Adaptive Loudness

     Hello Lu,

     By "simpler," do you mean, "fewer instructions?"  And is your DSP the ADAU1701?  If the answers are "Yes," the circuit below, while appearing more complex than the one you found, saves many instructions.  Here the necessary square-root and reciprocal calculations are preformed implicitly -- a technique taught in high school but soon forgotten since no creative use for it is ever presented.  I could go off on how math is taught generally (maybe write a blog post?), but for now -- to see how it works, see page 8 of the AD636 True RMS -- DC converter data sheet from which this idea came from.  Anyhow, this dynamic loudness thing operates very similarly to the one you have found.  It uses the ADI Loudness block to perform the actual loudness comp filtering.  Because this block also adjusts gain on its own, we must apply an inverse gain to keep the overall gain constant.  This fighting the gain changes ultimately limits the loudness comp's dynamic range to about 35 dB -- don't lower the Hard Clip's lower limit since it won't help.   BTW, the implicit circuit needs all three Feedback blocks as shown -- trying to save with a T junction won't work due to a glitch in the ADAU1701's compiler which hasn't been fixed as of SigmaStudio v. 4.5.

     Another way to provide dynamic loudness compensation is to have the audio level control two General Second-Order Index Selectable Filters.  It's more trouble to implement because you need to input a host of filter curves, but much less work for the DSP.  For an example, please see this post.

     Using the ADI Loudness block with a rotary encoder is actually quite straightforward -- just substitute it for your usual Slew External Volume Control block.  If you wish to make it switchable, place both blocks in your project and select the output of one or the other with a MUX.

     Best regards,

     Bob

dynamic-1701Loudness-comp-implicit.zip

     Hello Lu,

     By "simpler," do you mean, "fewer instructions?"  And is your DSP the ADAU1701?  If the answers are "Yes," the circuit below, while appearing more complex than the one you found, saves many instructions.  Here the necessary square-root and reciprocal calculations are preformed implicitly -- a technique taught in high school but soon forgotten since no creative use for it is ever presented.  I could go off on how math is taught generally (maybe write a blog post?), but for now -- to see how it works, see page 8 of the AD636 True RMS -- DC converter data sheet from which this idea came from.  Anyhow, this dynamic loudness thing operates very similarly to the one you have found.  It uses the ADI Loudness block to perform the actual loudness comp filtering.  Because this block also adjusts gain on its own, we must apply an inverse gain to keep the overall gain constant.  This fighting the gain changes ultimately limits the loudness comp's dynamic range to about 35 dB -- don't lower the Hard Clip's lower limit since it won't help.   BTW, the implicit circuit needs all three Feedback blocks as shown -- trying to save with a T junction won't work due to a glitch in the ADAU1701's compiler which hasn't been fixed as of SigmaStudio v. 4.5.

     Another way to provide dynamic loudness compensation is to have the audio level control two General Second-Order Index Selectable Filters.  It's more trouble to implement because you need to input a host of filter curves, but much less work for the DSP.  For an example, please see this post.

     Using the ADI Loudness block with a rotary encoder is actually quite straightforward -- just substitute it for your usual Slew External Volume Control block.  If you wish to make it switchable, place both blocks in your project and select the output of one or the other with a MUX.

     Best regards,

     Bob

dynamic-1701Loudness-comp-implicit.zip

hey bob, thanks for your fast reply!

im using a ADAU1451 exactly this: https://www.hifiberry.com/shop/beocreate/beocreate-4-channel-amplifier/

will this also work? i will check your DSP file and try to understand ;) and later just try to get it work on my test-DSP-board.

just attached my dsp file for you

greetings from germany,

Lu

DesktopSpeaker.rar

Hi Lu,

The short answer is yes- Bob's algorithm will work on the ADAU1451.

Please note that the ADAU1451 is based on a different architecture than the ADAU1701. So some of the blocks in SigmaStudio are slightly different between the two DSPs.

Because of this, I recommend that you re-draw Bob's schematic in your project from scratch.

Josh

Hi Lu & JoshuaB,

     My version of the adaptive algorithm should work fine with the ADAU1451; however you could substitute that chip's divide and square root blocks for the implicit calculation shown.  The ADAU1451 is much more capable, thus its divide block doesn't share the limitations of the ADAU1701 divide block.  Also it offers several times more MIPS to handle the calculations.  For the ADAU1451, your final project will appear much like the adaptive loudness schematic you have shown in your original post.

     Best regards.

     Bob

hey josh, hey bob,

short answer: it works!

thank you guys so much!

btw: i also tryed to increase the bass-output by adding a second rotary volume to the bass-stream (reversed table) and this also works. but anyhow, im using your system.

i have an other question:

my board dousnt start with a audio stream straigt after powering up/switch on.. means, i always need to click one time on "link compile download" then it starts.

my second smaller dsp board does this without problems.

do u guys have any idea why this happens?

greetings,

Luca

Hi,

In order for the ADAU1451 to start up by itself, it needs to have a program stored in the external EEPROM and the selfboot pin must be tied high.

You can do the first part in SigmaStudio by downloading the program; right-clicking the ADAU1451 in the hardware controls page; selecting "Self-boot Memory -> Write Latest Compilation through DSP"; and entering the parameters for your board's memory.

The following capture is from Beocreate's documentation. You should probably use their parameters.

Is there a jumper or switch on the board to tie the selfboot pin high?

Josh

hi josh,

thanks for your reply!

i do this way since i have my two dsp boards. one works "on the fly" (means loaded from eeprom) but not the second dsp-board.

thats the reason its very strange and im asking here ;)

maybe the audio source from the raspi (spotify or via bluetooth) is getting in something like a sleep mode??

thanks in advance

Hello TheLGMaster,

I would like to answer the question: "Is the DSP booting up from its EEPROM?" 

Are you using our eval boards? If you are then tell me the model numbers. 

Most of our eval boards will have a GPIO connected to LEDs. What you should do is place an oscillator in your project to turn a GPIO pin on and off. Then you can "see" if the program is loaded and running when you power up the board. If the program is running then we need to look more closely at other things. If it is not loading up then we need to look closer on how the EEPROM is being written or other hardware questions. 

There are too many variables right now, lets, narrow this down some.

Thanks,

Dave T

hello Bob, very nice block -- I have implemented this in my project. In listening tests, the effect is perhaps too pronounced in my setup, particularly in the bass response. Is there a parameter I can change to back off the effect somewhat? I thought the two "dials" in the Loudness L&H module itself would achieve this but I couldn't discern much of a change by live-auditioning different positions on the dial.

thanks, -Mike

     Hello Mike,

     According to the Sigmastudio Toolbox Wiki, the Loudness block's knobs are supposed to control the amount of boost -- if this doesn't work, we'll instead need to "fudge" the level-sensing part of this setup to make the measured level less sensitive to the actual input level.  This in turn would reduce the effect, hopefully to human size.  I have an idea for this but its implementation depends somewhat upon the function blocks available for the DSP you're using.  Thus I'll ask first, which SigmaDSP are you designing with?

     Best regards,

     Bob

Thanks for giving this a read and some thought. I'm using the 1701.

-Mike