Nice, this was a bit over my head at times but I have a firmer grasp on the process.
Thanks!
Thanks!
So if Dirac Live does full frequency range Time and Frequency domain correction- then one would assume it would do phase correction throughout the same range?No but I think there is a light version of Dirac with such a limit.
You can expect the video to take longer to process than audio. So audio delay is needed.…the problem with low frequency room corrections is it can result in rather large audio delays and subsequent audio/video lip sync issues. Until we have video processors than can deliberately add delay to the video signal to allow the audio DSP to catch up, its not much good to the HT guys.
Like when you go to your tv settings to adjust lip sync, its always 0-200ms audio delay. Like why not -200 to +200ms, and the minus is adding a delay to the video only? Most times the audio is lagging the video, why would I wanna add more audio delay?
They can delay but retarding the audio is only a feature of the latest AV products. A FIR filter 65K taps at 48 kHz will have a delay of more than a second. So you need to add a lot of frame delays to video. This is easy in a software player but not sure the hardware capable ones can go this far.All "surround" gear has had the ability to adjust audio delay since day dot.... (well at least since the 80's!) so I would not have thought that this would be a big deal!?
They can delay but retarding the audio is only a feature of the latest AV products. A FIR filter 65K taps at 48 kHz will have a delay of more than a second. So you need to add a lot of frame delays to video. This is easy in a software player but not sure the hardware capable ones can go this far.
Not trying to adjust audio delay. Trying to add video delay, to allow for possible seconds worth of outboard FIR filtering.All "surround" gear has had the ability to adjust audio delay since day dot.... (well at least since the 80's!) so I would not have thought that this would be a big deal!?
They can delay but retarding the audio is only a feature of the latest AV products. A FIR filter 65K taps at 48 kHz will have a delay of more than a second. So you need to add a lot of frame delays to video. This is easy in a software player but not sure the hardware capable ones can go this far.
Jriver has Lip sync Video delay adjustment, it does have a live input and DSP capabilities and some form of TV Integration but I have never used that to know if would work for you.Not trying to adjust audio delay. Trying to add video delay, to allow for possible seconds worth of outboard FIR filtering.
With a centred Impulse and rounded up to three decimal places or the nearest millisecondFIR filter with 64K taps (65536) at 48kHz would actually have a delay of 683ms (0.683 sec).
Bummer. That mini-DSP has always been a bit tempting; but without a computer in the audio chain the software options aren't a possibility. But that does explain why his after charts look better than any I've seen before.Dirac isn't close to these...watch the video as Mitch will explain it all.
That might make for a really interesting channel. Bar chats about audio. I find a lot of people's videos are much more engaging when they are talking to someone rather than just the camera...... I hate giving presentations as I am more of a sit at the bar and chat kind of guy......
For me, that is a big advantage to videos; I can coast through the bits that are too technical easier than when reading.Nice, this was a bit over my head at times but I have a firmer grasp on the process.
Thanks!
Bummer. That mini-DSP has always been a bit tempting; but without a computer in the audio chain the software options aren't a possibility.
I haven't had a chance to watch the video yet.
That might make for a really interesting channel. Bar chats about audio. I find a lot of people's videos are much more engaging when they are talking to someone rather than just the camera.
For me, that is a big advantage to videos; I can coast through the bits that are too technical easier than when reading.
Can you point to a specific sample and the detailed connection(s)? Thanks ...That computer can be as simple as a $50 RPi and there are very simple ways to get digital audio in to the RPi. In the simplest cases this can all be done in one USB DAC with digital input. Signal path looks something like: digital audio source -> DAC digital audio input (TOSLINK/SPDIF/AES) -> DAC USB to RPi -> processing in RPi -> RPi USB to DAC. Of course this requires just one USB connection between the RPi and DAC as USB is bi-directional.
Michael
Can you point to a specific sample and the detailed connection(s)? Thanks ...
Interesting that it still stands up. The best system I've personally owned was build around DRC-FIR + BruteFIR back in 2006 or so. DRC-FIR was as user-unfriendly as anything I've ever used with a myriad of lightly documented parameters to tweak and tune, but in this specific system (where I took care of early reflection and speaker positioning as best I could before correction), it was basically point-and-shoot. Although to be fair I *think* this was before Denis added the output plots which would probably help greatly in understanding the tweaking process.Dave, if you check out the video you will see that DRC-FIR makes the list for SOTA DRC If you decide to go digital XO's one day, then yah you would need to go with one of the commercial versions, which indeed are Windows programs.
Hi @dwkdnvr, cool! Yes, Denis got the basics of psychoacoustic filtering and frequency dependent windowing down pretty well. To help a bit with the user unfriendliness, gmad on diyAudio has some scripts and insights as well: https://www.diyaudio.com/forums/ful...ectrical-loudspeaker-correction-networks.htmlInteresting that it still stands up. The best system I've personally owned was build around DRC-FIR + BruteFIR back in 2006 or so. DRC-FIR was as user-unfriendly as anything I've ever used with a myriad of lightly documented parameters to tweak and tune, but in this specific system (where I took care of early reflection and speaker positioning as best I could before correction), it was basically point-and-shoot. Although to be fair I *think* this was before Denis added the output plots which would probably help greatly in understanding the tweaking process.
The one thing that I believe DRC-FIR lacked that Acourate emphasized was inter-channel similarity measure/correction. I think DRC-FIR was a single channel system where each channel was independently corrected to the target; I find it plausible that sacrificing 'absolute' compliance with the target to get better L/R matching could be an improvement.
Hi Mitch--thanks for the presentation. I was wondering why the latencies are so long? I was poking around and found this claim made by Brute-FIR:Hi @dwkdnvr, cool! Yes, Denis got the basics of psychoacoustic filtering and frequency dependent windowing down pretty well. To help a bit with the user unfriendliness, gmad on diyAudio has some scripts and insights as well: https://www.diyaudio.com/forums/ful...ectrical-loudspeaker-correction-networks.html
Also on diyAudio, wesayso (Ronald) has some excellent measurements and commentary on using DRC-FIR with his https://www.diyaudio.com/forums/full-range/242171-towers-25-driver-range-line-array.htm
Agreed on interchannel similarity. That was also noted in JJ's presentation and in the conclusions which I included in the video.
There is a widening gap between the commercial DRC applications that are being developed versus DRC-FIR. But if you like to tinker, and as you have already heard with your own ears, DRC-FIR can produce excellent results.
I guess I don't understand how dedicated 500MHz SHARC's can't seem to manage anything close? Granted my understanding is superficial, but it seems that most DSP speaker correction isn't coming close to this figure.With a massive convolution configuration file setting up BruteFIR to run 26 filters, each 131072 taps long, each connected to its own input and output (that is 26 inputs and outputs), meaning a total of 3407872 filter taps, a 1 GHz AMD Athlon with 266 MHz DDR RAM gets about 90% processor load, and can successfully run it in real time. The sample rate was 44.1 kHz, BruteFIR was compiled with 32 bit floating point precision, and the I/O delay was set to 375 ms. The sound card used was an RME Audio Hammerfall.
Interesting that it still stands up. The best system I've personally owned was build around DRC-FIR + BruteFIR back in 2006 or so. DRC-FIR was as user-unfriendly as anything I've ever used with a myriad of lightly documented parameters to tweak and tune, but in this specific system (where I took care of early reflection and speaker positioning as best I could before correction), it was basically point-and-shoot. Although to be fair I *think* this was before Denis added the output plots which would probably help greatly in understanding the tweaking process.
The one thing that I believe DRC-FIR lacked that Acourate emphasized was inter-channel similarity measure/correction. I think DRC-FIR was a single channel system where each channel was independently corrected to the target; I find it plausible that sacrificing 'absolute' compliance with the target to get better L/R matching could be an improvement.