DSP Memory Region

Some of this is known to be incorrect. I'll correct this in a few months after a bit more RE work. Merry (talk) 21:10, 28 January 2016 (CET)

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• Note that everything below will vary depending on the exact DSP firmware used and different variants have slightly different behaviours.*

The DSP communicates with the application through two shared memory areas 0x8000 bytes long each (at 0x1FF50000 and 0x1FF70000 respectively). The DSP alternates between the use of these two areas (much like a double-buffer). Each area has 15 structures within it. The location of these structures can be obtained by reading channel 2 of the DSP pipe. A list of structures in the order the DSP addresses are read from the pipe follows:

1. Frame count

2. Input configurations

3. Input status

4. Input ADPCM coefficients

5. DSP configuration

6. DSP status

7. Output samples

8. Intermediate mix samples

9. Compressor table

10. DSP debug statistics

11. Unknown Coefficients

12. Unknown Coefficients

13. Unknown Coefficients

14. Surround sound biquad filter 1

15. Surround sound biquad filter 2

The DSP has 24 inputs, each of which are individually configurable. These 24 inputs each produce three sets of 4 audio channels (two left, two right).

These four audio channels feed into three intermediate mixers. Two of these intermediate mixers are used for effects and aux.

Frame Count

The frame count of the first region must be even and the frame count of the second region must be odd.

The frame with the higher count is the "current region".

The DSP firmware only responds if the first frame count is 4.

Input Config

A 192 byte long structure. There are 24 of them.

Offset	Type	Description
0	u32	Dirty flags
4	f32[3][2][2]	Input Gain (Each input has 12 channels)
52	f32	Rate multiplier (1.0x == native DSP rate)
56	u8	Interpolation mode
57	u8	Polyphase filter select
58	u16	bit[0]: Simple Filter enabled, bit[1]: Biquadratic Filter enabled
60	SimpleFilter	Simple Filter (One pole normalized recursive linear filter)
64	BiquadFilter	Biquadratic Filter (Two poles two zeros normalized recursive linear filter)
74	u16	Bitmap of which buffers in queue are dirty
76	Buffer[4]	Buffer queue
156	u32	-
160	u16	Is Active
162	u16	Sync Count
164	u32	Play position
168	4	?
172	u32	Physical address of embedded buffer
176	u32	Number of samples in embedded buffer
180	u16	Format
182	AdpcmData	ADPCM data associated with embedded buffer
188	u16	bit[0]: ADPCM updated?; bit[1]: Is looping?r
190	u16	Buffer Id of embedded buffer

There is a 2 sample delay in this preprocessing stage, likely due to the interpolation step.

Format

This is a u16.

Bits	Desciption
0-1	Number of channels: 0,1,3 = mono; 2 = stereo
2-3	Buffer codec: 0:PCM8; 1:PCM16; 2:ADPCM
5	Fade

Buffer

Offset	Type	Description
0	u32	Physical Address
4	u32	Sample Count
8	AdpcmData	ADPCM data
14	u8	ADPCM data dirty?
15	u8	Looping?
16	u16	Buffer Id
18	u16	-

Adpcm Data

0	u8	ADPCM predictor/scale
1	u8	-
2	s16	ADPCM y[n-1]
4	s16	ADPCM y[n-2]

Simple Filter

This is a standard single-pole filter. The fall-off is 6dB per octave as you would expect.

Offset	Type	Description
0	s1.15	b0
2	s1.15	a1 (negated)

Biquad Filter

This is a biquadratic filter.

Offset	Type	Description
0	s2.14	a2 (negated)
2	s2.14	a1 (negated)
4	s2.14	b2
6	s2.14	b1
8	s2.14	b0

Input status

Read only. This structure is set by the DSP. This structure is 12 bytes long and there are 24 of them.

Offset	Type	Description
0	u8	Input Enabled?
1	u8	Dirty flag for buffer id, set to 1 when buffer (after the first) starts playing
2	u16	Sync count
4	u32	Position (number of samples) into current buffer playback
8	u16	Buffer id of the buffer that's just started playing.
10	u16	-

Input ADPCM coefficients

This is a 32 byte long structure. There are 24 of them.

Offset	Type	Description
0	s5.11[16]	ADPCM coefficents

DSP configuration

Delay Effect

Delay with feedback.

Length of delay is expressed in terms of number of audio frames (there are 160 samples per audio frame).

Feedback arm only has a gain on it. Under the feedback arm is a single-pole filter with the delay.

Reverb Effect

Reverb consists of two comb filters and one all-pass filter in standard configuration.

DSP status

Read only.

Output samples

Read only. This structure is 640 bytes long.

Offset	Type	Description
0	s16[160][2]	Samples (Stereo-output, 160 samples per channel)

Intermediate mix samples

Read/Write.

PCM32. Also serves an aux function, allowing the ARM11 to apply custom effects to audio.

This structure is 5120 bytes long.

Offset	Type	Description
0	s32[2][4][160]	Samples

Compressor table

A precomputed response curve lookup table for the compressor.