STM32F4 loudspeaker processing part 2

Second step

So you got that blinking light, maybe you did some stuff with on board sensors or an external control. Nice!
By this time you must have stumbled across the 'library' thing.

At this date I still do not understand the engineering/design philosophy behind the library concept:
Not structured. Not documented. No rules. Not standardized.
And as such a big cause of bugs and conceptual problems.(not only caused by yourself!)

Still you will need them: standing on the shoulders of others.

This alone took one month trial and error to get a working configuration going.
As I am using STM32 microcontrollers I use the HAL (hardware abstraction layer) libraries provided by STM.
More specifically: as these are generated by CubeMX

Next step is to get effluent in using CubeMX to generate the projects base code, then open/import this project in KEIL MdK to add your user code. Maybe get some knowledge about the build in debugging tools. These won't help you a lot in processing audio though, so don't waste to much time there..


Third step

Be brave and connect your first audio board DAC to your MCU board. Maybe you all ready have a DAC on your discovery board as with a STM32F4DISC board.

Now start to understand what is needed to stream i2s data to your DAC. Dig through all those data sheets to learn and find details about the several i2s protocols and how to configure both DAC and MCU. Don't despair! Just familiarize yourself with the keywords and trust the work others have done for you (at least at first).
Apart from the different protocols (use Philips!), the most important concept is: clock signals. To make things more confusing different names are used:

MCK  = CK = master clock
WS = LRCK = word select or LR clock
BCK = SCK = bit clock or serial clock

As audio freaks we do understand sample frequency and bit depth, and how they work out on the different clocks, don't we?
Now generate a sine-wave in code or use some table or whatever and see if you get a peep without cracks and or distortion. If you do consider yourselves lucky and talented. If not learn more about data formats! There is also some good youtube videos by Mutex if you get stuck.
Remember: this is bare metal coding so you have to control everything!
Again 1 more month of struggling: i2s audio data is a signed integer between -1 and 1 (times bithdepth -1) aka two's complement
You will have to control the data format all the time, casting stuff like:

 txBuf =(uint16_t)data_out //filling send buffer with 'typecasted' correct data//

And also be aware of overflow! You can't just multiply or divide your data.
These are helpful:

data_out>>1 //data halved as in 6 dB down in volume//

And hey, here's interesting stuff: ever participated in the floating point versus fixed point debate of audio processing ?? hmmm..


next to part 3

Powered Loudspeakers + DSP...

Why powered speakers?

Obviously, most loudspeaker boxes will contain individual drivers for maybe high, low, mid frequencies.
So we must provide a way to separate those frequencies.
Commonly this is done with 'passive' components (i.e. inductor / capacitor networks). I could write a post on that one day.
But for more control and that precious phase/time alignment, you will need some DSP.
In pro-audio this traditionally has lead to 19'amp racks filled with amps and  some DSP (maybe like one off those Soundwebs).
Anyway, things get rather bulky fast. So if you need just 1 (euhm, mostly 2 as in stereo) boxes, maybe as a fill-in or delay or just some soundscape thingie, something compact would be nice.
Hence the popularity of 'powered' boxes.

Now for the difficult part:

Most of  the powered loudspeakers in the market are not of a quality level we do appreciate, those plastic boxes will serve a purpose, but not ours...
So here's the design goal: make a powered box with a build in DSP//amplifier (no user controls!)
So first we checked the market to find a ready made 'plate-amp' to fit into our loudspeakers. People into this know the brands,yup I tested them all and I will not state my opinion on either off them .. :)
The main problem is that they are either limited in DSP (no FIR) or highly suffering from unstable software due to featurites.
So phase 1:

Build a DSP unit:

There you go, home brewn DSP with a Sharc SigmaStudio ADAU1701.
Check the mix of through hole audiophile components (WIMA, panasonic) and smd stuff (man, soldering these manually..aaarggh..)

Nothing really new here hardware wise, everybody knows those MiniDSP boards, but this one is programmed with the free Sigma Studio software, so we can do stuff like FIR.

Problem solved?

Hmmm, not really: apparently it is difficult to mix analog and digital circuitry. And those on board ADC's are not the most brilliant.
I did some test with SPdif and i2s inputs, which did improve the sonic quality, but things get complicated fast.

We could use some different DSP with a dedicated AD/DA section, but you will quickly find that the choice off self-boot chips is limited. All the rest will need some MCU to boot and control.
And hey, here's an idea:

Why not use a MCU  (you know like a arduino) to do all processing?
There is a zillion off boards available out there and they are all dead-cheap!

One tiny problem: not that many people are doing audio stuff with them, google doesn't really help

So stay tuned to find out more...

Working Distortion

Those friends that know me from real life, will have heard me sighing in despair:

Audio reproduction is all and only about hustling distortions around.

Aside from all (FFT-) techniques that should be common knowledge by now, we still have a long way to go in having our parameters to describe 'good sound'.
First statement to make:
Audio passing from one end to another can never be 'better'.
Be it a microphone, loudspeaker or any electronics.
So if it only can deteriorate: then audio engineering boils down to manipulating in which way it get's worse.
Isn't that sad  😥

If we manage everything correctly in the frequency domain, as described here in this blog, then what's left?

Distortion!!

Sure we all know this should be minimal, shouldn't it?
And we all express it as THD, oh wait let's add in N(oise)  too!
Not a very elaborate way to express what we are trying to study is it?

And why are all these Waves plugs so highly loved and wanted?
Tasty spices also known as harmonic distortion.

Check my current home-brew study object: