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(digital) audio delay/looper (proof of concept)

even if there are solid commercial solutions for the plausible use cases, the aim here is to create a working low-budget basis that can be customized for special needs. (also, the cheaper commercial solutions seem to have a delay/loop time or channel count that is not satisfactory)

also consider this a proof of concept for using stm32 for simple audio effect units..

also see file sample.wav: it somewhat works, but is rather noisy. testsetup (and interfacing circuits, which obviously could be better):

  • computer audio jack connected directly to mixer (A&H PA12) line in
  • bal (aux) line to ADC / controller (neg and ground connected)
  • DAC / controller to mixer line in (neg and ground connected)
  • mixer PFL to headphones

Fancy controller photo

basic architecture

An interrupt handler at sampling frequency performs the following operations:

  1. get new sample from ADC
  2. pass new sample to storage logic
  3. get old sample from storage logic (using given delay)
  4. pass a half-wet mix to DAC

The storage logic stores or retrieves single blocks (in polling mode) once the buffers as used by above interrupt handler are full/empty. Alternative/possible operation/optimization:

  • Use bigger buffers to allow for writes/reads of more than one block at a time -> singular delays exceeding the timing restrictions may thus be spread over several blocks.
  • Use non-blocking sdio operation.
  • Combine sdio read/write operation instead of separating logic -> if delay is align blockwise block buffers are full/empty at the same time

data throughput

for sampling rate of 48khz and 8bit sample resolution needs throughput of at least 48kHz x 1 bytes = 48kByte / sec / channel; or given storage blocks of 512 bytes: 48k x 1 / 512 = 93.75 combined average block read+writes / sec, or a max time limit of 10.666.. ms per read/write operation.

furthermore 48kB / sec / channel means a required storage space of 48kB * 60 = 2.81... MB / minute / channel or ~164.79... MB / hour / channel.

microcontrollers have quite the memory restrictions thus making (longer) storage of audio data impossible. SD cards offer themselves as a cheap and accessible way to provide the memory capacity needed (and could even be used as recorders).

the basic SPI interface of SD cards requires 7-10ms on average; as tested on a nucleo- f303k8.

spi test code: https://gist.github.com/tschiemer/c77ccae718c401fa7604f56b68034944

the 4bit SDIO mode (on nucleo-l476rg) seems to typically require roughly 3200 usec for one combined read/write operation occassionally significantly more (10+ ms) as tested on a stm32l476rg with suboptimal code.

Note: a recorder or player only requires write or read operations (obviously) and might have better sdio timing conditions

For basic throughput limitations given due to chosen testboards: stm32l476rg: "Data transfer up to 50 MHz for the 8 bit mode" (-> 25 MByte/s for 4 bit mode) stm32h476zi2: "Data transfer up to 208 Mbyte/s for the 8-bit mode." (104 MByte/s for 4 bit mode)

AD/DA

(some) stm32s provide 12bit ADCs and DACs, possibly opamps, so along with the SDIO interface they offer themselves for a budget friendly basis. nucleo-l476rg was chosen for the moment being.

Basic line in circuit that works for a mixer line level out signal:

                3V3
                 |
                 R1 100k
                 |
LineIn -----C1---x---- ADC
           0.1u  |
                 R2 100k
                 |
                GND

It is really suboptimal, especially if the 3V3 source is taken directly from the nucleo board and other components (like the sdio) are connected to it aswell: noise on the power line is clearly heard, also any noise from the power source (power adapter or computer) is passed on. But hey, it's a proof of concept.

Interesting: STM32 ADCs allow for differential input.

Also see: https://forum.arduino.cc/index.php?topic=567581.0

ADC optimziations as explained by stm: https://www.st.com/resource/en/application_note/cd00004444-understanding-and-minimising-adc-conversion-errors-stmicroelectronics.pdf

Differential biasing techniques: http://ww1.microchip.com/downloads/en/appnotes/00842a.pdf

A line out circuit isn't necessarily needed (no guarantees are given for any attached equipment) but are suggested. Essentially the DC bias should be removed with an inline capacitor, a post-capacitor resistor connecting line to ground has been suggested to avoid backflow/charging of capacitor (as I understand). Audio equipment often comes with an input DC block already - but relying on this might risk your equipment.

STM32s often have two (buffered) DACs if they have any, so outputting a differential signal would be possible aswell.

For more: https://forum.allaboutcircuits.com/threads/removing-dc-offset-from-a-signal.65288/

https://www.allaboutcircuits.com/technical-articles/dac-output-circuitry-for-an-arbitrary-waveform-generator/

pin assignment

NUCLEO-L476RG

connector pin function
CN7 1 / PC10 D2 (sdio)
2 / PC11 D3 (sdio)
3 / PC12 CLK (sdio)
4 / PD2 CMD (sdio)
CN10 1 / PC9 D1 (sdio)
2 / PC8 D2 (sdio)
CN8 1 / PA0 ADC
3 / PA4 DAC

Ground and 3v3 are self speaking.

SD card:

AD/DA: see above

further references