Markus,
I would just convert to unbalanced, as I don't see how you could possible get any benefit from it to begin with, which is less noise w/long cable runs.
It could even help you, as I believe the conversion will reduce gain by 6 dB (half the signal voltage from using only one polarity).
I would just convert to unbalanced, as I don't see how you could possible get any benefit from it to begin with, which is less noise w/long cable runs.
It could even help you, as I believe the conversion will reduce gain by 6 dB (half the signal voltage from using only one polarity).
markus76 said:
I did not follow the Yahoo group - only some of the diyaudio discussion. And yes I read that paper before. But do you really need 96 kHz oversampling and does your AV provide it? Is it worth to throw away a full digital connection? (No offence, just real interest since I never tried it)
I do not see how you are feeding a multichannel signal at the moment. Do you need more than the stereo signal to get your sub signals?
BTW: I´m only talking about your present stereo setup. If this is discussing your next steps for multichannel right now, please forget about my rant.
I agree with Rudolf. I'm only supplying my DEQ2496 with a locked in 44.1khz and it is very clean there is no digital clipping. It is infinitely better than the lack-lustre behringer analogue in/out. My main program material is CDs converted to flac though (i'm a music nut). I think Markus is more into DVD?
That isn't going to help Marcus though as he will still need to use the analogue output for his mains unless he deals with issue of the split balanced or "Y" cable.
Markus what receiver is it? I think the success of the Y cable will depend on what op-amps are in the output buffer driving the connection. If it has something like a NE5322 you might find that it works fine. The NE5322 is renowned for driving very long cables, so maybe it will drive a split cable too.
If the receiver is something cheap it will probably have the infamous JRC4558 in it. That should still work.
With your amp try the Jaycar balanced/unbalanced converter kit, it's cheap and always handy to have something like that around if it doesn't turn out to be optimal.
http://www.jaycar.com
Also, you could always start a topic in diyaudio about your dilema. There are heaps of high power electronics nerds in here who deal with issues like that blindfolded before breakfast......
😀
col.
That isn't going to help Marcus though as he will still need to use the analogue output for his mains unless he deals with issue of the split balanced or "Y" cable.
Markus what receiver is it? I think the success of the Y cable will depend on what op-amps are in the output buffer driving the connection. If it has something like a NE5322 you might find that it works fine. The NE5322 is renowned for driving very long cables, so maybe it will drive a split cable too.
If the receiver is something cheap it will probably have the infamous JRC4558 in it. That should still work.
With your amp try the Jaycar balanced/unbalanced converter kit, it's cheap and always handy to have something like that around if it doesn't turn out to be optimal.
http://www.jaycar.com
Also, you could always start a topic in diyaudio about your dilema. There are heaps of high power electronics nerds in here who deal with issues like that blindfolded before breakfast......
😀
col.
Guys, DCX and DEQ both use a buggy chip. This has nothing to do with clipping. It doesn't matter if you feed an upsampled signal or plain 44.1 kHz. Sometimes the effect is subtle but sometimes people (like me) get the "frying egg" sound.
I need to use the analog in because my setup is used for watching TV and DVD too. So you have to feed these multichannel signals to some sort of switcher and Dolby downmixer commonly known as an AV receiver. NO AV receiver that I know is capable of downmixing a digital multichannel signal into a DIGITAL stereo signal (there is a unit from Dolby but it costs just too much). But this is what I would need to feed the DCX digitally. The DCX accepts only digital PCM 2-channel signals.
Spoke to a fellow mixing engineer today and he said that if all devices are +4 dBu balanced then connecting them to a simple Y-cable works just fine. I'll only get a 3 dB lower level.
Best, Markus
I need to use the analog in because my setup is used for watching TV and DVD too. So you have to feed these multichannel signals to some sort of switcher and Dolby downmixer commonly known as an AV receiver. NO AV receiver that I know is capable of downmixing a digital multichannel signal into a DIGITAL stereo signal (there is a unit from Dolby but it costs just too much). But this is what I would need to feed the DCX digitally. The DCX accepts only digital PCM 2-channel signals.
Spoke to a fellow mixing engineer today and he said that if all devices are +4 dBu balanced then connecting them to a simple Y-cable works just fine. I'll only get a 3 dB lower level.
Best, Markus
Go for the Y cable mod! All you have to do is get the mains off of the DCX. Otherwise your going to end up in selectronic.fr land.
col.
col.
Marcus,
The "frying egg" sound was caused in my DCX by the flat cable connecting the circuit boards. I far as I can tell it's the result of thermal recycling causing the cable to move and the connectors coming a bit loose. I noticed that the lump of silicon stuff on the bottom of the case that should be holding the cable still was no longer stuck to it.
I "solved" the problem by pushing the connectors back down and leaving the device on all the time. If you do this while the device is on for heaven's sake turn the levels down to nothing because otherwise the transient spike could really hurt your tweeter.
Quality control with this device is not great. When I first got it there was an extremely low frequency rumble which I found was the result of the XLR connectors not being screwed tightly enough to the case.
Some DEQ devices will keep cycling their on process because the underside of the circuit board shorts to the case. I slid a piece of thin cardboard such as is used for greeting cards between it and the case. It's worked fine ever since.
I'm going to have lots of free time after next week to get my speakers finished.
The "frying egg" sound was caused in my DCX by the flat cable connecting the circuit boards. I far as I can tell it's the result of thermal recycling causing the cable to move and the connectors coming a bit loose. I noticed that the lump of silicon stuff on the bottom of the case that should be holding the cable still was no longer stuck to it.
I "solved" the problem by pushing the connectors back down and leaving the device on all the time. If you do this while the device is on for heaven's sake turn the levels down to nothing because otherwise the transient spike could really hurt your tweeter.
Quality control with this device is not great. When I first got it there was an extremely low frequency rumble which I found was the result of the XLR connectors not being screwed tightly enough to the case.
Some DEQ devices will keep cycling their on process because the underside of the circuit board shorts to the case. I slid a piece of thin cardboard such as is used for greeting cards between it and the case. It's worked fine ever since.
I'm going to have lots of free time after next week to get my speakers finished.
mult
Markus,
You can passively split any line level output, whether bal/unbal without any problem.
You do this all the time in the studio when using an analog patch bay and patch into a "mult" which after all is a passive Y. Remember that if you are dealing with balanced connections, you might want to telescope the shields, that is connect the grounds only to one side, leave the other side "telescoping" or not connected. You can't do this with unbalanced lines of course. Solder your own.
I passively split the outs (unbal) from my preamp to feed several different amps.
Does your AV rcvr have an effects or old fashioned tape loop? Just curious, I guess that won't help because you are using the DCX....
Note that signal level is independent of bal/unbal. It's just that balanced systems are usually aimed at +4 signal level users.
Markus,
You can passively split any line level output, whether bal/unbal without any problem.
You do this all the time in the studio when using an analog patch bay and patch into a "mult" which after all is a passive Y. Remember that if you are dealing with balanced connections, you might want to telescope the shields, that is connect the grounds only to one side, leave the other side "telescoping" or not connected. You can't do this with unbalanced lines of course. Solder your own.
I passively split the outs (unbal) from my preamp to feed several different amps.
Does your AV rcvr have an effects or old fashioned tape loop? Just curious, I guess that won't help because you are using the DCX....
Note that signal level is independent of bal/unbal. It's just that balanced systems are usually aimed at +4 signal level users.
markus76 said:
In this setup, the subs will be getting a signal delayed by the digital unit, compared to the mains. I don't know if that latency will be significant in the passband, but I probably would not do it that way.
New data - this is what it looks like with no subs at all:
With subs:
Settings:
Sub 1: 5.7dB, LP 40Hz/12dB But.
Sub 2: -15dB, LP 50Hz/12dB But., Phase inverted
Sub 3: -15dB, LP 40Hz/12dB But.
Earl, any chance to smooth 100-200 Hz without the subs locations becoming audible?
Best, Markus
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With subs:
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Settings:
Sub 1: 5.7dB, LP 40Hz/12dB But.
Sub 2: -15dB, LP 50Hz/12dB But., Phase inverted
Sub 3: -15dB, LP 40Hz/12dB But.
Earl, any chance to smooth 100-200 Hz without the subs locations becoming audible?
Best, Markus
Wow, that is some very impressive result. I wish my room/speakers have that sort of measurement result.
My gated measurement shows that the response of my system is ruler flat, but far field measurement shows over 6dB dip between 200Hz to 500Hz (most detrimental), 9dB dip at 110Hz to 140Hz, 6dB peak at 80Hz. It seems to me that the only solution is to change room dimensions.
My gated measurement shows that the response of my system is ruler flat, but far field measurement shows over 6dB dip between 200Hz to 500Hz (most detrimental), 9dB dip at 110Hz to 140Hz, 6dB peak at 80Hz. It seems to me that the only solution is to change room dimensions.
markus76 said:
Markus
How many spatial averged locations is this data? What kind of frequency smoothing is applied? The reason that I ask is because above about 100 Hz and certainly by 200 Hz the sound field is becoming statistical and the peaks and dips are to be expected as they are within the sampling errors inherent in all room measurements.
There's only one location shown - don't need to have more than one seat optimized. There's no smoothing applied. Resolution is better than 1 Hz.
Best, Markus
Best, Markus
Have you tried increasing your LP to 85Hz, or higher, depending on the rolloff? Maybe that will fill in those nulls, or reduce the peaks in that range.
Maybe try it....
Maybe try it....
markus76 said:
Markus
You don't seem to understand acoustic measurements. If thats only 1 location - no spatial averaging - and no frequency averaging, then the peaks and dips above 100 Hz are meaningless. You need to read my book or look up "Statistical Acoustics". There are also several papers at AES about this. But the point is that without some spatial and frequency averaging you are looking rather meaningless data - your data is about +- 6 dB from the "real" response. And It will likely change each time you measured it.
? My data is a realtime FFT averaging 10 measurements - sorry didn't write that earlier. It doesn't vary. Here are two measurements that I made. The setup was completely rebuilt between measurements.
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An externally hosted image should be here but it was not working when we last tested it.
Markus
Move the mic about 6 inches to where the "other" ear might be and rerun. You will see that the response above about 100 Hz. will probably be very different. Which one is correct? Neither. Then move it a couple more inches back, or forward. All different.
You really need to read up on frequncy response measurements in small rooms.
I cannot comment on aspects of things that I expect to see, such as the "rough" response from 100 to 200 Hz. I expect to see this kind of roughness and its not real in the sense that it is hyper-sensitive to location. Numerous studies have shown this and how you cannot use single point microphone measurements for assesment of sound fields in the "statistical" region. Thats why its called the statistical region. Each measurement is a sample of the field with a +-6 dB standard deviation.
Move the mic about 6 inches to where the "other" ear might be and rerun. You will see that the response above about 100 Hz. will probably be very different. Which one is correct? Neither. Then move it a couple more inches back, or forward. All different.
You really need to read up on frequncy response measurements in small rooms.
I cannot comment on aspects of things that I expect to see, such as the "rough" response from 100 to 200 Hz. I expect to see this kind of roughness and its not real in the sense that it is hyper-sensitive to location. Numerous studies have shown this and how you cannot use single point microphone measurements for assesment of sound fields in the "statistical" region. Thats why its called the statistical region. Each measurement is a sample of the field with a +-6 dB standard deviation.
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