Move Audio drivers from quantum to platform drivers folder (#14308)
* Move Audio drivers from quantum to platform drivers folder * fix path for audio drivers Co-authored-by: Ryan <fauxpark@gmail.com> Co-authored-by: Ryan <fauxpark@gmail.com>
This commit is contained in:
Drashna Jaelre
GitHub
parent
9f0e74802a
commit
ba8f1454f4
10 changed files with 5 additions and 10 deletions
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@ -26,17 +26,12 @@
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#if defined(__AVR__)
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# include <avr/io.h>
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# if defined(AUDIO_DRIVER_PWM)
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# include "driver_avr_pwm.h"
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# endif
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#endif
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#if defined(PROTOCOL_CHIBIOS)
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# if defined(AUDIO_DRIVER_PWM)
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# include "driver_chibios_pwm.h"
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# elif defined(AUDIO_DRIVER_DAC)
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# include "driver_chibios_dac.h"
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# endif
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#if defined(AUDIO_DRIVER_PWM)
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# include "audio_pwm.h"
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#elif defined(AUDIO_DRIVER_DAC)
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# include "audio_dac.h"
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#endif
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typedef union {
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@ -1,17 +0,0 @@
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/* Copyright 2020 Jack Humbert
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* Copyright 2020 JohSchneider
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*
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* This program is free software: you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation, either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*/
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#pragma once
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@ -1,332 +0,0 @@
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/* Copyright 2016 Jack Humbert
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* Copyright 2020 JohSchneider
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*
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* This program is free software: you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation, either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*/
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#if defined(__AVR__)
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# include <avr/pgmspace.h>
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# include <avr/interrupt.h>
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# include <avr/io.h>
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#endif
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#include "audio.h"
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extern bool playing_note;
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extern bool playing_melody;
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extern uint8_t note_timbre;
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#define CPU_PRESCALER 8
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/*
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Audio Driver: PWM
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drive up to two speakers through the AVR PWM hardware-peripheral, using timer1 and/or timer3 on Atmega32U4.
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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
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and an optional secondary channel_2 on either pin PB5, PB6 or PB7, with a PWM signal from timer1
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alternatively, the PWM pins on PORTB can be used as only/primary speaker
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*/
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#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)
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# error "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under the AVR settings for available options."
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#endif
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#if (AUDIO_PIN == C4) || (AUDIO_PIN == C5) || (AUDIO_PIN == C6)
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# define AUDIO1_PIN_SET
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# define AUDIO1_TIMSKx TIMSK3
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# define AUDIO1_TCCRxA TCCR3A
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# define AUDIO1_TCCRxB TCCR3B
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# define AUDIO1_ICRx ICR3
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# define AUDIO1_WGMx0 WGM30
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# define AUDIO1_WGMx1 WGM31
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# define AUDIO1_WGMx2 WGM32
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# define AUDIO1_WGMx3 WGM33
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# define AUDIO1_CSx0 CS30
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# define AUDIO1_CSx1 CS31
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# define AUDIO1_CSx2 CS32
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# if (AUDIO_PIN == C6)
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# define AUDIO1_COMxy0 COM3A0
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# define AUDIO1_COMxy1 COM3A1
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# define AUDIO1_OCIExy OCIE3A
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# define AUDIO1_OCRxy OCR3A
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# define AUDIO1_PIN C6
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# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPA_vect
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# elif (AUDIO_PIN == C5)
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# define AUDIO1_COMxy0 COM3B0
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# define AUDIO1_COMxy1 COM3B1
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# define AUDIO1_OCIExy OCIE3B
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# define AUDIO1_OCRxy OCR3B
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# define AUDIO1_PIN C5
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# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPB_vect
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# elif (AUDIO_PIN == C4)
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# define AUDIO1_COMxy0 COM3C0
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# define AUDIO1_COMxy1 COM3C1
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# define AUDIO1_OCIExy OCIE3C
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# define AUDIO1_OCRxy OCR3C
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# define AUDIO1_PIN C4
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# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPC_vect
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# endif
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#endif
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#if defined(AUDIO_PIN) && defined(AUDIO_PIN_ALT) && (AUDIO_PIN == AUDIO_PIN_ALT)
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# error "Audio feature: AUDIO_PIN and AUDIO_PIN_ALT on the same pin makes no sense."
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#endif
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#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)))
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# error "Audio feature: PORTB as AUDIO_PIN and AUDIO_PIN_ALT at the same time is not supported."
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#endif
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#if defined(AUDIO_PIN_ALT) && (AUDIO_PIN_ALT != B5) && (AUDIO_PIN_ALT != B6) && (AUDIO_PIN_ALT != B7)
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# error "Audio feature: the pin selected as AUDIO_PIN_ALT is not supported."
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#endif
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#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)
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# define AUDIO2_PIN_SET
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# define AUDIO2_TIMSKx TIMSK1
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# define AUDIO2_TCCRxA TCCR1A
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# define AUDIO2_TCCRxB TCCR1B
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# define AUDIO2_ICRx ICR1
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# define AUDIO2_WGMx0 WGM10
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# define AUDIO2_WGMx1 WGM11
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# define AUDIO2_WGMx2 WGM12
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# define AUDIO2_WGMx3 WGM13
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# define AUDIO2_CSx0 CS10
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# define AUDIO2_CSx1 CS11
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# define AUDIO2_CSx2 CS12
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# if (AUDIO_PIN == B5) || (AUDIO_PIN_ALT == B5)
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# define AUDIO2_COMxy0 COM1A0
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# define AUDIO2_COMxy1 COM1A1
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# define AUDIO2_OCIExy OCIE1A
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# define AUDIO2_OCRxy OCR1A
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# define AUDIO2_PIN B5
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# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPA_vect
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# elif (AUDIO_PIN == B6) || (AUDIO_PIN_ALT == B6)
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# define AUDIO2_COMxy0 COM1B0
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# define AUDIO2_COMxy1 COM1B1
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# define AUDIO2_OCIExy OCIE1B
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# define AUDIO2_OCRxy OCR1B
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# define AUDIO2_PIN B6
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# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPB_vect
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# elif (AUDIO_PIN == B7) || (AUDIO_PIN_ALT == B7)
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# define AUDIO2_COMxy0 COM1C0
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# define AUDIO2_COMxy1 COM1C1
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# define AUDIO2_OCIExy OCIE1C
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# define AUDIO2_OCRxy OCR1C
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# define AUDIO2_PIN B7
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# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPC_vect
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# elif (AUDIO_PIN == D5) && defined(__AVR_ATmega32A__)
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# pragma message "Audio support for ATmega32A is experimental and can cause crashes."
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# undef AUDIO2_TIMSKx
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# define AUDIO2_TIMSKx TIMSK
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# define AUDIO2_COMxy0 COM1A0
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# define AUDIO2_COMxy1 COM1A1
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# define AUDIO2_OCIExy OCIE1A
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# define AUDIO2_OCRxy OCR1A
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# define AUDIO2_PIN D5
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# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPA_vect
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# endif
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#endif
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// C6 seems to be the assumed default by many existing keyboard - but sill warn the user
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#if !defined(AUDIO1_PIN_SET) && !defined(AUDIO2_PIN_SET)
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# 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... :-)"
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// TODO: make this an error - go through the breaking-change-process and change all keyboards to the new define
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#endif
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// -----------------------------------------------------------------------------
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#ifdef AUDIO1_PIN_SET
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static float channel_1_frequency = 0.0f;
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void channel_1_set_frequency(float freq) {
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if (freq == 0.0f) // a pause/rest is a valid "note" with freq=0
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{
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// disable the output, but keep the pwm-ISR going (with the previous
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// frequency) so the audio-state keeps getting updated
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// Note: setting the duty-cycle 0 is not possible on non-inverting PWM mode - see the AVR data-sheet
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AUDIO1_TCCRxA &= ~(_BV(AUDIO1_COMxy1) | _BV(AUDIO1_COMxy0));
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return;
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} else {
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AUDIO1_TCCRxA |= _BV(AUDIO1_COMxy1); // enable output, PWM mode
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}
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channel_1_frequency = freq;
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// set pwm period
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AUDIO1_ICRx = (uint16_t)(((float)F_CPU) / (freq * CPU_PRESCALER));
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// and duty cycle
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AUDIO1_OCRxy = (uint16_t)((((float)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre / 100);
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}
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void channel_1_start(void) {
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// enable timer-counter ISR
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AUDIO1_TIMSKx |= _BV(AUDIO1_OCIExy);
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// enable timer-counter output
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AUDIO1_TCCRxA |= _BV(AUDIO1_COMxy1);
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}
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void channel_1_stop(void) {
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// disable timer-counter ISR
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AUDIO1_TIMSKx &= ~_BV(AUDIO1_OCIExy);
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// disable timer-counter output
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AUDIO1_TCCRxA &= ~(_BV(AUDIO1_COMxy1) | _BV(AUDIO1_COMxy0));
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}
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#endif
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#ifdef AUDIO2_PIN_SET
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static float channel_2_frequency = 0.0f;
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void channel_2_set_frequency(float freq) {
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if (freq == 0.0f) {
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AUDIO2_TCCRxA &= ~(_BV(AUDIO2_COMxy1) | _BV(AUDIO2_COMxy0));
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return;
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} else {
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AUDIO2_TCCRxA |= _BV(AUDIO2_COMxy1);
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}
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channel_2_frequency = freq;
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AUDIO2_ICRx = (uint16_t)(((float)F_CPU) / (freq * CPU_PRESCALER));
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AUDIO2_OCRxy = (uint16_t)((((float)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre / 100);
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}
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float channel_2_get_frequency(void) { return channel_2_frequency; }
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void channel_2_start(void) {
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AUDIO2_TIMSKx |= _BV(AUDIO2_OCIExy);
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AUDIO2_TCCRxA |= _BV(AUDIO2_COMxy1);
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}
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void channel_2_stop(void) {
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AUDIO2_TIMSKx &= ~_BV(AUDIO2_OCIExy);
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AUDIO2_TCCRxA &= ~(_BV(AUDIO2_COMxy1) | _BV(AUDIO2_COMxy0));
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}
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#endif
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void audio_driver_initialize() {
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#ifdef AUDIO1_PIN_SET
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channel_1_stop();
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setPinOutput(AUDIO1_PIN);
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#endif
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#ifdef AUDIO2_PIN_SET
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channel_2_stop();
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setPinOutput(AUDIO2_PIN);
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#endif
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// TCCR3A / TCCR3B: Timer/Counter #3 Control Registers TCCR3A/TCCR3B, TCCR1A/TCCR1B
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// Compare Output Mode (COM3An and COM1An) = 0b00 = Normal port operation
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// OC3A -- PC6
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// OC3B -- PC5
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// OC3C -- PC4
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// OC1A -- PB5
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// OC1B -- PB6
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// OC1C -- PB7
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// Waveform Generation Mode (WGM3n) = 0b1110 = Fast PWM Mode 14. Period = ICR3, Duty Cycle OCR3A)
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// OCR3A - PC6
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// OCR3B - PC5
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// OCR3C - PC4
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// OCR1A - PB5
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// OCR1B - PB6
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// OCR1C - PB7
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// Clock Select (CS3n) = 0b010 = Clock / 8
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#ifdef AUDIO1_PIN_SET
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// initialize timer-counter
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AUDIO1_TCCRxA = (0 << AUDIO1_COMxy1) | (0 << AUDIO1_COMxy0) | (1 << AUDIO1_WGMx1) | (0 << AUDIO1_WGMx0);
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AUDIO1_TCCRxB = (1 << AUDIO1_WGMx3) | (1 << AUDIO1_WGMx2) | (0 << AUDIO1_CSx2) | (1 << AUDIO1_CSx1) | (0 << AUDIO1_CSx0);
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#endif
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#ifdef AUDIO2_PIN_SET
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AUDIO2_TCCRxA = (0 << AUDIO2_COMxy1) | (0 << AUDIO2_COMxy0) | (1 << AUDIO2_WGMx1) | (0 << AUDIO2_WGMx0);
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AUDIO2_TCCRxB = (1 << AUDIO2_WGMx3) | (1 << AUDIO2_WGMx2) | (0 << AUDIO2_CSx2) | (1 << AUDIO2_CSx1) | (0 << AUDIO2_CSx0);
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#endif
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}
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void audio_driver_stop() {
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#ifdef AUDIO1_PIN_SET
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channel_1_stop();
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#endif
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#ifdef AUDIO2_PIN_SET
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channel_2_stop();
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#endif
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}
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void audio_driver_start(void) {
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#ifdef AUDIO1_PIN_SET
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channel_1_start();
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if (playing_note) {
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channel_1_set_frequency(audio_get_processed_frequency(0));
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}
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#endif
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#if !defined(AUDIO1_PIN_SET) && defined(AUDIO2_PIN_SET)
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channel_2_start();
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if (playing_note) {
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channel_2_set_frequency(audio_get_processed_frequency(0));
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}
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#endif
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}
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static volatile uint32_t isr_counter = 0;
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#ifdef AUDIO1_PIN_SET
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ISR(AUDIO1_TIMERx_COMPy_vect) {
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isr_counter++;
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if (isr_counter < channel_1_frequency / (CPU_PRESCALER * 8)) return;
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isr_counter = 0;
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bool state_changed = audio_update_state();
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if (!playing_note && !playing_melody) {
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channel_1_stop();
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# ifdef AUDIO2_PIN_SET
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channel_2_stop();
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# endif
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return;
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}
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if (state_changed) {
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channel_1_set_frequency(audio_get_processed_frequency(0));
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# ifdef AUDIO2_PIN_SET
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if (audio_get_number_of_active_tones() > 1) {
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channel_2_set_frequency(audio_get_processed_frequency(1));
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} else {
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channel_2_stop();
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}
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# endif
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}
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}
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#endif
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#if !defined(AUDIO1_PIN_SET) && defined(AUDIO2_PIN_SET)
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ISR(AUDIO2_TIMERx_COMPy_vect) {
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isr_counter++;
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if (isr_counter < channel_2_frequency / (CPU_PRESCALER * 8)) return;
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isr_counter = 0;
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bool state_changed = audio_update_state();
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if (!playing_note && !playing_melody) {
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channel_2_stop();
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return;
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}
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if (state_changed) {
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channel_2_set_frequency(audio_get_processed_frequency(0));
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}
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}
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#endif
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@ -1,126 +0,0 @@
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/* Copyright 2019 Jack Humbert
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* Copyright 2020 JohSchneider
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*
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* 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
|
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* (at your option) any later version.
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*
|
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* This program is distributed in the hope that it will be useful,
|
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* 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.
|
||||
*
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* You should have received a copy of the GNU General Public License
|
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*/
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#pragma once
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#ifndef A4
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# define A4 PAL_LINE(GPIOA, 4)
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#endif
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#ifndef A5
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# define A5 PAL_LINE(GPIOA, 5)
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#endif
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/**
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* Size of the dac_buffer arrays. All must be the same size.
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*/
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#define AUDIO_DAC_BUFFER_SIZE 256U
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/**
|
||||
* 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);
|
||||
|
|
@ -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 <http://www.gnu.org/licenses/>.
|
||||
*/
|
||||
|
||||
#include "audio.h"
|
||||
#include <ch.h>
|
||||
#include <hal.h>
|
||||
|
||||
/*
|
||||
Audio Driver: DAC
|
||||
|
||||
which utilizes the dac unit many STM32 are equipped with, to output a modulated waveform from samples stored in the dac_buffer_* array who are passed to the hardware through DMA
|
||||
|
||||
it is also possible to have a custom sample-LUT by implementing/overriding 'dac_value_generate'
|
||||
|
||||
this driver allows for multiple simultaneous tones to be played through one single channel by doing additive wave-synthesis
|
||||
*/
|
||||
|
||||
#if !defined(AUDIO_PIN)
|
||||
# error "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under 'ARM (DAC additive)' for available options."
|
||||
#endif
|
||||
#if defined(AUDIO_PIN_ALT) && !defined(AUDIO_PIN_ALT_AS_NEGATIVE)
|
||||
# pragma message "Audio feature: AUDIO_PIN_ALT set, but not AUDIO_PIN_ALT_AS_NEGATIVE - pin will be left unused; audio might still work though."
|
||||
#endif
|
||||
|
||||
#if !defined(AUDIO_PIN_ALT)
|
||||
// no ALT pin defined is valid, but the c-ifs below need some value set
|
||||
# define AUDIO_PIN_ALT PAL_NOLINE
|
||||
#endif
|
||||
|
||||
#if !defined(AUDIO_DAC_SAMPLE_WAVEFORM_SINE) && !defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE) && !defined(AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE) && !defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID)
|
||||
# define AUDIO_DAC_SAMPLE_WAVEFORM_SINE
|
||||
#endif
|
||||
|
||||
#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_SINE
|
||||
/* one full sine wave over [0,2*pi], but shifted up one amplitude and left pi/4; for the samples to start at 0
|
||||
*/
|
||||
static const dacsample_t dac_buffer_sine[AUDIO_DAC_BUFFER_SIZE] = {
|
||||
// 256 values, max 4095
|
||||
0x0, 0x1, 0x2, 0x6, 0xa, 0xf, 0x16, 0x1e, 0x27, 0x32, 0x3d, 0x4a, 0x58, 0x67, 0x78, 0x89, 0x9c, 0xb0, 0xc5, 0xdb, 0xf2, 0x10a, 0x123, 0x13e, 0x159, 0x175, 0x193, 0x1b1, 0x1d1, 0x1f1, 0x212, 0x235, 0x258, 0x27c, 0x2a0, 0x2c6, 0x2ed, 0x314, 0x33c, 0x365, 0x38e, 0x3b8, 0x3e3, 0x40e, 0x43a, 0x467, 0x494, 0x4c2, 0x4f0, 0x51f, 0x54e, 0x57d, 0x5ad, 0x5dd, 0x60e, 0x63f, 0x670, 0x6a1, 0x6d3, 0x705, 0x737, 0x769, 0x79b, 0x7cd, 0x800, 0x832, 0x864, 0x896, 0x8c8, 0x8fa, 0x92c, 0x95e, 0x98f, 0x9c0, 0x9f1, 0xa22, 0xa52, 0xa82, 0xab1, 0xae0, 0xb0f, 0xb3d, 0xb6b, 0xb98, 0xbc5, 0xbf1, 0xc1c, 0xc47, 0xc71, 0xc9a, 0xcc3, 0xceb, 0xd12, 0xd39, 0xd5f, 0xd83, 0xda7, 0xdca, 0xded, 0xe0e, 0xe2e, 0xe4e, 0xe6c, 0xe8a, 0xea6, 0xec1, 0xedc, 0xef5, 0xf0d, 0xf24, 0xf3a, 0xf4f, 0xf63, 0xf76, 0xf87, 0xf98, 0xfa7, 0xfb5, 0xfc2, 0xfcd, 0xfd8, 0xfe1, 0xfe9, 0xff0, 0xff5, 0xff9, 0xffd, 0xffe,
|
||||
0xfff, 0xffe, 0xffd, 0xff9, 0xff5, 0xff0, 0xfe9, 0xfe1, 0xfd8, 0xfcd, 0xfc2, 0xfb5, 0xfa7, 0xf98, 0xf87, 0xf76, 0xf63, 0xf4f, 0xf3a, 0xf24, 0xf0d, 0xef5, 0xedc, 0xec1, 0xea6, 0xe8a, 0xe6c, 0xe4e, 0xe2e, 0xe0e, 0xded, 0xdca, 0xda7, 0xd83, 0xd5f, 0xd39, 0xd12, 0xceb, 0xcc3, 0xc9a, 0xc71, 0xc47, 0xc1c, 0xbf1, 0xbc5, 0xb98, 0xb6b, 0xb3d, 0xb0f, 0xae0, 0xab1, 0xa82, 0xa52, 0xa22, 0x9f1, 0x9c0, 0x98f, 0x95e, 0x92c, 0x8fa, 0x8c8, 0x896, 0x864, 0x832, 0x800, 0x7cd, 0x79b, 0x769, 0x737, 0x705, 0x6d3, 0x6a1, 0x670, 0x63f, 0x60e, 0x5dd, 0x5ad, 0x57d, 0x54e, 0x51f, 0x4f0, 0x4c2, 0x494, 0x467, 0x43a, 0x40e, 0x3e3, 0x3b8, 0x38e, 0x365, 0x33c, 0x314, 0x2ed, 0x2c6, 0x2a0, 0x27c, 0x258, 0x235, 0x212, 0x1f1, 0x1d1, 0x1b1, 0x193, 0x175, 0x159, 0x13e, 0x123, 0x10a, 0xf2, 0xdb, 0xc5, 0xb0, 0x9c, 0x89, 0x78, 0x67, 0x58, 0x4a, 0x3d, 0x32, 0x27, 0x1e, 0x16, 0xf, 0xa, 0x6, 0x2, 0x1};
|
||||
#endif // AUDIO_DAC_SAMPLE_WAVEFORM_SINE
|
||||
#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE
|
||||
static const dacsample_t dac_buffer_triangle[AUDIO_DAC_BUFFER_SIZE] = {
|
||||
// 256 values, max 4095
|
||||
0x0, 0x20, 0x40, 0x60, 0x80, 0xa0, 0xc0, 0xe0, 0x100, 0x120, 0x140, 0x160, 0x180, 0x1a0, 0x1c0, 0x1e0, 0x200, 0x220, 0x240, 0x260, 0x280, 0x2a0, 0x2c0, 0x2e0, 0x300, 0x320, 0x340, 0x360, 0x380, 0x3a0, 0x3c0, 0x3e0, 0x400, 0x420, 0x440, 0x460, 0x480, 0x4a0, 0x4c0, 0x4e0, 0x500, 0x520, 0x540, 0x560, 0x580, 0x5a0, 0x5c0, 0x5e0, 0x600, 0x620, 0x640, 0x660, 0x680, 0x6a0, 0x6c0, 0x6e0, 0x700, 0x720, 0x740, 0x760, 0x780, 0x7a0, 0x7c0, 0x7e0, 0x800, 0x81f, 0x83f, 0x85f, 0x87f, 0x89f, 0x8bf, 0x8df, 0x8ff, 0x91f, 0x93f, 0x95f, 0x97f, 0x99f, 0x9bf, 0x9df, 0x9ff, 0xa1f, 0xa3f, 0xa5f, 0xa7f, 0xa9f, 0xabf, 0xadf, 0xaff, 0xb1f, 0xb3f, 0xb5f, 0xb7f, 0xb9f, 0xbbf, 0xbdf, 0xbff, 0xc1f, 0xc3f, 0xc5f, 0xc7f, 0xc9f, 0xcbf, 0xcdf, 0xcff, 0xd1f, 0xd3f, 0xd5f, 0xd7f, 0xd9f, 0xdbf, 0xddf, 0xdff, 0xe1f, 0xe3f, 0xe5f, 0xe7f, 0xe9f, 0xebf, 0xedf, 0xeff, 0xf1f, 0xf3f, 0xf5f, 0xf7f, 0xf9f, 0xfbf, 0xfdf,
|
||||
0xfff, 0xfdf, 0xfbf, 0xf9f, 0xf7f, 0xf5f, 0xf3f, 0xf1f, 0xeff, 0xedf, 0xebf, 0xe9f, 0xe7f, 0xe5f, 0xe3f, 0xe1f, 0xdff, 0xddf, 0xdbf, 0xd9f, 0xd7f, 0xd5f, 0xd3f, 0xd1f, 0xcff, 0xcdf, 0xcbf, 0xc9f, 0xc7f, 0xc5f, 0xc3f, 0xc1f, 0xbff, 0xbdf, 0xbbf, 0xb9f, 0xb7f, 0xb5f, 0xb3f, 0xb1f, 0xaff, 0xadf, 0xabf, 0xa9f, 0xa7f, 0xa5f, 0xa3f, 0xa1f, 0x9ff, 0x9df, 0x9bf, 0x99f, 0x97f, 0x95f, 0x93f, 0x91f, 0x8ff, 0x8df, 0x8bf, 0x89f, 0x87f, 0x85f, 0x83f, 0x81f, 0x800, 0x7e0, 0x7c0, 0x7a0, 0x780, 0x760, 0x740, 0x720, 0x700, 0x6e0, 0x6c0, 0x6a0, 0x680, 0x660, 0x640, 0x620, 0x600, 0x5e0, 0x5c0, 0x5a0, 0x580, 0x560, 0x540, 0x520, 0x500, 0x4e0, 0x4c0, 0x4a0, 0x480, 0x460, 0x440, 0x420, 0x400, 0x3e0, 0x3c0, 0x3a0, 0x380, 0x360, 0x340, 0x320, 0x300, 0x2e0, 0x2c0, 0x2a0, 0x280, 0x260, 0x240, 0x220, 0x200, 0x1e0, 0x1c0, 0x1a0, 0x180, 0x160, 0x140, 0x120, 0x100, 0xe0, 0xc0, 0xa0, 0x80, 0x60, 0x40, 0x20};
|
||||
#endif // AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE
|
||||
#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE
|
||||
static const dacsample_t dac_buffer_square[AUDIO_DAC_BUFFER_SIZE] = {
|
||||
[0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1] = 0, // first and
|
||||
[AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = AUDIO_DAC_SAMPLE_MAX, // second half
|
||||
};
|
||||
#endif // AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE
|
||||
/*
|
||||
// four steps: 0, 1/3, 2/3 and 1
|
||||
static const dacsample_t dac_buffer_staircase[AUDIO_DAC_BUFFER_SIZE] = {
|
||||
[0 ... AUDIO_DAC_BUFFER_SIZE/3 -1 ] = 0,
|
||||
[AUDIO_DAC_BUFFER_SIZE / 4 ... AUDIO_DAC_BUFFER_SIZE / 2 -1 ] = AUDIO_DAC_SAMPLE_MAX / 3,
|
||||
[AUDIO_DAC_BUFFER_SIZE / 2 ... 3 * AUDIO_DAC_BUFFER_SIZE / 4 -1 ] = 2 * AUDIO_DAC_SAMPLE_MAX / 3,
|
||||
[3 * AUDIO_DAC_BUFFER_SIZE / 4 ... AUDIO_DAC_BUFFER_SIZE -1 ] = AUDIO_DAC_SAMPLE_MAX,
|
||||
}
|
||||
*/
|
||||
#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID
|
||||
static const dacsample_t dac_buffer_trapezoid[AUDIO_DAC_BUFFER_SIZE] = {0x0, 0x1f, 0x7f, 0xdf, 0x13f, 0x19f, 0x1ff, 0x25f, 0x2bf, 0x31f, 0x37f, 0x3df, 0x43f, 0x49f, 0x4ff, 0x55f, 0x5bf, 0x61f, 0x67f, 0x6df, 0x73f, 0x79f, 0x7ff, 0x85f, 0x8bf, 0x91f, 0x97f, 0x9df, 0xa3f, 0xa9f, 0xaff, 0xb5f, 0xbbf, 0xc1f, 0xc7f, 0xcdf, 0xd3f, 0xd9f, 0xdff, 0xe5f, 0xebf, 0xf1f, 0xf7f, 0xfdf, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff,
|
||||
0xfff, 0xfdf, 0xf7f, 0xf1f, 0xebf, 0xe5f, 0xdff, 0xd9f, 0xd3f, 0xcdf, 0xc7f, 0xc1f, 0xbbf, 0xb5f, 0xaff, 0xa9f, 0xa3f, 0x9df, 0x97f, 0x91f, 0x8bf, 0x85f, 0x7ff, 0x79f, 0x73f, 0x6df, 0x67f, 0x61f, 0x5bf, 0x55f, 0x4ff, 0x49f, 0x43f, 0x3df, 0x37f, 0x31f, 0x2bf, 0x25f, 0x1ff, 0x19f, 0x13f, 0xdf, 0x7f, 0x1f, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0};
|
||||
#endif // AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID
|
||||
|
||||
static dacsample_t dac_buffer_empty[AUDIO_DAC_BUFFER_SIZE] = {AUDIO_DAC_OFF_VALUE};
|
||||
|
||||
/* keep track of the sample position for for each frequency */
|
||||
static float dac_if[AUDIO_MAX_SIMULTANEOUS_TONES] = {0.0};
|
||||
|
||||
static float active_tones_snapshot[AUDIO_MAX_SIMULTANEOUS_TONES] = {0, 0};
|
||||
static uint8_t active_tones_snapshot_length = 0;
|
||||
|
||||
typedef enum {
|
||||
OUTPUT_SHOULD_START,
|
||||
OUTPUT_RUN_NORMALLY,
|
||||
// path 1: wait for zero, then change/update active tones
|
||||
OUTPUT_TONES_CHANGED,
|
||||
OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE,
|
||||
// path 2: hardware should stop, wait for zero then turn output off = stop the timer
|
||||
OUTPUT_SHOULD_STOP,
|
||||
OUTPUT_REACHED_ZERO_BEFORE_OFF,
|
||||
OUTPUT_OFF,
|
||||
OUTPUT_OFF_1,
|
||||
OUTPUT_OFF_2, // trailing off: giving the DAC two more conversion cycles until the AUDIO_DAC_OFF_VALUE reaches the output, then turn the timer off, which leaves the output at that level
|
||||
number_of_output_states
|
||||
} output_states_t;
|
||||
output_states_t state = OUTPUT_OFF_2;
|
||||
|
||||
/**
|
||||
* Generation of the waveform being passed to the callback. Declared weak so users
|
||||
* can override it with their own wave-forms/noises.
|
||||
*/
|
||||
__attribute__((weak)) uint16_t dac_value_generate(void) {
|
||||
// DAC is running/asking for values but snapshot length is zero -> must be playing a pause
|
||||
if (active_tones_snapshot_length == 0) {
|
||||
return AUDIO_DAC_OFF_VALUE;
|
||||
}
|
||||
|
||||
/* doing additive wave synthesis over all currently playing tones = adding up
|
||||
* sine-wave-samples for each frequency, scaled by the number of active tones
|
||||
*/
|
||||
uint16_t value = 0;
|
||||
float frequency = 0.0f;
|
||||
|
||||
for (uint8_t i = 0; i < active_tones_snapshot_length; i++) {
|
||||
/* Note: a user implementation does not have to rely on the active_tones_snapshot, but
|
||||
* could directly query the active frequencies through audio_get_processed_frequency */
|
||||
frequency = active_tones_snapshot[i];
|
||||
|
||||
dac_if[i] = dac_if[i] + ((frequency * AUDIO_DAC_BUFFER_SIZE) / AUDIO_DAC_SAMPLE_RATE) * 2 / 3;
|
||||
/*Note: the 2/3 are necessary to get the correct frequencies on the
|
||||
* DAC output (as measured with an oscilloscope), since the gpt
|
||||
* timer runs with 3*AUDIO_DAC_SAMPLE_RATE; and the DAC callback
|
||||
* is called twice per conversion.*/
|
||||
|
||||
dac_if[i] = fmod(dac_if[i], AUDIO_DAC_BUFFER_SIZE);
|
||||
|
||||
// Wavetable generation/lookup
|
||||
uint16_t dac_i = (uint16_t)dac_if[i];
|
||||
|
||||
#if defined(AUDIO_DAC_SAMPLE_WAVEFORM_SINE)
|
||||
value += dac_buffer_sine[dac_i] / active_tones_snapshot_length;
|
||||
#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE)
|
||||
value += dac_buffer_triangle[dac_i] / active_tones_snapshot_length;
|
||||
#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID)
|
||||
value += dac_buffer_trapezoid[dac_i] / active_tones_snapshot_length;
|
||||
#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE)
|
||||
value += dac_buffer_square[dac_i] / active_tones_snapshot_length;
|
||||
#endif
|
||||
/*
|
||||
// SINE
|
||||
value += dac_buffer_sine[dac_i] / active_tones_snapshot_length / 3;
|
||||
// TRIANGLE
|
||||
value += dac_buffer_triangle[dac_i] / active_tones_snapshot_length / 3;
|
||||
// SQUARE
|
||||
value += dac_buffer_square[dac_i] / active_tones_snapshot_length / 3;
|
||||
//NOTE: combination of these three wave-forms is more exemplary - and doesn't sound particularly good :-P
|
||||
*/
|
||||
|
||||
// STAIRS (mostly usefully as test-pattern)
|
||||
// value_avg = dac_buffer_staircase[dac_i] / active_tones_snapshot_length;
|
||||
}
|
||||
|
||||
return value;
|
||||
}
|
||||
|
||||
/**
|
||||
* DAC streaming callback. Does all of the main computing for playing songs.
|
||||
*
|
||||
* Note: chibios calls this CB twice: during the 'half buffer event', and the 'full buffer event'.
|
||||
*/
|
||||
static void dac_end(DACDriver *dacp) {
|
||||
dacsample_t *sample_p = (dacp)->samples;
|
||||
|
||||
// work on the other half of the buffer
|
||||
if (dacIsBufferComplete(dacp)) {
|
||||
sample_p += AUDIO_DAC_BUFFER_SIZE / 2; // 'half_index'
|
||||
}
|
||||
|
||||
for (uint8_t s = 0; s < AUDIO_DAC_BUFFER_SIZE / 2; s++) {
|
||||
if (OUTPUT_OFF <= state) {
|
||||
sample_p[s] = AUDIO_DAC_OFF_VALUE;
|
||||
continue;
|
||||
} else {
|
||||
sample_p[s] = dac_value_generate();
|
||||
}
|
||||
|
||||
/* zero crossing (or approach, whereas zero == DAC_OFF_VALUE, which can be configured to anything from 0 to DAC_SAMPLE_MAX)
|
||||
* ============================*=*========================== AUDIO_DAC_SAMPLE_MAX
|
||||
* * *
|
||||
* * *
|
||||
* ---------------------------------------------------------
|
||||
* * * } AUDIO_DAC_SAMPLE_MAX/100
|
||||
* --------------------------------------------------------- AUDIO_DAC_OFF_VALUE
|
||||
* * * } AUDIO_DAC_SAMPLE_MAX/100
|
||||
* ---------------------------------------------------------
|
||||
* *
|
||||
* * *
|
||||
* * *
|
||||
* =====*=*================================================= 0x0
|
||||
*/
|
||||
if (((sample_p[s] + (AUDIO_DAC_SAMPLE_MAX / 100)) > AUDIO_DAC_OFF_VALUE) && // value approaches from below
|
||||
(sample_p[s] < (AUDIO_DAC_OFF_VALUE + (AUDIO_DAC_SAMPLE_MAX / 100))) // or above
|
||||
) {
|
||||
if ((OUTPUT_SHOULD_START == state) && (active_tones_snapshot_length > 0)) {
|
||||
state = OUTPUT_RUN_NORMALLY;
|
||||
} else if (OUTPUT_TONES_CHANGED == state) {
|
||||
state = OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE;
|
||||
} else if (OUTPUT_SHOULD_STOP == state) {
|
||||
state = OUTPUT_REACHED_ZERO_BEFORE_OFF;
|
||||
}
|
||||
}
|
||||
|
||||
// still 'ramping up', reset the output to OFF_VALUE until the generated values reach that value, to do a smooth handover
|
||||
if (OUTPUT_SHOULD_START == state) {
|
||||
sample_p[s] = AUDIO_DAC_OFF_VALUE;
|
||||
}
|
||||
|
||||
if ((OUTPUT_SHOULD_START == state) || (OUTPUT_REACHED_ZERO_BEFORE_OFF == state) || (OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE == state)) {
|
||||
uint8_t active_tones = MIN(AUDIO_MAX_SIMULTANEOUS_TONES, audio_get_number_of_active_tones());
|
||||
active_tones_snapshot_length = 0;
|
||||
// update the snapshot - once, and only on occasion that something changed;
|
||||
// -> saves cpu cycles (?)
|
||||
for (uint8_t i = 0; i < active_tones; i++) {
|
||||
float freq = audio_get_processed_frequency(i);
|
||||
if (freq > 0) { // disregard 'rest' notes, with valid frequency 0.0f; which would only lower the resulting waveform volume during the additive synthesis step
|
||||
active_tones_snapshot[active_tones_snapshot_length++] = freq;
|
||||
}
|
||||
}
|
||||
|
||||
if ((0 == active_tones_snapshot_length) && (OUTPUT_REACHED_ZERO_BEFORE_OFF == state)) {
|
||||
state = OUTPUT_OFF;
|
||||
}
|
||||
if (OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE == state) {
|
||||
state = OUTPUT_RUN_NORMALLY;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// update audio internal state (note position, current_note, ...)
|
||||
if (audio_update_state()) {
|
||||
if (OUTPUT_SHOULD_STOP != state) {
|
||||
state = OUTPUT_TONES_CHANGED;
|
||||
}
|
||||
}
|
||||
|
||||
if (OUTPUT_OFF <= state) {
|
||||
if (OUTPUT_OFF_2 == state) {
|
||||
// stopping timer6 = stopping the DAC at whatever value it is currently pushing to the output = AUDIO_DAC_OFF_VALUE
|
||||
gptStopTimer(&GPTD6);
|
||||
} else {
|
||||
state++;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
static void dac_error(DACDriver *dacp, dacerror_t err) {
|
||||
(void)dacp;
|
||||
(void)err;
|
||||
|
||||
chSysHalt("DAC failure. halp");
|
||||
}
|
||||
|
||||
static const GPTConfig gpt6cfg1 = {.frequency = AUDIO_DAC_SAMPLE_RATE * 3,
|
||||
.callback = NULL,
|
||||
.cr2 = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event. */
|
||||
.dier = 0U};
|
||||
|
||||
static const DACConfig dac_conf = {.init = AUDIO_DAC_OFF_VALUE, .datamode = DAC_DHRM_12BIT_RIGHT};
|
||||
|
||||
/**
|
||||
* @note The DAC_TRG(0) here selects the Timer 6 TRGO event, which is triggered
|
||||
* on the rising edge after 3 APB1 clock cycles, causing our gpt6cfg1.frequency
|
||||
* to be a third of what we expect.
|
||||
*
|
||||
* Here are all the values for DAC_TRG (TSEL in the ref manual)
|
||||
* TIM15_TRGO 0b011
|
||||
* TIM2_TRGO 0b100
|
||||
* TIM3_TRGO 0b001
|
||||
* TIM6_TRGO 0b000
|
||||
* TIM7_TRGO 0b010
|
||||
* EXTI9 0b110
|
||||
* SWTRIG 0b111
|
||||
*/
|
||||
static const DACConversionGroup dac_conv_cfg = {.num_channels = 1U, .end_cb = dac_end, .error_cb = dac_error, .trigger = DAC_TRG(0b000)};
|
||||
|
||||
void audio_driver_initialize() {
|
||||
if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
|
||||
palSetLineMode(A4, PAL_MODE_INPUT_ANALOG);
|
||||
dacStart(&DACD1, &dac_conf);
|
||||
}
|
||||
if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
|
||||
palSetLineMode(A5, PAL_MODE_INPUT_ANALOG);
|
||||
dacStart(&DACD2, &dac_conf);
|
||||
}
|
||||
|
||||
/* enable the output buffer, to directly drive external loads with no additional circuitry
|
||||
*
|
||||
* see: AN4566 Application note: Extending the DAC performance of STM32 microcontrollers
|
||||
* Note: Buffer-Off bit -> has to be set 0 to enable the output buffer
|
||||
* Note: enabling the output buffer imparts an additional dc-offset of a couple mV
|
||||
*
|
||||
* this is done here, reaching directly into the stm32 registers since chibios has not implemented BOFF handling yet
|
||||
* (see: chibios/os/hal/ports/STM32/todo.txt '- BOFF handling in DACv1.'
|
||||
*/
|
||||
DACD1.params->dac->CR &= ~DAC_CR_BOFF1;
|
||||
DACD2.params->dac->CR &= ~DAC_CR_BOFF2;
|
||||
|
||||
if (AUDIO_PIN == A4) {
|
||||
dacStartConversion(&DACD1, &dac_conv_cfg, dac_buffer_empty, AUDIO_DAC_BUFFER_SIZE);
|
||||
} else if (AUDIO_PIN == A5) {
|
||||
dacStartConversion(&DACD2, &dac_conv_cfg, dac_buffer_empty, AUDIO_DAC_BUFFER_SIZE);
|
||||
}
|
||||
|
||||
// no inverted/out-of-phase waveform (yet?), only pulling AUDIO_PIN_ALT to AUDIO_DAC_OFF_VALUE
|
||||
#if defined(AUDIO_PIN_ALT_AS_NEGATIVE)
|
||||
if (AUDIO_PIN_ALT == A4) {
|
||||
dacPutChannelX(&DACD1, 0, AUDIO_DAC_OFF_VALUE);
|
||||
} else if (AUDIO_PIN_ALT == A5) {
|
||||
dacPutChannelX(&DACD2, 0, AUDIO_DAC_OFF_VALUE);
|
||||
}
|
||||
#endif
|
||||
|
||||
gptStart(&GPTD6, &gpt6cfg1);
|
||||
}
|
||||
|
||||
void audio_driver_stop(void) { state = OUTPUT_SHOULD_STOP; }
|
||||
|
||||
void audio_driver_start(void) {
|
||||
gptStartContinuous(&GPTD6, 2U);
|
||||
|
||||
for (uint8_t i = 0; i < AUDIO_MAX_SIMULTANEOUS_TONES; i++) {
|
||||
dac_if[i] = 0.0f;
|
||||
active_tones_snapshot[i] = 0.0f;
|
||||
}
|
||||
active_tones_snapshot_length = 0;
|
||||
state = OUTPUT_SHOULD_START;
|
||||
}
|
||||
|
|
@ -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 <http://www.gnu.org/licenses/>.
|
||||
*/
|
||||
|
||||
#include "audio.h"
|
||||
#include "ch.h"
|
||||
#include "hal.h"
|
||||
|
||||
/*
|
||||
Audio Driver: DAC
|
||||
|
||||
which utilizes both channels of the DAC unit many STM32 are equipped with to output a modulated square-wave, from precomputed samples stored in a buffer, which is passed to the hardware through DMA
|
||||
|
||||
this driver can either be used to drive to separate speakers, wired to A4+Gnd and A5+Gnd, which allows two tones to be played simultaneously
|
||||
OR
|
||||
one speaker wired to A4+A5 with the AUDIO_PIN_ALT_AS_NEGATIVE define set - see docs/feature_audio
|
||||
|
||||
*/
|
||||
|
||||
#if !defined(AUDIO_PIN)
|
||||
# pragma message "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under 'ARM (DAC basic)' for available options."
|
||||
// TODO: make this an 'error' instead; go through a breaking change, and add AUDIO_PIN A5 to all keyboards currently using AUDIO on STM32 based boards? - for now: set the define here
|
||||
# define AUDIO_PIN A5
|
||||
#endif
|
||||
// check configuration for ONE speaker, connected to both DAC pins
|
||||
#if defined(AUDIO_PIN_ALT_AS_NEGATIVE) && !defined(AUDIO_PIN_ALT)
|
||||
# error "Audio feature: AUDIO_PIN_ALT_AS_NEGATIVE set, but no pin configured as AUDIO_PIN_ALT"
|
||||
#endif
|
||||
|
||||
#ifndef AUDIO_PIN_ALT
|
||||
// no ALT pin defined is valid, but the c-ifs below need some value set
|
||||
# define AUDIO_PIN_ALT -1
|
||||
#endif
|
||||
|
||||
#if !defined(AUDIO_STATE_TIMER)
|
||||
# define AUDIO_STATE_TIMER GPTD8
|
||||
#endif
|
||||
|
||||
// square-wave
|
||||
static const dacsample_t dac_buffer_1[AUDIO_DAC_BUFFER_SIZE] = {
|
||||
// First half is max, second half is 0
|
||||
[0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1] = AUDIO_DAC_SAMPLE_MAX,
|
||||
[AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = 0,
|
||||
};
|
||||
|
||||
// square-wave
|
||||
static const dacsample_t dac_buffer_2[AUDIO_DAC_BUFFER_SIZE] = {
|
||||
// opposite of dac_buffer above
|
||||
[0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1] = 0,
|
||||
[AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = AUDIO_DAC_SAMPLE_MAX,
|
||||
};
|
||||
|
||||
GPTConfig gpt6cfg1 = {.frequency = AUDIO_DAC_SAMPLE_RATE,
|
||||
.callback = NULL,
|
||||
.cr2 = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event. */
|
||||
.dier = 0U};
|
||||
GPTConfig gpt7cfg1 = {.frequency = AUDIO_DAC_SAMPLE_RATE,
|
||||
.callback = NULL,
|
||||
.cr2 = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event. */
|
||||
.dier = 0U};
|
||||
|
||||
static void gpt_audio_state_cb(GPTDriver *gptp);
|
||||
GPTConfig gptStateUpdateCfg = {.frequency = 10,
|
||||
.callback = gpt_audio_state_cb,
|
||||
.cr2 = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event. */
|
||||
.dier = 0U};
|
||||
|
||||
static const DACConfig dac_conf_ch1 = {.init = AUDIO_DAC_OFF_VALUE, .datamode = DAC_DHRM_12BIT_RIGHT};
|
||||
static const DACConfig dac_conf_ch2 = {.init = AUDIO_DAC_OFF_VALUE, .datamode = DAC_DHRM_12BIT_RIGHT};
|
||||
|
||||
/**
|
||||
* @note The DAC_TRG(0) here selects the Timer 6 TRGO event, which is triggered
|
||||
* on the rising edge after 3 APB1 clock cycles, causing our gpt6cfg1.frequency
|
||||
* to be a third of what we expect.
|
||||
*
|
||||
* Here are all the values for DAC_TRG (TSEL in the ref manual)
|
||||
* TIM15_TRGO 0b011
|
||||
* TIM2_TRGO 0b100
|
||||
* TIM3_TRGO 0b001
|
||||
* TIM6_TRGO 0b000
|
||||
* TIM7_TRGO 0b010
|
||||
* EXTI9 0b110
|
||||
* SWTRIG 0b111
|
||||
*/
|
||||
static const DACConversionGroup dac_conv_grp_ch1 = {.num_channels = 1U, .trigger = DAC_TRG(0b000)};
|
||||
static const DACConversionGroup dac_conv_grp_ch2 = {.num_channels = 1U, .trigger = DAC_TRG(0b010)};
|
||||
|
||||
void channel_1_start(void) {
|
||||
gptStart(&GPTD6, &gpt6cfg1);
|
||||
gptStartContinuous(&GPTD6, 2U);
|
||||
palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG);
|
||||
}
|
||||
|
||||
void channel_1_stop(void) {
|
||||
gptStopTimer(&GPTD6);
|
||||
palSetPadMode(GPIOA, 4, PAL_MODE_OUTPUT_PUSHPULL);
|
||||
palSetPad(GPIOA, 4);
|
||||
}
|
||||
|
||||
static float channel_1_frequency = 0.0f;
|
||||
void channel_1_set_frequency(float freq) {
|
||||
channel_1_frequency = freq;
|
||||
|
||||
channel_1_stop();
|
||||
if (freq <= 0.0) // a pause/rest has freq=0
|
||||
return;
|
||||
|
||||
gpt6cfg1.frequency = 2 * freq * AUDIO_DAC_BUFFER_SIZE;
|
||||
channel_1_start();
|
||||
}
|
||||
float channel_1_get_frequency(void) { return channel_1_frequency; }
|
||||
|
||||
void channel_2_start(void) {
|
||||
gptStart(&GPTD7, &gpt7cfg1);
|
||||
gptStartContinuous(&GPTD7, 2U);
|
||||
palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG);
|
||||
}
|
||||
|
||||
void channel_2_stop(void) {
|
||||
gptStopTimer(&GPTD7);
|
||||
palSetPadMode(GPIOA, 5, PAL_MODE_OUTPUT_PUSHPULL);
|
||||
palSetPad(GPIOA, 5);
|
||||
}
|
||||
|
||||
static float channel_2_frequency = 0.0f;
|
||||
void channel_2_set_frequency(float freq) {
|
||||
channel_2_frequency = freq;
|
||||
|
||||
channel_2_stop();
|
||||
if (freq <= 0.0) // a pause/rest has freq=0
|
||||
return;
|
||||
|
||||
gpt7cfg1.frequency = 2 * freq * AUDIO_DAC_BUFFER_SIZE;
|
||||
channel_2_start();
|
||||
}
|
||||
float channel_2_get_frequency(void) { return channel_2_frequency; }
|
||||
|
||||
static void gpt_audio_state_cb(GPTDriver *gptp) {
|
||||
if (audio_update_state()) {
|
||||
#if defined(AUDIO_PIN_ALT_AS_NEGATIVE)
|
||||
// one piezo/speaker connected to both audio pins, the generated square-waves are inverted
|
||||
channel_1_set_frequency(audio_get_processed_frequency(0));
|
||||
channel_2_set_frequency(audio_get_processed_frequency(0));
|
||||
|
||||
#else // two separate audio outputs/speakers
|
||||
// primary speaker on A4, optional secondary on A5
|
||||
if (AUDIO_PIN == A4) {
|
||||
channel_1_set_frequency(audio_get_processed_frequency(0));
|
||||
if (AUDIO_PIN_ALT == A5) {
|
||||
if (audio_get_number_of_active_tones() > 1) {
|
||||
channel_2_set_frequency(audio_get_processed_frequency(1));
|
||||
} else {
|
||||
channel_2_stop();
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// primary speaker on A5, optional secondary on A4
|
||||
if (AUDIO_PIN == A5) {
|
||||
channel_2_set_frequency(audio_get_processed_frequency(0));
|
||||
if (AUDIO_PIN_ALT == A4) {
|
||||
if (audio_get_number_of_active_tones() > 1) {
|
||||
channel_1_set_frequency(audio_get_processed_frequency(1));
|
||||
} else {
|
||||
channel_1_stop();
|
||||
}
|
||||
}
|
||||
}
|
||||
#endif
|
||||
}
|
||||
}
|
||||
|
||||
void audio_driver_initialize() {
|
||||
if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
|
||||
palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG);
|
||||
dacStart(&DACD1, &dac_conf_ch1);
|
||||
|
||||
// initial setup of the dac-triggering timer is still required, even
|
||||
// though it gets reconfigured and restarted later on
|
||||
gptStart(&GPTD6, &gpt6cfg1);
|
||||
}
|
||||
|
||||
if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
|
||||
palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG);
|
||||
dacStart(&DACD2, &dac_conf_ch2);
|
||||
|
||||
gptStart(&GPTD7, &gpt7cfg1);
|
||||
}
|
||||
|
||||
/* enable the output buffer, to directly drive external loads with no additional circuitry
|
||||
*
|
||||
* see: AN4566 Application note: Extending the DAC performance of STM32 microcontrollers
|
||||
* Note: Buffer-Off bit -> has to be set 0 to enable the output buffer
|
||||
* Note: enabling the output buffer imparts an additional dc-offset of a couple mV
|
||||
*
|
||||
* this is done here, reaching directly into the stm32 registers since chibios has not implemented BOFF handling yet
|
||||
* (see: chibios/os/hal/ports/STM32/todo.txt '- BOFF handling in DACv1.'
|
||||
*/
|
||||
DACD1.params->dac->CR &= ~DAC_CR_BOFF1;
|
||||
DACD2.params->dac->CR &= ~DAC_CR_BOFF2;
|
||||
|
||||
// start state-updater
|
||||
gptStart(&AUDIO_STATE_TIMER, &gptStateUpdateCfg);
|
||||
}
|
||||
|
||||
void audio_driver_stop(void) {
|
||||
if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
|
||||
gptStopTimer(&GPTD6);
|
||||
|
||||
// stop the ongoing conversion and put the output in a known state
|
||||
dacStopConversion(&DACD1);
|
||||
dacPutChannelX(&DACD1, 0, AUDIO_DAC_OFF_VALUE);
|
||||
}
|
||||
|
||||
if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
|
||||
gptStopTimer(&GPTD7);
|
||||
|
||||
dacStopConversion(&DACD2);
|
||||
dacPutChannelX(&DACD2, 0, AUDIO_DAC_OFF_VALUE);
|
||||
}
|
||||
gptStopTimer(&AUDIO_STATE_TIMER);
|
||||
}
|
||||
|
||||
void audio_driver_start(void) {
|
||||
if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
|
||||
dacStartConversion(&DACD1, &dac_conv_grp_ch1, (dacsample_t *)dac_buffer_1, AUDIO_DAC_BUFFER_SIZE);
|
||||
}
|
||||
if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
|
||||
dacStartConversion(&DACD2, &dac_conv_grp_ch2, (dacsample_t *)dac_buffer_2, AUDIO_DAC_BUFFER_SIZE);
|
||||
}
|
||||
gptStartContinuous(&AUDIO_STATE_TIMER, 2U);
|
||||
}
|
||||
|
|
@ -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 <http://www.gnu.org/licenses/>.
|
||||
*/
|
||||
#pragma once
|
||||
|
||||
#if !defined(AUDIO_PWM_DRIVER)
|
||||
// NOTE: Timer2 seems to be used otherwise in QMK, otherwise we could default to A5 (= TIM2_CH1, with PWMD2 and alternate-function(1))
|
||||
# define AUDIO_PWM_DRIVER PWMD1
|
||||
#endif
|
||||
|
||||
#if !defined(AUDIO_PWM_CHANNEL)
|
||||
// NOTE: sticking to the STM data-sheet numbering: TIMxCH1 to TIMxCH4
|
||||
// default: STM32F303CC PA8+TIM1_CH1 -> 1
|
||||
# define AUDIO_PWM_CHANNEL 1
|
||||
#endif
|
||||
|
||||
#if !defined(AUDIO_PWM_PAL_MODE)
|
||||
// pin-alternate function: see the data-sheet for which pin needs what AF to connect to TIMx_CHy
|
||||
// default: STM32F303CC PA8+TIM1_CH1 -> 6
|
||||
# define AUDIO_PWM_PAL_MODE 6
|
||||
#endif
|
||||
|
||||
#if !defined(AUDIO_STATE_TIMER)
|
||||
// timer used to trigger updates in the audio-system, configured/enabled in chibios mcuconf.
|
||||
// Tim6 is the default for "larger" STMs, smaller ones might not have this one (enabled) and need to switch to a different one (e.g.: STM32F103 has only Tim1-Tim4)
|
||||
# define AUDIO_STATE_TIMER GPTD6
|
||||
#endif
|
||||
|
|
@ -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 <http://www.gnu.org/licenses/>.
|
||||
*/
|
||||
|
||||
/*
|
||||
Audio Driver: PWM
|
||||
|
||||
the duty-cycle is always kept at 50%, and the pwm-period is adjusted to match the frequency of a note to be played back.
|
||||
|
||||
this driver uses the chibios-PWM system to produce a square-wave on specific output pins that are connected to the PWM hardware.
|
||||
The hardware directly toggles the pin via its alternate function. see your MCUs data-sheet for which pin can be driven by what timer - looking for TIMx_CHy and the corresponding alternate function.
|
||||
|
||||
*/
|
||||
|
||||
#include "audio.h"
|
||||
#include "ch.h"
|
||||
#include "hal.h"
|
||||
|
||||
#if !defined(AUDIO_PIN)
|
||||
# error "Audio feature enabled, but no pin selected - see docs/feature_audio under the ARM PWM settings"
|
||||
#endif
|
||||
|
||||
extern bool playing_note;
|
||||
extern bool playing_melody;
|
||||
extern uint8_t note_timbre;
|
||||
|
||||
static PWMConfig pwmCFG = {
|
||||
.frequency = 100000, /* PWM clock frequency */
|
||||
// CHIBIOS-BUG? can't set the initial period to <2, or the pwm (hard or software) takes ~130ms with .frequency=500000 for a pwmChangePeriod to take effect; with no output=silence in the meantime
|
||||
.period = 2, /* initial PWM period (in ticks) 1S (1/10kHz=0.1mS 0.1ms*10000 ticks=1S) */
|
||||
.callback = NULL, /* no callback, the hardware directly toggles the pin */
|
||||
.channels =
|
||||
{
|
||||
#if AUDIO_PWM_CHANNEL == 4
|
||||
{PWM_OUTPUT_DISABLED, NULL}, /* channel 0 -> TIMx_CH1 */
|
||||
{PWM_OUTPUT_DISABLED, NULL}, /* channel 1 -> TIMx_CH2 */
|
||||
{PWM_OUTPUT_DISABLED, NULL}, /* channel 2 -> TIMx_CH3 */
|
||||
{PWM_OUTPUT_ACTIVE_HIGH, NULL} /* channel 3 -> TIMx_CH4 */
|
||||
#elif AUDIO_PWM_CHANNEL == 3
|
||||
{PWM_OUTPUT_DISABLED, NULL},
|
||||
{PWM_OUTPUT_DISABLED, NULL},
|
||||
{PWM_OUTPUT_ACTIVE_HIGH, NULL}, /* TIMx_CH3 */
|
||||
{PWM_OUTPUT_DISABLED, NULL}
|
||||
#elif AUDIO_PWM_CHANNEL == 2
|
||||
{PWM_OUTPUT_DISABLED, NULL},
|
||||
{PWM_OUTPUT_ACTIVE_HIGH, NULL}, /* TIMx_CH2 */
|
||||
{PWM_OUTPUT_DISABLED, NULL},
|
||||
{PWM_OUTPUT_DISABLED, NULL}
|
||||
#else /*fallback to CH1 */
|
||||
{PWM_OUTPUT_ACTIVE_HIGH, NULL}, /* TIMx_CH1 */
|
||||
{PWM_OUTPUT_DISABLED, NULL},
|
||||
{PWM_OUTPUT_DISABLED, NULL},
|
||||
{PWM_OUTPUT_DISABLED, NULL}
|
||||
#endif
|
||||
},
|
||||
};
|
||||
|
||||
static float channel_1_frequency = 0.0f;
|
||||
void channel_1_set_frequency(float freq) {
|
||||
channel_1_frequency = freq;
|
||||
|
||||
if (freq <= 0.0) // a pause/rest has freq=0
|
||||
return;
|
||||
|
||||
pwmcnt_t period = (pwmCFG.frequency / freq);
|
||||
pwmChangePeriod(&AUDIO_PWM_DRIVER, period);
|
||||
pwmEnableChannel(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1,
|
||||
// adjust the duty-cycle so that the output is for 'note_timbre' duration HIGH
|
||||
PWM_PERCENTAGE_TO_WIDTH(&AUDIO_PWM_DRIVER, (100 - note_timbre) * 100));
|
||||
}
|
||||
|
||||
float channel_1_get_frequency(void) { return channel_1_frequency; }
|
||||
|
||||
void channel_1_start(void) {
|
||||
pwmStop(&AUDIO_PWM_DRIVER);
|
||||
pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
|
||||
}
|
||||
|
||||
void channel_1_stop(void) { pwmStop(&AUDIO_PWM_DRIVER); }
|
||||
|
||||
static void gpt_callback(GPTDriver *gptp);
|
||||
GPTConfig gptCFG = {
|
||||
/* a whole note is one beat, which is - per definition in musical_notes.h - set to 64
|
||||
the longest note is BREAVE_DOT=128+64=192, the shortest SIXTEENTH=4
|
||||
the tempo (which might vary!) is in bpm (beats per minute)
|
||||
therefore: if the timer ticks away at .frequency = (60*64)Hz,
|
||||
and the .interval counts from 64 downwards - audio_update_state is
|
||||
called just often enough to not miss any notes
|
||||
*/
|
||||
.frequency = 60 * 64,
|
||||
.callback = gpt_callback,
|
||||
};
|
||||
|
||||
void audio_driver_initialize(void) {
|
||||
pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
|
||||
|
||||
// connect the AUDIO_PIN to the PWM hardware
|
||||
#if defined(USE_GPIOV1) // STM32F103C8
|
||||
palSetLineMode(AUDIO_PIN, PAL_MODE_ALTERNATE_PUSHPULL);
|
||||
#else // GPIOv2 (or GPIOv3 for f4xx, which is the same/compatible at this command)
|
||||
palSetLineMode(AUDIO_PIN, PAL_MODE_ALTERNATE(AUDIO_PWM_PAL_MODE));
|
||||
#endif
|
||||
|
||||
gptStart(&AUDIO_STATE_TIMER, &gptCFG);
|
||||
}
|
||||
|
||||
void audio_driver_start(void) {
|
||||
channel_1_stop();
|
||||
channel_1_start();
|
||||
|
||||
if (playing_note || playing_melody) {
|
||||
gptStartContinuous(&AUDIO_STATE_TIMER, 64);
|
||||
}
|
||||
}
|
||||
|
||||
void audio_driver_stop(void) {
|
||||
channel_1_stop();
|
||||
gptStopTimer(&AUDIO_STATE_TIMER);
|
||||
}
|
||||
|
||||
/* a regular timer task, that checks the note to be currently played
|
||||
* and updates the pwm to output that frequency
|
||||
*/
|
||||
static void gpt_callback(GPTDriver *gptp) {
|
||||
float freq; // TODO: freq_alt
|
||||
|
||||
if (audio_update_state()) {
|
||||
freq = audio_get_processed_frequency(0); // freq_alt would be index=1
|
||||
channel_1_set_frequency(freq);
|
||||
}
|
||||
}
|
||||
|
|
@ -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 <http://www.gnu.org/licenses/>.
|
||||
*/
|
||||
|
||||
/*
|
||||
Audio Driver: PWM
|
||||
|
||||
the duty-cycle is always kept at 50%, and the pwm-period is adjusted to match the frequency of a note to be played back.
|
||||
|
||||
this driver uses the chibios-PWM system to produce a square-wave on any given output pin in software
|
||||
- a pwm callback is used to set/clear the configured pin.
|
||||
|
||||
*/
|
||||
#include "audio.h"
|
||||
#include "ch.h"
|
||||
#include "hal.h"
|
||||
|
||||
#if !defined(AUDIO_PIN)
|
||||
# error "Audio feature enabled, but no pin selected - see docs/feature_audio under the ARM PWM settings"
|
||||
#endif
|
||||
extern bool playing_note;
|
||||
extern bool playing_melody;
|
||||
extern uint8_t note_timbre;
|
||||
|
||||
static void pwm_audio_period_callback(PWMDriver *pwmp);
|
||||
static void pwm_audio_channel_interrupt_callback(PWMDriver *pwmp);
|
||||
|
||||
static PWMConfig pwmCFG = {
|
||||
.frequency = 100000, /* PWM clock frequency */
|
||||
// CHIBIOS-BUG? can't set the initial period to <2, or the pwm (hard or software) takes ~130ms with .frequency=500000 for a pwmChangePeriod to take effect; with no output=silence in the meantime
|
||||
.period = 2, /* initial PWM period (in ticks) 1S (1/10kHz=0.1mS 0.1ms*10000 ticks=1S) */
|
||||
.callback = pwm_audio_period_callback,
|
||||
.channels =
|
||||
{
|
||||
// software-PWM just needs another callback on any channel
|
||||
{PWM_OUTPUT_ACTIVE_HIGH, pwm_audio_channel_interrupt_callback}, /* channel 0 -> TIMx_CH1 */
|
||||
{PWM_OUTPUT_DISABLED, NULL}, /* channel 1 -> TIMx_CH2 */
|
||||
{PWM_OUTPUT_DISABLED, NULL}, /* channel 2 -> TIMx_CH3 */
|
||||
{PWM_OUTPUT_DISABLED, NULL} /* channel 3 -> TIMx_CH4 */
|
||||
},
|
||||
};
|
||||
|
||||
static float channel_1_frequency = 0.0f;
|
||||
void channel_1_set_frequency(float freq) {
|
||||
channel_1_frequency = freq;
|
||||
|
||||
if (freq <= 0.0) // a pause/rest has freq=0
|
||||
return;
|
||||
|
||||
pwmcnt_t period = (pwmCFG.frequency / freq);
|
||||
pwmChangePeriod(&AUDIO_PWM_DRIVER, period);
|
||||
|
||||
pwmEnableChannel(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1,
|
||||
// adjust the duty-cycle so that the output is for 'note_timbre' duration HIGH
|
||||
PWM_PERCENTAGE_TO_WIDTH(&AUDIO_PWM_DRIVER, (100 - note_timbre) * 100));
|
||||
}
|
||||
|
||||
float channel_1_get_frequency(void) { return channel_1_frequency; }
|
||||
|
||||
void channel_1_start(void) {
|
||||
pwmStop(&AUDIO_PWM_DRIVER);
|
||||
pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
|
||||
|
||||
pwmEnablePeriodicNotification(&AUDIO_PWM_DRIVER);
|
||||
pwmEnableChannelNotification(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1);
|
||||
}
|
||||
|
||||
void channel_1_stop(void) {
|
||||
pwmStop(&AUDIO_PWM_DRIVER);
|
||||
|
||||
palClearLine(AUDIO_PIN); // leave the line low, after last note was played
|
||||
|
||||
#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
|
||||
palClearLine(AUDIO_PIN_ALT); // leave the line low, after last note was played
|
||||
#endif
|
||||
}
|
||||
|
||||
// generate a PWM signal on any pin, not necessarily the one connected to the timer
|
||||
static void pwm_audio_period_callback(PWMDriver *pwmp) {
|
||||
(void)pwmp;
|
||||
palClearLine(AUDIO_PIN);
|
||||
|
||||
#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
|
||||
palSetLine(AUDIO_PIN_ALT);
|
||||
#endif
|
||||
}
|
||||
static void pwm_audio_channel_interrupt_callback(PWMDriver *pwmp) {
|
||||
(void)pwmp;
|
||||
if (channel_1_frequency > 0) {
|
||||
palSetLine(AUDIO_PIN); // generate a PWM signal on any pin, not necessarily the one connected to the timer
|
||||
#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
|
||||
palClearLine(AUDIO_PIN_ALT);
|
||||
#endif
|
||||
}
|
||||
}
|
||||
|
||||
static void gpt_callback(GPTDriver *gptp);
|
||||
GPTConfig gptCFG = {
|
||||
/* a whole note is one beat, which is - per definition in musical_notes.h - set to 64
|
||||
the longest note is BREAVE_DOT=128+64=192, the shortest SIXTEENTH=4
|
||||
the tempo (which might vary!) is in bpm (beats per minute)
|
||||
therefore: if the timer ticks away at .frequency = (60*64)Hz,
|
||||
and the .interval counts from 64 downwards - audio_update_state is
|
||||
called just often enough to not miss anything
|
||||
*/
|
||||
.frequency = 60 * 64,
|
||||
.callback = gpt_callback,
|
||||
};
|
||||
|
||||
void audio_driver_initialize(void) {
|
||||
pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
|
||||
|
||||
palSetLineMode(AUDIO_PIN, PAL_MODE_OUTPUT_PUSHPULL);
|
||||
palClearLine(AUDIO_PIN);
|
||||
|
||||
#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
|
||||
palSetLineMode(AUDIO_PIN_ALT, PAL_MODE_OUTPUT_PUSHPULL);
|
||||
palClearLine(AUDIO_PIN_ALT);
|
||||
#endif
|
||||
|
||||
pwmEnablePeriodicNotification(&AUDIO_PWM_DRIVER); // enable pwm callbacks
|
||||
pwmEnableChannelNotification(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1);
|
||||
|
||||
gptStart(&AUDIO_STATE_TIMER, &gptCFG);
|
||||
}
|
||||
|
||||
void audio_driver_start(void) {
|
||||
channel_1_stop();
|
||||
channel_1_start();
|
||||
|
||||
if (playing_note || playing_melody) {
|
||||
gptStartContinuous(&AUDIO_STATE_TIMER, 64);
|
||||
}
|
||||
}
|
||||
|
||||
void audio_driver_stop(void) {
|
||||
channel_1_stop();
|
||||
gptStopTimer(&AUDIO_STATE_TIMER);
|
||||
}
|
||||
|
||||
/* a regular timer task, that checks the note to be currently played
|
||||
* and updates the pwm to output that frequency
|
||||
*/
|
||||
static void gpt_callback(GPTDriver *gptp) {
|
||||
float freq; // TODO: freq_alt
|
||||
|
||||
if (audio_update_state()) {
|
||||
freq = audio_get_processed_frequency(0); // freq_alt would be index=1
|
||||
channel_1_set_frequency(freq);
|
||||
}
|
||||
}
|
||||
Loading…
Add table
Add a link
Reference in a new issue