Linkwitz links to the Grimm white paper. I believe the underlying reasoning is here
http://www.grimmaudio.com/whitepapers/speakers.pdf
In particular look at the comment at the end of section 2.2 where Bruno Putzeys says:
"The audibility of pre-ringing of >20 kHz filters in digital audio has always been a matter of some debate. Audibility of pre-ringing at 2 kHz is not."
Bruno's a smart guy, and he may have a point about pre-ringing at 2 kHz (although correcting a diffraction blip in any manner except proper cabinet design is a fool's errand).
However, the frequency response dip pointed out in Bruno's section 2.2 is not a filter artifact, but instead a product of the 30 degree vertical measurement showing the mids working against each other, and indeed the blue frequency response line shows a first cancellation right where one would expect it for an off-axis path length difference. The trains have wrecked, all right, but the same would happen with an LR4 filter. The summed impulse response shows a hump at that point because that's the tweeter output without the mids's output counteracting it. Note the very last impulse response showing extensive pre-ringing would be objectionable even though the steeper filters, reducing crossover overlap, would have a narrower dip in the frequency response. Pre-ringing at audible frequencies sounds quite unnatural, best described as a "scratchiness" around the range where the oscillation occurs. On the other hand, how many people listen to their speakers at 30 degrees off the vertical axis? Room contributions to the listening position will have sufficient delays that the Haas precedence effect might make this anomaly a moot point.
The argument fails when the wavelength is considerably larger than driver separation, for example an WMW arrangement crossed over at 100 or 200 Hz. Cabinet diffraction, apart from baffle step (broadband anyway so its impulse response is docile), is not an issue here, nor is driver dispersion, so off-axis response would be about as flat as on-axis. At that point a reasonable complementary FIR is your friend, because it both keeps mid frequencies away from nasty high-Q woofer resonances and limits mid excursion by keeping bass energy out of mid drivers.
Moreover, FIR design methods such as windowed sinc show better pre-ringing characteristics than the omnipresent Parks-McClellan, which really shouldn't be used for crossovers. I'll confess to a preference for distinct crossover and correction filters for that reason, since specifying a crossover frequency response characteristic mixed in with correction could produce the dubious results Bruno points out. This doesn't mean FIR crossovers are to be shunned, but using them properly means looking at the time domain response first then checking to make sure the frequency response is what you want.
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I would recommend to anyone trying to design a crossover (DSP or not) to read at least the enlightening paper from Mr Floyd Toole here
http://www.harman.com/EN-US/OurCompany/Innovation/Documents/White Papers/LoudspeakersandRoomsPt2.pdf
Jean Claude
http://www.harman.com/EN-US/OurCompany/Innovation/Documents/White Papers/LoudspeakersandRoomsPt2.pdf
Jean Claude
Dr Toole, actually. Reading his stuff is good for the soul.
Note particularly his treatment of resonances -- drivers before they're crossed over are always minimum-phase systems, so correcting anomalies can be done with judicious parametric EQ. He's also not a fan of fixing diffraction problems with EQ since that will mess up the room response.
I'm looking at the 2-channel analogue expansion.
There are 3 options:
1. Cheapest is no analogue volume. Volume is handled by the DACs but output level is fixed to some value.
2. More expensive: we fit an analogue volume chip like CS3310 or PGA2311. Problem is that max output level will be less than on the 8 main outs due to lower admissible rails (+/- 5 V vs +/- 8.5 V).
3. We fit an expensive PGA2310. This will blow the bill however.
What do you think (Steve)?
There are 3 options:
1. Cheapest is no analogue volume. Volume is handled by the DACs but output level is fixed to some value.
2. More expensive: we fit an analogue volume chip like CS3310 or PGA2311. Problem is that max output level will be less than on the 8 main outs due to lower admissible rails (+/- 5 V vs +/- 8.5 V).
3. We fit an expensive PGA2310. This will blow the bill however.
What do you think (Steve)?
Thanks Nick,
I certainly only need digi volume control just as for the standard 8 outputs - no different handling at all.
Personally I can't see why you would need a separate control for them.
Do need all the same output settings and controls as the others have of course. Looks like that is fine as there are 2 more blanks there in the existing control panel begging to be used
I see them as just two more channels to make 5 way work perfectly.
Nice and simple
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Thanks Nick,
I certainly only need digi volume control just as for the standard 8 outputs - no different handling at all.
Personally I can't see why you would need a separate control for them.
Do need all the same output settings and controls as the others have of course. Looks like that is fine as there are 2 more blanks there in the existing control panel begging to be used
I see them as just two more channels to make 5 way work perfectly.
Nice and simple
Najda actually has analogue volume on the 8 outputs. Analogue volume is managed by the CS3318 chip which has these +/-8.5 V rails that I mentioned. (So it's analogue volume digitally controlled).
Keeping the same type of volume is indeed the ideal way to make it - but it's also the most expensive one (i.e. that's point 3 in my previous post).
To avoid quantisation when the volume is turned down. You'd be surprised at how much quantisation can be covered by proper dither [1], and 2x resampling means it can be spread across an entire octave above the audible range.
[1] Just for kicks & grins, I bit reduced a test file to 6 bits; it was ghastly without dither, but with dither it sounded pretty darned good.
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Can I ask how this could be implemented in a stereo home theatre set-up? I am interested in getting blu-ray sound but understand that over spdif I would be getting only the lesser soundtrack as described;
"Dolby TrueHD has a companion DD5.1 track, usually at 640Kbs which would be sent out via SPDIF. DTS HD Master consists of a DTS core and additional audio information to make it lossless. Over SPDIF only the core would be sent."
But I see some are using HDMI over I2S, which this board has. My question therefore is how difficult this would be to implement, or would it even be possible with issues such as the copy protection?
Thx
P.S. If need be I could let the player do the decoding and use the analogue input of this board. Would be simpler at least.
"Dolby TrueHD has a companion DD5.1 track, usually at 640Kbs which would be sent out via SPDIF. DTS HD Master consists of a DTS core and additional audio information to make it lossless. Over SPDIF only the core would be sent."
But I see some are using HDMI over I2S, which this board has. My question therefore is how difficult this would be to implement, or would it even be possible with issues such as the copy protection?
Thx
P.S. If need be I could let the player do the decoding and use the analogue input of this board. Would be simpler at least.
I think the main issue is, how are you going to get a stereo content out of your blu-ray player?
Blu-ray content is usually multichannel and what you want is a stereo signal. You cannot just extract the Left and Right components out of the L-C-R-SL-SR-SUB mix and hook these to a stereo sound system. You need to perform a downmix of all channels into a pair of stereo signals.
Can your decoder perform this downmix? That's the first thing I would look at.
Second thing is that Najda doesn't include a DD decoder. So Najda won't understand the DD stream on SPDIF - this is just not going to work.
It would be a pity to go through an additional DAC/ADC stage if what you're after is specifically blu-ray
I searched this thread for info on the presets and found post 556. On my computer using Najda Under Control, I am unable to save a file to one of the nine presets. I know, I am probably not doing it right.... Could I get a nudge in the right direction? Thanking you in advance for your patience.
Aha - did not understand that.
So best way to get all 10 outputs controlled by the same controller is???
Hi Nick,
Yeah, I want to avoid the DAC-ADC route if possible.
Unfortunately down-mixing is inevitable when listening to blu-ray codecs over stereo, but I would hope it is still lossless audio.
Although there is no DD decoder, I do not think this is a problem. All new players will decode before outputting over HDMI, if the option is selected (output as PCM or bitstream). Correct me if this is wrong anyone!
Assuming the decoded output carried on HDMI is acceptable, is it theoretically possible to use a HDMI connection to this board?
Yeah, I want to avoid the DAC-ADC route if possible.
Unfortunately down-mixing is inevitable when listening to blu-ray codecs over stereo, but I would hope it is still lossless audio.
Although there is no DD decoder, I do not think this is a problem. All new players will decode before outputting over HDMI, if the option is selected (output as PCM or bitstream). Correct me if this is wrong anyone!
Assuming the decoded output carried on HDMI is acceptable, is it theoretically possible to use a HDMI connection to this board?
Extract volume to dac board
Hi, I just ordered a Najda
As I plan to use it in combination with my BIII dac I'd like to be able to "tap in" to the analog volume control on Nadja and use it for controlling the volume on the dac board. I think someone had the same question earlier in this thread, but I don't think I've seen any answer.
So, is it possible to extract the volume setting from Nadja (maybe be reading a voltage somewhere)? Then it should be possible to read this with another microprocessor and communicating this to the the dac board using I2C.
Best regards,
Mattias
Hi, I just ordered a Najda
As I plan to use it in combination with my BIII dac I'd like to be able to "tap in" to the analog volume control on Nadja and use it for controlling the volume on the dac board. I think someone had the same question earlier in this thread, but I don't think I've seen any answer.
So, is it possible to extract the volume setting from Nadja (maybe be reading a voltage somewhere)? Then it should be possible to read this with another microprocessor and communicating this to the the dac board using I2C.
Best regards,
Mattias
Would this pass for the +-12V and 5V required (have one already)?
Adjustable Bench Power Supply 0-30 VDC 0-5A, 1A 5VDC fixed CSI 3005X5
0-30VDC 0-3Amp / 5VDC 1Amp, Dual Output Bench Power Supply
Adjustable Bench Power Supply 0-30 VDC 0-5A, 1A 5VDC fixed CSI 3005X5: Amazon.co.uk: DIY & Tools
1A, 5VDC fixed output is on the rear panel.
Adjustable Bench Power Supply 0-30 VDC 0-5A, 1A 5VDC fixed CSI 3005X5
0-30VDC 0-3Amp / 5VDC 1Amp, Dual Output Bench Power Supply
Adjustable Bench Power Supply 0-30 VDC 0-5A, 1A 5VDC fixed CSI 3005X5: Amazon.co.uk: DIY & Tools
1A, 5VDC fixed output is on the rear panel.
Hi Mark,
FIR presets can involve quite many coefficients. For this reason they can only be stored in slots 1 to 3.
IIR presets can be stored anywhere, slots 1 to 9.
If you have a custom FIR preset that you want to store, then you need to overwrite one of the 3 factory presets that are all pass-through.
So if we go the route PGA2320, then the 2 additional analogue outs will be perfectly integrated. I.e. there will be no difference compared to the board 8 outs. It's a bit more expensive but I think that's the way to do it.
If you can get a I2S signal out of a HDMI receiver, then it's going to work without a doubt, like with any USB/I2S converters.
Hi, I just ordered a Najda
As I plan to use it in combination with my BIII dac I'd like to be able to "tap in" to the analog volume control on Nadja and use it for controlling the volume on the dac board. I think someone had the same question earlier in this thread, but I don't think I've seen any answer.
So, is it possible to extract the volume setting from Nadja (maybe be reading a voltage somewhere)? Then it should be possible to read this with another microprocessor and communicating this to the the dac board using I2C.
Best regards,
Mattias
Hi Mattias,
Rikkitikitavi (ehm... not sure about spelling) suggested to re-inject the analogue signal out of external DACs back onto the board. I think it's a nice way to explore - that would allow proper volume control with external DACs at little cost (just removing one coupling capacitor for each output).
You would also keep the benefit of individual analogue output level on each channel.
Otherwise volume commands are transferred via SPI.
I think it should work alright. 5V 1A at the rear is great. If this power supply can also output +/- 12 V on the front connectors, then it's perfect.
Best,
Nick
It is great that it has presets, but 3 for FIR sounds like not much, would it be much more costly to increase the number of presets?
Thx