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First Post Edit:
This thread has resulted in a replacement output PCB to address many of the issues covered in this thread.
Details on the PCB can be found in this post:
http://www.diyaudio.com/forums/digi...improvement-stock-dcx2496-12.html#post4424533
Measurements can be found here:
http://www.diyaudio.com/forums/digi...improvement-stock-dcx2496-13.html#post4424612
Boards can be purchased by signing up here (please read instructions in the spreadsheet):
https://docs.google.com/spreadsheets/d/1bRIlMzzoHpJYvS8kQ_fVN-UgCuzX58TXHaJvsF43iLg/edit?usp=sharing
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Hi All,
Being a long time owner of a DCX2496, I have always wanted to tackle a long list of needed improvements to some of the common issues I have had with this slick little XO.
I finally have a need for this crossover in a more critical application than the usual development role it has found itself in for the past few years, so I've finally got some good motivation to get a board done!
The goal here is to start by summarizing all the things I've noticed while listening to the DCX over the years, and then hopefully quantifying those impressions with measurements.
The measurements should provide the baseline for performance, and help to zero in on the most critical issues with the DCX. I want every change to provide a tangible improvement that can be measured and qualified by listening.
So where do we start? How about a list of subjective things I don't like about the DCX as it is now:
1. Noise - this is one of my biggest issues with the DCX. The outputs suffer from a very high level of output noise which is further exacerbated by the gain in whatever amplifier stage is being used after the DCX. I have also noticed that there is significantly more noise when the input is done via AES/EBU, which is my preferred input type.
2. Output levels - This unit was clearly designed to be used with pro level inputs and outputs. The input voltage required to hit 0dBFS on the ADC is 10VRMS, and the output from the DCX at 0dBFS is a whopping 10VRMS as well! In a normal home HiFi, I think aiming for 2VRMS on both ends is a little more realistic and makes much better use of the 24-bit processing by using more of the available dynamic range.
3. It sounds dull. There is something very wrong with the way the upper mid and high frequencies sound and it's hard to pinpoint exactly what is causing it. I find the issue to be glaringly obvious in some recordings, and not as noticeable in others, but it always seems to be there to some degree. I noticed it most while working on a speaker using a pair of NeoPro5i ribbons which sounded astonishing with a passive crossover, and completely bland and flat when crossed over with this unit.
4. It's fatiguing to listen to. Again, I don't know exactly what causes this, but I find when it's in the signal chain, I'm reaching to turn the volume down more frequently than when it's not in the chain. There's something a bit grating to the way it sounds, and I don't like it. This is the main reason I've been using it primarily for development.
5. It clicks and pops. I have noticed this come and go under different use cases, but I find it occasionally pops on startup and clicks when the digital signal locks or unlocks. This is made worse no doubt by the obscene amounts of output gain, but it's also generally poor design.
Enough about the negatives, let's talk about why this little box is worth putting some effort into. Here's what I like about it:
1. It's an incredibly powerful and user friendly active crossover. It can provide pretty much any crossover type, slope and frequency with great ease, and I find it very intuitive to use. The built in EQ is also extremely powerful and allows the end user to achieve a flat response almost regardless of the drivers chosen in the system.
2. It's self contained. No dragging around a PC with 8 DACS and buying expensive software that has to run in the background, limiting your source to pretty much only your PC.
3. It's incredibly cheap given how powerful of a tool it is. You can buy the DCX for less than the price of your average 3-way passive crossover, which is an impressive feat from Behringer. It's also a fraction of the cost of competing solutions, and in my opinion, offers one of the most powerful DSP sections of the competing products I've seen.
4. The processing itself happens at 24/96 which most other options don't support. The last thing I want to do is downsample my 24/96 recordings back to 16/48 for processing - like other solutions do.
5. It supports both analog and digital inputs which is nice if you have multiple sources (a turntable and a music server for example)
In the next post, I'll provide all the baseline measurements and discuss exactly what might be causing some of the above issues, and what we can do to fix it.
Cheers,
Owen
First Post Edit:
This thread has resulted in a replacement output PCB to address many of the issues covered in this thread.
Details on the PCB can be found in this post:
http://www.diyaudio.com/forums/digi...improvement-stock-dcx2496-12.html#post4424533
Measurements can be found here:
http://www.diyaudio.com/forums/digi...improvement-stock-dcx2496-13.html#post4424612
Boards can be purchased by signing up here (please read instructions in the spreadsheet):
https://docs.google.com/spreadsheets/d/1bRIlMzzoHpJYvS8kQ_fVN-UgCuzX58TXHaJvsF43iLg/edit?usp=sharing
------------------------------------------------------------------------------------------------------------------------------------------------
Hi All,
Being a long time owner of a DCX2496, I have always wanted to tackle a long list of needed improvements to some of the common issues I have had with this slick little XO.
I finally have a need for this crossover in a more critical application than the usual development role it has found itself in for the past few years, so I've finally got some good motivation to get a board done!
The goal here is to start by summarizing all the things I've noticed while listening to the DCX over the years, and then hopefully quantifying those impressions with measurements.
The measurements should provide the baseline for performance, and help to zero in on the most critical issues with the DCX. I want every change to provide a tangible improvement that can be measured and qualified by listening.
So where do we start? How about a list of subjective things I don't like about the DCX as it is now:
1. Noise - this is one of my biggest issues with the DCX. The outputs suffer from a very high level of output noise which is further exacerbated by the gain in whatever amplifier stage is being used after the DCX. I have also noticed that there is significantly more noise when the input is done via AES/EBU, which is my preferred input type.
2. Output levels - This unit was clearly designed to be used with pro level inputs and outputs. The input voltage required to hit 0dBFS on the ADC is 10VRMS, and the output from the DCX at 0dBFS is a whopping 10VRMS as well! In a normal home HiFi, I think aiming for 2VRMS on both ends is a little more realistic and makes much better use of the 24-bit processing by using more of the available dynamic range.
3. It sounds dull. There is something very wrong with the way the upper mid and high frequencies sound and it's hard to pinpoint exactly what is causing it. I find the issue to be glaringly obvious in some recordings, and not as noticeable in others, but it always seems to be there to some degree. I noticed it most while working on a speaker using a pair of NeoPro5i ribbons which sounded astonishing with a passive crossover, and completely bland and flat when crossed over with this unit.
4. It's fatiguing to listen to. Again, I don't know exactly what causes this, but I find when it's in the signal chain, I'm reaching to turn the volume down more frequently than when it's not in the chain. There's something a bit grating to the way it sounds, and I don't like it. This is the main reason I've been using it primarily for development.
5. It clicks and pops. I have noticed this come and go under different use cases, but I find it occasionally pops on startup and clicks when the digital signal locks or unlocks. This is made worse no doubt by the obscene amounts of output gain, but it's also generally poor design.
Enough about the negatives, let's talk about why this little box is worth putting some effort into. Here's what I like about it:
1. It's an incredibly powerful and user friendly active crossover. It can provide pretty much any crossover type, slope and frequency with great ease, and I find it very intuitive to use. The built in EQ is also extremely powerful and allows the end user to achieve a flat response almost regardless of the drivers chosen in the system.
2. It's self contained. No dragging around a PC with 8 DACS and buying expensive software that has to run in the background, limiting your source to pretty much only your PC.
3. It's incredibly cheap given how powerful of a tool it is. You can buy the DCX for less than the price of your average 3-way passive crossover, which is an impressive feat from Behringer. It's also a fraction of the cost of competing solutions, and in my opinion, offers one of the most powerful DSP sections of the competing products I've seen.
4. The processing itself happens at 24/96 which most other options don't support. The last thing I want to do is downsample my 24/96 recordings back to 16/48 for processing - like other solutions do.
5. It supports both analog and digital inputs which is nice if you have multiple sources (a turntable and a music server for example)
In the next post, I'll provide all the baseline measurements and discuss exactly what might be causing some of the above issues, and what we can do to fix it.
Cheers,
Owen
Last edited:
I went passive on the output (adaption of Jan's circuit)- ended any issues with noise or gain, and (talking out of my rear because I don't have solid ears-only data) I think it sounds better. Certainly a lot of measured distortion artifacts at treble frequencies went away. No remote, but I didn't spend what his remote-controlled board costs! 
Alright... on to the measurements!
Let's start with the noise floor of the outputs. I set all 6 channels to bypass the crossovers, set the gain in each channel and the inputs to be 0dB, and measured with both analog inputs and AES/EBU.
Measurements 1 and 2 show the noise floor with analog input, and with AES/EBU respectively. As I had thought, the noise floor is significantly worse using digital input than it is using the analog inputs. My best guess is that the AES/EBU signal is polluting the analog outputs since they all run together on the same un-shielded ribbon cable. The only other possible culprit is the CS8420 which is present on the digital input but not on the ADCs. I doubt this is the source of the noise, however, since that level of noise is significantly worse than what's specified in the datasheet for the CS8420.
The remaining noise present in both the digital and analog input setups is almost certainly lower harmonics from the SMPS switching frequency.
There are large spikes present at 88Hz, 180Hz, 270Hz, 440Hz etc... which are not related to normal linear line rectified supplies. This noise is clearly present right up into the 1-3kHz region.
Based on the above, it looks like a great deal of the noise could be reduced by doing the following:
1. Wire the AES/EBU input directly from a dedicated input connector to the isolation transformer on the digital board using a shielded TP cable.
2. Remove the questionable SMPS and replace it with a fully linear supply using toroidal transformers and ultra-low noise regulators for all rails.
Next up is THD and THD+N. The last 8 measurements show THD and THD+N on the outputs using both analog and digital inputs.
Overall, distortion is reasonable at 1kHz, but it still falls short of the spec for the DAC which would imply that the limitation is the output stage design. Distortion also climbs to rather high levels at higher frequencies which, although probably not audible, points towards an output stage that isn't all that well designed.
These measurements are as follows:
3. Digital Input - THD @ 1kHz
4. Digital Input - THD+N @ 1kHz
5. Digital Input - FFT 1kHz, 2VRMS
6. Digital Input - THD VS FR 2VRMS
7. Analog Input - THD @ 1kHz
8. Analog Input - THD+N @ 1kHz
9. Analog Input - FFT 1kHz, 2VRMS
10. Analog Input - THD VS FR 2VRMS
Based on the above, it's probably best to redo the output stage to maximize the performance to at least the same level as what the DAC is capable of. It's a shame to have the analog output stage being the bottleneck when it doesn't cost much to do it right!
More to follow...
Cheers,
Owen
Let's start with the noise floor of the outputs. I set all 6 channels to bypass the crossovers, set the gain in each channel and the inputs to be 0dB, and measured with both analog inputs and AES/EBU.
Measurements 1 and 2 show the noise floor with analog input, and with AES/EBU respectively. As I had thought, the noise floor is significantly worse using digital input than it is using the analog inputs. My best guess is that the AES/EBU signal is polluting the analog outputs since they all run together on the same un-shielded ribbon cable. The only other possible culprit is the CS8420 which is present on the digital input but not on the ADCs. I doubt this is the source of the noise, however, since that level of noise is significantly worse than what's specified in the datasheet for the CS8420.
The remaining noise present in both the digital and analog input setups is almost certainly lower harmonics from the SMPS switching frequency.
There are large spikes present at 88Hz, 180Hz, 270Hz, 440Hz etc... which are not related to normal linear line rectified supplies. This noise is clearly present right up into the 1-3kHz region.
Based on the above, it looks like a great deal of the noise could be reduced by doing the following:
1. Wire the AES/EBU input directly from a dedicated input connector to the isolation transformer on the digital board using a shielded TP cable.
2. Remove the questionable SMPS and replace it with a fully linear supply using toroidal transformers and ultra-low noise regulators for all rails.
Next up is THD and THD+N. The last 8 measurements show THD and THD+N on the outputs using both analog and digital inputs.
Overall, distortion is reasonable at 1kHz, but it still falls short of the spec for the DAC which would imply that the limitation is the output stage design. Distortion also climbs to rather high levels at higher frequencies which, although probably not audible, points towards an output stage that isn't all that well designed.
These measurements are as follows:
3. Digital Input - THD @ 1kHz
4. Digital Input - THD+N @ 1kHz
5. Digital Input - FFT 1kHz, 2VRMS
6. Digital Input - THD VS FR 2VRMS
7. Analog Input - THD @ 1kHz
8. Analog Input - THD+N @ 1kHz
9. Analog Input - FFT 1kHz, 2VRMS
10. Analog Input - THD VS FR 2VRMS
Based on the above, it's probably best to redo the output stage to maximize the performance to at least the same level as what the DAC is capable of. It's a shame to have the analog output stage being the bottleneck when it doesn't cost much to do it right!
More to follow...
Cheers,
Owen
Attachments
I went passive on the output (adaption of Jan's circuit)- ended any issues with noise or gain, and (talking out of my rear because I don't have solid ears-only data) I think it sounds better. Certainly a lot of measured distortion artifacts at treble frequencies went away. No remote, but I didn't spend what his remote-controlled board costs!
I would imagine the significantly lower gain (almost 21dB less!) would certainly help reduce the output noise, but given the poor PSRR of the DAC itself (50dB), I think the 5VA rail for the DAC would also have to be looked at to address the noise from the PSU.
The DAC itself also has a rather low differential output of 0.85VRMS at 0dBFS, so I think I would opt for a single OPA1632 per channel with a gain of 2.4 to get the differential output up to 2VRMS and the SE output up to 1VRMS.
Are you using balanced or SE outputs? If you're using SE then you're giving up the CMRR needed to maximize the DAC performance and you're only getting 0.425VRMS at 0dBFS
I think a well executed output stage would still be beneficial compared to just running straight out of the DAC, although running straight out of the DAC is almost certainly better than running it stock
Cheers,
Owen
Next up is frequency response...
Everything here looks pretty good, and inter-channel gain matching is actually much better than I was expecting.
Using the digital input, all six channels are within 0.1dB of each other which is quite good. Each respective channel is down about 0.13dB at 20Hz and 0.2dB at 20kHz.
Using the analog inputs pushes the channel matching a little broader, but still within 0.2dB which is pretty good. Each respective channel in this case is down about 0.2dB at both 20Hz and 20kHz.
Overall, no problems with the FR. I still prefer DC coupled outputs, but there are no serious red flags here, and certainly nothing to explain the poor high frequency performance!
1. FR and Inter-channel matching with Digital input
2. FR and Inter-channel matching with Analog input
So we've covered most of the basics... noise, distortion, frequency response, and we still don't have an explanation for the dull sound and the poor high frequency performance. After performing all the above measurements, I figured it would be best to see if I could get some meaningful jitter measurements from a system level to see if maybe that was the culprit.
It turns out this thing happens to have by far the worst jitter artifacts I have ever seen in anything I've ever measured. I can't be sure until I fix it, but I think this might have to something to do with the less than desirable sound.
3. Jitter Artifacts - Digital input at 44.1kHz
4. Jitter Artifacts - Digital input at 48kHz
5. Jitter Artifacts - Digital input at 96kHz
6. Jitter Artifacts - Analog input at 96kHz
Unfortunately I can only measure for artifacts at 96kHz using the analog input because that's the only frequency it operates at. I was able to measure at each of the common frequencies for the digital input.
Overall, this is very poor performance. As a reference, the NTD1 and the ESS9018 in the same measurement setup only have two tiny artifacts at -155dBr. The DCX produces a plethora of artifacts, some at a whopping -86dBr which is higher than most of the distortion harmonics!
Interestingly, the issue is mostly confined to the digital input, which tells me it probably has something to do with the CS8420. The fact that it gets worse as we approach a 1:1 ration also hints at this.
This issue definitely needs to be addressed, either through the use of a better ASRC, or possibly a better clock, although the more reasonable jitter levels on the analog input side would imply that the existing clock isn't really to blame.
I was really hoping not to have to alter the digital board, but it looks like that might not be an option.
More to come!
Cheers,
Owen
Everything here looks pretty good, and inter-channel gain matching is actually much better than I was expecting.
Using the digital input, all six channels are within 0.1dB of each other which is quite good. Each respective channel is down about 0.13dB at 20Hz and 0.2dB at 20kHz.
Using the analog inputs pushes the channel matching a little broader, but still within 0.2dB which is pretty good. Each respective channel in this case is down about 0.2dB at both 20Hz and 20kHz.
Overall, no problems with the FR. I still prefer DC coupled outputs, but there are no serious red flags here, and certainly nothing to explain the poor high frequency performance!
1. FR and Inter-channel matching with Digital input
2. FR and Inter-channel matching with Analog input
So we've covered most of the basics... noise, distortion, frequency response, and we still don't have an explanation for the dull sound and the poor high frequency performance. After performing all the above measurements, I figured it would be best to see if I could get some meaningful jitter measurements from a system level to see if maybe that was the culprit.
It turns out this thing happens to have by far the worst jitter artifacts I have ever seen in anything I've ever measured. I can't be sure until I fix it, but I think this might have to something to do with the less than desirable sound.
3. Jitter Artifacts - Digital input at 44.1kHz
4. Jitter Artifacts - Digital input at 48kHz
5. Jitter Artifacts - Digital input at 96kHz
6. Jitter Artifacts - Analog input at 96kHz
Unfortunately I can only measure for artifacts at 96kHz using the analog input because that's the only frequency it operates at. I was able to measure at each of the common frequencies for the digital input.
Overall, this is very poor performance. As a reference, the NTD1 and the ESS9018 in the same measurement setup only have two tiny artifacts at -155dBr. The DCX produces a plethora of artifacts, some at a whopping -86dBr which is higher than most of the distortion harmonics!
Interestingly, the issue is mostly confined to the digital input, which tells me it probably has something to do with the CS8420. The fact that it gets worse as we approach a 1:1 ration also hints at this.
This issue definitely needs to be addressed, either through the use of a better ASRC, or possibly a better clock, although the more reasonable jitter levels on the analog input side would imply that the existing clock isn't really to blame.
I was really hoping not to have to alter the digital board, but it looks like that might not be an option.
More to come!
Cheers,
Owen
Attachments
Last edited:
Im in!.
I have balanced passive output - and coaxial straight to trafo.
I think that passive output has 1,7v but im not sure. For sure it is enough.
When i went to passive - i start to have problem with digital signal - so i went to coaxial direct to transformer - but it didnt help. It was very strange grounding problem - i must cut some grounds in ribbon cables and left some. Just one combination works. I use spdif /i change resistor behind trafo to 75ohm/ and with some sound cards - coaxial connector must be connected to DCX chassis otherwise it did not play and with other sound card it must float...
Basicaly grounding problem behind my comprehension.
Do you guys thinking about Najda board? For ca 250E maybe better point to start. Also it has 8 channels... /but only SE/
Back to DCX. One of ultimate digital input solutions is GOOD new clock in DCX
optical in and optical out module back to source to slave source to DCX clock.
Many sound cards has this option /f.e. ESI july/ this way get total isolation from PC /optical/ and utilize good local clock in DCX - without need for PLL clock reconstruction. You also bypass ASRC in DCX - so you must send 24/96 from PC. But you can configure f.e. WIN7 to do that. And use very hiquality resampling in PC f.e. SOX
Other solution is Asynchronous USB module
I have balanced passive output - and coaxial straight to trafo.
I think that passive output has 1,7v but im not sure. For sure it is enough.
When i went to passive - i start to have problem with digital signal - so i went to coaxial direct to transformer - but it didnt help. It was very strange grounding problem - i must cut some grounds in ribbon cables and left some. Just one combination works. I use spdif /i change resistor behind trafo to 75ohm/ and with some sound cards - coaxial connector must be connected to DCX chassis otherwise it did not play and with other sound card it must float...
Basicaly grounding problem behind my comprehension.
Do you guys thinking about Najda board? For ca 250E maybe better point to start. Also it has 8 channels... /but only SE/
Back to DCX. One of ultimate digital input solutions is GOOD new clock in DCX
optical in and optical out module back to source to slave source to DCX clock.
Many sound cards has this option /f.e. ESI july/ this way get total isolation from PC /optical/ and utilize good local clock in DCX - without need for PLL clock reconstruction. You also bypass ASRC in DCX - so you must send 24/96 from PC. But you can configure f.e. WIN7 to do that. And use very hiquality resampling in PC f.e. SOX
Other solution is Asynchronous USB module
Impossible.
No way that behringer PCB survive. And besides AKM 4393 are quite good.
some links:
Ergo Audio :: Oettle digi-in mod
I did modify my DX2496 by gutting the analouge parts.
Replaced them with a simple single ended - differential buffer with Vcom offset for the ADC input, and a simple 2nd order differential filter on each DAC out, cutting at 70 kHz aboutish. In total 5 OPA2134. All built on Veroboard (but with some consideration to layout)
Gain structure is designed that 2 Volt peaks ADC input.
Sonically it was a great improvement of course with far less noise. I really encourage you to proceed in this.
In progress is a linear powersupply with external transformer box , both for noise reasons but also because I want to remove the chassis from protective earth due to ground loops, and in that I will construct a new chassis and run the DSP main board bare (no display front panel card) Second to that is a 6 channel volume control and I am home free.
Replaced them with a simple single ended - differential buffer with Vcom offset for the ADC input, and a simple 2nd order differential filter on each DAC out, cutting at 70 kHz aboutish. In total 5 OPA2134. All built on Veroboard (but with some consideration to layout)
Gain structure is designed that 2 Volt peaks ADC input.
Sonically it was a great improvement of course with far less noise. I really encourage you to proceed in this.
In progress is a linear powersupply with external transformer box , both for noise reasons but also because I want to remove the chassis from protective earth due to ground loops, and in that I will construct a new chassis and run the DSP main board bare (no display front panel card) Second to that is a 6 channel volume control and I am home free.
Alright... this is the last of the measurements for the stock unit that I wanted to get posted. These are mostly a mishmash of things I measured while I had everything set up.
1. SNR @ 2VRMS output, 20kHz BW
- I wanted to measure this to get an idea of what the baseline SNR is with a 2VRMS output. This is more in-line with what most hifi users will actually be getting from the unit since most people can't make use of the 10VRMS output. As expected, it's not great and ranges anywhere from 91dB to 98dB.
2. Distortion Product Ratio - 1kHz 2VRMS
- This just makes it easy to see the distortion signature of the stock output stage. It has a dominant 3rd harmonic that is about 88dB down, and a second harmonic that is about 97dB down.
3. FFT of Noise Floor - Full Band 20Hz to 80kHz
- I wanted to see if there was anything particularly nasty a little farther out, but everything looks alright here. The noise floor starts to climb after 40kHz, but nothing too scary.
4. Output noise level - 20kHz BW
- Self explanatory
5. Output noise level - 90kHz BW
- Self explanatory
6. THD VS FR with XO on and off.
- I wanted to make sure that enabling the crossovers didn't impact distortion. Thankfully, there is no increase in distortion levels with the filters on or off. This is very good news, as it indicates that the DSP section itself is not contributing to distortion.
7. THD VS Level - Analog Input
- Self explanatory
8. THD VS Level - Digital Input
- Self explanatory
Next up - a plan on how I'd like to address some of the issues found in the measurements, and hopefully a discussion of what people would like to keep in the DCX, and what they can live without.
Cheers,
Owen
1. SNR @ 2VRMS output, 20kHz BW
- I wanted to measure this to get an idea of what the baseline SNR is with a 2VRMS output. This is more in-line with what most hifi users will actually be getting from the unit since most people can't make use of the 10VRMS output. As expected, it's not great and ranges anywhere from 91dB to 98dB.
2. Distortion Product Ratio - 1kHz 2VRMS
- This just makes it easy to see the distortion signature of the stock output stage. It has a dominant 3rd harmonic that is about 88dB down, and a second harmonic that is about 97dB down.
3. FFT of Noise Floor - Full Band 20Hz to 80kHz
- I wanted to see if there was anything particularly nasty a little farther out, but everything looks alright here. The noise floor starts to climb after 40kHz, but nothing too scary.
4. Output noise level - 20kHz BW
- Self explanatory
5. Output noise level - 90kHz BW
- Self explanatory
6. THD VS FR with XO on and off.
- I wanted to make sure that enabling the crossovers didn't impact distortion. Thankfully, there is no increase in distortion levels with the filters on or off. This is very good news, as it indicates that the DSP section itself is not contributing to distortion.
7. THD VS Level - Analog Input
- Self explanatory
8. THD VS Level - Digital Input
- Self explanatory
Next up - a plan on how I'd like to address some of the issues found in the measurements, and hopefully a discussion of what people would like to keep in the DCX, and what they can live without.
Cheers,
Owen
Attachments
So here's the rough initial plan, broken down into things that will definitely be done, and things I would like to do, but that may require a little more effort to implement.
Everything here is open to debate and suggestions, so don't be shy.
Replace the entire analog output and input section
I would propose the use of a single OPA1632 per channel for all 6 outputs and the two inputs. The balanced outputs from the DAC would run directly into the OPA1632 and then directly to the output connectors. The gain would be set to give 2VRMS at 0dBFS for the output and 0dBFS @2VRMS for the input.
The above has the following advantages:
1. It maintains a fully differential signal path, which is what the DAC wants.
2. It allows the use of a single phase of the output for SE, while still maintaining the CMRR the DAC wants to see.
3. It brings the gain in-line with what we need for hifi use
4. The MUTE functionality can be maintained by connecting to the enable pin on the OPA.
5. Performance of the OPA is worlds better than the stock op-amps.
6. Can be DC coupled, and allows for an optional LPF to be populated.
Replace the entire SMPS with a linear supply
I would like to strip out the existing SMPS entirely and replace it with a fully linear supply. A trio of small 10VA toroids (one for digital and two for analog) could be used to feed a dedicated PSU section with a good rectifier section and full suite of low noise regulators. The TPS7A3301 in a TO-220 pack could be used for all first stage supplies (+/-15VA, +5VA +3.3VA +5VD +3.3VD) and then smaller positive voltage regs could be used for any subsequent supplies.
All of the above would have to fit inside the chassis as I'm not a big fan of making anything external. The toroids themselves are small enough that they can me made to fit and be kept a good distance from the sensitive output circuitry.
Overall, the above PSU changes would dramatically improve the noise floor of the unit, and reduce EMI/EMC concerns.
Remove all input circuitry from input C and make it a dedicated AES/EBU input
Since I have never in my life connected a MIC to the DCX, I would propose the removal of the input circuitry for input C, and instead dedicate it to be the AES/EBU input which would run directly to the digital board. The UI would remain exactly as is (switching between analog and digital input would still work as it did before) but the option to plug in a MIC would no longer be there. This should reduce some of the noise being coupled in from digital input to the analog outputs.
More to come...
Cheers,
Owen
Everything here is open to debate and suggestions, so don't be shy.
Replace the entire analog output and input section
I would propose the use of a single OPA1632 per channel for all 6 outputs and the two inputs. The balanced outputs from the DAC would run directly into the OPA1632 and then directly to the output connectors. The gain would be set to give 2VRMS at 0dBFS for the output and 0dBFS @2VRMS for the input.
The above has the following advantages:
1. It maintains a fully differential signal path, which is what the DAC wants.
2. It allows the use of a single phase of the output for SE, while still maintaining the CMRR the DAC wants to see.
3. It brings the gain in-line with what we need for hifi use
4. The MUTE functionality can be maintained by connecting to the enable pin on the OPA.
5. Performance of the OPA is worlds better than the stock op-amps.
6. Can be DC coupled, and allows for an optional LPF to be populated.
Replace the entire SMPS with a linear supply
I would like to strip out the existing SMPS entirely and replace it with a fully linear supply. A trio of small 10VA toroids (one for digital and two for analog) could be used to feed a dedicated PSU section with a good rectifier section and full suite of low noise regulators. The TPS7A3301 in a TO-220 pack could be used for all first stage supplies (+/-15VA, +5VA +3.3VA +5VD +3.3VD) and then smaller positive voltage regs could be used for any subsequent supplies.
All of the above would have to fit inside the chassis as I'm not a big fan of making anything external. The toroids themselves are small enough that they can me made to fit and be kept a good distance from the sensitive output circuitry.
Overall, the above PSU changes would dramatically improve the noise floor of the unit, and reduce EMI/EMC concerns.
Remove all input circuitry from input C and make it a dedicated AES/EBU input
Since I have never in my life connected a MIC to the DCX, I would propose the removal of the input circuitry for input C, and instead dedicate it to be the AES/EBU input which would run directly to the digital board. The UI would remain exactly as is (switching between analog and digital input would still work as it did before) but the option to plug in a MIC would no longer be there. This should reduce some of the noise being coupled in from digital input to the analog outputs.
More to come...
Cheers,
Owen
Linear supply will be nice.
I dont care about mic. But i think most people want SPDIF /or better USB/ instead Aes/Ebu. Also without changing ASRC and clock almost not worth to do.
Please if you can measure also balanced direct voltage out - with simple passive low pass and DC cap. Load ca. 1k - very easy mod and maybe good enough /for out section only/.
thank you very much
I dont care about mic. But i think most people want SPDIF /or better USB/ instead Aes/Ebu. Also without changing ASRC and clock almost not worth to do.
Please if you can measure also balanced direct voltage out - with simple passive low pass and DC cap. Load ca. 1k - very easy mod and maybe good enough /for out section only/.
thank you very much
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