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Old 4th January 2013, 10:55 PM   #241
Pano is offline Pano  United States
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Some reverb impulses I have (convolved reverb) are 80K samples long at 44.1Khz. If they are truncated to less, you lose a lot of the room/venue sound. That does slow the convolution engine down a little, but not too bad.
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Old 4th January 2013, 11:15 PM   #242
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Quote:
Originally Posted by Pano View Post
Some reverb impulses I have (convolved reverb) are 80K samples long at 44.1Khz. If they are truncated to less, you lose a lot of the room/venue sound. That does slow the convolution engine down a little, but not too bad.
Yes. The last output of the reverb will occur 79999 samples after the last input, or 1.81 sec after the input stops. Add to that the latency of the processing, I/O buffering, etc. The processing can be speeded up considerable by using partitioned convolution.
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Old 5th January 2013, 07:02 AM   #243
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What's the difference between writing DSP on a PC and a dedicated DSP?
On a DSC or DSP you spend more time looking at dissasemblies to see if the compiler generated the instructions you want and less time worrying about interruptions to real time processing.
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Old 5th January 2013, 11:16 AM   #244
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On a DSC or DSP you spend more time looking at dissasemblies to see if the compiler generated the instructions you want...
Doesn't sound like my idea of fun!
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...and less time worrying about interruptions to real time processing.
One would hope that between the PC, sound card and a suitable buffering system, the PC can go off and do what it wants whenever it wants, within reason. They use PCs in professional recording studios these days, so I'm guessing that they've got it covered.

How are you going to link your audio source to the DSP card? Can you avoid opening a whole new can of worms, worrying about real or imaginary jitter, resampling etc.?
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Old 5th January 2013, 11:50 AM   #245
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How are you going to link your audio source to the DSP card?
I2S in all communication with either sources or targets.
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Old 5th January 2013, 12:03 PM   #246
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I know of two GPU convolution engines, GPU Impulse Reverb VST | NVidia/ATI GPUs used as DSP for convolution reverb calculation using OpenCL and Reverberate by Liquidsonics using CUDA.

GPUs are really beasts these days. I'm a finishing grad student in lattice QCD. Supercomputers for high energy physics (and everything else I assume) are all based on GPUs now. The cluster here in Frankfurt has about 700 ATI cards in it.

I haven't had a time critical application so I'm still just using my i5 for convolution.
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Old 5th January 2013, 12:11 PM   #247
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Doesn't sound like my idea of fun!

One would hope that between the PC, sound card and a suitable buffering system, the PC can go off and do what it wants whenever it wants, within reason. They use PCs in professional recording studios these days, so I'm guessing that they've got it covered.
Ceratinly there have not been problems with the PC based UE provided the PC is within the required specifications.
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Old 5th January 2013, 01:41 PM   #248
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FFT convolution is not the solution to all problems :-(

What is the relation between FFT length and frequency resolution?:

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The important point is that at a fixed sampling rate, increasing frequency resolution decreases temporal resolution. That is the more accurate your measurement in the frequency domain, the less accurate you can be in the time domain. You effectively lose all time information inside the FFT length.
I will stay in time domain.

In the filter case, for each passband I will need to downsample to some apropriate frequency related to upper filter frequency. I am thinking 24 dB per octave filter slope, sampling at 3 octaves above should be ok I guess. Halving 96 kHz I get these frequencies:

2.5k - 20k : 96 kHz
300 - 2.5k : 24 kHz
80 - 300 : 3 kHz
30 - 80 : 750 Hz

Accurate Estimation of Minimum Filter Length for Optimum FIR Digital Filters. Koichi Ichige, Mamoru Iwaki and Rokuya Ishii. Interesting reading on FIR filter length optimization. Filter length is determined by the amount of ripple you can tolerate. How much ripple is acceptable in hifi?
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Old 5th January 2013, 01:58 PM   #249
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One would hope that between the PC, sound card and a suitable buffering system, the PC can go off and do what it wants whenever it wants, within reason. They use PCs in professional recording studios these days, so I'm guessing that they've got it covered.
Windows (I belive we are talking Microsoft here) has absolutely no realtime support whatsoever. One has to put in a lot of effort make it perform stable in a DSP application.

However, in our world it is acceptable with a few mishaps as long as you don't destroy the speakers.

Personally I will not enjoy listening to a system not knowing exactly what it is doing.
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Old 5th January 2013, 02:11 PM   #250
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Originally Posted by more10 View Post
In the filter case, for each passband I will need to downsample to some apropriate frequency related to upper filter frequency. I am thinking 24 dB per octave filter slope, sampling at 3 octaves above should be ok I guess. Halving 96 kHz I get these frequencies:

2.5k - 20k : 96 kHz
300 - 2.5k : 24 kHz
80 - 300 : 3 kHz
30 - 80 : 750 Hz
You should add a short FIR to do a brickwall LP at Nyquist nonetheless, to avoid any problem.
That is what Rainer Thaden does in the Four Audio HD2:

http://www.four-audio.com/en/technic...e-the-hd2.html
http://www.studitech.ru/resque/manua...ES32_rev-5.pdf


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Interesting reading on FIR filter length optimization. Filter length is determined by the amount of ripple you can tolerate. How much ripple is acceptable in hifi?
You can see that when playing with different window functions (and impulse length) in rePhase.
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