That's funny...
I watch BBC 1 and BBC Entertainment at home, as well as CNN amd Bloomberg, plus the inescapable Sky, and I find BBC has the best sound overall and in general. Clean and clear like a whistle, with good dynamics.
CNN is also good, but somehow it's a little too dynamically flat for its own good. Bloomberg is below par, altthough still acceptable.
Sky is surprisingly good, I suspect in part because they also show Fox Crime and Fox Life, plus they own one of local national TV stations, so I suspect they saw reason in investing into quality gear and training local staff.
I watch BBC 1 and BBC Entertainment at home, as well as CNN amd Bloomberg, plus the inescapable Sky, and I find BBC has the best sound overall and in general. Clean and clear like a whistle, with good dynamics.
CNN is also good, but somehow it's a little too dynamically flat for its own good. Bloomberg is below par, altthough still acceptable.
Sky is surprisingly good, I suspect in part because they also show Fox Crime and Fox Life, plus they own one of local national TV stations, so I suspect they saw reason in investing into quality gear and training local staff.
It depends on what DVB standard you receive.
For "normal" DVB-T it's MPEG2 for HD DVB-T usually MPEG4.
But the bitrates used vary wildly (and so does the quality). A 64QAM 8K CR2/3 GI1/4 DVB-T multiplex is ~ 31Mbit /sec (from the top of my head) usually containing up to 5 TV channels.
DVB-C usually does a little better with regard to bandwith (modulations used are usually QAM64,128 & 256 depending on the S/N ratio of the cable network).
For "normal" DVB-T it's MPEG2 for HD DVB-T usually MPEG4.
But the bitrates used vary wildly (and so does the quality). A 64QAM 8K CR2/3 GI1/4 DVB-T multiplex is ~ 31Mbit /sec (from the top of my head) usually containing up to 5 TV channels.
DVB-C usually does a little better with regard to bandwith (modulations used are usually QAM64,128 & 256 depending on the S/N ratio of the cable network).
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I was wrong, the effective bit rate of a DVB-T multplex 64QAM 8K CR2/3 GI1/4 is ~20Mbit /sec. This would yield ~3.5Mbit /sec if 5 TV stations are in the same multiplex. Not a big surprise the quality isn't always top notch..
http://www.dvb.org/technology/fact_sheets/index.xml
http://www.dvb.org/technology/fact_sheets/index.xml
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Yes, very good stuff. But other national broadcasters do a great job. Having spent a bit of time at la Maison de la Radio (Radio France) in Paris and many years listening to the FM feeds, I can say they do a very good job. To my ear, as good or better than the Beeb.
Radio Netherlands also do great work.
@Pano
As I said, the greatest problem as is are the high range speakers, which are greatly subdued to my mind. I need a +8 dB treble lift to get them into what I hear as a normally balanced top to bottom range. Having experienced similar feelings elsewhere, and knowing that once you change the factory installed speakers for something half decent, +8 dB would cause screaming and demand I lower this to +2 dB worst case.
At least, that's what happened with the old car.
On the plus side, new speakers also afforded me a lot more general clarity and coherence, not like at home, but rather good for a car. Certainly good enough to satisfy my car listening requirements. And trust me, I really don't demand a lot from a car audio system, just decent is good enough.
As I said, the greatest problem as is are the high range speakers, which are greatly subdued to my mind. I need a +8 dB treble lift to get them into what I hear as a normally balanced top to bottom range. Having experienced similar feelings elsewhere, and knowing that once you change the factory installed speakers for something half decent, +8 dB would cause screaming and demand I lower this to +2 dB worst case.
At least, that's what happened with the old car.
On the plus side, new speakers also afforded me a lot more general clarity and coherence, not like at home, but rather good for a car. Certainly good enough to satisfy my car listening requirements. And trust me, I really don't demand a lot from a car audio system, just decent is good enough.
Driving around Europe especially Germany is excellent . The bosses RDS Sony was excellent . I listened all night driving from the Austrian boarder to North Germany . The RDS seamless . If we loose FM we loose that which for most is the best hi fi they ever had . That Sony and Quad 63 's would interest me . I bet it would be stunning .
My little Sony portable ( $60 ) is almost as good as my old Quad FM3 . Certainly better than all digital alternatives . The now defunct TV 10 bit Nicam was sometimes the best . I put that down to balance on a particular night ( eg R3 and BBC TV simultaneous transmission ) . I think I could detect that the FM ( 13 bit ) had more detail when almost equal . Apparently the BBC did transmit some 13 bit . 10 bit was chosen because it is more durable ( Wales , multipath ) . A test transmission to Brighton gave no picture yet excellent sound in 10 bit as the first public demonstration ( 10 miles outside of TV range ) . I would guess 2 uV signal . This is from memory , someone from the BBC might know more accurate information . If only we could have the 13 bit's to listen to in digital . My FM3 I am sure is not the reason why it sounds good . It must be adding 1% distrotion and plenty of hiss ? I was looking at 10 bit encoding . Whilst not perfect it might be interesting .
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Just lifted this from a discussion about Chaos Theroy .
Chaos was observed by a number of experimenters before it was recognized; e.g., in 1927 by van der Pol[52] and in 1958 by R.L. Ives.[53][54] However, as a graduate student in Chihiro Hayashi's laboratory at Kyoto University, Yoshisuke Ueda was experimenting with analog computers and noticed, on Nov. 27, 1961, what he called "randomly transitional phenomena". Yet his advisor did not agree with his conclusions at the time, and did not allow him to report his findings until 1970.[55][56]
Op amps were intended to be this . I wonder if polynomials explain all ? The starting point is always the difficulty . Chaos is not random .
Chaos was observed by a number of experimenters before it was recognized; e.g., in 1927 by van der Pol[52] and in 1958 by R.L. Ives.[53][54] However, as a graduate student in Chihiro Hayashi's laboratory at Kyoto University, Yoshisuke Ueda was experimenting with analog computers and noticed, on Nov. 27, 1961, what he called "randomly transitional phenomena". Yet his advisor did not agree with his conclusions at the time, and did not allow him to report his findings until 1970.[55][56]
Op amps were intended to be this . I wonder if polynomials explain all ? The starting point is always the difficulty . Chaos is not random .
As nobody seems to be doing this thread , how about this . Do valves/tubes add useful colour to the sound and make it nice ? I find they don't if we say to a reasonable hi fi standard ( DIN45500 ) . When they do it , it is not my cup of tea . Some valve amps were very low distortion . Accepted wisdom says we should not be hearing coloration if below 0.1% THD espeacailly when class A amps . Doubtless low damping factor is a coulouration and might sometimes be heard as boom or poor frequency response . Also transformers or circuits with poor frequency response . To me voice quality is better and distortion can be low . Camouflage is not what they do , or do well if they do . Marantz Model 9 followed by the Dynaco ST70 as low colouration designs of note .
Here's a crazy idea: What about actually trying to find ways to advance the state of the art, in terms of discovering or developing (or just sharing) measurements or methods that could be used to help predict "sound quality", with the goal being something like "developing some understanding and some methods that could be useful in improving everyone's ability to repeatably generate better audio system designs"?
I "accidentally" posted a rough initial outline of one possible type of scheme for a process for trying to do that, somewhat collectively, earlier tonight, starting at the fourth paragraph of post 878, which is at:
http://www.diyaudio.com/forums/solid-state/216409-power-supply-resevoir-size-88.html#post3145013
I also posted a method for measuring the distortion of only the tops and bottoms of a pulse train, within an LT-Spice simulation, which is illustrated at
http://www.diyaudio.com/forums/solid-state/216409-power-supply-resevoir-size-85.html#post3142618
In that case, it was only for use as a tool to examine the output error levels of automated sequences of simulation runs where PSU reservoir capacitance was varied, in order to more-easily be able to find the largest capacitance value at which the power supply could still "mangle" the output waveform, by any amount.
But I plan to eventually try to transform that type of processing method into something that can be used to help evaluate any simulated system's transient response accuracy (i.e. not just the tops and bottoms of the pulses), especially for automated sequences of simulation runs where various circuit parameters are step-swept through ranges of values.
Measuring Transient Response accuracy in different ways should be important because: a) steady-state performance measurement is already relatively easy and b) low sine distortion is relatively easy to achieve, and c) low steady-state (sine) distortion seems to not be a good predictor of "sound quality", anyway (although I would probably want to assume that having some acceptable steady-state distortion level would be necessary but not sufficient), and (also: ) therefore the Transient Response must be involved.
Before anyone pooh-poohs spice modeling as a tool for such purposes, think about how repeatable it is. People far apart can test and compare with exactly the same circuit, rather than with different implementations of the same circuit. That might be important in speeding up a development process by leveraging through the use of multiple people, connected by the internet (or even simply multiple instances of a circuit, being simulated on multiple PCs at the same location). Simulation is cheaper, faster, and much more repeatable and controllable. I do understand that there are also many limitations.
Chers,
Tom
I "accidentally" posted a rough initial outline of one possible type of scheme for a process for trying to do that, somewhat collectively, earlier tonight, starting at the fourth paragraph of post 878, which is at:
http://www.diyaudio.com/forums/solid-state/216409-power-supply-resevoir-size-88.html#post3145013
I also posted a method for measuring the distortion of only the tops and bottoms of a pulse train, within an LT-Spice simulation, which is illustrated at
http://www.diyaudio.com/forums/solid-state/216409-power-supply-resevoir-size-85.html#post3142618
In that case, it was only for use as a tool to examine the output error levels of automated sequences of simulation runs where PSU reservoir capacitance was varied, in order to more-easily be able to find the largest capacitance value at which the power supply could still "mangle" the output waveform, by any amount.
But I plan to eventually try to transform that type of processing method into something that can be used to help evaluate any simulated system's transient response accuracy (i.e. not just the tops and bottoms of the pulses), especially for automated sequences of simulation runs where various circuit parameters are step-swept through ranges of values.
Measuring Transient Response accuracy in different ways should be important because: a) steady-state performance measurement is already relatively easy and b) low sine distortion is relatively easy to achieve, and c) low steady-state (sine) distortion seems to not be a good predictor of "sound quality", anyway (although I would probably want to assume that having some acceptable steady-state distortion level would be necessary but not sufficient), and (also: ) therefore the Transient Response must be involved.
Before anyone pooh-poohs spice modeling as a tool for such purposes, think about how repeatable it is. People far apart can test and compare with exactly the same circuit, rather than with different implementations of the same circuit. That might be important in speeding up a development process by leveraging through the use of multiple people, connected by the internet (or even simply multiple instances of a circuit, being simulated on multiple PCs at the same location). Simulation is cheaper, faster, and much more repeatable and controllable. I do understand that there are also many limitations.
Chers,
Tom
Careful with the damping factor, Nige. Accepted wisdom has it that it should be as high as possibel, i.e. the output impedance should be as low as possible, but from personal experience I know that this is not quite so linear.
For a start, there's the question of how was that DF obtained. Could be through multiple output series/parallel devices, through lots of global NFB, etc, and I find that some devices with a nominally smaller damping factor actually sound better than others with a higher damping factor, and in the bass region, too.
As for tubes, and I now risk my life from tube camp hit squads, I find that far too many of them sound like a call girl telling you in the morning that last night you were the world's bets lover ever - you know she is lying, but it's good to hear. They sound mostly "syrupy" to me, sticky, too sweet.
On the other hand, I do not believe this is the way it has to be, rather that this is what has come to be accepted as "tube sound", as what is customer anticipated. After all this time, I feel reasonably certain that say Waveborn could do a nicely neutral tube amp I could really enjoy without the usual "sticky" character of commercial audio. I have heard a tube amp from an Italian company (I forget the name) which was supremely neutral and had an uncanny tendency to disappear sonically, leaving you and the music. Its nominal DF was just 10:1, and let me assure you, it had no problems whatsoever with the bass.
As most things in audio, everything is very much relative.
@gootee
Tom, I haven't read the other thread yet, but I will. That said, I find that a reasonable rule of thumb (no science, just quick'n'dirty) for transformers is to use the number of watts you need from it in any case, add say 15-20% reserve, and multiply the number of watts by sq. root of 2, i.e. 1.41, for the required VA rating. Thus, for 200W/4 Ohms, I'd use (200*1.2)1.41, or 338 VA. Since I cannot have just that locally, only 300 or 400 VA, I'd go for the 400 VA. More never hurts.
As for the capacitors, 10,000 uF is good enough for 100W/8 Ohms, so for 200W/4 Ohms, as I see it, I'd need double that, as the current is double the initial.
What I'd like to know is how actually useful would it be to switch from one bridge rectifier for both + and - sides, as compared to using two full wave bridge rectifiers, one for the + side, and the other for the - side. Any thoughts?
Tom, I haven't read the other thread yet, but I will. That said, I find that a reasonable rule of thumb (no science, just quick'n'dirty) for transformers is to use the number of watts you need from it in any case, add say 15-20% reserve, and multiply the number of watts by sq. root of 2, i.e. 1.41, for the required VA rating. Thus, for 200W/4 Ohms, I'd use (200*1.2)1.41, or 338 VA. Since I cannot have just that locally, only 300 or 400 VA, I'd go for the 400 VA. More never hurts.
As for the capacitors, 10,000 uF is good enough for 100W/8 Ohms, so for 200W/4 Ohms, as I see it, I'd need double that, as the current is double the initial.
What I'd like to know is how actually useful would it be to switch from one bridge rectifier for both + and - sides, as compared to using two full wave bridge rectifiers, one for the + side, and the other for the - side. Any thoughts?
I think damping factor is interesting and undiscovered by most . Add a 0R47 to a modern amp and see what you get . Keep going up to 8R if you like . The amp might be very happy with 3R3 also . If theory is correct ( 1950's ) then that it where we start to notice if an 8R load ( boom ) . Remember most 8 R speakers are not exactly that so 2R2 might be more appropriate . If a speaker sounds a bit lean and dry it might help ?
Yes, but you also get, in the same package tour, bass boom, overhang and whatnot. God knows I love bass dearly, but I can't stand imprecise and uncontrolled bass.
0R47 I can stomach, but 3R3 is, in my view, far too high, it will deteriorate the DF to below 10:1, which is the acceptable minimum for speaker Q control.
Anyone remember the first (Kensonic) Accuphase integrated amp from the early 70ies? Well, it had a damping factor switch contro at the back, allowing for a DF of 66, 12 and 6, if memory serves. I remember the 66 or 12 test was a good one, but if memory serves, nobody was happy with 6.
...when particular speaker design does not rely on certain electrical damping. Most of modern woofers do, so damping factor for them is needed, however not of crazy numbers, just enough to damp resonances on main resonant frequency.
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