Using rev D audio board as simple DAC with variable sampling rate

R

rvh

Well-known member
Hi,
Can the codec on the rev D audio shield (for teensy 4) be used as a simple DAC with variable sample rate (i.e. more or less continuous sampling frequencies so I don't need to resample/interpolate to adjust playback frequency)? I would like to be able to go as low as a few hundred Hz, and up to about 16kHz.

If so, does anyone know of some example code that sets up the codec in that manner to get me started (presumably not using the audio library) ?

I had a bit of a search but couldn't see anything suitable. Thanks for any pointers.

Cheers,
Richard.
 
Frank B said:
The question is, what your exact definition of "simple" is. Best is to take a look at the SGTL-5000 datasheet - and the audio shield schematic - if there are open questions.

Easiest is to use the Audio Library.
You can change it's frequency: https://github.com/TeensyUser/doc/w...-to-change-the-i2s-frequency-on-t4x--micromod
Click to expand...

That's excellent Frank. The setI2SFreq code snippet is just what I was looking for. The only thing I have found in my brief test is that while it worked well from 48kHz down to a little under 10kHz, it may have overflowed a register as I continued to go lower because the sample rate skipped way up high again (at 9887hz) and then reduced gradually again form there.

I'll have a look to see if I can do anything about that (please let me know if you already know the answer). But even if not, this method will solve my problem because I can easily add a simple linear 2,4,8, or 16 point interpolator to reach lower frequencies.

Thanks very much.

Cheers,
Richard
 
rvh said:
Hi,
Can the codec on the rev D audio shield (for teensy 4) be used as a simple DAC with variable sample rate (i.e. more or less continuous sampling frequencies so I don't need to resample/interpolate to adjust playback frequency)? I would like to be able to go as low as a few hundred Hz, and up to about 16kHz.
Click to expand...

In short, probably not. This is a synchronous I2S chip and relies on quartz-accurate clocking to work - nominally it
supports 8/11.025/12/16/22.05/24/32/44.1/48 and 96kSPS, although I think people have overclocked it, and I
guess it will underclock to some extent too (has anyone tried this?), but you cannot expect to vary the frequency while
it is running/clocking as these synchronous pipelined sigma-delta converters (which I believe both ADC and DAC to be)
rely on precise low jitter clocking for their basic function. However I'm not certain about this, the datasheet doesn't
give much away about the precise implementation other than mentioning its 24 bit, which indicates sigma-delta.

You can stop and restart the clocking at different rates though, but I don't think that quite counts as "simple DAC
with variable sample rate", and I don't know if it will handle 100Hz kinds of rates, internal capacitances might
be present that prevent this, the datasheet doesn't say anything but minimum Fs = 8000

Its basically designed expressly for audio. There is likely no precision in the output gain, for instance, there would be
no point using a voltage reference with such a chip for instance, the two channels won't gain-match exactly...
Furthermore there is gain variation of a percent or so across the pass band due to the internal digital filtering,
so absolute accuracy is about that of a 8 bit converter, but not an issue for audio as this is only 0.12dB pass-band
ripple and inaudible.

What application are you thinking of?
The audio adapter AC-couples all the signals anyway, so its not usable at DC anyway.
 
rvh said:
That's excellent Frank. The setI2SFreq code snippet is just what I was looking for. The only thing I have found in my brief test is that while it worked well from 48kHz down to a little under 10kHz, it may have overflowed a register as I continued to go lower because the sample rate skipped way up high again (at 9887hz) and then reduced gradually again form there.

I'll have a look to see if I can do anything about that (please let me know if you already know the answer). But even if not, this method will solve my problem because I can easily add a simple linear 2,4,8, or 16 point interpolator to reach lower frequencies.

Thanks very much.

Cheers,
Richard
Click to expand...

Yes, it should be possible to reach lower frequencies. There are several prescalers in the chain. Best is to take a look at the reference manual.
However, from memory I don't know which ones are already at "maximum", I simply forgot. :)
If it doesn't work to set e.g. "n" to 8, you have to work yourself in there. But it is not too extensive or difficult. Just concentrate on the divisors of the audio PLL and SAI.
 
MarkT said:
In short, probably not. This is a synchronous I2S chip and relies on quartz-accurate clocking to work - nominally it
supports 8/11.025/12/16/22.05/24/32/44.1/48 and 96kSPS, although I think people have overclocked it, and I
guess it will underclock to some extent too (has anyone tried this?), but you cannot expect to vary the frequency while
it is running/clocking as these synchronous pipelined sigma-delta converters (which I believe both ADC and DAC to be)
rely on precise low jitter clocking for their basic function. However I'm not certain about this, the datasheet doesn't
give much away about the precise implementation other than mentioning its 24 bit, which indicates sigma-delta.

You can stop and restart the clocking at different rates though, but I don't think that quite counts as "simple DAC
with variable sample rate", and I don't know if it will handle 100Hz kinds of rates, internal capacitances might
be present that prevent this, the datasheet doesn't say anything but minimum Fs = 8000

Its basically designed expressly for audio. There is likely no precision in the output gain, for instance, there would be
no point using a voltage reference with such a chip for instance, the two channels won't gain-match exactly...
Furthermore there is gain variation of a percent or so across the pass band due to the internal digital filtering,
so absolute accuracy is about that of a 8 bit converter, but not an issue for audio as this is only 0.12dB pass-band
ripple and inaudible.

What application are you thinking of?
The audio adapter AC-couples all the signals anyway, so its not usable at DC anyway.
Click to expand...

Thanks Mark, much appreciated. Funny enough, my application is actually an 8-bit vintage audio emulation. I don't need to get to DC, somewhere around 300-500 Hz would probably be fine. I saw the codec is able to do 8kHZ, so I'm hoping the I2S routine method proposed by Frank should be plausible down to at least 8kHz, if not lower, which I could combine with a fixed 16pt interpolator (but ideally not).

I haven't looked at how feasible fast changes in sample rate are with that method though. You may well be right that say pitch modulation is not going to work well using that approach. But I'll give it a go.

The alternatives I'm looking at are to wire up a different I2S DAC, either on the Teensy, or possibly even an arduino as the code doesn't really need the speed of a teensy.
 
I tried a few things but no luck still getting Frank's code snippet to work in my code as intended below about Fs=9.9kHz with the rev D codec. And it seems that sample-rate modulation even at modest speeds produces clicks, so I probably will need to find an alternative solution anyway.
 
On a related note, I would be grateful if anyone has suggestions for alternative platforms for me to consider (maybe earlier Teensy boards have simple dac shield options?).

My requirements are not demanding (I thought). I am looking for an Arduino/teensy platform with an option to have continuously variable output rates to a dac that is say at least 10bits, and can accommodate rates from say 500hz to about 22kHz. I hadn't looked into I2C options before, and wrongly assumed the MCP4725 shield would be plenty fast for modest audio like that. But it's appallingly slow as it turns out...

I'm considering an Arduino due, but only as a last resort as it has a lot of features I wouldn't be using.
 
Yup, that code snipped is not intended to be used with constantly changing samplerates or even modulating. Its meant to switch to a fixed frequency.
 
Frank B said:
The 3.x Teensys have DACs
Click to expand...

Thanks Frank, yes I discovered that last night and have ordered a few 3.2s. The 3.x look like my most promising option so far because, if my understanding is correct, I can just use an 'analogwrite' to the dac whenever I like.

Should we be expecting the teensy 4.x to have buillt in dacs at some point?
 
So it's a year down the track and I have my project running very nicely on a Teensy 3.6. But the board is now out of stock everywhere, so I'm looking or possible alternatives. And the extra CPU power of a Teensy 4.x would be nice anyway. So I'm again soliciting for suggestions for how to add a DAC to a Teensy 4.x that I can write to at any time, rather than at a steady sampling rate (i.e. non i2s). Thanks.

p.s. I'm after just 10bits, but in stereo, and up to about 40-50kHz.
 
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