How do I calculate input impedance of this?
- By jimk04
- Analog Line Level
- 15 Replies
I have this SE to balanced converter using signal transformer. It is buffered with an OPA551.
The circuit looks something like this.
4.7uf series and a 10k shunt before the op amp.
Do I need to be looking at the OP551 DS to know this?
Thanks
The circuit looks something like this.
4.7uf series and a 10k shunt before the op amp.
Do I need to be looking at the OP551 DS to know this?
Thanks
Compact SMPS filter using AmyAlice circuit design
- By avdesignguru
- Power Supplies
- 6 Replies
Attachments
Measuring Loop Gain on the Bench - Best Way?
- By PB2
- Solid State
- 46 Replies
I'm looking for the best, simple, way to do this on the bench without more than
a function generator (my HP goes to 11 MHz) and a scope. Our scope has a
readout of the signal amplitude and is 200 MHz. This amp is a prototype so soldering
minor changes is not a problem.
I used LTSpice to draw what I'm planning to do on the bench to measure the loop gain
of the original Universal Tiger (UT) amp. This is NOT the way to do it in simulation but
of course it will run. The best way in simulation is to use a Tian probe as explained in
these videos by one of our members:
https://www.diyaudio.com/community/...al-tiger-inconsistencies.365751/#post-7737499
Let me say that I understand that this method introduces errors but from what I
see/understand I don't think that they are significant and I welcome comments to
improve it or use another method.
The main feedback path is broken at the output and the SPICE ideal signal source drives
it. The obvious errors are that the loading on the output by the feedback network
is removed (more parts could be added to emulate it) but since the load is 4-8 ohms I
believe that the error is insignificant. The other error is that the source impedance of
the amp output is not accounted for. Now amps like this are (should be) high damping
factor and therefore the output impedance is low in the audio band. The zobel is in
parallel and should keep the impedance low at high frequencies. The circuit needs a
stable DC output and therefore a feedback network matching the pos input resistance
to ground is added with two resistors (R29, R28) and a large cap to ground to shunt
the signal. This introduces error below 1 KHz but that is not where the analysis takes
place. There was significant DC offset at the output, as is seen in real hardware, and
a very large resistor, R30, was added to trim the offset. This is not how to do it in real
hardware but with an ideal power supply in sim it works fine.
The output to measure is the point where the loop is broken with the OUT label, not
the SPK labeled net. Plotting frequency response in LTSpice is OUT/Vin from the source
which is what we want. Anyway, here's the diagram:
Ran it to see if the results are reasonable and what to expect on the bench:
It shows an unstable amp, but perhaps I'm missing something, we do know that these
amps certainly do oscillate in real hardware.
Odd that LTSpice does not make the two cursors different colors in the plot.
==== PHASE MARGIN
Cursor 1, furthest to the right, is at the 0dB gain point (the horizontal cursor is at 0dB)
where we should look for phase margin on the phase plot (the vertical cursor). Note
that we are driving the negative input and therefore, in the passband, we see -180 deg
as would be expected on the inverting input. The point where there is zero phase margin
is another -180 for a total of -360 bringing the signal back in phase to cause oscillation.
The amount less than -360 is the phase margin.
Note that for cursor 1 the phase is greater than -360 at about -390 and according to
this there is about -30 deg of phase margin - there is none. Keep in mind that this is
device and model dependent so real hardware could be better or worse.
Or is this not a valid method?
==== GAIN MARGIN
Cursor 2, on the left, is where the phase reaches -360 deg (vertical) and the gain
margin is how much less than 0dB the gain curve shows. The gain is actually greater
than 0dB and therefore there's negative gain margin with it showing 12.47 dB of gain
(horizontal) - it should be negative in order to have gain margin.
It should have been obvious that the stability of this amp is highly dependent on the
process variation of the semiconductors which is demonstrated by the fact that I have
one amp that readily oscillates and another of the same revision that does not.
a function generator (my HP goes to 11 MHz) and a scope. Our scope has a
readout of the signal amplitude and is 200 MHz. This amp is a prototype so soldering
minor changes is not a problem.
I used LTSpice to draw what I'm planning to do on the bench to measure the loop gain
of the original Universal Tiger (UT) amp. This is NOT the way to do it in simulation but
of course it will run. The best way in simulation is to use a Tian probe as explained in
these videos by one of our members:
https://www.diyaudio.com/community/...al-tiger-inconsistencies.365751/#post-7737499
Let me say that I understand that this method introduces errors but from what I
see/understand I don't think that they are significant and I welcome comments to
improve it or use another method.
The main feedback path is broken at the output and the SPICE ideal signal source drives
it. The obvious errors are that the loading on the output by the feedback network
is removed (more parts could be added to emulate it) but since the load is 4-8 ohms I
believe that the error is insignificant. The other error is that the source impedance of
the amp output is not accounted for. Now amps like this are (should be) high damping
factor and therefore the output impedance is low in the audio band. The zobel is in
parallel and should keep the impedance low at high frequencies. The circuit needs a
stable DC output and therefore a feedback network matching the pos input resistance
to ground is added with two resistors (R29, R28) and a large cap to ground to shunt
the signal. This introduces error below 1 KHz but that is not where the analysis takes
place. There was significant DC offset at the output, as is seen in real hardware, and
a very large resistor, R30, was added to trim the offset. This is not how to do it in real
hardware but with an ideal power supply in sim it works fine.
The output to measure is the point where the loop is broken with the OUT label, not
the SPK labeled net. Plotting frequency response in LTSpice is OUT/Vin from the source
which is what we want. Anyway, here's the diagram:
Ran it to see if the results are reasonable and what to expect on the bench:
It shows an unstable amp, but perhaps I'm missing something, we do know that these
amps certainly do oscillate in real hardware.
Odd that LTSpice does not make the two cursors different colors in the plot.
==== PHASE MARGIN
Cursor 1, furthest to the right, is at the 0dB gain point (the horizontal cursor is at 0dB)
where we should look for phase margin on the phase plot (the vertical cursor). Note
that we are driving the negative input and therefore, in the passband, we see -180 deg
as would be expected on the inverting input. The point where there is zero phase margin
is another -180 for a total of -360 bringing the signal back in phase to cause oscillation.
The amount less than -360 is the phase margin.
Note that for cursor 1 the phase is greater than -360 at about -390 and according to
this there is about -30 deg of phase margin - there is none. Keep in mind that this is
device and model dependent so real hardware could be better or worse.
Or is this not a valid method?
==== GAIN MARGIN
Cursor 2, on the left, is where the phase reaches -360 deg (vertical) and the gain
margin is how much less than 0dB the gain curve shows. The gain is actually greater
than 0dB and therefore there's negative gain margin with it showing 12.47 dB of gain
(horizontal) - it should be negative in order to have gain margin.
It should have been obvious that the stability of this amp is highly dependent on the
process variation of the semiconductors which is demonstrated by the fact that I have
one amp that readily oscillates and another of the same revision that does not.
Building a vintage speaker?
Lately I've been depressed (no, don't worry, this isn't a sob story) and I try to distract myself somehow. I also noticed there gathered lots of drivers and horns and damping material and other speaker stuff, a lot of chinese amp boards, PSs and I realized that's a lot more than I really need or I'll use for my 'serious' projects. A lot of cheap stuff but also good ones I've often got by bidding on eBay, thinking "eh, just enter the minimum bid, it will sell for much more anyway" without really planning to do something with it or getting it with a project in mind. But now I thought I should really get rid of something or actually do a project with it.
I don't want to do the 5738th 2-way with a 17cm/25mm, I find that rather boring. I like big speakers. I like vintage HiFi but not everything has to come from the 70s or 80s. So, my question is, would you invest time, wood etc to build a speaker that's a reminiscence to vintage speakers?But would it be a good idea to use the cheap-ish drivers and horns etc you know aren't optimal? Would you build (to name a few well known speakers) a Klipsch Cornwall clone, LaScala, Heresy or vintage JBL (etc) if it re-creates the look very close, even if i.e. the horns don't have a modern dispersion pattern? With 'original' front cloth? Without? Or would you use something less close to the classic speaker? Modern color/finish or vintage all the way? Would you like it to sound as neutral as possible or prefer the sound of the original? Modify the principle? Like a LaScala with BR for LF extension? What classic/vintage speaker do you think would be nice to build?
Just to show one example, here some pictures of the Klipsch Hersey but I'm not fixated on the Hersey or Klipsch in general.
I don't want to do the 5738th 2-way with a 17cm/25mm, I find that rather boring. I like big speakers. I like vintage HiFi but not everything has to come from the 70s or 80s. So, my question is, would you invest time, wood etc to build a speaker that's a reminiscence to vintage speakers?But would it be a good idea to use the cheap-ish drivers and horns etc you know aren't optimal? Would you build (to name a few well known speakers) a Klipsch Cornwall clone, LaScala, Heresy or vintage JBL (etc) if it re-creates the look very close, even if i.e. the horns don't have a modern dispersion pattern? With 'original' front cloth? Without? Or would you use something less close to the classic speaker? Modern color/finish or vintage all the way? Would you like it to sound as neutral as possible or prefer the sound of the original? Modify the principle? Like a LaScala with BR for LF extension? What classic/vintage speaker do you think would be nice to build?
Just to show one example, here some pictures of the Klipsch Hersey but I'm not fixated on the Hersey or Klipsch in general.
DSP Active Crossover to Analog Active on a budget - Almost there, but need help with CD Horn Equalization - Altec 14s
Hi folks,
Background:
Have been working on a DIY project to replace the DSP functionality with analog, because I am OCD about SQ, and was able to hear immediately noticeable differences in Analog vs. Digital crossovers. Easily A/Bd at the same time as I was able to run the left channel through 1 crossover and right through another and easily level match using REW.
I've been using a Driverack PA2 with Altec Model 14s, but felt I could hear that was a definite difference in mid-bass texture/detail, soundstage and crispness of highs. Using the Driverack, I was able to follow the lead of the Audiophile Nirvana guys (https://www.audiophilenirvana.com/a...ws/altec-lansing-model-14-part-4-ultimate-14/) and get fantastic results.
2019 thread of that effort: https://audiokarma.org/forums/index...h-driverack-pa2-project-and-questions.870225/
With addtional tweaking I laned on LR 24 for the sub/mids and LR 18 for the mid/high, levels matched using the DSPs gain controls on the outputs and further EQ using the AutoEQ as a starting point. Was also able to perform time alignment.
I tried upgrading to a Venu360 and the new minidsp flex 8 (not balanced, unfortunately, but was still decently quiet), but that wasn't really much better. Digital signals are more noticeably impacted detail wise and analog signals seemed to lose some life. Tried a few pre-built pro analog crossovers and they were much better, but had a bit of noise and also I didn't like that the levels and in some cases crossover frequencies could easily be altered by an errant finger. The card versions are much better in this regard.
One thing's for sure... You can pretty much ignore the snr figures published by minidsp. Upon closer inspection they show the digital input to digital output and digital input to analog output figures, but never seem to put the analog in/analog out figures, lol. It was actually the worst of the bunch. Awesome for home theatre though, but the UI on the Harman dbx products is far superior and those include RTA and signal generators built in, so no having to setup and flipping back/forth to REW for anything except time alignment.
Results so far:
In the end, I settled on the Xkitz- 3-way FET opamp based balanced boards and linear ps, as seen here. Even with external EQ, quieter, more detailed, bigger soundstage, way better bass/mid-bass textures, clearer highs.
I used and old, broken SD dvd player as a donor for the power receptacle and case , with 3d printed front/back panels (still need to tweak the makeshift back) and just purchased a few standoffs to make this very budget friendly. This was a drastic improvement over the DSP xovers and quieter and more detailed than the pro analog crossovers. Completely silent, might be the slightest combo of buzz/hiss if it's very, very quiet and I wedge my head into the CD horn. Has never been this quiet. Balanced cables helped a lot here, but these xovers really cleaned it up.
So far I've been able to accomplish the following.
-crossover points
-levels
-eq (using a Kenwood GE-7030, absolutely fantastic!), with the exception of a dip at the mid/hi crossover
-phase adjustment to eradicate nulls and reduce dip at 1500hz xover point (changed wires at mid/tweeter as required using REW, sub has selectable phase 0, 90, 180, 270)
compromise...
-Only LR4 24db slopes available on the diy crossover.. I knew that and trying to be budget friendly and can't justify the Marchand crossover expense, particularly when this one gets me 90% there)
What remains is
1. CD equalization - the EQs 12db range handled the rest of the frequency response, but doesn't have enough differential to raise this dip enough.
2. Time alignment (not too worried about this as the mid/hi are under 2ms, and the sub is about 18ms, but moveable and a lot of is ringing from the 22' ceiling and gobs of windows)
So the questions:
I have a dip at the 1500 Hz region (and 10 to 20Hz needed to be lifted, the EQ took care of that) that I want to take care of.
1. Can I alter the circuit by inserting a capacitor to adjust for the CD horn, as is done in passive networks? (passive example here: https://techtalk.parts-express.com/forum/tech-talk-forum/1410233-cd-horn-passive-equalization)
2. Would it be better to add a circuit? (plenty of room in the case to add a circuit board... something like this? .... https://sound-au.com/project173.htm, but this seems to have minimal impact on the 1.5K region. Also wondering, that even so, might the overall flattening of the HF range help raise it enough for the external EQ to do the rest)
2. As a 3rd alternative, would I be able to easily alter the circuit to have perhaps an LR2 12db or LR3 18 slope for the tweeter low pass?
3. Should I forget about trying to make this work? I hate to do this as I'm almost there and like the price! Of course, if there's other similarly priced active crossover diy boards, I'm still in the 45 day trial period.
Trying to keep it budget friendly/diy friendly. Retired, so I have time.
Glad to post REW or Driverack analysis screenshots, if it helps.
Other info for context:
Thank you for any and all advice, suggestions or comments!
Background:
Have been working on a DIY project to replace the DSP functionality with analog, because I am OCD about SQ, and was able to hear immediately noticeable differences in Analog vs. Digital crossovers. Easily A/Bd at the same time as I was able to run the left channel through 1 crossover and right through another and easily level match using REW.
I've been using a Driverack PA2 with Altec Model 14s, but felt I could hear that was a definite difference in mid-bass texture/detail, soundstage and crispness of highs. Using the Driverack, I was able to follow the lead of the Audiophile Nirvana guys (https://www.audiophilenirvana.com/a...ws/altec-lansing-model-14-part-4-ultimate-14/) and get fantastic results.
2019 thread of that effort: https://audiokarma.org/forums/index...h-driverack-pa2-project-and-questions.870225/
With addtional tweaking I laned on LR 24 for the sub/mids and LR 18 for the mid/high, levels matched using the DSPs gain controls on the outputs and further EQ using the AutoEQ as a starting point. Was also able to perform time alignment.
I tried upgrading to a Venu360 and the new minidsp flex 8 (not balanced, unfortunately, but was still decently quiet), but that wasn't really much better. Digital signals are more noticeably impacted detail wise and analog signals seemed to lose some life. Tried a few pre-built pro analog crossovers and they were much better, but had a bit of noise and also I didn't like that the levels and in some cases crossover frequencies could easily be altered by an errant finger. The card versions are much better in this regard.
One thing's for sure... You can pretty much ignore the snr figures published by minidsp. Upon closer inspection they show the digital input to digital output and digital input to analog output figures, but never seem to put the analog in/analog out figures, lol. It was actually the worst of the bunch. Awesome for home theatre though, but the UI on the Harman dbx products is far superior and those include RTA and signal generators built in, so no having to setup and flipping back/forth to REW for anything except time alignment.
Results so far:
In the end, I settled on the Xkitz- 3-way FET opamp based balanced boards and linear ps, as seen here. Even with external EQ, quieter, more detailed, bigger soundstage, way better bass/mid-bass textures, clearer highs.
I used and old, broken SD dvd player as a donor for the power receptacle and case , with 3d printed front/back panels (still need to tweak the makeshift back) and just purchased a few standoffs to make this very budget friendly. This was a drastic improvement over the DSP xovers and quieter and more detailed than the pro analog crossovers. Completely silent, might be the slightest combo of buzz/hiss if it's very, very quiet and I wedge my head into the CD horn. Has never been this quiet. Balanced cables helped a lot here, but these xovers really cleaned it up.
So far I've been able to accomplish the following.
-crossover points
-levels
-eq (using a Kenwood GE-7030, absolutely fantastic!), with the exception of a dip at the mid/hi crossover
-phase adjustment to eradicate nulls and reduce dip at 1500hz xover point (changed wires at mid/tweeter as required using REW, sub has selectable phase 0, 90, 180, 270)
compromise...
-Only LR4 24db slopes available on the diy crossover.. I knew that and trying to be budget friendly and can't justify the Marchand crossover expense, particularly when this one gets me 90% there)
What remains is
1. CD equalization - the EQs 12db range handled the rest of the frequency response, but doesn't have enough differential to raise this dip enough.
2. Time alignment (not too worried about this as the mid/hi are under 2ms, and the sub is about 18ms, but moveable and a lot of is ringing from the 22' ceiling and gobs of windows)
So the questions:
I have a dip at the 1500 Hz region (and 10 to 20Hz needed to be lifted, the EQ took care of that) that I want to take care of.
1. Can I alter the circuit by inserting a capacitor to adjust for the CD horn, as is done in passive networks? (passive example here: https://techtalk.parts-express.com/forum/tech-talk-forum/1410233-cd-horn-passive-equalization)
2. Would it be better to add a circuit? (plenty of room in the case to add a circuit board... something like this? .... https://sound-au.com/project173.htm, but this seems to have minimal impact on the 1.5K region. Also wondering, that even so, might the overall flattening of the HF range help raise it enough for the external EQ to do the rest)
2. As a 3rd alternative, would I be able to easily alter the circuit to have perhaps an LR2 12db or LR3 18 slope for the tweeter low pass?
3. Should I forget about trying to make this work? I hate to do this as I'm almost there and like the price! Of course, if there's other similarly priced active crossover diy boards, I'm still in the 45 day trial period.
Trying to keep it budget friendly/diy friendly. Retired, so I have time.
Glad to post REW or Driverack analysis screenshots, if it helps.
Other info for context:
- Asked maker of xkitz, if about this several days ago, and no response yet. Also asked about timing circuits and looking at Rod Elliots pages, but it's a bit above my mental pay grade.
- Voltage readings of outputs at crossover in volts AC are.100 sub, .035 mid, .006 tweeter to get these leveled when fed sine waves for peak frequencies for each output (I believe it was 40hz, 800hz and ~4khz)
- Crossed over at 80Hz for the mid-woofer/subwoofer (Mid: 200wpc rms amp with 150mV sensitivity, Sub: 1200wpc rms Velodyne, unknown input sensitivity, but leveling puts this about 3x the AC output voltage of the crossover LF outputs)
- Crossed at 1500Hz for the midwoofer/horn (Horn: 70wpc rms, 150mV sensitivity)
- Crossovers tried: digital- driverack PA2, Venu360 (locally available at guitar center!), MiniDSP flex 8 (unbalanced) Analog: BSS360, Rane AC23B, Ashly XR2001, xkitz - DIY
- Subwoofer has it's own equalization and phase adjustments.
- crossover circuit below:
Thank you for any and all advice, suggestions or comments!
passive radiator sim (hornresp)
- By Booger weldz
- Subwoofers
- 4 Replies
rookie attempt atchecking this sectiion of HR out..... so far i have a strange looking overdamped super low tuned ....yuck. (I’m sure I’ve screwed up something (s))
it’s the 7” dayton epique driver and ‘one’ of the matching radiators at this point .
I need to add another. shoukd I add another Sd and Mmd in the PR section to perk up the low end extension section?
Modification of Triode Dicks Topcap PP amplifier. "Mono Bills Topcap"
- By jayrodge
- Tubes / Valves
- 13 Replies
I'm putting this out there as my next project. During my covid layoff I built Triode Dicks Topcap 807PP amp from parts off ebay. They turned out pretty darn good for my first ever build. Fun project. I found out though that, for my DACs output, the 6SL7 had too high mu and not enough swing at the OP. 6SN7s fixed that but the OP is really low. The 807 THD was also high. So, looking at Triode Dicks later projects, I decided that I could adopt what he learned over time and make some big improvements to the early 807 design. The attachment is a pdf of what I'm about to do. I decided late to make a PCB for the CCS and Neg Bias circuits so while I'm waiting I thought I'd share this and have others poke holes in it so I end up with the best mods possible. What do you think of Mono Bills Topcap?? Please let me know!
Nakamichi 1000ZXL Playback and Recording chains isolated schematics + CR7A/Dragon playback diff
- By edgarsls
- Analogue Source
- 0 Replies
The Nakamichi 1000ZXL is the best cassette deck for recording, based on specs.
For playback the CR7A is regarded as the most accurate one.
Here's the playback and recording chains isolated from the messy schematic.
The deck isn't very new so there's no Dolby C, S, SR. No HX Pro or DBX.
It could've likely been upgraded with HX Pro capability since the recording bias level is already set by the CPU. There's an A/D converter. So all the circuitry is there that would allow HX Pro to be implemented in software.
Today you probably could drive the recording heads directly from a DAC and avoid all the analog complexity
The RAMM code is an audio signal encoded on the cassette that stores the calibration data when recorded on this deck so you dont need to run the auto-cal procedure.
For the CR7A/Dragon schematic; The schematic is from the CR7A, the text in red is the differences in the Dragon.
as you can see it's nearly the exact same circuit. Biggest differences probably are the 4066 switch which are problematic and add noise. And the change in the feedback circuit resistor on the first opamp from 280k to 316k.
For playback the CR7A is regarded as the most accurate one.
Here's the playback and recording chains isolated from the messy schematic.
The deck isn't very new so there's no Dolby C, S, SR. No HX Pro or DBX.
It could've likely been upgraded with HX Pro capability since the recording bias level is already set by the CPU. There's an A/D converter. So all the circuitry is there that would allow HX Pro to be implemented in software.
Today you probably could drive the recording heads directly from a DAC and avoid all the analog complexity
The RAMM code is an audio signal encoded on the cassette that stores the calibration data when recorded on this deck so you dont need to run the auto-cal procedure.
For the CR7A/Dragon schematic; The schematic is from the CR7A, the text in red is the differences in the Dragon.
as you can see it's nearly the exact same circuit. Biggest differences probably are the 4066 switch which are problematic and add noise. And the change in the feedback circuit resistor on the first opamp from 280k to 316k.
8uF Gudeman Hermetically Sealed Cap
Picture is a Gudeman 8uF 115Vac 200VDC capacitor that I got (QTY 8) surplus in, I think, the 1980s.
I think that I also have some Spragues that look exactly the same but maybe .5" shorter, same value.
115 Vac does this suggest a motor start or run cap?
I might use them in a crossover network if I could confirm that they're film.
Does anyone know if these are paper in oil, or film, they measure like a film in terms of value is
very close within 2% and DF is the same as a film.
Normally I'd throw these away but they look so darn high quality, thought I'd ask first.
The end terminals come out through what looks like glass.
Markings are value as above, and:
GUDEMAN
F54-019JT-10
72928-7101
Date code looks like 1971 first week. The picture brings out the surface imperfections, it is actually a nice shiny silver:
I think that I also have some Spragues that look exactly the same but maybe .5" shorter, same value.
115 Vac does this suggest a motor start or run cap?
I might use them in a crossover network if I could confirm that they're film.
Does anyone know if these are paper in oil, or film, they measure like a film in terms of value is
very close within 2% and DF is the same as a film.
Normally I'd throw these away but they look so darn high quality, thought I'd ask first.
The end terminals come out through what looks like glass.
Markings are value as above, and:
GUDEMAN
F54-019JT-10
72928-7101
Date code looks like 1971 first week. The picture brings out the surface imperfections, it is actually a nice shiny silver:
Advice for Developing a Tube Preamp with Warm Sound
- Tubes / Valves
- 22 Replies
Hello everyone,
I’m new to audio equipment development and would appreciate some recommendations and ideas. I’m currently working on a tube preamp project with the goal of achieving the warm sound characteristic of these devices. I’m implementing a +17V and -17V power supply, as well as a 48V phantom power.
I plan to use this voltage phatom power to bias a tube stage that I want to amplify to 60dB (equivalent to 1000 times gain). I’ve looked into the 12AX7, but I’ve heard it can present distortion at low voltages. I’m currently testing the 12AU7, which is said to have less noise, although its amplification factor is much lower.
I’m wondering how I could achieve the desired gain to get that "warm" sound. I was considering a dual amplification stage using the 12AU7 along with a FET. What recommendations would you have for tackling this issue?
Sorry if my question seems basic; I’m new to this. Thanks.
Here is a basic dual-stage preamp with a gain of 15 per stage, along with some curves I obtained from the 12AU7 in Multisim.
I’m new to audio equipment development and would appreciate some recommendations and ideas. I’m currently working on a tube preamp project with the goal of achieving the warm sound characteristic of these devices. I’m implementing a +17V and -17V power supply, as well as a 48V phantom power.
I plan to use this voltage phatom power to bias a tube stage that I want to amplify to 60dB (equivalent to 1000 times gain). I’ve looked into the 12AX7, but I’ve heard it can present distortion at low voltages. I’m currently testing the 12AU7, which is said to have less noise, although its amplification factor is much lower.
I’m wondering how I could achieve the desired gain to get that "warm" sound. I was considering a dual amplification stage using the 12AU7 along with a FET. What recommendations would you have for tackling this issue?
Sorry if my question seems basic; I’m new to this. Thanks.
Here is a basic dual-stage preamp with a gain of 15 per stage, along with some curves I obtained from the 12AU7 in Multisim.
Monarchy Audio Digital Interface Processer 24/96 aka DIP fault issue
- By JKiriakis
- Digital Line Level
- 9 Replies
The 'Lock' LED on the front panel blinks intermittently and coincident with that the signal fails. Waiting awhile, sometimes the led and signal will return, then fail again.
Any ideas on how to fix, what might be the root cause? CC Poon the proprietor is retired, but the company was formerly in South San Francisco. I happen to live nearby so could certainly deliver to any former tech available locally.
Thanks for your insights in advance. No schematic available that I am able to access.
Any ideas on how to fix, what might be the root cause? CC Poon the proprietor is retired, but the company was formerly in South San Francisco. I happen to live nearby so could certainly deliver to any former tech available locally.
Thanks for your insights in advance. No schematic available that I am able to access.
Change power supply voltage for Audible Illusions 3A preamp
- By theanhngn
- Power Supplies
- 19 Replies
Hi all,
I got an Audible Illusions 3A with 120v power supply version. Now i would like to change to 220V power supply.
Does any of you have done it before or does any of you have the its PSU schematic diagram?
Thanks and regards,
I got an Audible Illusions 3A with 120v power supply version. Now i would like to change to 220V power supply.
Does any of you have done it before or does any of you have the its PSU schematic diagram?
Thanks and regards,
APT Holman Preamps and relay repair
- By Marksd
- Analog Line Level
- 79 Replies
Attachments
-
APT Holman Preamp Factory modified relay installed under the mod 1 factory repair1.JPG -
APT Holman Preamp Factory modified relay installed under the mod 1 factory repair2.JPG -
APT Holman Preamp Factory modified relay installed under the mod 1 factory repair3.JPG -
APT Holman Preamp Factory modified relay installed under the mod 1 factory repair4.JPG -
APT Holman Preamp Factory modified relay installed under the mod 1 factory repair5.JPG -
APT Holman Preamp Factory modified relay installed under the mod 1 factory repair6.JPG -
APT Holman Preamp Factory modified relay installed under the mod 1 factory repair7.JPG -
APT Holman Preamp with standard relay1.JPG -
APT Holman Preamp with standard relay2.JPG -
APT Holman Preamp with standard relay3.JPG
Bliesma T34B-4 on Waveguide No.2439
- By Joseph Crowe
- Multi-Way
- 0 Replies
Introduction
In this post I test a new waveguide design that I came up with for the Bliesma T34B-4 Beryllium dome tweeter. The specification page for the T34B-4 can be found here. The tweeter retails for around $500 USD. I directly compare this new waveguide/tweeter combo against the RCF CD350 compression driver with the ES-800 Biradial. I chose this comparison because the RCF/ES-800 combo represents a pretty standard setup, and a good all-around performer. The RCF retails for $90 USD, and so there "should" be a considerable step up in performance with the Bliesma.
Waveguide Design Elements
I wanted a waveguide that was based on my ES-Biradial horn flare geometry so that I could control vertical and horizontal directivity independently. The biradial shape also provides closer driver spacing to the low frequency driver.
The horn is 3D printed in two components. The first component is the throat piece which closely follows the shape of surround. This requires high precision printing which is why this is a separate piece. The second piece is the horn mouth which is printed in a different axis than the first piece.
Measurements
I began by measuring the raw 2439 frequency response at 1m (see red below). I then installed a capacitor to flatten the response for further testing. I found that a 6.8uF capacitor was ideal (see red response below). The raw impedance sweep is also shown as an overlay in black.
I then conducted the same response with the RCF CD350/ES-800 Biradial combo. A 3.3uF capacitor was enough to provide a reasonably flat response.
I then measured the horizontal and vertical directivity of the T34B/2439 combo.
Horizontal coverage is characterized as a consistent and wide coverage of 120 degrees however it narrows somewhat starting at 8kHz where we see directivity at 70 degrees for the 11kHz region. Pattern control is maintained as low as 1.2kHz.
The vertical directivity is characterized as a gradual narrowing starting at 2kHz narrowing to 40 degrees at 11kHz. Overall behavior is devoid of any diffraction or spurious resonances.
I then measured the RFC CD350/ES-800 Biradial combo which provides constant directivity across the operating range. Coverage starts out at 120 degrees at 1kHz and ends at 100 degrees for the 11kHz region.
The vertical is characterized as a gradual narrowing starting at 2kHz and ending at 60 degrees coverage for the 11kHz region.
Generally we see a more optimal polar map for the ES-800 biradial. One factor contributing to this is the larger throat of the bliesma at 1.33" versus 1".
Time Domain
I then measured the Burst Decay for the Bliesma. We see a perfectly clean result within the audible band.
The CSD plot also is very commendable.
I then measured the RCF CD350/ES-800 Biradial combo. The burst decay shows some ringing in the 10kHz region out to 15 periods. Personally I like to see a clean burst decay to within 18 periods. So this is acceptable.
The CSD plot is shown as well for reference.
Distortion
I then looked at distortion using a 95dB test signal level referenced at 1m. The mic was placed at 20cm from the horn mouth. H3 and H4 measure at around 0.008% for the 2kHz region.
If we change the vertical scale to dB instead of percent, this translates into -82dB.
I then measured the RCF CD350/ES-800 Biradial combo under the same conditions and achieved 0.007% for H3 and 0.014% for H4.
Again, changing the vertical scale to dB results in H3 at -83dB and H4 at -77dB.
Conclusion on Harmonic
Both the RCF CD350 and T34B have similar harmonic distortion profiles for H3 and H4.
Intermodulation Distortion using Multiband Multitone
I then measured the IMD of the Bliesma at the same 95dB test SPL that we used for the Harmonic test. IMD is at -60dB for the 1.5kHz region increasing to -70dB for the 5kHz region. Upper treble is around -65dB.
I then tested the RCF CD350/ES-800 Biradial combo. We -64dB for the 1.5kHz region -61dB for the 5kHz region and -55dB for the upper treble.
The Bliesma seems to outperform the RCF marginally on the IMD by about +5 to +8dB dB. This may be more academic since both results are far below the threshold of audibility.
Gedlee Distortion (Gm)
Below is direct comparion using Gedlee Distortion for the 95dB test SPL. I'm using Virtins Multi-Instrument Software to conduct this test. The RCF is shown in red with the Bliesma in Blue.
Conclusions
The RCF CD350/ES-800 Biradial combo is a tough act to follow but the Bliesma pulls it off by offering a smoother response, lower distortion, and a cleaner time domain. However the larger diameter of the Bliesa introduces some narrowing of directivity in the critical upper treble where soundstage width could be compromised. The Bliesma has no trouble digging down deep into the lower midrange without sounding stressed or resonate, which is often the case when mounting a dome tweeter in a waveguide.
My subjective evaluation was brief and conducted in mono. Overall I would say that the Bliesma sounded excellent, with just a slight lack of upper treble energy compared to the RCF. Directly comparing the two revealed that the compression driver is more expressive and colorful, highlighting dynamic contrast more readily. Both had similar levels of clarity through the midrange and treble.
In this post I test a new waveguide design that I came up with for the Bliesma T34B-4 Beryllium dome tweeter. The specification page for the T34B-4 can be found here. The tweeter retails for around $500 USD. I directly compare this new waveguide/tweeter combo against the RCF CD350 compression driver with the ES-800 Biradial. I chose this comparison because the RCF/ES-800 combo represents a pretty standard setup, and a good all-around performer. The RCF retails for $90 USD, and so there "should" be a considerable step up in performance with the Bliesma.
Waveguide Design Elements
I wanted a waveguide that was based on my ES-Biradial horn flare geometry so that I could control vertical and horizontal directivity independently. The biradial shape also provides closer driver spacing to the low frequency driver.
The horn is 3D printed in two components. The first component is the throat piece which closely follows the shape of surround. This requires high precision printing which is why this is a separate piece. The second piece is the horn mouth which is printed in a different axis than the first piece.
Measurements
I began by measuring the raw 2439 frequency response at 1m (see red below). I then installed a capacitor to flatten the response for further testing. I found that a 6.8uF capacitor was ideal (see red response below). The raw impedance sweep is also shown as an overlay in black.
I then conducted the same response with the RCF CD350/ES-800 Biradial combo. A 3.3uF capacitor was enough to provide a reasonably flat response.
I then measured the horizontal and vertical directivity of the T34B/2439 combo.
Horizontal coverage is characterized as a consistent and wide coverage of 120 degrees however it narrows somewhat starting at 8kHz where we see directivity at 70 degrees for the 11kHz region. Pattern control is maintained as low as 1.2kHz.
The vertical directivity is characterized as a gradual narrowing starting at 2kHz narrowing to 40 degrees at 11kHz. Overall behavior is devoid of any diffraction or spurious resonances.
I then measured the RFC CD350/ES-800 Biradial combo which provides constant directivity across the operating range. Coverage starts out at 120 degrees at 1kHz and ends at 100 degrees for the 11kHz region.
The vertical is characterized as a gradual narrowing starting at 2kHz and ending at 60 degrees coverage for the 11kHz region.
Generally we see a more optimal polar map for the ES-800 biradial. One factor contributing to this is the larger throat of the bliesma at 1.33" versus 1".
Time Domain
I then measured the Burst Decay for the Bliesma. We see a perfectly clean result within the audible band.
The CSD plot also is very commendable.
I then measured the RCF CD350/ES-800 Biradial combo. The burst decay shows some ringing in the 10kHz region out to 15 periods. Personally I like to see a clean burst decay to within 18 periods. So this is acceptable.
The CSD plot is shown as well for reference.
Distortion
I then looked at distortion using a 95dB test signal level referenced at 1m. The mic was placed at 20cm from the horn mouth. H3 and H4 measure at around 0.008% for the 2kHz region.
If we change the vertical scale to dB instead of percent, this translates into -82dB.
I then measured the RCF CD350/ES-800 Biradial combo under the same conditions and achieved 0.007% for H3 and 0.014% for H4.
Again, changing the vertical scale to dB results in H3 at -83dB and H4 at -77dB.
Conclusion on Harmonic
Both the RCF CD350 and T34B have similar harmonic distortion profiles for H3 and H4.
Intermodulation Distortion using Multiband Multitone
I then measured the IMD of the Bliesma at the same 95dB test SPL that we used for the Harmonic test. IMD is at -60dB for the 1.5kHz region increasing to -70dB for the 5kHz region. Upper treble is around -65dB.
I then tested the RCF CD350/ES-800 Biradial combo. We -64dB for the 1.5kHz region -61dB for the 5kHz region and -55dB for the upper treble.
The Bliesma seems to outperform the RCF marginally on the IMD by about +5 to +8dB dB. This may be more academic since both results are far below the threshold of audibility.
Gedlee Distortion (Gm)
Below is direct comparion using Gedlee Distortion for the 95dB test SPL. I'm using Virtins Multi-Instrument Software to conduct this test. The RCF is shown in red with the Bliesma in Blue.
Conclusions
The RCF CD350/ES-800 Biradial combo is a tough act to follow but the Bliesma pulls it off by offering a smoother response, lower distortion, and a cleaner time domain. However the larger diameter of the Bliesa introduces some narrowing of directivity in the critical upper treble where soundstage width could be compromised. The Bliesma has no trouble digging down deep into the lower midrange without sounding stressed or resonate, which is often the case when mounting a dome tweeter in a waveguide.
My subjective evaluation was brief and conducted in mono. Overall I would say that the Bliesma sounded excellent, with just a slight lack of upper treble energy compared to the RCF. Directly comparing the two revealed that the compression driver is more expressive and colorful, highlighting dynamic contrast more readily. Both had similar levels of clarity through the midrange and treble.
how to get +/- supply from 2 SMPS?
- By mrjayviper
- Power Supplies
- 7 Replies
is it as simple as connecting -V of 1st unit to the +v the 2nd unit to get ground? Thanks
pic for reference
pic for reference
Working on Technics SA1000, wanting to replace TA7136P opamps
- By saabracer23
- Solid State
- 7 Replies
This receiver is fairly highly regarded, it was the largest to come out of the receiver wars of the 70s, making 330w per channel. Pretty outstanding if it isn’t the prettiest girl at the dance.
It uses 6 TA7136P seven pin single opamps. Now this opamp isn’t highly regarded, not even slightly. It’s known to be a pretty poor performer. I see that there are a couple of people that sell adapter boards to where you can attach your own opamp, but at $5-$7 per adapter board I’m going to look at other options. I am currently trying to get accustomed to Kicad and EasyEDA so that I can make up my own adapter boards and order 50 for less than the cost of the 6 I would have to buy (I will share the gerber when I do).
One board is supposed to be used with the OPA604, but it’s generic pinout where they just attach the appropriate pins, V+ to V+, output to output, etc. So this board can be used with any single opamp with the same pinout like the TL071. Speaking of, I’ve also the TA7136 repalced with a TL071.
The other board that is available can be used with single opamps like the OPA604 and NE5534AP, but can also be used with dual opamps like the TL072, NE5532AP. With the dual opamps you’re just supposed to wire the two opamps in parallel, so parallel the outputs, + and - inputs and such. I still don’t know what the advantage of using a dual opamp is over a single opamp. Is wiring a dual opamp in parallel give better performance?
Until I get well enough acquainted with Kicad or EasyEDA I’m thinking I’ll just use these small adapter boards I have on hand and wire leads that go to the appropriate spots on the TA7136 pinout.
I can use standard 8 pin opamps or I could use a smaller SOIC-8 package. For standard packages I have TL072CP, NE5532AP and then the single NE5534AP. For the smaller packages I have some OPA1656IDR and then for a single option I have OPA1641AID.
Im thinking for simplicity I’ll use of the single opamp options, unless someone has a reason as to why I should go dual.
It should fairly simple, joining the correct pins.
Just omit pins 1 and 5 and remove the components on the board at 1 and 5 as stated in the photo. Also attach a 100 nF from pins 4 to pin 7, the supplies.
So my questions, any reason to use a dual opamp over a single? And do all of my choices look like they would work well? That being NE5524 and OPA1641 for the singles and NE5532, OPA1656, or TL072 for the duals?
Appreciate any and all thoughts on this.
Dan
It uses 6 TA7136P seven pin single opamps. Now this opamp isn’t highly regarded, not even slightly. It’s known to be a pretty poor performer. I see that there are a couple of people that sell adapter boards to where you can attach your own opamp, but at $5-$7 per adapter board I’m going to look at other options. I am currently trying to get accustomed to Kicad and EasyEDA so that I can make up my own adapter boards and order 50 for less than the cost of the 6 I would have to buy (I will share the gerber when I do).
One board is supposed to be used with the OPA604, but it’s generic pinout where they just attach the appropriate pins, V+ to V+, output to output, etc. So this board can be used with any single opamp with the same pinout like the TL071. Speaking of, I’ve also the TA7136 repalced with a TL071.
The other board that is available can be used with single opamps like the OPA604 and NE5534AP, but can also be used with dual opamps like the TL072, NE5532AP. With the dual opamps you’re just supposed to wire the two opamps in parallel, so parallel the outputs, + and - inputs and such. I still don’t know what the advantage of using a dual opamp is over a single opamp. Is wiring a dual opamp in parallel give better performance?
Until I get well enough acquainted with Kicad or EasyEDA I’m thinking I’ll just use these small adapter boards I have on hand and wire leads that go to the appropriate spots on the TA7136 pinout.
I can use standard 8 pin opamps or I could use a smaller SOIC-8 package. For standard packages I have TL072CP, NE5532AP and then the single NE5534AP. For the smaller packages I have some OPA1656IDR and then for a single option I have OPA1641AID.
Im thinking for simplicity I’ll use of the single opamp options, unless someone has a reason as to why I should go dual.
It should fairly simple, joining the correct pins.
Just omit pins 1 and 5 and remove the components on the board at 1 and 5 as stated in the photo. Also attach a 100 nF from pins 4 to pin 7, the supplies.
So my questions, any reason to use a dual opamp over a single? And do all of my choices look like they would work well? That being NE5524 and OPA1641 for the singles and NE5532, OPA1656, or TL072 for the duals?
Appreciate any and all thoughts on this.
Dan
For Sale Lii Song F-12 Full Range Drivers
$200 plus shipping to wherever you are.
1 pair of Lii Audio F-12 full range drivers. Used briefly in a temporary baffle for approximately 100 hours. They sounded great, but were too big for my listening space. These will ship in the original packaging. These are also posted on US Audiomart.
1 pair of Lii Audio F-12 full range drivers. Used briefly in a temporary baffle for approximately 100 hours. They sounded great, but were too big for my listening space. These will ship in the original packaging. These are also posted on US Audiomart.
CEC DX51 DAC, R channel pops
- By penmarker
- Digital Line Level
- 7 Replies
Hi everyone hope everyone is doing great.
I acquired this DAC and looking for a popping solution.
The right channel will pop whenever audio starts playing. It can be replicated by either play/pause or turning volume up from 0.
There are a pair of CHAMP Modules (pictured, horizontal with many LEDs) which I assumed was the cause. After swapping the cards the popping did not swap so it could be up stream.
The three boards (vertical) are Decoder board, Digital Filter board, and DAC board.
The dac uses 4 PCM1704 chips.
No service manuals are available and information is sparse.
Aside from the popping and slightly higher noise floor compared to modern DACs, it sounds great. I use it for my tv as well so the pops between quiet passages doesn’t sound great.
Ps; this dac has a quirk of never turning off. When you press the front power button everything stays lit but the input/output relays are disengaged. I assume they’re subscribed to the idea of leaving your equipment turned on makes it sound better.
I acquired this DAC and looking for a popping solution.
The right channel will pop whenever audio starts playing. It can be replicated by either play/pause or turning volume up from 0.
There are a pair of CHAMP Modules (pictured, horizontal with many LEDs) which I assumed was the cause. After swapping the cards the popping did not swap so it could be up stream.
The three boards (vertical) are Decoder board, Digital Filter board, and DAC board.
The dac uses 4 PCM1704 chips.
No service manuals are available and information is sparse.
Aside from the popping and slightly higher noise floor compared to modern DACs, it sounds great. I use it for my tv as well so the pops between quiet passages doesn’t sound great.
Ps; this dac has a quirk of never turning off. When you press the front power button everything stays lit but the input/output relays are disengaged. I assume they’re subscribed to the idea of leaving your equipment turned on makes it sound better.
Attachments
Guitar speaker cab series/parallel switching
- Multi-Way
- 9 Replies
I have two single extension cabs with a 10" 8ohm speaker each. Would the following doodled diagram work if I wanted to run one into the other as either series or parallel? I have amps that are 4 or 8 ohms only as well as a few that are selectable between 4,8, and 16 ohms so it would be nice to have it as versatile as possible. So the three jacks (in, series out and parallel out) on one cab and just a single jack on the second cab.
Edit:replaced crappy doodle. 🙄
Edit:replaced crappy doodle. 🙄
