Hi Andrew
It is possible to get far-field data from non-far-field measurements. I do it.
You have to decompose the full 360 degree field into its radiation modes, which can then be reassembled at a further distance (say 1 meter out to 10 meters) since the propagation characteristics of the modes is know even though those of an arbitrary source are not.
One other benefit is that one now has very high angular resolution of the source - basically infinite from a finite number of angular measurements.
Some details on Geddes measurement technique here:
http://www.diyaudio.com/forums/multi-way/212394-real-farfield-distance-5.html#post3024323
For a realistic measurement of FR in the actual listening room at the listening position I use pink noise and 1/3 (or 1/6) octave filtered Leq. Leq is equivalent noise level, the long-term average of noise. Good SPL meters are capable of doing it. It is quite immune to speech, cars, dogs, etc. It gives what you can hear at the actual listening environment, including room influences. But if you want to separate the contribution of the loudspeaker and the contribution of the room, do an anechoic (outdoor) or gated pulse measurement.
Unfortunately this is not what we perceive in terms of timbre/tonality. If this would be true then a diffuse field equalization would make every speaker sound the same (more or less), which it does not.
The diffuse field is certainly a great contributor to perceived tonality but the direct sound field and certain early reflections are more important than power response. It is still unknown how our hearing weights the specific contributors.
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The process LMS uses (applying a time gate to stepped sine tones) is mathematically equivalent to applying a window of the same duration to an impulse response derived from a continuous sweep or 2-channel FFT measurement, just much, much less efficient.
This thread has made this process way more complicated than it needs to be. If you want the repsonse in your listening position, do what professional sound engineers do(For concert venues). Pink noise and a real time analizer. Forget sweeps, chirps and all the stuff that requires complex vector calculus to decipher. There is no such thing as a perfect room or perfect response, so don't bother looking for it. Set your mic up where you listen, measure the pink noise response of the system and eq for the response curev you want, or think you want, in that listening space. It will never be perfect from every point in the room, that is not physically possible. Optimize it for where you listen. Many people have touched on the need to start with measurment, and tune by ear. That is solid advice, because regardless of the response curve of the room, what matters most is whether or not YOU enjoy the sound. 
This thread has made this process way more complicated than it needs to be. If you want the repsonse in your listening position, do what professional sound engineers do(For concert venues). Pink noise and a real time analizer. Forget sweeps, chirps and all the stuff that requires complex vector calculus to decipher. There is no such thing as a perfect room or perfect response, so don't bother looking for it. Set your mic up where you listen, measure the pink noise response of the system and eq for the response curev you want, or think you want, in that listening space. It will never be perfect from every point in the room, that is not physically possible. Optimize it for where you listen. Many people have touched on the need to start with measurment, and tune by ear. That is solid advice, because regardless of the response curve of the room, what matters most is whether or not YOU enjoy the sound.
This doesn't work. Please see http://www.diyaudio.com/forums/mult...-loudspeaker-home-accuracy-5.html#post3541714
I respect your opinion Markus, but your supplied link only goes to this page of the thread. I have over 20 year of experience in live sound reinforcement, home theater install, and mobile audio. I was a certified IASCA and USAC sound competition judge for many years, built several factory demo vehicles for the likes of Rockford, JBL and Alpine. I have also spent countless hours in recording stuidios working with the sound engineers that practically live in the studio. That's the industry standard that I have been exposed to. That's how the recordings you listen to are done Before any recording session, the system is checked, and they all (in my experience) use pink noise to measure frequency response before beginnning any recording or concert session. They then eq for the response they are looking for, and make adjustments by ear during initial sound check. Please explain how or why that doesn't work, as I have seen it done that way for more than 20 years. It's always worked for me, and IASCA and USAC have also found it the most acceptable way to compare audio systems for judgement purpoeses for more than a couple of decades.
I really don't know where to start because it is such a widely accepted fact that simple steady-state 1/3 octave equalization is not desirable, especially in acoustically small rooms (free field is a completely different story).
Toole talks about it extensively in "Sound reproduction", Newell in "Recording Studio Design", even Allen writes:
"Steady-State vs. First-Arrival Measurement Techniques
The dominance of dialog in a typical movie was discussed previously. If the words are the
lifeblood of the movie and 80% of the movie is words, and if many of the sounds that make for
speech intelligibility are of short duration, one may ask why steady-state measurement techniques
are used to calibrate theatres, as opposed to measurements of first-arrival signals.
There are two answers. First, in the mid-1970s, when theatre equalization became practical, pink
noise and a realtime analyzer were the only tools and technique available and at a cost practical in
the commercial theatre world.
But a more interesting possibility is that setting a flat first-arrival frequency response may not be
a good thing. Ear and brain do not have a flat frequency response integration time; the combination
takes much longer to determine loudness at low frequencies than at mid and high frequencies.15
Indeed, it seems possible that as yet, no perfect technique for B-chain measurement exists,
requiring, it would seem, some combination of first-arrival and steady-state analysis."
("The X-Curve" by Ioan Allen, SMPTE Motion Imaging Journal)
Toole talks about it extensively in "Sound reproduction", Newell in "Recording Studio Design", even Allen writes:
"Steady-State vs. First-Arrival Measurement Techniques
The dominance of dialog in a typical movie was discussed previously. If the words are the
lifeblood of the movie and 80% of the movie is words, and if many of the sounds that make for
speech intelligibility are of short duration, one may ask why steady-state measurement techniques
are used to calibrate theatres, as opposed to measurements of first-arrival signals.
There are two answers. First, in the mid-1970s, when theatre equalization became practical, pink
noise and a realtime analyzer were the only tools and technique available and at a cost practical in
the commercial theatre world.
But a more interesting possibility is that setting a flat first-arrival frequency response may not be
a good thing. Ear and brain do not have a flat frequency response integration time; the combination
takes much longer to determine loudness at low frequencies than at mid and high frequencies.15
Indeed, it seems possible that as yet, no perfect technique for B-chain measurement exists,
requiring, it would seem, some combination of first-arrival and steady-state analysis."
("The X-Curve" by Ioan Allen, SMPTE Motion Imaging Journal)
Thank you Markus,
I will do some research based on what you have supplied. Maybe the recording insudtry needs to make an update to procedure. I know that the studios eq everything on the 1/3 ocatve standard. Maybe the old school methods need some updating. I hope I did not come across as hostile, it was not my intention. I appreciate the info and will use it accordingly. I hope to learn some new methods.
I will do some research based on what you have supplied. Maybe the recording insudtry needs to make an update to procedure. I know that the studios eq everything on the 1/3 ocatve standard. Maybe the old school methods need some updating. I hope I did not come across as hostile, it was not my intention. I appreciate the info and will use it accordingly. I hope to learn some new methods.
Harman - Scientific Publications
Link to Floyd toole's white papers on the Harman web site.
Very interesting stuff.
Here's a good one relating directly to this thread: http://www.wghwoodworking.com/audio/LoudspeakersandRooms-WorkingTogether.pdf
Thanks again Markus.
Link to Floyd toole's white papers on the Harman web site.
Very interesting stuff.
Here's a good one relating directly to this thread: http://www.wghwoodworking.com/audio/LoudspeakersandRooms-WorkingTogether.pdf
Thanks again Markus.
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Discussion of what might be the ideal method aside, setting up a room for good sound overall, particularly when it comes to just final EQ tweaking, is an entirely different thing than what the OP asked about. This thread was a person stating they wanted to try designing a crossover based on measurements (need suitable reflection free-ish measurements, one way or another), and then measure whether it was working as expected (again, need to eliminate room for the most part).
They simply asked how to tell whether their measurements were accurate ("accurate" in this case meaning reasonably free of room influence, among other things). The answer (which they did get) is that it would be easier to point out how to see the problems if the impulse was shown, and that the room described is too small to get very low at all, even if empty.
They simply asked how to tell whether their measurements were accurate ("accurate" in this case meaning reasonably free of room influence, among other things). The answer (which they did get) is that it would be easier to point out how to see the problems if the impulse was shown, and that the room described is too small to get very low at all, even if empty.
Markus - I am just going to stop commenting because you are always there first - "that's what I was going to say."
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