EMU 1616M uses 4398 but only 6 channels + SPIF
Old 1818M has 8 channels out with 4398
I am using 1616M now and would love to see a modification decription
Direct out
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http://www.diyaudio.com/forums/digital-line-level/137976-experience-diy-dac-351.html#post2387730
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Search "direct out mod". I have been doing them for 5 years on the Behringer DCX/ DEQ. It is very simple. Any of the AKM or CS dac chips have built in I/V conversion and gain stages which output a signal that is ready to drive a volume control or amp except for the need to block 2.5v dc that is riding on the outputs. Simply grab the signal at a ribbon cable or the next component after the dac chip and block the dc with a high quality cap such as the Dayton Foil or, a good transformer such as the Jensen JT-11- EMCF or FMCF. I send this straight out. Some guys add a simple rc filter to reduce the ultrasonic noise. This thread turned into some good discussion of transformers for a direct out mod.EMU 1616M uses 4398 but only 6 channels + SPIF
Old 1818M has 8 channels out with 4398
I am using 1616M now and would love to see a modification description
.
http://www.diyaudio.com/forums/digital-line-level/137976-experience-diy-dac-351.html#post2387730
.
Two soundcards will have different clocks. Big problem!
In Linux world this has been solved by netjack tools.
-Fist start jackd for master soundcard.
-Add second soundcard as slave with alsa_in/alsa_out commands
-Brutefir is used for xover. In brutefir config file, instead “alsa_pcm”, the name of the slave jackd devices are “jack_pcm”:
Alsa_in/alsa_out tries to resample the slave output stream in an attempt to compensate for drift between the two clocks. Depending on quality/latency parameters, this may computationally huge task. In my case, it is bigger than brutefir process itself. And I have only 15 xover filters in my brutefir setup.
I don’t know if musiland soundcards are supported by alsa?
In Linux world this has been solved by netjack tools.
-Fist start jackd for master soundcard.
-Add second soundcard as slave with alsa_in/alsa_out commands
-Brutefir is used for xover. In brutefir config file, instead “alsa_pcm”, the name of the slave jackd devices are “jack_pcm”:
Alsa_in/alsa_out tries to resample the slave output stream in an attempt to compensate for drift between the two clocks. Depending on quality/latency parameters, this may computationally huge task. In my case, it is bigger than brutefir process itself. And I have only 15 xover filters in my brutefir setup.
I don’t know if musiland soundcards are supported by alsa?
Digital files
Did you come up with a good routing scheme in order to play wave files from the same hard drive that is running the crossover?
Did you come up with a good routing scheme in order to play wave files from the same hard drive that is running the crossover?
Yes. There are two approaches what have been tested on my XP machine. The first is the use of a third party software called Virtural Audio Cable . This allows the digital signal form the media played to be sent directly to the Ultimate Equalizer (UE). However, when doing this the processing must remain in the native 16 bit wave format. The other approach, which I have not thoroughly tested yet, is which the use two sound cards and route the analog output of the the medial player sound card to the analog input of the cards used with the UE. The down side is the additional D>A, A>D conversions but the up side is that the UE can be operated is 24 bit/48k mode. On XP machines VAC V3 must be used. There is also a VAC V4 for which some success on W7 PCs has been reported, but it has not been fully tested.
Impulse?
Have you posted an impulse or step response of a finished system using Bodzio UE? Maybe even run a new set up with a smaller speaker system elevated and summed at a 1 meter mic location and then take a normal, windowed impulse of the finished system at the same location so that we can have something to look at which is comparable to the commonly seen measurements of other passive cross speaker systems. Show a comparison of Ultimate Equalizer versus passive cross?
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I posted that some time ago in a different thread I guess. Here it is again:
An externally hosted image should be here but it was not working when we last tested it.
This was for a 2-way system. Tahe top trace in each case is the pulse input to the UE. The lower trace is the maesured acoustic pulse generated by the speraker. The top set is for a UE linear phase system where the acoustic response has LR4 amplitude. The lower set is for the same speaker, again using the UE with LR4 amplitude but with the phase linearization option turned off.
Clean
The processing in the Bodzio UE program looks very clean. I have seen screen shots of pulses running through other pc cross programs that show severe digital ringing artifacts. How is the processor demand? Can I run a four way on a laptop?I posted that some time ago in a different thread I guess. Here it is again:
This was for a 2-way system. The top trace in each case is the pulse input to the UE. The lower trace is the measured acoustic pulse generated by the speaker. The top set is for a UE linear phase system where the acoustic response has LR4 amplitude. The lower set is for the same speaker, again using the UE with LR4 amplitude but with the phase linearization option turned off.
Processor demand depends on what processor speed is. There is not problem with a P4, 2.8G Hz processor. Less that 40% load for a 4 way. Laptops are a different issue. The problem is sound cards. USB connections don't see to work well. It has been reported that the UE works with a Firewire sound card but that was on a desktop. SO I really don't know hat to tell you about running on a laptop.
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Firewire 410
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http://www.interdomain.net.au/~bodzio/UE_Manual.zip
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I have an M-Audio firewire 410 so hopefully that will work. I just read through the excellent reference manual for Ultimate Equalizer and was happy to see that the measurement process is very much streamlined and has been laid out step by step with screen shots. It can't get much easier than that. Anyone that thought in the past that Sound Easy was too hard to learn needs to check out the tutorial to see how easy it has become in Ultimate Equalizer.
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http://www.interdomain.net.au/~bodzio/UE_Manual.zip
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Please Help A Newbie Out
I read through the thread (well a good chunk of it, it is sooo long!) as well as the DRC Wiki. I would like to do a simple PC XO with DRC, before I even start with XO, I am trying to get the whole procedure tested on my system, unfortunately, I am not getting the expected results. So please help me identify my mistakes so I can get onto the next step.
I am running Win XP, SP3, Foobar -> Convolve or MPC -> Convolver after recording the sweep for my speakers, and generating the correction files with DRC. Plug these files into the convolution plug-ins and out comes garbled sound, echoy and phasey... I checked the record levels - they are not distorted and channels are not mixed up, etc. What else should I be paying attention to?
Thanks in advance,
Jaz
I read through the thread (well a good chunk of it, it is sooo long!) as well as the DRC Wiki. I would like to do a simple PC XO with DRC, before I even start with XO, I am trying to get the whole procedure tested on my system, unfortunately, I am not getting the expected results. So please help me identify my mistakes so I can get onto the next step.
I am running Win XP, SP3, Foobar -> Convolve or MPC -> Convolver after recording the sweep for my speakers, and generating the correction files with DRC. Plug these files into the convolution plug-ins and out comes garbled sound, echoy and phasey... I checked the record levels - they are not distorted and channels are not mixed up, etc. What else should I be paying attention to?
Thanks in advance,
Jaz
Never mind! Forgot to run lsconv to get the impulse responses, duh!
But I do have a question - does anyone know of a multi-channel convolver VST for the MAC?
Jaz
But I do have a question - does anyone know of a multi-channel convolver VST for the MAC?
Jaz
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