Perhaps what you meant to say was, "I am unaware of any theoretical or experimental support for that assertion."
its not hard to find subjective comments trashing 1990's BitStream DACs in general
and some technical criticism too
also the TDA1547 has internal hard wired I/V op amps - does the CD filter circuit totally bypass them?
the full scale single bit BitStream steps are hard on op amp inputs - passive prefiltering, large GBW, jfet input in older op amps or the new linearized input AD8099 are beter in such large signal out of band applications, did I forget to mention passive prefiltering
what does the sw-C modulator Clock feedthru look like? what is the order of the noise shaper? - external analog filtering should be several poles higher order roll off
and some technical criticism too
also the TDA1547 has internal hard wired I/V op amps - does the CD filter circuit totally bypass them?
the full scale single bit BitStream steps are hard on op amp inputs - passive prefiltering, large GBW, jfet input in older op amps or the new linearized input AD8099 are beter in such large signal out of band applications, did I forget to mention passive prefiltering
what does the sw-C modulator Clock feedthru look like? what is the order of the noise shaper? - external analog filtering should be several poles higher order roll off
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Those old desks had no opamps, what gave them there "sound" , according to Rupert himself, was the transformers. From mic to final mix, the signal easily went thru a dozen transformers. Every I/O on every pro piece of gear, including a lot of mics used transformers.
Here is the schematic of my CD-Player:
It is built very good for 1993. The TDA1547 is combined with SAA7350.
http://ultrashare.de/f/1234/DP1001_SM_KENWOOD_EN.pdf.html
cbdb, I'm glad you mentioned that, and not just because I like transformers. 😉
Anyone who has spent some time behind mixing consoles will know that they all have different "sonic signatures". Often this sound is brand consistent and over many years.
If you are forced to mix on a console you don't like, you may be fighting the console the whole time, never getting what you really hoped for. Other consoles can make it dead easy.
So while the argument that "The signal went thru dozens of opamps before it got to you" may be quite true for some gear, you don't know that the mixing and mastering engineers weren't fighting it the whole way. The music may well have sounded a lot better if it had not gone thru all those opamp circuits.
Anyone who has spent some time behind mixing consoles will know that they all have different "sonic signatures". Often this sound is brand consistent and over many years.
If you are forced to mix on a console you don't like, you may be fighting the console the whole time, never getting what you really hoped for. Other consoles can make it dead easy.
So while the argument that "The signal went thru dozens of opamps before it got to you" may be quite true for some gear, you don't know that the mixing and mastering engineers weren't fighting it the whole way. The music may well have sounded a lot better if it had not gone thru all those opamp circuits.
SY, I understand you. Sorry, this sounds a little bit arrogant to me. I have already said that I have drawn the wrong conclusion with feedback!
Yes, the phenomenon of manuplated soundstage seems not explainable. I think the problem is we are focusing on the common datas or culprits. That is the point.
What about phase anomolies? How does the NJM4565D handle the square wave from the DAC?
Does the NJM4565D settle to slow down so that it seems to add a kind of decay/reverb?
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Here's a crazy idea:
(Many) People could do listening tests AND make measurements of all of this feedback and phase anomaly and soundstage stuff, both at the same time, by using LTSpice. It can use WAV files for inputs and outputs!
Maybe it could be used to create some part of a uniform "experiment space" for multiple participants.
From one input WAV file, a separate listenable output file could be created for any topology or circuit that anyone wanted to compare (e.g. feedback vs no feedback or less feedback). The input file (and optionally the output files) and the LTSpice schematics could be distributed to participants.
The output WAV files would be listenable! Maybe everyone wouldn't hear the same thing, due to differences between participants' (non-PC) audio systems. But they might hear the same differences, most of the time, between either the input and output files, or between two different output files, depending on the particular experiment being performed.
And, with the simultaneous schematics, (each) one could then also perform standard (or non-standard) system-type testing. Things like phase anomalies and degraded transient response, FFTs, etc would be easy to plot and see (or not) and measure.
It's free from linear.com...
Tom
(Many) People could do listening tests AND make measurements of all of this feedback and phase anomaly and soundstage stuff, both at the same time, by using LTSpice. It can use WAV files for inputs and outputs!
Maybe it could be used to create some part of a uniform "experiment space" for multiple participants.
From one input WAV file, a separate listenable output file could be created for any topology or circuit that anyone wanted to compare (e.g. feedback vs no feedback or less feedback). The input file (and optionally the output files) and the LTSpice schematics could be distributed to participants.
The output WAV files would be listenable! Maybe everyone wouldn't hear the same thing, due to differences between participants' (non-PC) audio systems. But they might hear the same differences, most of the time, between either the input and output files, or between two different output files, depending on the particular experiment being performed.
And, with the simultaneous schematics, (each) one could then also perform standard (or non-standard) system-type testing. Things like phase anomalies and degraded transient response, FFTs, etc would be easy to plot and see (or not) and measure.
It's free from linear.com...
Tom
I will expand. There are audio processors that artificially widen the sound stage of even mono signals. I believe they do this by freq dependent time/phase shifting of one or both channels so they are no longer identical (not sure of details). I was suggesting that if there really is a soundstage difference between 2 amps (which I dont believe if there decent amps) it may be caused by a phase anomaly (distortion) in one channel of one of amps. The unfortunate thing is that a lot of people would choose the distorted amp because it "sounds better". (some people even prefer one channel completely out of phase because it "sounds better".) As I have said a number of times, most people like a bit of distortion (not necessarily THD, but some sort of inacuricies).
If you listen to a mono signal there is no sound stage width, it would then make sense that this width is only dependent on the differences between the left and right channel. So if your amps 2 channels are identical (mono signals always sound right in the middle), even if its the worst amp on the planet it will not effect your sound stage width.
Did you mean feedback causing some phase anomalies is from its lag response?
Here some input voltage plot (between V+ input/ noninverting and V- input/inverting) and output voltage at my CFA NFB amplifier. first plot (worse) using 100pf Cc, other plot using 50pf Cc otherwise with no Cc text.
(Sorry, the polarity is inverted)
Here some input voltage plot (between V+ input/ noninverting and V- input/inverting) and output voltage at my CFA NFB amplifier. first plot (worse) using 100pf Cc, other plot using 50pf Cc otherwise with no Cc text.
(Sorry, the polarity is inverted)
Attachments
Nice plots. Of course we all know that capacitors cause phase shift, so less cap, less phase shift. Now you feed back a sample of the output signal to the (inverting) input. If there was no phase shift, the fed back signal and the input signal (on the non-inverting input) would be in phase, and would nicely cancel (almost completely, if there's enough feedback), so there is not much 'input error' voltage. You increase Cc, more phase shift, less complete cancellation, more error.
Now lets think this through. Because of the higher Cc, the output signal would start to drop quicker. But, thanks to the feedback, the larger Cc causes a larger error signal at the input that counteracts the quicker fall-off of the output signal. The result is that the amp more faithfully reproduces our music, up to higher frequencies. I think we should like that a lot. Or not?
jan didden
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Not at all. They do that by inserting some of the L signal into the R channel, in opposite polarity, and vice versa. There's no mysterious time shifting (whatever that means) involved.
jan didden
The operation of an effects box (which does a gross amount of interchannel mixing) is unrelated to feedback in an amplifier. There are no phase "anomalies."
Sorry, that sounds arrogant again.
Brr.
I have said several times that it isn't the feedback. How many times do I have to say it?
The opamp alone has problems to handle square waves.
I recommend that a second person makes a test with the opamps. I am afraid the test will be not objective.
SY could claim he heard no difference to prove him correct.
SY, you are moderator?
Be fair.
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