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337 lines
19 KiB
C
337 lines
19 KiB
C
/* Copyright 2016-2019 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
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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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#include "audio.h"
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#include <ch.h>
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#include <hal.h>
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/*
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Audio Driver: DAC
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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
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it is also possible to have a custom sample-LUT by implementing/overriding 'dac_value_generate'
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this driver allows for multiple simultaneous tones to be played through one single channel by doing additive wave-synthesis
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*/
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#if !defined(AUDIO_PIN)
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# error "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under 'ARM (DAC additive)' for available options."
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#endif
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#if defined(AUDIO_PIN_ALT) && !defined(AUDIO_PIN_ALT_AS_NEGATIVE)
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# 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."
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#endif
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#if !defined(AUDIO_PIN_ALT)
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// no ALT pin defined is valid, but the c-ifs below need some value set
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# define AUDIO_PIN_ALT PAL_NOLINE
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#endif
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#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)
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# define AUDIO_DAC_SAMPLE_WAVEFORM_SINE
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#endif
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#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_SINE
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/* one full sine wave over [0,2*pi], but shifted up one amplitude and left pi/4; for the samples to start at 0
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*/
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static const dacsample_t dac_buffer_sine[AUDIO_DAC_BUFFER_SIZE] = {
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// 256 values, max 4095
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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,
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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};
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#endif // AUDIO_DAC_SAMPLE_WAVEFORM_SINE
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#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE
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static const dacsample_t dac_buffer_triangle[AUDIO_DAC_BUFFER_SIZE] = {
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// 256 values, max 4095
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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,
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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};
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#endif // AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE
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#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE
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static const dacsample_t dac_buffer_square[AUDIO_DAC_BUFFER_SIZE] = {
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[0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1] = 0, // first and
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[AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = AUDIO_DAC_SAMPLE_MAX, // second half
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};
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#endif // AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE
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/*
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// four steps: 0, 1/3, 2/3 and 1
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static const dacsample_t dac_buffer_staircase[AUDIO_DAC_BUFFER_SIZE] = {
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[0 ... AUDIO_DAC_BUFFER_SIZE/3 -1 ] = 0,
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[AUDIO_DAC_BUFFER_SIZE / 4 ... AUDIO_DAC_BUFFER_SIZE / 2 -1 ] = AUDIO_DAC_SAMPLE_MAX / 3,
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[AUDIO_DAC_BUFFER_SIZE / 2 ... 3 * AUDIO_DAC_BUFFER_SIZE / 4 -1 ] = 2 * AUDIO_DAC_SAMPLE_MAX / 3,
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[3 * AUDIO_DAC_BUFFER_SIZE / 4 ... AUDIO_DAC_BUFFER_SIZE -1 ] = AUDIO_DAC_SAMPLE_MAX,
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}
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*/
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#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID
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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,
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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};
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#endif // AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID
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static dacsample_t dac_buffer_empty[AUDIO_DAC_BUFFER_SIZE] = {AUDIO_DAC_OFF_VALUE};
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/* keep track of the sample position for for each frequency */
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static float dac_if[AUDIO_MAX_SIMULTANEOUS_TONES] = {0.0};
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static float active_tones_snapshot[AUDIO_MAX_SIMULTANEOUS_TONES] = {0, 0};
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static uint8_t active_tones_snapshot_length = 0;
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typedef enum {
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OUTPUT_SHOULD_START,
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OUTPUT_RUN_NORMALLY,
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// path 1: wait for zero, then change/update active tones
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OUTPUT_TONES_CHANGED,
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OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE,
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// path 2: hardware should stop, wait for zero then turn output off = stop the timer
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OUTPUT_SHOULD_STOP,
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OUTPUT_REACHED_ZERO_BEFORE_OFF,
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OUTPUT_OFF,
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OUTPUT_OFF_1,
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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
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number_of_output_states
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} output_states_t;
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output_states_t state = OUTPUT_OFF_2;
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/**
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* Generation of the waveform being passed to the callback. Declared weak so users
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* can override it with their own wave-forms/noises.
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*/
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__attribute__((weak)) uint16_t dac_value_generate(void) {
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// DAC is running/asking for values but snapshot length is zero -> must be playing a pause
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if (active_tones_snapshot_length == 0) {
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return AUDIO_DAC_OFF_VALUE;
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}
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/* doing additive wave synthesis over all currently playing tones = adding up
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* sine-wave-samples for each frequency, scaled by the number of active tones
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*/
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uint16_t value = 0;
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float frequency = 0.0f;
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for (uint8_t i = 0; i < active_tones_snapshot_length; i++) {
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/* Note: a user implementation does not have to rely on the active_tones_snapshot, but
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* could directly query the active frequencies through audio_get_processed_frequency */
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frequency = active_tones_snapshot[i];
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dac_if[i] = dac_if[i] + ((frequency * AUDIO_DAC_BUFFER_SIZE) / AUDIO_DAC_SAMPLE_RATE) * 2 / 3;
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/*Note: the 2/3 are necessary to get the correct frequencies on the
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* DAC output (as measured with an oscilloscope), since the gpt
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* timer runs with 3*AUDIO_DAC_SAMPLE_RATE; and the DAC callback
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* is called twice per conversion.*/
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dac_if[i] = fmod(dac_if[i], AUDIO_DAC_BUFFER_SIZE);
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// Wavetable generation/lookup
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uint16_t dac_i = (uint16_t)dac_if[i];
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#if defined(AUDIO_DAC_SAMPLE_WAVEFORM_SINE)
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value += dac_buffer_sine[dac_i] / active_tones_snapshot_length;
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#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE)
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value += dac_buffer_triangle[dac_i] / active_tones_snapshot_length;
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#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID)
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value += dac_buffer_trapezoid[dac_i] / active_tones_snapshot_length;
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#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE)
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value += dac_buffer_square[dac_i] / active_tones_snapshot_length;
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#endif
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/*
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// SINE
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value += dac_buffer_sine[dac_i] / active_tones_snapshot_length / 3;
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// TRIANGLE
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value += dac_buffer_triangle[dac_i] / active_tones_snapshot_length / 3;
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// SQUARE
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value += dac_buffer_square[dac_i] / active_tones_snapshot_length / 3;
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//NOTE: combination of these three wave-forms is more exemplary - and doesn't sound particularly good :-P
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*/
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// STAIRS (mostly usefully as test-pattern)
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// value_avg = dac_buffer_staircase[dac_i] / active_tones_snapshot_length;
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}
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return value;
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}
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/**
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* DAC streaming callback. Does all of the main computing for playing songs.
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*
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* Note: chibios calls this CB twice: during the 'half buffer event', and the 'full buffer event'.
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*/
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static void dac_end(DACDriver *dacp) {
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dacsample_t *sample_p = (dacp)->samples;
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// work on the other half of the buffer
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if (dacIsBufferComplete(dacp)) {
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sample_p += AUDIO_DAC_BUFFER_SIZE / 2; // 'half_index'
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}
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for (uint8_t s = 0; s < AUDIO_DAC_BUFFER_SIZE / 2; s++) {
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if (OUTPUT_OFF <= state) {
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sample_p[s] = AUDIO_DAC_OFF_VALUE;
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continue;
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} else {
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sample_p[s] = dac_value_generate();
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}
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/* zero crossing (or approach, whereas zero == DAC_OFF_VALUE, which can be configured to anything from 0 to DAC_SAMPLE_MAX)
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* ============================*=*========================== AUDIO_DAC_SAMPLE_MAX
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* * *
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* * *
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* ---------------------------------------------------------
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* * * } AUDIO_DAC_SAMPLE_MAX/100
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* --------------------------------------------------------- AUDIO_DAC_OFF_VALUE
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* * * } AUDIO_DAC_SAMPLE_MAX/100
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* ---------------------------------------------------------
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* *
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* * *
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* * *
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* =====*=*================================================= 0x0
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*/
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if (((sample_p[s] + (AUDIO_DAC_SAMPLE_MAX / 100)) > AUDIO_DAC_OFF_VALUE) && // value approaches from below
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(sample_p[s] < (AUDIO_DAC_OFF_VALUE + (AUDIO_DAC_SAMPLE_MAX / 100))) // or above
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) {
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if ((OUTPUT_SHOULD_START == state) && (active_tones_snapshot_length > 0)) {
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state = OUTPUT_RUN_NORMALLY;
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} else if (OUTPUT_TONES_CHANGED == state) {
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state = OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE;
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} else if (OUTPUT_SHOULD_STOP == state) {
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state = OUTPUT_REACHED_ZERO_BEFORE_OFF;
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}
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}
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// still 'ramping up', reset the output to OFF_VALUE until the generated values reach that value, to do a smooth handover
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if (OUTPUT_SHOULD_START == state) {
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sample_p[s] = AUDIO_DAC_OFF_VALUE;
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}
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if ((OUTPUT_SHOULD_START == state) || (OUTPUT_REACHED_ZERO_BEFORE_OFF == state) || (OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE == state)) {
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uint8_t active_tones = MIN(AUDIO_MAX_SIMULTANEOUS_TONES, audio_get_number_of_active_tones());
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active_tones_snapshot_length = 0;
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// update the snapshot - once, and only on occasion that something changed;
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// -> saves cpu cycles (?)
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for (uint8_t i = 0; i < active_tones; i++) {
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float freq = audio_get_processed_frequency(i);
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if (freq > 0) { // disregard 'rest' notes, with valid frequency 0.0f; which would only lower the resulting waveform volume during the additive synthesis step
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active_tones_snapshot[active_tones_snapshot_length++] = freq;
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}
|
|
}
|
|
|
|
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(void) {
|
|
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;
|
|
}
|