, 2017, 08:58:05 pmObviously true, but the stuff they put on CDs (for example) is bandwidth limited (ie. filtered) to remove everything above the Nyquist limitbeforethey sample it.Everything above the 22kHz Nyquist limit of CDs is inaudible anyway. It adds nothing to the sound.Bottom line: 16bit, 44.1kHz is enough to perfectly reproduce audio, despite what the "golden ears" might tell tell you.In theory a casecouldbe made for 48kHz sampling because it allows the low pass filter applied at the mastering stage to be a bit less aggressive. In practice it doesn't seem to make much difference. 96 or 192kHz? Not needed. 24 bits? Not needed.It's all in the video.

author: Loboscope
author_rank: Regular Contributor
timestamp: November 10, 2017, 12:20:23 am
content: Be aware, that anything in human audio is bandwith-limited. So are our ears, even if some members here seem to be anatomic miracles, at least exceptions. For further information please look here:https://www.amazon.com/Auditory-Neuroscience-Making-Sense-Sound/dp/0262518023.But allnaturalsources of sound are bandwith-limited too. Because of the mass-inertia of strings, reeds, resonance-wood, drum-membranes, speaker-membranes a.s.o., it will take some times for the transients to rise. Therefore audio will never deal with square-waves, the rise-times of human audio signals will be limited by nature. Even the compressibility of the air itself will prohibit square-waves being transferred.Additionally all high-frequency content of Instruments/Voices are of low level in relation to the fundamental frequencies.So no ADC / DAC in audio-systems has to deal with square-waves and/or a 20 KHz-wave at 0 dB for example.A tiny bit more than 2*f as Nyquist-frequency will be enough to sample all of our wonderful music we here all days, 44,1 KHz will be enough. And as distribution format the CD with 16 bit / 44.1 KHz will be sufficient.The audio quality depends only from the quality of the sources (first and most of all the musicians!), the recording an the responsible recording engineer and his mixing/mastering qualities.I have lots of excellent CD´s here and I make recordings as sideline profession. My main profession is musician with classic-academic studies and I know how instruments, how music will sound.After I bought my first CD-Player in the late eighties last century I no more touched my Vinyl again.

author: coppice
author_rank: Super Contributor
timestamp: November 10, 2017, 01:01:13 am
content: Quote from: Fungus on November 10, 2017, 12:19:01 am16 bits properly dithered is fine for distribution. Without dithering its a bit iffy. Its a pity they didn't go for something like 17 or 18 bits, and put the adequacy of the dynamic range beyond question. People seem to get hooked on round numbers like 16 and 24.. The bottom line is 16 bits was not great in the early 80s, when 16 bit/44.1ksps was introduced, but as our understanding of dithering techniques has developed, it has become adequate.44.1ksps also seems to be perfectly adequate. Being able to hear a tone at 23kHz or 24kHz with reasonable sensitivity doesn't mean that content at those frequencies will have any material effect on your perception of complex musical sounds. In my youth, when I could hear those frequencies pretty well, I was never aware of the lack of them affecting my perception of music. A nice flat response to 20kHz seems fine. Its actually at the other end of the spectrum where systems are usually lacking. Typically 20Hz is considered the bottom of the range of human hearing, However, the lowest note on the pedals of a large organ is in the 13 to 16Hz range (depending on the organ). You feel more than hear those notes, but they are a very material part of the musical experience in a cathedral. Even today, when most of the technical reasons for poor bass response have been dealt with, few audio system reproduce that stuff well. A honk at the resonance of the speaker is all most systems seem to muster.

author: BrianHG
author_rank: Super Contributor
timestamp: November 10, 2017, 04:21:45 am
content: Quote from: Fungus on November 09, 2017, 11:19:35 amQuoteHe is not completely right, unless the audio output equipment has proper analog filtering, or sin(x)/x correction. However, since sound samples in a PC rarely match the DAC's output sample rate today, this filtering must also be done in Windows or in the sound card drivers. Do you trust Microsoft, or all the apps you installed on you PC.Look here, I measured these an hour ago, see how it looks for real on my scope: (I matched his setup, but specified my sample rates...)#1, 20Khz sine wave sampled and played at 96KHz:#2, 20Khz sine wave sampled and played at 44.1KHz:#3, 20Khz sine wave sampled and played at 44.1KHz, however, I used a DSP feature built into Foobar2000 music player called PPHS resampler set to 96KHZ, ultra mode:Lesson, your video is only correct if proper analog filtering practice is applied, which I already knew and also knew about how today, only a few % of audio equipment bother to attend to this issue properly accommodating multiple source sample rates. Lesson 2, if you have a PC as your music player with a 96KHz DAC, and you want to bypass the system re sampling to guarantee a good output sine wave at the high above 16KHz when playing CD 44.1KHz music, use Foobar2000 player with the PPHS resampler enabled and set to 96K, ultra mode and your analog output will be guaranteed have a properly clean sin(x)/x filtered 20Khz signal reconstruction.In other words, if and only if you playback hardware has properly tuned analog output filtering, or may be done in combination with digitally filtered upsampling and analog filtering for that new sample rate, under these circumstances, a stereo 44.1KHz sample should achieve at least a 22Khz bandwidth reproduction with proper stereo image. For those who try to nit-pick specific lemon multiple tones, I'm sorry to say that no human instrument performance would ever sit exactly on a lemon frequency exactly. For those who sample and play at 96KHz, you are just pretty much guaranteeing that errors in sampling and playback hardware wont produce the nasty problem in my scope shot #2 for frequencies between around 16KHz and 20KHz. Errors now at the new higher frequencies due to poor design practices are way too far outside our hearing to bother with and microphones in the recording studio have already lost soo much db at 40Khz anyways that there isn't much signal left.Source audio files available upon request.Freeware Foobar2000 media player with SDK:http://www.foobar2000.org/

author: BrianHG
author_rank: Super Contributor
timestamp: November 10, 2017, 07:03:22 am
content: I know how Monty Montgomery @ xiph.org says true that the sample data represent a smooth linear signal and properly designed AD/DA systems should reconstruct the original smooth analog wave of the source up to 1/2 the sample frequency. But, unfortunately, today, consumer grade equipment don't quite measure up in reconstructing the original source signal's true image. It's funny how older non-oversampling DACs in really old CD players with good analog filters at the output stage may do this reconstruction better than some of today's technology when it comes to dealing specifically with 44.1k 16 bit audio.

author: hamster_nz
author_rank: Super Contributor
timestamp: November 10, 2017, 07:14:59 am
content: @BrianHG, what happens if you play the 44.1 sample, and turn on the channel's low pass filter, set to 22kHz (like in a good reconstruction filter)?If you then flip it to high pass you should see the 24.1kHz alias in there...

author: AndyC_772
author_rank: Super Contributor
timestamp: November 10, 2017, 07:26:51 am
content: Quote from: paulca on November 09, 2017, 08:58:05 pmI think you're just describing aliasing. If you have a component in the original analogue signal at a frequency exceeding 0.5*fs, then it'll appear in the reconstructed signal at a lower frequency, and is impossible to distinguish from a real, wanted signal at that same lower frequency.That's why a sampling system has to include an analogue low-pass filter in front of the ADC.

author: BrianHG
author_rank: Super Contributor
timestamp: November 10, 2017, 07:30:14 am
content: Quote from: hamster_nz on November 10, 2017, 07:14:59 amI chose 20KHz since it had really pronounced garbage on the scope. I assume generating a 22.05KHz sine would show perfectly.As for Foobar2000's PPHS resampler, is has no option other than to select an output sample rate and 'ultra' mode. I assume when I send it a 44.1k source audio, it is precisely tuned to properly limit filter that to 22.05khz.As for your idea, I grabbed a frozen scope shot of my #2 illustration: However, this is not the same as over shooting the source frequency bandwidth. Thus again, sampling at 96k allows for a super refined low pass filter to get rid of anything above 22.05khz when downsampling to 44.1k. Again, some programs do a better job than others when doing this but I don't trust Window's built in resampling. I would always run my audio acquisition software at the specified rating of it's ADC, the do a thorough downsample to my operating frequency. With hard drive space today, I'm happy to just stick with 96k. 192k is just a waste...

author: Fungus
author_rank: Super Contributor
timestamp: November 10, 2017, 08:51:10 am
content: Quote from: BrianHG on November 10, 2017, 07:03:22 amWeird how Monty's 15 year old consumer grade sound card can do it perfectly...

author: BrianHG
author_rank: Super Contributor
timestamp: November 10, 2017, 09:17:53 am
content: Quote from: Fungus on November 10, 2017, 08:51:10 am15 years ago, he has a 192Khz capable sound card, back then, that's not consumer grade. That's a studio grade sound card. (You can see he is able to select higher sample rate settings....)The other possible side of the coin goes to my other statement above: Older hardware specifically tuned to 44.1k only has hard wired analog output filters specifically limited to 22.05KHz. Modern engineers have gotten sloppy with DACs which have built in voltage out driving the line out/headphones directly without an analog op-amp and sin(x)/x filter.Even my old Soundblaster AWE64Gold from 98 had op-amp I-V converter circuit with a filter stage. I know because I modified the gain resistor to get an excellent 5vp-p output with almost no noise at all. (Great for DVD movies at the time with very low volume recordings, or wide dynamic ranges...)

author: Zero999
author_rank: Super Contributor
timestamp: November 10, 2017, 09:41:35 am
content: Quote from: coppice on November 10, 2017, 01:01:13 amI agree with you that more than 16-bits is required for recording purposes, because it avoids the need to keep adjusting the gain control or a limiting circuit, so the normal level can be set so it never clips. However, for playback, I think 16-bit is more than adequate, because music doesn't use the full undithered 90.3dB dynamic range anyway.QuoteI agree. Those faint high pitch sounds, will be drowned out by the lower frequency content. I also believe pitch perception declines at higher frequencies, so it's less important.How about recording at much higher sample rates? It makes more sense to avoid rounding errors when doing intermediate processing but is it needed for recording? I suppose the master should be as perfect as possible and not having the lag and phase distortion of a filter could make it easier.QuoteIt's possible that many people can't hear the fundamental of a <16Hz organ note, only the harmonics. The lower frequencies aren't important for pitch perception at the bass end of the spectrum. The fundamental can be missing, yet the brain will still perceive the correct pitch.https://en.wikipedia.org/wiki/Missing_fundamentalThe ears and the brain can be easily fooled, no matter how cleaver one is. The key is to beware of this, especially when someone tries to part you from your hard earned money.

author: BrianHG
author_rank: Super Contributor
timestamp: November 10, 2017, 10:10:43 am
content: Quote from: Hero999 on November 10, 2017, 09:41:35 amNow I'm not saying you need slightly better quality than a CD, I'm saying with today's technology, you should be able to get at least slightly better quality than CD quality for free, at no extra cost. Anyone going off the deep end trying to sell you a 1MHZ amp, picovolt accuracy, silver plated copper speaker cable are full of audiophoolery BS. I just expect that at a minimum, you shouldn't see my second scope shot illustration #2 from my laptop, just as fungus said, a 15 year old sound card didn't exhibit that modulation problem with a 20KHz tone and I could only get rid of the problem with special dedicated software. So much for a laptop with a 192KHz, 24 bit audio. It plays the Hi-def fine, but, cant even cope with CD 44.1k 16bit without special software countermeasures.

author: Loboscope
author_rank: Regular Contributor
timestamp: November 10, 2017, 10:42:45 am
content: Quote from: Hero999 on November 10, 2017, 09:41:35 am16 Hz will only be reproduced by organs which have a 32´-stop. There are not so much organs with 32´, only the greatest ones have these deep going stops. I guess 99% of all organs in the world will only have 16´-stops as deepest stop. Will say, that they ´only´ will reproduce 32 Hz as deepest tone.There are stops which use the phenomenon of the "missing fundamental" by creating deep notes with two pipes sounding together at a distance of a fifth (Quinte). With the proper intonation, this lets sound the sub-octave of the deeper tone of the pipes, a so called combination-tone (Kombinationston). The level of this sub-octave will be lower than the level of of the original tones of the pipes, but our ear/brain still recognizes/reconstructs the deeper tone.Also the fundamentals of great bells have lower levels than their harmonics, but here also our ear/brain recognizes the fundamental and the bell will sound deep and sonorous.Concerning the capabilities of modern ADC/DAC-converters I did run a simple test on my Behringer X32-Rack. Behringer is known as a brand that stands for less expensive but normally good constructed gear and equipment for PA and recording intended for semi-professional and professional use. (For the X32 family and the digital mixer lines look here:http://www.music-group.com/Categories/Behringer/Mixers/Digital-Mixers/c/1234111?group=Digital%20Mixers&colExpFlag=,Digital%20Mixers)I did run a 20, 21 and 22 KHz test signal through the mixer analog in and analog out, so it has to pass the ADC and the DAC (at a samling rate of 44,1 KHz). The level was a tiny bit below 0 dB, short before clipping. These are signals, the converters has not to deal with in real life, because only very few instruments will have harmonics up to 20 KHz or even more, and if, their level will be very, very low.Below, there are the three samples as scope-snapshot. You will see, that 20 KHz is fine, 21. KHz is a little bit distorted and 22 KHz (at ca. -0,5 dB at the X32!) will be distorted obviously by an artifact.But as I said, this will never disturb any audio-source coming from a natural instrument/sound.As far as I know, Behringer uses ADC´s/DAC´s from AKM and certainly not the most expensive ones. If you by ultra-cheap crap, you will run into problems, don´t matter if it is an amp, a CD-Player, a speaker a.s.o.But any gear from the middle-range upwards shall guarantee a good to very good sound quality. My nearfield Monitor-Speakers are Neumann KH120A [http://www.neumann-kh-line.com/neumann-kh/home_en.nsf/root/prof-monitoring_studio-monitors_nearfield-monitors_KH120A] and they sound precise, clean and neutral as a Monitor for mixing/mastering should do. They are not cheap, but also not expensive, mid-range in price I would call them. So nowadays you will not have to invest a fortune to have excellent audio equipment.

author: BrianHG
author_rank: Super Contributor
timestamp: November 10, 2017, 11:29:15 am
content: Quote from: Loboscope on November 10, 2017, 10:42:45 amEven AKM's entry level DACs are really good and properly designed. If I remember correctly, they have their own proprietary I/V technology stage built into their audio voltage output DACs.

author: Fungus
author_rank: Super Contributor
timestamp: November 10, 2017, 12:32:25 pm
content: Oh, come on. The reconstruction filter has been built into DACs for decades.(that's why they've started calling them "codecs").

author: paulca
author_rank: Super Contributor
timestamp: November 10, 2017, 01:28:26 pm
content: Quote from: BrianHG on November 10, 2017, 09:17:53 amYou have to remember that in audio production you don't "sample" to just record.If you wanted to sample, say, a piano, you could take a multi-part sample, one sample per octave (or half or quarter octave depending on your equipment/sampler and it's memory). But a lot of basic samples are single samples which when used in music might be played across multiple octaves.So it is common practice to over sample at much higher rates to retain quality when played on lower notes.A 44.xKhz sample of a piano at middle C might start to sound rubbish two octaves down when you replay it at 11Khz.

author: David Hess
author_rank: Super Contributor
timestamp: November 10, 2017, 01:38:09 pm
content: The 44.1/48 kHz 16 bit samples are resampled at a multiple of the sample rate so the requirements of the analog antialiasing filter are considerably relaxed. Digital integration makes this more economical than an analog solution.

author: paulca
author_rank: Super Contributor
timestamp: November 10, 2017, 01:44:32 pm
content: Quote from: David Hess on November 10, 2017, 01:38:09 pmI believe that is one way of doing it. The other is to use high sample rates. I confess it's been a decade or two since I've used a professional hardware sampler though.

author: coppice
author_rank: Super Contributor
timestamp: November 10, 2017, 03:33:55 pm
content: Quote from: David Hess on November 10, 2017, 01:38:09 pmSample rate conversion like that is normally done by using a polyphase filter that bandlimits to the new Shannon rate and produces output samples at the required rate all in one go.

author: Buriedcode
author_rank: Super Contributor
timestamp: November 10, 2017, 05:22:51 pm
content: It seems most DACs for audio, and - at least as far as I have seen in consumer grade audio gear - ADCs are Sigma Delta, which greatly lessens the burden on the antialiasing filter. It seems many have forgotten this when talking about 44.1kHz as a sampling frequency.https://en.wikipedia.org/wiki/Delta-sigma_modulationhttp://www.ni.com/white-paper/11342/en/You can buy perfectly adequate ADCs, DACs /Codecs for <$2 that require only a clock source and a simple RC filter. Often the less-than-stellar results one gets from PC and laptop soundcards is nothing to do with the codec itself, but rather the power supply used (for example, they rarely have a decent mic bias, and headphone amps tend to be right next to noisy parts). This is why external cards - even ones with *worse* specced codecs often perform much better, they don't have to cope with all the EMI generated inside a computer.Also, the fact a converter can do 192kHz sample rate, does not mean it is 'studio grade' or 'consumer grade' or say anything about its ENOB.

author: David Hess
author_rank: Super Contributor
timestamp: November 10, 2017, 05:59:19 pm
content: I was just keeping things simple. Oversampling is implemented in the ADC/DAC which considerably relaxes the requirements of the analog antialias filtering. Delta-sigma converters also do this to an extreme to relax the requirements of the modulator and implement noise shaping to improve dynamic range. But the result is that 44.1/48 KHz and 16 bits are sufficient for transparent audio because of this processing being done by the ADC/DAC.There are all kinds of other limitations including noisy power, ground loops, crummy analog signal paths including the power amplifier, and poor clocking. The later is especially a problem with interfaces which do not provide the sampling clock like USB although I wonder if any common interface on a PC does. Like power amplifiers, clocking is a solved problem but there are lots of examples out there which are broken.

author: Fungus
author_rank: Super Contributor
timestamp: November 11, 2017, 11:45:26 am
content: The weird thing about all this theory is that most of it was figured out in the 19th century. Long before "electronics".

author: coppice
author_rank: Super Contributor
timestamp: November 11, 2017, 08:47:37 pm
content: Quote from: Fungus on November 10, 2017, 12:32:25 pmThe term codec is a contraction of code and decode. It was a term that started with telephony ADC + DAC + A-law/u-law conversion chips, I believe. By any measure a DAC only or an ADC only device is not a codec, whatever bits of its part of the signal chain it includes or misses out.

author: Zero999
author_rank: Super Contributor
timestamp: November 13, 2017, 09:08:54 am
content: All this talk of sampling is interesting and very relevant if one is using a sound card to record something, but for CD playback, all of the sampling and anti-aliasing will have already done before the CD was recorded. The DAC in the playback device doesn't have to worry about filtering or anti-aliasing. All it has to do is convert the 1s and 0s on the disc to the correct analogue value at the right speed.

author: coppice
author_rank: Super Contributor
timestamp: November 13, 2017, 10:07:35 am
content: Quote from: Hero999 on November 13, 2017, 09:08:54 amRubbish. What comes out of the DAC needs band limiting to recover the original signal. That filter might be integrated with the DAC on a single chip, but its still a vital filter beyond the actual DAC.

author: Zero999
author_rank: Super Contributor
timestamp: November 13, 2017, 10:20:51 am
content: Quote from: coppice on November 13, 2017, 10:07:35 amI thought that the only stuff which needed filtering, on the analogue side of the DAC was above the sample rate of 44.1kHz and therefore ultrasonic in this case, so the filter isn't critical, or have I missed something?

author: Dubbie
author_rank: Unknown
timestamp: November 13, 2017, 10:29:17 am
content: Yes you have hero999. Read a primer on DACs and you will see that filtering before digitising is essential as is filtering after converting back to analog. With both of these taken care of you can get practically a perfect reconstruction of the original waveform within the bandwidth constraints you are working in.If you want to learn more, I can recommend this book:http://www.dspguide.com/I learnt (and still am learning) a lot from this book.You can read the whole book for free on the site. Chapter 3 will fill in the blanks for you.

author: coppice
author_rank: Super Contributor
timestamp: November 13, 2017, 10:32:02 am
content: Quote from: Hero999 on November 13, 2017, 10:20:51 amYou've missed that if all that ultrasonic stuff reaches a typical audio amplifier the results won't be pretty. Also, if anything downstream re-digitizes the signal (perhaps as part of a digital power amp) its going to digitize something pretty funky if you haven't band limited the signal.

author: Zero999
author_rank: Super Contributor
timestamp: November 13, 2017, 10:49:06 am
content: Quote from: coppice on November 13, 2017, 10:32:02 amI see your point, but I think you've overlooked the fact that the output filter, on a DAC, is far less critical, than the input filter, on an ADC.Any decent audio amplifier should already have a low pass filter, guard against demodulation of high frequency signals and protect a piezo tweeter, connected to the output. Anything which digitises the signal again (which isn't best design practise anyway: ideally a digital amplifier will convert the digital input from the DAC directly to analogue PWM to drive the speaker) should already have a good low pass filter.

author: coppice
author_rank: Super Contributor
timestamp: November 13, 2017, 11:21:42 am
content: Quote from: Hero999 on November 13, 2017, 10:49:06 amIn practice its a little less critical, but it still needs to be taken seriously.Quote from: Hero999 on November 13, 2017, 10:49:06 amShow me an audio amp with a brick wall filter that will clean off everything beyond the Shannon rate of an incoming 44.1ksps signal. As for re-digitizing a signal, what happens when you re-digitize an unfiltered 44.1ksps signal at 96ksps or 192ksps? Those digitizers won't filter away the mess below about 48kHz or 96kHz.I expect the reason you don't see this filtering as important is you haven't experienced what happens when it isn't there. Most audio DACs have had the filtering integrated since the mid 80s. in early CD players they used to up sample by 4x or 8x the 44.1ksps rate during the ZOH pre-compensation, just to make the analogue post filter simpler. Now most DACs are sigma-delta, the filtering requirements are must easier to meet, but still important.

author: Fungus
author_rank: Super Contributor
timestamp: November 13, 2017, 11:39:09 am
content: Quote from: coppice on November 13, 2017, 11:21:42 amI don't think an audio amp has any business doing anything at all to a signal other than making it louder.Any filtering should be in the source. The source knows exactly what sort of filtering is needed (if any), the amplifier doesn't.

author: alexanderbrevig
author_rank: Frequent Contributor
timestamp: November 13, 2017, 11:52:11 am
content: Quote from: Fungus on November 13, 2017, 11:39:09 amhttps://en.wikipedia.org/wiki/Class-D_amplifierYou would be very disappointed with a Class-D that has no filtering.

author: Zero999
author_rank: Super Contributor
timestamp: November 13, 2017, 11:58:25 am
content: Quote from: Fungus on November 13, 2017, 11:39:09 amThe audio amplifier should filter out any ultrasonic frequencies on the input, which the designer knows can cause unintended operation. It is important for EMC purposes. If poor/inadequate filtering is used, the speakers will buzz loudly when: placed near a switched mode power supply or a nearby mobile phone receives a call, or it might pick up and demodulate the signal from a nearby AM transmitter. To stop this, a filter with a cut-off above 20kHz should be added to the inputs, along with ferrite beads. This is more important with amplifiers, connected to long leads, carrying small signals, such as microphones.Quote from: alexanderbrevig on November 13, 2017, 11:52:11 amWhat filtering? On the input or output?Filtering is often omitted from the output, especially when the speaker is close by. The inductance of the voice coil, acts as a low pass filter preventing large currents from flowing at the PWM frequency.There should be some filtering in the input, for EMC reasons, mentioned above.

author: Fungus
author_rank: Super Contributor
timestamp: November 13, 2017, 12:00:41 pm
content: Quote from: alexanderbrevig on November 13, 2017, 11:52:11 amA class D filters itsoutput, obviously.

author: Zero999
author_rank: Super Contributor
timestamp: November 13, 2017, 12:19:54 pm
content: Quote from: coppice on November 13, 2017, 11:21:42 amThere are no audio amplifiers with a brick wall filter at that frequency, because it's not necessary. If the amplifier is properly designed, then a small amount of 44.1kHz and harmonics present in the input, will not cause any problems. Generally most amplifier designs don't misbehave, until the frequency is well above that anyway.QuoteI see your point about re-digitising the signal and the problems it may cause.

author: oldway
author_rank: Super Contributor
timestamp: November 13, 2017, 02:55:10 pm
content: All this has nothing to do with the subject of this topic ....After 14 pages of audiofoolery, I'm still waiting for someone to tell me what a "high end" amplifier is, to know where we can find the list of these "high end" amplifiers, what sets them apart from low end amplifiers, (NB: apart from perlinpin powder that can not be measured or confirmed in an objective and impartial double-blind test)If these tests can not be done because the hering memory is limited to 10 seconds, how can we say that one amplifier sounds better than another ....?How is it possible that there is a hearing difference between 2 amplifiers, while the measurements made with high precision instruments are equivalent?Unlike a microphone, or a loudspeaker where there are complex phenomena like resonances, an amplifier is an electronic component whose characteristics are measurable with high precision ... where would the magic of the different sound come from? perlinpinpin powder?

author: Fungus
author_rank: Super Contributor
timestamp: November 13, 2017, 04:27:51 pm
content: Quote from: oldway on November 13, 2017, 02:55:10 pmA mythical beast.Quote from: oldway on November 13, 2017, 02:55:10 pmIt isn't.(not unless people are measuring the wrong things)

author: trys
author_rank: Regular Contributor
timestamp: November 13, 2017, 04:59:14 pm
content: Cotton buds ('Q Tips' in the US) make a big difference.

author: Buriedcode
author_rank: Super Contributor
timestamp: November 13, 2017, 05:06:53 pm
content: Quote from: coppice on November 13, 2017, 11:21:42 amAs you stated, most, if not all modern audio DACs are sigma delta, - their output is updated at 64/128x the sampling rate, pushing the sample rate waaay up the spectrum, meaning the a simple output filter, combined with the frequency response of an amp greatly attenuates it. I won't go into interpolation/digital filters etc.. as ultimately it just means that at the output, the sample frequency is now in the MHz range, and is already attenuated. Unless one is plugging the output of their DAC into an RF amp, I don't see why output filtering is particularly important. Yes, it is needed, but I can't see it being something that requires a great deal of design?The conventional (say, R2R ladder DACs) that really do require strong attenuating filters are, these days, the ones designed and built by 'audiophiles' because they believe that sigma delta DACs sound 'sterile' and because they've seen 1-bit SD modulation waveform and thought it doesn't 'look' like nice clean audio. The irony being that in an effort to bash modern audio converters, they've resorted to older technology that performs measurably worse, but 'looks nicer' on a scope (and also costs a lot more because of very tight component tolerances, which justifies the high cost for those silly enough to pay for them).Some sources of that misunderstand the process about it:http://www.mother-of-tone.com/conversion.htmhttp://www.msbtechnology.com/faq/why-ladder-dacs/https://hifiduino.wordpress.com/2014/10/12/r2r-for-the-rest-of-us/Note the last one isn't snake oil or anything, I'm sure its a very well designed R2R DAC, but seems like a LOT of effort and money for something that performs very similarly to a $2 IC.

author: Fungus
author_rank: Super Contributor
timestamp: November 13, 2017, 06:22:19 pm
content: Quote from: Buriedcode on November 13, 2017, 05:06:53 pmYep, I've done that and it works.Ideally you should output curves, not straight lines though.

author: coppice
author_rank: Super Contributor
timestamp: November 13, 2017, 06:48:38 pm
content: Quote from: Buriedcode on November 13, 2017, 05:06:53 pmI didn't say it was a tough job to do the filtering these days. I said it was needed in response to someone saying it was unnecessary. A huge number of audio amps really do not like RF, and are poorly protected from its effects. Everybody knows the burbling pattern of GSM interactions because of this.Quote from: Buriedcode on November 13, 2017, 05:06:53 pmGood stereo DACs haven't cost $2 for a very long time. 20 cents is more like the going rate. Its crazy these days. A good op-amp to buffer the output of the DAC can cost more than the DAC.

author: David Hess
author_rank: Super Contributor
timestamp: November 14, 2017, 08:14:37 am
content: The output of the DAC still needs to be filtered because intermodulation from non-linearity in the output amplifier which increases with frequency will result in mixing products in the audible band and a general increase in noise.The same thing happens with speakers when a single driver handles a broad range of frequencies and has a large displacement. This is especially a problem with woofers at low frequencies where displacement may be large and the intermodulation produces a "boomy" sound. Base reflex and horn designs help prevent this by lowering mechanical impedance to produce more power with less cone movement.

author: Zero999
author_rank: Super Contributor
timestamp: November 14, 2017, 08:57:15 am
content: Quote from: coppice on November 13, 2017, 06:48:38 pmYes cheap amplifiers often overlook filtering and good EMC design practise.Quote from: David Hess on November 14, 2017, 08:14:37 amA high end amplifier shouldn't do that, because it will behave at 44.1kHz and have good filtering for the harmonics which will cause problems.

author: David Hess
author_rank: Super Contributor
timestamp: November 14, 2017, 04:04:35 pm
content: Quote from: Hero999 on November 14, 2017, 08:57:15 amA high end amplifier has the same problem filtering the aliasing products outside of a 20 kHz bandwidth sampled at 44.1 kHz as the DAC reconstruction filter does. Oversampling DACs make this into a trivial issue where a simple low order filter is sufficient but in the past, 11th order filters (1) were used to produce enough rolloff between 20 and 22 kHz.The amplifier does not need to misbehave at higher frequencies to cause problems. Non-linearity simply increases at higher frequencies producing mixing products which may end up in the audio band.(1) At least that is what Burr-Brown required for their DACs to meet their specifications.

author: Zero999
author_rank: Super Contributor
timestamp: November 15, 2017, 09:16:48 am
content: Quote from: David Hess on November 14, 2017, 04:04:35 pmThat's pretty much what I meant by misbehaving and is probably one of the reasons why many audio amplifiers are designed to work well above the audio band, so ultrasonic noise from CDs and other digital sources don't cause buzzing. For example: the LM3886 is specified for low distortion up to 20kHz and has a full power bandwidth of 80kHz and the NTE1380 has a FPBW of 140kHz. Somehow I doubt they're going to produce audible noise from 44.kHz.http://www.ti.com/lit/ds/symlink/lm3886.pdfhttp://www.nteinc.com/specs/1300to1399/pdf/nte1380.pdfIt's true a good filter will be required to block the higher 44.1kHz harmonics, which could do something funky, but I doubt the fundamental needs to be attenuated to ridiculously low levels, to avoid noise.

author: paulca
author_rank: Super Contributor
timestamp: November 16, 2017, 04:06:33 pm
content: The high end amp costs more. That's about all you know for sure.

author: David Hess
author_rank: Super Contributor
timestamp: November 19, 2017, 10:40:23 pm
content: Quote from: Hero999 on November 15, 2017, 09:16:48 amFPBW (full power bandwidth) is just another name for large signal response and depends on slew rate and output voltage. The gain-bandwidth product or small signal bandwidth is more important for reducing distortion unless the full power bandwidth/large signal response/slew rate is insufficient which should never happen.The NTE1380 datasheet does not show a schematic but the LM3886 does. See those 1.1k resistors used for emitter degeneration of the input PNP differential pair? They lower the input stage transconductance while maintaining the same tail current. This allows the compensation capacitor marked as 10 picofarads to be smaller then it otherwise would need to be improving the output slew rate and full power or large signal bandwidth. As a side effect, those 1.1k resistors in series with the emitters increase the input noise considerably. If a schematic is not available or is incomplete, then the input noise specification can be used to determine if emitter/source degeneration was used. Contrast the 318 and the 833; the former has 10 times the slew rate (and full power bandwidth) at the cost of 3 times higher voltage noise do to emitter degeneration to reduce transconductance.The old 318 operational amplifier used emitter degeneration resistors to get its high frequency performance and was noisy and lower precision as a result. Most audio power amplifiers do as well. FET input amplifiers have lower transconductance to start with. There are also other ways to handle transconductance reduction.For audio power amplifiers, input noise is not normally a concern because it is assumed that a high level signal is available but I have noticed that many modern amplifiers produce a really annoying hiss with no input signal when connected to high efficiency speakers.The excess gain at frequencies above the audio band is just a side effect of maximizing the gain at high frequencies to limit distortion. I do not think anybody does it to limit intermodulation of content above the audio band which should not exist anyway and would not have been a problem before digitally sourced music. But I do wonder if early inconsistent reports of digital audio quality came about because of inadequate anti-alias filtering allowing high frequency alias products to mix in the power amplifier. The anti-aliasing filters were an obvious place to save money and lots of cheap audio gear did so.QuoteBurr-Brown and others were not recommending 11th order anti-alias filtering with a transition band between 20 and 22 kHz to make their 44.1 kHz ADCs and DACs easier to sell. This was such a big problem that oversampling ADCs and DACs with their much simpler anti-aliasing requirements replaced them quickly and made this a solved problem even on the economic side.

author: Zero999
author_rank: Super Contributor
timestamp: November 20, 2017, 06:44:55 pm
content: Quote from: David Hess on November 19, 2017, 10:40:23 pmI agree about power audio amplifiers being a bit more noisy than op-amps. The NTE1380 is same as the old TDA2030, which I've tested and the noise was not detectable, even with headphones, although the gain was only 16, so that's hardly surprising: the spec is typically 3µV, over the audio bandwidth, which would give 48µV out.EDIT:I've just calculated the noise for the TDA2030/NTE1380 and it comes to 20.2nV/Hz0.5typical, 7.5nV/Hz0.5. The current noise isn't too bad though: 1.35pA/Hz0.5If you need significantly more than the minimum allowed gain, use a pre-amplifier.http://www.changpuak.ch/electronics/datasheets/TDA2030.pdfQuoteSounds overkill. There are other sources of ultrasonic noise than digital audio: switch mode power supplies and RF, especially the IF in a receiver, which would have been a potential issue, long before digital audio was widespread. I couldn't get the old TDA2030 to produce any audible noise, when the input frequency was swept from 22kHz to 200kHz. The only problem I had was the zobel network resistor overheating, when I had the output level too high. I suppose that's a constant signal though, so that's not a fair test. I should've done it with an FM and AM modulated 1kHz tone.

author: paulca
author_rank: Super Contributor
timestamp: November 20, 2017, 08:25:49 pm
content: Quote from: David Hess on November 19, 2017, 10:40:23 pmSlightly related, but I knew I'd see this name somewhere before.My digital USB headphone amp identifies as a Burr-Brown. I googled them, they appear to be a US audio IC company.However, the USB ident information goes on to say "Burr-Brown Japan"... and the actual DAC chip idents are "Texas Instruments PCM2702 16 bit stereo audio DAC"Out of interest this was purchased as a "Pro-Ject Head Box" headphone amp.https://www.amazon.co.uk/Pro-Ject-Head-Headphone-Amplifier-Black/dp/B00AMKZ7C4Anyway, it's gorgeous, fat, clean, quiet and clear. I love it.

author: Buriedcode
author_rank: Super Contributor
timestamp: November 20, 2017, 09:22:24 pm
content: Burr Brown was aquired by TI in .. 2000 ? I'm guessing they just kept the VID/PID

author: David Hess
author_rank: Super Contributor
timestamp: November 21, 2017, 04:40:40 pm
content: Quote from: Hero999 on November 20, 2017, 06:44:55 pmI was just pointing out why they are noisier and they are not noisier than all operational amplifier; the same limitations apply to both. Emitter degeneration is the easiest way to increase the full power bandwidth and the added noise is not a problem with low efficiency speakers intended for solid state amplifiers. But a solid state amplifier like this used with high efficiency speakers intended for a lower power vacuum tube amplifier results in an annoyingly loud hiss.QuoteIt *was* overkill for many manufacturers and they either left the anti-aliasing filters out or used cheaper lower order filters. Besides the other causes like using the wrong equalization for mastering, I speculate that this was the cause of early "harsh" sounding CD audio.An intermodulation test would be required. With such a filter installed, there would be no test signal so no intermodulation. There is little reason to test for this in an audio power amplifier; to borrow an aphorism from programming, do not trap errors that you cannot handle.Quote from: paulca on November 20, 2017, 08:25:49 pmBefore Texas Instruments bought Burr-Brown in 2000 and erased their legacy, they were a premium analog and mixed-signal manufacturer similar to Linear Technology but originating from the time of Analog Devices and Analogic. Burr-Brown was an early player in the market for high end audio ADCs, DACs, operational amplifiers, and other functions. They published a rich assortment of application notes many or most of which are no longer available.

author: paulca
author_rank: Super Contributor
timestamp: November 21, 2017, 09:00:18 pm
content: Quote from: David Hess on November 21, 2017, 04:40:40 pmI think this is one of the re-incarnations today:http://www.box-designs.com/index.phpWhen you poke about at several of their amps and look up the datasheets they still say Burr-Brown on the headers.My wallet ran away and hid in the corner when I browsed that site.

author: quietcat
author_rank: Newbie
timestamp: December 19, 2017, 07:54:44 pm
content: Quote from: Hero999 on November 01, 2017, 11:37:24 amYou forgot to multiply by 20 to get the dB, so 0.6 dB may or may not be noticeable, depending on the situation.

author: 6PTsocket
author_rank: Regular Contributor
timestamp: January 16, 2018, 07:42:15 am
content: I agree that any decent amp has specs that far excede the distortion of the speakers that follow it. On the subject of soft THD, you are confusing an amp used for performance with one used for playback. Some musicians intentiinally overdrive tube amps beyond their linear range because the like the effect. Solid state amps clip sharply when the limit is reached and it sounds harsh and unpleasant. Tubes clip more gradually and some musicians use it as an audio effect. In the other hand, if you are trying to reproduce music, recordrd clean or intentionally distorted, you want a clean reproduction. You do not want to distort it again and add your own special effects. Therefore, whether tube or solid state you want to stay out of the clipped signal region and use an amp with enough power to drive the speakers, which also must be sufficiently rated, to the desired level without overdriving them.Sent from my SM-G900V using Tapatalk

author: Zero999
author_rank: Super Contributor
timestamp: January 16, 2018, 09:42:00 am
content: Quote from: quietcat on December 19, 2017, 07:54:44 pmI stand corrected on my calculations but my point still stands. The example I used was extreme: one cable double the length of the other and both around 10% of the speaker impedance. Ideally, the cable should have a much lower resistance, so any differences in attenuation will be much less.Quote from: 6PTsocket on January 16, 2018, 07:42:15 amI believe this thread is about Hi-Fi amplifier, i.e. playback only. Distortion was mentioned, but the idea is to minimise it by not overdriving the amplifier.