Here is the actual room air sound SPL response measured at my listening position; you would please visit my post here for the details of present/latest system setup including the Fq-SPL response of air sound at LP and "measured" at upstream digital domain, analog line-levels as well as SP high-levels;Do you have the measured (not calculated) responses NF or at the LP?
- Frequency response measurements by "cumulative white noise averaging": #392, #404, especially the end portion of #297(remote thread) by Dr. Floyd Toole, #315(remote thread) by Dr. Floyd Toole, #125(remote thread)
- Where in my multichannel multi-driver (multi-way) multi-amplifier stereo system should I measure/check frequency (Fq) Responses? #393
- Frequency (Fq) responses in the completed system measured by using “cumulative white noise averaging method” under the present standard crossover configurations and relative gains_Part-1_Fq Responses in EKIO’s digital output level: #394
- Frequency (Fq) responses in the completed system measured by using “cumulative white noise averaging method” under the present standard crossover configurations and relative gains_Part-2_Fq Responses in DAC8PRO’s analog output level: #396
- Frequency (Fq) responses in the completed system measured by using “cumulative white noise averaging method” under the present standard crossover configurations and relative gains_Part-3_Fq Responses in amplifiers’ SP output level before protection capacitors: #401
- Frequency (Fq) responses in the completed system measured by using “cumulative white noise averaging method” under the present standard crossover configurations and relative gains_Part-4_Fq Responses in amplifiers’ SP output level after protection capacitors: #402
- Frequency (Fq) responses in the completed system measured by using “cumulative white noise averaging method” under the present standard crossover configurations and relative gains_Part-5_Fq Responses in actual SP room sound at listening position using one measurement microphone: #403
- Frequency (Fq) responses in the completed system measured by using “cumulative white noise averaging method” under the present standard crossover configurations and relative gains_Part-6_Summary, discussions, and a little step forward: #404, #405-#409
As for present Fq response (calibration curve) of my "specially selected in 2008" BEHRINGER ECM8000 measurement microphone, please refer to my post here.
- Frequency response of my BEHRINGER ECM8000 measurement microphone (specially selected unit in 2008): #831
In my post #404 on my project thread, I wrote;
Throughout these careful measurements, I confirmed and validated that primitive "cumulative (recorded) white noise averaging method" is really powerful and reliable in terms of;
1. the method is universally applicable in the stages of digital out of crossover software (EKIO), DAC's analog out, amplifier SP out, and of course in the actual room SP sound,
2. the method is accurate, sensitive and reproducible, having little or no statistical fluctuation, because of the FFT averaging analysis on the "accumulated rich data" of the recorded sound,
3. the recorded "white noise tracks" can be re-analyzed any way, anytime, afterwards,
4. flexible mix-paste (sound mixing) can be done to virtually simulate any combination of the channels, especially in amplifiers' SP out signals before going into SP drivers,
5. if needed, the environmental "continuous room back ground noise" can be reduced/removed by the Adobe Audition's "noise capture - noise reduction" function,
6. if needed, suitable gain/level adjustment can be applied for "level matched comparison" of Fq response shapes between the different series of the recorded data,
7. flexible and suitable FFT size (as smoothing intensity) can be selected depending on the frequency zone of interest.
Throughout these careful measurements, I confirmed and validated that primitive "cumulative (recorded) white noise averaging method" is really powerful and reliable in terms of;
1. the method is universally applicable in the stages of digital out of crossover software (EKIO), DAC's analog out, amplifier SP out, and of course in the actual room SP sound,
2. the method is accurate, sensitive and reproducible, having little or no statistical fluctuation, because of the FFT averaging analysis on the "accumulated rich data" of the recorded sound,
3. the recorded "white noise tracks" can be re-analyzed any way, anytime, afterwards,
4. flexible mix-paste (sound mixing) can be done to virtually simulate any combination of the channels, especially in amplifiers' SP out signals before going into SP drivers,
5. if needed, the environmental "continuous room back ground noise" can be reduced/removed by the Adobe Audition's "noise capture - noise reduction" function,
6. if needed, suitable gain/level adjustment can be applied for "level matched comparison" of Fq response shapes between the different series of the recorded data,
7. flexible and suitable FFT size (as smoothing intensity) can be selected depending on the frequency zone of interest.
Dr. Toole kindly wrote here in response to my inquiry;
> If properly done both swept tone and noise analysis should give identical answers. It is a choice. The principal difference is in the heating of the drivers in sustained tests at high sound levels - power compression. Low frequencies require longer averaging times.
> If properly done both swept tone and noise analysis should give identical answers. It is a choice. The principal difference is in the heating of the drivers in sustained tests at high sound levels - power compression. Low frequencies require longer averaging times.
Yes, you would please carefully read my post here.Are you using 96kHz sampling rate to apply these 25kHz brickwall filters?
- Summary of rationales for "on-the-fly (real-time)" conversion of all music tracks (including 1 bit DSD tracks) into 88.2 kHz or 96 kHz PCM format for DSP (XO/EQ) processing: #532
Recently (nowadays) I almost always convert (on-the-fly) sampling rate of all the tracks of my digital music library (ref. here) into 88.2 kHz 24 bit PCM by JRiver MC's DSP Studio settings (see the below attached screen capture); 88.2 kHz (i.e. up to Fq of 44.1 kHz in L & R channel) is much more than enough since I have -48 dB/Oct low-pass (high-cut) Linkwitz-Riley filters at 25 kHz, as you pointed.
Just for your convenience and reference, please find here and here (exactly the same contents) the Hyperlink Index for my PC-DSP-based multichannel multi-SP-driver multi-amplifier fully active stereo audio project.
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