Sound chips that natively go to DC?

Folks,

The subject pretty much says it. Are there any that can at least be coaxed into DC input and output by disabling internal cancelers and such? Should not be too drifty. The digital interface doesn't matter much, USB, SPI, whatever. Preferably not I2C but even that would be ok.

16 bit would suffice. Of course, more is better. Input to output synchronicity is important, as usual. Also synchronicity between the stereo input channels.
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Regards, Joerg 

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Joerg
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Analog ?>;-} ...Jim Thompson

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| James E.Thompson                                 |    mens     | 
| Analog Innovations                               |     et      | 
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Reply to
Jim Thompson

The all do dc, you need to jump the cap inside..

as for output, that may not be such a good idea because many of them do not operate dual rail supplies, they have dc isolation cap.

Best thing to do is what I did years ago., AM in AM out. write the software to decode the modulation on the input and modulate AM for the output. You need to make an interface board, but that isn't really that bad.

I did it via a 324 chip, one unit for the CW oscillator, a unit for the modulator, one input from the OSC and the other input from your reference and finally, an AM detector that uses a third unit with a integrator in the feed back and that forms your detection. The last unit can be used as a unity stage.

Jamie

Reply to
Maynard A. Philbrook Jr.

Do you mean PC sound cards?

I doubt that any sound cards use dual rails. In any case, these will have lousy DC characteristics (offset). It would take some work to straighten that out.

If you're going to go that far, why PWM from a digital port, of some shape?

Reply to
krw

Not exactly sure what you mean there? are you saying creating a custom digital port (USB for example) ? If so, that would me you need software drivers. Something that gets messy.. Recently I was experimenting with inserting 2 FM carriers via a simple board interface. Using both channels. Looks interesting any ..

Jamie

Reply to
Maynard A. Philbrook Jr.

T'is analog somewhere's in there.

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Regards, Joerg 

http://www.analogconsultants.com/
Reply to
Joerg

I found that to be problematic in the ones I looked at, some sort of DC-scrubber circuit that couldn't be disabled.

That would be no problem at all.

I was contemplating the old HC4066 trick for chopping and synchronous rectification but it does get old.

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Regards, Joerg 

http://www.analogconsultants.com/
Reply to
Joerg

Almost, I mean the chips on there.

That would be ok, I can take care of that. As long as the offset is very stable and determined, like a bandgap or two.

Well, that jinxes it :-(

Sure, and I could also just use regular ADCs and DACs on another port. But it makes a mess of things. Lots more parts, muxing (signals go via the same connector), and most of all now the processor must have this on a high priority interrupt because unlike sound chips there's no buffer. Cost is not a big concern on this project but simplicity and size is.

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Regards, Joerg 

http://www.analogconsultants.com/
Reply to
Joerg

many of them have a small DC current put at the mic input that allows you to power up a microphones

Jamie

Reply to
Maynard A. Philbrook Jr.

Often only on one input and I can deal with that. But if, as Keith hinted, the DC characteristics in general are lousy then I can't use sound chips anyhow.

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Regards, Joerg 

http://www.analogconsultants.com/
Reply to
Joerg

What is the object? That is, define the gozinta/gozouta requirements. ...Jim Thompson

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| James E.Thompson                                 |    mens     | 
| Analog Innovations                               |     et      | 
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Jim Thompson

Regular sound chips (codecs) have no buffer either. The buffering occurs in the digital interface which is part of the host chip. SPI and I2C are likely not to work. You'd need to look for a typical DSP-style codec interface like AC97, I2S, etc.

Removing the DC blocking capacitors should make a codec go to DC but you'll need some differential amplifiers to make the codec deal with ground centered signals. If you are lucky the codec outputs a reference voltage which is used to reference the signals.

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Reply to
Nico Coesel

Goes in: Signals from DC to 10kHz, amplitude up to 3Vpp but this gets divided down. Input swings around ground so in single-supply chips there'd have to be an external DC-shifter (no big deal).

Goes out: Signal of very same frequency and phase as goes it but slowly amplitude modulated. Amplitude range much lower, generally a few millivolts. DC stability must be impeccable. But siganl can be divided down from FS in order to ease that requirement a bit. Modulation BW less than 50 Hoitz and modulation depth no more than 50%.

Synchronicity between in- and output is critical and ideally the processor should not have to take care of this. Sound chips have buffers to deal with USB transmission gaps and the like.

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Regards, Joerg 

http://www.analogconsultants.com/
Reply to
Joerg

They usually do. For example, right now one from this family is in there:

formatting link

Quote "When receiving the audio data, the PCM2900C/2902C stores the first audio packet, which contains 1-ms audio data, into the internal storage buffer. The PCM2900C/2902C starts to play the audio data when detecting the next start of frame (SOF) packet, as illustrated in Figure

32".

Unfortunately it has the usual in there, poorly documented digital high-pass filters.

It usually doesn't because they put digital HPFs in there.

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Regards, Joerg 

http://www.analogconsultants.com/
Reply to
Joerg

SOF comes from USB. So yes a USB codec needs some buffering. But a codec intended to be connected to a SoC or DSP doesn't have buffering. At least I never came across one that did.

I've Never seen one with an HPF. Its not necessary because they are intended to be used with DC blocking capacitors. A HPF would impair the boom-boom sound.

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Failure does not prove something is impossible, failure simply 
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Reply to
Nico Coesel

Most (if not all) of the Delta-Sigma A/D converters for audio that I have seen and used have a HPF built in but allow you to bypass them. For instance, this one from TI, similar to the earlier link posted but this data sheet shows it in the block diagram...

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I'm pretty sure that the reason they put the HPF in there is because their DC accuracy or drift is not very good. I have wanted to use one of these regular old audio ADCs for DC before but I have gone to slower delta-sigma converters that are optimized for DC.

I think that the old Crystal semiconductor and AKM delta-sigmas had some kind of HPF that could not be defeated but can't remember for sure now. The newer ones I think can be bypassed for DC operation.

One of the converters I use for DC does a DC offset calibration in-between every conversion which works great. Probably not so easy to do with a regular audio ADC but I'd love to be able to make one work or hear how well an audio ADC works for someone at DC.

boB K7IQ

Reply to
boB

Interesting chip. that PCM1804. my Dell XPS has the same options on it for PCM/Digitial with the same sample rates and bits.

I'll have to check my dell next time I boot it to see what the sound chip set is.

Jamie

Reply to
Maynard A. Philbrook Jr.

My new (ish) dell laptop uses an IDT codec and is a company I am not familiar with. I think I found a block diagram of it somewhere but I find that it is usually very difficult to find real documents for the high volume chipsets that computer makers use these days. This is especially true for Broadcom chips where they do not want the everyday joe to be able to program them... Graphics chips too like Nvidia. Not like it used to be where you could just get the document for a chip and be able to program it. That's why I stick with smaller embedded microcontroller applications where I know what is going on inside.

Happy new Year from the yechy cold and rainy pacific northwest !

boB

Reply to
boB

Happy New Year from San Tan Valley, AZ, where the temperature is heading back toward normal... 70+ predicted for the next 7 days ;-) ...Jim Thompson

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| James E.Thompson                                 |    mens     | 
| Analog Innovations                               |     et      | 
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Reply to
Jim Thompson

Therein lies the main problem. Even If I found a trick to disable the HPF the DC specs are either not properly given at all or they are very dismal. Like with the internal "reference" in some ATMegas where I seriously wondered how someone could possibly botch an analog circuit so badly.

An alternative would be some other AD/DA device that isn't called sound chip but relieves the processor of having to baby its timing.

Hey, send us some of that rain. We are dying for it here, seriously.

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Happy New Year, Joerg 

http://www.analogconsultants.com/
Reply to
Joerg

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