High Resolution Multi-Channel Digital Interface

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This is an update on my experiments with a real-time direct-digital audio interface.

Highlights of my present implementation:

- Excellent sound quality (IMHO).
- Multichannel support (six channels).
- Good electrical noise immunity.
- ESD protection.
- Power supply sequencing between interconnected units can occur in any order.
- Tolerant of live plug-in and disconnect.
- Master (low jitter) clock can be located next to DAC or digital amplifier.
- Flexibility with multiple source formats (DSD, I2S, RJ24, various sampling frequencies).
- Destination format is standardized at 24bit I2S (sampling frequency up to 200KHz, dependent on master clock).
- Low cost, standardized interconnect cables and connectors (CAT5).
- Suitable for short-distance (six inch box-to-box) as well as longer distances (I’ve tried up to twenty-five feet, but it should be able to go even further).

Background:

A couple of years ago, I was inspired by the availability of new digital amplifier chipsets, audio processors, as well as the coming of multichannel DVD-A and SACD. I had long wanted to implement an all-digital signal path, including digital speaker crossovers and room correction. Digital amplification offered a more practical implementation of the large number of separate amplifiers that I would require. I’d already formed a rather strong opinion that for a given cost, a digital amplifier could offer better sound quality than a traditional DAC / analog amplifier combination.

I spent most of the first year studying digital amplifiers and working on some preliminary designs. It became very apparent that the lack of a standard commercial multichannel digital interface was going to be a roadblock to both my digital processing and digital amplifier experiments.

I bought a Panasonic SA-XR10 receiver as an experimental platform after discovering that it utilized TI’s Equibit digital amplifier chip set. One of the things that really startled me was how much better CDs and 2-channel DVD-As sounded through S/PDIF compared to using the analog link: Disk player >>> DAC >>> ADC >>> Digital Amp. (I knew that bypassing this analog patch job was going to be an improvement, but I didn’t expect it to be such a big one.)

This revelation caused me to drop everything else and focus my attention on a new digital interface.

Considerations:

The present S/PDIF standard only supports Stereo PCM (up to 96KHz) without compression. I wasn’t interested in compressed multi-channel formats such as AC3, DTS, MP3, etc. (If I happen to want to play an AC3, DTS, or MP3 disk, I’ll use the player to decode it before sending it over the interface.) Also, S/PDIF uses an imbedded clock that has some data-dependant clock recovery issues.

The only direct-digital SACD and DVD-A interfaces that I’m aware of are proprietary to certain manufacturers and will only work with their equipment.

The upcoming IEEE-1394 digital audio interface requires a tremendous amount of overhead. It probably will offer excellent sound quality, but it will be very DIY-unfriendly.

USB looks promising for DIY. TI’s TUSB3200A supports eight channels of USB to PCM interface. Their reference design should help reduce the work required to get this up and running. Still, it requires a lot of overhead. I may reconsider it in the future, but I decided that I wanted to stay with a synchronous interface for now.

LVDS over twisted-pair appeared to be the best type of signaling to use for an interface. There are many support chips available for it, as well as it becoming a standard on newer FPGA’s. It isn’t dependent on a standardized supply voltage, it has good noise immunity for this type of application, the timing accuracy is very good, and the bandwidth exceeds the requirements of digital audio.

Initially I’d thought to use discreet parallel LVDS links for each of the I2S lines. The biggest problem with this is that for multi-channel, it would require expensive cable and connectors. There also could be a problem with timing skew between the links, especially with longer distances.

A Serializer-Deserializer (SERDES) approach offers quite a few advantages over parallel: Multiple signals can be sent over a single twisted pair. This allows a much cheaper CAT5 cable and modular connector system to be used. Since the data is reclocked, buffered, and synchronized at the receiving end, SERDES is virtually immune to skewing, and offers additional jitter rejection.

SERDES can be implemented either with a clock embedded into the data (a single LVDS pair required), or a clock separate from the data (two LVDS pairs required). (Most SERDES chipsets with imbedded clocks have managed to avoid the data-dependant clock recovery problems that are associated with S/PDIF.)

The SERDES transmitter requires a control clock that is synchronous to and a multiple of the data being transmitted. In this case that clock would be the MCLK of the disk player. Most disk players use either 256fs or 384fs for their MCLK. The possible frequency range for MCLK then becomes 11.2896MHz (44.1KHz CD w/ 256fs) to 73.728MHz (192KHz MLP DVD-A w/ 384fs). This turned out to be the primary consideration in choosing a particular SERDES chipset. The SERDES’ PLLs need to be able to operate over this frequency range.

The only SERDES chipset that I was able to identify with this PLL frequency range was TI’s MuxIt devices. This is a four chip solution for one data link: a PLL (SN65LVDS150) and a Multiplexer (SN65LVDS151) for the transmitter side, and a PLL (SN65LVDS150) and Demultiplexer (SN65LVDS152) for the receiver side. MuxIt uses separate LVDS clock and data pairs. It can be configured to pass between four and ten parallel data lines (not including MCLK).

A CAT5 cable has four twisted pairs. A MuxIt link requires only two of these pairs.
I use one of the extra twisted pairs as a ground link between the transmitter and receiver units. The jury’s still out on whether it’s better to have this ground connection or not. It can prevent the LVDS signals from exceeding the common mode range of the devices, but it also could possibly introduce a ground loop. So far, I haven’t experienced any problems with it connected or disconnected.

There are a couple of possible uses for the fourth twisted pair: remote power, or a master clock signal to the transmitter from the receiver. This master clock could be used to synchronize the transmitter (such as a disk player) to the DAC or digital amp. There are two constraints with this: the transmitter would have to be able to synch off of an external clock, and the receiver would need to know what frequency the transmitter required (this might change for different disks). Because these constraints make it harder to implement a ‘universal’ interface, I choose not to send back a master clock at this time.

I tried running a MuxIt link from a DVD-A player to the SA-XR10 digital amp, using the recovered clock sent from the DVD-A to clock the Equibit section. Not only was I able to hear multichannel surround without the analog link for the first time (wow!), I found there to be a substantial sonic improvement on regular stereo CD’s compared to the S/PDIF interface (the thing that inspired me to focus on this exercise in the first place). I realized that the S/PDIF link on these units was hardly optimized (lots of board-to-board interconnects, etc.), but I still was surprised. To help convince myself that this wasn’t purely psychological, I A/B’d the interfaces (using a button on the remote) for several people. Everyone preferred the MuxIt interface, saying that they could hear better detail, or that things sounded clearer. By contrast, the difference going from the analog link to S/PDIF was more along the lines of improved depth and imaging. (I want to avoid debates on comparison testing or subjective descriptions. This is what I tried, this is the best I can do to describe it in print.)

(continued in part 2...)
 
(...continued from part 1)

I then added three (for six channels) Asynchronous Sample Rate Converters (ASRC) after the MuxIt receiver. This allowed the use of a local MCLK at the digital amp for further jitter reduction (fixed at 96KHz), as well as accommodating PCM format conversions. The MCLK from the transmitter isn’t used for anything other than for the receiver PLL. I configured the MuxIt multiplier as times six and send five audio data lines (BCLK, LRCLK, FDATA, SDATA, and CSWDATA). I have a jumper on the transmitter for the sixth data line to indicate whether the source is I2S or Right-justified 24bit format. This format line is fed to the mode pins of the ASRC devices, allowing them to automatically handle format conversions.

I laid the board out to accommodate either the AD1896 or SRC4192 ASRC devices. They’re mostly footprint compatible. The AD1896 requires a daisy chain link to keep multiple devices synchronized in Matched Phase Mode (The SRC4192 doesn’t require these links and ignores them). The SRC4192 requires an external oscillator for its internal logic (the AD1896 can also use a crystal).

So far, I’ve only used the AD1896 devices (I’ve got SRC4192 devices on backorder). It’s a little strange: on some recordings they seem to make a big improvement, on others there’s little or no difference. This distinction doesn’t seem to fall along a boundary of what I would consider good quality recordings vs. lessor quality recordings. Some ‘good’ recordings seem to sound better with the ASRC, some don’t. Some ‘lesser’ recordings seem to sound better with the ASRC, some don’t. At least there haven’t been any instances where I thought the sound was degraded. I have to point out that these observations are even more subjective than the MuxIt vs. S/PDIF comparisons. I don’t currently have the capability to do a pushbutton A/B comparison.

The ASRC devices do definitely make it much easier to interface to a variety of units.

For SACD, I added a SM5816AF DSD to PCM converter to the transmitter board. This chip outputs 8fs PCM (six mono 352.8KHz 28bit bitstreams) or 2fs PCM (88.2KHz 28bit I2S). I’m using the latter, since it’s I2S. (There’s an upcoming SM5819AF that offers 4fs PCM (176.4KHz I2S), but I haven’t got my hands on any yet.)

I decided to stick with I2S PCM for the time being because it’s what I know. I still don’t have a strong opinion on the SACD vs. DVD-A debate. I think they both sound better than any of the previous available consumer formats. For me, the important thing is that some recordings are coming out on SACD only, and some are DVD-A only. I want to be able to play either.

That’s about it for the overview description. I’ll try to get some pictures and schematics posted later this weekend.

Regards,
Brian.:cubist:
 
PCM MuxIt Transmitter Board:
 

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All I can say, wow! You are doing some really cool work there. Sending multiple channels of PCM data directly to the digital amp is something very desireable. I work in pro audio world. We are using DSP processors that can network with eachother using a single CAT5 cable. They can usually carry 8 channels of 24 bit audio at 96kHZ or 16 channels at 48kHz. My guess is, they are using the same chips that you are utilizing for your TX/RX duties.

What would be awesome is the ability to play back a CD or DVD in a computer, do a digital crossover in the same computer, come out of Ethernet card into a digital amp with a CAT5 connector and your receiver board. Some folks on this very board are already writing applications for digital crossovers inside a PC. It seems like a perfect logical step. The question is how to make an Ethernet card output raw PCM. I guess a special driver or a translating application would be needed.

BTW, how does the Equibit amplifier do volume adjustments?
 
Thunau said:
I work in pro audio world. We are using DSP processors that can network with eachother using a single CAT5 cable. They can usually carry 8 channels of 24 bit audio at 96kHZ or 16 channels at 48kHz. My guess is, they are using the same chips that you are utilizing for your TX/RX duties.

They probably are doing something different. Having a proprietary interface doesn't require as much flexibility.

What would be awesome is the ability to play back a CD or DVD in a computer, do a digital crossover in the same computer, come out of Ethernet card into a digital amp with a CAT5 connector and your receiver board. Some folks on this very board are already writing applications for digital crossovers inside a PC. It seems like a perfect logical step. The question is how to make an Ethernet card output raw PCM. I guess a special driver or a translating application would be needed.

Ethernet would be completely different from what I am doing, even though it can use a CAT5 cable.

The approach I described would easily work by tapping into the I2S lines of a computer sound card. The biggest problem with using a soundcard for a digital crossover is that most non-professional units are limited to eight channels. That's enough for stereo, but not multichannel.

I'm planning on focusing on a non-PC approach. I think that FPGAs (and their ability to accommodate large parallel data paths) are the ideal platform to implement massive time-domain FIR filters. How's this for a thought: one second of 192KHz fs requires 192000 taps per channel (to correct down to 1Hz). For 24bit data, each tap should be 72bit to avoid rounding errors and provide headroom for normalization. Six two-way speakers plus two subs will require fourteen channels (20channels for three-way). That's 2688000 72bit taps and 2688000 24x24 multiplies per second! This is the benchmark goal I have for a true brute force FIR without the use of FFTs.

More near term, I'm taking a less ambitious intermediate step using TAS3103 digital audio processor chips to implement an all-digital solution, including crossovers. This will limit me to IIR filters and digital delays for driver time-alignment. I think this should still be quite an improvement over what I have now. I need to get some practice with MLS measurement and correction techniques. Once I have that in place, along with a full complement of digital amps, it should be much easier to transition to the FIR approach.

BTW, how does the Equibit amplifier do volume adjustments?

They can use either digital attenuation, variable power supply voltage, or a combination of both.

Regards,
Brian.:cubist:
 
I just have to let you know that I'm REALLY impressed! And to let you know that we appreciate you sharing your knowledge and experience with this new concept that makes a lot of old cronies (and even a youngster like me) who are attached to a "pure" analog signal take a step back. There's a lot of talk with this type of approach, but not a lot of action. Kudos to you on a fanciful amount of labor of love. Even if the system sounded like crap, you should still be proud that you managed to integrate the whole thing on such a universal level. Outstanding work.... simply outstanding!
 
Outstanding stuff, Brian. Very elegant.

Coupling this with an XR-25 or 45 for 6 full digital channels has some very attractive possibilities.

Do you have any interest/intent of doing any small production runs of the boards? It looks like you've provided enough artwork to get the pcb's done at any of the online places, but assembly by us duffers would still be a challenge.

Time for me to try to reverse-engineer my Delta 1010 expansion connector. If I can reliably isolate the I2S lines, I might have to try this.
 
Digital Room Equalization at EACH speaker

Digital domain room equalization is impressive when properly executed. Several topologies are possible based upon how you answer the questions:

Where do the digital amps go?

Where does the digital crossover go?

Where does the digital room equalization go?

Can digital room equalization optimize several speakers simultaneously for an optimized summation? or just one speaker at a time against the room? This last question seems the most challenging. Do algorithms exist for optimizing seveal speakers simultaneously to get the best summation at the listening position?


My current thinking is to put the digital amp + crossover + room equalization at each speaker, most likely serially connecting a PC to each speaker to calculate and download room equalization until this algorithm can be built into each speaker's DSP. Separately equalize each speaker against the room.

Has anyone tackled digital room equalization?
 
dwk123 said:
Coupling this with an XR-25 or 45 for 6 full digital channels has some very attractive possibilities.

Do you have any interest/intent of doing any small production runs of the boards? It looks like you've provided enough artwork to get the pcb's done at any of the online places, but assembly by us duffers would still be a challenge.

I have thought about it. I don't want to make a regular business out of making this type of thing, but it would be nice to sell enough to recover some of my development cost.

In this particular case, I feel that patching these boards into existing commercial products is a bigger challenge than soldering the parts onto the board. It didn't seem to me that anyone who could figure out where the I2S signals were located and could tap into them would care about having the board preassembled. I think that if I tried to sell these, the amount of support people would probably want would be overwhelming. This is a hobby and I want to keep it fun.

In the future, I've thought about designing boards that would just require plugging in. Perhaps a receiver board for the Panasonic SA-XRxx units, and a transmitter board that could plug into the I2S lines of a six or eight channel sound card. These types of devices could be made so that they are fairly straightforward to use for people that don't want to make and troubleshoot their own circuitry.

Unfortunately, I think that interfacing to any CD, DVD-A, or SACD player will always require tapping into PC board vias, traces, pads, and/or IC pins. Every player would need to be individually figured out. Careful layout and routing of the interface wires is necessary to prevent signal ringing and reflections.

Right now, I want to focus my time on new circuits. This interface was just the first step of my new all-digital system goals. I'm presently brewing up a multi-channel digital processor based on the TAS3103.

In the meantime, here's a few more details for anybody that might be interested in trying this on their own:

******************************

All of the parts for the PCM MuxIt Transmitter and PCM MuxIt Receiver are available from Digikey (prices shown are US$ at time of posting).

PCM MuxIt Transmitter BOM:
(1) 296-9737-5-ND.................SN65LVDS150PW MuxIt PLL.............$06.160
(1) 296-9738-5-ND.................SN65LVDS151DA MuxIt Transmitter...$05.020
(6) P49.9HCT-ND...................49.9ohm 0603 resistor.....................$00.090
(2) P100HCT-ND....................100ohm 0603 resistor......................$00.090
(1) P10.0KHCT-ND..................10Kohm 0603 resistor......................$00.090
(6) PCC1762CT-ND.................0.1uF 16V 0603 cap........................$00.103
(6) PCC2250CT-ND.................4.7uF 16V 1206 cap........................$00.594

TOTAL: $16.72


PCM MuxIt Receiver BOM:
(1) 296-9737-5-ND.................SN65LVDS150 MuxIt PLL...................$06.160
(1) 296-9739-5-ND.................SN65LVDS152DA MuxIt Receiver........$05.020
(3) 296-15234-5-ND...............SRC4192IDB 192KHz ASRC.................$15.740
(1) ECS-P53-B-ND (30.0 MHZ)..30.0MHz 3.3V oscillator....................$08.400
(1) ZXCM209TFCT-ND.............ZXCM209TFTA 3.08V Reset generator.$00.710
(5) P49.9HCT-ND...................49.9ohm 0603 resistor.......................$00.090
(3) P100HCT-ND....................100ohm 0603 resistor........................$00.090
(6) P10.0KHCT-ND.................10Kohm 0603 resistor.........................$00.090
(12) PCC1762CT-ND...............0.1uF 16V 0603 cap..........................$00.103
(11) PCC2250CT-ND...............4.7uF 16V 1206 cap..........................$00.594

TOTAL: $76.54

*********************************

I had my bare boards made at Advanced Circuits with their Bare Bones PCB service:
http://www.barebonespcb.com

I've had excellent results with their service. The boards don't have soldermask or silkscreen, but the price is great, especially for low volume boards. This service has one day turn time as standard! Now that this is available, I rarely do any hand-wired protos.
They charge a $10 handling fee, plus $5 or more for shipping (depending on the delivery option you choose).

The Gerber zip files I posted earlier can be directly loaded into the Bare Bones website.

Use the following info:

PCM MuxIt Transmitter:
X dimension: 0.864"
Y dimension: 1.272"
.zip file name: MX2.zip
Part name: MX
revision: 2
The price for one board is $25.55. Additional boards are then $0.55 each.

PCM MuxIt Receiver
X dimension: 2.034"
Y dimension: 1.472"
.zip file name: MR2.zip
Part name: MR
revision: 2
The price for one board is $26.50. Additional boards are then $1.50 each.

*********************************

I use 30AWG wire-wrap wire to tap into the I2S signals.

Generally it's best to connect as close as possible to the IC that generates the signals. Reflections are usually less of a problem with this approach. This is especially true for the MCLK signal.

I tap a ground wire as close as possible to each point that I tap a signal from the board. I then twist the wires together to minimize the loop-area impedance of the signal and its return ground path. This is very important to preserve the signal quality and to minimize ringing. If two or more signals being tapped are close to the same ground wire tap point, they can share and be twisted together with the same ground wire. Also keep all of the wires as short as possible.

Some examples of this can be seen in the pictures I posted, although much of the wiring is hidden under the boards (to keep the length short).

*********************************

These boards both use a 3.3V supply. Many times you can use a 3.3V supply from the unit you are tapping into. I don't have the measured current draw for these boards handy (I'll try to post it later). Some units are sensitive to having additional power drawn from their 3.3V supply.

For example, if the 3.3V supply is generated off a 5V supply by a small surface mount linear regulator, it may not have enough capacity.

Another case to watch out for is if the 3.3V supply is the regulated leg of a multitap switching supply. In this situation, extra current draw on the 3.3V supply will be accommodated, but the other supply voltages will increase beyond their spec.

Some units I've tried seem to have plenty of reserve (I've had good luck with Panasonic). Others seem to have their supplies closely engineered to accommodate the original circuitry and nothing more (this was the case on a Sony I did).

If necessary, you can put in a separate 3.3V supply.

*********************************

I always recommend getting the schematic and other service documentation for the unit you are modifying.

*********************************

For the transmitter, it's probably best to tap into the I2S signals that feed the disk player's DAC. It isn't necessary to disable the DAC. If a disk player sends the I2S signals through a separate base management / surround processing chip, then you have the option of tapping into them before they get this extra processing (my preference).

For the receiver, there can be a number of different places to tap into. In the case of the SA-XR10 that I am using, there are two choices:
a.) The connector going directly to the digital amplifier board.
b.) Tapping into the signals that originally came from the ADC (this approach requires that the ADC be disabled).

Initially, I'm using the second approach. This allows me to use the receiver's volume control. Once I have my digital processor working, I plan to feed the amp directly.


*********************************

I have a couple hundred hours of listening to this interface now. I have to say that I'm very pleased with it.

I just wish I could get new stuff done faster.

Regards,
Brian.:cubist:
 
AMAZING

Howdy Brian,

well done, that is some very fine work. I've been using the ad1896 myself in a variety of projects, I love it. I strongly suggest you loose the simple oscillator circuit for the ad1896's and adopt something like the Kwak clock, the resultant jitter from the ad1896 is only governed by the jitter on your master clock source. The sonic differences are quite amazing. Like everything else the ad1896 likes very clean separate supplies.


I'm currently designing a board to slip into my dvd-a player to give me 3 aes/ebu streams, ad1896 and tx chips, planning to use the ad1853 for analog outs as well. I found a dvd-a player with a Zoran Vaddis 5 chip for less then 200 USD, getting any info from Zoran is impossible, but I managed to get a datasheet from AVS Tech, the manufacturer of the 6 channel dac chip. Standard form its nothing exciting, switch mode psu and all, but a good bunch of power supplies, a decent clock ( uses 27 Mhz, mpeg clock freq, makes it easy) and a digital out and it'll be a rocket. Does dvd-a MLP. I've not looked inside many other players, so I'm not sure what all-in-one chips they use, but I'm guessing Zoran has a big market share.

How good is the Equibit amp? Have you compared it to anything? Tripath? a really good analog amp? e.g current feedback, no odd order distortion, 900khz bandwidth. I've a schematic if your interested. I only say this as the topology I have is as good if not better then Tact Audio's Millennium amp (equibit implementation, back in the toccota (sp?) days) which is considered a very good digital amp.


As you have said, IEEE-1394 is not a nice monster. From what I've heard and read, it sounds like crap compared to aes/ebu

Best bet is doing away with a tx/rx setup completely, or using the aes/ebu or your MixIt methods. Have you considered merging source and amp into one box and doing away with any transmitter/receiver setup? Players decoder into ad1896(s) into Equibit chip(s)


Once I've got this going I'll tackle a sacd player, again starting with an inexpensive implementation and adding to it. I haven't looked into sacd players for a few years now, but from what I'd seen, Sony was using the same chips everywhere and changing psu and component quality to differentiate models.


I'm following the DAX Groups project with close interest, a dvd-a and/or sacd player straight into a digital amp would be great. So far I think great analog amps are still a little in front of digital amps, but I'd love to be proven wrong.


Mark Hathaway
 
Re: AMAZING

Thanks, Mark.
Mark Hathaway said:
I strongly suggest you loose the simple oscillator circuit for the ad1896's and adopt something like the Kwak clock, the resultant jitter from the ad1896 is only governed by the jitter on your master clock source. The sonic differences are quite amazing. Like everything else the ad1896 likes very clean separate supplies.
I'm using the ASRC with both the input and output ports in slave mode, so the oscillator is only running the internal ratio estimation logic. This allows me to keep the critical low-jitter MCLK for the DAC (in my case a digital amp) completely off of the interface board.

If the ASRC is an output master, then the quality of the oscillator is much more critical (as you are describing).

I'm currently designing a board to slip into my dvd-a player to give me 3 aes/ebu streams, ad1896
The problem with this approach is that there will be three different recovered clocks from the separate AES/EBU streams. They will be asynchronous to each other. There also will be three different group delays. Once this happens, there isn't any practical way to get the three streams phase-aligned with each other again.
How good is the Equibit amp? Have you compared it to anything? Tripath? a really good analog amp? e.g current feedback, no odd order distortion, 900khz bandwidth. I've a schematic if your interested. I only say this as the topology I have is as good if not better then Tact Audio's Millennium amp (equibit implementation, back in the toccota (sp?) days) which is considered a very good digital amp.
I haven't heard the Millennium yet, but it seems to have a very good reputation.

For a couple hundred bucks, I think the Panasonic SA-XRxx receivers are surprisingly good. These are the only Equibit units that I have heard so far. Since they don't represent an optimal Equibit implementation (especially the power supply), I haven't been able to experience the full potential of TI's Equibit chip sets. However, my expectations are very high.
As you have said, IEEE-1394 is not a nice monster. From what I've heard and read, it sounds like crap compared to aes/ebu.
My main concerns with IEEE-1394 is that it is very complicated to implement, and it's also likely that high resolution audio will be encrypted.

IEEE-1394 fully buffers the data at the receiving end. The master clock can then bring the data directly out of the buffer. The quality should be as good as the clock and the rest of the circuitry that is downstream from this point.

AES/EBU uses a recovered clock (and it doesn't support multichannel unless some type of compression is used). For this reason I think that IEEE-1394 offers the potential for much higher quality.

Actual implementations will vary, of course.
Have you considered merging source and amp into one box and doing away with any transmitter/receiver setup?
That certainly would be one very good and practical way to implement a final design.

At this point, I'm not even close to achieving a full system design. One of the reasons I was compelled to make this interface was that I wanted the flexibility to work on one section at a time.
I'm following the DAX Groups project with close interest, a dvd-a and/or sacd player straight into a digital amp would be great. So far I think great analog amps are still a little in front of digital amps, but I'd love to be proven wrong.
I feel that digital amps are already competitive with more traditional DAC / analog amp designs (for a given price). Right now there seems to be a big gap between consumer products (like Panasonic) and the high-priced stuff (like TacT). I suspect that most of the people here will be concerned with how this gap fills out.

Regards,
Brian.:cubist:
 
Transmitter Board Correction

The dimensions I posted for the PCM MuxIt Transmitter board were incorrect.

Use this information instead of what I previously posted:

PCM MuxIt Transmitter:
X dimension: 0.952"
Y dimension: 1.360"
.zip file name: MX2.zip
Part name: MX
revision: 2
The price for one board is $25.65. Additional boards are then $0.65 each.

I'm sorry about the error.

Brian.:cubist:
 
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