From ba8f1454f46537609f65a6abb4bb0e82fecbc2f1 Mon Sep 17 00:00:00 2001 From: Drashna Jaelre Date: Tue, 5 Oct 2021 18:01:45 -0700 Subject: 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 Co-authored-by: Ryan --- common_features.mk | 2 +- platforms/avr/drivers/audio_pwm.h | 17 ++ platforms/avr/drivers/audio_pwm_hardware.c | 332 ++++++++++++++++++++++++ platforms/chibios/drivers/audio_dac.h | 126 ++++++++++ platforms/chibios/drivers/audio_dac_additive.c | 335 +++++++++++++++++++++++++ platforms/chibios/drivers/audio_dac_basic.c | 245 ++++++++++++++++++ platforms/chibios/drivers/audio_pwm.h | 40 +++ platforms/chibios/drivers/audio_pwm_hardware.c | 144 +++++++++++ platforms/chibios/drivers/audio_pwm_software.c | 164 ++++++++++++ quantum/audio/audio.h | 13 +- quantum/audio/driver_avr_pwm.h | 17 -- quantum/audio/driver_avr_pwm_hardware.c | 332 ------------------------ quantum/audio/driver_chibios_dac.h | 126 ---------- quantum/audio/driver_chibios_dac_additive.c | 335 ------------------------- quantum/audio/driver_chibios_dac_basic.c | 245 ------------------ quantum/audio/driver_chibios_pwm.h | 40 --- quantum/audio/driver_chibios_pwm_hardware.c | 144 ----------- quantum/audio/driver_chibios_pwm_software.c | 164 ------------ 18 files changed, 1408 insertions(+), 1413 deletions(-) create mode 100644 platforms/avr/drivers/audio_pwm.h create mode 100644 platforms/avr/drivers/audio_pwm_hardware.c create mode 100644 platforms/chibios/drivers/audio_dac.h create mode 100644 platforms/chibios/drivers/audio_dac_additive.c create mode 100644 platforms/chibios/drivers/audio_dac_basic.c create mode 100644 platforms/chibios/drivers/audio_pwm.h create mode 100644 platforms/chibios/drivers/audio_pwm_hardware.c create mode 100644 platforms/chibios/drivers/audio_pwm_software.c delete mode 100644 quantum/audio/driver_avr_pwm.h delete mode 100644 quantum/audio/driver_avr_pwm_hardware.c delete mode 100644 quantum/audio/driver_chibios_dac.h delete mode 100644 quantum/audio/driver_chibios_dac_additive.c delete mode 100644 quantum/audio/driver_chibios_dac_basic.c delete mode 100644 quantum/audio/driver_chibios_pwm.h delete mode 100644 quantum/audio/driver_chibios_pwm_hardware.c delete mode 100644 quantum/audio/driver_chibios_pwm_software.c diff --git a/common_features.mk b/common_features.mk index a71bcace02..d7b0f826aa 100644 --- a/common_features.mk +++ b/common_features.mk @@ -83,7 +83,7 @@ ifeq ($(strip $(AUDIO_ENABLE)), yes) SRC += $(QUANTUM_DIR)/process_keycode/process_audio.c SRC += $(QUANTUM_DIR)/process_keycode/process_clicky.c SRC += $(QUANTUM_DIR)/audio/audio.c ## common audio code, hardware agnostic - SRC += $(QUANTUM_DIR)/audio/driver_$(PLATFORM_KEY)_$(strip $(AUDIO_DRIVER)).c + SRC += $(PLATFORM_PATH)/$(PLATFORM_KEY)/$(DRIVER_DIR)/audio_$(strip $(AUDIO_DRIVER)).c SRC += $(QUANTUM_DIR)/audio/voices.c SRC += $(QUANTUM_DIR)/audio/luts.c endif diff --git a/platforms/avr/drivers/audio_pwm.h b/platforms/avr/drivers/audio_pwm.h new file mode 100644 index 0000000000..d6eb3571da --- /dev/null +++ b/platforms/avr/drivers/audio_pwm.h @@ -0,0 +1,17 @@ +/* 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 . + */ +#pragma once diff --git a/platforms/avr/drivers/audio_pwm_hardware.c b/platforms/avr/drivers/audio_pwm_hardware.c new file mode 100644 index 0000000000..df03a4558c --- /dev/null +++ b/platforms/avr/drivers/audio_pwm_hardware.c @@ -0,0 +1,332 @@ +/* Copyright 2016 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 . + */ + +#if defined(__AVR__) +# include +# include +# include +#endif + +#include "audio.h" + +extern bool playing_note; +extern bool playing_melody; +extern uint8_t note_timbre; + +#define CPU_PRESCALER 8 + +/* + Audio Driver: PWM + + drive up to two speakers through the AVR PWM hardware-peripheral, using timer1 and/or timer3 on Atmega32U4. + + the primary channel_1 can be connected to either pin PC4 PC5 or PC6 (the later being used by most AVR based keyboards) with a PMW signal generated by timer3 + and an optional secondary channel_2 on either pin PB5, PB6 or PB7, with a PWM signal from timer1 + + alternatively, the PWM pins on PORTB can be used as only/primary speaker +*/ + +#if defined(AUDIO_PIN) && (AUDIO_PIN != C4) && (AUDIO_PIN != C5) && (AUDIO_PIN != C6) && (AUDIO_PIN != B5) && (AUDIO_PIN != B6) && (AUDIO_PIN != B7) && (AUDIO_PIN != D5) +# error "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under the AVR settings for available options." +#endif + +#if (AUDIO_PIN == C4) || (AUDIO_PIN == C5) || (AUDIO_PIN == C6) +# define AUDIO1_PIN_SET +# define AUDIO1_TIMSKx TIMSK3 +# define AUDIO1_TCCRxA TCCR3A +# define AUDIO1_TCCRxB TCCR3B +# define AUDIO1_ICRx ICR3 +# define AUDIO1_WGMx0 WGM30 +# define AUDIO1_WGMx1 WGM31 +# define AUDIO1_WGMx2 WGM32 +# define AUDIO1_WGMx3 WGM33 +# define AUDIO1_CSx0 CS30 +# define AUDIO1_CSx1 CS31 +# define AUDIO1_CSx2 CS32 + +# if (AUDIO_PIN == C6) +# define AUDIO1_COMxy0 COM3A0 +# define AUDIO1_COMxy1 COM3A1 +# define AUDIO1_OCIExy OCIE3A +# define AUDIO1_OCRxy OCR3A +# define AUDIO1_PIN C6 +# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPA_vect +# elif (AUDIO_PIN == C5) +# define AUDIO1_COMxy0 COM3B0 +# define AUDIO1_COMxy1 COM3B1 +# define AUDIO1_OCIExy OCIE3B +# define AUDIO1_OCRxy OCR3B +# define AUDIO1_PIN C5 +# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPB_vect +# elif (AUDIO_PIN == C4) +# define AUDIO1_COMxy0 COM3C0 +# define AUDIO1_COMxy1 COM3C1 +# define AUDIO1_OCIExy OCIE3C +# define AUDIO1_OCRxy OCR3C +# define AUDIO1_PIN C4 +# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPC_vect +# endif +#endif + +#if defined(AUDIO_PIN) && defined(AUDIO_PIN_ALT) && (AUDIO_PIN == AUDIO_PIN_ALT) +# error "Audio feature: AUDIO_PIN and AUDIO_PIN_ALT on the same pin makes no sense." +#endif + +#if ((AUDIO_PIN == B5) && ((AUDIO_PIN_ALT == B6) || (AUDIO_PIN_ALT == B7))) || ((AUDIO_PIN == B6) && ((AUDIO_PIN_ALT == B5) || (AUDIO_PIN_ALT == B7))) || ((AUDIO_PIN == B7) && ((AUDIO_PIN_ALT == B5) || (AUDIO_PIN_ALT == B6))) +# error "Audio feature: PORTB as AUDIO_PIN and AUDIO_PIN_ALT at the same time is not supported." +#endif + +#if defined(AUDIO_PIN_ALT) && (AUDIO_PIN_ALT != B5) && (AUDIO_PIN_ALT != B6) && (AUDIO_PIN_ALT != B7) +# error "Audio feature: the pin selected as AUDIO_PIN_ALT is not supported." +#endif + +#if (AUDIO_PIN == B5) || (AUDIO_PIN == B6) || (AUDIO_PIN == B7) || (AUDIO_PIN_ALT == B5) || (AUDIO_PIN_ALT == B6) || (AUDIO_PIN_ALT == B7) || (AUDIO_PIN == D5) +# define AUDIO2_PIN_SET +# define AUDIO2_TIMSKx TIMSK1 +# define AUDIO2_TCCRxA TCCR1A +# define AUDIO2_TCCRxB TCCR1B +# define AUDIO2_ICRx ICR1 +# define AUDIO2_WGMx0 WGM10 +# define AUDIO2_WGMx1 WGM11 +# define AUDIO2_WGMx2 WGM12 +# define AUDIO2_WGMx3 WGM13 +# define AUDIO2_CSx0 CS10 +# define AUDIO2_CSx1 CS11 +# define AUDIO2_CSx2 CS12 + +# if (AUDIO_PIN == B5) || (AUDIO_PIN_ALT == B5) +# define AUDIO2_COMxy0 COM1A0 +# define AUDIO2_COMxy1 COM1A1 +# define AUDIO2_OCIExy OCIE1A +# define AUDIO2_OCRxy OCR1A +# define AUDIO2_PIN B5 +# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPA_vect +# elif (AUDIO_PIN == B6) || (AUDIO_PIN_ALT == B6) +# define AUDIO2_COMxy0 COM1B0 +# define AUDIO2_COMxy1 COM1B1 +# define AUDIO2_OCIExy OCIE1B +# define AUDIO2_OCRxy OCR1B +# define AUDIO2_PIN B6 +# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPB_vect +# elif (AUDIO_PIN == B7) || (AUDIO_PIN_ALT == B7) +# define AUDIO2_COMxy0 COM1C0 +# define AUDIO2_COMxy1 COM1C1 +# define AUDIO2_OCIExy OCIE1C +# define AUDIO2_OCRxy OCR1C +# define AUDIO2_PIN B7 +# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPC_vect +# elif (AUDIO_PIN == D5) && defined(__AVR_ATmega32A__) +# pragma message "Audio support for ATmega32A is experimental and can cause crashes." +# undef AUDIO2_TIMSKx +# define AUDIO2_TIMSKx TIMSK +# define AUDIO2_COMxy0 COM1A0 +# define AUDIO2_COMxy1 COM1A1 +# define AUDIO2_OCIExy OCIE1A +# define AUDIO2_OCRxy OCR1A +# define AUDIO2_PIN D5 +# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPA_vect +# endif +#endif + +// C6 seems to be the assumed default by many existing keyboard - but sill warn the user +#if !defined(AUDIO1_PIN_SET) && !defined(AUDIO2_PIN_SET) +# pragma message "Audio feature enabled, but no suitable pin selected - see docs/feature_audio under the AVR settings for available options. Don't expect to hear anything... :-)" +// TODO: make this an error - go through the breaking-change-process and change all keyboards to the new define +#endif +// ----------------------------------------------------------------------------- + +#ifdef AUDIO1_PIN_SET +static float channel_1_frequency = 0.0f; +void channel_1_set_frequency(float freq) { + if (freq == 0.0f) // a pause/rest is a valid "note" with freq=0 + { + // disable the output, but keep the pwm-ISR going (with the previous + // frequency) so the audio-state keeps getting updated + // Note: setting the duty-cycle 0 is not possible on non-inverting PWM mode - see the AVR data-sheet + AUDIO1_TCCRxA &= ~(_BV(AUDIO1_COMxy1) | _BV(AUDIO1_COMxy0)); + return; + } else { + AUDIO1_TCCRxA |= _BV(AUDIO1_COMxy1); // enable output, PWM mode + } + + channel_1_frequency = freq; + + // set pwm period + AUDIO1_ICRx = (uint16_t)(((float)F_CPU) / (freq * CPU_PRESCALER)); + // and duty cycle + AUDIO1_OCRxy = (uint16_t)((((float)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre / 100); +} + +void channel_1_start(void) { + // enable timer-counter ISR + AUDIO1_TIMSKx |= _BV(AUDIO1_OCIExy); + // enable timer-counter output + AUDIO1_TCCRxA |= _BV(AUDIO1_COMxy1); +} + +void channel_1_stop(void) { + // disable timer-counter ISR + AUDIO1_TIMSKx &= ~_BV(AUDIO1_OCIExy); + // disable timer-counter output + AUDIO1_TCCRxA &= ~(_BV(AUDIO1_COMxy1) | _BV(AUDIO1_COMxy0)); +} +#endif + +#ifdef AUDIO2_PIN_SET +static float channel_2_frequency = 0.0f; +void channel_2_set_frequency(float freq) { + if (freq == 0.0f) { + AUDIO2_TCCRxA &= ~(_BV(AUDIO2_COMxy1) | _BV(AUDIO2_COMxy0)); + return; + } else { + AUDIO2_TCCRxA |= _BV(AUDIO2_COMxy1); + } + + channel_2_frequency = freq; + + AUDIO2_ICRx = (uint16_t)(((float)F_CPU) / (freq * CPU_PRESCALER)); + AUDIO2_OCRxy = (uint16_t)((((float)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre / 100); +} + +float channel_2_get_frequency(void) { return channel_2_frequency; } + +void channel_2_start(void) { + AUDIO2_TIMSKx |= _BV(AUDIO2_OCIExy); + AUDIO2_TCCRxA |= _BV(AUDIO2_COMxy1); +} + +void channel_2_stop(void) { + AUDIO2_TIMSKx &= ~_BV(AUDIO2_OCIExy); + AUDIO2_TCCRxA &= ~(_BV(AUDIO2_COMxy1) | _BV(AUDIO2_COMxy0)); +} +#endif + +void audio_driver_initialize() { +#ifdef AUDIO1_PIN_SET + channel_1_stop(); + setPinOutput(AUDIO1_PIN); +#endif + +#ifdef AUDIO2_PIN_SET + channel_2_stop(); + setPinOutput(AUDIO2_PIN); +#endif + + // TCCR3A / TCCR3B: Timer/Counter #3 Control Registers TCCR3A/TCCR3B, TCCR1A/TCCR1B + // Compare Output Mode (COM3An and COM1An) = 0b00 = Normal port operation + // OC3A -- PC6 + // OC3B -- PC5 + // OC3C -- PC4 + // OC1A -- PB5 + // OC1B -- PB6 + // OC1C -- PB7 + + // Waveform Generation Mode (WGM3n) = 0b1110 = Fast PWM Mode 14. Period = ICR3, Duty Cycle OCR3A) + // OCR3A - PC6 + // OCR3B - PC5 + // OCR3C - PC4 + // OCR1A - PB5 + // OCR1B - PB6 + // OCR1C - PB7 + + // Clock Select (CS3n) = 0b010 = Clock / 8 +#ifdef AUDIO1_PIN_SET + // initialize timer-counter + AUDIO1_TCCRxA = (0 << AUDIO1_COMxy1) | (0 << AUDIO1_COMxy0) | (1 << AUDIO1_WGMx1) | (0 << AUDIO1_WGMx0); + AUDIO1_TCCRxB = (1 << AUDIO1_WGMx3) | (1 << AUDIO1_WGMx2) | (0 << AUDIO1_CSx2) | (1 << AUDIO1_CSx1) | (0 << AUDIO1_CSx0); +#endif + +#ifdef AUDIO2_PIN_SET + AUDIO2_TCCRxA = (0 << AUDIO2_COMxy1) | (0 << AUDIO2_COMxy0) | (1 << AUDIO2_WGMx1) | (0 << AUDIO2_WGMx0); + AUDIO2_TCCRxB = (1 << AUDIO2_WGMx3) | (1 << AUDIO2_WGMx2) | (0 << AUDIO2_CSx2) | (1 << AUDIO2_CSx1) | (0 << AUDIO2_CSx0); +#endif +} + +void audio_driver_stop() { +#ifdef AUDIO1_PIN_SET + channel_1_stop(); +#endif + +#ifdef AUDIO2_PIN_SET + channel_2_stop(); +#endif +} + +void audio_driver_start(void) { +#ifdef AUDIO1_PIN_SET + channel_1_start(); + if (playing_note) { + channel_1_set_frequency(audio_get_processed_frequency(0)); + } +#endif + +#if !defined(AUDIO1_PIN_SET) && defined(AUDIO2_PIN_SET) + channel_2_start(); + if (playing_note) { + channel_2_set_frequency(audio_get_processed_frequency(0)); + } +#endif +} + +static volatile uint32_t isr_counter = 0; +#ifdef AUDIO1_PIN_SET +ISR(AUDIO1_TIMERx_COMPy_vect) { + isr_counter++; + if (isr_counter < channel_1_frequency / (CPU_PRESCALER * 8)) return; + + isr_counter = 0; + bool state_changed = audio_update_state(); + + if (!playing_note && !playing_melody) { + channel_1_stop(); +# ifdef AUDIO2_PIN_SET + channel_2_stop(); +# endif + return; + } + + if (state_changed) { + channel_1_set_frequency(audio_get_processed_frequency(0)); +# ifdef AUDIO2_PIN_SET + if (audio_get_number_of_active_tones() > 1) { + channel_2_set_frequency(audio_get_processed_frequency(1)); + } else { + channel_2_stop(); + } +# endif + } +} +#endif + +#if !defined(AUDIO1_PIN_SET) && defined(AUDIO2_PIN_SET) +ISR(AUDIO2_TIMERx_COMPy_vect) { + isr_counter++; + if (isr_counter < channel_2_frequency / (CPU_PRESCALER * 8)) return; + + isr_counter = 0; + bool state_changed = audio_update_state(); + + if (!playing_note && !playing_melody) { + channel_2_stop(); + return; + } + + if (state_changed) { + channel_2_set_frequency(audio_get_processed_frequency(0)); + } +} +#endif 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 . + */ +#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 . + */ + +#include "audio.h" +#include +#include + +/* + 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 . + */ + +#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 . + */ +#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 . + */ + +/* +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 . + */ + +/* +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); + } +} diff --git a/quantum/audio/audio.h b/quantum/audio/audio.h index 56b9158a1a..290d461f5a 100644 --- a/quantum/audio/audio.h +++ b/quantum/audio/audio.h @@ -26,17 +26,12 @@ #if defined(__AVR__) # include -# if defined(AUDIO_DRIVER_PWM) -# include "driver_avr_pwm.h" -# endif #endif -#if defined(PROTOCOL_CHIBIOS) -# if defined(AUDIO_DRIVER_PWM) -# include "driver_chibios_pwm.h" -# elif defined(AUDIO_DRIVER_DAC) -# include "driver_chibios_dac.h" -# endif +#if defined(AUDIO_DRIVER_PWM) +# include "audio_pwm.h" +#elif defined(AUDIO_DRIVER_DAC) +# include "audio_dac.h" #endif typedef union { diff --git a/quantum/audio/driver_avr_pwm.h b/quantum/audio/driver_avr_pwm.h deleted file mode 100644 index d6eb3571da..0000000000 --- a/quantum/audio/driver_avr_pwm.h +++ /dev/null @@ -1,17 +0,0 @@ -/* 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 . - */ -#pragma once diff --git a/quantum/audio/driver_avr_pwm_hardware.c b/quantum/audio/driver_avr_pwm_hardware.c deleted file mode 100644 index df03a4558c..0000000000 --- a/quantum/audio/driver_avr_pwm_hardware.c +++ /dev/null @@ -1,332 +0,0 @@ -/* Copyright 2016 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 . - */ - -#if defined(__AVR__) -# include -# include -# include -#endif - -#include "audio.h" - -extern bool playing_note; -extern bool playing_melody; -extern uint8_t note_timbre; - -#define CPU_PRESCALER 8 - -/* - Audio Driver: PWM - - drive up to two speakers through the AVR PWM hardware-peripheral, using timer1 and/or timer3 on Atmega32U4. - - the primary channel_1 can be connected to either pin PC4 PC5 or PC6 (the later being used by most AVR based keyboards) with a PMW signal generated by timer3 - and an optional secondary channel_2 on either pin PB5, PB6 or PB7, with a PWM signal from timer1 - - alternatively, the PWM pins on PORTB can be used as only/primary speaker -*/ - -#if defined(AUDIO_PIN) && (AUDIO_PIN != C4) && (AUDIO_PIN != C5) && (AUDIO_PIN != C6) && (AUDIO_PIN != B5) && (AUDIO_PIN != B6) && (AUDIO_PIN != B7) && (AUDIO_PIN != D5) -# error "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under the AVR settings for available options." -#endif - -#if (AUDIO_PIN == C4) || (AUDIO_PIN == C5) || (AUDIO_PIN == C6) -# define AUDIO1_PIN_SET -# define AUDIO1_TIMSKx TIMSK3 -# define AUDIO1_TCCRxA TCCR3A -# define AUDIO1_TCCRxB TCCR3B -# define AUDIO1_ICRx ICR3 -# define AUDIO1_WGMx0 WGM30 -# define AUDIO1_WGMx1 WGM31 -# define AUDIO1_WGMx2 WGM32 -# define AUDIO1_WGMx3 WGM33 -# define AUDIO1_CSx0 CS30 -# define AUDIO1_CSx1 CS31 -# define AUDIO1_CSx2 CS32 - -# if (AUDIO_PIN == C6) -# define AUDIO1_COMxy0 COM3A0 -# define AUDIO1_COMxy1 COM3A1 -# define AUDIO1_OCIExy OCIE3A -# define AUDIO1_OCRxy OCR3A -# define AUDIO1_PIN C6 -# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPA_vect -# elif (AUDIO_PIN == C5) -# define AUDIO1_COMxy0 COM3B0 -# define AUDIO1_COMxy1 COM3B1 -# define AUDIO1_OCIExy OCIE3B -# define AUDIO1_OCRxy OCR3B -# define AUDIO1_PIN C5 -# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPB_vect -# elif (AUDIO_PIN == C4) -# define AUDIO1_COMxy0 COM3C0 -# define AUDIO1_COMxy1 COM3C1 -# define AUDIO1_OCIExy OCIE3C -# define AUDIO1_OCRxy OCR3C -# define AUDIO1_PIN C4 -# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPC_vect -# endif -#endif - -#if defined(AUDIO_PIN) && defined(AUDIO_PIN_ALT) && (AUDIO_PIN == AUDIO_PIN_ALT) -# error "Audio feature: AUDIO_PIN and AUDIO_PIN_ALT on the same pin makes no sense." -#endif - -#if ((AUDIO_PIN == B5) && ((AUDIO_PIN_ALT == B6) || (AUDIO_PIN_ALT == B7))) || ((AUDIO_PIN == B6) && ((AUDIO_PIN_ALT == B5) || (AUDIO_PIN_ALT == B7))) || ((AUDIO_PIN == B7) && ((AUDIO_PIN_ALT == B5) || (AUDIO_PIN_ALT == B6))) -# error "Audio feature: PORTB as AUDIO_PIN and AUDIO_PIN_ALT at the same time is not supported." -#endif - -#if defined(AUDIO_PIN_ALT) && (AUDIO_PIN_ALT != B5) && (AUDIO_PIN_ALT != B6) && (AUDIO_PIN_ALT != B7) -# error "Audio feature: the pin selected as AUDIO_PIN_ALT is not supported." -#endif - -#if (AUDIO_PIN == B5) || (AUDIO_PIN == B6) || (AUDIO_PIN == B7) || (AUDIO_PIN_ALT == B5) || (AUDIO_PIN_ALT == B6) || (AUDIO_PIN_ALT == B7) || (AUDIO_PIN == D5) -# define AUDIO2_PIN_SET -# define AUDIO2_TIMSKx TIMSK1 -# define AUDIO2_TCCRxA TCCR1A -# define AUDIO2_TCCRxB TCCR1B -# define AUDIO2_ICRx ICR1 -# define AUDIO2_WGMx0 WGM10 -# define AUDIO2_WGMx1 WGM11 -# define AUDIO2_WGMx2 WGM12 -# define AUDIO2_WGMx3 WGM13 -# define AUDIO2_CSx0 CS10 -# define AUDIO2_CSx1 CS11 -# define AUDIO2_CSx2 CS12 - -# if (AUDIO_PIN == B5) || (AUDIO_PIN_ALT == B5) -# define AUDIO2_COMxy0 COM1A0 -# define AUDIO2_COMxy1 COM1A1 -# define AUDIO2_OCIExy OCIE1A -# define AUDIO2_OCRxy OCR1A -# define AUDIO2_PIN B5 -# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPA_vect -# elif (AUDIO_PIN == B6) || (AUDIO_PIN_ALT == B6) -# define AUDIO2_COMxy0 COM1B0 -# define AUDIO2_COMxy1 COM1B1 -# define AUDIO2_OCIExy OCIE1B -# define AUDIO2_OCRxy OCR1B -# define AUDIO2_PIN B6 -# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPB_vect -# elif (AUDIO_PIN == B7) || (AUDIO_PIN_ALT == B7) -# define AUDIO2_COMxy0 COM1C0 -# define AUDIO2_COMxy1 COM1C1 -# define AUDIO2_OCIExy OCIE1C -# define AUDIO2_OCRxy OCR1C -# define AUDIO2_PIN B7 -# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPC_vect -# elif (AUDIO_PIN == D5) && defined(__AVR_ATmega32A__) -# pragma message "Audio support for ATmega32A is experimental and can cause crashes." -# undef AUDIO2_TIMSKx -# define AUDIO2_TIMSKx TIMSK -# define AUDIO2_COMxy0 COM1A0 -# define AUDIO2_COMxy1 COM1A1 -# define AUDIO2_OCIExy OCIE1A -# define AUDIO2_OCRxy OCR1A -# define AUDIO2_PIN D5 -# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPA_vect -# endif -#endif - -// C6 seems to be the assumed default by many existing keyboard - but sill warn the user -#if !defined(AUDIO1_PIN_SET) && !defined(AUDIO2_PIN_SET) -# pragma message "Audio feature enabled, but no suitable pin selected - see docs/feature_audio under the AVR settings for available options. Don't expect to hear anything... :-)" -// TODO: make this an error - go through the breaking-change-process and change all keyboards to the new define -#endif -// ----------------------------------------------------------------------------- - -#ifdef AUDIO1_PIN_SET -static float channel_1_frequency = 0.0f; -void channel_1_set_frequency(float freq) { - if (freq == 0.0f) // a pause/rest is a valid "note" with freq=0 - { - // disable the output, but keep the pwm-ISR going (with the previous - // frequency) so the audio-state keeps getting updated - // Note: setting the duty-cycle 0 is not possible on non-inverting PWM mode - see the AVR data-sheet - AUDIO1_TCCRxA &= ~(_BV(AUDIO1_COMxy1) | _BV(AUDIO1_COMxy0)); - return; - } else { - AUDIO1_TCCRxA |= _BV(AUDIO1_COMxy1); // enable output, PWM mode - } - - channel_1_frequency = freq; - - // set pwm period - AUDIO1_ICRx = (uint16_t)(((float)F_CPU) / (freq * CPU_PRESCALER)); - // and duty cycle - AUDIO1_OCRxy = (uint16_t)((((float)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre / 100); -} - -void channel_1_start(void) { - // enable timer-counter ISR - AUDIO1_TIMSKx |= _BV(AUDIO1_OCIExy); - // enable timer-counter output - AUDIO1_TCCRxA |= _BV(AUDIO1_COMxy1); -} - -void channel_1_stop(void) { - // disable timer-counter ISR - AUDIO1_TIMSKx &= ~_BV(AUDIO1_OCIExy); - // disable timer-counter output - AUDIO1_TCCRxA &= ~(_BV(AUDIO1_COMxy1) | _BV(AUDIO1_COMxy0)); -} -#endif - -#ifdef AUDIO2_PIN_SET -static float channel_2_frequency = 0.0f; -void channel_2_set_frequency(float freq) { - if (freq == 0.0f) { - AUDIO2_TCCRxA &= ~(_BV(AUDIO2_COMxy1) | _BV(AUDIO2_COMxy0)); - return; - } else { - AUDIO2_TCCRxA |= _BV(AUDIO2_COMxy1); - } - - channel_2_frequency = freq; - - AUDIO2_ICRx = (uint16_t)(((float)F_CPU) / (freq * CPU_PRESCALER)); - AUDIO2_OCRxy = (uint16_t)((((float)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre / 100); -} - -float channel_2_get_frequency(void) { return channel_2_frequency; } - -void channel_2_start(void) { - AUDIO2_TIMSKx |= _BV(AUDIO2_OCIExy); - AUDIO2_TCCRxA |= _BV(AUDIO2_COMxy1); -} - -void channel_2_stop(void) { - AUDIO2_TIMSKx &= ~_BV(AUDIO2_OCIExy); - AUDIO2_TCCRxA &= ~(_BV(AUDIO2_COMxy1) | _BV(AUDIO2_COMxy0)); -} -#endif - -void audio_driver_initialize() { -#ifdef AUDIO1_PIN_SET - channel_1_stop(); - setPinOutput(AUDIO1_PIN); -#endif - -#ifdef AUDIO2_PIN_SET - channel_2_stop(); - setPinOutput(AUDIO2_PIN); -#endif - - // TCCR3A / TCCR3B: Timer/Counter #3 Control Registers TCCR3A/TCCR3B, TCCR1A/TCCR1B - // Compare Output Mode (COM3An and COM1An) = 0b00 = Normal port operation - // OC3A -- PC6 - // OC3B -- PC5 - // OC3C -- PC4 - // OC1A -- PB5 - // OC1B -- PB6 - // OC1C -- PB7 - - // Waveform Generation Mode (WGM3n) = 0b1110 = Fast PWM Mode 14. Period = ICR3, Duty Cycle OCR3A) - // OCR3A - PC6 - // OCR3B - PC5 - // OCR3C - PC4 - // OCR1A - PB5 - // OCR1B - PB6 - // OCR1C - PB7 - - // Clock Select (CS3n) = 0b010 = Clock / 8 -#ifdef AUDIO1_PIN_SET - // initialize timer-counter - AUDIO1_TCCRxA = (0 << AUDIO1_COMxy1) | (0 << AUDIO1_COMxy0) | (1 << AUDIO1_WGMx1) | (0 << AUDIO1_WGMx0); - AUDIO1_TCCRxB = (1 << AUDIO1_WGMx3) | (1 << AUDIO1_WGMx2) | (0 << AUDIO1_CSx2) | (1 << AUDIO1_CSx1) | (0 << AUDIO1_CSx0); -#endif - -#ifdef AUDIO2_PIN_SET - AUDIO2_TCCRxA = (0 << AUDIO2_COMxy1) | (0 << AUDIO2_COMxy0) | (1 << AUDIO2_WGMx1) | (0 << AUDIO2_WGMx0); - AUDIO2_TCCRxB = (1 << AUDIO2_WGMx3) | (1 << AUDIO2_WGMx2) | (0 << AUDIO2_CSx2) | (1 << AUDIO2_CSx1) | (0 << AUDIO2_CSx0); -#endif -} - -void audio_driver_stop() { -#ifdef AUDIO1_PIN_SET - channel_1_stop(); -#endif - -#ifdef AUDIO2_PIN_SET - channel_2_stop(); -#endif -} - -void audio_driver_start(void) { -#ifdef AUDIO1_PIN_SET - channel_1_start(); - if (playing_note) { - channel_1_set_frequency(audio_get_processed_frequency(0)); - } -#endif - -#if !defined(AUDIO1_PIN_SET) && defined(AUDIO2_PIN_SET) - channel_2_start(); - if (playing_note) { - channel_2_set_frequency(audio_get_processed_frequency(0)); - } -#endif -} - -static volatile uint32_t isr_counter = 0; -#ifdef AUDIO1_PIN_SET -ISR(AUDIO1_TIMERx_COMPy_vect) { - isr_counter++; - if (isr_counter < channel_1_frequency / (CPU_PRESCALER * 8)) return; - - isr_counter = 0; - bool state_changed = audio_update_state(); - - if (!playing_note && !playing_melody) { - channel_1_stop(); -# ifdef AUDIO2_PIN_SET - channel_2_stop(); -# endif - return; - } - - if (state_changed) { - channel_1_set_frequency(audio_get_processed_frequency(0)); -# ifdef AUDIO2_PIN_SET - if (audio_get_number_of_active_tones() > 1) { - channel_2_set_frequency(audio_get_processed_frequency(1)); - } else { - channel_2_stop(); - } -# endif - } -} -#endif - -#if !defined(AUDIO1_PIN_SET) && defined(AUDIO2_PIN_SET) -ISR(AUDIO2_TIMERx_COMPy_vect) { - isr_counter++; - if (isr_counter < channel_2_frequency / (CPU_PRESCALER * 8)) return; - - isr_counter = 0; - bool state_changed = audio_update_state(); - - if (!playing_note && !playing_melody) { - channel_2_stop(); - return; - } - - if (state_changed) { - channel_2_set_frequency(audio_get_processed_frequency(0)); - } -} -#endif diff --git a/quantum/audio/driver_chibios_dac.h b/quantum/audio/driver_chibios_dac.h deleted file mode 100644 index 07cd622ead..0000000000 --- a/quantum/audio/driver_chibios_dac.h +++ /dev/null @@ -1,126 +0,0 @@ -/* 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 . - */ -#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/quantum/audio/driver_chibios_dac_additive.c b/quantum/audio/driver_chibios_dac_additive.c deleted file mode 100644 index db304adb87..0000000000 --- a/quantum/audio/driver_chibios_dac_additive.c +++ /dev/null @@ -1,335 +0,0 @@ -/* 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 . - */ - -#include "audio.h" -#include -#include - -/* - 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/quantum/audio/driver_chibios_dac_basic.c b/quantum/audio/driver_chibios_dac_basic.c deleted file mode 100644 index fac6513506..0000000000 --- a/quantum/audio/driver_chibios_dac_basic.c +++ /dev/null @@ -1,245 +0,0 @@ -/* 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 . - */ - -#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/quantum/audio/driver_chibios_pwm.h b/quantum/audio/driver_chibios_pwm.h deleted file mode 100644 index 86cab916e1..0000000000 --- a/quantum/audio/driver_chibios_pwm.h +++ /dev/null @@ -1,40 +0,0 @@ -/* 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 . - */ -#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/quantum/audio/driver_chibios_pwm_hardware.c b/quantum/audio/driver_chibios_pwm_hardware.c deleted file mode 100644 index cd40019ee7..0000000000 --- a/quantum/audio/driver_chibios_pwm_hardware.c +++ /dev/null @@ -1,144 +0,0 @@ -/* 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 . - */ - -/* -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/quantum/audio/driver_chibios_pwm_software.c b/quantum/audio/driver_chibios_pwm_software.c deleted file mode 100644 index 15c3e98b6a..0000000000 --- a/quantum/audio/driver_chibios_pwm_software.c +++ /dev/null @@ -1,164 +0,0 @@ -/* 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 . - */ - -/* -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); - } -} -- cgit v1.2.3