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dac_additive: Decouple the buffer length from the waveform length (#22276)

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* dac_additive: Decouple the buffer length from the waveform length

* Formatting changes for the previous commit

* Reformat waveform tables with rows of 16 entries, ending at column 116

* Revert "Reformat waveform tables with rows of 16 entries, ending at column 116"

This reverts commit 6f2d37908d.
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Nebuleon 2023-12-12 14:06:56 -05:00 committed by GitHub
parent 02c5afc7d5
commit 229a1690a7
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3 changed files with 82 additions and 31 deletions
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31
docs/audio_driver.md
View file

@ -116,19 +116,32 @@ Additionally, in the board config, you'll want to make changes to enable the DAC
| Define | Defaults | Description |
| -------------------------------- | -------------------------- | --------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
| `AUDIO_DAC_SAMPLE_MAX` | `4095U` | Highest value allowed. Lower value means lower volume. And 4095U is the upper limit, since this is limited to a 12 bit value. Only effects non-pregenerated samples. |
| `AUDIO_DAC_OFF_VALUE` | `AUDIO_DAC_SAMPLE_MAX / 2` | The value of the DAC when notplaying anything. Some setups may require a high (`AUDIO_DAC_SAMPLE_MAX`) or low (`0`) value here. |
| `AUDIO_DAC_OFF_VALUE` | `AUDIO_DAC_SAMPLE_MAX / 2` | The value of the DAC when not playing anything. Some setups may require a high (`AUDIO_DAC_SAMPLE_MAX`) or low (`0`) value here. |
| `AUDIO_MAX_SIMULTANEOUS_TONES` | __see next table__ | The number of tones that can be played simultaneously. A value that is too high may freeze the controller or glitch out when too many tones are being played. |
| `AUDIO_DAC_SAMPLE_RATE` | __see next table__ | Effective bit rate of the DAC (in hertz), higher limits simultaneous tones, and lower sacrifices quality. |
| `AUDIO_DAC_BUFFER_SIZE` | __see next table__ | Number of samples generated every refill. Too few may cause excessive CPU load; too many may cause freezes, RAM or flash exhaustion or lags during matrix scanning. |
There are a number of predefined quality settings that you can use, with "sane minimum" being the default. You can use custom values by simply defining the sample rate and number of simultaneous tones, instead of using one of the listed presets.
There are a number of predefined quality settings that you can use, with "sane minimum" being the default. You can use custom values by simply defining the sample rate, number of simultaneous tones and buffer size, instead of using one of the listed presets.
| Define | Sample Rate | Simultaneous tones |
| --------------------------------- | ----------- | ------------------- |
| `AUDIO_DAC_QUALITY_VERY_LOW` | `11025U` | `8` |
| `AUDIO_DAC_QUALITY_LOW` | `22040U` | `4` |
| `AUDIO_DAC_QUALITY_HIGH` | `44100U` | `2` |
| `AUDIO_DAC_QUALITY_VERY_HIGH` | `88200U` | `1` |
| `AUDIO_DAC_QUALITY_SANE_MINIMUM` | `16384U` | `8` |
| Define | Sample Rate | Simultaneous tones | Buffer size |
| --------------------------------- | ----------- | ------------------- | ----------- |
| `AUDIO_DAC_QUALITY_VERY_LOW` | `11025U` | `8` | `64U` |
| `AUDIO_DAC_QUALITY_LOW` | `22050U` | `4` | `128U` |
| `AUDIO_DAC_QUALITY_HIGH` | `44100U` | `2` | `256U` |
| `AUDIO_DAC_QUALITY_VERY_HIGH` | `88200U` | `1` | `256U` |
| `AUDIO_DAC_QUALITY_SANE_MINIMUM` | `16384U` | `8` | `64U` |
#### Notes on buffer size :id=buffer-size
By default, the buffer size attempts to keep to these constraints:
* The interval between buffer refills can't be too short, since the microcontroller would then only be servicing buffer refills and would freeze up.
* On the additive driver, the interval between buffer refills can't be too long, since matrix scanning would suffer lengthy pauses every so often, which would delay key presses or releases or lose some short taps altogether.
* The interval between buffer refills is kept to a minimum, which allows notes to stop as soon as possible after they should.
* For greater compatibility, the buffer size should be a power of 2.
* The buffer size being too large causes resource exhaustion leading to build failures or freezing at runtime: RAM usage (on the additive driver) or flash usage (on the basic driver).
You can lower the buffer size if you need a bit more space in your firmware, or raise it if your keyboard freezes up.
```c

34
platforms/chibios/drivers/audio_dac.h
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@ -23,11 +23,6 @@
# define A5 PAL_LINE(GPIOA, 5)
#endif
/**
* Size of the dac_buffer arrays. All must be the same size.
*/
#define AUDIO_DAC_BUFFER_SIZE 256U
/**
* Highest value allowed sample value.
@ -96,6 +91,35 @@
# define AUDIO_DAC_SAMPLE_RATE 44100U
#endif
/**
* Size of the dac_buffer array. This controls the length of the runtime buffer
* which accumulates the data to be sent to the DAC every few milliseconds, and
* it does not need to correspond to the length of the wavetable for the chosen
* waveform defined by AUDIO_DAC_SAMPLE_WAVEFORM_* in the additive DAC driver.
* By default, this is set to be as close to 3.3 ms as possible, giving 300 DAC
* interrupts per second. Any smaller and the interrupt load gets too heavy and
* this results in crackling due to buffer underrun in the additive DAC driver;
* too large and the RAM (additive driver) or flash (basic driver) usage may be
* too high, causing build failures, and matrix scanning is liable to have long
* periodic pauses that delay key presses or releases or fully lose short taps.
* Large buffers also cause notes to take longer to stop after they should from
* music mode or MIDI input.
* This should be a power of 2 for maximum compatibility.
*/
#ifndef AUDIO_DAC_BUFFER_SIZE
# if AUDIO_DAC_SAMPLE_RATE < 5100U
# define AUDIO_DAC_BUFFER_SIZE 16U
# elif AUDIO_DAC_SAMPLE_RATE < 9900U
# define AUDIO_DAC_BUFFER_SIZE 32U
# elif AUDIO_DAC_SAMPLE_RATE < 19500U
# define AUDIO_DAC_BUFFER_SIZE 64U
# elif AUDIO_DAC_SAMPLE_RATE < 38700U
# define AUDIO_DAC_BUFFER_SIZE 128U
# else
# define AUDIO_DAC_BUFFER_SIZE 256U
# endif
#endif
/**
* The number of tones that can be played simultaneously. If too high a value
* is used here, the keyboard will freeze and glitch-out when that many tones

48
platforms/chibios/drivers/audio_dac_additive.c
View file

@ -53,35 +53,39 @@
#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_SINE
/* one full sine wave over [0,2*pi], but shifted up one amplitude and left pi/4; for the samples to start at 0
*/
static const dacsample_t dac_buffer_sine[AUDIO_DAC_BUFFER_SIZE] = {
static const dacsample_t dac_buffer_sine[] = {
// 256 values, max 4095
0x0, 0x1, 0x2, 0x6, 0xa, 0xf, 0x16, 0x1e, 0x27, 0x32, 0x3d, 0x4a, 0x58, 0x67, 0x78, 0x89, 0x9c, 0xb0, 0xc5, 0xdb, 0xf2, 0x10a, 0x123, 0x13e, 0x159, 0x175, 0x193, 0x1b1, 0x1d1, 0x1f1, 0x212, 0x235, 0x258, 0x27c, 0x2a0, 0x2c6, 0x2ed, 0x314, 0x33c, 0x365, 0x38e, 0x3b8, 0x3e3, 0x40e, 0x43a, 0x467, 0x494, 0x4c2, 0x4f0, 0x51f, 0x54e, 0x57d, 0x5ad, 0x5dd, 0x60e, 0x63f, 0x670, 0x6a1, 0x6d3, 0x705, 0x737, 0x769, 0x79b, 0x7cd, 0x800, 0x832, 0x864, 0x896, 0x8c8, 0x8fa, 0x92c, 0x95e, 0x98f, 0x9c0, 0x9f1, 0xa22, 0xa52, 0xa82, 0xab1, 0xae0, 0xb0f, 0xb3d, 0xb6b, 0xb98, 0xbc5, 0xbf1, 0xc1c, 0xc47, 0xc71, 0xc9a, 0xcc3, 0xceb, 0xd12, 0xd39, 0xd5f, 0xd83, 0xda7, 0xdca, 0xded, 0xe0e, 0xe2e, 0xe4e, 0xe6c, 0xe8a, 0xea6, 0xec1, 0xedc, 0xef5, 0xf0d, 0xf24, 0xf3a, 0xf4f, 0xf63, 0xf76, 0xf87, 0xf98, 0xfa7, 0xfb5, 0xfc2, 0xfcd, 0xfd8, 0xfe1, 0xfe9, 0xff0, 0xff5, 0xff9, 0xffd, 0xffe,
0xfff, 0xffe, 0xffd, 0xff9, 0xff5, 0xff0, 0xfe9, 0xfe1, 0xfd8, 0xfcd, 0xfc2, 0xfb5, 0xfa7, 0xf98, 0xf87, 0xf76, 0xf63, 0xf4f, 0xf3a, 0xf24, 0xf0d, 0xef5, 0xedc, 0xec1, 0xea6, 0xe8a, 0xe6c, 0xe4e, 0xe2e, 0xe0e, 0xded, 0xdca, 0xda7, 0xd83, 0xd5f, 0xd39, 0xd12, 0xceb, 0xcc3, 0xc9a, 0xc71, 0xc47, 0xc1c, 0xbf1, 0xbc5, 0xb98, 0xb6b, 0xb3d, 0xb0f, 0xae0, 0xab1, 0xa82, 0xa52, 0xa22, 0x9f1, 0x9c0, 0x98f, 0x95e, 0x92c, 0x8fa, 0x8c8, 0x896, 0x864, 0x832, 0x800, 0x7cd, 0x79b, 0x769, 0x737, 0x705, 0x6d3, 0x6a1, 0x670, 0x63f, 0x60e, 0x5dd, 0x5ad, 0x57d, 0x54e, 0x51f, 0x4f0, 0x4c2, 0x494, 0x467, 0x43a, 0x40e, 0x3e3, 0x3b8, 0x38e, 0x365, 0x33c, 0x314, 0x2ed, 0x2c6, 0x2a0, 0x27c, 0x258, 0x235, 0x212, 0x1f1, 0x1d1, 0x1b1, 0x193, 0x175, 0x159, 0x13e, 0x123, 0x10a, 0xf2, 0xdb, 0xc5, 0xb0, 0x9c, 0x89, 0x78, 0x67, 0x58, 0x4a, 0x3d, 0x32, 0x27, 0x1e, 0x16, 0xf, 0xa, 0x6, 0x2, 0x1};
0xfff, 0xffe, 0xffd, 0xff9, 0xff5, 0xff0, 0xfe9, 0xfe1, 0xfd8, 0xfcd, 0xfc2, 0xfb5, 0xfa7, 0xf98, 0xf87, 0xf76, 0xf63, 0xf4f, 0xf3a, 0xf24, 0xf0d, 0xef5, 0xedc, 0xec1, 0xea6, 0xe8a, 0xe6c, 0xe4e, 0xe2e, 0xe0e, 0xded, 0xdca, 0xda7, 0xd83, 0xd5f, 0xd39, 0xd12, 0xceb, 0xcc3, 0xc9a, 0xc71, 0xc47, 0xc1c, 0xbf1, 0xbc5, 0xb98, 0xb6b, 0xb3d, 0xb0f, 0xae0, 0xab1, 0xa82, 0xa52, 0xa22, 0x9f1, 0x9c0, 0x98f, 0x95e, 0x92c, 0x8fa, 0x8c8, 0x896, 0x864, 0x832, 0x800, 0x7cd, 0x79b, 0x769, 0x737, 0x705, 0x6d3, 0x6a1, 0x670, 0x63f, 0x60e, 0x5dd, 0x5ad, 0x57d, 0x54e, 0x51f, 0x4f0, 0x4c2, 0x494, 0x467, 0x43a, 0x40e, 0x3e3, 0x3b8, 0x38e, 0x365, 0x33c, 0x314, 0x2ed, 0x2c6, 0x2a0, 0x27c, 0x258, 0x235, 0x212, 0x1f1, 0x1d1, 0x1b1, 0x193, 0x175, 0x159, 0x13e, 0x123, 0x10a, 0xf2, 0xdb, 0xc5, 0xb0, 0x9c, 0x89, 0x78, 0x67, 0x58, 0x4a, 0x3d, 0x32, 0x27, 0x1e, 0x16, 0xf, 0xa, 0x6, 0x2, 0x1,
};
#endif // AUDIO_DAC_SAMPLE_WAVEFORM_SINE
#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE
static const dacsample_t dac_buffer_triangle[AUDIO_DAC_BUFFER_SIZE] = {
static const dacsample_t dac_buffer_triangle[] = {
// 256 values, max 4095
0x0, 0x20, 0x40, 0x60, 0x80, 0xa0, 0xc0, 0xe0, 0x100, 0x120, 0x140, 0x160, 0x180, 0x1a0, 0x1c0, 0x1e0, 0x200, 0x220, 0x240, 0x260, 0x280, 0x2a0, 0x2c0, 0x2e0, 0x300, 0x320, 0x340, 0x360, 0x380, 0x3a0, 0x3c0, 0x3e0, 0x400, 0x420, 0x440, 0x460, 0x480, 0x4a0, 0x4c0, 0x4e0, 0x500, 0x520, 0x540, 0x560, 0x580, 0x5a0, 0x5c0, 0x5e0, 0x600, 0x620, 0x640, 0x660, 0x680, 0x6a0, 0x6c0, 0x6e0, 0x700, 0x720, 0x740, 0x760, 0x780, 0x7a0, 0x7c0, 0x7e0, 0x800, 0x81f, 0x83f, 0x85f, 0x87f, 0x89f, 0x8bf, 0x8df, 0x8ff, 0x91f, 0x93f, 0x95f, 0x97f, 0x99f, 0x9bf, 0x9df, 0x9ff, 0xa1f, 0xa3f, 0xa5f, 0xa7f, 0xa9f, 0xabf, 0xadf, 0xaff, 0xb1f, 0xb3f, 0xb5f, 0xb7f, 0xb9f, 0xbbf, 0xbdf, 0xbff, 0xc1f, 0xc3f, 0xc5f, 0xc7f, 0xc9f, 0xcbf, 0xcdf, 0xcff, 0xd1f, 0xd3f, 0xd5f, 0xd7f, 0xd9f, 0xdbf, 0xddf, 0xdff, 0xe1f, 0xe3f, 0xe5f, 0xe7f, 0xe9f, 0xebf, 0xedf, 0xeff, 0xf1f, 0xf3f, 0xf5f, 0xf7f, 0xf9f, 0xfbf, 0xfdf,
0xfff, 0xfdf, 0xfbf, 0xf9f, 0xf7f, 0xf5f, 0xf3f, 0xf1f, 0xeff, 0xedf, 0xebf, 0xe9f, 0xe7f, 0xe5f, 0xe3f, 0xe1f, 0xdff, 0xddf, 0xdbf, 0xd9f, 0xd7f, 0xd5f, 0xd3f, 0xd1f, 0xcff, 0xcdf, 0xcbf, 0xc9f, 0xc7f, 0xc5f, 0xc3f, 0xc1f, 0xbff, 0xbdf, 0xbbf, 0xb9f, 0xb7f, 0xb5f, 0xb3f, 0xb1f, 0xaff, 0xadf, 0xabf, 0xa9f, 0xa7f, 0xa5f, 0xa3f, 0xa1f, 0x9ff, 0x9df, 0x9bf, 0x99f, 0x97f, 0x95f, 0x93f, 0x91f, 0x8ff, 0x8df, 0x8bf, 0x89f, 0x87f, 0x85f, 0x83f, 0x81f, 0x800, 0x7e0, 0x7c0, 0x7a0, 0x780, 0x760, 0x740, 0x720, 0x700, 0x6e0, 0x6c0, 0x6a0, 0x680, 0x660, 0x640, 0x620, 0x600, 0x5e0, 0x5c0, 0x5a0, 0x580, 0x560, 0x540, 0x520, 0x500, 0x4e0, 0x4c0, 0x4a0, 0x480, 0x460, 0x440, 0x420, 0x400, 0x3e0, 0x3c0, 0x3a0, 0x380, 0x360, 0x340, 0x320, 0x300, 0x2e0, 0x2c0, 0x2a0, 0x280, 0x260, 0x240, 0x220, 0x200, 0x1e0, 0x1c0, 0x1a0, 0x180, 0x160, 0x140, 0x120, 0x100, 0xe0, 0xc0, 0xa0, 0x80, 0x60, 0x40, 0x20};
0xfff, 0xfdf, 0xfbf, 0xf9f, 0xf7f, 0xf5f, 0xf3f, 0xf1f, 0xeff, 0xedf, 0xebf, 0xe9f, 0xe7f, 0xe5f, 0xe3f, 0xe1f, 0xdff, 0xddf, 0xdbf, 0xd9f, 0xd7f, 0xd5f, 0xd3f, 0xd1f, 0xcff, 0xcdf, 0xcbf, 0xc9f, 0xc7f, 0xc5f, 0xc3f, 0xc1f, 0xbff, 0xbdf, 0xbbf, 0xb9f, 0xb7f, 0xb5f, 0xb3f, 0xb1f, 0xaff, 0xadf, 0xabf, 0xa9f, 0xa7f, 0xa5f, 0xa3f, 0xa1f, 0x9ff, 0x9df, 0x9bf, 0x99f, 0x97f, 0x95f, 0x93f, 0x91f, 0x8ff, 0x8df, 0x8bf, 0x89f, 0x87f, 0x85f, 0x83f, 0x81f, 0x800, 0x7e0, 0x7c0, 0x7a0, 0x780, 0x760, 0x740, 0x720, 0x700, 0x6e0, 0x6c0, 0x6a0, 0x680, 0x660, 0x640, 0x620, 0x600, 0x5e0, 0x5c0, 0x5a0, 0x580, 0x560, 0x540, 0x520, 0x500, 0x4e0, 0x4c0, 0x4a0, 0x480, 0x460, 0x440, 0x420, 0x400, 0x3e0, 0x3c0, 0x3a0, 0x380, 0x360, 0x340, 0x320, 0x300, 0x2e0, 0x2c0, 0x2a0, 0x280, 0x260, 0x240, 0x220, 0x200, 0x1e0, 0x1c0, 0x1a0, 0x180, 0x160, 0x140, 0x120, 0x100, 0xe0, 0xc0, 0xa0, 0x80, 0x60, 0x40, 0x20,
};
#endif // AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE
#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE
static const dacsample_t dac_buffer_square[AUDIO_DAC_BUFFER_SIZE] = {
[0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1] = AUDIO_DAC_OFF_VALUE, // first and
[AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = AUDIO_DAC_SAMPLE_MAX, // second half
static const dacsample_t dac_buffer_square[] = {
AUDIO_DAC_OFF_VALUE, // first and
AUDIO_DAC_SAMPLE_MAX, // second steps
};
#endif // AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE
/*
// four steps: 0, 1/3, 2/3 and 1
static const dacsample_t dac_buffer_staircase[AUDIO_DAC_BUFFER_SIZE] = {
[0 ... AUDIO_DAC_BUFFER_SIZE/3 -1 ] = 0,
[AUDIO_DAC_BUFFER_SIZE / 4 ... AUDIO_DAC_BUFFER_SIZE / 2 -1 ] = AUDIO_DAC_SAMPLE_MAX / 3,
[AUDIO_DAC_BUFFER_SIZE / 2 ... 3 * AUDIO_DAC_BUFFER_SIZE / 4 -1 ] = 2 * AUDIO_DAC_SAMPLE_MAX / 3,
[3 * AUDIO_DAC_BUFFER_SIZE / 4 ... AUDIO_DAC_BUFFER_SIZE -1 ] = AUDIO_DAC_SAMPLE_MAX,
static const dacsample_t dac_buffer_staircase[] = {
0,
AUDIO_DAC_SAMPLE_MAX / 3,
2 * AUDIO_DAC_SAMPLE_MAX / 3,
AUDIO_DAC_SAMPLE_MAX,
}
*/
#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID
static const dacsample_t dac_buffer_trapezoid[AUDIO_DAC_BUFFER_SIZE] = {0x0, 0x1f, 0x7f, 0xdf, 0x13f, 0x19f, 0x1ff, 0x25f, 0x2bf, 0x31f, 0x37f, 0x3df, 0x43f, 0x49f, 0x4ff, 0x55f, 0x5bf, 0x61f, 0x67f, 0x6df, 0x73f, 0x79f, 0x7ff, 0x85f, 0x8bf, 0x91f, 0x97f, 0x9df, 0xa3f, 0xa9f, 0xaff, 0xb5f, 0xbbf, 0xc1f, 0xc7f, 0xcdf, 0xd3f, 0xd9f, 0xdff, 0xe5f, 0xebf, 0xf1f, 0xf7f, 0xfdf, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff,
0xfff, 0xfdf, 0xf7f, 0xf1f, 0xebf, 0xe5f, 0xdff, 0xd9f, 0xd3f, 0xcdf, 0xc7f, 0xc1f, 0xbbf, 0xb5f, 0xaff, 0xa9f, 0xa3f, 0x9df, 0x97f, 0x91f, 0x8bf, 0x85f, 0x7ff, 0x79f, 0x73f, 0x6df, 0x67f, 0x61f, 0x5bf, 0x55f, 0x4ff, 0x49f, 0x43f, 0x3df, 0x37f, 0x31f, 0x2bf, 0x25f, 0x1ff, 0x19f, 0x13f, 0xdf, 0x7f, 0x1f, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0};
static const dacsample_t dac_buffer_trapezoid[] = {
0x0, 0x1f, 0x7f, 0xdf, 0x13f, 0x19f, 0x1ff, 0x25f, 0x2bf, 0x31f, 0x37f, 0x3df, 0x43f, 0x49f, 0x4ff, 0x55f, 0x5bf, 0x61f, 0x67f, 0x6df, 0x73f, 0x79f, 0x7ff, 0x85f, 0x8bf, 0x91f, 0x97f, 0x9df, 0xa3f, 0xa9f, 0xaff, 0xb5f, 0xbbf, 0xc1f, 0xc7f, 0xcdf, 0xd3f, 0xd9f, 0xdff, 0xe5f, 0xebf, 0xf1f, 0xf7f, 0xfdf, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff,
0xfff, 0xfdf, 0xf7f, 0xf1f, 0xebf, 0xe5f, 0xdff, 0xd9f, 0xd3f, 0xcdf, 0xc7f, 0xc1f, 0xbbf, 0xb5f, 0xaff, 0xa9f, 0xa3f, 0x9df, 0x97f, 0x91f, 0x8bf, 0x85f, 0x7ff, 0x79f, 0x73f, 0x6df, 0x67f, 0x61f, 0x5bf, 0x55f, 0x4ff, 0x49f, 0x43f, 0x3df, 0x37f, 0x31f, 0x2bf, 0x25f, 0x1ff, 0x19f, 0x13f, 0xdf, 0x7f, 0x1f, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
};
#endif // AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID
static dacsample_t dac_buffer[AUDIO_DAC_BUFFER_SIZE];
@ -124,20 +128,30 @@ __attribute__((weak)) uint16_t dac_value_generate(void) {
uint_fast16_t value = 0;
float frequency = 0.0f;
#if defined(AUDIO_DAC_SAMPLE_WAVEFORM_SINE)
const size_t wavetable_length = ARRAY_SIZE(dac_buffer_sine);
#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE)
const size_t wavetable_length = ARRAY_SIZE(dac_buffer_triangle);
#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID)
const size_t wavetable_length = ARRAY_SIZE(dac_buffer_trapezoid);
#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE)
const size_t wavetable_length = ARRAY_SIZE(dac_buffer_square);
#endif
for (size_t i = 0; i < active_tones_snapshot_length; i++) {
/* Note: a user implementation does not have to rely on the active_tones_snapshot, but
* could directly query the active frequencies through audio_get_processed_frequency */
frequency = active_tones_snapshot[i];
float new_dac_if = dac_if[i];
new_dac_if += frequency * ((float)AUDIO_DAC_BUFFER_SIZE / AUDIO_DAC_SAMPLE_RATE * 2.0f / 3.0f);
new_dac_if += frequency * ((float)wavetable_length / AUDIO_DAC_SAMPLE_RATE * 2.0f / 3.0f);
/*Note: the 2/3 are necessary to get the correct frequencies on the
* DAC output (as measured with an oscilloscope), since the gpt
* timer runs with 3*AUDIO_DAC_SAMPLE_RATE; and the DAC callback
* is called twice per conversion.*/
while (new_dac_if >= AUDIO_DAC_BUFFER_SIZE)
new_dac_if -= AUDIO_DAC_BUFFER_SIZE;
while (new_dac_if >= wavetable_length)
new_dac_if -= wavetable_length;
dac_if[i] = new_dac_if;
// Wavetable generation/lookup