Bottlehead Forum
Bottlehead Finished Products => Legacy Products => Bottlehead DAC => Topic started by: Doc B. on October 13, 2014, 11:48:48 AM
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Here's a description of the Bottlehead DAC -
Nearly four years of design and research headed by John Swenson has brought us to the release of this very special Digital to Analog Converter, the Bottlehead DAC.
For those of you who just need the highlights, here are the bullet points-
• 32 bits, up to 384kHz USB, 192kHz S/PDIF
• USB, coax digital, TOSLINK inputs
• Approx 2V output via RCA jacks
• State of the art digital filtering
The Bottlehead DAC can handle 16,24 and 32 bit audio data at sample rates of 44.1, 48, 88.2, 96, 176.4, 192kHz, and (via USB) 352.8 and 384kHz. Sample rates through 192kHz can be handled by the USB (asynchronous UA2), coax digital and TOSLINK optical inputs (assuming your TOSLINK source is capable of transmitting 192 kHz). No switches need to be thrown to select different sample rates, the sample rate is automatically sensed by the DAC and displayed on a classic 7 segment red LED display inside the 4.5†x 3†x 7.25†(approx.) cabinet.
The DAC chip used is the Texas Instruments Burr Brown PCM5142 Delta Sigma DAC with 112dB S/N, exceptionally low out of band noise (for reduced EMI and aliasing), integrated negative charge pump line driver (which means no output coupling capacitors, ICs or tubes, and a low output impedance) and intelligent muting (no pops). On its own the PCM5142 is a remarkable sounding DAC. However John has created supporting circuitry and firmware around this DAC that take its sonic signature to an even more refined state. A Field Programmable Gate Array chip and an XMOS processor are employed to achieve this.
As John puts it,
The FPGA is the "brains" of the whole thing. It is the S/PDIF decoder, buffer controller, output format generator, adjustable oscillator interface, display controller and several others I can't think of at the moment. The XMOS processor is for the USB interface.
One of the most significant aspects of the DAC is the oversampling reconstruction filter that John has designed. We actually turn off the filter that comes in the DAC and use John’s instead. Here’s John on filters:
For those few that haven't heard my thoughts on digital filters, a quick synopsis: As far as I know all digital filters inside DAC chips use special DSP "tricks" to get really good spec sheet numbers at low cost, BUT these tricks wreak havoc with the "musicality" of the sound. This is why people like NOS DACs. BUT with a NOS you still have the aliases which cause a "dirty" sound. So what I am doing is relaxing the requirements of the filter, so it doesn't have to meet those insane numbers in the DAC spec sheets, the result is I can implement a simple filter that fits in an inexpensive FPGA.
The result is something most people have never heard, the musicality of NOS without the "dirty" sound of the aliases. It's simply stunning. Listening to your favorite music with this filter is a whole new experience. There HAS to be a filter, if you just send a 44.1 or 96 or whatever, the DAC chip will use its own. The only way to turn it off is to upsample to 352 or 384.
BTW a reconstruction filter is not evil; it is a very good thing. It's just that all the DAC chip implementations have made compromises that adversely affect the sound quality. That is what my filter is all about, doing it right so you get the good things from a proper reconstruction filter without the bad side effects of the DAC chip implementations.
As to buzz words, the filter is intermediate phase. A linear phase filter has equal amounts of pre and post ringing but fairly short amount of time in each. The minimum phase has no pre ringing, but in order to get the same over all functionality it has to have a much longer amount of post ringing. The intermediate phase splits the difference and has a small amount of pre ringing and post ringing that is a little bit longer than linear phase but much less than minimum phase. This is also deliberately a short filter. That means the amount of time it spends ringing is a very short amount of time compared to other filters. The tradeoff is that the ultimate amount of alias reduction is only about 80db, whereas those in modern DAC chips are going to 130 db or more. This is the tradeoff I mentioned, by not aiming for nearly as much ultimate image rejection I get a filter that sounds much better.
There is no dither involved; it simply is not needed in the DAC.
The overall summary is that not having an oversampling reconstruction filter is NOT the best sound. It may be better than some of the horrible sounding filters used in DAC chips, but a properly done oversampling reconstruction filter sounds way better than either no filter or the ones built into the DAC chips. The secret is to find the RIGHT filter and how to implement it without degrading the sound. I have been working on this for many years and what is in the Bottlehead DAC is the culmination of all that research and listening.â€
In order to fully realize the benefits of this filter the DAC needs to be constructed in such a manner that the benefits are not masked by power supply noise, jitter, or unnecessary buffer or gain stages. And so these areas have received special care as well.
This DAC has 6 ultra low noise regulators. It has ground plane isolation with special high quality isolators on the signals. This does a very good job minimizing the ground plane noise from the digital chips from getting into the analog side of things. This gives a marvelous very "black" background. It also significantly cuts down on the outside world affects such as source component interaction and cable affects. It doesn't completely eliminate them, but it does significantly reduce these interactions.
And we have future plans to offer a battery supply in kit form that improves the black background even further.
As for jitter:
There are two different clock "topologies" for a DAC:
Source is master, DAC is slave
Source is slave, DAC is master
What matters is the jitter at the DAC chip (in the DAC). Having a fixed frequency clock in the DAC, right next to the DAC chip is the best way to implement this. You can do this when the DAC is master; it has the "master clock". USB can do this in "asynchronous mode". The BH DAC uses asynchronous mode so it can do the DAC as master. In this mode the source (usually a computer) sends the data out, but the DAC can tell it to speed up or slow down so the average data rate matches the clock in the DAC.
There is another USB mode called adaptive, in which DAC is the slave, but the BH DAC does not use this. SOME other DACs use this mode.
The S/PDIF inputs (coax and optical) just work with the source as master and the DAC as slave. Thus the DAC has to somehow synchronize its clock to the data rate from the source. This is traditionally done with a device called a PLL, which is built in to all the S/PDIF receiver chips. PLLs have much higher jitter than a good fixed frequency clock. The BH DAC does not do it this way. It cleans up the S/PDIF signal, and sends it into an FPGA (field programmable gate array) which does the S/PDIF decoding. But the special part is a digitally controlled ultra low jitter clock. This is almost as good as the best fixed frequency clocks. The FPGA tells this clock to speed up or slow down so it is synchronized to the average data rate of the source.
The result of this is that both S/PDIF and USB produce ultra low jitter to the DAC chip. This combination of ultra low jitter from BOTH S/PDIF and USB doesn't exist in any other DAC. On other DACs one or the other will be significantly worse than the other input.
I have been listening to all three inputs (S/PDIF coax and TOSLINK and USB) and can tell hardly any difference between them. There are some very slight differences, but they are very small and hard to determine. There is only a very tiny difference in sound with different source components (different computers, OS etc, at least the ones I have on hand) I've spent a lot of work trying to decrease these sensitivities and I think I've done a pretty good job. Normally I would say that a good asynchronous USB implementation will beat a good S/PDIF implementation, BUT in this case I have spent a huge amount of effort (and I really do mean huge) making an exceptional S/PDIF input. The result is that the SQ from all the inputs should be very similar. The upshot is that if you already have an S/PDIF output from whatever, you can keep on using it with the BH DAC and know that you are not loosing anything compared to one of the other inputs.
Interestingly when we started this project with John we just assumed that the DAC would have a tube output stage. We even built two early prototypes with tube outputs that used our best technology. But the introduction of the PCM5142 into the mix radically altered our perspective. The negative charge pump line driver in the chip (nope, we don’t really understand how it works either) sounded so good that anything we added after it just colored the sound. So there is NO active output stage, the output from the DAC drives a simple final analog RC filter, which is what mostly determines the output impedance of around 500 ohms.
Another great feature of this DAC is upgradeability. There are lots of interesting ideas brewing for future upgrades:
The firmware for the FPGA is stored in a special flash memory that is soldered to the board. When new firmware is needed you simply ship your DAC to Bottlehead and we will install the upgrade for you for a modest fee.
I'm designing the USB input to handle 8 channels, but the hardware and firmware to couple multiple DAC boxes together will not be available at first, it's going to take some time to get that ready.
I HAVE built a really good DSD DAC, and have been comparing the two and I can say that the BH DAC with my custom filter sounds very similar to the really good DSD DAC. I have been coming up with a scheme to add the DSD output DAC to the aux connector, so there MIGHT be a DSD add on board later on.
So why is this not a kit?
As you may gather from the description, this is not just a DAC chip slapped on a minor variant of a developer’s board made to compete with the cheap kits on ebay. There are many complex connections between many chips, regulators and other components on a multilayer board, and firmware must be installed and tested after assembly. Thus the DAC PC board needs to be manufactured in a facility that specializes in that kind of production. That board costs many hundreds of dollars. We got high blood pressure just thinking about the first phone call from an unlucky customer who blew that expensive board due to a wiring error or a slip with the meter probes, and we decided that for everyone’s sanity it made the most sense for us to assemble these DACs in house.
There will be at least one supporting kit in the future, a battery supply. We have already tested the idea and it creates an audible improvement in sonics over the stock universal wall wart supplied with the DAC.
And then there is the Nixie tube display that we have been dreaming about. That just might make it as a kit too.
OK so how so how do you get one?
Because of the high PC board production costs, and high volume parts purchases we must make to qualify for some necessary software licenses, we find it necessary to have twenty people commit to the first production run of these DACs. The buyer will commit to a $1550 order (plus $15 for shipping in the US, $25 outside the US). We are working to have a web page up to start taking pre-orders this week.
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Can't find a picture of this anywhere so far (?)
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Try the thread titled prototype images
http://bottlehead.com/smf/index.php?topic=6875.15 (http://bottlehead.com/smf/index.php?topic=6875.15)
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What is the "tap" rate with sampling ?
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Good stuff. I have a question on the ground plane isolation, does that include the SPDIF and USB input? i.e. any isolation transformers or optical isolators to separate the computer ground from the analog audio out ground?
Cheers,
Mark
P.S. Since i seen the first prototype with the segment displays i have been playing with low voltage IV-3 VFD tubes. ....you really need to make a kit :D
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Looks sexy. I thought I saw a HDMI port on the prototype? Does this mean my SACD collection won't be pumped into this device? Will it do DSD over USB in that DoP embedding DSD into PCM, or will DSD files need to be converted?
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Put it up for order, I'll buy one.
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Will there be options for custom enclosures or building your own and transferring the internals?
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Will there be options for custom enclosures or building your own and transferring the internals?
You are more than welcome to do whatever you like with the DAC, but this would void the warranty.
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Looks sexy. I thought I saw a HDMI port on the prototype? Does this mean my SACD collection won't be pumped into this device? Will it do DSD over USB in that DoP embedding DSD into PCM, or will DSD files need to be converted?
Yes there is an HDMI jack, but it is NOT an "HDMI stream", it is the aux connector for attaching other devices in the future. I needed 4 high speed differential signals and the easiest way to do that is with HDMI connectors and cables. Other solutions cost way more money.
So I'm sorry if it gets confusing, thinking you can hook it up to a home theater receiver etc over HDMI, but it is the way to do what I need without adding hardly any extra cost to the product. I didn't want to add a lot of money for an aux connector that many people would not use.
As far as DSD goes, this version is NOT a DSD DAC, it is strictly a PCM DAC. I am working on a really good DSD DAC that will be designed to plug into the aux port. It will get data from any of the inputs, either DoP or ASIO. This DSD add on is still under development, it will be a while before it is up and running.
John S.
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Good stuff. I have a question on the ground plane isolation, does that include the SPDIF and USB input? i.e. any isolation transformers or optical isolators to separate the computer ground from the analog audio out ground?
Cheers,
Mark
The coax input has its own isolation transformer, and the TOSLINK input is of coarse optically isolated. ALL of digital side of things (FPGA, XMOS chip, coax receiver etc) are on a separate ground plane from the DAC chips. I use GMR (Giant MagnetoResistive) isolators, I find they offer the lowest jitter and lowest noise of any of the isolator technologies. The "clean" side cconsists of the DAC chip, main oscillator and reclocking flops. The I2S data from the FPGA goes through the isolators, gets reclocked and fed to the DAC chip.
John S.
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What is the "tap" rate with sampling ?
I'm not quite sure what you are asking, the filter is an oversampling filter whose output frequency is 352.8 or 384. The "X" number (2X 4X etc) is determined by the input sample rate. If the input is 44.1 or 48 it is 8X, if the input is 88.2 or 96, it is 4X, if the input i s 176.4 or 192 it is a 2X, if the input is 352.8 or 384 no filter is used at all.
I hope that covers your question, let me know if that wasn't what you were after.
John S.
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The coax input has its own isolation transformer, and the TOSLINK input is of coarse optically isolated. ALL of digital side of things (FPGA, XMOS chip, coax receiver etc) are on a separate ground plane from the DAC chips. I use GMR (Giant MagnetoResistive) isolators, I find they offer the lowest jitter and lowest noise of any of the isolator technologies. The "clean" side cconsists of the DAC chip, main oscillator and reclocking flops. The I2S data from the FPGA goes through the isolators, gets reclocked and fed to the DAC chip.
John S.
Fantastic as that is one area i have always had issues with, you really have thought of everything :)
For the DC supply i assume the wallwart supplied will be a switching unit? I'm planning to build a nice power supply for it and Paul confirmed its needs 6-9V ~400mA. Previously i have been powering my DAC's through a USB isolator fed from a TI TPS7A4700, the rest of the power supply before it is a clone of your squeezebox supply. My plan is to reconfigure the regulator output so i would be interested to hear your thoughts on the most appropriate voltage, and regulator IC to use for the job.
Thanks again,
Mark
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For the DC supply i assume the wallwart supplied will be a switching unit?
Yes, we will supply a switching supply.
I'm planning to build a nice power supply for it
The battery power supply sounded quite a bit better than the CLCLCLC power supply loaded with common mode chokes and very tightly regulated. I wouldn't go too nuts on linear supplies until you hear it.
-PB
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I wouldn't go too nuts on linear supplies until you hear it.
The battery upgrade kit is a given, i have experimented with SLA's and Li-Pos and seen lower noise floors with them. This supply is just to fill the gap until its ready for prime time ;)
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I'm not quite sure what you are asking, the filter is an oversampling filter whose output frequency is 352.8 or 384. The "X" number (2X 4X etc) is determined by the input sample rate. If the input is 44.1 or 48 it is 8X, if the input is 88.2 or 96, it is 4X, if the input i s 176.4 or 192 it is a 2X, if the input is 352.8 or 384 no filter is used at all.
I hope that covers your question, let me know if that wasn't what you were after.
John S.
FIR tap refers to a quantifiable array of memory based delay filters, intrinsically minimal with off shelf chips. Basically the more taps, the better the source wave is reconstructed in memory. This too, is a component of jitter, beyond the more basic timing issues. So my question reworded is, are you relying on the DAC chip's organic wave reconstruct, or is their added tap filtering going on ?
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FIR tap refers to a quantifiable array of memory based delay filters, intrinsically minimal with off shelf chips. Basically the more taps, the better the source wave is reconstructed in memory. This too, is a component of jitter, beyond the more basic timing issues. So my question reworded is, are you relying on the DAC chip's organic wave reconstruct, or is their added tap filtering going on ?
I am not using the reconstruction filter in the DAC chip. I've implemented my own reconstruction filter as an FIR with about 1200 taps. At 44.1 (or 48) input it is an 8X upsampling filter.
The FIR is implemented with a single 52 bit MAC running a loop going through the coefficients and intermediate results. This loop runs at 96MHz no matter what the sample rate.
The filter itself is an intermediate phase, it has a little bit of pre-ringing and some post ringing. This sounds significantly better than either straight linear phase or minimal phase.
That's about it for the basics of the filter.
John S.
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I am not using the reconstruction filter in the DAC chip. I've implemented my own reconstruction filter as an FIR with about 1200 taps. At 44.1 (or 48) input it is an 8X upsampling filter.
The FIR is implemented with a single 52 bit MAC running a loop going through the coefficients and intermediate results. This loop runs at 96MHz no matter what the sample rate.
The filter itself is an intermediate phase, it has a little bit of pre-ringing and some post ringing. This sounds significantly better than either straight linear phase or minimal phase.
That's about it for the basics of the filter.
Ok, what benchmark or constraints did you apply to your filter ? Basically why only 1200 taps ? Was the reconstruct measured with a time domain limit and might the ringing side effect be mitigated with further filtering ? I ask, as researching other implementation of FPGA, tap rates are now 10x the order of magnitude of 1200.
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Ok, what benchmark or constraints did you apply to your filter ? Basically why only 1200 taps ? Was the reconstruct measured with a time domain limit and might the ringing side effect be mitigated with further filtering ? I ask, as researching other implementation of FPGA, tap rates are now 10x the order of magnitude of 1200.
I did a LOT of listening to a large range of filter parameters and found that the long filters do not sound as good as short filters. The long filters let you get very low alias attenuation (better than 130db) which a lot of people focus on, but my listening indicates that these long filters don't sound as good. By relaxing the alias attenuation to 80db I was able to get much better sound. Relaxing it even more starts letting enough aliasing through that sound quality starts going down. 80db seems to be a good compromise. The result of the relaxed alias attenuation is a much shorter filter. That is most likely significant.
Other parameters of the filter were worked out with my friend Alex and I spending hours tweaking parameters and listening to the results with a wide range of source material. This was very definitely a tuned by ear design rather than meet some mathematical constraint.
John S.
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Hi John,
1) Your DAC has no output stage. My experience says an output stage can change the sound from detailed and airy to a more robust and weighty sound. Not just in the macro sense but more by giving for example a piano more body and energy transmission in each tone even when played subtle. A superior output stage can do this and still keep the details and airiness much like a superior preamp can.
How are your views on these observations and if you agree did you manage to keep the weightiness even without the output stage?
Did you experience any disadvantages at all by not using an output stage or did you find it better in general but not necessarily in every aspect?
2) Owning a NOS DAC without digital filtering I'm interested to know if I can emulate the sound your DAC is providing by running a test on some converted files using your filter settings.
I know you've played around with the SOX upsampler and like to know if you can expose the SOX command line to convert a CD format to eg. a 24 bit 4x44.1=176.4Khz file offline where the parameters follow your filter settings and will this emulate the sound of your DAC?
I don't understand the SOX in more than general terms so I don't know how to apply a general description on filtering to a command.
Even if I can emulate the sound by converting files offline I could still be interested in supporting the project for the following reasons
# I don't have to spend 100+ hours to convert my music collection
# I don't have to spend another 100+ hours to convert my music collection if you find a better filter setting
# I don't have to spend $ on two extra hard drives to keep my converted collection together with the original rip
# I don't have to spend $- on power running two extra hard drives for years
# I don't have to spend $ every year on more more cloud backup storage for the converted files
# I don't have to spend $$ on a planned power cable for my DAC as I'll opt for the battery instead and I don't have to worry if a $$$ power cable might be better than a $$ power cable
# Using only one shelf in the rack section my DAC is at I can skip a complete section of my rack and free space for an art sculpture or another eye pleaser as your DAC can fit in the first rack section
Tx for your effort on improving digital playback.
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This DAC may very well upset some conventional notions. I don't think that you will see John or me the least bit upset if it does. As I explain in the description we had assumed - when the DAC project was initiated with a very different design using a very different DAC chip - that there would be a tube output stage. When this design was put together it was put up against earlier tube output designs and it was a clear winner in terms of weight, dynamics, extension, noise floor and imaging.
A few other commercial DACs have been brought by during the listening stages of this project and there was some disparity in quality levels that made it difficult to relate the two. My hope is to head to our studio in the next couple of weeks so that we can compare against a Pacific Microsonics Model2 and hopefully some other studio grade gear.
I would offer that there may be many other things that are different between your DAC and this one besides the filter. Even if there was a way to emulate the filter that is just one component of the design and I don't think you could really get a full sense of what this DAC might sound like by emulating that one element.
We are hoping to do some demos of the DAC in the near future. I don't know where or when yet, but we will certainly announce any opportunity here on the forum.
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Theres a lot of questions flying around and rightly so at the size of the investment but i also know if Doc et al spent 4 years developing it that it will sound amazing and still be a bargain! Taran
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Thanks Taran, and thanks for being one of the first to commit. Thank you too Dave, your fearless support of us the minute we release a new product is really special to us! You guys rock!
I've been absolutely buried trying to get everything lined up today, so I'll just post here before I get to the official post,
the crowdfund has started
here's a link -
http://bottlehead.com/product/bottlehead-dac/ (http://bottlehead.com/product/bottlehead-dac/)
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Did you experience any disadvantages at all by not using an output stage or did you find it better in general but not necessarily in every aspect?
Having built several of the prototypes (including the last one with a tube output stage), I can say that there was a distinct disadvantage in the case of this DAC when adding any of the tube output stages we tried.
This was not the case with the earliest prototypes that required an output stage of some sort to function.
Owning a NOS DAC without digital filtering I'm interested to know if I can emulate the sound your DAC is providing by running a test on some converted files using your filter.
I can't speak for Doc and John, but this isn't an open source project, and a request like this comes across strangely.
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I am in Australia...sending the DAC back for unsoldering one chip & upgrading or inserting a replacement would be an expensive, inconvenient & the least preferred option.
Is the a Plan B for OS sales...Thanks
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I am in Australia...sending the DAC back for unsoldering one chip & upgrading or inserting a replacement would be an expensive, inconvenient & the least preferred option.
It's about $60 each way to ship. That is expensive, but not incredibly so.
This still boils down to wanting to maintain your warranty. If something happens and your DAC board is damaged in the upgrade process, you really want that to be our responsibility, not yours!
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It's about $60 each way to ship. That is expensive, but not incredibly so.
This still boils down to wanting to maintain your warranty. If something happens and your DAC board is damaged in the upgrade process, you really want that to be our responsibility, not yours!
I would be nervous about getting stung for a second lot of import duty on it return! Not sure how that works if anyone can clarify the procedure of shipping a purchased item from the UK for warranty/service/upgrade work for customs and receiving it back it would be appreciated.
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Yeah, we can slip an invoice in with the shipping paperwork. This has come up when we've retubed and serviced our Tape Repro amps internationally, and so far there haven't been any problems.
-PB
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I got hit with that before when i sent a DAC back to a manufacturer in California. The return shipping label clearly said "returned from warranty repair", $0 value, and $500 insured value so some wise ass in customs charged me duty on the $500 + shipping + handling charge + vat, and i had to go the depot to collect it myself. I had a colorful vocabulary that day.
I got a refund in the end after submitting a claim, had to get a letter from the manufacturer to verify it was a warranty repair with a copy of the original invoice to prove i owned it. Two months later i got a cheque in the post refunding the cost - the handling charge >:(
The worst part is every single package i have received from the US since then has been sezied and had duty charged on it. As the postman in my local depot put it "your on somebodys shit list!"
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The worst part is every single package i have received from the US since then has been sezied and had duty charged on it. As the postman in my local depot put it "your on somebodys shit list!"
Maybe we can mail it to your neighbor instead? (Or to your work address?)
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Is the a Plan B for OS sales...Thanks
Off the top of my head I don't see any way that would be cheaper. If we ask customers too attempt it themselves, they run the risk of roaching a $750 main board. We are trying to make it so the cost difference between DIYing the update and us doing it is minimal.
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I can't speak for Doc and John, but this isn't an open source project, and a request like this comes across strangely.
Well it's a question you can choose to answer or not. Anyway I ordered a bh dac and are looking forward to hear how it compares to my modern nos dac ;)
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Thanks so much for your support! Since we are licensing the digital side of this project from John, it's completely up to him when it comes to a decision about sharing his IP. Apologies if we came off a little abrupt there. I think we're all just a bit wound up from finally getting this project out of the lab and onto the street. As you guys know it's been quite a long haul while everyone else is showing great enthusiasm with their questions our perspective tends to lean a bit towards
"Auughh! No more changes! We have to deliver this thing!"
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I currently use a Cambridge NP30 to play digital files from a USB memory or NAS (all a bit new, and slightly mysterious to me.)
I run this into a "Tubemagic D1" DAC, upgraded OpAmps, NOS output tube (via Optical)
Then though my amp/old favorite speakers
Or
For a more "revealing" listen, through my BH Crack/speedball and Senheiser HD650s
Listening to FLAC files this way sounds superb to me. [some 44.1/16bit, up to 96/24bit (max for current streamer and DAC)]
I also enjoy vinyl.
I don't think I'm a terribly "sensitive" listener. Too much exposure to helicopters, motorbikes and rock music, maybe.
Anyone care to offer an opinion on how audible an upgrade to the new BH DAC would likely be for me?
Just comments/suggestions/opinions sought. Not guarantees :)
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"Auughh! No more changes! We have to deliver this thing!"
I understand this statement completely Doc --- draw the line in the sand, FPGA in this case, and move on.
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Well my blood presure is back to normal this afternoon. This morning I put the DAC prototype back in the system. It had been sitting in my office for the past week or two as we had been so busy with sales that there was not really any time to listen. Thinking I would put on a little music today and relax a bit, I plugged the DAC in and it started acting very strangely. It wouldn't autoselect the sample rate, the display was a little buggy and the sound was distorted. Oh My God!!! We've started production and this thing is broken!!!
Peebs comes in to work a little later and says -"You know, that's not the right power supply. It's way too puny". He goes downstairs and grabs the proper, beefy power supply and it starts right up, beautifully. It was very nice to listen to some really fine sound to calm my nerves. TGIF.
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I know the sinking feeling. Glad you are back to normal.
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When this design was put together it was put up against earlier tube output designs and it was a clear winner in terms of weight, dynamics, extension, noise floor and imaging.
Did you experience this with an active pre amp as the Beepre or also when using a passive pre amp?
I'm thinking that an active pre amp might work like an active output stage and adding an active stage ahead of an active pre amp will just be an extra stage not doing anything but smearing the signal. I'm using a transformer based attenuator so an output stage might be beneficial in my system?
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Did you experience this with an active pre amp as the Beepre or also when using a passive pre amp?
Both.
I'm thinking that an active pre amp might work like an active output stage and adding an active stage ahead of an active pre amp will just be an extra stage not doing anything but smearing the signal. I'm using a transformer based attenuator so an output stage might be beneficial in my system?
In our earlier design, with a different DAC chip, gain after the DAC output was essential to the DAC functioning. In the new design, sufficient voltage output and low output impedance are available right at the output of the DAC chips themselves. When adding an active stage, generally the reasons are to increase the gain or decrease the output impedance.
A transformer based attenuator in general is a pretty low impedance load, so the focus is most often on low output impedance of sources beyond all else.
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Dear Paul; I imagine that your last comment in the previous post means that running the DAC out of my OPPO and into the TVC's fronting my BH linestage will be optimum?
Cheers, Hank
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Yes, the DAC will drive your TVC's nicely.
-PB