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UliBru

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  1. Hello Phil, Acourate expects an ASIO driver for a soundcard. In your case you run two different devices (Intel HD audio spdif output and USB input for mic). In such a case please use Asio4All as an Asio wrapper for Windows sound. In the Windows sound dialog you can select the USB mic as standard input and Intel HD spdif as standard output. It also makes sense to open the Asio4All control panel by the according button in the logsweep recorder, to click the tool button in the control panel and to inspect the selected input/outputs in the tree displayed at the left side of the control panel. Select a buffer size of 1024 samples. Then the IO channel should also be listed correctly in the logsweep recorder and you can record the sweep as intended. Cheers Uli
  2. Hello Jaques, if the 65 Hz simply get attenuated by standing wave cancellation the solution is to avoid the cancellation. Opening the window destroys much of the wave reflected from the back wall. There are different solutions: - take the back wall away (like opening the window), but this is cold in winter time :-) - destroy the energy of the wave at the back wall by absorption (e.g. plate absorber, Helmholtz absorber, bass traps) - actively use subwoofers at the back wall with delayed signal and opposite polarity (double bass array) If you record your speaker with and without subs you will get two pulse responses (with/without sub). You can subtract the pulse response without sub from the response with sub and thus you get the pulse response of the sub but with the correct time relationship. Then you can convolve each pulse response with a 65 Hz sinewave. You will clearly find the low amplitude in the result as a steady-state oscillation. But now you can think about how to shift the sub signal so that the sum of the speaker and the sub gets the maximum amplitude. If the result is not good enough you can place the sub at another position and repeat the procedure. Finally you should find an acceptable position where speaker and sub add as desired, at least with some improvement. At the end you can feed speaker and sub with the input signal but with the detected required delay and gains. This can then be done by DSP. I hope the recipe is clear. I have already carried out such installations with good result.
  3. In Acourate, anything above the target curve (not necessarily a line) gets reduced and anything below the target curve does NOT get corrected. The resulting filter is normalized to max. 0 dB for the filter frequency range.
  4. I have even never used Reverberate :-). So I have just brought Reverberate on the list because I got a report by another Acourate user with a Mac: He also gave a hint to pay some attention on the setting: JRiver Mediacenter also has a normalization checkbox (on by default). By normalization bit accuracy may get lost.
  5. 1) According to the website of Roland there is an ASIO driver for the UA-5 soundcard. Even if a soundcard is without ASIO driver it can be used by the Asio4All driver, which simulates an ASIO interface. 2) In most cases I use 48 kHz for recording. A samplerate of 96 kHz allows to record up to 48 kHz = fs/2. But most users do not have a microphone and a speaker good for this frequency range. How to correct something that you cannot measure very well? 3) there is a free logsweep recorder AcourateLSR2 on the Acourate website. You can also download the Acourate trial, it allows a recording too. 4) an efficient test of different plugins can be carried out by using test filters. I prefer a test filter that simply creates an echo. Thus you can test the filter data format and also the function. I can create such filters for you. Just tell me what format you need.
  6. There is a trial version on the website. But because of bad experiences in the past with hackers it is pretty much stripped down. You can also try the test offer
  7. Mitch has written in his article: Your .frd file is also a text file with ascending pairs of frequency and amplitude in each line. Thus simply rename the file extension to .txt and follow the article.
  8. Hello Alex, please see a post by Bob Stern at http://www.computeraudiophile.com/f3-article-comments/article-acourate-digital-room-and-loudspeaker-correction-software-walkthrough-16452/#post234728 about Mac plugins. In addition I know that Liquid Sonic Reverberate or SIR2 are other possible convolution plugis. As I do not use a Mac I cannot tell you which plugin is the best. Acourate supports them by generating wav files as correction filters. I guess this format is accepted by all of them. It is no problem to supply a required bit depth (I would prefer 64 bit float).
  9. So what? Either you can explain your statement about Dirac's mixed phase filters and give some more backgound information or you cannot. If you can't then you prove by yourself just to churn out hollow phrases.
  10. I truly believe that 99.9% of Acourate users do not have a problem with high frequency correction. Personally I would not run without it. But it is also possible to limit the correction, see the picture.
  11. Can you please explain mixed phase filters and why they are better? What are the other filter types you compare to and why are they worse? What filters do you exactly know are used by Acourate?
  12. Acourate is the toolbox to measure the audio system and to calculate FIR filters. The filters have to be applied by a convolution software. The target is to simply get good filters (or very good or damn good). Applying the filters is a second step. There are different convolution programs on the market. Either as plugins (e.g. FooConvolver), stand-alone pograms (AcourateConvolver, Brutefir, JConvolver) or embedded in a player (JRiver Mediacenter, HQPlayer). You can get free programs or you have to purchase them. A first test has shown that the Signalyst HQPlayer can apply Acourate filters also for DSD. It is anyway good to know that it works. I do not know if Signalyst will reveal details.
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