Merge pull request #9776 from relic-se/audiofilters_distortion

audiofilters: Add Distortion effect and implement LFO ticking

Author: gamblor21

ports/unix/variants/coverage/mpconfigvariant.mk

@@ -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 \
@@ -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 \

py/circuitpy_defns.mk

@@ -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 \

shared-bindings/audiofilters/Distortion.c

@@ -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);

shared-bindings/audiofilters/Distortion.h

@@ -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} },

shared-bindings/audiofilters/Filter.c

@@ -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
@@ -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);

shared-bindings/audiofilters/__init__.c

@@ -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;

shared-bindings/synthio/__init__.c

@@ -7,6 +7,7 @@
 
 #include <stdint.h>
 #include "py/runtime.h"
+#include <math.h>
 
 void common_hal_audiodelays_echo_construct(audiodelays_echo_obj_t *self, uint32_t max_delay_ms,
     mp_obj_t delay_ms, mp_obj_t decay, mp_obj_t mix,
@@ -57,17 +58,17 @@ 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);
 
@@ -77,14 +78,17 @@ 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);
         m_malloc_fail(self->max_echo_buffer_len);
     }
     memset(self->echo_buffer, 0, self->max_echo_buffer_len);
 
+    // calculate the length of a single sample in milliseconds
+    self->sample_ms = MICROPY_FLOAT_CONST(1000.0) / self->sample_rate;
+
     // calculate everything needed for the current delay
     mp_float_t f_delay_ms = synthio_block_slot_get(&self->delay_ms);
     recalculate_delay(self, f_delay_ms);
@@ -127,6 +131,9 @@ void common_hal_audiodelays_echo_set_delay_ms(audiodelays_echo_obj_t *self, mp_o
 }
 
 void recalculate_delay(audiodelays_echo_obj_t *self, mp_float_t f_delay_ms) {
+    // Require that delay is at least 1 sample long
+    f_delay_ms = MAX(f_delay_ms, self->sample_ms);
+
     if (self->freq_shift) {
         // Calculate the rate of iteration over the echo buffer with 8 sub-bits
         self->echo_buffer_rate = (uint32_t)MAX(self->max_delay_ms / f_delay_ms * MICROPY_FLOAT_CONST(256.0), MICROPY_FLOAT_CONST(1.0));
@@ -153,7 +160,7 @@ void recalculate_delay(audiodelays_echo_obj_t *self, mp_float_t f_delay_ms) {
         memset(self->echo_buffer + self->echo_buffer_len, 0, self->max_echo_buffer_len - self->echo_buffer_len);
     }
 
-    self->current_delay_ms = (uint32_t)f_delay_ms;
+    self->current_delay_ms = f_delay_ms;
 }
 
 mp_obj_t common_hal_audiodelays_echo_get_decay(audiodelays_echo_obj_t *self) {
@@ -251,48 +258,13 @@ void common_hal_audiodelays_echo_stop(audiodelays_echo_obj_t *self) {
     return;
 }
 
-#define RANGE_LOW_16 (-28000)
-#define RANGE_HIGH_16 (28000)
-#define RANGE_SHIFT_16 (16)
-#define RANGE_SCALE_16 (0xfffffff / (32768 * 2 - RANGE_HIGH_16)) // 2 for echo+sample
-
-// dynamic range compression via a downward compressor with hard knee
-//
-// When the output value is within the range +-28000 (about 85% of full scale),
-// it is unchanged. Otherwise, it undergoes a gain reduction so that the
-// largest possible values, (+32768,-32767) * 2 (2 for echo and sample),
-// still fit within the output range
-//
-// This produces a much louder overall volume with multiple voices, without
-// much additional processing.
-//
-// https://en.wikipedia.org/wiki/Dynamic_range_compression
-static
-int16_t mix_down_sample(int32_t sample) {
-    if (sample < RANGE_LOW_16) {
-        sample = (((sample - RANGE_LOW_16) * RANGE_SCALE_16) >> RANGE_SHIFT_16) + RANGE_LOW_16;
-    } else if (sample > RANGE_HIGH_16) {
-        sample = (((sample - RANGE_HIGH_16) * RANGE_SCALE_16) >> RANGE_SHIFT_16) + RANGE_HIGH_16;
-    }
-    return sample;
-}
-
 audioio_get_buffer_result_t audiodelays_echo_get_buffer(audiodelays_echo_obj_t *self, bool single_channel_output, uint8_t channel,
     uint8_t **buffer, uint32_t *buffer_length) {
 
     if (!single_channel_output) {
         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;
 
@@ -303,16 +275,6 @@ 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) {
@@ -334,9 +296,38 @@ 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));
+
+        mp_float_t f_delay_ms = synthio_block_slot_get(&self->delay_ms);
+        if (MICROPY_FLOAT_C_FUN(fabs)(self->current_delay_ms - f_delay_ms) >= self->sample_ms) {
+            recalculate_delay(self, f_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 {
@@ -400,13 +391,10 @@ 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];
@@ -440,7 +428,7 @@ audioio_get_buffer_result_t audiodelays_echo_get_buffer(audiodelays_echo_obj_t *
                     }
 
                     if (MP_LIKELY(self->bits_per_sample == 16)) {
-                        word = mix_down_sample(word);
+                        word = synthio_mix_down_sample(word, SYNTHIO_MIX_DOWN_SCALE(2));
                         if (self->freq_shift) {
                             for (uint32_t j = echo_buffer_pos >> 8; j < next_buffer_pos >> 8; j++) {
                                 echo_buffer[j % echo_buf_len] = (int16_t)word;
@@ -465,14 +453,15 @@ audioio_get_buffer_result_t audiodelays_echo_get_buffer(audiodelays_echo_obj_t *
                     }
 
                     word = echo + sample_word;
+                    word = synthio_mix_down_sample(word, SYNTHIO_MIX_DOWN_SCALE(2));
 
                     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 {
@@ -500,13 +489,13 @@ 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;
+            }
         }
     }
 

shared-module/audiodelays/Echo.c

@@ -8,6 +8,7 @@
 #include "py/obj.h"
 
 #include "shared-module/audiocore/__init__.h"
+#include "shared-module/synthio/__init__.h"
 #include "shared-module/synthio/block.h"
 
 extern const mp_obj_type_t audiodelays_echo_type;
@@ -16,7 +17,8 @@ typedef struct {
     mp_obj_base_t base;
     uint32_t max_delay_ms;
     synthio_block_slot_t delay_ms;
-    uint32_t current_delay_ms;
+    mp_float_t current_delay_ms;
+    mp_float_t sample_ms;
     synthio_block_slot_t decay;
     synthio_block_slot_t mix;