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audiofilters: Add Distortion effect and implement LFO ticking #9776

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audiofilters: Add Distortion effect and implement LFO ticking #9776

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16575fc
Added `Distortion` and `DistortionMode` classes to `audiofilters` mod…
relic-se Oct 23, 2024
46ebae1
Remove separate DistortionMode code files.
relic-se Oct 31, 2024
3a16daf
Remove `audiofilters.DistortionMode.ATAN`
relic-se Oct 31, 2024
1008dd5
Simplify `audiofilters.DistortionMode.LOFI` sample processing with bi…
relic-se Oct 31, 2024
37b6b70
Fix error with null sample handling in `audiofilters.Distortion`.
relic-se Nov 4, 2024
064c3f3
Merge branch 'adafruit:main' into audiofilters_distortion
relic-se Nov 13, 2024
a7060f0
Merge branch 'adafruit:main' into audiofilters_distortion
relic-se Nov 27, 2024
6481b4e
Merge branch 'adafruit:main' into audiofilters_distortion
relic-se Dec 10, 2024
31c9095
Implement `synthio_block_slot_get_limited`.
relic-se Dec 10, 2024
155f197
Convert default float values from null checks to MP_ROM_INT.
relic-se Dec 10, 2024
89f2ae1
Remove unnecessary kwarg setters.
relic-se Dec 10, 2024
5c981f0
Use `MICROPY_FLOAT_CONST` and `MICROPY_FLOAT_C_FUN` within floating p…
relic-se Dec 10, 2024
222ce2c
Apply similar updates to audiofilters.Filter and audiodelays.Echo: MI…
relic-se Dec 10, 2024
0410d22
Added `soft_clip` property to toggle between hard clipping (default) …
relic-se Dec 11, 2024
57022f9
Implemented soft clipping and continued optimization of distortion al…
relic-se Dec 11, 2024
48ca21d
Add Distortion to unix port and make type conversions explicit.
relic-se Dec 11, 2024
4257c62
Variable number of samples within `shared_bindings_synthio_lfo_tick`.
relic-se Dec 12, 2024
0e64e1c
Implement block ticking within audio effects.
relic-se Dec 12, 2024
5fbbeed
Call `shared_bindings_synthio_lfo_tick` on audioeffects in `SYNTHIO_M…
relic-se Dec 12, 2024
e18e5b2
Merge branch 'adafruit:main' into audiofilters_distortion
relic-se Jan 17, 2025
cef94d7
Remove unnecessary deinit check.
relic-se Jan 17, 2025
b796f0d
Only calculate lofi bit mask when necessary.
relic-se Jan 17, 2025
84f8e31
Update `shared_bindings_synthio_lfo_tick` within `audiomixer` to use …
relic-se Jan 17, 2025
d84cdbc
Update Distortion class docstring.
relic-se Jan 17, 2025
8c40c56
Move `shared_bindings_synthio_lfo_tick` to avoid error if using unsig…
relic-se Jan 17, 2025
05db339
Remove truncation of `delay_ms` within buffer processing loop.
relic-se Jan 17, 2025
ec5b1e8
Add `mix_down_sample` to echo output.
relic-se Jan 17, 2025
48f272e
Fix build errors.
relic-se Jan 17, 2025
99b4fae
Remove unnecessary copies of `mix_down_sample`.
relic-se Jan 17, 2025
e049337
Allow variable mix down sample scale.
relic-se Jan 18, 2025
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2 changes: 2 additions & 0 deletions ports/unix/variants/coverage/mpconfigvariant.mk
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Original file line number Diff line number Diff line change
Expand Up @@ -35,6 +35,7 @@ SRC_BITMAP := \
shared-bindings/audiocore/WaveFile.c \
shared-bindings/audiodelays/Echo.c \
shared-bindings/audiodelays/__init__.c \
shared-bindings/audiofilters/Distortion.c \
shared-bindings/audiofilters/Filter.c \
shared-bindings/audiofilters/__init__.c \
shared-bindings/audiomixer/__init__.c \
Expand Down Expand Up @@ -77,6 +78,7 @@ SRC_BITMAP := \
shared-module/audiocore/WaveFile.c \
shared-module/audiodelays/Echo.c \
shared-module/audiodelays/__init__.c \
shared-module/audiofilters/Distortion.c \
shared-module/audiofilters/Filter.c \
shared-module/audiofilters/__init__.c \
shared-module/audiomixer/__init__.c \
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1 change: 1 addition & 0 deletions py/circuitpy_defns.mk
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Original file line number Diff line number Diff line change
Expand Up @@ -625,6 +625,7 @@ SRC_SHARED_MODULE_ALL = \
audiocore/__init__.c \
audiodelays/Echo.c \
audiodelays/__init__.c \
audiofilters/Distortion.c \
audiofilters/Filter.c \
audiofilters/__init__.c \
audioio/__init__.c \
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386 changes: 386 additions & 0 deletions shared-bindings/audiofilters/Distortion.c
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47 changes: 47 additions & 0 deletions shared-bindings/audiofilters/Distortion.h
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Original file line number Diff line number Diff line change
@@ -0,0 +1,47 @@
// This file is part of the CircuitPython project: https://circuitpython.org
//
// SPDX-FileCopyrightText: Copyright (c) 2024 Cooper Dalrymple
//
// SPDX-License-Identifier: MIT

#pragma once

#include "shared-module/audiofilters/Distortion.h"

extern const mp_obj_type_t audiofilters_distortion_type;
extern const mp_obj_type_t audiofilters_distortion_mode_type;

void common_hal_audiofilters_distortion_construct(audiofilters_distortion_obj_t *self,
mp_obj_t drive, mp_obj_t pre_gain, mp_obj_t post_gain,
audiofilters_distortion_mode mode, bool soft_clip, mp_obj_t mix,
uint32_t buffer_size, uint8_t bits_per_sample, bool samples_signed,
uint8_t channel_count, uint32_t sample_rate);

void common_hal_audiofilters_distortion_deinit(audiofilters_distortion_obj_t *self);
bool common_hal_audiofilters_distortion_deinited(audiofilters_distortion_obj_t *self);

uint32_t common_hal_audiofilters_distortion_get_sample_rate(audiofilters_distortion_obj_t *self);
uint8_t common_hal_audiofilters_distortion_get_channel_count(audiofilters_distortion_obj_t *self);
uint8_t common_hal_audiofilters_distortion_get_bits_per_sample(audiofilters_distortion_obj_t *self);

mp_obj_t common_hal_audiofilters_distortion_get_drive(audiofilters_distortion_obj_t *self);
void common_hal_audiofilters_distortion_set_drive(audiofilters_distortion_obj_t *self, mp_obj_t arg);

mp_obj_t common_hal_audiofilters_distortion_get_pre_gain(audiofilters_distortion_obj_t *self);
void common_hal_audiofilters_distortion_set_pre_gain(audiofilters_distortion_obj_t *self, mp_obj_t arg);

mp_obj_t common_hal_audiofilters_distortion_get_post_gain(audiofilters_distortion_obj_t *self);
void common_hal_audiofilters_distortion_set_post_gain(audiofilters_distortion_obj_t *self, mp_obj_t arg);

audiofilters_distortion_mode common_hal_audiofilters_distortion_get_mode(audiofilters_distortion_obj_t *self);
void common_hal_audiofilters_distortion_set_mode(audiofilters_distortion_obj_t *self, audiofilters_distortion_mode mode);

bool common_hal_audiofilters_distortion_get_soft_clip(audiofilters_distortion_obj_t *self);
void common_hal_audiofilters_distortion_set_soft_clip(audiofilters_distortion_obj_t *self, bool soft_clip);

mp_obj_t common_hal_audiofilters_distortion_get_mix(audiofilters_distortion_obj_t *self);
void common_hal_audiofilters_distortion_set_mix(audiofilters_distortion_obj_t *self, mp_obj_t arg);

bool common_hal_audiofilters_distortion_get_playing(audiofilters_distortion_obj_t *self);
void common_hal_audiofilters_distortion_play(audiofilters_distortion_obj_t *self, mp_obj_t sample, bool loop);
void common_hal_audiofilters_distortion_stop(audiofilters_distortion_obj_t *self);
2 changes: 1 addition & 1 deletion shared-bindings/audiofilters/Filter.c
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Original file line number Diff line number Diff line change
Expand Up @@ -72,7 +72,7 @@ static mp_obj_t audiofilters_filter_make_new(const mp_obj_type_t *type, size_t n
enum { ARG_filter, ARG_mix, ARG_buffer_size, ARG_sample_rate, ARG_bits_per_sample, ARG_samples_signed, ARG_channel_count, };
static const mp_arg_t allowed_args[] = {
{ MP_QSTR_filter, MP_ARG_OBJ | MP_ARG_KW_ONLY, {.u_obj = MP_OBJ_NULL} },
{ MP_QSTR_mix, MP_ARG_OBJ | MP_ARG_KW_ONLY, {.u_obj = MP_OBJ_NULL} },
{ MP_QSTR_mix, MP_ARG_OBJ | MP_ARG_KW_ONLY, {.u_obj = MP_ROM_INT(1)} },
{ MP_QSTR_buffer_size, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 512} },
{ MP_QSTR_sample_rate, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 8000} },
{ MP_QSTR_bits_per_sample, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 16} },
Expand Down
5 changes: 5 additions & 0 deletions shared-bindings/audiofilters/__init__.c
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Original file line number Diff line number Diff line change
Expand Up @@ -10,6 +10,7 @@
#include "py/runtime.h"

#include "shared-bindings/audiofilters/__init__.h"
#include "shared-bindings/audiofilters/Distortion.h"
#include "shared-bindings/audiofilters/Filter.h"

//| """Support for audio filter effects
Expand All @@ -21,6 +22,10 @@
static const mp_rom_map_elem_t audiofilters_module_globals_table[] = {
{ MP_ROM_QSTR(MP_QSTR___name__), MP_ROM_QSTR(MP_QSTR_audiofilters) },
{ MP_ROM_QSTR(MP_QSTR_Filter), MP_ROM_PTR(&audiofilters_filter_type) },
{ MP_ROM_QSTR(MP_QSTR_Distortion), MP_ROM_PTR(&audiofilters_distortion_type) },

// Enum-like Classes.
{ MP_ROM_QSTR(MP_QSTR_DistortionMode), MP_ROM_PTR(&audiofilters_distortion_mode_type) },
};

static MP_DEFINE_CONST_DICT(audiofilters_module_globals, audiofilters_module_globals_table);
Expand Down
2 changes: 1 addition & 1 deletion shared-bindings/synthio/__init__.c
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Original file line number Diff line number Diff line change
Expand Up @@ -292,7 +292,7 @@ MP_DEFINE_CONST_FUN_OBJ_1(synthio_voct_to_hz_obj, voct_to_hz);

#if CIRCUITPY_AUDIOCORE_DEBUG
static mp_obj_t synthio_lfo_tick(size_t n, const mp_obj_t *args) {
shared_bindings_synthio_lfo_tick(48000);
shared_bindings_synthio_lfo_tick(48000, SYNTHIO_MAX_DUR);
mp_obj_t result[n];
for (size_t i = 0; i < n; i++) {
synthio_block_slot_t slot;
Expand Down
79 changes: 43 additions & 36 deletions shared-module/audiodelays/Echo.c
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Original file line number Diff line number Diff line change
Expand Up @@ -57,17 +57,17 @@ void common_hal_audiodelays_echo_construct(audiodelays_echo_obj_t *self, uint32_

// If we did not receive a BlockInput we need to create a default float value
if (decay == MP_OBJ_NULL) {
decay = mp_obj_new_float(0.7);
decay = mp_obj_new_float(MICROPY_FLOAT_CONST(0.7));
}
synthio_block_assign_slot(decay, &self->decay, MP_QSTR_decay);

if (delay_ms == MP_OBJ_NULL) {
delay_ms = mp_obj_new_float(250.0);
delay_ms = mp_obj_new_float(MICROPY_FLOAT_CONST(250.0));
}
synthio_block_assign_slot(delay_ms, &self->delay_ms, MP_QSTR_delay_ms);

if (mix == MP_OBJ_NULL) {
mix = mp_obj_new_float(0.5);
mix = mp_obj_new_float(MICROPY_FLOAT_CONST(0.5));
}
synthio_block_assign_slot(mix, &self->mix, MP_QSTR_mix);

Expand All @@ -77,7 +77,7 @@ void common_hal_audiodelays_echo_construct(audiodelays_echo_obj_t *self, uint32_

// Allocate the echo buffer for the max possible delay, echo is always 16-bit
self->max_delay_ms = max_delay_ms;
self->max_echo_buffer_len = (uint32_t)(self->sample_rate / 1000.0f * max_delay_ms) * (self->channel_count * sizeof(uint16_t)); // bytes
self->max_echo_buffer_len = (uint32_t)(self->sample_rate / MICROPY_FLOAT_CONST(1000.0) * max_delay_ms) * (self->channel_count * sizeof(uint16_t)); // bytes
self->echo_buffer = m_malloc(self->max_echo_buffer_len);
if (self->echo_buffer == NULL) {
common_hal_audiodelays_echo_deinit(self);
Expand Down Expand Up @@ -284,15 +284,6 @@ audioio_get_buffer_result_t audiodelays_echo_get_buffer(audiodelays_echo_obj_t *
channel = 0;
}

// get the effect values we need from the BlockInput. These may change at run time so you need to do bounds checking if required
mp_float_t mix = MIN(1.0, MAX(synthio_block_slot_get(&self->mix), 0.0));
mp_float_t decay = MIN(1.0, MAX(synthio_block_slot_get(&self->decay), 0.0));

uint32_t delay_ms = (uint32_t)synthio_block_slot_get(&self->delay_ms);
if (self->current_delay_ms != delay_ms) {
recalculate_delay(self, delay_ms);
}

// Switch our buffers to the other buffer
self->last_buf_idx = !self->last_buf_idx;

Expand All @@ -303,16 +294,6 @@ audioio_get_buffer_result_t audiodelays_echo_get_buffer(audiodelays_echo_obj_t *

// The echo buffer is always stored as a 16-bit value internally
int16_t *echo_buffer = (int16_t *)self->echo_buffer;
uint32_t echo_buf_len = self->echo_buffer_len / sizeof(uint16_t);

// Set our echo buffer position accounting for stereo
uint32_t echo_buffer_pos = 0;
if (self->freq_shift) {
echo_buffer_pos = self->echo_buffer_left_pos;
if (channel == 1) {
echo_buffer_pos = self->echo_buffer_right_pos;
}
}

// Loop over the entire length of our buffer to fill it, this may require several calls to get data from the sample
while (length != 0) {
Expand All @@ -334,9 +315,38 @@ audioio_get_buffer_result_t audiodelays_echo_get_buffer(audiodelays_echo_obj_t *
}
}

// Determine how many bytes we can process to our buffer, the less of the sample we have left and our buffer remaining
uint32_t n;
if (self->sample == NULL) {
n = MIN(length, SYNTHIO_MAX_DUR * self->channel_count);
} else {
n = MIN(MIN(self->sample_buffer_length, length), SYNTHIO_MAX_DUR * self->channel_count);
}

// get the effect values we need from the BlockInput. These may change at run time so you need to do bounds checking if required
shared_bindings_synthio_lfo_tick(self->sample_rate, n / self->channel_count);
mp_float_t mix = synthio_block_slot_get_limited(&self->mix, MICROPY_FLOAT_CONST(0.0), MICROPY_FLOAT_CONST(1.0));
mp_float_t decay = synthio_block_slot_get_limited(&self->decay, MICROPY_FLOAT_CONST(0.0), MICROPY_FLOAT_CONST(1.0));

uint32_t delay_ms = (uint32_t)synthio_block_slot_get(&self->delay_ms);
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We may want to check that delay >= 0. Never did in the original code (my bad) but a LFO could go negative and weird things happen.

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Casting it to an unsigned integer does force it to be >= 0, but a negative result might cause it to wrap around. If it's 0, I believe it will do a minimum of 1 sample of delay. I'll look into it.

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maybe it should be synthio_block_slot_get_limited(1, some_calculated_maximum) then?

Technically under the C99 standard, conversion from a negative floating point value to an unsigned integer value is undefined behavior.

When a finite value of real floating type is converted to an integer type other than _Bool,
the fractional part is discarded (i.e., the value is truncated toward zero). If the value of
the integral part cannot be represented by the integer type, the behavior is undefined. (6.3.1.4.1)

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I've taken a different approach here. Storing this value and current_delay_ms as uint32_t was an arbitrary choice, so I've instead reverted both back to floats. I've added in another pre-calculated variable for the length of one audio frame (sample) in milliseconds, and I'm checking against that determine if we need to recalculate delay buffer parameters. The minimum value of the delay is handled within recalculate_delay. Before, delays could only be manipulated by an LFO in millisecond intervals because of this, whereas now they can be controlled with sample-level precision by an LFO. This could be useful for chorus/phaser type effects.

As for some_calculated_maximum, it's difficult to sum that up because that maximum value depends on the freq_shift property. If using the standard delay, it is limited to max_effect_ms, but with freq_shift, I believe it is max_effect_ms*256 because of the 8 sub-bits in the buffer pointer. You lose a lot of audio quality when going that long, though.

I'm tempted to move this logic into recalculate_delay if it needs more refinement.

if (self->current_delay_ms != delay_ms) {
recalculate_delay(self, delay_ms);
}

uint32_t echo_buf_len = self->echo_buffer_len / sizeof(uint16_t);

// Set our echo buffer position accounting for stereo
uint32_t echo_buffer_pos = 0;
if (self->freq_shift) {
echo_buffer_pos = self->echo_buffer_left_pos;
if (channel == 1) {
echo_buffer_pos = self->echo_buffer_right_pos;
}
}

// If we have no sample keep the echo echoing
if (self->sample == NULL) {
if (mix <= 0.01) { // Mix of 0 is pure sample sound. We have no sample so no sound
if (mix <= MICROPY_FLOAT_CONST(0.01)) { // Mix of 0 is pure sample sound. We have no sample so no sound
if (self->samples_signed) {
memset(word_buffer, 0, length * (self->bits_per_sample / 8));
} else {
Expand Down Expand Up @@ -400,13 +410,10 @@ audioio_get_buffer_result_t audiodelays_echo_get_buffer(audiodelays_echo_obj_t *
length = 0;
} else {
// we have a sample to play and echo
// Determine how many bytes we can process to our buffer, the less of the sample we have left and our buffer remaining
uint32_t n = MIN(self->sample_buffer_length, length);

int16_t *sample_src = (int16_t *)self->sample_remaining_buffer; // for 16-bit samples
int8_t *sample_hsrc = (int8_t *)self->sample_remaining_buffer; // for 8-bit samples

if (mix <= 0.01) { // if mix is zero pure sample only
if (mix <= MICROPY_FLOAT_CONST(0.01)) { // if mix is zero pure sample only
for (uint32_t i = 0; i < n; i++) {
if (MP_LIKELY(self->bits_per_sample == 16)) {
word_buffer[i] = sample_src[i];
Expand Down Expand Up @@ -467,12 +474,12 @@ audioio_get_buffer_result_t audiodelays_echo_get_buffer(audiodelays_echo_obj_t *
word = echo + sample_word;
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if (MP_LIKELY(self->bits_per_sample == 16)) {
word_buffer[i] = (int16_t)((sample_word * (1.0 - mix)) + (word * mix));
word_buffer[i] = (int16_t)((sample_word * (MICROPY_FLOAT_CONST(1.0) - mix)) + (word * mix));
if (!self->samples_signed) {
word_buffer[i] ^= 0x8000;
}
} else {
int8_t mixed = (int16_t)((sample_word * (1.0 - mix)) + (word * mix));
int8_t mixed = (int16_t)((sample_word * (MICROPY_FLOAT_CONST(1.0) - mix)) + (word * mix));
if (self->samples_signed) {
hword_buffer[i] = mixed;
} else {
Expand Down Expand Up @@ -500,13 +507,13 @@ audioio_get_buffer_result_t audiodelays_echo_get_buffer(audiodelays_echo_obj_t *
self->sample_remaining_buffer += (n * (self->bits_per_sample / 8));
self->sample_buffer_length -= n;
}
}

if (self->freq_shift) {
if (channel == 0) {
self->echo_buffer_left_pos = echo_buffer_pos;
} else if (channel == 1) {
self->echo_buffer_right_pos = echo_buffer_pos;
if (self->freq_shift) {
if (channel == 0) {
self->echo_buffer_left_pos = echo_buffer_pos;
} else if (channel == 1) {
self->echo_buffer_right_pos = echo_buffer_pos;
}
}
}

Expand Down
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