Using JRiver convolver for external sources

[Ofer]

Thanks, I am. But it is a somewhat crooked workaround. Changing the Asio spdif to a wdm recording device locks it at 48khz/24bit so I can't play anything in it's original bitrate. I think it also intruduces latency as there are lipsync issuses even with only Frequency correction filters. Having said all that with the right filters the sound improves dramticly and I am just starting to explore the options. Overall creating filters and crossovers in AL is rather intuative. The rest of computer DSP is far from it.

 

[mccarty350]

I would agree on all counts. I ended up hard coding my bitrate at 96/24. Only again from being used to that setting from MiniDSP then out of anything I think I can hear above 48khz.

So having generated quite a few recent convolvers here are some things I noticed:

1. You always want to use total time domain correction and if you're fully active do a per driver correction on top of that.
2. Minimum phase and linear phase seems to be case by case, I don't feel that I prefer one over the other strongly.
3. Here is something that matters far more than phase it appears - your crossover slope in the speaker setup. I tried generating different convolvers over the holiday weekend and tried 2, 1.5, 1, .8, .6, 5, .2, and .1 octave crossovers. You can set different setups up and then use the same measurement with each setup using a different octave slope.
4. The number of taps have diminishing returns, the difference between 64k and 113k taps when you're running at 96khz sample rate doesn't seem to be audible or at least is not dramatic enough for me to detect.

In this case on my setup I felt that a .5 octave steep as hell crossover or even a .2 sound great. .1 sounded to my ears slightly processed. The gentler slopes of 2 and 1 made the sound more diffuse and airy but I felt I lost dynamics, etc.

Let me know your experience! I have a ton of curves if there is a way to post some for folks to try. B&K, Harman, etc. etc.

 

[Ofer]

Hi @mccarty350, so as I still struggle with DSP issues and this is the living room, my window for experimentation is rather small. Still a few things I did notice. My overall felling is still that I don't really know what I'm doing and I'm shooting in the dark most of the time. I will refer to your points.
1. True. TTD sounds the best. One obvious thing is phantom center. It is much more, well centered. In the miniDSP+REW corrections it was diffused and changing according to frequency. Now it is dead centered. The whole sound is more natural and the speakers tend to disappear.
2. Yet to experiment. All are minimum phase.
3. Agree. My system is 3way active with sub in the center. The sub is a PA sub, the idea behind PA subs is different to HT subs. PA subs have high SPL but limited bandwidth, most music ends at ~40hz (the low E of the base). So the sub starts to drop at ~35hz, in fact my woofers go lower then the sub. In the miniDSP the best set up was to have the sub HPF@35hz and LPF @100hz. Trying to do that in AL by setting it as a 2 driver sub with both hpf and lpf was a disaster. it sounded muffled. Using long slopes 3 or 4 octave for the sub lpf and the woofers hpf and enabling overlap sounds great. On the midrange/tweeter front I have tried steep slopes of 1 octave but it didn't sound so good.
4. Haven't tried playing with taps.
One thing that is needed is to experiment with TTD windows and see how in effects the step response. Maybe it's time to go back to Mitch's book and walk-through. BTW how did you try Herman curves etc. Is there a way to import them from text files like in REW. Another thing left to do is measure the system after applying convolution. As far as I understand it is not possible in AL, only simulation. In REW it now requires asio and loopback and frankly I can't deal with anymore of this now.
To sum it up for now, we have a saying were I come from, maybe there is an equivalent in English "The whole is larger then the sum of it's parts (or components)". With AL it feels like the audio system is not an assortment of DACs, amps, speakers etc but a whole system playing in unison.

 

[mccarty350]

Honestly, most of us are but even though I feel like I'm not incredibly skilled I'm producing incredible results. It does not take incredible knowledge to get excellent results. Could someone that is an expert get more out of it? Sure. But it's great even without that.

1. It's insane how perfect the image gets centered. I've never been able to get result this good from tweaking balance, etc. etc.
2. To try linear phase, go to your target, hit the radio button that says linear in the target creater, save it as a different file name that says linear in the name just to keep track. Then in your filter make sure you uncheck minimum phase in the correction procedure. Give it a shot. Super crisp and clean sound, if you don't have things set up well it can make things more gritty or overly revealing, under good circumstances it moves things even more into a state of improvement.
3. Did you try increasing the amount of time in your correction procedure at the bass frequency and then upping to more taps? That might help. I'm actually using slopes of .5. The wider the octaves the more diffuse the sound becomes, the tighter (.5) make it hit harder/sharper. Finding a sweet spot where it's got a good soundstage and is sharp/hits hard without being fatiguing is what I have focused on.
4. Taps is the granularity of the filter. The more taps the more the dsp is imposed, the less taps the more of the natural character of the speakers that comes through is my take.

I hit tons of forums and downloaded the .txt files for room curves for room eq wizard. Others I looked at and created manually based on what was shown visually in forums. If you want to pm me your email I'll send you a bunch of them that I've gathered.

 

[Ofer]

Thanks for the suggestions. Will try them all eventually.
3. If you mean playing with the size of the window then yes. Every change I have made was for the worse. Despite all my reading I still haven't figured out the logic of playing with the windows. I will try the taps though. I do use sharp XO filters between the midrange and tweeter to make it more sharp. Will also try between the woofer and the midrange. As for the sub/woofer, the wide xo serves to blend the woofer and the sub better.
As for the txt curves. I have a few from REW as well. My question is how do you use them in AL? in REW you can import the text file, can you do this in AL?