analog signal processor, front panel, circa 2013

the single biggest difference between 'hacky' and 'solid' audio gear is the box it comes in.

this isn't just vanity. well-constructed equipment lasts, which justifies its cost in cash and time. you won't worry when you move it. you won't worry when you touch it. the interconnects won't wiggle out of place, the volume control won't crackle or free-spin. the bloody thing won't catch fire.

if it's performing for you, or you're performing with it, it won't give up, turncoat and die.

it is a real pity that most hackers half-ass their boxes. above is my own attempt, in 2013, to turn a cigar box into an electronics chassis. there's no symphony to the position of anything. it's hacked together in all senses of the word.

the inside is worse.

analog signal processor, inside view, circa 2013

are the circuit boards anchored by the tension of the wires? yes.

are the jacks Loctite-ed with hot glue? yes.

is the volume pot protoboarded and bolted to scrap plastic? yes.

is every input and output tight-roped on unshielded wires? yes.

this isn't even all. there were two stereo amplifiers and three separate power supplies, all with their own hacked-up boxes, and cables strung in between.

it even took ages to hand-cobble this disaster. i'd completely overlooked mechanicals during the design phase. here was the result. never again.

DSP 01 would be a singular, hyper-integrated thing. i would nestle the user interface, DSP core and quad-channel amplifier board into a single box. one 24V power supply would power the amplifiers as well as the DSP, through a 5V buck regulator.

the UI board plugs directly into the main DSP board, giving it RCA inputs, RCA/3.5mm outputs, two adjustment pots and indicator lights. i added a tiny knob to dim the power light. i lasercut holes in the front panel with my mechanical Eagle CAD technique, to exactly fit the pots and RCA jacks. the amplifier's heat sink would orient vertically to maximize passive cooling.

then i went to lasers.

those screws are too high. i'll have to cut holes around them.

all precision projects must ebb to barbarism.

next update, i will describe my acoustic measurements and DSP programming.

Tshen2 2014

here are the goals i set for the DSP:

stereo inputs with clip-detect lights. DSPs are notorious for gain staging problems, so i wanted the user to give it the largest possible signal without clipping the inputs.
six outputs to support stereo triamplified speakers or stereo biamplified speakers + subwoofer.
modular user interface for an arbitrary configuration of switches or dials.
easy to integrate with off-the-shelf amplifier boards, into a single chassis.
can be powered by a single-cell Li-ion battery for portable use.

the heart of my DSP is Analog Device's ADAU1701. it is an amazing piece of work. two 24-bit ADCs and four 24-bit DACs. integrated 50MHz DSP core. self-boot capability. auxiliary 8-bit ADC to read potentiometers.

coming from a place of analog skepticism, i expected it to sound awful. i planned to use digipots on the outputs to do volume control in analog, and avoid digital attenuation. i threw in a microcontroller (ATtiny) for clip-detect and to adjust the digipots. i added output amplifiers.

then i actually listened to the evaluation board and it sounded fine. the digipots and output amplifiers were completely extraneous. clip-detect could be built into the DSP itself - i could leave out the microcontroller too. this led to a much simplified version 2:

this includes:

ADAU1701: the DSP core, including 2 inputs and 4 outputs.
AK4430: additional 2 outputs, for 6 outputs total.
DIR9001: SPDIF receiver to receive digital audio.
USBi: Analog Device's SigmaDSP programmer.

after more listening, i decided the SPDIF input was also extraneous. i would include the hooks to add it later. it could live on the modular user interface, and connect through a header.

these changes led to version 3:

why a 24.576MHz clock? the AK4430 delivers maximum THD+N if your master clock is at least 512*(sample rate). at a 48kHz sample rate, that's a minimum clock of 24.576MHz. this relationship exists because the AK4430 uses a switch-cap output filter. like all switch-cap filters, its corner frequency is affected by the clock frequency.

why the AK4430 at all? it's good enough, and quite cheap, and runs at 3.3V.

now let's calculate our power requirements:

looks like our worst-case current draw is about 200mA. what about power supply noise?

ADAU1701 power supply noise requirements:

PSRR is 50dB at 1kHz, with 200mV signal on AVDD
FS (full scale voltage) is 2.5 Vpp
SNR is 104dB
PSRR noise is PSN (power supply noise) - 50dB
PSN - 50dB << FS - SNR
PSN << FS - 54dB
PSN << 5 mVpp

AK4430 power supply noise requirements:

PSRR is 62dB at 1kHz, 50mVpp signal on VDD and CVDD
FS is 5.66 Vpp
SNR is 104dB
PSN - 62dB << FS - SNR
PSN << FS - 42dB
PSN << 40 mVpp

the ADP3335 LDO has excellent load regulation, 500mA maximum output current and a peak voltage noise of under 0.3 mVpp. it will do fine.

i'll also throw on a 30-pin header to connect the Main Board and the UI Board.

time to bang it out!

OSHpark time!

next update, i will describe the user interface and lasercut chassis.

Tshen2 2014

DSP 01, part 3: UI and lasercut chassis

DSP 01, part 2: architecture & electronics