Difference between revisions of "DSP (File Format)"
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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 (File Format)|AGSC]]; the [[CSMP (File Format)|CSMP]] format actually embeds the entire DSP format within it. | 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 (File Format)|AGSC]]; the [[CSMP (File Format)|CSMP]] format actually embeds the entire DSP format within it. | ||
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{{todo|An explanation of how ADPCM works would be nice to have somewhere on this page. Also, a better text explanation for the decoding process to go along with the example code.}} | {{todo|An explanation of how ADPCM works would be nice to have somewhere on this page. Also, a better text explanation for the decoding process to go along with the example code.}} | ||
Revision as of 15:42, 25 January 2015
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.
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This file format has been completely documented This file format is now completely understood and no further research is needed. |
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To do: An explanation of how ADPCM works would be nice to have somewhere on this page. Also, a better text explanation for the decoding process to go along with the example code. |
Header
| Offset | Size | Description |
|---|---|---|
| 0x0 | 4 | Sample count |
| 0x4 | 4 | ADPCM nibble count; includes frame headers |
| 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 | Current address; 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 | Reserved |
| 0x60 | End of DSP header | |
ADPCM 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 header, the bottom 4 bits are the scale value, and the top 4 bits are the coefficient index to use for the current frame.
Example C Decoding Function
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(u8 *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][0]; s16 coef2 = d.coefs[coef_index][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; } } }
