Dual channel 16bit dac PT8211

Hm, looks exactly like the picture in the Datasheet now:
screenshot.jpg
(i've set the right channel to output zeros. top: BitClock, mid: LR-Clock, last: Data)

Image1.jpg

I'll make a new "output_pt8211" in my github and a pullrequest as soon i have verified that it works. Should have the chips in a few days - have some sop-adapters that i can use for testing..
 
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This format is called "right justified", IIRC. Maybe this more generic name is better suited for the object?
 
Frank B said:
let's hear what Paul says..

https://github.com/FrankBoesing/output_pt8211

I'm very curoius wether it works or not..

edit: there are some more changes:
- i've disabled MCLK (not needed)
- i've deleted the i2s input
Click to expand...

First, @BEN, thank you for the chips!

I soldered one to an SOP-Adpterboard, output is connected directly to a chinese amplifier-board for a few $ - and it WORKS.
Ok, at least it sounds ok - i have a small speaker connected and can't say wether the qualtiy is good or not.

Maybe, someone with an oscilloskope can check the waveforms ?

@Paul, want a pullrequest ? The waveform-check is the only thing missing, and maybe it could go into the next beta ?

DSC_0318[1].jpg

 
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Oh that's great news! I can help with the scope waveform, but not before next week. Do you power the DAC from 5 or 3V3?
 
Ben said:
Oh that's great news! I can help with the scope waveform, but not before next week. Do you power the DAC from 5 or 3V3?
Click to expand...

Both works. I measured 5.1 Volt.
Don't hurry.. :)

Maybe next week, we can speak about filters and a Vreg (if required...)
I want to make a board (I'm still undecided: With amplifer or not, headphone?..)

I need a good output for another, very big (well, amount of work..) project. (mono)
 
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Frank B said:
Correction: Some problems with 5V: after some minutes of playing it stops.
3.3 Volt works better.
Click to expand...
Eww that's nasty. Can you try adding some capacity to the 5V rail? I think 10µF is safe for the USB port, and another 100nF across the DAC?
I managed to get some SO8 adapters from the friendly guys at our physics department, so I'll be able to hook up the scope sometime next week, and I found a schematic and layout for a 3rd order low pass, sadly the layout is a bit big by todays standards, SO8 OP and 1206 R and C, single sided layout; that would need an update. But let me look at the scope first before we rush to conclusions :D
 
There are already 470µF on the Amplifier-board :) But i can try it tomorrow.

The datasheet shows a simple RC-Filter (15k+470pf) - is that ok ?
I want to add a amplifier to my project-board, maybe mono, and if possible with optional headphone connector. Which chip can i use ? >1mm pitch would be best. ~ 1 watt.
 
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Frank B said:
There are already 470µF on the Amplifier-board :) But i can try it tomorrow.
Click to expand...
Oh ok. That's strange, 5.1V is totally within spec. Maybe the 100n will help...
Frank B said:
The datasheet shows a simple RC-Filter (15k+470pf) - is that ok ?
Click to expand...
Thats -3dB at about 22.5kHz, surely won't hurt, but maybe won't help either. I'll have to look at the FFT when the scope is hooked up.
Frank B said:
I want to add a amplifier to my project-board, maybe mono, and if possible with optional headphone connector. Which chip can i use ? >1mm pitch would be best. ~ 1 watt.
Click to expand...
Hard to say, maybe the Ti LM4861 (http://www.ti.com/lit/ds/symlink/lm4861.pdf)? Available at reichelt in SO8 (http://www.reichelt.de/ICs-LM-2000-...TION=3&GROUPID=5466&ARTICLE=39469&OFFSET=500&).
But adding virtually any IC to this project destroys the idea of a cheap 25ct DAC, so there really is no "correct" choice IMHO. On the other hand, a hobby doesn't need to make sense, and who am I to judge :D
 
Ben said:
Oh ok. That's strange, 5.1V is totally within spec. Maybe the 100n will help...
Thats -3dB at about 22.5kHz, surely won't hurt, but maybe won't help either. I'll have to look at the FFT when the scope is hooked up.
Hard to say, maybe the Ti LM4861 (http://www.ti.com/lit/ds/symlink/lm4861.pdf)? Available at reichelt in SO8 (http://www.reichelt.de/ICs-LM-2000-...TION=3&GROUPID=5466&ARTICLE=39469&OFFSET=500&).
But adding virtually any IC to this project destroys the idea of a cheap 25ct DAC, so there really is no "correct" choice IMHO. On the other hand, a hobby doesn't need to make sense, and who am I to judge :D
Click to expand...

I added 10uF, 100nF and the resistor. Works good now.
But, possibly, it could have been a bad connection on my breadboard.

LOL, yes..sense...yes there is none. From this point of view, my whole project does not make sense. And i could use the teensy-dac.
I want to add the DAC+amplifier+a small speaker to my ili-9341 display-board.
 
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Frank, do you have a working sketch using your PT8211 object you can share? My Teensy stalls before a single loop() is completed (checked with Serial), so a known-good example would be great.

I installed a fresh Arduino 1.6.9 + Teensyduino 1.29 Beta #4 (minutes before Paul announced the official release :rolleyes: ) and copied the files from your git. I checked the Audio.h manually and the only difference is the #include "output_pt8211.h", so this should be ok. Maybe my Teensy is faulty, it's one of the first T3.2 and sat in a drawer for quite some time...
 
Hi Ben,
every sketch should work. You can use the audio-lib examples. But i can post a tested one this evening after work.
If there is a SGTL5000 control in your sketch, remove it. It tries to talk to the chip via i2c, maybe this is the problem (does it block?)

Frank.
 
Frank B said:
Hi Ben,
every sketch should work. You can use the audio-lib examples. But i can post a tested one this evening after work.
Click to expand...
I'd appreciate it. I didn't start with the examples but from blank, maybe it was just to late in the evening to generate proper code :D

Frank B said:
If there is a SGTL5000 control in your sketch, remove it. It tries to talk to the chip via i2c, maybe this is the problem (does it block?)

Frank.
Click to expand...
No, just two sine blocks and the PT8211 (and patch cords). I will try again tomorrow, I'm sorry I won't have time today though.
Ben
 
Ok,

here are detailed instructions :)

Paul wants to merge both outputs into one, so let's try to use output_i2s instead of my stripped down variant.

1. Edit output_i2s.cpp
in Line 320/321, remove the Frame-Sync-Early-Bit :
Code: 
 I2S0_TCR4 = I2S_TCR4_FRSZ(1) | I2S_TCR4_SYWD(15) | I2S_TCR4_MF
        |[COLOR=#ff0000][B] /* I2S_TCR4_FSE | */[/B][/COLOR] I2S_TCR4_FSP | I2S_TCR4_FSD;
That's all needed to bring the PT8211 to work. And, hey, it STILL works with the SGTL-Chip after that change.


2. Use this Sketch:

Code: 
#include <Audio.h>
#include <Wire.h>
#include <SPI.h>
#include <SD.h>
#include <SerialFlash.h>

// GUItool: begin automatically generated code
AudioSynthWaveformSine   sine2;          //xy=147,346
AudioSynthWaveformSine   sine1;          //xy=181,270
AudioOutputI2S           i2s1;           //xy=353,308
AudioConnection          patchCord1(sine2, 0, i2s1, 1);
AudioConnection          patchCord2(sine1, 0, i2s1, 0);
// GUItool: end automatically generated code


void setup() {
 AudioMemory(10);
 sine1.frequency(1000);
 sine1.phase(00);
 sine1.amplitude(0.9); 

 sine2.frequency(1000);
 sine2.phase(90);
 sine2.amplitude(0.9);  
}
void loop() {}

Thats all.

have fun,
Frank.

p.s. no need to hurry..
I have big problems to get I2S on the overclocked K66 to work... but for my project i need heavy overclocking - so ... might be easier to use the DAC.
 
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Frank, thank you so much. I edited the *.cpp and I'm running your test sketch. It worked right of the bat. I don't know what I did wrong on Saturday, must have been a coding error, I use the same hardware as before.

Have a look:

ref_fft2.png
This is the output waveform without any external circuitry apart from 47nF across the rail, powered from Vusb. Note that the scope applies averaging and due to trigger jitter the waveform is a bit distorted. In reality the waveform is even more "stepped" and slews between individual voltage levels in about 400ns, holding the voltage very stable for the rest of the sample time. So in conclusion, there is no output filtering at all in the DAC.

Also interesting: with the sketch you provided (amplitude=0.9) I get a signal of about 920mV peak-peak, which means barely over 1V full-scale. From the datasheet I expected more than double the amplitude. Any idea?



This is an FFT (done with the scope, so no high precision measurement, but the DAC is bad enough to have it's flaws exposed even with that...)

ref_fft.png


ref_fft1.png

The leftmost peak is what we want, at 1 kHz. Scaling is 10dB per division, so the lower harmonics are about 50db below the signal. Bad, very bad, but what do we expect for 25 cent... Notice the horrendous amount of nyquist errors mirroring around the sample frequency to the right of the spectrum, 1kHz before and after 44.1kHz. I placed the scopes cursor on one of those peaks in the third picture. I also did a FFT up to several MHz and I can see signals up to about 2MHz.

Conclusion: We definitely need a reconstruction filter, and I'd say we go with butterworth instead of the more "audiophile" bessel to get those nyquist images as low as possible and because any attempt to treat this signal with care is a waste of time and effort.

I'll do some tests regarding the DACs sensitivity to power quality and 5V vs. 3V3 operation later this week, and I'll cobble together some filtering on the output to see what passive filtering can do.
 
Ben said:
From the datasheet I expected more than double the amplitude. Any idea?
Click to expand...
Is it ignoring the first bit ? We could try to invert the bitclock and see if that helps.. In the datasheet-diagrams is a pause before the first bit.
attachment.php

Ben said:
I'll do some tests regarding the DACs sensitivity to power quality and 5V vs. 3V3 operation later this week, and I'll cobble together some filtering on the output to see what passive filtering can do.
Click to expand...

I could try to use 4 x oversampling, if that makes things easier (less external parts?) - is it worth to try it ?
 
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Frank B said:
Is it ignoring the first bit ? We could try to invert the bitclock and see if that helps.. In the datasheet-diagrams is a pause before the first bit.
Click to expand...
Not ignoring, I think it treats the MSB as the second-most SB and defaults the MSB to 0. But yeah, certainly a problem with the digital interface.
Frank B said:
I could try to use 4 x oversampling, if that makes things easier (less external parts?) - is it worth to try it ?
Click to expand...
If you can run the output object with higher sample rate than the rest of the audio lib, sure, would be great. You'd need to decide on the interpolation method though, a high order low pass would eat up a lot CPU time. All in all, if you just want to drive a small speaker integrated in a display, this is too much effort. But I can totally see two of these DACs used with the Quad-Output object for sensor-actor-applications (think PID control loops), and in those applications oversampling might be worth the effort. I suppose we should ask Paul if he's ok with us developing a low-cost alternative to the audio shield for analog output. Maybe a single board with two PT8211 for cheap quad out?
 
I can not imagine many use-cases for four channel audio on the teensy...

I look if I can do anything re: the digital interface. But I don't own an oscilloscope .
Your screenshot shows less than 32 steps... ?!
Im not audiophile and my ears are not the best- for me , the sound is ok.. even with a larger speaker (?)
On the other hand, sometime ago i read of china-chips (i2s) that had 8 bit R2R instead16...

Edit: The max. value for amplitude is 1.0, not 0.9
 
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Oh no my fault regarding the amplitude. The output scales with the supply voltage. I think I got that previous measurement when I had it running off of the 3V3 rail. At 5V the output swings 2.3 Vpp, thats in spec according to the datasheet. The interface is ok as it is!
So Frank, I guess it's ok to use the chip as it is for your display application. If you want, add a low pass with 8.2kOhm and 1nF to the output.

I will include resistors and caps for a 3rd order Sallen-Key LP filter in my next order, just for the fun of it. And I ran out of rail2rail OPs in DIP, so I can't continue with the filter right now.

Regarding the few steps: At 1kHz the slew rate is more than 1 LSB per sample time (1/fs), that's why it looks so coarse. Look at the minima and maxima of the sine, the steps are much smaller there. It may still be that the DAC has less than 16 bit resolution, but I'd need a precision multimeter to measure this. I have some at work, but we are currently moving into another building and everything is packed away in boxes, so I won't be able to do that measurement for at least another month.
 
Ben said:
Also interesting: with the sketch you provided (amplitude=0.9) I get a signal of about 920mV peak-peak, which means barely over 1V full-scale. From the datasheet I expected more than double the amplitude. Any idea?
Click to expand...

I have 2.42 Volt. Measured with output of constant values 32767 vs(minus) -32768 - hacked the I2S output-code for it.. :)
For 16383 / -16384 , it is the half - so...

- peak-peak is ok
- MSB is ok

Do you have an audioshield ? Could you pls. check and do your measuring with the osc. with it ? Just to make sure that it's no software-bug somewhere in the audio-lib.
 
Ben said:
it's all good, it was my fault, see my post #45.
Click to expand...

No Problem... :D

Maybe, if I can find some time, I try to add upsampling to the output-objects. It will make the external filtering a bit easier, I hope. (4x 44= 176KHz )
Let's see...

Is linear interpolation sufficiant, or do we we need a more advanced interpolation-filter ? CIC ? FIR? (but at the moment I don't know how to calculate them ;-)
I don't want to waste more CPU-cycles than necessary !
 
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Ok just a quick feedback, I really need to go to bed. This looks EXCELLENT. The linear interpolation can be seen clearly on the scope, works as expected. I measured 2.22% max CPU time with the sketch attached below. Very promising. Good work Frank.


Code: 
#include <Audio.h>
//#include <Wire.h>
//#include <SPI.h>
//#include <SD.h>
//#include <SerialFlash.h>


AudioSynthWaveformSine			sine1;
AudioOutputPT8211				PT8211_1;
//AudioConnection				patchCord1(sine1, 0, PT8211_1, 1);
AudioConnection					patchCord2(sine1, 0, PT8211_1, 0);


void setup() {
	Serial.begin(9600);
	
	AudioMemory(10);
	sine1.frequency(10000);
	sine1.phase(0);
	sine1.amplitude(0.9); 
}

void loop() {
	Serial.print(AudioProcessorUsage());
	Serial.print("   ");
	Serial.println(AudioProcessorUsageMax());
	delay(100);
}

I'll do some FFT's for comparison tomorrow, with and without RC filter. If you have any sketches you want me to view on the scope please go ahead and post them, I'll be happy to share scope screenshots, raw data, whatever you need.

Ben

PS: Unfortunately I'm no expert with digital filters either. I know CIC is super easy and fast on FPGAs, but has it advantages over "normal" FIR/IIR when implemented on a micro? For FIR/IIR calculation I know some tools, but not for CIC, and I don't know how to efficiently implement CIC filters in C++.
 
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