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Since you seem to think my current setup is "overkill" for my "modest" room, you may want to try this calculator I found over the Internet.In my case, my speakers are 2.3m away from the listening position, and the front speakers are rated at 90dB SPL @ 1 watt @ 1 meter.
Music can be as loud as 110dB SPL
Plugging these figures into the calculator shows that I need an amplifier with a staggering power of 1055 watts just to get to 110dB!
However, that's just for one channel. In a stereo setup, I need 3dB less (since 107 + 107 dB adds up to 110 dB). Also, I place my speakers fairly close to the wall, so I'm probably getting an additional 3dB in terms of sound reinforcement from reflections.
So, I probably only need each amplifier to generate the equivalent of 104dB from each speaker. Recalculating shows that I need a power amplifier that can generate 265 watts.
Perhaps by coincidence, my Musical Fidelity A5cr power amps are rated at 255 watts per channel in 8 ohms.
So, do you still think my amplifiers are overkill? It seems to me that they just barely scrape through with enough power to handle the full dynamic range of music in my "modest" room.
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Follow Ups:
Hello,sorry for butting in on your conversation.Can I just ask,does one get the full dynamic range of audio by having a certain amount of watts driving your speakers?What I'm asking is,do you you have to crank it up to get the full dynamic range,or does having 200 odd watts of amp power give the full dynamic range from noise floor to ceiling,no matter if you listen quietly,or upset the neighbours?I'm asking because DVD-A has a wider dynamic range,and I'm sure my gear isn't up to it.I've got about 35watts per channel(JVC A-x2 in silver) and possibly a low spl pair of speakers(Goodmans Magnum K2-maybe circa 1980's).Pardon for interupting.
A typical listening room has a "noise floor" of around 40-50dB SPL. Just to get down to 40dB would require you to turn off all sources of noise in your house (including air-conditioning, fridges, dishwashers) plus an attempt at soundproofing your room. To give you an example, my own room measures just a tad below 50dB (but that's with my HTPC turned on to do the measurement!).Most people listening at average levels between 70-90dSPL, which means the music peaks at around 90-110dBSPL.
which means usable dynamic range can be as low as 20dB or as high as 70dB.
You can get more dynamic range by going multi-channel, simply because you have more speakers, therefore overall volume is louder.
But a realistic maximum is around 120dB SPL, which is the threshold of pain.
I've calculated that my system could possibly reach a maximum of 119.7dB, but that's with ALL 7.1 channels at maximum level, and my amplifiers will be drawing almost 2000 watts from my dedicated 20A circuit. And even then, I won't be able to sustain this level for any length of time (even if my ears could withstand it!) - the amplifiers will be draining any reserve capacity in the choke regulated transformers.
As an aside, this analysis shows why DSD ultrasonic noise is irrelevant. Even if our ears can hear beyond 20kHz (which has never been proven), DSD ultrasonic noise reaches a maximum of -70dBFS at 50kHz, which as you can see is below the noise floor of most set ups. People who complain that they can "hear" DSD ultrasonic noise are more likely to be hearing artefacts generated by limitations in their equipment.
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I tend to agree with that statement. I certainly can't hear it. However, I also tend to believe that non-integral sample rate conversion cannot be audibly distinguished from integral sample rate conversion. And there is certainly no conlusive evidence to the contrary for that either.There is a psychological effect for me however. Just knowing that there will be temporal errors introduced in non-integral SRC, regardless of how infinitesimally small (inaudible) they may be, just rubs me the wrong way. So I prefer integral SRC.
For the same reason I prefer high bit/sample rate PCM to DSD. Why bother with a system that generates measurable noise-shaping artifacts, whether audible or not, when you can have a measuably cleaner PCM reproduction.
But that's just the engineer in me I guess. I always prefer the most elegant solution to a problem.
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ALL sample rate conversion generate quantization noise and artefacts - one day we will have DSPs with sufficient power (in terms of filter taps and numerical precision) to make this a non-issue, but that day hasn't quite arrived yet.*** Why bother with a system that generates measurable noise-shaping artifacts, whether audible or not, when you can have a measuably cleaner PCM reproduction. ***
The "system" that you are not bothering is the same "system" that's embedded in just about every PCM ADC and DAC so the artefacts are not confined to DSD. Look at spec sheets.
The "measurably cleaner" PCM reproduction is because the modulation typically happens at 128x rather than 64x but the ultrasonic artefacts are still there. The only way to avoid it is to use a ladder type DAC, but that introduces other problems which is why they are becoming a rarity these days.
DSD is an interesting idea, comparable to the idea of capturing the "raw" image from a digital camera instead of converting to TIFF (PCM) or JPEG (compressed audio). There are benefits of capturing data "raw" (in this case, the undecimated delta sigma bitstream) and defer conversion to PCM, but the acquired signal is not easily editable without conversion.
However, 64fs may be good enough at one stage, but these days 128fs is the "norm" for both ADCs and DACs, and "hybrid" designs (multi-bit plus sigma delta) is common, so "raw" really should be a 5 bit 128fs bitstream rather than a 1 bit 64fs bitstream. It's kind of like forcing digital cameras to capture 1 megapixel "raw" images when the CCD is actually a 10 megapixel wafer, so the original "benefit" of DSD (not throwing away data) becomes a moot point. To me, that's a far more serious objection to DSD than ultrasonic artefacts.
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"We make all our recordings in PCM. And we try to fit them into a very short total time because the problem is if you make more than 65 minutes on a hybrid SACD you have a problem with the dynamics. And that is the reason why we try to reduce the total time. Then it is possible to make a good sound in SACD.You must know that the DSD signal is very bad. They say you have a frequency response up to 100kHz. But the problem is when you come over 40kHz you have a lot of distortion, and it is not possible to cut this distortion. You must take it onto the SACD disc itself. And when you have a lot of dynamics, and a lot of total playing time, then you don’t have enough room on the SACD. So you must reduce the dynamics. And this is the reason why you find a lot of SACDs that are poor in dynamics. And that is very bad. And therefore the SACD is a big big technical problem. But when you make a PCM recording and when the playing time is not too long, you can realise a good result.
The reason why we have DSD is that they have the license for the CD. But this ability to earn money stopped two years ago, because the 25 years was over. And this is the reason why they think, ‘well, what can we do to get more money?’ And the reason is DSD, but DSD is a very problematic and a difficult format.
And when you read the instructions for recording they say that DSD is only a ‘consumer’ format, i.e. you should make PCM recordings, and when you bring it on a disc, you can use DSD. But most of the record companies don’t speak about it. It is not a good format, but the problem is that the marketing for the SACD was very good. They spent more than 50million euros for the labels to make recordings on SACD. They started to buy recordings. They went to all the independents saying: "Well, we spend the money for three or five recordings and you make it, we pay it." After three years they stopped it — at the end of last year."
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Tsk, tsk... Martin, are you quoting KEKL? This sounds SO like him. If so, very funny! However, if you intended to submit the quote as a serious critcism of DSD... uhhh the word RUBBISH comes to mind. English being a second language is a fine excuse for shaky grammar, but what's the excuse for having no F***ING CLUE about what he/she's talking about?"You must know that the DSD signal is very bad"
Just another paranoid Sony hater with NOTHING to back up what he/she says. And not even willing to take credit for the ill-formed opinion that's being spewed. Try again using some ACTUAL information from a REAL person.
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He was a very real person, who actually makes classical hybrid SACDs. And somebody who was actually offered "incentives" by the DSD consortium.I phoned him 'out of the blue'. He was assumming I was a typically non-technical member of the public. Sometimes that's the best way to find out how people in the industry really feel.
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nt
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"You must know that the DSD signal is very bad."This guy sure sounds like he knows he's talking about something.
"It is not a good format, but the problem is that the marketing for the SACD was very good."
Oh yes I've never seen such fantastically successful marketing before.
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English is not his first language.
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Although I have no basis for refuting the remarks.
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. . . so I will respect the wishes of the source not to have their identity repeated here. I do understand that some people here will want a "byline" so to speak. But I'm afraid for the reason I stated, I will not name my interviewee.
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Chances are, if you are listening to a PCM recording that has undergone dither at some stage in it's lifecycle, it will contain ultrasonic noise, in many cases similar to DSD ultrasonic noise.Check out the following URL for comparisons of noise shaping curves for various dithering algorithms. What the diagrams don't show of course is that the noise shaping continues into the ultrasonics, and most algorithms work on the principle of shifting noise from the audible band up into the ultrasonic spectrum.
In this respect, DSD is no better or worse than any of the commonly used algorithms.
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"What the diagrams don't show of course is that the noise shaping continues into the ultrasonics"The noise energy can only be distributed within the systems bandwidth if pcm is used.
(Upper part being ultrasonic of course)Beyond the upper bandwidth limit of a pcm system the noise will be filtered.
Also in this repect DSD has the disadvantage that the noise shaping needs to be done in the ad converter and as a consequence the noise is *always* included in the recording.
With pcm this depends on the hardware/software algorithms used.
Frank
*** "What the diagrams don't show of course is that the noise shaping continues into the ultrasonics"
The noise energy can only be distributed within the systems bandwidth if pcm is used. ***If you look at the diagrams, they all cut off at 30kHz. What I said was (and I think you are agreeing with me) that the noise shaping continutes beyond 30kHz into whatever the upper limit is.
*** With pcm this depends on the hardware/software algorithms used. ***
Well, that is a moot point, since there will be embedded noise in PCM whenever delta sigma A/D is used. The only way you can avoid ultrasonic noise is by using ladder A/D (which has linearity issues) or by generating waveforms synthetically.
There is also additional ultrasonic noise generalted by sigma delta DACs, above and beyond noise embedded in the recording. My sound card implements a filter that is down -2dB at 50kHz to counteract this noise.
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..Well, that is a moot point, since there will be embedded noise in PCM whenever delta sigma A/D is used. The only way you can avoid ultrasonic noise is by using ladder A/D (which has linearity issues) or by generating waveforms synthetically...Not quite...
Modern AD converters using 1 bit conversion can have higer internal sample rates and can have a flat noise floor for the entire usable PCM bandwidth.
Any rise in noise floor due to a noise shaper beyond the nyquist frequency is filtered out.
With DSD this rising noise floor is always found within the usable bandwidth.
With PCM the noise floor can be flat across the useable bandwidth.
(The latest BB ADC goes flat up to 50kHz. pdf page 10)
Frank
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'If you look at the diagrams, they all cut off at 30kHz. What I said was (and I think you are agreeing with me) that the noise shaping continutes beyond 30kHz into whatever the upper limit is.'It really cannot continue beyond the upper band limit with PCM.
(Unless a bitstream dac with a noise shaper is used. But that's a moot point in a discussion abbout about adding noise shaping during 'processing' of the audio data. )
... you know that there *are* actually options for ultrasonic noise distribution and noise shaping methods.
I know.But that isn't an issue here.
What is is that in principle noise shaping isn't part of PCM and can be avoided with hires.
With DSD the noise shaping is required in principle. The noise shaping option in DSD tooling is on top of the noise shaping in the analog to DSD converter.
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Just as DSD noise-shaping artifacts are inaudible (I think we all agree here), so too are PCM noise-shaping artifacts, espcially considering how much more pervasive the noise-shaping is in DSD.So we again come back to my original premise: high-rate PCM is more elegant than DSD. If DSD were audibly superior to high-rate PCM then the issue of elegance would be moot. But of course no one has shown conclusively that DSD can be audibly distinguished from high-rate PCM.
Another thing I've been wondering lately...if the original CD patent had specified 44.1/20 instead of 44.1/16 would we even be having this conversation? I'm beginning to think anything above 48/24 may be overkill...
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*** So we again come back to my original premise: high-rate PCM is more elegant than DSD ***I'm not sure what you mean by "elegance" but in the case of ultrasonic noise it's a completely moot point since as I've pointed out real life PCM recordings have just as much ultrasonic noise as DSD so I don't know what you mean when you say noise shaping is more pervasive in DSD.
Using a different criteria, one can easily argue that DSD is more "elegant" than PCM. In fact, I think Sony and Phillips *are* advancing such arguments. But their arguments are equally moot because many SA-CDs are mastered from PCM recordings.
I would say - forget about the elegance and just enjoy the music!
*** if the original CD patent had specified 44.1/20 instead of 44.1/16 ***Be thankful they did not use 14-bits as originally proposed. It's easy to moan and whinge in hindsight but you must remember at the time it was impossible even to achieve 14 bit linearity, much less 16 or 20. The only reason we can get 20-bit accuracy these days is due to delta sigma (which DSD is based on, so you are accusing something for not being elegant when it is in fact the technology that broke the linearity barrier!)
*** I'm beginning to think anything above 48/24 may be overkill ***
On my soundcard, 44.1/24 generates the "best" results, as you can see from the attached link
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"On my soundcard, 44.1/24 generates the "best" results, as you can see from the attached link"
'so too are PCM noise-shaping artifacts'There is no such thing as 'PCM noise shaping artifacts' those are just plain noise shaping artifacts.
24 bits PCM at 48ks or higher doesn't require noise shaping at all.
Just plain old dither would be sufficient and probably the best option to use with 48ks. Higher sampling rates give the option to move the noise spectrum higher up the inaudible frequency range and use a less agressive noise shaping algorithm.Noise shaping is only a trick to enhance the perceived resolution on delivery formats with lower resolution.
Frank
That’s a 10,000 to 1 different between the initial impact and the average output. With more power music has more impact and thus sounds more real although you only need the mega-wattage for milliseconds at a time. It is worth the investment.
Also keep in mind that most solid state amplifiers are capable of delivering significantly more than their continuous rated power on transient peaks. If you look at the peak current output rating for your amp and the minimum load impedance presented by your speakers, that will give you some idea of how large the available instantaneous power delivery might be for handling brief transients like the attack of a triangle hit.
In the instruction manual for my Arnie Nudel designed Infinity Reference Standard Kappa 7's it says to “buy an amp with as much power as you can afford.”With my 4 ohm Kappa 7's I discovered I needed not only lots of amp power but also one that was stable down to 1 ohm, because the Kappa's 7's hit a low of 2 ohms.
I had a Denon 80 channel watts per channel receiver that ran very hot with these speakers and didn't have the "oomph" they did at the store sort of like they were running out of steam. That was in 1993 and I talked to Arnie and he suggested I select an amp with at least 150 watts per channel that was stable down to 2 ohms and that would solve my problems. I worked at the Good Guys back then and I was discussing the problem with representative from Adcom. He suggested the GFA-555II as a solution as it "measured" steady down to 1 ohm and outputted 325 Watts into 4 ohms and he correctly said I would never hear strain or overload again. The speakers have performed flawlessly for the past 13 years!
So Chirstine at 90dB @ 1 watt per meter your speakers are 2dB more efficient than mine so you amp power seems to be just about right.
I too have always preferred the sound of inefficient speakers.
Most speakers seldom stay at their "nominal" impedance and dip way below nominal for certain frequencies/volume, mine for example goes as low as 4.6 ohm, so even more amplifier power is required.*** didn't have the "oomph" they did at the store sort of like they were running out of steam ***
This is a really astute and insightful observation and is a good sign that your previous amp was clipping. I tend to describe it as a "compression" or "hardening" of the sound esp. at peaks.
I hear this when I tested my NAD 50w amp on my speakers. I also heard this on my previous amp (Denon rated at 170w per channel, but I doubt it delivered 7x this with all channels driven).
A really good test is i think track 3 on the Gladiator soundtrack. When I heard this at the dealer on the Halcro and the B&W Signature 800s it was glorious. Then when I listened to it on the Denon I can hear obvious signs of compression/hardness/clipping. With my new amps, I am happy to report i hear no problems :-)
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Clip a good voltmeter with peak hold capability across outputs of one channel of your amp in parallel with the speaker. Crank it up and see what the amp is really putting out in your room. If you know the impedence of you speaker, you can get a rough calculation of your power output.
Indeed, the jist of your post closely mirrors some of the copy Musical Fidelity is using to sell their new line of amps. Specifically (about the A5):"In our opinion almost all hi-fi amplifiers available are grossly underpowered. With speakers of anything less than 95dB efficiency it is a fact that amplifiers with less than 200 wpc are clipping (and that means distortion and non-linearities) frequently."
The above from their website.
I tend to agree with this approach. When specifying power amps for PA systems, headroom is VERY important. In audio, too much is always better than too little.
I've never noticed that line before, thanks for pointing it out. That statement is included in MF's blurb on the A5 integrated, which I've never read because I always click straight through the web page on the A5 power, which was what I was interested in.Surfing around, I noticed MF has a web page that explains exactly what they mean by that statement, and the reasoning is very similar to the reasoning I used (see below).
Serendipity!
I guess this highlights the fact that the majority of amplifiers sold today are in fact grossly underpowered and even the A5 is barely sufficient, despite MF's hype (I noticed their "comparison" to the "competitor" amp is a bit misleading, since the competitor amp had less than 1/3rd the distortion which accounts for the high asking price!)
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This is the correct one
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An amplifier "power" is usually measured at the point where the amplifier notionally "clips" (ie. 1% THD + noise).However, if you do a plot of THD vs power for most amplifiers, distortion rises very rapidly way below the notional clip point. An example is the graph linked below, which is Stereophile's measurement of the Musical Fidelity Nu-Vista 300.
Note that although the amplifier's power is measured at over 300 watts, distortion rises rapidly above 200 watts.
Arguably, if you care about distortion, this amplifier doesn't really deliver much above 200 watts.
Distortion is important because I've measured my speakers and they are capable to reproducing music with as a little as 0.1% THD in the presence region (where our ears are the most sensitive). I certainly wouldn't want an amplifier generating distortion at 1% and negate all the advantages of my speakers.
Using this more conservative approach, I doubt my A5cr can deliver much more than say 150 watts, in which case the maximum I can listen to music without distortion or clipping is only about 80dB SPL.
By coincidence this is actually the level that I listen to most music, so I could argue my amplifier is really just powerful enough for my listening needs.
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Let's assume that I don't need to listen at 110dB. In fact, let's assume I only need to listen to music at around 85dB, since I like to preserve my hearing.Uncompressed music have transient peaks up to 25dB above average. So, entering a desired level of 85dB into the calculator, but with a headroom of 25dB gives the result of 529 watts.
So, in other words, my "overkill" amplifiers are not even capable of playing back music at 85dB without some clipping of transients.
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