I always fix the added mass with a special glue, and i never heard any weird sound caused by that
I have a small precision digital weight meter with a precision of 0,1 gramms!
The closed chamber was made of a piece of a pipe and some plywood
. to be honest the precise volume was 1,35 liter, becouse I was not calculated for the volume for the driver (it is fixed into the small cabinet that way, that the magnet is outside)
but even with the 1,35 liter volume if you calculate the TSPs, they are very different to the added mass methode....
which one is more precise method?
and what causes the difference?
Thanks,
Tamas
I have a small precision digital weight meter with a precision of 0,1 gramms!
The closed chamber was made of a piece of a pipe and some plywood
. to be honest the precise volume was 1,35 liter, becouse I was not calculated for the volume for the driver (it is fixed into the small cabinet that way, that the magnet is outside)
but even with the 1,35 liter volume if you calculate the TSPs, they are very different to the added mass methode....
which one is more precise method?
and what causes the difference?
Thanks,
Tamas
Closed box SHOULD give better results, but only if the box is really closed. With a lot of drivers, it can't be closed because the dust caps or surrounds leak so much. Happens more commonly than you'd think. If that's the case, added mass is better.
It is possible to determine using nearfield sound pressure, but for that you need a calibrated microphone and a measurement system that measures absolute (not just relative) pressures, and know drive voltage levels. And software that can do the math, of course (I don't know if Arta does that).
It is possible to determine using nearfield sound pressure, but for that you need a calibrated microphone and a measurement system that measures absolute (not just relative) pressures, and know drive voltage levels. And software that can do the math, of course (I don't know if Arta does that).
Some problems getting started!
Hi!
I have just downloaded and registered for ARTA!
I will use it to optimize my High End stereo - which is computer based.
My set up is:
- Computer with MS Vista
- Lynx AES 16e SRC
- DAD AX24 (6 channnel DA converter + 1 channel Mic preamp/AD convert)
- Earthworks M30 measurement mic
(The rest of my system is 3 x Dynamic Precision DP A1S power amps and Overkill Audio Finale speakers).
I have some problems getting started. So I will post some questions in the end of this post.
I am using digital XO for my 3-way speakers. The way I do that is to run a filter in Convolver in Console.
My starting questions....
1) Are there any good places to read about more ARTA experience, youtube videos with "getting started" etc..? (Unfortunately my german is not very good)
2) How detailed can I set up my outputs. I would like the following set up:
Left Channel
0-150Hz Output 1
150-800Hz Output 2
800- Output 3
Right Channel Output 4,5,6
Is this possible....? Like a global setting....so no matter what i do - there is always a crossover function in the signal....
3) If question 2 is not possible - I will have to run the crossover in my digital loop in Console....this will cause some latency - will that affect measurements...?
4) The measurements I am most interested in (at least in the beginning) are:
*Frequency respons (waterfall)
*Room decay measurements
Which measurement will give the most reliable results.....?
5) Does this program work for 88/96 kHz sampling rate...?
Thanks!
Best regards
Ole Martin (audiophile in Norway)
Hi!
I have just downloaded and registered for ARTA!
I will use it to optimize my High End stereo - which is computer based.
My set up is:
- Computer with MS Vista
- Lynx AES 16e SRC
- DAD AX24 (6 channnel DA converter + 1 channel Mic preamp/AD convert)
- Earthworks M30 measurement mic
(The rest of my system is 3 x Dynamic Precision DP A1S power amps and Overkill Audio Finale speakers).
I have some problems getting started. So I will post some questions in the end of this post.
I am using digital XO for my 3-way speakers. The way I do that is to run a filter in Convolver in Console.
My starting questions....
1) Are there any good places to read about more ARTA experience, youtube videos with "getting started" etc..? (Unfortunately my german is not very good)
2) How detailed can I set up my outputs. I would like the following set up:
Left Channel
0-150Hz Output 1
150-800Hz Output 2
800- Output 3
Right Channel Output 4,5,6
Is this possible....? Like a global setting....so no matter what i do - there is always a crossover function in the signal....
3) If question 2 is not possible - I will have to run the crossover in my digital loop in Console....this will cause some latency - will that affect measurements...?
4) The measurements I am most interested in (at least in the beginning) are:
*Frequency respons (waterfall)
*Room decay measurements
Which measurement will give the most reliable results.....?
5) Does this program work for 88/96 kHz sampling rate...?
Thanks!
Best regards
Ole Martin (audiophile in Norway)
Hi Ole,
The English pdf manuals are pretty good (especially considering the author is from Croatia) -- same page where you downloaded the software. Start at the beginning and it will walk you through calibrating your sound card, setting up the hardware, etc. Once you get to real measurements you will want to do an impulse response measurement and save the impulse. From the impulse you can do all sorts of post-processing to get frequency response, room parameters, etc. You can use any sampling rate your hardware supports.
The English pdf manuals are pretty good (especially considering the author is from Croatia) -- same page where you downloaded the software. Start at the beginning and it will walk you through calibrating your sound card, setting up the hardware, etc. Once you get to real measurements you will want to do an impulse response measurement and save the impulse. From the impulse you can do all sorts of post-processing to get frequency response, room parameters, etc. You can use any sampling rate your hardware supports.
Arta 1.6
ARTA 1.6 will be published on the 6th of december:
New/modified items:
Regards
Heinrich
ARTA 1.6 will be published on the 6th of december:
New/modified items:
- Transient recorder as storage scope with spectrum analyzer. Option for triggering both channels with predelay
- Imported wav and ascii files can be analysed as impulse response or time record
- New contour plot for directivity patterns
- All time windows now with additional 8 x zoom, zoom report on top of graph
- Command for setting cursor on position of max amplitude in Imp window
- 1/3 oct. band monitor simultaneously for both input channels
- Enhanced FR target filter generation (var Q, combined BP)
- Load target command accepts now files with arbitrary number of points (restriction reg. 720 points removed)
- online calculation of differences (measurement - overlay) in FR1, FR2 and Spa mode
- .....
Regards
Heinrich
I have downloaded the latest version of Arta. I have been reading the STEPS manual. I am using a M-Audio Mobile Pre USB sound card and a Dayton Electret Microphone. Is there an easier way to calibrate the input and output of the sound card and the microphone?
I have a calibration file for the microphone. Is there any way I can use that file to calibrate the mic instead of a calibrator?
Thanks
I have a calibration file for the microphone. Is there any way I can use that file to calibrate the mic instead of a calibrator?
Thanks
Hi,
I use a free version of the last ARTA software; Problem is that I cannot get a correct magnitude scale in the step response window thus getting a tiny image, despite pushing the gain max button (i.e. range from -263 to + 263)
A friend of mine does use a licensed version and is able to get a significant reduced magnitude scale (i.e. -25 to + 25)
Any thought about?
I use a free version of the last ARTA software; Problem is that I cannot get a correct magnitude scale in the step response window thus getting a tiny image, despite pushing the gain max button (i.e. range from -263 to + 263)
An externally hosted image should be here but it was not working when we last tested it.
A friend of mine does use a licensed version and is able to get a significant reduced magnitude scale (i.e. -25 to + 25)
Any thought about?
You need to go through the calibration setup, chapter 1 of the manual.
Same as above.
Go to File - Options and check the box that imported ascii and wav files should be treated as impulse responses.
Hi catapult,
My set-up is effectively still not calibrated (hopefully soon...) But I don't understand why it needs to be so to render the proper (readed) magnitude of a step response that is extrapolated from an impulse...?