DSP (File Format)

The .dsp format is a common GameCube/Wii format for audio that comes with the SDK. It encodes sound using Nintendo's ADPCM codec. The same ADPCM codec is also embedded into several Retro Studios format, like AGSC; the CSMP format actually embeds the entire DSP format within it.

Header

Offset	Size	Description
0x0	4	Sample count
0x4	4	ADPCM nibble count
0x8	4	Sample rate
0xC	2	Loop flag; 1 means looped, 0 means not looped
0xE	2	Format; always 0
0x10	4	Loop start offset
0x14	4	Loop end offset
0x18	4	Always 0
0x1C	2 × 16	Decode coefficients; this is 8 pairs of signed 16-bit values
0x3C	2	Gain; always 0
0x3E	2	Initial predictor/scale; always matches first frame header
0x40	2	Initial sample history 1
0x42	2	Initial sample history 2
0x44	2	Loop context predictor/scale
0x46	2	Loop context sample history 1
0x48	2	Loop context sample history 2
0x4A	2 × 11	Padding
0x60	End of DSP header

Audio Data

The ADPCM audio data is split up into multiple frames. Each frame is 8 bytes; it starts with a one-byte header, then has 7 bytes (or 14 samples) of audio data. For each frame, the bottom 4 bits are the scale value, and the top 4 bits are the coefficient index to use for the current frame.

Sample decoding code (vgmstream used as reference):

static const s8 nibble_to_s8[16] = {0,1,2,3,4,5,6,7,-8,-7,-6,-5,-4,-3,-2,-1};

s8 get_low_nibble(u8 byte) {
    return nibble_to_s8[byte & 0xF];
}

s8 get_high_nibble(u8 byte) {
    return nibble_to_s8[(byte >> 4) & 0xF];
}

s16 clamp(s32 val) {
    if (val < -32768) val = -32768;
    if (val > 32767) val = 32767;
    return s16(val);
}

void DecodeADPCM(char *src, s16* dst, const DSPHeader& d)
{
  s16 hist1 = d.initial_hist1;
  s16 hist2 = d.initial_hist2;
  s16 *dst_end = dst + d.num_samples;

  while (dst < dst_end)
  {
    // Each frame, we need to read the header byte and use it to set the scale and coefficient values:
    u8 header = *src++;

    u16 scale = 1 << (header & 0xF);
    u8 coef_index = (header >> 4);
    s16 coef1 = d.coefs[coef_index * 2];
    s16 coef2 = d.coefs[coef_index * 2 + 1];

    // 7 bytes per frame
    for (u32 b = 0; b < 7; b++)
    {
      u8 byte = *src++;

      // 2 samples per byte
      for (u32 s = 0; s < 2; s++)
      {
        s8 adpcm_nibble = (s == 0) ? get_high_nibble(byte) : get_low_nibble(byte);
        s16 sample = clamp(((adpcm_nibble * scale) << 11) + 1024 + ((coef1 * hist1) + (coef2 * hist2)) >> 11);

        hist2 = hist1;
        hist1 = sample;
        *dst++ = sample;

        if (dst >= dst_end) break;
      }
      if (dst >= dst_end) break;
    }
  }
}