Move Audio drivers from quantum to platform drivers folder (#14308)

* Move Audio drivers from quantum to platform drivers folder

* fix path for audio drivers

Co-authored-by: Ryan <fauxpark@gmail.com>

Co-authored-by: Ryan <fauxpark@gmail.com>

6 files changed, 1054 insertions, 0 deletions

diff --git a/platforms/chibios/drivers/audio_dac.h b/platforms/chibios/drivers/audio_dac.h
new file mode 100644
index 0000000000..07cd622ead
--- /dev/null
+++ b/platforms/chibios/drivers/audio_dac.h
@@ -0,0 +1,126 @@
+/* Copyright 2019 Jack Humbert
+ * Copyright 2020 JohSchneider
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ */
+#pragma once
+
+#ifndef A4
+#    define A4 PAL_LINE(GPIOA, 4)
+#endif
+#ifndef A5
+#    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.
+
+ * since the DAC is limited to 12 bit, the absolute max is 0xfff = 4095U;
+ * lower values adjust the peak-voltage aka volume down.
+ * adjusting this value has only an effect on a sample-buffer whose values are
+ * are NOT pregenerated - see square-wave
+ */
+#ifndef AUDIO_DAC_SAMPLE_MAX
+#    define AUDIO_DAC_SAMPLE_MAX 4095U
+#endif
+
+#if !defined(AUDIO_DAC_SAMPLE_RATE) && !defined(AUDIO_MAX_SIMULTANEOUS_TONES) && !defined(AUDIO_DAC_QUALITY_VERY_LOW) && !defined(AUDIO_DAC_QUALITY_LOW) && !defined(AUDIO_DAC_QUALITY_HIGH) && !defined(AUDIO_DAC_QUALITY_VERY_HIGH)
+#    define AUDIO_DAC_QUALITY_SANE_MINIMUM
+#endif
+
+/**
+ * These presets allow you to quickly switch between quality settings for
+ * the DAC. The sample rate and maximum number of simultaneous tones roughly
+ * has an inverse relationship - slightly higher sample rates may be possible.
+ *
+ * NOTE: a high sample-rate results in a higher cpu-load, which might lead to
+ *       (audible) discontinuities and/or starve other processes of cpu-time
+ *       (like RGB-led back-lighting, ...)
+ */
+#ifdef AUDIO_DAC_QUALITY_VERY_LOW
+#    define AUDIO_DAC_SAMPLE_RATE 11025U
+#    define AUDIO_MAX_SIMULTANEOUS_TONES 8
+#endif
+
+#ifdef AUDIO_DAC_QUALITY_LOW
+#    define AUDIO_DAC_SAMPLE_RATE 22050U
+#    define AUDIO_MAX_SIMULTANEOUS_TONES 4
+#endif
+
+#ifdef AUDIO_DAC_QUALITY_HIGH
+#    define AUDIO_DAC_SAMPLE_RATE 44100U
+#    define AUDIO_MAX_SIMULTANEOUS_TONES 2
+#endif
+
+#ifdef AUDIO_DAC_QUALITY_VERY_HIGH
+#    define AUDIO_DAC_SAMPLE_RATE 88200U
+#    define AUDIO_MAX_SIMULTANEOUS_TONES 1
+#endif
+
+#ifdef AUDIO_DAC_QUALITY_SANE_MINIMUM
+/* a sane-minimum config: with a trade-off between cpu-load and tone-range
+ *
+ * the (currently) highest defined note is NOTE_B8 with 7902Hz; if we now
+ * aim for an even even multiple of the buffer-size, we end up with:
+ * ( roundUptoPow2(highest note / AUDIO_DAC_BUFFER_SIZE) * nyquist-rate * AUDIO_DAC_BUFFER_SIZE)
+ *                              7902/256 = 30.867        *       2      * 256 ~= 16384
+ * which works out (but the 'scope shows some sampling artifacts with lower harmonics :-P)
+ */
+#    define AUDIO_DAC_SAMPLE_RATE 16384U
+#    define AUDIO_MAX_SIMULTANEOUS_TONES 8
+#endif
+
+/**
+ * Effective bit-rate of the DAC. 44.1khz is the standard for most audio - any
+ * lower will sacrifice perceptible audio quality. Any higher will limit the
+ * number of simultaneous tones. In most situations, a tenth (1/10) of the
+ * sample rate is where notes become unbearable.
+ */
+#ifndef AUDIO_DAC_SAMPLE_RATE
+#    define AUDIO_DAC_SAMPLE_RATE 44100U
+#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
+ * are being played.
+ */
+#ifndef AUDIO_MAX_SIMULTANEOUS_TONES
+#    define AUDIO_MAX_SIMULTANEOUS_TONES 2
+#endif
+
+/**
+ * The default value of the DAC when not playing anything. Certain hardware
+ * setups may require a high (AUDIO_DAC_SAMPLE_MAX) or low (0) value here.
+ * Since multiple added sine waves tend to oscillate around the midpoint,
+ * and possibly never/rarely reach either 0 of MAX, 1/2 MAX can be a
+ * reasonable default value.
+ */
+#ifndef AUDIO_DAC_OFF_VALUE
+#    define AUDIO_DAC_OFF_VALUE AUDIO_DAC_SAMPLE_MAX / 2
+#endif
+
+#if AUDIO_DAC_OFF_VALUE > AUDIO_DAC_SAMPLE_MAX
+#    error "AUDIO_DAC: OFF_VALUE may not be larger than SAMPLE_MAX"
+#endif
+
+/**
+ *user overridable sample generation/processing
+ */
+uint16_t dac_value_generate(void);
diff --git a/platforms/chibios/drivers/audio_dac_additive.c b/platforms/chibios/drivers/audio_dac_additive.c
new file mode 100644
index 0000000000..db304adb87
--- /dev/null
+++ b/platforms/chibios/drivers/audio_dac_additive.c
@@ -0,0 +1,335 @@
+/* Copyright 2016-2019 Jack Humbert
+ * Copyright 2020 JohSchneider
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include "audio.h"
+#include <ch.h>
+#include <hal.h>
+
+/*
+  Audio Driver: DAC
+
+  which utilizes the dac unit many STM32 are equipped with, to output a modulated waveform from samples stored in the dac_buffer_* array who are passed to the hardware through DMA
+
+  it is also possible to have a custom sample-LUT by implementing/overriding 'dac_value_generate'
+
+  this driver allows for multiple simultaneous tones to be played through one single channel by doing additive wave-synthesis
+*/
+
+#if !defined(AUDIO_PIN)
+#    error "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under 'ARM (DAC additive)' for available options."
+#endif
+#if defined(AUDIO_PIN_ALT) && !defined(AUDIO_PIN_ALT_AS_NEGATIVE)
+#    pragma message "Audio feature: AUDIO_PIN_ALT set, but not AUDIO_PIN_ALT_AS_NEGATIVE - pin will be left unused; audio might still work though."
+#endif
+
+#if !defined(AUDIO_PIN_ALT)
+// no ALT pin defined is valid, but the c-ifs below need some value set
+#    define AUDIO_PIN_ALT PAL_NOLINE
+#endif
+
+#if !defined(AUDIO_DAC_SAMPLE_WAVEFORM_SINE) && !defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE) && !defined(AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE) && !defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID)
+#    define AUDIO_DAC_SAMPLE_WAVEFORM_SINE
+#endif
+
+#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] = {
+    // 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};
+#endif  // AUDIO_DAC_SAMPLE_WAVEFORM_SINE
+#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE
+static const dacsample_t dac_buffer_triangle[AUDIO_DAC_BUFFER_SIZE] = {
+    // 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};
+#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]                     = 0,                     // first and
+    [AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = AUDIO_DAC_SAMPLE_MAX,  // second half
+};
+#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,
+}
+*/
+#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};
+#endif  // AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID
+
+static dacsample_t dac_buffer_empty[AUDIO_DAC_BUFFER_SIZE] = {AUDIO_DAC_OFF_VALUE};
+
+/* keep track of the sample position for for each frequency */
+static float dac_if[AUDIO_MAX_SIMULTANEOUS_TONES] = {0.0};
+
+static float   active_tones_snapshot[AUDIO_MAX_SIMULTANEOUS_TONES] = {0, 0};
+static uint8_t active_tones_snapshot_length                        = 0;
+
+typedef enum {
+    OUTPUT_SHOULD_START,
+    OUTPUT_RUN_NORMALLY,
+    // path 1: wait for zero, then change/update active tones
+    OUTPUT_TONES_CHANGED,
+    OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE,
+    // path 2: hardware should stop, wait for zero then turn output off = stop the timer
+    OUTPUT_SHOULD_STOP,
+    OUTPUT_REACHED_ZERO_BEFORE_OFF,
+    OUTPUT_OFF,
+    OUTPUT_OFF_1,
+    OUTPUT_OFF_2,  // trailing off: giving the DAC two more conversion cycles until the AUDIO_DAC_OFF_VALUE reaches the output, then turn the timer off, which leaves the output at that level
+    number_of_output_states
+} output_states_t;
+output_states_t state = OUTPUT_OFF_2;
+
+/**
+ * Generation of the waveform being passed to the callback. Declared weak so users
+ * can override it with their own wave-forms/noises.
+ */
+__attribute__((weak)) uint16_t dac_value_generate(void) {
+    // DAC is running/asking for values but snapshot length is zero -> must be playing a pause
+    if (active_tones_snapshot_length == 0) {
+        return AUDIO_DAC_OFF_VALUE;
+    }
+
+    /* doing additive wave synthesis over all currently playing tones = adding up
+     * sine-wave-samples for each frequency, scaled by the number of active tones
+     */
+    uint16_t value     = 0;
+    float    frequency = 0.0f;
+
+    for (uint8_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];
+
+        dac_if[i] = dac_if[i] + ((frequency * AUDIO_DAC_BUFFER_SIZE) / AUDIO_DAC_SAMPLE_RATE) * 2 / 3;
+        /*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.*/
+
+        dac_if[i] = fmod(dac_if[i], AUDIO_DAC_BUFFER_SIZE);
+
+        // Wavetable generation/lookup
+        uint16_t dac_i = (uint16_t)dac_if[i];
+
+#if defined(AUDIO_DAC_SAMPLE_WAVEFORM_SINE)
+        value += dac_buffer_sine[dac_i] / active_tones_snapshot_length;
+#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE)
+        value += dac_buffer_triangle[dac_i] / active_tones_snapshot_length;
+#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID)
+        value += dac_buffer_trapezoid[dac_i] / active_tones_snapshot_length;
+#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE)
+        value += dac_buffer_square[dac_i] / active_tones_snapshot_length;
+#endif
+        /*
+        // SINE
+        value += dac_buffer_sine[dac_i] / active_tones_snapshot_length / 3;
+        // TRIANGLE
+        value += dac_buffer_triangle[dac_i] / active_tones_snapshot_length / 3;
+        // SQUARE
+        value += dac_buffer_square[dac_i] / active_tones_snapshot_length / 3;
+        //NOTE: combination of these three wave-forms is more exemplary - and doesn't sound particularly good :-P
+        */
+
+        // STAIRS (mostly usefully as test-pattern)
+        // value_avg = dac_buffer_staircase[dac_i] / active_tones_snapshot_length;
+    }
+
+    return value;
+}
+
+/**
+ * DAC streaming callback. Does all of the main computing for playing songs.
+ *
+ * Note: chibios calls this CB twice: during the 'half buffer event', and the 'full buffer event'.
+ */
+static void dac_end(DACDriver *dacp) {
+    dacsample_t *sample_p = (dacp)->samples;
+
+    // work on the other half of the buffer
+    if (dacIsBufferComplete(dacp)) {
+        sample_p += AUDIO_DAC_BUFFER_SIZE / 2;  // 'half_index'
+    }
+
+    for (uint8_t s = 0; s < AUDIO_DAC_BUFFER_SIZE / 2; s++) {
+        if (OUTPUT_OFF <= state) {
+            sample_p[s] = AUDIO_DAC_OFF_VALUE;
+            continue;
+        } else {
+            sample_p[s] = dac_value_generate();
+        }
+
+        /* zero crossing (or approach, whereas zero == DAC_OFF_VALUE, which can be configured to anything from 0 to DAC_SAMPLE_MAX)
+         * ============================*=*========================== AUDIO_DAC_SAMPLE_MAX
+         *                          *       *
+         *                        *           *
+         * ---------------------------------------------------------
+         *                     *                 *                  } AUDIO_DAC_SAMPLE_MAX/100
+         * --------------------------------------------------------- AUDIO_DAC_OFF_VALUE
+         *                  *                       *               } AUDIO_DAC_SAMPLE_MAX/100
+         * ---------------------------------------------------------
+         *               *
+         * *           *
+         *   *       *
+         * =====*=*================================================= 0x0
+         */
+        if (((sample_p[s] + (AUDIO_DAC_SAMPLE_MAX / 100)) > AUDIO_DAC_OFF_VALUE) &&  // value approaches from below
+            (sample_p[s] < (AUDIO_DAC_OFF_VALUE + (AUDIO_DAC_SAMPLE_MAX / 100)))     // or above
+        ) {
+            if ((OUTPUT_SHOULD_START == state) && (active_tones_snapshot_length > 0)) {
+                state = OUTPUT_RUN_NORMALLY;
+            } else if (OUTPUT_TONES_CHANGED == state) {
+                state = OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE;
+            } else if (OUTPUT_SHOULD_STOP == state) {
+                state = OUTPUT_REACHED_ZERO_BEFORE_OFF;
+            }
+        }
+
+        // still 'ramping up', reset the output to OFF_VALUE until the generated values reach that value, to do a smooth handover
+        if (OUTPUT_SHOULD_START == state) {
+            sample_p[s] = AUDIO_DAC_OFF_VALUE;
+        }
+
+        if ((OUTPUT_SHOULD_START == state) || (OUTPUT_REACHED_ZERO_BEFORE_OFF == state) || (OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE == state)) {
+            uint8_t active_tones         = MIN(AUDIO_MAX_SIMULTANEOUS_TONES, audio_get_number_of_active_tones());
+            active_tones_snapshot_length = 0;
+            // update the snapshot - once, and only on occasion that something changed;
+            // -> saves cpu cycles (?)
+            for (uint8_t i = 0; i < active_tones; i++) {
+                float freq = audio_get_processed_frequency(i);
+                if (freq > 0) {  // disregard 'rest' notes, with valid frequency 0.0f; which would only lower the resulting waveform volume during the additive synthesis step
+                    active_tones_snapshot[active_tones_snapshot_length++] = freq;
+                }
+            }
+
+            if ((0 == active_tones_snapshot_length) && (OUTPUT_REACHED_ZERO_BEFORE_OFF == state)) {
+                state = OUTPUT_OFF;
+            }
+            if (OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE == state) {
+                state = OUTPUT_RUN_NORMALLY;
+            }
+        }
+    }
+
+    // update audio internal state (note position, current_note, ...)
+    if (audio_update_state()) {
+        if (OUTPUT_SHOULD_STOP != state) {
+            state = OUTPUT_TONES_CHANGED;
+        }
+    }
+
+    if (OUTPUT_OFF <= state) {
+        if (OUTPUT_OFF_2 == state) {
+            // stopping timer6 = stopping the DAC at whatever value it is currently pushing to the output = AUDIO_DAC_OFF_VALUE
+            gptStopTimer(&GPTD6);
+        } else {
+            state++;
+        }
+    }
+}
+
+static void dac_error(DACDriver *dacp, dacerror_t err) {
+    (void)dacp;
+    (void)err;
+
+    chSysHalt("DAC failure. halp");
+}
+
+static const GPTConfig gpt6cfg1 = {.frequency = AUDIO_DAC_SAMPLE_RATE * 3,
+                                   .callback  = NULL,
+                                   .cr2       = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event.  */
+                                   .dier      = 0U};
+
+static const DACConfig dac_conf = {.init = AUDIO_DAC_OFF_VALUE, .datamode = DAC_DHRM_12BIT_RIGHT};
+
+/**
+ * @note The DAC_TRG(0) here selects the Timer 6 TRGO event, which is triggered
+ * on the rising edge after 3 APB1 clock cycles, causing our gpt6cfg1.frequency
+ * to be a third of what we expect.
+ *
+ * Here are all the values for DAC_TRG (TSEL in the ref manual)
+ * TIM15_TRGO 0b011
+ * TIM2_TRGO  0b100
+ * TIM3_TRGO  0b001
+ * TIM6_TRGO  0b000
+ * TIM7_TRGO  0b010
+ * EXTI9      0b110
+ * SWTRIG     0b111
+ */
+static const DACConversionGroup dac_conv_cfg = {.num_channels = 1U, .end_cb = dac_end, .error_cb = dac_error, .trigger = DAC_TRG(0b000)};
+
+void audio_driver_initialize() {
+    if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
+        palSetLineMode(A4, PAL_MODE_INPUT_ANALOG);
+        dacStart(&DACD1, &dac_conf);
+    }
+    if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
+        palSetLineMode(A5, PAL_MODE_INPUT_ANALOG);
+        dacStart(&DACD2, &dac_conf);
+    }
+
+    /* enable the output buffer, to directly drive external loads with no additional circuitry
+     *
+     * see: AN4566 Application note: Extending the DAC performance of STM32 microcontrollers
+     * Note: Buffer-Off bit -> has to be set 0 to enable the output buffer
+     * Note: enabling the output buffer imparts an additional dc-offset of a couple mV
+     *
+     * this is done here, reaching directly into the stm32 registers since chibios has not implemented BOFF handling yet
+     * (see: chibios/os/hal/ports/STM32/todo.txt '- BOFF handling in DACv1.'
+     */
+    DACD1.params->dac->CR &= ~DAC_CR_BOFF1;
+    DACD2.params->dac->CR &= ~DAC_CR_BOFF2;
+
+    if (AUDIO_PIN == A4) {
+        dacStartConversion(&DACD1, &dac_conv_cfg, dac_buffer_empty, AUDIO_DAC_BUFFER_SIZE);
+    } else if (AUDIO_PIN == A5) {
+        dacStartConversion(&DACD2, &dac_conv_cfg, dac_buffer_empty, AUDIO_DAC_BUFFER_SIZE);
+    }
+
+    // no inverted/out-of-phase waveform (yet?), only pulling AUDIO_PIN_ALT to AUDIO_DAC_OFF_VALUE
+#if defined(AUDIO_PIN_ALT_AS_NEGATIVE)
+    if (AUDIO_PIN_ALT == A4) {
+        dacPutChannelX(&DACD1, 0, AUDIO_DAC_OFF_VALUE);
+    } else if (AUDIO_PIN_ALT == A5) {
+        dacPutChannelX(&DACD2, 0, AUDIO_DAC_OFF_VALUE);
+    }
+#endif
+
+    gptStart(&GPTD6, &gpt6cfg1);
+}
+
+void audio_driver_stop(void) { state = OUTPUT_SHOULD_STOP; }
+
+void audio_driver_start(void) {
+    gptStartContinuous(&GPTD6, 2U);
+
+    for (uint8_t i = 0; i < AUDIO_MAX_SIMULTANEOUS_TONES; i++) {
+        dac_if[i]                = 0.0f;
+        active_tones_snapshot[i] = 0.0f;
+    }
+    active_tones_snapshot_length = 0;
+    state                        = OUTPUT_SHOULD_START;
+}
diff --git a/platforms/chibios/drivers/audio_dac_basic.c b/platforms/chibios/drivers/audio_dac_basic.c
new file mode 100644
index 0000000000..fac6513506
--- /dev/null
+++ b/platforms/chibios/drivers/audio_dac_basic.c
@@ -0,0 +1,245 @@
+/* Copyright 2016-2020 Jack Humbert
+ * Copyright 2020 JohSchneider
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include "audio.h"
+#include "ch.h"
+#include "hal.h"
+
+/*
+  Audio Driver: DAC
+
+  which utilizes both channels of the DAC unit many STM32 are equipped with to output a modulated square-wave, from precomputed samples stored in a buffer, which is passed to the hardware through DMA
+
+  this driver can either be used to drive to separate speakers, wired to A4+Gnd and A5+Gnd, which allows two tones to be played simultaneously
+  OR
+  one speaker wired to A4+A5 with the AUDIO_PIN_ALT_AS_NEGATIVE define set - see docs/feature_audio
+
+*/
+
+#if !defined(AUDIO_PIN)
+#    pragma message "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under 'ARM (DAC basic)' for available options."
+// TODO: make this an 'error' instead; go through a breaking change, and add AUDIO_PIN A5 to all keyboards currently using AUDIO on STM32 based boards? - for now: set the define here
+#    define AUDIO_PIN A5
+#endif
+// check configuration for ONE speaker, connected to both DAC pins
+#if defined(AUDIO_PIN_ALT_AS_NEGATIVE) && !defined(AUDIO_PIN_ALT)
+#    error "Audio feature: AUDIO_PIN_ALT_AS_NEGATIVE set, but no pin configured as AUDIO_PIN_ALT"
+#endif
+
+#ifndef AUDIO_PIN_ALT
+// no ALT pin defined is valid, but the c-ifs below need some value set
+#    define AUDIO_PIN_ALT -1
+#endif
+
+#if !defined(AUDIO_STATE_TIMER)
+#    define AUDIO_STATE_TIMER GPTD8
+#endif
+
+// square-wave
+static const dacsample_t dac_buffer_1[AUDIO_DAC_BUFFER_SIZE] = {
+    // First half is max, second half is 0
+    [0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1]                     = AUDIO_DAC_SAMPLE_MAX,
+    [AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = 0,
+};
+
+// square-wave
+static const dacsample_t dac_buffer_2[AUDIO_DAC_BUFFER_SIZE] = {
+    // opposite of dac_buffer above
+    [0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1]                     = 0,
+    [AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = AUDIO_DAC_SAMPLE_MAX,
+};
+
+GPTConfig gpt6cfg1 = {.frequency = AUDIO_DAC_SAMPLE_RATE,
+                      .callback  = NULL,
+                      .cr2       = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event.    */
+                      .dier      = 0U};
+GPTConfig gpt7cfg1 = {.frequency = AUDIO_DAC_SAMPLE_RATE,
+                      .callback  = NULL,
+                      .cr2       = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event.    */
+                      .dier      = 0U};
+
+static void gpt_audio_state_cb(GPTDriver *gptp);
+GPTConfig   gptStateUpdateCfg = {.frequency = 10,
+                               .callback  = gpt_audio_state_cb,
+                               .cr2       = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event.    */
+                               .dier      = 0U};
+
+static const DACConfig dac_conf_ch1 = {.init = AUDIO_DAC_OFF_VALUE, .datamode = DAC_DHRM_12BIT_RIGHT};
+static const DACConfig dac_conf_ch2 = {.init = AUDIO_DAC_OFF_VALUE, .datamode = DAC_DHRM_12BIT_RIGHT};
+
+/**
+ * @note The DAC_TRG(0) here selects the Timer 6 TRGO event, which is triggered
+ * on the rising edge after 3 APB1 clock cycles, causing our gpt6cfg1.frequency
+ * to be a third of what we expect.
+ *
+ * Here are all the values for DAC_TRG (TSEL in the ref manual)
+ * TIM15_TRGO 0b011
+ * TIM2_TRGO  0b100
+ * TIM3_TRGO  0b001
+ * TIM6_TRGO  0b000
+ * TIM7_TRGO  0b010
+ * EXTI9      0b110
+ * SWTRIG     0b111
+ */
+static const DACConversionGroup dac_conv_grp_ch1 = {.num_channels = 1U, .trigger = DAC_TRG(0b000)};
+static const DACConversionGroup dac_conv_grp_ch2 = {.num_channels = 1U, .trigger = DAC_TRG(0b010)};
+
+void channel_1_start(void) {
+    gptStart(&GPTD6, &gpt6cfg1);
+    gptStartContinuous(&GPTD6, 2U);
+    palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG);
+}
+
+void channel_1_stop(void) {
+    gptStopTimer(&GPTD6);
+    palSetPadMode(GPIOA, 4, PAL_MODE_OUTPUT_PUSHPULL);
+    palSetPad(GPIOA, 4);
+}
+
+static float channel_1_frequency = 0.0f;
+void         channel_1_set_frequency(float freq) {
+    channel_1_frequency = freq;
+
+    channel_1_stop();
+    if (freq <= 0.0)  // a pause/rest has freq=0
+        return;
+
+    gpt6cfg1.frequency = 2 * freq * AUDIO_DAC_BUFFER_SIZE;
+    channel_1_start();
+}
+float channel_1_get_frequency(void) { return channel_1_frequency; }
+
+void channel_2_start(void) {
+    gptStart(&GPTD7, &gpt7cfg1);
+    gptStartContinuous(&GPTD7, 2U);
+    palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG);
+}
+
+void channel_2_stop(void) {
+    gptStopTimer(&GPTD7);
+    palSetPadMode(GPIOA, 5, PAL_MODE_OUTPUT_PUSHPULL);
+    palSetPad(GPIOA, 5);
+}
+
+static float channel_2_frequency = 0.0f;
+void         channel_2_set_frequency(float freq) {
+    channel_2_frequency = freq;
+
+    channel_2_stop();
+    if (freq <= 0.0)  // a pause/rest has freq=0
+        return;
+
+    gpt7cfg1.frequency = 2 * freq * AUDIO_DAC_BUFFER_SIZE;
+    channel_2_start();
+}
+float channel_2_get_frequency(void) { return channel_2_frequency; }
+
+static void gpt_audio_state_cb(GPTDriver *gptp) {
+    if (audio_update_state()) {
+#if defined(AUDIO_PIN_ALT_AS_NEGATIVE)
+        // one piezo/speaker connected to both audio pins, the generated square-waves are inverted
+        channel_1_set_frequency(audio_get_processed_frequency(0));
+        channel_2_set_frequency(audio_get_processed_frequency(0));
+
+#else  // two separate audio outputs/speakers
+       // primary speaker on A4, optional secondary on A5
+        if (AUDIO_PIN == A4) {
+            channel_1_set_frequency(audio_get_processed_frequency(0));
+            if (AUDIO_PIN_ALT == A5) {
+                if (audio_get_number_of_active_tones() > 1) {
+                    channel_2_set_frequency(audio_get_processed_frequency(1));
+                } else {
+                    channel_2_stop();
+                }
+            }
+        }
+
+        // primary speaker on A5, optional secondary on A4
+        if (AUDIO_PIN == A5) {
+            channel_2_set_frequency(audio_get_processed_frequency(0));
+            if (AUDIO_PIN_ALT == A4) {
+                if (audio_get_number_of_active_tones() > 1) {
+                    channel_1_set_frequency(audio_get_processed_frequency(1));
+                } else {
+                    channel_1_stop();
+                }
+            }
+        }
+#endif
+    }
+}
+
+void audio_driver_initialize() {
+    if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
+        palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG);
+        dacStart(&DACD1, &dac_conf_ch1);
+
+        // initial setup of the dac-triggering timer is still required, even
+        // though it gets reconfigured and restarted later on
+        gptStart(&GPTD6, &gpt6cfg1);
+    }
+
+    if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
+        palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG);
+        dacStart(&DACD2, &dac_conf_ch2);
+
+        gptStart(&GPTD7, &gpt7cfg1);
+    }
+
+    /* enable the output buffer, to directly drive external loads with no additional circuitry
+     *
+     * see: AN4566 Application note: Extending the DAC performance of STM32 microcontrollers
+     * Note: Buffer-Off bit -> has to be set 0 to enable the output buffer
+     * Note: enabling the output buffer imparts an additional dc-offset of a couple mV
+     *
+     * this is done here, reaching directly into the stm32 registers since chibios has not implemented BOFF handling yet
+     * (see: chibios/os/hal/ports/STM32/todo.txt '- BOFF handling in DACv1.'
+     */
+    DACD1.params->dac->CR &= ~DAC_CR_BOFF1;
+    DACD2.params->dac->CR &= ~DAC_CR_BOFF2;
+
+    // start state-updater
+    gptStart(&AUDIO_STATE_TIMER, &gptStateUpdateCfg);
+}
+
+void audio_driver_stop(void) {
+    if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
+        gptStopTimer(&GPTD6);
+
+        // stop the ongoing conversion and put the output in a known state
+        dacStopConversion(&DACD1);
+        dacPutChannelX(&DACD1, 0, AUDIO_DAC_OFF_VALUE);
+    }
+
+    if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
+        gptStopTimer(&GPTD7);
+
+        dacStopConversion(&DACD2);
+        dacPutChannelX(&DACD2, 0, AUDIO_DAC_OFF_VALUE);
+    }
+    gptStopTimer(&AUDIO_STATE_TIMER);
+}
+
+void audio_driver_start(void) {
+    if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
+        dacStartConversion(&DACD1, &dac_conv_grp_ch1, (dacsample_t *)dac_buffer_1, AUDIO_DAC_BUFFER_SIZE);
+    }
+    if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
+        dacStartConversion(&DACD2, &dac_conv_grp_ch2, (dacsample_t *)dac_buffer_2, AUDIO_DAC_BUFFER_SIZE);
+    }
+    gptStartContinuous(&AUDIO_STATE_TIMER, 2U);
+}
diff --git a/platforms/chibios/drivers/audio_pwm.h b/platforms/chibios/drivers/audio_pwm.h
new file mode 100644
index 0000000000..86cab916e1
--- /dev/null
+++ b/platforms/chibios/drivers/audio_pwm.h
@@ -0,0 +1,40 @@
+/* Copyright 2020 Jack Humbert
+ * Copyright 2020 JohSchneider
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ */
+#pragma once
+
+#if !defined(AUDIO_PWM_DRIVER)
+// NOTE: Timer2 seems to be used otherwise in QMK, otherwise we could default to A5 (= TIM2_CH1, with PWMD2 and alternate-function(1))
+#    define AUDIO_PWM_DRIVER PWMD1
+#endif
+
+#if !defined(AUDIO_PWM_CHANNEL)
+// NOTE: sticking to the STM data-sheet numbering: TIMxCH1 to TIMxCH4
+// default: STM32F303CC PA8+TIM1_CH1 -> 1
+#    define AUDIO_PWM_CHANNEL 1
+#endif
+
+#if !defined(AUDIO_PWM_PAL_MODE)
+// pin-alternate function: see the data-sheet for which pin needs what AF to connect to TIMx_CHy
+// default: STM32F303CC PA8+TIM1_CH1 -> 6
+#    define AUDIO_PWM_PAL_MODE 6
+#endif
+
+#if !defined(AUDIO_STATE_TIMER)
+// timer used to trigger updates in the audio-system, configured/enabled in chibios mcuconf.
+// Tim6 is the default for "larger" STMs, smaller ones might not have this one (enabled) and need to switch to a different one (e.g.: STM32F103 has only Tim1-Tim4)
+#    define AUDIO_STATE_TIMER GPTD6
+#endif
diff --git a/platforms/chibios/drivers/audio_pwm_hardware.c b/platforms/chibios/drivers/audio_pwm_hardware.c
new file mode 100644
index 0000000000..cd40019ee7
--- /dev/null
+++ b/platforms/chibios/drivers/audio_pwm_hardware.c
@@ -0,0 +1,144 @@
+/* Copyright 2020 Jack Humbert
+ * Copyright 2020 JohSchneider
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+/*
+Audio Driver: PWM
+
+the duty-cycle is always kept at 50%, and the pwm-period is adjusted to match the frequency of a note to be played back.
+
+this driver uses the chibios-PWM system to produce a square-wave on specific output pins that are connected to the PWM hardware.
+The hardware directly toggles the pin via its alternate function. see your MCUs data-sheet for which pin can be driven by what timer - looking for TIMx_CHy and the corresponding alternate function.
+
+ */
+
+#include "audio.h"
+#include "ch.h"
+#include "hal.h"
+
+#if !defined(AUDIO_PIN)
+#    error "Audio feature enabled, but no pin selected - see docs/feature_audio under the ARM PWM settings"
+#endif
+
+extern bool    playing_note;
+extern bool    playing_melody;
+extern uint8_t note_timbre;
+
+static PWMConfig pwmCFG = {
+    .frequency = 100000, /* PWM clock frequency  */
+    // CHIBIOS-BUG? can't set the initial period to <2, or the pwm (hard or software) takes ~130ms with .frequency=500000 for a pwmChangePeriod to take effect; with no output=silence in the meantime
+    .period   = 2,    /* initial PWM period (in ticks) 1S (1/10kHz=0.1mS 0.1ms*10000 ticks=1S) */
+    .callback = NULL, /* no callback, the hardware directly toggles the pin */
+    .channels =
+        {
+#if AUDIO_PWM_CHANNEL == 4
+            {PWM_OUTPUT_DISABLED, NULL},   /* channel 0 -> TIMx_CH1 */
+            {PWM_OUTPUT_DISABLED, NULL},   /* channel 1 -> TIMx_CH2 */
+            {PWM_OUTPUT_DISABLED, NULL},   /* channel 2 -> TIMx_CH3 */
+            {PWM_OUTPUT_ACTIVE_HIGH, NULL} /* channel 3 -> TIMx_CH4 */
+#elif AUDIO_PWM_CHANNEL == 3
+            {PWM_OUTPUT_DISABLED, NULL},
+            {PWM_OUTPUT_DISABLED, NULL},
+            {PWM_OUTPUT_ACTIVE_HIGH, NULL}, /* TIMx_CH3 */
+            {PWM_OUTPUT_DISABLED, NULL}
+#elif AUDIO_PWM_CHANNEL == 2
+            {PWM_OUTPUT_DISABLED, NULL},
+            {PWM_OUTPUT_ACTIVE_HIGH, NULL}, /* TIMx_CH2 */
+            {PWM_OUTPUT_DISABLED, NULL},
+            {PWM_OUTPUT_DISABLED, NULL}
+#else /*fallback to CH1 */
+            {PWM_OUTPUT_ACTIVE_HIGH, NULL}, /* TIMx_CH1 */
+            {PWM_OUTPUT_DISABLED, NULL},
+            {PWM_OUTPUT_DISABLED, NULL},
+            {PWM_OUTPUT_DISABLED, NULL}
+#endif
+        },
+};
+
+static float channel_1_frequency = 0.0f;
+void         channel_1_set_frequency(float freq) {
+    channel_1_frequency = freq;
+
+    if (freq <= 0.0)  // a pause/rest has freq=0
+        return;
+
+    pwmcnt_t period = (pwmCFG.frequency / freq);
+    pwmChangePeriod(&AUDIO_PWM_DRIVER, period);
+    pwmEnableChannel(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1,
+                     // adjust the duty-cycle so that the output is for 'note_timbre' duration HIGH
+                     PWM_PERCENTAGE_TO_WIDTH(&AUDIO_PWM_DRIVER, (100 - note_timbre) * 100));
+}
+
+float channel_1_get_frequency(void) { return channel_1_frequency; }
+
+void channel_1_start(void) {
+    pwmStop(&AUDIO_PWM_DRIVER);
+    pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
+}
+
+void channel_1_stop(void) { pwmStop(&AUDIO_PWM_DRIVER); }
+
+static void gpt_callback(GPTDriver *gptp);
+GPTConfig   gptCFG = {
+    /* a whole note is one beat, which is - per definition in musical_notes.h - set to 64
+       the longest note is BREAVE_DOT=128+64=192, the shortest SIXTEENTH=4
+       the tempo (which might vary!) is in bpm (beats per minute)
+       therefore: if the timer ticks away at .frequency = (60*64)Hz,
+       and the .interval counts from 64 downwards - audio_update_state is
+       called just often enough to not miss any notes
+    */
+    .frequency = 60 * 64,
+    .callback  = gpt_callback,
+};
+
+void audio_driver_initialize(void) {
+    pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
+
+    // connect the AUDIO_PIN to the PWM hardware
+#if defined(USE_GPIOV1)  // STM32F103C8
+    palSetLineMode(AUDIO_PIN, PAL_MODE_ALTERNATE_PUSHPULL);
+#else  // GPIOv2 (or GPIOv3 for f4xx, which is the same/compatible at this command)
+    palSetLineMode(AUDIO_PIN, PAL_MODE_ALTERNATE(AUDIO_PWM_PAL_MODE));
+#endif
+
+    gptStart(&AUDIO_STATE_TIMER, &gptCFG);
+}
+
+void audio_driver_start(void) {
+    channel_1_stop();
+    channel_1_start();
+
+    if (playing_note || playing_melody) {
+        gptStartContinuous(&AUDIO_STATE_TIMER, 64);
+    }
+}
+
+void audio_driver_stop(void) {
+    channel_1_stop();
+    gptStopTimer(&AUDIO_STATE_TIMER);
+}
+
+/* a regular timer task, that checks the note to be currently played
+ * and updates the pwm to output that frequency
+ */
+static void gpt_callback(GPTDriver *gptp) {
+    float freq;  // TODO: freq_alt
+
+    if (audio_update_state()) {
+        freq = audio_get_processed_frequency(0);  // freq_alt would be index=1
+        channel_1_set_frequency(freq);
+    }
+}
diff --git a/platforms/chibios/drivers/audio_pwm_software.c b/platforms/chibios/drivers/audio_pwm_software.c
new file mode 100644
index 0000000000..15c3e98b6a
--- /dev/null
+++ b/platforms/chibios/drivers/audio_pwm_software.c
@@ -0,0 +1,164 @@
+/* Copyright 2020 Jack Humbert
+ * Copyright 2020 JohSchneider
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+/*
+Audio Driver: PWM
+
+the duty-cycle is always kept at 50%, and the pwm-period is adjusted to match the frequency of a note to be played back.
+
+this driver uses the chibios-PWM system to produce a square-wave on any given output pin in software
+- a pwm callback is used to set/clear the configured pin.
+
+ */
+#include "audio.h"
+#include "ch.h"
+#include "hal.h"
+
+#if !defined(AUDIO_PIN)
+#    error "Audio feature enabled, but no pin selected - see docs/feature_audio under the ARM PWM settings"
+#endif
+extern bool    playing_note;
+extern bool    playing_melody;
+extern uint8_t note_timbre;
+
+static void pwm_audio_period_callback(PWMDriver *pwmp);
+static void pwm_audio_channel_interrupt_callback(PWMDriver *pwmp);
+
+static PWMConfig pwmCFG = {
+    .frequency = 100000, /* PWM clock frequency  */
+    // CHIBIOS-BUG? can't set the initial period to <2, or the pwm (hard or software) takes ~130ms with .frequency=500000 for a pwmChangePeriod to take effect; with no output=silence in the meantime
+    .period   = 2, /* initial PWM period (in ticks) 1S (1/10kHz=0.1mS 0.1ms*10000 ticks=1S) */
+    .callback = pwm_audio_period_callback,
+    .channels =
+        {
+            // software-PWM just needs another callback on any channel
+            {PWM_OUTPUT_ACTIVE_HIGH, pwm_audio_channel_interrupt_callback}, /* channel 0 -> TIMx_CH1 */
+            {PWM_OUTPUT_DISABLED, NULL},                                    /* channel 1 -> TIMx_CH2 */
+            {PWM_OUTPUT_DISABLED, NULL},                                    /* channel 2 -> TIMx_CH3 */
+            {PWM_OUTPUT_DISABLED, NULL}                                     /* channel 3 -> TIMx_CH4 */
+        },
+};
+
+static float channel_1_frequency = 0.0f;
+void         channel_1_set_frequency(float freq) {
+    channel_1_frequency = freq;
+
+    if (freq <= 0.0)  // a pause/rest has freq=0
+        return;
+
+    pwmcnt_t period = (pwmCFG.frequency / freq);
+    pwmChangePeriod(&AUDIO_PWM_DRIVER, period);
+
+    pwmEnableChannel(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1,
+                     // adjust the duty-cycle so that the output is for 'note_timbre' duration HIGH
+                     PWM_PERCENTAGE_TO_WIDTH(&AUDIO_PWM_DRIVER, (100 - note_timbre) * 100));
+}
+
+float channel_1_get_frequency(void) { return channel_1_frequency; }
+
+void channel_1_start(void) {
+    pwmStop(&AUDIO_PWM_DRIVER);
+    pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
+
+    pwmEnablePeriodicNotification(&AUDIO_PWM_DRIVER);
+    pwmEnableChannelNotification(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1);
+}
+
+void channel_1_stop(void) {
+    pwmStop(&AUDIO_PWM_DRIVER);
+
+    palClearLine(AUDIO_PIN);  // leave the line low, after last note was played
+
+#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
+    palClearLine(AUDIO_PIN_ALT);  // leave the line low, after last note was played
+#endif
+}
+
+// generate a PWM signal on any pin, not necessarily the one connected to the timer
+static void pwm_audio_period_callback(PWMDriver *pwmp) {
+    (void)pwmp;
+    palClearLine(AUDIO_PIN);
+
+#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
+    palSetLine(AUDIO_PIN_ALT);
+#endif
+}
+static void pwm_audio_channel_interrupt_callback(PWMDriver *pwmp) {
+    (void)pwmp;
+    if (channel_1_frequency > 0) {
+        palSetLine(AUDIO_PIN);  // generate a PWM signal on any pin, not necessarily the one connected to the timer
+#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
+        palClearLine(AUDIO_PIN_ALT);
+#endif
+    }
+}
+
+static void gpt_callback(GPTDriver *gptp);
+GPTConfig   gptCFG = {
+    /* a whole note is one beat, which is - per definition in musical_notes.h - set to 64
+       the longest note is BREAVE_DOT=128+64=192, the shortest SIXTEENTH=4
+       the tempo (which might vary!) is in bpm (beats per minute)
+       therefore: if the timer ticks away at .frequency = (60*64)Hz,
+       and the .interval counts from 64 downwards - audio_update_state is
+       called just often enough to not miss anything
+    */
+    .frequency = 60 * 64,
+    .callback  = gpt_callback,
+};
+
+void audio_driver_initialize(void) {
+    pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
+
+    palSetLineMode(AUDIO_PIN, PAL_MODE_OUTPUT_PUSHPULL);
+    palClearLine(AUDIO_PIN);
+
+#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
+    palSetLineMode(AUDIO_PIN_ALT, PAL_MODE_OUTPUT_PUSHPULL);
+    palClearLine(AUDIO_PIN_ALT);
+#endif
+
+    pwmEnablePeriodicNotification(&AUDIO_PWM_DRIVER);  // enable pwm callbacks
+    pwmEnableChannelNotification(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1);
+
+    gptStart(&AUDIO_STATE_TIMER, &gptCFG);
+}
+
+void audio_driver_start(void) {
+    channel_1_stop();
+    channel_1_start();
+
+    if (playing_note || playing_melody) {
+        gptStartContinuous(&AUDIO_STATE_TIMER, 64);
+    }
+}
+
+void audio_driver_stop(void) {
+    channel_1_stop();
+    gptStopTimer(&AUDIO_STATE_TIMER);
+}
+
+/* a regular timer task, that checks the note to be currently played
+ * and updates the pwm to output that frequency
+ */
+static void gpt_callback(GPTDriver *gptp) {
+    float freq;  // TODO: freq_alt
+
+    if (audio_update_state()) {
+        freq = audio_get_processed_frequency(0);  // freq_alt would be index=1
+        channel_1_set_frequency(freq);
+    }
+}