General presentation

The AWM2 is in charge of handling the individual channels. It manages reading the rom, decoding the samples, applying volume and pitch envelopes and lfos and filtering the result. Each channel is then sent to the mixer for further processing.

The sound data can be four formats (8 bits, 12 bits, 16 bits, and a 8-bits log format with roughly 10 bits of dynamic). The rom bus is 25 bits address and 32 bits data wide. It applies four filters to the sample data, two of fixed type (low pass then highpass) and two free 3-point FIR filters (used for yet another lowpass and highpass). Envelopes are handled automatically, and the final panned result is sent to the mixer.

Sample addressing

For each channel, four registers directly control the sample addressing:

  • (32b) rom information
    • bits 31-30: sample type
    • bits 29-25: loop sample count, decimal part
    • bits 24-0: loop start or end of sample address
  • (32b) pre-loop sample count
  • (32b) loop sample count
  • (16b) replay frequency

Sample format


The MEG is a DSP with 384 program steps connected to a 0x40000 samples ram. Instructions are 64 bits wide, and to each instruction is associated a 2.14 fixed point value, Every third instruction (pc multiple of 3) can initiate a memory access to the reverb buffer which will be completed two instructions later. Each of those instructions is associated to a 16-bits address offset value.

The DSP also sports 256 rotating registers (e.g. register 1 at run <n> becomes register 0 at run <n+1>) and 64 fixed registers. The fixed registers are used to store the results of reading the samples ram and also communicate with the mixer.

Every 44100th of a second the 384 program steps are run once in order (no branches) to compute everything.

24 LFO registers are available (possibly more). The LFO registers internal counters are 22 bits wide. The LSB of the register gives the increment per sample, encoded in a special 3.5 format. With scale = 3bits and v = 5bits,

   step  = base[scale] + (v << shift[scale])
   base  = { 0, 32, 64, 128, 256, 512,  1024, 2048 }
   shift = { 0,  0,  1,   2,   3,   4,     5,    6 }

The 21th bit of the counter inverts bits 20-0 on read, those are interpreted as a 0-1 value, giving a sawtooth wave.

8 mappings can be setup, which allow to manage rotating buffers in the samples ram easily by automating masking and offset adding. The register format is: tttttsss oooooooo. 't' is not understood yet. 's' is the sub-buffer size, defined as 1 << (10+s). The base offset is o << 10. There are no alignment issues, e.g. you can have a buffer at 0x28000 which is 0x10000 samples long.


The SWP30 is the combination of a rompler called AWM2 (Advanced Wave Memory 2) and an effects DSP called MEG (Multiple Effects Generator). It also includes some routing/mixing capabilities, moving data between AWM2, MEG and serial inputs and outputs with volume management capabilities everywhere. Its clock is 33.9MHz and the output is at 44100Hz stereo (768 cycles per sample pair) per dac output.

I/O wise, the chip has 8 generic audio serial inputs and 8 outputs for external plugins, and two dac outputs. The DAC outputs are stereo, and so is the first generic input. It's unclear whether the outputs and the other inputs are stereo. The MU100 connects a stereo ADC to the first input, and routes the third input and output to the plugin boards, but not the left/right input clock, arguing for mono.


The chip interface presents 4096 16-bits registers in a 64x64 grid. They all seem to be read/write. Some of this grid is for per-channel values for AWM2, but parts are isolated and renumbered for MEG regisrers or for general control functions.

Names we'll use in the rest of the text:

  • reg(y, x) is the register at address 2*(y*0x40 + x)
  • ch<nn> is reg(channel, xx) for a given AWG2 channel
  • sy<nn> is reg(nn/2, 0xe + (nn % 2))
  • fp<nnn> is reg(nn/6, 0x21 + 2*(nn % 6))
  • of<nn> is reg(nn/2, 0x30 + (nn % 2))
  • lfo<nn> is reg(nn/2, 0x3e + (nn % 2)) for nn = 0..17

 The mixer gets the outputs of the AWM2, the MEG (for the previous
 sample) and the external inputs, attenuates and sums them according
 to its mapping instructions, and pushes the results to the MEG, the
 DACs and the external outputs.  The attenuations are 8-bits values
 is 4.4 floating point format (multiplies by (1-mant/2)*2**(-exp)).
 The routing is indicated through triplets of 16-bits values.
 ch33       dry (msb) and reverb (lsb) attenuation for an AWM2 channel
 ch34       chorus (msb) and variation (lsb) atternuation
 ch35-37    routing for an AWM2 channel