Marantz CD73
- By Pillo69
- Digital Source
- 13 Replies
Hello, although I do not participate much in the forum I am a regular reader.
I have purchased a Marantz cd73.
The player does not work, the drawer opens and closes, the "Laser" led is on and the green LEDs on the top cover light up. But it does nothing else.
You can hear the noise of the drawer motor running, but the arm does not work either.
The recommendations that I have read are about the power supply, I have measured and I have the voltages equal to the diagram.
Someone knows and can guide me where the problem may come from.
The image is borrowed from the web.
I have purchased a Marantz cd73.
The player does not work, the drawer opens and closes, the "Laser" led is on and the green LEDs on the top cover light up. But it does nothing else.
You can hear the noise of the drawer motor running, but the arm does not work either.
The recommendations that I have read are about the power supply, I have measured and I have the voltages equal to the diagram.
Someone knows and can guide me where the problem may come from.
The image is borrowed from the web.
synergy.....Take #7
Or at least I'm calling it Take 7.
It's really the 3rd completely different cabinet.
Other 'takes' were trying out different size ports or different drivers, etc.
They've all been desinged to go with subs at 100Hz like my previous synergy thread 100Hz two-way synergy project.
That thread ended up turning into this 60x40
I've truly enjoyed this speaker, to the point I wanted to build another for stereo, but it's too damn big and heavy to move around. About 100 lbs.
One of the biggest joys in audio for me, is being able to move gear around, listen outside, swap stuff around, always making comparisons and learning.
So I've spent a few months researching how to get the synergy goodness, but shed alot of weight.
Anyway here's the outcome.
It's a 90x60, that has very close to the same primary horn width and height, but of course less depth due to its wider pattern.
It weighs in at a glorious 45 lbs !!!
Uses two dirt cheap faital 10FE200's as mids, and one not so cheap b&c dcx 464 coax cd, ...but getting the CD down to 500Hz is what lets it work.
I did a few things different from my previous attempts that all tried to adhere to the conventional synergy wisdom.
Biggest change was putting the mid ports in the center of the horn instead of in the corners.
I made a test horn where I could block ports easily and experimented a bit. I couldn't find a compelling case for corner ports, in terms of how the HF and VHF were effected.
That was a peasant surprise really, because I wanted the ports in the horn center as part of a weight reduction design. It allowed the mid drivers to go on the top and bottom in small cavities, and the horn side walls be all the cabinent there is.
Like this
Plus, ports in the center of the horn are just plain easier to make 🙂
Also made a simple box to test port sizes and shapes, port thickness, and how much port frustrum matters. Picture a sealed box where you just lay a plate with various port tries on top, and move it around over the cone.
I used plates out of 12mm BB and thin steel sheet, and ended up choosing a 10:1 compression ratio in terms of hole size.
Surprisingly, I could not find anything to beat a simple round hole aligned with the center of the cone.
Neither could i find a reason to use thin ports, or even do anything to the wood hole other than a little roundover smoothing.
The port holes' center is 7.5" from the throat. Basically, the 10" drivers sit as close to the horn apex as possible, which also puts the port nearly under the center of the cone.
I always have to give sincere thanks to BWaslo for his speadsheet....I simply could not make a synergy without it.
Also need to thank Makita for my track saw Lol. It makes the bevel/miter cuts so easy and true...much better than i can do by table saw and sleds.
Oh, I've found I like to make the horn start at 1.4" square, and not have to round out the walls to mate the CD. It's been easier to add museum wax to the horn corners in the mouth.
Syn7 may or may not get a secondary flare depending on outdoor testing.
I should be able to make a detachable flare like in the first pic fairly easily.
What I really want to do is add a tractrix flare as the secondary flare.
Ala the K-402. I figure the K-402 first 2/3's from the throat is pretty much straight sided conical, and the last 1/3 tractrix. A conatrix or a tractical, depending on which end you look from haha
It's really the 3rd completely different cabinet.
Other 'takes' were trying out different size ports or different drivers, etc.
They've all been desinged to go with subs at 100Hz like my previous synergy thread 100Hz two-way synergy project.
That thread ended up turning into this 60x40
I've truly enjoyed this speaker, to the point I wanted to build another for stereo, but it's too damn big and heavy to move around. About 100 lbs.
One of the biggest joys in audio for me, is being able to move gear around, listen outside, swap stuff around, always making comparisons and learning.
So I've spent a few months researching how to get the synergy goodness, but shed alot of weight.
Anyway here's the outcome.
It's a 90x60, that has very close to the same primary horn width and height, but of course less depth due to its wider pattern.
It weighs in at a glorious 45 lbs !!!
Uses two dirt cheap faital 10FE200's as mids, and one not so cheap b&c dcx 464 coax cd, ...but getting the CD down to 500Hz is what lets it work.
I did a few things different from my previous attempts that all tried to adhere to the conventional synergy wisdom.
Biggest change was putting the mid ports in the center of the horn instead of in the corners.
I made a test horn where I could block ports easily and experimented a bit. I couldn't find a compelling case for corner ports, in terms of how the HF and VHF were effected.
That was a peasant surprise really, because I wanted the ports in the horn center as part of a weight reduction design. It allowed the mid drivers to go on the top and bottom in small cavities, and the horn side walls be all the cabinent there is.
Like this
Plus, ports in the center of the horn are just plain easier to make 🙂
Also made a simple box to test port sizes and shapes, port thickness, and how much port frustrum matters. Picture a sealed box where you just lay a plate with various port tries on top, and move it around over the cone.
I used plates out of 12mm BB and thin steel sheet, and ended up choosing a 10:1 compression ratio in terms of hole size.
Surprisingly, I could not find anything to beat a simple round hole aligned with the center of the cone.
Neither could i find a reason to use thin ports, or even do anything to the wood hole other than a little roundover smoothing.
The port holes' center is 7.5" from the throat. Basically, the 10" drivers sit as close to the horn apex as possible, which also puts the port nearly under the center of the cone.
I always have to give sincere thanks to BWaslo for his speadsheet....I simply could not make a synergy without it.
Also need to thank Makita for my track saw Lol. It makes the bevel/miter cuts so easy and true...much better than i can do by table saw and sleds.
Oh, I've found I like to make the horn start at 1.4" square, and not have to round out the walls to mate the CD. It's been easier to add museum wax to the horn corners in the mouth.
Syn7 may or may not get a secondary flare depending on outdoor testing.
I should be able to make a detachable flare like in the first pic fairly easily.
What I really want to do is add a tractrix flare as the secondary flare.
Ala the K-402. I figure the K-402 first 2/3's from the throat is pretty much straight sided conical, and the last 1/3 tractrix. A conatrix or a tractical, depending on which end you look from haha
Add a second output to a passive summing mixer
- By mfoucault
- Instruments and Amps
- 11 Replies
Hi,
I plan to build a passive summing mixer with 8 inputs and 2 outputs.
I follow this diagram :
All inputs resistors are 6.8k ohms
To calculate the summing resistor I use the New York Dave formula (I/N)*R/((I/N)-R, where I is the input impedance, R the desired output impedance, and N the number of channels.
So, here, the summing resistor is 165 ohms
How to calculate the output summing resistors needed if I want to add a second output to this diagram ? Each output need to be at 150 ohm for the mic pre amp that after.
Thanks
I plan to build a passive summing mixer with 8 inputs and 2 outputs.
I follow this diagram :
All inputs resistors are 6.8k ohms
To calculate the summing resistor I use the New York Dave formula (I/N)*R/((I/N)-R, where I is the input impedance, R the desired output impedance, and N the number of channels.
So, here, the summing resistor is 165 ohms
How to calculate the output summing resistors needed if I want to add a second output to this diagram ? Each output need to be at 150 ohm for the mic pre amp that after.
Thanks
Equal-pressure loudness contours
- By guferr
- Everything Else
- 14 Replies
So, everyone knows the equal-loudness curves, which show the sound pressure (in dB SPL) needed to achieve a particular loudness in phons through different frequencies.
This curve is fundamental for many areas of acoustics and hearing, however, I wanted to bring more information out of it.
Firstly, they don't show the variation in loudness of sounds with a same sound pressure through different frequencies. That is very important, specially if you want to understand how a "flat response" should sound like when you feed it with pure tones of the same amplitude.
Secondly, the unit phon itself isn't the closest to be linear with perceived loudness. The unit sone was made for that, but needs corrections for very loud and very quiet sound (and it's a bit of an approximation anyway, loudness is very subjective, but this is still better for a psychophysical approach than phon).
So, I took the data from ISO 226, converting the values from phons to sones and the values from db SPL to Pa (pascals) to find a loudness curve for each frequency, relating the perceived loudness with sound pressure.
For example, for the 100 Hz frequency, the data is:
I then performed a fitting to find the values for other sound pressures. The model I used is a simple power-law expression:
L = A·pγ
Where L is the loudness, A is a constant, p is the sound pressure and γ the exponent (a dimensionless constant)
Not only this model provided a near-perfect fit (R² > 0.999 always), it is also the same used for gamma correction, and the value of γ (gamma) tells a lot about dynamics.
If γ = 1 for a given frequency, then the perceived loudness is directly proportional to the sound pressure. Which only happens in one frequency, around 60 Hz.
If γ < 1, then the perceived loudness is compressed, meaning that doubling the sound pressure doesn't double the perceived loudness. Everything above 60 Hz is compressed, and the most compressed frequencies are exactly those we are most sensitive to, between 2 and 6 kHz.
If γ > 1, which is the case for everything below 60 Hz, then the perceived loudness is expanded, so doubling the sound pressure more-than-doubles the perceived loudness.
The constant A is a different one for each frequency, and also has a different dimension for each. Its unit has to be sone·Pa-γ, so it's only the same dimension if γ is the same. Nevertheless, it's still useful to compare their values, for they tell a lot about how each frequency is "intrinsically loud".
Here's a graph of how the A constants and γ values vary with frequency
The graph on the left will give the overall shape of the equal-pressure loudness contours. The graph on the right tells about the dynamics of each frequency and how the contours will change with different sound pressures.
So, using these data, it's possible to make the contours I wished for.
Here they are:
Each curve in the graph on the left shows the variation in perceived loudness for pure tones of different frequencies with the same sound pressure.
The curves on the graph on the right has been normalized so the loudest frequency of each curve has a value 1, and it helps visualizing how the proportions between the loudness of each frequency changes with sound pressure (due to their different dynamics).
In the curve for 20 mPa (the cyan curve), the frequencies around 15 kHz are just about 40% as loud as the frequencies around 3 kHz (the loudest in all cases). But, for 2 Pa (black curve), they're 80% as loud as 3 kHz. Their loudness relative to 3 kHz double.
The bass frequencies change even more. In this same range (from 20 mPa to 2 Pa), frequencies around 40 Hz go from being 1% as loud as 3 kHz to 10%.
The frequencies between 2 and 6 kHz keep their proportional loudness because their γ are nearly the same, around 0.585 (they have the same dynamics).
Although the data may be imprecise, this is just an approximation when it comes to your actual loudness perception. But in a way or the other, this is still useful for a ton of qualitative observations.
The main observation being that, no mater how an equipment is good at keeping its profile (flat or whatever other it has), the equalization will sound different at different SPL, increasing the overall loudness will also increase the bass and mid-bass (from 20 to 600 Hz) and treble (12-17 kHz) relative to mid-highs(2-6 kHz).
These contours also help "equalizing by ear", as they give a notion of how it should sound, taking the loudness of 3 kHz as a reference. The only thing you need to know then it's the sound pressure level you're dealing with.
For a reference, here's a little table I took from wikipedia:
This curve is fundamental for many areas of acoustics and hearing, however, I wanted to bring more information out of it.
Firstly, they don't show the variation in loudness of sounds with a same sound pressure through different frequencies. That is very important, specially if you want to understand how a "flat response" should sound like when you feed it with pure tones of the same amplitude.
Secondly, the unit phon itself isn't the closest to be linear with perceived loudness. The unit sone was made for that, but needs corrections for very loud and very quiet sound (and it's a bit of an approximation anyway, loudness is very subjective, but this is still better for a psychophysical approach than phon).
So, I took the data from ISO 226, converting the values from phons to sones and the values from db SPL to Pa (pascals) to find a loudness curve for each frequency, relating the perceived loudness with sound pressure.
For example, for the 100 Hz frequency, the data is:
I then performed a fitting to find the values for other sound pressures. The model I used is a simple power-law expression:
L = A·pγ
Where L is the loudness, A is a constant, p is the sound pressure and γ the exponent (a dimensionless constant)
Not only this model provided a near-perfect fit (R² > 0.999 always), it is also the same used for gamma correction, and the value of γ (gamma) tells a lot about dynamics.
If γ = 1 for a given frequency, then the perceived loudness is directly proportional to the sound pressure. Which only happens in one frequency, around 60 Hz.
If γ < 1, then the perceived loudness is compressed, meaning that doubling the sound pressure doesn't double the perceived loudness. Everything above 60 Hz is compressed, and the most compressed frequencies are exactly those we are most sensitive to, between 2 and 6 kHz.
If γ > 1, which is the case for everything below 60 Hz, then the perceived loudness is expanded, so doubling the sound pressure more-than-doubles the perceived loudness.
The constant A is a different one for each frequency, and also has a different dimension for each. Its unit has to be sone·Pa-γ, so it's only the same dimension if γ is the same. Nevertheless, it's still useful to compare their values, for they tell a lot about how each frequency is "intrinsically loud".
Here's a graph of how the A constants and γ values vary with frequency
The graph on the left will give the overall shape of the equal-pressure loudness contours. The graph on the right tells about the dynamics of each frequency and how the contours will change with different sound pressures.
So, using these data, it's possible to make the contours I wished for.
Here they are:
Each curve in the graph on the left shows the variation in perceived loudness for pure tones of different frequencies with the same sound pressure.
The curves on the graph on the right has been normalized so the loudest frequency of each curve has a value 1, and it helps visualizing how the proportions between the loudness of each frequency changes with sound pressure (due to their different dynamics).
In the curve for 20 mPa (the cyan curve), the frequencies around 15 kHz are just about 40% as loud as the frequencies around 3 kHz (the loudest in all cases). But, for 2 Pa (black curve), they're 80% as loud as 3 kHz. Their loudness relative to 3 kHz double.
The bass frequencies change even more. In this same range (from 20 mPa to 2 Pa), frequencies around 40 Hz go from being 1% as loud as 3 kHz to 10%.
The frequencies between 2 and 6 kHz keep their proportional loudness because their γ are nearly the same, around 0.585 (they have the same dynamics).
Although the data may be imprecise, this is just an approximation when it comes to your actual loudness perception. But in a way or the other, this is still useful for a ton of qualitative observations.
The main observation being that, no mater how an equipment is good at keeping its profile (flat or whatever other it has), the equalization will sound different at different SPL, increasing the overall loudness will also increase the bass and mid-bass (from 20 to 600 Hz) and treble (12-17 kHz) relative to mid-highs(2-6 kHz).
These contours also help "equalizing by ear", as they give a notion of how it should sound, taking the loudness of 3 kHz as a reference. The only thing you need to know then it's the sound pressure level you're dealing with.
For a reference, here's a little table I took from wikipedia:
-
Locked
LM3886 - LINTRON Toroidal Transformer 15V @ 30VA??
Hi Guys,
Just managed to pick up:
LINTRON Toroidal Mains Transformer 15V @ 30VA
Is this going to sufficient for driving a single LM3886?
Just managed to pick up:
LINTRON Toroidal Mains Transformer 15V @ 30VA
Is this going to sufficient for driving a single LM3886?
Trouble Shoot Relay Switching
- By Steve Luck
- Solid State
- 1 Replies
All ready to give away a very sweet sounding amp and speaker system to my sister and brother in law when I discovered one channel on one of the three sources not working at all.
Three pairs of RCA input each pair connected to a relay - I managed to desolder the 10 pin relay and replace it with a socket and a new relay but still no joy on the same input source. Relays are triggered by three way toggle switch on the front of the amp. Relays are TQ2 5V DPDT Type Form C
https://b2b-api.panasonic.eu/file_stream/pids/fileversion/4514
Output from the circuit board is the same pair of cables sharing a ground connection and L and R for all three relays. There's a couple of resistors and transistor on each relay - I don't understand why I only get one channel if the relay is working.
Any Ideas very welcome Pictures from Ali Express site with source relay board in the middle.
Three pairs of RCA input each pair connected to a relay - I managed to desolder the 10 pin relay and replace it with a socket and a new relay but still no joy on the same input source. Relays are triggered by three way toggle switch on the front of the amp. Relays are TQ2 5V DPDT Type Form C
https://b2b-api.panasonic.eu/file_stream/pids/fileversion/4514
Output from the circuit board is the same pair of cables sharing a ground connection and L and R for all three relays. There's a couple of resistors and transistor on each relay - I don't understand why I only get one channel if the relay is working.
Any Ideas very welcome Pictures from Ali Express site with source relay board in the middle.
Happy New Year for 2021!
Happy New Year to all of the Pass Labs followers out there! It's nice to have a refuge where we can distract ourselves during difficult times.
I hope things settle down and return to a sense of normalcy in 2021. Until they do, stay healthy my friends!
I hope things settle down and return to a sense of normalcy in 2021. Until they do, stay healthy my friends!
Rotel RA-930BX
- By Karl vd Berg
- Solid State
- 31 Replies
Hi all,
I have a problem with my Rotel RA-930BX. It simply is totally off!
Fuses are ok, I measured the AC in the board and it is ok also.
Maybe it's an easy thing, but I can't find where's the problem!
Just stopped listening to the music! 🙁
Any help will be much appreciated.😱
I have a problem with my Rotel RA-930BX. It simply is totally off!
Fuses are ok, I measured the AC in the board and it is ok also.
Maybe it's an easy thing, but I can't find where's the problem!
Just stopped listening to the music! 🙁
Any help will be much appreciated.😱
Attachments
Simulation stability with different models - Dado's 200W CFA
- By Onemangang
- Solid State
- 4 Replies
Hi everyone, and happy newyear 
Sorry if this has been discussed before, could not find anything on it.
I have very much enjoyed dado's 200W CFA thread and downloaded the .asc file from post #1582 assuming it is the newest.
I did not have the IRFP240Ckst/IRFP240Ckst models handy, so instead I used the following models from Ian Hegglun.
*VDMOS with subthreshold (c) Ian Hegglun
.model IRFP240h VDMOS (Rg=17 Vto=4.0 Kp=6 Lambda=3m
+ Rs=40m Ksubthres=0.15 Mtriode=0.35 Rd=0.15
+ Bex=-2.3 Vtotc=-6m Tksubthres1=4m Trs1=3.5m Trd1=5m
+ Cgdmax=1.3n Cgdmin=10p a=0.35 Cgs=1.2n Cjo=1n
+ m=0.4 VJ=0.75 IS=10n N=1.5 Eg=1.1 Rb=10m Trb1=3m
+ Vds=200 Ron=0.15 Qg=45nC mfg=VishIH1907)
*VDMOS with subthreshold (c) Ian Hegglun
.model IRFP9240h VDMOS (pchan Rg=9 Vto=-3.76 Kp=9
+ Rs=64m Ksubthres=0.15 Mtriode=0.2 Rd=0.25 Lambda=4m
+ Bex=-2.3 Vtotc=+6m Tksubthres1=4m Trs1=3.5m Trd1=5m
+ Cgdmax=1.6n Cgdmin=30p a=0.5 Cgs=1.4n Cjo=1n
+ m=0.4 Vj=0.75 N=7 Is=10u Eg=3.15 Rb=50m Trb1=0
+ Vds=-200 Ron=0.5 Qg=44nC mfg=VishIH1907)
I then ran the bode plot as is, without changing anything else. It gave strange results, with a phasemargin of only ~39°.
Went backwards in the thread and found dado's latest zip file with schematic and models. It sims fine with expected results.
Then I changed the schematic from dado's zipfile to use Ian Hegglun's models, and I get the above strange behavior.
My impression was that the models from Hegglun and Cordell are all good, but they clearly behave differently.
Is there a consensus on which models to use for IRFP 240/9240 ?
Regards
Jørgen
Sorry if this has been discussed before, could not find anything on it.
I have very much enjoyed dado's 200W CFA thread and downloaded the .asc file from post #1582 assuming it is the newest.
I did not have the IRFP240Ckst/IRFP240Ckst models handy, so instead I used the following models from Ian Hegglun.
*VDMOS with subthreshold (c) Ian Hegglun
.model IRFP240h VDMOS (Rg=17 Vto=4.0 Kp=6 Lambda=3m
+ Rs=40m Ksubthres=0.15 Mtriode=0.35 Rd=0.15
+ Bex=-2.3 Vtotc=-6m Tksubthres1=4m Trs1=3.5m Trd1=5m
+ Cgdmax=1.3n Cgdmin=10p a=0.35 Cgs=1.2n Cjo=1n
+ m=0.4 VJ=0.75 IS=10n N=1.5 Eg=1.1 Rb=10m Trb1=3m
+ Vds=200 Ron=0.15 Qg=45nC mfg=VishIH1907)
*VDMOS with subthreshold (c) Ian Hegglun
.model IRFP9240h VDMOS (pchan Rg=9 Vto=-3.76 Kp=9
+ Rs=64m Ksubthres=0.15 Mtriode=0.2 Rd=0.25 Lambda=4m
+ Bex=-2.3 Vtotc=+6m Tksubthres1=4m Trs1=3.5m Trd1=5m
+ Cgdmax=1.6n Cgdmin=30p a=0.5 Cgs=1.4n Cjo=1n
+ m=0.4 Vj=0.75 N=7 Is=10u Eg=3.15 Rb=50m Trb1=0
+ Vds=-200 Ron=0.5 Qg=44nC mfg=VishIH1907)
I then ran the bode plot as is, without changing anything else. It gave strange results, with a phasemargin of only ~39°.
Went backwards in the thread and found dado's latest zip file with schematic and models. It sims fine with expected results.
Then I changed the schematic from dado's zipfile to use Ian Hegglun's models, and I get the above strange behavior.
My impression was that the models from Hegglun and Cordell are all good, but they clearly behave differently.
Is there a consensus on which models to use for IRFP 240/9240 ?
Regards
Jørgen
Attachments
TDA2030 BTL with output transistors
Hi!
I am trying to build a mono amp based on bridged TDA2030 (2050) with output transistors TIP35/TIP36. I found a schematic that seems ok here https://320volt.com/wp-content/uploads/2009/03/200watt-anfi-devre-semasi.png and I copied it in EasyEDA to build a PCB. But I am not sure if I've done it correctly, particulary the transistors connections.
Here is my schematic:
and here is an autorouted PCB I want to use as a reference:
BUT with a similar schematic (I don't have the link, but is quite similar, without any input filterning and decoupling), the PCB looks entirely different, regarding the connections of the transistors (BDA249/TIP35C and BDA250/TIP36C):
I used schematics/footprints contributed by other users for TDA2030 and TIP35C/TIP36C and I din't change anything in pins/footprints.
Is my procedure that wrong?
TDA2030 schematic and footprint:
TIP35C schematic and footprint:
TIP36C schematic and footprint:
I am trying to build a mono amp based on bridged TDA2030 (2050) with output transistors TIP35/TIP36. I found a schematic that seems ok here https://320volt.com/wp-content/uploads/2009/03/200watt-anfi-devre-semasi.png and I copied it in EasyEDA to build a PCB. But I am not sure if I've done it correctly, particulary the transistors connections.
Here is my schematic:
and here is an autorouted PCB I want to use as a reference:
BUT with a similar schematic (I don't have the link, but is quite similar, without any input filterning and decoupling), the PCB looks entirely different, regarding the connections of the transistors (BDA249/TIP35C and BDA250/TIP36C):
I used schematics/footprints contributed by other users for TDA2030 and TIP35C/TIP36C and I din't change anything in pins/footprints.
Is my procedure that wrong?
TDA2030 schematic and footprint:
TIP35C schematic and footprint:
TIP36C schematic and footprint:
Attachments
Room DSP Discovery
- By EliGuy
- Room Acoustics & Mods
- 10 Replies
Hi, I've been playing around with the DEQX for about a year now and
I've tried many speakers and combinations of measurements/corrections.
While I'm not completely thrilled with the DAC capabilities of the Hd Express II
I think that the room correction capabilities cannot be ignored.
So I wanted to share something I tried that I think sounds really good and also ask a question about it.
The usual method to correct the drivers (in this case Kef LS50's) with open baffle 15" woofers below)
is to measure each driver from up close and then correct and only then do eq corrections on the room.
Well I decided to try something different this time and I was pleasantly surprised. 🙂
I wired BOTH speakers (highs/lows separately) to the same amps so that both speakers
were playing in mono. Then I did my measurements from the listening position and then cut and pasted
the one side side to the the other so that both speakers are using the same FIR filter. I then separated
the drivers so they each get their own amp. And then I took the average response on both speakers playing simultaneously
and applied EQ.
And well it does sound really good. Im wondering if tehre's any explanation for this? Only thing I could think of is that using the same filter sort of 'averages' out the response in the room better.
Anyways there it is, hopefully someone more knowledgable can chime in on whether this is a good idea or a bad one.
I've tried many speakers and combinations of measurements/corrections.
While I'm not completely thrilled with the DAC capabilities of the Hd Express II
I think that the room correction capabilities cannot be ignored.
So I wanted to share something I tried that I think sounds really good and also ask a question about it.
The usual method to correct the drivers (in this case Kef LS50's) with open baffle 15" woofers below)
is to measure each driver from up close and then correct and only then do eq corrections on the room.
Well I decided to try something different this time and I was pleasantly surprised. 🙂
I wired BOTH speakers (highs/lows separately) to the same amps so that both speakers
were playing in mono. Then I did my measurements from the listening position and then cut and pasted
the one side side to the the other so that both speakers are using the same FIR filter. I then separated
the drivers so they each get their own amp. And then I took the average response on both speakers playing simultaneously
and applied EQ.
And well it does sound really good. Im wondering if tehre's any explanation for this? Only thing I could think of is that using the same filter sort of 'averages' out the response in the room better.
Anyways there it is, hopefully someone more knowledgable can chime in on whether this is a good idea or a bad one.
FS:Subbu DAC V3
ES9023,just unmonted it from the case.
It comes with the original PSU and the JG Buffer.
All that is neded is a PSU for the buffer.
150 Euro or 180$ + shipping.
It comes with the original PSU and the JG Buffer.
All that is neded is a PSU for the buffer.
150 Euro or 180$ + shipping.
Attachments
6v6 Musical Machine - Clipping & power expectations
- By paultheg33k
- Tubes / Valves
- 29 Replies
