INTERACT FORUM
More => Old Versions => JRiver Media Center 18 for Windows => Topic started by: NickF on January 19, 2012, 05:02:22 pm
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This is new territory for me but I am really fascinated and want to give it a try. The trouble is, I don't think I completely understand the process for getting things set up. The one thing I have done is to order a measurement mic. I have gone for the MP-1r-KIT Measurement kit from Juice. Bernt has confirmed that these are individually calibrated. I now need to know what software I need to do the setup. Many products have been mentioned in this thread. Could one of you experienced guys outline the steps and what product is needed at each step?
Nick.
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Could one of you experienced guys outline the steps and what product is needed at each step?
You need MC plus a Digital Room Correction program.
The leading options are:
Audiolense http://www.juicehifi.com/index.html (http://www.juicehifi.com/index.html)
Acourate http://www.acourate.com/ (http://www.acourate.com/)
DRC http://drc-fir.sourceforge.net/ (http://drc-fir.sourceforge.net/)
Ultimate Equalizer http://www.bodziosoftware.com.au/ (http://www.bodziosoftware.com.au/)
Another solution, more of an automated equalizer:
Room EQ Wizard http://www.hometheatershack.com/roomeq/ (http://www.hometheatershack.com/roomeq/)
As far as I know they all achieve good results. They will not achieve the same results, even from the same measurement.
So your results may vary. The most important thing to remember is it all comes down to developing a target response for your system in your room that you like. So you end up depending on your ears.
The posts referenced here are a great read on how to get good results and combine your ears with verification measurements.
http://yabb.jriver.com/interact/index.php?topic=68828.msg463993#msg463993 (http://yabb.jriver.com/interact/index.php?topic=68828.msg463993#msg463993)
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This is new territory for me but I am really fascinated and want to give it a try. The trouble is, I don't think I completely understand the process for getting things set up. The one thing I have done is to order a measurement mic. I have gone for the MP-1r-KIT Measurement kit from Juice. Brent has confirmed that these are individually calibrated. I now need to know what software I need to do the setup. Many products have been mentioned in this thread. Could one of you experienced guys outline the steps and what product is needed at each step?
Nick.
I don't consider myself experienced on this, but some basic steps are:
1) You hook up your microphone to your PC and put it at your listening position / Lazy Boy or whatever
2) You start the software of your choice (Audiolense in your case I presume)
3) Perform a new measurement using the buttons on the right. The software will start a whooping sound (frequency sweep) from the very low to the very high frequencies through the speakers, while the expected sound pressure level is considered to be the 0db point.
This takes place once for each speaker. The microphone picks up the resulting sound as it reaches your listening position and the software records it.
4) Using the software, an analysis of the measurement is used to determine the equalisation required to improve the sound in your listening area. For example, if there is a dip of 6db at 2.3 KHz, this means that vocals may be sounding hollow.
Aiming to eliminate this anomally, a correction is required. In detail, a boost of 6db at 2.3 KHz, if our goal is to achieve 0db.
These infinite equalisations across the frequency spectrum can be viewed like a graph.This graph (with some limitations) is like an inverse mirror of the measurement graph.
The result is called a filter and this filter (a PC file actually) is saved on your PC.
5) Using MC17, you go to Player-> DSP Studio -> Convolution and check its box.
In Settings of the same pop up window, you select the filter file you have created, and from that point onward MC17 applies this correction to all music you play.
This is simplistic, but I hope you get a rough idea!
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There are also inifinite number of things you can do and traps to fall into when doing the measurement. Two nice things to consider, which I found important:
1) In a system with center channel, the mic will most probably be located directly in front of the speaker, with the sensor in your nose-tip location and yourself outside the room. The back of your chair will reflect significantly, and results impulse response which is (almost the) magnitude of the original signal. And it will be 1-3ms delayed. This should be avoided, as it does not reflect the room response when you are in the room. It also creates results in strange filters. Use an absorber to replace your body.
2) I have a very difficult room, and filters based on sweet spot only measurements tend to result in pre-ringing anomalies. Creating the filter based on multiple measurements in a 3D array around the sweet spot gives a tremedously better result. E.g. sweet spot + 30cm away in all directions (back/in front/left/right/above/below). Audiolense supports multiseat filters.
These are just some things found by experience, there are more.
Then you will need to spend some time finding the targer response of your taste. My preference for music is a 3dB slope from the bass roll-off to 20kHz. For movies, using LFE and two subs, the bass rolloff is at a lower frequency, and I use an almost 8dB linear slope.
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Thanks for the advice guys. I am looking forward to getting started but face a frustrating 2 week wait for the measurement mic to arrive!
A couple of questions:
Does the Audiolense created filter and convolution take account of delays due to speaker distance or is this done separately in "Room Correction" in MC?
I assume, from comments above, that I can set up a different filter for music and movies. Are these filters selected automatically and, if so, how?
Nick.
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Thanks for the advice guys. I am looking forward to getting started but face a frustrating 2 week wait for the measurement mic to arrive!
A couple of questions:
Does the Audiolense created filter and convolution take account of delays due to speaker distance or is this done separately in "Room Correction" in MC?
I assume, from comments above, that I can set up a different filter for music and movies. Are these filters selected automatically and, if so, how?
Nick.
AL will automatically correct for distance for the individual speakers. You can manually edit the correction in terms of ms delay if required. I have never found that necessary.
You can define different zones in MC. The convolver can be set up with individual filters for individual zones. This works really good!
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Do folks think it's OK to try to correct all dips and peaks with DRC, or is there a rule-of-thumb that says a dip or peak correction beyond a certain number dBs is better or must be done with room treatment? I think I get it that room treatments with DRC is optimal in general, but given the range of options it would be great for some tips on what kinds of correction can't be handled well with DRC and room treatment is essentially mandatory.
Related to that, I have a vague understanding that if a peak or dip is related to a room mode (resonant frequency of a room, generally in the bass region), then room treatment is the only way to address this, but knowing when a dip or peak is related to a room mode or not is beyond my current understanding.
thanks for any insight!
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I am no expert in this but I had assumed that this technique, as it is end-to-end, will compensate for frequency and time based deficiencies anywhere in the chain, whether amp, speakers or room. I guess where I am a bit nervous, not having mine set up yet, is that I may lose some of the "warmth" of my Tannoys. I suppose time will tell. But why, otherwise, wouldn't we want to correct for any deficiencies?
Nick.
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I just had a thought about steps in the process for setting this up. I am sure many of us have receivers of one type/brand or another, and many of these have mechanisms for compensating for room and speaker setup. My Denon certainly does. So we need to disable any setup of the receiver and have all of the compensation in MC Convolution.
Nick.
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Another thought - I shall need to go and have a lie down!! I have seen mention of taking measurements at mutiple positions in the listening room. The Audyssey system on my Denon supports this although I have never used it. How can this work? Surely, the system can only create a correct audio image for one point (the sweet spot) in the room. What is the thinking behind this and how can it work?
Nick.
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Another thought - I shall need to go and have a lie down!! I have seen mention of taking measurements at mutiple positions in the listening room. The Audyssey system on my Denon supports this although I have never used it. How can this work? Surely, the system can only create a correct audio image for one point (the sweet spot) in the room. What is the thinking behind this and how can it work?
Nick.
That was me. The short answer is that multiposition measurements do create one filter, but averages the correction.
A room will give slightly or very different response at different locations. My room is difficult, and even 30cm from the ideal sweet spot, bass response in particular is very different. Ideally, if you are the only one listening, you should make one measurement in the sweet spot and create the filter. In difficult rooms with lots of nodes and bass problems, single-point correction can create quite unwanted anomalies at and outside the sweet spot. In my difficult room, the DRC will then try and compensate a lot in the 50-80Hz range. The resulting filter is not good, and gives pre-ringing in the bass (sounds like swoooopp! for certain sounds, like kick drums). To avoid this, I had to 1) find the optimum placement of speakers and sofa, 2) create 1 (one) filter made from multiple measurements in a grid around ideal sweet spot. Step 2) is what is called multiseat (Audysseay does that, as do Audiolense), and is usually intended to create a filter that averages the correction such that several people at least have some kind of audio improvement. The recommendation is then to measure in a quite large area of the room. My step 2) involves measurements up to 1m from sweet spot, and works really well in sweet spot and for the two adjacent seats.
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Do folks think it's OK to try to correct all dips and peaks with DRC, or is there a rule-of-thumb that says a dip or peak correction beyond a certain number dBs is better or must be done with room treatment? I think I get it that room treatments with DRC is optimal in general, but given the range of options it would be great for some tips on what kinds of correction can't be handled well with DRC and room treatment is essentially mandatory.
DRC can not make magic. Narrow freq-band dips are not disturbing the perception of audio, and you should not worry much about these. A general rule of thumb would be to not try and increase dips more than around 6dB, even if I have seen people reporting good results with 9 and 12dB increases. A 6dB increase of a dip means you have lowered all other frequencies by 6dB, i.e. halved your power amplifier's headroom. Also, remember that the x dB increase has to be played by the speaker. You can get really unwanted results from your speaker if you force it to play 6, 9, 12 dB higher at one or several narrow dips.
Peaks is a different story. A typical room often have several dB peaks in the 40 to 50Hz range. This is easily (?) flattened out with DRC. In my room, this 40Hz peak is accompanied by a broad dip from 50-80Hz, which cannot fully be brought up in a 2.0 setup. With the immense power of two subs in the 7.1 setup, this is less of a problem, as the two subs can be placed such as to minimize the dip in the first place. And then DRC uses some head room of the subs to increase the response up to the target curve.
Related to that, I have a vague understanding that if a peak or dip is related to a room mode (resonant frequency of a room, generally in the bass region), then room treatment is the only way to address this, but knowing when a dip or peak is related to a room mode or not is beyond my current understanding.
Again, peaks can the lowered with good results. Dips, and especially when you are at or close to a null, should not be corrected with DRC. Despite the name, bass traps can be helpful in increasing bass response in a room, as they absorb sound waves that otherwise would be reflected and would cause negative interference.
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There are also inifinite number of things you can do and traps to fall into when doing the measurement. Two nice things to consider, which I found important:
1) In a system with center channel, the mic will most probably be located directly in front of the speaker, with the sensor in your nose-tip location and yourself outside the room. The back of your chair will reflect significantly, and results impulse response which is (almost the) magnitude of the original signal. And it will be 1-3ms delayed. This should be avoided, as it does not reflect the room response when you are in the room. It also creates results in strange filters. Use an absorber to replace your body.
I have four separate seats that are not massive in a big room (sound can travel under and between them). I get good results if I recline the center seats to avoid this reflection problem (measure where your head is when setting more upright).
I get even better results if I remove the seats and then average multiple measurements in the center seating area. This way I am correcting for mainly the lower frequency room boundary effects without irregular short seat surface reflections screwing up the measurements at higher frequencies. Those short reflections change totally if the seat is occupied, empty, reclined, etc. Removing that variable completely seems to work in my case. I don't hear the difference if someone sits next to me so I don't want to screw up the direct response of the speakers in an ineffective attempt to compensate for a seat reflection.
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Yippeee! My mic has arrived! It should be an interesting weekend getting my system set up.
Trumpetguy and hulkss, surely the measurements should be done with the listeners in their seats, otherwise you will be compensating for effects which won't exist in a real situation. You have left me somewhat confused. ?
Anyway, that sort of fine tuning is somewhere down the road. I'll start with the basics and get used to Convolution first.
Nick.
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surely the measurements should be done with the listeners in their seats, otherwise you will be compensating for effects which won't exist in a real situation.
You should attempt to correct for low frequency room effects (boundary reflections). At higher frequencies (above the Schroeder frequency) correct the direct sound from the speakers. Get anything away from the measurement mic that would cause a short distance reflection into it and screw up your measurements. People don't reflect much bass, they are too small.
Time windowing of your measurements is needed with a shorter window at high frequencies.
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Trumpetguy and hulkss, surely the measurements should be done with the listeners in their seats, otherwise you will be compensating for effects which won't exist in a real situation. You have left me somewhat confused. ?
I have seen that this topic can cause debate temperature to rise :) , but personally I do not have any strong reasoning for either/or. I did some early test in/out of the room, and it didn't make a noticable difference so I leave the room simply because this is simpler. I find it difficult to place the mic in the middle of my own head ;), that is one thing. Placing it just beside will cause reflections, causing impulses which will disturb high frequencies simply because of the close proximity to the mic. On the other hand, your body would absorb and reflect energy, an effect that could be relevant for your filter. You will need to figure this out yourself.
Anyway, that sort of fine tuning is somewhere down the road. I'll start with the basics and get used to Convolution first.
Wise strategy :)
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Please assume a live event in your living room, a band is playing nicely. The arriving sound will float around you, reflected by ears, shoulders and the body. Despite this influence on the sound I'm pretty sure you will here a perfect live event and be happy with the (assumed nice) sound. Because your brain has learned to interprete the influence of your body and your ears. It simply ignores it or even does not recognize it.
Now assume that you are able to measure with a microphone in the middle of your head. It will record all the influences of your body on the sound. A later correction will compensate for the influences. Thus the sound of a speaker reproducing perfectly the live event of the band will be altered by the correction. And you will hear a wrong sound !
IMO it is not correct to record a sweep sitting in the chair and with te microphone in the mouth or in the ears.
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IMO it is not correct to record a sweep sitting in the chair and with te microphone in the mouth or in the ears.
LOL, you could be locked up or severly hurt to put a microphone into your head, regardless of opening...
Thanks for the reasonable argument on why the room should be left empty during measurement.
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Well I'm on the verge of giving up already! I have downloaded both Acourate and Audiolense to try them. I rashly thought that I would get a 30 day trial of a functioning product, like we do with JRiver. No such luck. Audiolense produces a file but it contains no correction. What is the use of that? I opened Acourate and could make no sense of it. At least I was able to make progress with Audiolense so I decided to buy it - 250 Euros!! Now I can't even activate it. The product wants to send an email with the hardware id, whatever that is. My HTPC doesn't have an email client - why would it? So I have now spent 411 Euros and have got nowhere. I can't even post a message asking for help on the Audiolense Forum without the topic being approved by a moderator. What a mistake!!
It makes you realise what a brilliant organisation JRiver is.
Nick.
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Thanks, Nick, but I'm sure Audiolense will find a way to get you going.
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Nick,
I had the same issue installing Audiolense since the computer had no internet connection. Here's the instructions I received:
"Next step for you is to install Audiolense and generate a HardwareID for license activation. You can generate a hardware ID from the license menu or by running AudiolenseHID.exe that sits in the program folder."
Once you have that generated, send it to Audiolense and you'll receive your activation.
Good luck,
Bill
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Nick,
I had the same issue installing Audiolense since the computer had no internet connection. Here's the instructions I received:
"Next step for you is to install Audiolense and generate a HardwareID for license activation. You can generate a hardware ID from the license menu or by running AudiolenseHID.exe that sits in the program folder."
Once you have that generated, send it to Audiolense and you'll receive your activation.
Good luck,
Bill
Thanks Bill, I'll give that a try now.
Nick.
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I believe Audiolense trial is fully functional (once you get it going, and you will it is just not 100% intuitive...), but with a heavily attenuated filter. Some 10 or 20dB. Should make you able to evaluate it, but you will of course get limited dynamics.
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Well I'm on the verge of giving up already! I have downloaded both Acourate and Audiolense to try them.
Nick,
Don't give up, the rewards are too great. These two programs do work and produce excellent results, however, they are also capable of making a mess. You will need to do a lot of reading before you understand what you are actually doing. I can not listen to audio without digital correction now - it is amazing when you get it right.
Start reading here: http://yabb.jriver.com/interact/index.php?topic=68828.msg463993#msg463993 (http://yabb.jriver.com/interact/index.php?topic=68828.msg463993#msg463993)
and then some about room acoustics and technical digital filter stuff if you can handle it. Just developing a good target response is a fair amount of work and has a profound impact on results.
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Well I'm on the verge of giving up already! I have downloaded both Acourate and Audiolense to try them. I rashly thought that I would get a 30 day trial of a functioning product, like we do with JRiver. No such luck.
Unfortunately it is not possible to get filters that run as trial filters for 30 days. No chance that a filter stops working after a trial period. If you know such a solution then please tell me.
As a previous full working Acourate trial version also has been hacked in the past I have been forced to create just a "dummy" version that does not allow to create the final filters.
So now there is a free logsweep recorder (www.acourate/AcourateLSR2Setup.exe) and I offer to compute filters based on the measured pulse responses and to treat two stereo music tracks of your own choice. This allows you to listen to the filtered tracks and to compare the result with the original tracks or other correction products. This service is free of charge. Isn't that fair?
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Unfortunately it is not possible to get filters that run as trial filters for 30 days. No chance that a filter stops working after a trial period. If you know such a solution then please tell me.
As a previous full working Acourate trial version also has been hacked in the past I have been forced to create just a "dummy" version that does not allow to create the final filters.
So now there is a free logsweep recorder (www.acourate/AcourateLSR2Setup.exe) and I offer to compute filters based on the measured pulse responses and to treat two stereo music tracks of your own choice. This allows you to listen to the filtered tracks and to compare the result with the original tracks or other correction products. This service is free of charge. Isn't that fair?
Sorry Uli, I didn't mean to be rude, I just found myself a bit bewildered and disappointed. Yes, I appreciate the problems of time expiry of filter files. I didn't realise you offered this service. It does seem very reasonable. I may try your logsweep recorder as I get more experienced. Thanks.
Nick.
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Well, I'm up and running, or should I say walking!! :D It's a bit of a steep learning curve but I've made a start. Bernt was quick off the mark this morning with my activation file. Thanks Bernt.
I'm starting with stereo for my music. I have taken Trumpetguy's advice and set up a Zone for Movies and one for Music so I can use different filters. I can switch between these automatically with my iPad remote.
I have done my first set of measurements after much fiddling with mic mounting and positioning, connections to my RME sound card and adjustments to volume and pre-amp gain.
I created a target, probably one of the most critical steps, I think. I really like the sound of my Tannoy Berkleys. They are vintage speakers but have been re-coned and I have upgraded the crossovers with high spec components and they are superb. I have shaped the target to retain some of their natural behaviour.
I produced the correction file and loaded it into Convolver and...... it works!! Even this first attempt reveals a much sharper and more detailed sound stage with so much less clutter from the room reflections. I am now having a problem dragging myself away from listening to go back to tuning!
Thanks for all the help and advice so far.
Nick.
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I'm a firm believer in quality interconnects. So how important are the interconnects between mic and pre-amp and pre-amp and sound card? The mic to pre-amp is about four metres long so that could get quite expensive!
Nick.
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I'm a firm believer in quality interconnects.
Quality interconnects are important. I will give you an Engineer's recommendation, not Audiophile.
This is what I do:
1. Use all balanced interconnections. Read this: http://www.jensen-transformers.com/an/an003.pdf (http://www.jensen-transformers.com/an/an003.pdf) and this: http://www.rane.com/note110.html (http://www.rane.com/note110.html)
2. Use brand name professional cable. I use Mogami: http://www.mogamicable.com/additional/best_cable.php (http://www.mogamicable.com/additional/best_cable.php)
3. If you want to make your own cables to custom lengths, use Mogami with connectors from: http://www2.neutrik.com/us/en/audio/203_11/XLR_Cable_Connectors_group.aspx (http://www2.neutrik.com/us/en/audio/203_11/XLR_Cable_Connectors_group.aspx)
4. Use hardware that takes full advantage of balanced cables. I recommend this design developed by Bill Whitlock of Jensen Transformers: http://www.thatcorp.com/datashts/THAT_1200-Series_Datasheet.pdf (http://www.thatcorp.com/datashts/THAT_1200-Series_Datasheet.pdf)
I use two of these (shorter length) to connect my 16 channel DAC to the amps:
(http://www.markertek.com/productImage/HI-RES/25MA-XM-M05.JPG)
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hulkss, du you use the original breakout cables from Lynx, or are these some other brand? The original cables are probably splendid, but they seem kind of thin and unprotected (don't know how to put it).
A side question: I have unused input and digital connectors hanging which are not terminated. Should there be some end resistor or other type of termination of these?
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Well it's been an interesting day playing with this solution. There have been some distinct audible improvements in the stereo image of my music and improved clarity of very low frequencies. I have the CD 'This Fire' by Paula Cole and the first track, Tiger, has some incredible low frequencies. Without Convolution, you feel the bass but with Convolution, you can clearly hear the cadence and shape of the sound. But as I continued to listen, it became clear that there is a distorted edge to many sounds, a harsh, fuzziness. I have swapped back and forth between Convolution and the pure path through MC and there is no doubt which I prefer. Something is wrong with Convolution.
Tomorrow I will try ConvolverVST to see if that is any better but, at the moment, I couldn't live with Convolution. Sorry Guys.
Nick.
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hulkss, du you use the original breakout cables from Lynx, or are these some other brand?
A side question: I have unused input and digital connectors hanging which are not terminated. Should there be some end resistor or other type of termination of these?
I bought these cables: http://www.markertek.com/Cables/Audio-Cables/ADAT-Cables/TecNec-Cables-Connectors/25MA-XM-M05.xhtml?25MA-XM-M05 (http://www.markertek.com/Cables/Audio-Cables/ADAT-Cables/TecNec-Cables-Connectors/25MA-XM-M05.xhtml?25MA-XM-M05)
I'm sure the Lynx ones are fine.
Never seen anyone use xlr terminator caps, you can buy them though.
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Something is wrong with Convolution.
Nick,
More likely something is wrong with your filter. Poor filter = poor convolution result.
I hope you have a decent target response curve. Read this: https://docs.google.com/viewer?a=v&pid=explorer&chrome=true&srcid=0B97zTRsdcJTfY2U4ODhiZmUtNDEyNC00ZDcyLWEzZTAtMGJiODQ1ZTUxMGQ4&hl=en_US (https://docs.google.com/viewer?a=v&pid=explorer&chrome=true&srcid=0B97zTRsdcJTfY2U4ODhiZmUtNDEyNC00ZDcyLWEzZTAtMGJiODQ1ZTUxMGQ4&hl=en_US)
Note that the best sounding response (top line p24) has a full 10 dB decrease from 20 - 20kHz.
Depending on the power response (not frequency response) many loudspeaker systems need a "midrange compensation dip" of a few dB centered at 2 kHz to remove harshness. Keep in mind that the microphone and computer do not hear like your ear and brain, the target curve has to compensate. A flat target will sound very lame.
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Thanks, hulkss. It's very early days for me so I haven't reached any conclusions yet, just first impressions.
I spent quite some time investigating and shaping my target. It is 10dB down across the spectrum so I'm fairly sure I have enough digital headroom for the correction to avoid clipping. Is there any way to show whether MC is seeing or dealing with clipping?
I will take another look at the shape of, it today. The effect that I am hearing isn't due to an over-emphasis of mid-range frequencies. It is a distorting effect added by the process. It manifests as a fizz on the edge of purish mid-range sounds, particularly female vocals.
I will take some more measurements today, look more closely at the target and do some comparisons between different convolution engines.
Nick.
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Please note that the correction counts but not the target !
Now
correction = target - (interpreted) measurement
This means that depending on the interpretation of the measurement (like smoothing, frequency dependent windowing and other treatments) also a target must change to get an identical correct correction. It does not help to read articles as mentioned above because each correction algorithm has its own philosophy to interprete the measurement (indeed this also causes different targets and results).
So you should always study the correction curve. If it has a rising slope then you should not wonder about more brightness. If it is strongly falling you should not wonder about a dull sound.
Try to understand what the correction is changing, the correction curve show this clearly.
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Thanks for the advice, Uli. As a complete novice, I welcome all input.
I had assumed that the target, a component of the process over which the user has complete control, would be an important area to focus. Surely it is intended to drive the final behaviour of the solution. I assumed that the correction, created by the tool, not the user, is essentially a response to the target, which is what your formula says.
So you are saying study the correction and understand what effect it will have on the solution. Are you also saying that we should modify the correction if we think it is necessary? I am using Audiolense. Does it allow me to modify the correction?
Nick.
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So you are saying study the correction and understand what effect it will have on the solution.
Yes.
Are you also saying that we should modify the correction if we think it is necessary?
Yes.
I am using Audiolense. Does it allow me to modify the correction?
Yes, modify the target accordingly.
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Thanks for the advice, Uli. As a complete novice, I welcome all input.
I had assumed that the target, a component of the process over which the user has complete control, would be an important area to focus. Surely it is intended to drive the final behaviour of the solution. I assumed that the correction, created by the tool, not the user, is essentially a response to the target, which is what your formula says.
So you are saying study the correction and understand what effect it will have on the solution. Are you also saying that we should modify the correction if we think it is necessary? I am using Audiolense. Does it allow me to modify the correction?
Nick.
With AL you should also play around with window widths in the filtering section. Different windowing (number of cycles) give different correction results. You should make sure your correction curve does not have any nulls (or very sharp and deep dips).
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If your not doing digital XO, here'e the Audyssey Multeq Pro product in a VST plug-in.
http://www.ikmultimedia.com/arc/features/ (http://www.ikmultimedia.com/arc/features/)
I've used Multeq Pro with good results. Their "secret" is to combine a number of measurements to average out measurement anomalies. This also applies to DRC software solutions.
http://www.audyssey.com/sites/default/files/attachments/onesheet_howtomulteq.pdf (http://www.audyssey.com/sites/default/files/attachments/onesheet_howtomulteq.pdf)
You can combine multiple measurements in Audiolense and see the effect in the simulated response. Multiple measurements in different central seating locations will help with measurement confidence, understanding of room response issues, and should help to provide a better overall filter solution.
Another very good article, it's 21 years old and describes the digital correction challenges we are just beginning to work now in our home theaters.
http://www.audiosignal.co.uk/Resources/Digital_room_equalisation_A4.pdf (http://www.audiosignal.co.uk/Resources/Digital_room_equalisation_A4.pdf)
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I had assumed that the target, a component of the process over which the user has complete control, would be an important area to focus. Surely it is intended to drive the final behaviour of the solution.
Nick.
Nick, yes, the target is the key. I have found the following target from this excellent paper: http://www.bksv.com/doc/17-197.pdf to have the most natural timbre at the listening position. The target points are -0.5db at 200Hz, -3 at 2KHz and -6 at 20KHz.
(http://i1217.photobucket.com/albums/dd381/mitchatola/bandk.jpg)
Using REW http://www.hometheatershack.com/roomeq/ (awesome measurement software and fantastic support on their forum, just like JRiver!) I measured the following frequency response at the listening postion with the Audiolense filters loaded into JRiver's fantastic Convolution:
(http://i1217.photobucket.com/albums/dd381/mitchatola/jan29leftspeakerwithaudiolensedrc.jpg)
As hulkss has mentioned, a flat frequency response at the listening position will sound (etch a sketch) bright. The B&K target curve above produces the most natural timbre and soundstage at the listening position IMHO. I have tried many variants of this target and end up coming back to this one as being the most natural sounding to my ears. YMMV.
I suggest you head over to the Audiolense forum http://groups.google.com/group/audiolense as there are many tips being shared that you may find beneficial.
Happy Listening!
Mitch
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Very nice result. From 40 hz to 20kHz a 10 dB tilt is what your measurement shows and is almost identical to the paper I referenced by Harman.
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Very nice result. From 40 hz to 20kHz a 10 dB tilt is what your measurement shows and is almost identical to the paper I referenced by Harman:
hulkss, thanks. I had a look at the Harman paper you referenced: https://docs.google.com/viewer?a=v&pid=explorer&chrome=true&srcid=0B97zTRsdcJTfY2U4ODhiZmUtNDEyNC00ZDcyLWEzZTAtMGJiODQ1ZTUxMGQ4&hl=en_US Slide 24, the top curve:
(http://i1217.photobucket.com/albums/dd381/mitchatola/Harman.jpg)
And yes, it is almost identical - good correlation to 2 different papers showing the same preferred target response. Btw, combining the left and right responses = flat at 20Hz. So relative to the target, I am easily within +-3db from 20Hz to 20KHz. I am so satisfied with the response of my system, I am done. Just listening to music now. With Convolution built-in to JRiver, I try and feed everything through it - sounds great!
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Truly a nice (or impressive) frequency response. Would you care to post impulse response and/or ETC as well? Is your room acoustically treated, or do you do huge digital enhancements?
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Well I have made some progress. Firstly, thanks for all the support and encouragement!
The distortion - I suspected there was something going on within the solution to cause this so I decided to drop the whole target by 6 dB. By the way, thanks Mitchco, I have used your B&K target as the basis and like it. This improved the situation but the problem was finally resolved by turning off Clip Protection and, epecially, Normalise filter volume. The latter is a killer, it completely destroys the quality. With these corrections, the distortion is now gone. The problem is that it raises the noise floor by 6 dB, not that this is a real problem, but I now need to go back to the target and see what the minimum reduction needs to be. That will take a few iterations.
I note that hulkss has posted something on clipping within the Convolution path and that clip protection is currently applied at the end rather than during. Are you hearing something too? My view is that clip protection is attacking an effect, not the cause. You need to ensure that you never reach a clipping situation. However elegantly clip protection works, it is no different from AGC and that is to be avoided at all costs. The original recording fitted within the digital sample ceiling and you need to keep it there.
One thing about Audiolense which is confusing me at present is the shape of the correction filter. There are some dips in my room/speaker response which it is trying to correct but the correction waveform flattens out in several places as though it hits a max threshold for correction. I thought that dropping the target would help this. In theory, it should but it doesn't. Anyone know why this is? I can't find any option to adjust it. Is it a defect or is there some other reason?
Well my enthusiasm is back up and, hopefully, over the next few days, I will get my stereo solution right and I can move on to the surround correction for my movies.
Nick.
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There are some dips in my room/speaker response which it is trying to correct but the correction waveform flattens out in several places as though it hits a max threshold for correction. I thought that dropping the target would help this. In theory, it should but it doesn't. Anyone know why this is? I can't find any option to adjust it. Is it a defect or is there some other reason?
The amount of + correction in the Audiolense filters defaults to a limit of +6 dB. This can be changed in the correction procedure designer (maybe not in all software versions). This needs to be increased with caution.
Clipping is bad for sure. The JRiver clip protection is not bad though. I believe it will only mess with the audio if you are actually clipping. Something is off with the gain levels in DSP somewhere.
Normalize filter volume should be OK too. Maybe Matt can comment. I don't have issues with clipping or the convolver normalization. I was pointing out that I could see levels above 100% in the DSP chain yet I was not activating clip protection. Matt explained that the floating point processing allows this without problems. The signal has to be back down below 100% only at the last step in the chain.
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Truly a nice (or impressive) frequency response. Would you care to post impulse response and/or ETC as well? Is your room acoustically treated, or do you do huge digital enhancements?
ETC’s using Audiolense TDD: http://www.hometheatershack.com/forums/rew-forum/51865-thank-you-rew-2.html#post477314
Latest ETC, but using only frequency correction: http://i1217.photobucket.com/albums/dd381/mitchatola/jan29etcfrequencycorrection.jpg
What is interesting to note when you compare the TTD versus frequency correction only ETC’s, is that the TTD does a good job of supressing the early specular reflections.
Waterfall Full range: http://i1217.photobucket.com/albums/dd381/mitchatola/jan29waterfallfullrange.jpg
Waterfall 15Hz to 300Hz: http://i1217.photobucket.com/albums/dd381/mitchatola/jan29waterfall15to200hz.jpg
You can read the details, plus more graphs: http://yabb.jriver.com/interact/index.php?topic=68828.msg463993#msg463993
No acoustic treatment, save a heavy carpet between the listening position and speakers. Not because I don’t think acoustic treatment works, it does, and I have experience with acoustic treatment as can be seen here: http://www.computeraudiophile.com/blogs/Hear-music-way-it-was-intended-be-reproduced-part-5 The issue for me is cost. To really get the room right using acoustic treatment costs in the thousands…
I use 12db of correction, but could get away with 6db and get -3db at 30Hz. I have tried up to 18db of correction, but settled on 12.
Hope that helps.
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Again, the ETCs are just as impressive as your frequency responses. No impulses above -15dB before ~20ms. And all that without any particular acoustic treatment. I believe you should consider yourself lucky with your room. Or maybe luck has nothing to do with it?
With 12dB increase, do you experience any loss of dynamics? Audiolense has a default setting of max +6dB. It sure helps explaining the straight frequency response (in addition to the graph smoothing), but I would expect some really noticeable loss of headroom. I know I would in my room. With a room with narrow-band dips, and even having sufficient amplifier power, it will be a really challenging task for a speaker to play +12 and even +18dB louder in these narrow frequency ranges. How do you solve these things?
Only asking because you seem to have solved somthing that I have been struggling with.
BTW, I tested some target curve close to the B&K curve yesterday. Sounds very different from what I am used to, but it created a really pleasant, warm, still dynamic soundscape. Flat -0.5dB <200Hz, then sloping to -10dB at 20kHz. It is a real challenger to my old near-flat target.
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BTW, I tested some target curve close to the B&K curve yesterday. Sounds very different from what I am used to, but it created a really pleasant, warm, still dynamic soundscape. Flat -0.5dB <200Hz, then sloping to -10dB at 20kHz. It is a real challenger to my old near-flat target.
Can you post some pictures of the correction for your old target and the B&K target?
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The ETCs are good even before DRC - in fact I don't see much improvement but then, I'm no expert at all. So please forgive my ignorance :)
In regard to B&K house curve: I personally find it dull. With especially many classical recordings the atmosphere from the concert room disappears. As far as I've read in the white-paper from B&K the curve resembles the acoustics from a concert hall/larger music arena, which apparently makes it ideal for playback of close-up recordings.
While probably true, I nevertheless prefer a completely flat curve for most recordings (mind you, I listen to classical music where far-field recording isn't uncommon... as far as I know).
With movies I have no idea. I'm still trying to figure out if dvd/blu-ray discs are remastered with a flat frequency respons or is attenuated in the upper range to compensate for screen dampening and the distance from speaker to listeners in the cinema room.
Best regards,
Mikkel
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Can you post some pictures of the correction for your old target and the B&K target?
I can provide the correction curve, but I yet do not have any way to measure actual room response with filter activated. Is it the first you are requesting? If it is the latter, could you please describe how I could accomplish that?
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The ETCs are good even before DRC - in fact I don't see much improvement but then, I'm no expert at all. So please forgive my ignorance :)
You are probably right. DRC seem to give good results in most rooms, but also seems to give the little extra in already good listening environments. The already good ETC suggests this room is aready quite all right from the start.
In regard to B&K house curve: I personally find it dull. With especially many classical recordings the atmosphere from the concert room disappears. As far as I've read in the white-paper from B&K the curve resembles the acoustics from a concert hall/larger music arena, which apparently makes it ideal for playback of close-up recordings.
While probably true, I nevertheless prefer a completely flat curve for most recordings (mind you, I listen to classical music where far-field recording isn't uncommon... as far as I know).
I find my taste to be changing every now and then. I do not believe one curve is the ultimate answer for all recordings and situations. For rock and pop, B&K sounded really good. But I can imagine the curve not being quite as good for classical music. I listen to classical music a lot, and want to test it on large orchestral works and chamber music. Looking forward to that.
With movies I have no idea. I'm still trying to figure out if dvd/blu-ray discs are remastered with a flat frequency respons or is attenuated in the upper range to compensate for screen dampening and the distance from speaker to listeners in the cinema room.
I know what my preference is - flat curve on movies is a boring experience. With movies, I am not after what is correct, only what is entertaining. To get that, my experience is that using the ears is the best thing. I have been using something close to a pear shaped curve for a couple of years. An 8dB hump below 500Hz. Now I am going to test B&K. It may give more flesh to voices.
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I can provide the correction curve, but I yet do not have any way to measure actual room response with filter activated. Is it the first you are requesting? If it is the latter, could you please describe how I could accomplish that?
I have asked for the correction curve itself, but not for the corrected room response curve.
As stated in a previous message I advise to check for the correction (influenced by measurement and the defined target). If e.g. someone says that the B&K curve sounds dull he can easily see this in the correction curve.
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...Or maybe luck has nothing to do with it?
I designed it that way or tried to. I have high efficiency speakers http://i1217.photobucket.com/albums/dd381/mitchatola/barnettestylec.jpg At 100db SPL @ 1w/1m sensitivity, even with 12db correction, the sound is still quite dynamic.
Here is an overlay of the measured uncorrected (purple) and with DRC enabled (blue) frequency response at the listening position. http://i1217.photobucket.com/albums/dd381/mitchatola/jan29rawanddrcfreqrespoverlay.jpg I could get away with 6db of correction if I did not want to extend the low end down to 20Hz. I think 6db correction got me to 32Hz. No sign of overload yet and I like to listen reasonably loud.
I attribute this to the upfront work I did using a distance laser measure to accurately and precisely set up an equilateral triangle between listening position and speaker symmetry. This sets up DRC for success.
I have a ~10ft equilateral triangle in my 30’ x 15’6” x 8’ room. My speakers are set up along the long wall. http://i1217.photobucket.com/albums/dd381/mitchatola/Roomside.jpg With respect to the ETC and diffuse field, I tried to set up a reflection free zone from the speakers to the listening position. Which means the listening position needs to be away from the rear wall. http://i1217.photobucket.com/albums/dd381/mitchatola/carpetrear.jpg I am about 1/3 of the width of the room from the back wall and the speakers are 1/6 of the width of the room from the front wall.
I voiced the speakers to the room by listening to music (in the bass range) and moving each speaker slowly to and from the front wall until I found the spot where all the bass notes from the song sound the most even in loudness. Use the balance control to voice one speaker at a time. You know you don’t have the spot when some bass notes disappear (null) or are very loud (peak). Once I got them in the ballpark, I used REW to measure and verify and fine tune. After a 3 iterations, I was done. Here is the uncrorrected frequency response at the listening position. http://i1217.photobucket.com/albums/dd381/mitchatola/jan29leftspeakerrawfreqresp.jpg Pretty good response on its own given the constraints of my room setup.
How did I measure? As you saw in the rom pics, I have a good size table and couch. I moved those off to the side of the room and set up the calibrated omni mic (isolated from floor vibrations) at the listening position facing straight up and ear height. No multi-seat measures yet as it seems to sound good anywhere in the room.
Re: narrow peaks and dips. Here is the unsmoothed frequency response at the listening position with DRC. http://i1217.photobucket.com/albums/dd381/mitchatola/jan29drcrawfreqrespDRC.jpg I as understand it, the measurement software has more measuring resolution than our ears can hear. We can start discriminating frequency differences between 1/3 and 1/6 octave range. I could be wrong, but that is my understanding.
So when you see my 1/6 octave smoothed frequency response, it seems like every little narrow dip and peak was corrected. That is not the case as you can see in the raw response. But for our ears purpose, 1/6 octave smoothing is just outside our ears ability to differentiate. At least that is my understanding.
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I as understand it, the measurement software has more measuring resolution than our ears can hear. We can start discriminating frequency differences between 1/3 and 1/6 octave range. I could be wrong, but that is my understanding.
So when you see my 1/6 octave smoothed frequency response, it seems like every little narrow dip and peak was corrected. That is not the case as you can see in the raw response. But for our ears purpose, 1/6 octave smoothing is just outside our ears ability to differentiate. At least that is my understanding.
The pulse response underlying the frequency plot has a certain length. The frequency chart shows the behaviour over the full pulse length. Imagine the duration of the pulse response e.g. with 1 hour and you will get even more peaks and dips.
So the frequency plot has to be understood as showing a steady state situation. There is no information in the chart at which time a certain frequency has happened.
But in reality we hear a stream of incoming and continuously changing soundwaves. So our ear does not take a snapshot for a certain time, compute the frequency response and then uses it. It is a common misunderstanding of the frequency plot.
Even a 1/n-octave smoothing based on averaging the frequency plot for a 1/n octave does not show the reality because the underlying data are depending again on the pulse length. The error is just not as obvious because of the averaging algorithm.
A better way to interprete a pulse response is the frequency dependent windowing. By nature in the windowed pulse response the data with a bigger time distance from a given point of interest are windowed out and do not influence the frequency plot anymore. So you put more weight on the soundwave the actually happens. By nature such a frequency plot looks like smoothed but it is not a smoothing. It is simply the FFT result of a momentary situation.
This way a correction based on such a frequency dependent windowed pulse also is correcting the momentary sound. It does not correct a measured pulse response over its full length.
Of course now there are again 1001 strategies of how to carry out the frequency dependent windowing, leading to different results.
Anyway FDW is now more and more common. So Audiolense and also Acourate basically apply FDW.
So just use the unsmoothed frequency plot as an indicator on how much noise and reflections are present in your room. With a better acoustical room treatment you will get a better plot = less peaks and dips. In other cases check the smooth frequency response, preferably a result of a frequency dependent windowed pulse.
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I have asked for the correction curve itself, but not for the corrected room response curve.
As stated in a previous message I advise to check for the correction (influenced by measurement and the defined target). If e.g. someone says that the B&K curve sounds dull he can easily see this in the correction curve.
Here is the filtered frequency responses, correction curves and simulated final frequency responses (Audiolense model). I used time domain correction and linear phase option.
The first curve with almost flat target:
(http://img838.imageshack.us/img838/1783/flattarget.th.png) (http://imageshack.us/photo/my-images/838/flattarget.png/)
The second with something similar to the Bruel & Kjär target:
(http://img815.imageshack.us/img815/498/bktarget.th.png) (http://imageshack.us/photo/my-images/815/bktarget.png/)
It was not me that said B&K sounds dull, but I admit it is quite different from my old near-flat curve. I may end up preferring it all the same, though. I would appreciate any opinions on the corrections. What puzzles me is that there is a limit of +6dB correction, but even so the correction curve amplitude is way larger at some frequencies.
Here is a ETC of the actual measurement:
(http://img21.imageshack.us/img21/998/etcf.th.png) (http://imageshack.us/photo/my-images/21/etcf.png/)
My room is almost square (HxWxD 2.35x3.95x4.2m). Both reflective windows (behind the speakers) and absorbers at first refection points on the sides and in the ceiling. Some would say a bit too much dampening. My sofa is placed about 1.2m from the back wall. I do not have space for an equilateral triangle, but almost, and the distance to each speaker is as good as identical. The eagle-eyed will see that the measurement was done with slightly different distance to the speakers, as the two first ETC peaks are about 0.04ms apart.
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Of course now there are again 1001 strategies of how to carry out the frequency dependent windowing, leading to different results.
I thought fixed point per octave (FPPO) was a common method that works well:
From SMAART documentation:
(http://dl.dropbox.com/u/45539942/FPPO.png)
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Here is the filtered frequency responses, correction curves and simulated final frequency responses (Audiolense model). I used time domain correction and linear phase option.
Thanks for posting your graphs. Looks good!
If you get a chance and want to try an experiment, open up Audiolense and click new target. Click on the minimum phase radio button. Right click on the red point on the chart. That should pop up a dialog with several tabs. One tab should be labelled Data (thanks hulkss for the tip) Click on it and enter the following:
200 -0.5
2000 -3
20000 -6
That's what I use as a minimum phase target and my interpretation of the B&K target looking closely at the chart from the article.
To my ears, on my system, it produces the best tonal (or spectral) balance compared to several other target variations I have tried. Not too bright, not too dull, but just right.
The big suprise for me was how this target resolves the best 3D soundstage as well. Not too far forward, not too far back, but just right. At least that is how it sounds on my system, to my ears.
Relative to other targets, I did not think a couple of db adjustment either way would make a big audible difference, but it did, not only in tone quality, but the soundstage seems sensitive to small adjustments as well. At least on my system and to my ears. That's why I think it is important to enter the B&K target numbers directly into the Target Designer's Data page for accuracy, as opposed to drawing on the graph with the mouse.
I listen mostly to both low and hi res rock, pop, blues, world, folk, and some classical.
Curious to hear your (and others) feedback if you try this target out.
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Mitcho: I have made the correction curves, and will report back during the weekend. Could you explain one thing - you suggest a 6dB decrease at 20kHz. In your measured room response in one earlier post, the attenuation was about 10dB. How can you apply a -6dB target and then measure -10dB? BTW I tried -10dB, and that sounded dull :)
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Mitcho: I have made the correction curves, and will report back during the weekend. Could you explain one thing - you suggest a 6dB decrease at 20kHz. In your measured room response in one earlier post, the attenuation was about 10dB. How can you apply a -6dB target and then measure -10dB? BTW I tried -10dB, and that sounded dull :)
Trumpetguy - great question, I wondered that myself. If you look at the measurement again, it is closer to -9db at 20KHz. Here is a link to my Audiolense target and simulated response: http://i1217.photobucket.com/albums/dd381/mitchatola/Jan29Audiolense.jpg You see the target is indeed set at -6dB at 20KHz. However, my horn tweeter rolls off right at 20KHz. One could argue it is actually closer to -8db right at 20Khz. That would mean there is 1dB or so discrepancy at 20KHz between simulated and measured.
While there are many variability’s, I think the fundamental issue is that I used a completely different computer (laptop) and sound card combo to measure the frequency response using REW versus the much more expensive HTPC and sound card (Lynx L22) combo I used for taking the Audiolense measurements.
Here is the frequency response of my Lynx L22 card. http://i1217.photobucket.com/albums/dd381/mitchatola/lynxl22-1.jpg It is ruler flat right out to 50KHz., which is the measurement limit in REW. I used this to take the Audiolense measurements. Now have a look at the frequency response of the onboard sound chip in the laptop I used to measure the frequency response with REW: http://i1217.photobucket.com/albums/dd381/mitchatola/thinkpadx301soundcardfreqresp.jpg Yes, I used the corresponding sound card cal's for both measuring in REW and Audiolense.
Additionally, in order to take the measurement in REW, I used the loopback capabilities of my Lynx card, but rather than going through Convolution in JRiver, since I can’t get at its input (would be a great feature to add in JRiver btw), I had to host ConvolverVST in Savihost, which then hosted the Audiolense generated filters. More variability…
In hindsight, I should have used the exact same gear (i.e. HTPC and Lynx L22) for both REW and Audiolense measurements to remove this variability from the equation. Next time…
Personally, I believe it is more the slope of the B&K target curve than anything else that contributes to what I am hearing and would recommend going with the B&K target numbers (i.e. -0.5dB @ 200Hz, -3@2Khz, and -6@20KHz) as the baseline. Do you have any way (i.e., soundcard loopback or virtual audio cable) of measuring the actual frequency response?
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Do you have any way (i.e., soundcard loopback or virtual audio cable) of measuring the actual frequency response?
I just posted a guide for using the RTA function of REW. It is useful for measuring the actual frequency response of your signal chain with EQ or convolution.
How to use the RTA feature of REW with JRiver (http://yabb.jriver.com/interact/index.php?topic=69725.0)
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I just posted a guide for using the RTA function of REW. It is useful for measuring the actual frequency response of your signal chain with EQ or convolution.
How to use the RTA feature of REW with JRiver (http://yabb.jriver.com/interact/index.php?topic=69725.0)
Thanks, mojave!
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I just posted a guide for using the RTA function of REW. It is useful for measuring the actual frequency response of your signal chain with EQ or convolution.
How to use the RTA feature of REW with JRiver (http://yabb.jriver.com/interact/index.php?topic=69725.0)
Ask and yee shall receive! Thank you mojave! This should work perfect!
PS. I should have listed the other parameters that may have helped contribute to my 20KHz variability:
- mic cal file ends at 19,998Hz. I wonder what happens right at 20KHz? :)
- band limited the laptop sound card REW loopback measurement with a swept sine wave to 20KHz.
- band limited the REW speaker measurement by sweeping to 20KHz.
In contrast, the Audiolense measurements were band limited to 24KHz by the swept sine wave.
Next time I will increase everything to 25KHz and try the RTA!
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I just posted a guide for using the RTA function of REW. It is useful for measuring the actual frequency response of your signal chain with EQ or convolution.
I would much prefer to be able to feed an REW sweep (impulse response measurement) through MC DSP.
An RTA measurement is much different than measuring the system impulse response. Yes, the results will be the same if you set a long rectangular window for the FTT, however, this approach does not exclude any reflected sound from the measurement.
This is not at all how your brain perceives sound, there is much more importance on the direct sound which arrives first than reflections which arrive later to your ear. An RTA treats it all the same.
A sweep with Frequency Dependent Windowing (FDW) for processing the FFT (as posted by Uli below) is the best way to measure your system and is how Acourate and Audiolense work. This is also illustrated in the Fixed Point Per Octave (FPPO) example I posted below. Impulse response analysis in REW provides similar capability.
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Hi guys. I'm new here! First post actually and I have to say this topic is really interesting!
Sorry to hijack the thread like this, I'm sorry for it, but I need some help. Just like NickF was, I'm waiting for my microphone kit to arrive so i can start playing.
But i have some questions that i really haven't found answer by searching.
1 - First of all which audiolense version should I invest in? I need at least the Surround version. But should I buy the XO ? Do you guys think is it worth the extra money?
2 - The mic position !! I haven't found any picture ! Horizontal position ? vertical position ? angled ? For both calibrations? I mean which calibration position for music (2.1) and 5.1 for movies i have to use?
3 - I use hdmi from ati/amd 6970 card into a onkyo 609, thats for output but how about input? I have to use the onboard realtek mic jack, is this alright ?? I took out a Asus D2x when i started using hdmi, should i use it instead for mic in ?
4 - What volume level do I configure in windows for the mic ? 100% ? and boost at zero ? and what about the kit pre-amp volume knob ?
5 - audiolense likes asio but since i'm using hdmi out from ati/amd i don't have asio. Should I use asio4all ? Pity it doesn't work with wasapi like J River, or ?
6 - onkyo volume !! how do i set it? So i can measure correctly without causing damage to the speakers?
7 - What do I do about the calibration/speaker settings that were made using Onkyo MCACC EQ ? Do i keep the speaker distance and levels it measured ? makes sense leaving it alone but...
A lot of questions, I know, but it's my first time doing this !!
Thank you for all for this great forum.
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1 - First of all which audiolense version should I invest in? I need at least the Surround version. But should I buy the XO ? Do you guys think is it worth the extra money?
Active XO is a major benefit definitely worth the investment IMHO (I assume you know what Active XO requires to implement).
2 - The mic position !! I haven't found any picture ! Horizontal position ? vertical position ? angled ?
Get an omnidirectional mic and aim it straight up at ear level when seated. Compensate for high frequency directional fall-off in frequency response of the mic in the mic cal file or in your target response. Most mics come with data like this:
(http://www.earthworksaudio.com/wp-content/uploads/M50PolarPattern.png)
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I'm only a beginner here but I'll give my answers from recent experience!
1 - which audiolense version should I invest in?
Good question. I went for Surround because I wanted to cover stereo for my music and surround for my movies but didn't want active crossovers. However, I soon found that there were features I needed in XO such as corrections of more than 6 dB so i paid for an upgrade. After all, it's only money!!
2 - The mic position
The mic is omnidirectional and needs to be mounted vertically. It comes with a clip with a large thread which I didn't recognise so I have taped it to an old camera tripod. I place the tripod on my listening seat and put cushions around it to avoid any reflections and vibrations. The mic and pre-amp come with no cables. All connections are RCA Phono. I asked a question about the quality of the cable but hulkss answered with a general answer about using quality cables. My view, and I am a novice in convolution, is that this measurement path is critical in the setup. It needs to hear every deficiency in the playback path and environment so it is important not to introduce any additional deficiencies or interference into the measurement path. So I have made some good quality cables.
3 - I use hdmi from ati/amd 6970 card into a onkyo 609, thats for output but how about input? I have to use the onboard realtek mic jack, is this alright ?? I took out a Asus D2x when i started using hdmi, should i use it instead for mic in ?
I am lucky to have an RME card which has inputs and outputs. I have never rated Realtek very highly so I don't know about the quality of the mic input. Remember that you are using a mic pre-amp so you really want a line input. I will leave it to others to advise here.
4 - What volume level do I configure in windows for the mic ? 100% ? and boost at zero ? and what about the kit pre-amp volume knob ?
I set my amp to a normal listening level and the pre-amp to about mid position. When you do the Measurement with Audiolense, when the measurement ends, it will display the highest level and, consequently, the part most likely to be clipped or distorted. If you get either, back off the volume so you get a nice sine wave. You need to reach a point where the volume is loud enough to expose resonances in the room without distortion. It's a case of trial and error.
5 - audiolense likes asio but since i'm using hdmi out from ati/amd i don't have asio. Should I use asio4all ? Pity it doesn't work with wasapi like J River, or ?
You need to get the best quality path from mic to Audiolense. I have no experience with asio4all. Someone else may help here.
6 - onkyo volume !! how do i set it? So i can measure correctly without causing damage to the speakers?
See my comments in 4 above. Start low until you know what levels Audiolense will output then increase to the level that you would listen at.
7 - What do I do about the calibration/speaker settings that were made using Onkyo MCACC EQ ? Do i keep the speaker distance and levels it measured ? makes sense leaving it alone but...
See my post further up this thread. You need to remove all corrections in your Onkyo. I don't know whether you can keep them stored and just switch them off.
Nick
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8) !! thank you all ! Really nice of you guys to help
I felt silly, I forgot I will be using a pre-amp so obviously I'll have to use line in, thank you for that one ! So still leaves, realtek line-in or asus d2x? New question. Line in expects stereo input or mono will also work ? When making the cables I'll use a mono jack? ; stereo jack and only use the tip connection? ; stereo jack and short/use both channel inputs ? What did you do Nick?
"I assume you know what Active XO requires to implement" nop, no idea, i think.. !! I presume using separated amplifiers per speaker type with a electronic crossover between the pre-amp and the power amps? I'm I making any sense here? ? ? That is out of my league :D
I asked about surround vs XO because of the extra functions like Audiolense True Time Domain correction and multiseat readings, if that is another thing that could help my sound system then i'll go for XO version.
To disable mcacc, by reading Onkyo user manual i just need to use pure audio mode..hmm but that disables routing the bass to the subwoofer. I'll have to wait for the Mic and experiment on this one.
Any input and correction on any thing is welcome !
Again thank you
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Line in expects stereo input or mono will also work ?
Yes, mono will work. Audiolense just found my input and only needs a mono input so it's stereo jack and only use the tip connection.
"I assume you know what Active XO requires to implement"
Hulkss mentioned this. I don't know what he meant so maybe he could elaborate. I just installed Audiolense and placed the licence file in the program folder and it worked.
I presume using separated amplifiers per speaker type with a electronic crossover between the pre-amp and the power amps?
This is an option in XO but you don't have to do this and I would advise that you don't try to start with this level of complexity. You need to look at the Speaker Setup and choose the right match for your setup. Remember that you will need two filters, one for stereo and one for surround, so you need to go through the whole process twice. Having done my first attempt at surround yesterday, I now know that multiseat is essential. I tend to listen to music alone and want the best possible stereo image and single measuring point gives this. For movies, I need two seats. With single point, there was absolutely no image in the second seat. I will try multiseat today.
To disable mcacc, by reading Onkyo user manual i just need to use pure audio mode..hmm but that disables routing the bass to the subwoofer.
I can't help here. I use 6 channel analogue input to my Denon. There are other Onkyo users here or on other forums who may be able to help.
Nick.
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1 - First of all which audiolense version should I invest in? I need at least the Surround version. But should I buy the XO ? Do you guys think is it worth the extra money?
Active XO is a major benefit definitely worth the investment IMHO (I assume you know what Active XO requires to implement).
2 - The mic position !! I haven't found any picture ! Horizontal position ? vertical position ? angled ?
Get an omnidirectional mic and aim it straight up at ear level when seated. Compensate for high frequency directional fall-off in frequency response of the mic in the mic cal file or in your target response. Most mics come with data like this:
(http://www.earthworksaudio.com/wp-content/uploads/M50PolarPattern.png)
The XO version is the only with True Time Domain correction, the other versions are frequency corrections only (2.0 or multichannel). TTD is worth the xtra cash. I use XO in an ordinary 7.1 multichannel setup, with two subs on the .1 channel. Makes it really simple to integrate the subs, with separate cross-over filters for all the 7 high-pass channels. Maybe to surround version have that functionality, I don't know.
hulkss: I have never understood why the mic should be pointed upwards. I have however noticed that the mic direction is a much debated subject. All my measurements have been done with the mic pointed horizontally forward, exactly in the middle of the main speakers. Should I redo all my measurements? I am not worried, I do that from time to time...
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About mic-direction: I strongly recommend reading this thread http://www.hometheatershack.com/forums/rew-forum/20293-full-range-eq-no-go-3.html
Take good notice of Wayne A. Pflughaupt's comments.
Vertical orientation is fine with an omnidirectional mic if you have a correction file to compensate for the roll-off at the high frequencies. It probably also is the only convenient orientation if using a multi-channel setup. Otherwise you'd have to turn the mic between each sweep...
As Wayne in the long thread points out, if pointing the mic upwards you get a lot of reflections from the ceiling (if not treated) instead of a more direct loudspeaker response (if pointed towards each speaker and positioned horizontally). But since DRC is about room correction I guess that isn't a problem. But do remember to use a correction-file.
Best regards,
Mikkel
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I have never understood why the mic should be pointed upwards.
The need exists in Digital Room Correction with Time Domain Correction to measure all speakers (including surrounds and rears) from a single point with similar measurement characteristics. By pointing the mic up, you can aim it with the same relative angle (about 90 degrees) to all of your speakers. The best average angle may not be exactly straight up.
You may have noticed that measurement mics are very small at the tip where the measurement sensor is. This is to achieve good omnidirectional measurement characteristics. Human hearing is the most sensitive to sounds from the front and sides. You want the mic to measure all those speakers the same so they will have the same timbre after correction.
As to measuring reflections from the ceiling: The Time Domain Correction Technique is expressly intended to correct for low frequency reflections, so you do want to measure them from all directions as equally as possible. At higher frequencies the measurement time window is shortened so those ceiling reflections are not measured at all. This is possible because you are usually measuring about three feet off the floor. The direct sound path length from speaker to mic is much shorter than a reflected ceiling path so the reflection is easy to eliminate with a shortened measurement window.
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Vertical orientation is fine with an omnidirectional mic if you have a correction file to compensate for the roll-off at the high frequencies. It probably also is the only convenient orientation if using a multi-channel setup.
So just for clarity for adolfotregosa, he has bought the MP-1r-KIT Measurement kit from Audiolense which does come calibrated so he should use it vertically, confirmed by hulkss. Adolfotregosa needs to remember to load the calibration file, supplied on a CD, into Audiolense before he starts to take measurements.
A question, though. Is the mic calibrated in a vertical position? Presumably it is.
Nick.
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A question, though. Is the mic calibrated in a vertical position? Presumably it is.
I would guess the mic is calibrated in a very narrow, completely anechioc chamber. Like a small box with absorbant walls. Orientation should not be an issue then.
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Seriously, thank you all. I'm really anxious to get the Mic and start working on this. I really want to see what I can pull of my sound system. I've been learning so much since I started building it. lol it never ends ;D
Does JRiver MC also use convolution while bit-streaming dts, dts-ma, etc movies files? Or it only works for music?
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Does JRiver MC also use convolution while bit-streaming dts, dts-ma, etc movies files? Or it only works for music?
You can use the DSP engine for everything. It's up to you. I have a zone for 2 ch music playback and a zone for multi-channel video playback that are slightly different.
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It would be great if MC allowed to directly record from the mic a played sweep.
Actually I tried using Audacity (ASIO version) to record the sweep played by MC but since MC is using ASIO as well the sound cards figures as in use and I cannot record from any other external program.
I'm doing this because MC allows me to mute channels and apply xover filters useful when measuring any non full range speaker, like the sub or the side speakers. E.g.: joining sub and center to have a correct impulse response to be processed by DRC.
Actually I had to use DirectSound to do so but it's rather tedious to play from MC, quickly switch to Audacity and press REC, and then stop it. Not to mention the record would take place in the same environment in which will be applied; also I would so keep everything in the ASIO realm.
Am I thinking it wrong?
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It would be great if MC allowed to directly record from the mic a played sweep.
Actually I tried using Audacity (ASIO version) to record the sweep played by MC but since MC is using ASIO as well the sound cards figures as in use and I cannot record from any other external program.
I'm doing this because MC allows me to mute channels and apply xover filters useful when measuring any non full range speaker, like the sub or the side speakers. E.g.: joining sub and center to have a correct impulse response to be processed by DRC.
Actually I had to use DirectSound to do so but it's rather tedious to play from MC, quickly switch to Audacity and press REC, and then stop it. Not to mention the record would take place in the same environment in which will be applied; also I would so keep everything in the ASIO realm.
Am I thinking it wrong?
First - using ASIO you will always lock audio stream to one application. You will need to generate and record from the same application. Why not use REW? Or download Audiolense and do it there, with full cross-over functionality. The test license allows you to, but has an attenuation built into the filter you would generate (until you pay :) ). With Audiolense doing the full-range and/or non-full range sweeps is straight forward.
The main reason I see for MC to generate sweeps and record room response from an analog in channel would be to measure the actual room response with a DRC filter and JRiver convolver in the signal chain.
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First - using ASIO you will always lock audio stream to one application. You will need to generate and record from the same application.
Yeah, I knew that, that's why I wrote my post.
Why not use REW? Or download Audiolense and do it there, with full cross-over functionality. The test license allows you to, but has an attenuation built into the filter you would generate (until you pay :) ). With Audiolense doing the full-range and/or non-full range sweeps is straight forward.
REW doesn't use ASIO, REW max allowed sample rate is 48kHz and I didn't see options for measuring a 5.1 system, just left/right and sub. Did I overlook the feature?
Audiolense is not free and I'm not certainly taking it into account right now. Besides, what's the purpose of measuring everything with it when you can't put it into "production" because of the attenuation thing...
The main reason I see for MC to generate sweeps and record room response from an analog in channel would be to measure the actual room response with a DRC filter and JRiver convolver in the signal chain.
That is another interesting pro.
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what's the purpose of measuring everything with it when you can't put it into "production" because of the attenuation thing...
I have not had any trouble with an "attenuation thing."
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Audiolense is not free and I'm not certainly taking it into account right now.
Why, because its not free? Is that a requirement? Does that apply to hardware also?
Besides, what's the purpose of measuring everything with it when you can't put it into "production" because of the attenuation thing...
The measurement is as good as anything in the trial version, but the filter is attenuated. Even with a 20dB attenuation you will get a good impression of the performance, but with reduced dynamic range. If you get a license, you can use the same measurement and regenerate the filter. I think that is a pretty good deal.
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Thanks guys for your answers but they don't address any of my current concerns.
I said I'm not interested in using/buying Audiolense - I'm using DRC right now and I've already done the 5.1 DRC configuration by using together JRiver and Audacity for recording the sweeps. I just wanted to deal with less programs and keeping the exact environment I'm using when doing the playback with JRiver - that is distances SPL leveling etc. etc.
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Thanks guys for your answers but they don't address any of my current concerns.
I said I'm not interested in using/buying Audiolense - I'm using DRC right now and I've already done the 5.1 DRC configuration by using together JRiver and Audacity for recording the sweeps. I just wanted to deal with less programs and keeping the exact environment I'm using when doing the playback with JRiver - that is distances SPL leveling etc. etc.
It isn't clear to me whether you are using or planning to use any convolution product in your solution. You talk about doing DRC but are you just using the Equaliser and Room Correction features in MC's DSP Studio? The real value-add of products like Audiolense is the "intelligence" in the creation of the correction filters to correct for the behaviour of the whole listening environment. You aren't going to duplicate or replace this with the other DSP components.
I can certainly see value in MC generating the sweep and recording the result to create the parameters required for Room Correction and Equalisers without getting into the full complexity of convolution. This would simplify the setup of these DSP components and could be a first step to a complete convolution solution. I suspect that Matt may have this in mind.
Nick.
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It isn't clear to me whether you are using or planning to use any convolution product in your solution. You talk about doing DRC but are you just using the Equaliser and Room Correction features in MC's DSP Studio? The real value-add of products like Audiolense is the "intelligence" in the creation of the correction filters to correct for the behaviour of the whole listening environment. You aren't going to duplicate or replace this with the other DSP components.
Nick.
Drc is also the name of a free program to create filters similar to audiolense etc.
http://drc-fir.sourceforge.net/
I also use it. I especially like the result I get when I use its option to compute a target frequency response based on a model of the psychoacoustic perception of the corrected frequency response. see:
http://drc-fir.sourceforge.net/doc/drc.html#htoc31
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Drc is also the name of a free program to create filters similar to audiolense etc.
Sorry, my mistake. Not enough acronyms, apparently!
Nick.
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It isn't clear to me whether you are using or planning to use any convolution product in your solution. You talk about doing DRC but are you just using the Equaliser and Room Correction features in MC's DSP Studio? The real value-add of products like Audiolense is the "intelligence" in the creation of the correction filters to correct for the behaviour of the whole listening environment. You aren't going to duplicate or replace this with the other DSP components.
I'm already using it and I'm certainly not talking about eq and room correction features. But, having them perfectly set up help me when recording the sweep, because I'm replicating the exact environment in which the filter will be applied.
The free DRC solution though, hasn't been created having the multi-channel setup in mind. So, in order to create correct filters for SW and other speakers I need to do some tricks:
e.g. for measuring the center channel:
Copy L to C
High filter 80hz on C
Copy L to SW
Adjust gain (- or + db) for SW to keep C and SW balanced
Low filter 80Hz on SW
Mute all the speakers except for C and SW
Play the sweep from MC and Record the Sweep from Audacity
e.g. for measuring the SW:
Copy L to C
High filter 320hz on C
Copy L to SW
Low filter 320Hz on SW (generally the xover is 2/octave above the normal playback setting)
Mute all the speakers except for C and SW
Play the sweep from MC and Record the Sweep from Audacity
I would like to better the last step and play and record the sweep through ASIO from JRiver.
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Sorry, my mistake. Not enough acronyms, apparently!
Nick.
yes, exactly. I'm referring to DRC as the homonym solution: http://drc-fir.sourceforge.net/
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REW doesn't use ASIO, REW max allowed sample rate is 48kHz and I didn't see options for measuring a 5.1 system, just left/right and sub. Did I overlook the feature?
ASIO supported was introduced in REW in version 5.01 on 7/11/11. You can use any sample rate supported by your audio device and you can select which input and outputs to use. By only changing the output in the soundcard setup in REW, I can measure any speaker in my system.
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ASIO supported was introduced in REW in version 5.01 on 7/11/11. You can use any sample rate supported by your audio device and you can select which input and outputs to use. By only changing the output in the soundcard setup in REW, I can measure any speaker in my system.
Could you please explain the relationship between MC, REW and your soundcard for this testing? Since we can't send the REW sweeps to MC I assume you still use a pink noise file in MC and the RTA in REW with no direct interaction between the two programs. So it would seem that you are measuring the speakers without DSP in MC, thus not seeing the results of any EQ done in MC?
I'm confused (nothing new!)
Thanks,
Rod
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I was just talking about using REW to measure your system to see what is going on before making changes. After you make changes, the RTA method is the only one that works for me and you can't use REW in ASIO mode anymore. In case you missed it, I made a guide on How to use the RTA feature of REW with JRiver (http://yabb.jriver.com/interact/index.php?topic=69725.0).
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I was just talking about using REW to measure your system to see what is going on before making changes. After you make changes, the RTA method is the only one that works for me and you can't use REW in ASIO mode anymore. In case you missed it, I made a guide on How to use the RTA feature of REW with JRiver (http://yabb.jriver.com/interact/index.php?topic=69725.0).
Thanks for the clarification. I had seen your helpful guide - thanks for your efforts on it!
I have been using the RTA in REW with the pink noise files for awhile now. Just wondered if there was some trick you were using that I missed.
I don't fully understand this whole convolution thing, but am wondering if there might be some way using it to get the REW Sweeps into MC. Then we could use REW to measure the system performance after corrections instead of only before. And of course we could use Waterfall, impulse and the full compliment of REW features.
Rod
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ASIO supported was introduced in REW in version 5.01 on 7/11/11. You can use any sample rate supported by your audio device and you can select which input and outputs to use. By only changing the output in the soundcard setup in REW, I can measure any speaker in my system.
Interesting but, still, it doesn't allow to play 2 speakers contemporaneously. e.g. center + sub
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To be more specific on the last post, I know how to measure my room, I know how to create filters in REW but I've not seen a set of instructions as to how to load a filter into JRiver. Maybe I need to just play around some more in JRiver but I have access only via RDP on an iPad... I do not want to purchase Audiolense. I would rather apply the money to Theta's forthcoming implementation of Dirac Live.
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Read it and the threads it linked to but unfortunately (unless I missed something) it didn't really help at all. Maybe I need an idiot's guide.
What format are you generating the REW filters in? How many channels are you trying to convolve at once?
If it's just stereo you should be able to generate a stereo WAV filter that you can load directly into JRiver's DSP studio "Convolution" module. If you're trying to do multichannel convolution you'll need to write a config file. Convolution (in my experience) is complicated to do right, and easy to do wrong, but if you can provide more specific info I can talk you through some of the ins and outs.
One solution that bypasses convolution (and only sacrifices some functionality) is to look at the correction filters that REW is dialing in and to manually transcribe them into the parametric equalizer in JRiver. You can find REW's proposed filters in REW's "EQ Filters" window: http://www.hometheatershack.com/roomeq/wizardhelpv5/help_en-GB/html/eqfilters.html#top
In the linked example, you'd just make a new entry in the JRiver DSP Studio parametric equalizer module for each of the those filters. You'd probably want to choose "adjust a frequency," and then you'd enter the relevant frequency, gain, and bandwidth (or Q). That would effectively replicate what REW is trying to accomplish without convolution. That has some advantages (much less latency and processor use) and some disadvantages (can't correct phase as readily), but is much easier for multichannel setups.
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What format are you generating the REW filters in? How many channels are you trying to convolve at once?
Just stereo for now. I'm dabbling while I wait for Theta to implement Dirac Live.
If it's just stereo you should be able to generate a stereo WAV filter that you can load directly into JRiver's DSP studio "Convolution" module.
........
One solution that bypasses convolution (and only sacrifices some functionality) is to look at the correction filters that REW is dialing in and to manually transcribe them into the parametric equalizer in JRiver. You can find REW's proposed filters in REW's "EQ Filters" window: http://www.hometheatershack.com/roomeq/wizardhelpv5/help_en-GB/html/eqfilters.html#top
In the linked example, you'd just make a new entry in the JRiver DSP Studio parametric equalizer module for each of the those filters. You'd probably want to choose "adjust a frequency," and then you'd enter the relevant frequency, gain, and bandwidth (or Q). That would effectively replicate what REW is trying to accomplish without convolution. That has some advantages (much less latency and processor use) and some disadvantages (can't correct phase as readily), but is much easier for multichannel setups.
Thanks. I've made it as far as observing the EQ Filters window and studying "house curves". I was not sure how to then take the information I see in REW and get it into JRiver. When I save the filters I do not get a file with the suffix .WAV.
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Just stereo for now. I'm dabbling while I wait for Theta to implement Dirac Live.
Thanks. I've made it as far as observing the EQ Filters window and studying "house curves". I was not sure how to then take the information I see in REW and get it into JRiver. When I save the filters I do not get a file with the suffix .WAV.
What suffix are you seeing? You may need to change your export settings. When I get home I'll open up REW and try and be more specific.
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Let me respond when I get home ;D
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What suffix are you seeing?
.req
It's not editable in a text editor
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.req
It's not editable in a text editor
.req is just REW's storage format, not a usable filter format. Under the file menu in REW, look for the option to export the filters as a WAV. You'll want to make sure that you're exporting the filters as a stereo WAV and not a mono WAV.
http://www.hometheatershack.com/roomeq/wizardhelpv5/help_en-GB/html/file.html#top
Once you've got a .WAV saved, try loading it in JRiver's DSP studio convolution module. One important thing to keep in mind is that convolution filters are sample rate specific. A convolution filter generated using a 44.1 KHz sample rate will not sound right with 48 KHz files. There are two solutions to that problem:
1) Pick the sample rate you use most often, and generate your REW filter using that, and then use JRiver's DSP engine to resample everything to that sample rate (this sometimes introduces problems, but works for some folks), or
2) Make a different convolution filter for every sample rate you plan on using, and then use JRiver's automatic convolution sample rate switcher (a check box in the convolution module). In order to work, the switcher requires that the files be named in an extremely specific fashion:
xxxx2.0_441
xxxx5.1_48
etc.
Where:
xxxx = Arbitrary Filter Name
2.0 or 5.1 = number of channels, and
441 or 48 = Sample rate
If that all starts to sound like a lot of work (and it can be to get it right), you can try the previous approach I mentioned (above) and just manually transfer the EQ filters and see how that sounds before worrying about phase. Phase can be important, but uneven frequency response is a much more noticeable issue in my experience.
Let me know if you hit more roadblocks, I'm happy to help!
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Hey thanks a lot. I will give this a solid look over the weekend! I guess I will have issues generating a sweep at higher bit rates to generate filters for high-res audio but 44.1 and 48 kHz should not be a problem.
Very much appreciated!