DAC AD1862: Almost THT, I2S input, NOS, R-2R

This is a personal qualification against the rules of the forum. I am asking the moderator to react to this arrogant behavior.

We always ask that all posts are respectful, however this one does not really fall foul of any rules and that line as posted could equally be taken as a comment in general. As always, think before you post.

Also please do not alter the implied meaning of quotations by changing, underlining or highlighting the original text as that does change the original meaning.
 
I am newb so I am not understanding what you suggest, could you tell what exactly suggest without I2s?
And I would like to know what is your best DAC and connection you built,designed or bought?
I did not suggest you do without I2S. You can use whatever you want. I just explained why I don't use it.
I designed a 64-channel DAC using PCM1704 with a 1228.8K sample rate via USB-3.0. I never built it because my PCM1704 stash disappeared. I am working on a followup that uses 64 AD1862 with a 846.72K sample rate. Fortunately, I still have a stash of Black Gates.
What's your best DAC and connection you built, designed or bought?
 
Sorry for my ignorance. Am I correct that all DACs receive data through I2S connection? USB/SPDIF inputs are all converted to I2S before feeding into DACs.
You are not correct. In your ignorance you have conflated I2S with the more common left-justified and right-justified audio sample data interfaces that preceded it. You probably never read a DAC chip datasheet and studied the diagrams that illustrate the different interface formats available. The most common are left-justified, right-justidied, I2S, and the DSP interface. I like the latter because there are separate wires for data channels.
 
Please dont feed the troll.
"Ignore" feature:
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Why not learn to use PCB software.
Then you can customise anything to your own needs.


Cheers,
Patrick
I have been thinking about what Patrick posted on 07-19, and I think he made a very good point. First of all, I do not know what Miro does for a living, but I do know it is NOT designing pcb's for us. Miro has posted several times that he is trying to make it available for us to make a basic 1862 DAC with a standard opamp I/V stage, so that we can enjoy such a good sounding system like he has, and most of us very much appreciate that. He has clearly established the scope for this thread. It is bound to happen that somebody will want or need something a little different for his personal situation. However, it is not reasonable to expect Miro to make up customized solutions for all people, or he will become burned out. I think we agree that this thread will succeed better and be the most helpful if Miro does not get burned out. As Patrick said, if a person learns pcb software, he can customize to his own needs. He is no longer at the mercy of someone else. Plus he then could design more boards for others to try out with the same needs/desires.
Personally, I purchased a low-cost professional pcb software several years ago called Target 3001. It has good documentation and has more than met my needs, and I know they have a free version for hobbyists which is powerful. I know there are several other brands out there that are free also, but I have not used them so I cannot recommend them one way or the other. In addition, there must exist several threads already about getting started using pcb software. Maybe someone can post a link to another thread, because that subject should be continued in another thread to avoid hijacking this thread. I do not think any of you would regret taking the time to learn pcb software, and you will actually have another way to contribute to the cause with your new designs...and no one gets burned out.
 
@propitious My initial thought was one post for AD1862 PCB and go away from this electronics hobby, leave something behind. This hobby has many disadvantages (it is pricey, it generates a lot of clutter in the house, it consumes more time than is healthy, and so on), but it is addictive and for curious people like me it is a magnet 🤣 It all depends on my free time, sometimes more of that, sometimes none. I am currently working on the PSU-3 - if it turns out good, it will be added here 🤓
The very professional software for PCB design is Altium (perhaps the only one where you can get a job - if you would like to make living from designing pcb in a company) ... cheaper and very user friendly alternative to this is DipTrace (recommendation from my friend) ... another alternatives are Eagle (also user friendly, but free edition has significant limitations), Kicad (this is the best opensource and completely free, but the learning curve is large). Target 3001 is enough for you, why not 😉 ... you can switch to another platform at any time.
Making a PCB on a functional diagram is easy (diagrams from datasheet, cloning schematic from service manuals, verified schematic diagrams from the web, magazines, books) ... Designing something on its own can be very difficult and painful, full of non-functional prototypes and wasted money 😛
 
@Vunce In my DACs is no delay between channels because both channels are latched together 😉
"Inaudible" delay is present only if 4 shift registers (32-bit buffer) would be omitted and LRCK inverted for the second channel - and this is not case for my DACs 🤓
Only this stopped-clock version has the delay: #3969 ... but I created also another stopped-clock version with 32-bit buffer where the delay is not present (only PCB with shift registers is somewhere in the forum, I haven't done the whole DAC PCB yet 🤣). But I made the PCB for the PCM58P (seems like PCM63’s father 🤔 ), I'm just preparing the materials for the post LOOL 🤣
"Delay" of digital data inside the FS widow (inside the latches) do not have connection with "delay" of analog outcome. Not even in I2S format.
This is only one possible loss of 1/44100 sec ONCE in the start of digital stream at one channel... (Because in the first "shift" we have onlu one channel under the latch-enable) That is nothing rally.
As long as Fs pulse is correct. Inverting LE, for one channel, adding delay of propagation delay time of the inverter only. Not data. But all these Fs is low frequencies for any inverter and it will be lower that propagation delay given in datasheet because it is measured close to max F of logic gate...
If You want cal it Yours OK. 🙂 But it seems to me that everything is straight forward like in the DACs PDF-s and so many pre-designs using the same shift sch? for instance DIYINHK has many boards just like you did years before. And others, Pavouk, Andrea, Eric, DDD etc.. Just nobody payed attention... Anyway good work, dont take it offensive please 🤐
 
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Bat-hearing is not needed to hear the effects of inter-channel delay. It distroys the coherence of the central sound stage image. Why? The central image is the sum of the left and right channel. Delaying one channel sets up a comb filter that enhances some and diminishes other frequencies to different degrees. If you can't hear that it means your audio system is not coherent to begin with.
What you technically state is correct under certain circumstances, none of which apply to loudspeaker listening. You confidently assert a subjective loudspeaker listening consequence which can not be taking place, for the simple reason that your muscles can't hold your head stationary to within 0.15 inch error for long, if at all. Plus the speakers have to be positioned equidistant from it's respective ear by less than 0.15 inches. That's not happening. In addition, the position of your ears are being involuntarily and continuously repositioned well outside of the 0.15 inch distance equating to the maximum delay we've been discussing. Assuming that you are as technically competent as you like to indicate you are, you are worse than someone who unknowingly misleads out of ignorance, because you knowingly mislead out of knowledge. Whatever your reason, stop injecting confusion in to the thread.
 
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Bat-hearing is not needed to hear the effects of inter-channel delay. It distroys the coherence of the central sound stage image. Why? The central image is the sum of the left and right channel. Delaying one channel sets up a comb filter that enhances some and diminishes other frequencies to different degrees. If you can't hear that it means your audio system is not coherent to begin with.

This is not true - In general, there is no delay between the channels with I2S interface.
But I have seen some devices, where R and L channels of the same sample were located at the different I2S frames.
frame_error_I2S.png


In this case yes, it will be delay between channels.

May be Tam Lin faced with this?