I would guess that putting the scope on the reset line adds capacitance and slows down the reset.
Dale
Dale
Dale,
You're some kinda genius!
Lifting pin 1 of micro and connecting it to +5V on terminal block via 10k works. The board seems fine and IR1 now changes volume (just on the bench; it is not connected to stereo system).
What happened?
Thanks, Robert
You're some kinda genius!
Lifting pin 1 of micro and connecting it to +5V on terminal block via 10k works. The board seems fine and IR1 now changes volume (just on the bench; it is not connected to stereo system).
What happened?
Thanks, Robert
Now that the board is working, I plugged in Vinnie's code for the Philips (1423) in my 5-In-One RS ($15 ?) remote. Thanks, Vinnie, this works perfectly.
It doesn't interfer with the Sony (which I turned off in the IR1) and it also doesn't interfer with our Philips 963SA SACD player, which uses another Philips code.
Very nice.
It doesn't interfer with the Sony (which I turned off in the IR1) and it also doesn't interfer with our Philips 963SA SACD player, which uses another Philips code.
Very nice.
Robert,
I just stayed at that hotel last night 🙄
Actually, I did not think it would solve the problem. I just wanted to see the frequency of the resets. All that I can conclude is that the capacitor on the reset circuit was damaged. You could try taking cap C10 out (leave it out) and see if the board still works with pin one back in the socket.
Dale
P.S. Thanks for your help debugging.
I just stayed at that hotel last night 🙄
Actually, I did not think it would solve the problem. I just wanted to see the frequency of the resets. All that I can conclude is that the capacitor on the reset circuit was damaged. You could try taking cap C10 out (leave it out) and see if the board still works with pin one back in the socket.
Dale
P.S. Thanks for your help debugging.
rljones said:
No problem! 😎
I looked in the back of my remote and typed in the radio shack model number on the websight search to find the documentation for it (programming instructions, codes, etc.). Glad is works.
I also disabled the Sony and am now using the Philips, and it doesn't interfere with my TV. Works great! Turns on/off my amp via remote! Gotta love being a lazy-*** 🙂
Dale, those APOX-SHM boards look NICE! I may break down and get them to replace my apox-2. Let me know how the testing goes.
I have to take some pictures today of my setup. I'll post the link later today.
-Vinnie
link to pics
Okay,
Here is my AKSA 55 amp with an APOX IR-1 and APOX-2 integrated into it. Obviously, the front panel is still
under contruction and needs to have the LCD and IR
sensor mounted in the panel, along with an LED or two
to indicate PWR/Standby. The knobs are el-cheapo rat
shack ones, but serve their purpose for now.
http://photos.yahoo.com/vinnie822
I put the APOX-2 in "pseudo-shunt-mode" by connecting the
INs and OUTs together, and then having the signal
coming in pass through a 6.2K resistor before connecting
to the input terminal on the Apox-2 (which is shorted to
the output terminal). In this fashion, the signal "technically"
doesn't pass through any relays at all, only the shunt to
gnd goes through one relay per channel. In this mode, the
sound is a few steps better than my DACT CT-2. Without this
tweak, the sound of the apox-2 still bests my DACT, especially
in bass extension.
As I mentioned in a previous post, I added a MOSFET to the standby terminals on the IR-1 board in order to have enough current to drive the 5V, 27ohm relay coils that turn on each of the power supplies to the AKSA 55. These are 250VAC, 30A Potter and Brumsfield relays.
I may go obsessive-complusive and order two APOX-SHM boards to replace the APOX-2 board. I wonder if I will gain any improvements?
These are just preliminary pics, as this is still a work in progress.
-Vinnie
Okay,
Here is my AKSA 55 amp with an APOX IR-1 and APOX-2 integrated into it. Obviously, the front panel is still
under contruction and needs to have the LCD and IR
sensor mounted in the panel, along with an LED or two
to indicate PWR/Standby. The knobs are el-cheapo rat
shack ones, but serve their purpose for now.
http://photos.yahoo.com/vinnie822
I put the APOX-2 in "pseudo-shunt-mode" by connecting the
INs and OUTs together, and then having the signal
coming in pass through a 6.2K resistor before connecting
to the input terminal on the Apox-2 (which is shorted to
the output terminal). In this fashion, the signal "technically"
doesn't pass through any relays at all, only the shunt to
gnd goes through one relay per channel. In this mode, the
sound is a few steps better than my DACT CT-2. Without this
tweak, the sound of the apox-2 still bests my DACT, especially
in bass extension.
As I mentioned in a previous post, I added a MOSFET to the standby terminals on the IR-1 board in order to have enough current to drive the 5V, 27ohm relay coils that turn on each of the power supplies to the AKSA 55. These are 250VAC, 30A Potter and Brumsfield relays.
I may go obsessive-complusive and order two APOX-SHM boards to replace the APOX-2 board. I wonder if I will gain any improvements?
These are just preliminary pics, as this is still a work in progress.
-Vinnie
APOX-SHM is alive
Well, after a trying afternnon, the APOX-SHM is now in my system and working quite nicely.
So far, I have only tested functionality such as:
1) Switching noise - none
2) Levels - works as advertised
3) Mute - mutes
I still need to evaluate hum (none noticed) and SE mode.
Now, from first impression, it seems to have better channel separation as measured by imaging. This may be due to my use of the Holco 0.1% resistors instead of BC 1%.
Best Regards,
Dale
P.S. Petter, looks like we have a winner!
Well, after a trying afternnon, the APOX-SHM is now in my system and working quite nicely.
So far, I have only tested functionality such as:
1) Switching noise - none
2) Levels - works as advertised
3) Mute - mutes
I still need to evaluate hum (none noticed) and SE mode.
Now, from first impression, it seems to have better channel separation as measured by imaging. This may be due to my use of the Holco 0.1% resistors instead of BC 1%.
Best Regards,
Dale
P.S. Petter, looks like we have a winner!
This is indeed GREAT news. I continue to be impressed with your work. Please do keep us updated on the most important thing: Sound quality.
One note: I don't seem to find Resistor set pricing for the SHM's. Is that not going to be offered (I am not sure whether to go for sets or get my own).
Petter
One note: I don't seem to find Resistor set pricing for the SHM's. Is that not going to be offered (I am not sure whether to go for sets or get my own).
Petter
Cobra2,
Once I get this stuff in and built up (+ my stereo back on track), you (and other takers who get near Oslo, Norway) can contact me if you need to verify product before a potential purchase.
Petter
Once I get this stuff in and built up (+ my stereo back on track), you (and other takers who get near Oslo, Norway) can contact me if you need to verify product before a potential purchase.
Petter
So what's the story with the hum on the Apox2? I lost track...
efficient communication has been a problem recently, to put it mildly.
Are there new relays for the APOX-SHM?
efficient communication has been a problem recently, to put it mildly.
Are there new relays for the APOX-SHM?
In the interest of speed, I will jump in here with (perhaps) incomplete information:
The short on Apox hum: All complaints in this board have been resolved from what I have been able to notice from my daily monitoring of this board 🙂 I believe Craig has worked extensively to resolve such issues. It would perhaps be opportune to post compiled findings under FAQ or in manuals under "if you should have this problem -- these are the recommended approaches" etc. Anyway, it was pretty quick for me to check out the last few pages to find info on this.
Relays for SHM: Yes, these are different. The other boards use stereo relays. For obvious reasons, the mono board uses mono relays which are MUCH easier to work with in terms of PCB layout.
Petter
The short on Apox hum: All complaints in this board have been resolved from what I have been able to notice from my daily monitoring of this board 🙂 I believe Craig has worked extensively to resolve such issues. It would perhaps be opportune to post compiled findings under FAQ or in manuals under "if you should have this problem -- these are the recommended approaches" etc. Anyway, it was pretty quick for me to check out the last few pages to find info on this.
Relays for SHM: Yes, these are different. The other boards use stereo relays. For obvious reasons, the mono board uses mono relays which are MUCH easier to work with in terms of PCB layout.
Petter
Hi Everyone,
In case anyone is trying to reach Craig, he is on vacation and will not be back until Wednesday.
For aprox. current requirements:
IR1 - 200mA
IS1 - 100mA each
APOX-2 - 75 mA
APOX-1 - 150mA
APOX-SHM - 150mA
Dale
In case anyone is trying to reach Craig, he is on vacation and will not be back until Wednesday.
For aprox. current requirements:
IR1 - 200mA
IS1 - 100mA each
APOX-2 - 75 mA
APOX-1 - 150mA
APOX-SHM - 150mA
Dale
manual updates
The APOX-2, APOX-IS1, and APOX-IR1 assembly instructions
have all been updated with a little more info added, and a few mistakes have been fixed. (Such as LED orientation)
-Craig
Apox Web Page
The APOX-2, APOX-IS1, and APOX-IR1 assembly instructions
have all been updated with a little more info added, and a few mistakes have been fixed. (Such as LED orientation)
-Craig
Apox Web Page
Reality check
Could people answer the following questions:
1) Who has completed at least one board?
2) What were the major issues? (if you haven;t already posted them)
3) Have you actually fired it up? Is it working?
4) How does it sound (compared to ?)
Just curious...
Dale
Could people answer the following questions:
1) Who has completed at least one board?
2) What were the major issues? (if you haven;t already posted them)
3) Have you actually fired it up? Is it working?
4) How does it sound (compared to ?)
Just curious...
Dale
Dale,
I've built five IS1 boards (all work), one IR1 and 2 APOX-2. I've done no more listening tests as I'm trying to organize the layout in the 3 chassis and need to make a template to route an opening in one for the display.
Petter,
I have a question about the SHM resistor sets. First off, my desired preamp scheme is: source -> IS1 -> SHM -> gain stage -> IS1 -> amp. my gain stage has an input impedance of 100k and an output impedance of 50R; the amp I use has an input impedance of 10k.
The spreadsheet that you and Craig posted suggests and plots a 4K or a 10k set of resistors. I'm more tempted to have a 10k set, but the max impedance is a little high at 100K and the min a little low at 3.8k. The suggested 10K set used is R1/12 10k, R10/11 7k, R36/37 3.5k, R32 100k, R2 10k, R3 5k, R4 2.5k, R5 1.25k, R6 625R, R7 313R, R8 156R, and R9 78R.
I'd like to keep a min impedance of 5k to 6k and a max of no more than 30-40k--considering my above scheme. I came up with the following resistor changes that alter R1, R10, R36 and R32 but leaves R2-9 the same. If I understand your spreadsheet, it raises the min input impedance to 5.5k and lowers max input impedance to 35k:
R1/12 15k, R10/11 10k, R36/37 5k, R32 5k, R2 10k, R3 5k, R4 2.5k, R5 1.25k, R6 625R, R7 313R, R8 156R, and R9 78R.
Drawback seems to be that the max attenuation is 58 dB in balanced and 52 dB in single ended. Would this work, or not be a good idea?
Thanks, Robert
I've built five IS1 boards (all work), one IR1 and 2 APOX-2. I've done no more listening tests as I'm trying to organize the layout in the 3 chassis and need to make a template to route an opening in one for the display.
Petter,
I have a question about the SHM resistor sets. First off, my desired preamp scheme is: source -> IS1 -> SHM -> gain stage -> IS1 -> amp. my gain stage has an input impedance of 100k and an output impedance of 50R; the amp I use has an input impedance of 10k.
The spreadsheet that you and Craig posted suggests and plots a 4K or a 10k set of resistors. I'm more tempted to have a 10k set, but the max impedance is a little high at 100K and the min a little low at 3.8k. The suggested 10K set used is R1/12 10k, R10/11 7k, R36/37 3.5k, R32 100k, R2 10k, R3 5k, R4 2.5k, R5 1.25k, R6 625R, R7 313R, R8 156R, and R9 78R.
I'd like to keep a min impedance of 5k to 6k and a max of no more than 30-40k--considering my above scheme. I came up with the following resistor changes that alter R1, R10, R36 and R32 but leaves R2-9 the same. If I understand your spreadsheet, it raises the min input impedance to 5.5k and lowers max input impedance to 35k:
R1/12 15k, R10/11 10k, R36/37 5k, R32 5k, R2 10k, R3 5k, R4 2.5k, R5 1.25k, R6 625R, R7 313R, R8 156R, and R9 78R.
Drawback seems to be that the max attenuation is 58 dB in balanced and 52 dB in single ended. Would this work, or not be a good idea?
Thanks, Robert
Robert,
I fully understand your wish to have a high enough input impedance so that the previous stage (which I don't believe you mentioned the output impedance of) is not loaded too much. At the same time, one wants to transfer as much energy as possible through the volume control into the next stage (i.e a low output impedance of the volume controller and as little "shunting" as possible). These are indeed conflicting goals.
As far as maximum input impedance, I am OK with that also. However, the 100K input impedance of your gain stage is likely the result of a 100K resistor to ground .... which can in many cases be replaced by something bigger if desired.
I have been looking at your suggestions. It would work, there is no problem there. The only thing that worries me a little bit is the volume switching that Craig has been doing to enable more volume levels (ref C++ lookup table) which might lead to sub-optimal progression of levels. Craig is really the best person to talk to about that.
Regarding max/minimum input impedance, bear in mind that not all possible settings are used, and so it can be a little misleading. The highest input impedance will be at the highest value of series resistor + the highest value of shunt resistor -- not a particularly likely scenario (because high value of series resistor means a desire for high attenuation which means low value of shunt resistor). What I think you should be eyeballing is for optimal sound quality near your optimal listening levels (and will I be able to offer any useful advice here -- probably not ... 🙂).
To recap: Pay more attention to the minimum input resistance than the high end. If you need to look at the high end, look for the lowest volume setting at the highest series resistance to compute what it actually is (i.e all series relays open, and the highest volume setting before Craig takes you to the next range).
An alternative to your scheme is to ask Craig (the Excel Genius) for a universal resistor scheme Excel file which can calculate resistor values for ANY desired impedance/starting point (i.e in your case change all resistors) where readers can substitute in the values that are closest from their favourite brands.
I hope this has been helpful. If you need more assistance, please email me privately.
Petter
I fully understand your wish to have a high enough input impedance so that the previous stage (which I don't believe you mentioned the output impedance of) is not loaded too much. At the same time, one wants to transfer as much energy as possible through the volume control into the next stage (i.e a low output impedance of the volume controller and as little "shunting" as possible). These are indeed conflicting goals.
As far as maximum input impedance, I am OK with that also. However, the 100K input impedance of your gain stage is likely the result of a 100K resistor to ground .... which can in many cases be replaced by something bigger if desired.
I have been looking at your suggestions. It would work, there is no problem there. The only thing that worries me a little bit is the volume switching that Craig has been doing to enable more volume levels (ref C++ lookup table) which might lead to sub-optimal progression of levels. Craig is really the best person to talk to about that.
Regarding max/minimum input impedance, bear in mind that not all possible settings are used, and so it can be a little misleading. The highest input impedance will be at the highest value of series resistor + the highest value of shunt resistor -- not a particularly likely scenario (because high value of series resistor means a desire for high attenuation which means low value of shunt resistor). What I think you should be eyeballing is for optimal sound quality near your optimal listening levels (and will I be able to offer any useful advice here -- probably not ... 🙂).
To recap: Pay more attention to the minimum input resistance than the high end. If you need to look at the high end, look for the lowest volume setting at the highest series resistance to compute what it actually is (i.e all series relays open, and the highest volume setting before Craig takes you to the next range).
An alternative to your scheme is to ask Craig (the Excel Genius) for a universal resistor scheme Excel file which can calculate resistor values for ANY desired impedance/starting point (i.e in your case change all resistors) where readers can substitute in the values that are closest from their favourite brands.
I hope this has been helpful. If you need more assistance, please email me privately.
Petter
Here's an idea that has been swimming in my head for a while. I was going to wait and do something more substantial with the idea, but given the great work that has been done connected to this thread I figured I would throw the idea out there and see what comes of it. Maybe others can iron out the kinks before I start mucking around with it...or maybe it's a stupid idea and not worth trying. Anyway I haven't seen it mentioned before, but maybe it's already been presented.
The upper-level PICs such as the 16F877 have a few analog inputs available--up to 8 channels of 10 bit accuracy. Therefore, each audio channel can be fed into a dual ganged motorized pot, with the other gang going between a regulated voltage and the µC. This provides constant feedback to the PIC regarding the volume level. The PIC changes the volume via motor-control output pins. Up to 8 channels can be volume controlled via the PIC, with balance functionality easily built in via software. No relays to worry about and only one device in the signal (or shunt) path. Purists might even shut off the voltage to the other gang when the volume is not being switched.
Furthermore, the resistance of the "important" gang can be calibrated to the voltage seen by the PIC. This would eliminate the worry about resistance variability. I bet even a linear taper could be used--and thus perhaps better pot for the $ (reminds me of college!)--with the PIC determining the proper settings to end up with logarithmic output via software math or lookup tables.
If not all 8 channels are used, one could use another motorized pot for the volume control interface knob, in place of the encoder. This would allow you to make the knob spin while pressing the remote control-Whoo hah! 😉
So--seems easy; dumb idea?
The upper-level PICs such as the 16F877 have a few analog inputs available--up to 8 channels of 10 bit accuracy. Therefore, each audio channel can be fed into a dual ganged motorized pot, with the other gang going between a regulated voltage and the µC. This provides constant feedback to the PIC regarding the volume level. The PIC changes the volume via motor-control output pins. Up to 8 channels can be volume controlled via the PIC, with balance functionality easily built in via software. No relays to worry about and only one device in the signal (or shunt) path. Purists might even shut off the voltage to the other gang when the volume is not being switched.
Furthermore, the resistance of the "important" gang can be calibrated to the voltage seen by the PIC. This would eliminate the worry about resistance variability. I bet even a linear taper could be used--and thus perhaps better pot for the $ (reminds me of college!)--with the PIC determining the proper settings to end up with logarithmic output via software math or lookup tables.
If not all 8 channels are used, one could use another motorized pot for the volume control interface knob, in place of the encoder. This would allow you to make the knob spin while pressing the remote control-Whoo hah! 😉
So--seems easy; dumb idea?
motorized volume control!
JHertz,
Thats not a dumb idea. In fact someone already makes something like that. (It involved a stepper motor, but I forget the web address)
One difficult part is the mechanical interface from the
motors to the pots and the knobs.
Measuring the current volume level might be a little tricky.
As the signal would have to be sampled over some time window
to determine the current output.
You also would probably have to put the signal through an absolute value circuit to be compatible with the PIC's single supply
ADC.
The other problem is that you will probably need a PGA (programmable gain amplifier) before the absolute value circuit. otherwise the resolution of the DAC will not be high enough at high attenuation.
If you want to design a board we could easily add in support software into the APOX-IR1.
Good Luck,
Craig Beiferman
JHertz,
Thats not a dumb idea. In fact someone already makes something like that. (It involved a stepper motor, but I forget the web address)
One difficult part is the mechanical interface from the
motors to the pots and the knobs.
Measuring the current volume level might be a little tricky.
As the signal would have to be sampled over some time window
to determine the current output.
You also would probably have to put the signal through an absolute value circuit to be compatible with the PIC's single supply
ADC.
The other problem is that you will probably need a PGA (programmable gain amplifier) before the absolute value circuit. otherwise the resolution of the DAC will not be high enough at high attenuation.
If you want to design a board we could easily add in support software into the APOX-IR1.
Good Luck,
Craig Beiferman
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