From 474d100b567ba6d8862cd67831efe4f5cb815f08 Mon Sep 17 00:00:00 2001 From: Jack Humbert Date: Sun, 2 Jun 2019 16:42:36 -0400 Subject: [PATCH] refined a bit --- keyboards/planck/ez/config.h | 3 + keyboards/planck/rev6/config.h | 2 + keyboards/planck/rev6/rev6.c | 4 +- quantum/audio/audio_arm.c | 218 ++++++++++++++++++--------------- quantum/stm32/mcuconf.h | 4 +- 5 files changed, 127 insertions(+), 104 deletions(-) diff --git a/keyboards/planck/ez/config.h b/keyboards/planck/ez/config.h index c449d171926..c37b6ca3018 100644 --- a/keyboards/planck/ez/config.h +++ b/keyboards/planck/ez/config.h @@ -51,6 +51,9 @@ #undef AUDIO_VOICES #undef C6_AUDIO +#define A5_AUDIO +#define DAC_OFF_VALUE 4095 + /* Debounce reduces chatter (unintended double-presses) - set 0 if debouncing is not needed */ #define DEBOUNCE 6 diff --git a/keyboards/planck/rev6/config.h b/keyboards/planck/rev6/config.h index 4713d9d233d..da4750974cf 100644 --- a/keyboards/planck/rev6/config.h +++ b/keyboards/planck/rev6/config.h @@ -51,6 +51,8 @@ #undef AUDIO_VOICES #undef C6_AUDIO +#define A5_AUDIO + /* Debounce reduces chatter (unintended double-presses) - set 0 if debouncing is not needed */ #define DEBOUNCE 6 diff --git a/keyboards/planck/rev6/rev6.c b/keyboards/planck/rev6/rev6.c index 650e1a194d6..ccf5a329c35 100644 --- a/keyboards/planck/rev6/rev6.c +++ b/keyboards/planck/rev6/rev6.c @@ -16,7 +16,9 @@ #include "rev6.h" void matrix_init_kb(void) { - matrix_init_user(); + palSetPadMode(GPIOA, 4, PAL_MODE_OUTPUT_PUSHPULL ); + palSetPad(GPIOA, 4); + matrix_init_user(); } void matrix_scan_kb(void) { diff --git a/quantum/audio/audio_arm.c b/quantum/audio/audio_arm.c index 5d25083f44f..7f842da3b57 100644 --- a/quantum/audio/audio_arm.c +++ b/quantum/audio/audio_arm.c @@ -26,19 +26,54 @@ // ----------------------------------------------------------------------------- +/** + * Size of the dac_buffer arrays. All must be the same size. + */ +#define DAC_BUFFER_SIZE 256U + +/** + * Highest value allowed by our 12bit DAC. + */ +#ifndef DAC_SAMPLE_MAX + #define DAC_SAMPLE_MAX 4095U +#endif + +/** + * Effective bitrate 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 voices. In most situations, a tenth (1/10) of the + * sample rate is where notes become unbearable. + */ +#ifndef DAC_SAMPLE_RATE + #define DAC_SAMPLE_RATE 44100U +#endif + +/** + * The number of voices (in polyphony) that are supported. If too high a value + * is used here, the keyboard will freeze and glitch-out when that many voices + * are being played. + */ +#ifndef DAC_VOICES_MAX + #define DAC_VOICES_MAX 2 +#endif + +/** + * The default value of the DAC when not playing anything. Certain hardware + * setups may require a high (DAC_SAMPLE_MAX) or low (0) value here. + */ +#ifndef DAC_OFF_VALUE + #define DAC_OFF_VALUE DAC_SAMPLE_MAX / 2 +#endif + int voices = 0; int voice_place = 0; float frequency = 0; float frequency_alt = 0; -int volume = 0; -long position = 0; float frequencies[8] = {0, 0, 0, 0, 0, 0, 0, 0}; int volumes[8] = {0, 0, 0, 0, 0, 0, 0, 0}; bool sliding = false; -float place = 0; - uint8_t * sample; uint16_t sample_length = 0; @@ -77,43 +112,6 @@ bool glissando = true; #endif float startup_song[][2] = STARTUP_SONG; -/** Size of the dac_buffer arrays. All must be the same size. */ -#define DAC_BUFFER_SIZE 256U - -/** Highest value allowed by our 12bit DAC */ -#ifndef DAC_SAMPLE_MAX - #define DAC_SAMPLE_MAX 4095U -#endif - -/** Effective bitrate 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 voices. - */ -#ifndef DAC_SAMPLE_RATE - #define DAC_SAMPLE_RATE 44100U -#endif - -/** The number of voices (in polyphony) that are supported. Certain voices will - * glitch out at different values - most (the look-ups) survive 5. - */ -#ifndef DAC_VOICES_MAX - #define DAC_VOICES_MAX 5 -#endif - -/** The default value of the DAC when not playing anything. Certain hardware - * setups may require a high (DAC_SAMPLE_MAX) or low (0) value here. - */ -#ifndef DAC_OFF_VALUE - #define DAC_OFF_VALUE DAC_SAMPLE_MAX / 2 -#endif - -GPTConfig gpt7cfg1 = { - .frequency = DAC_SAMPLE_RATE, - .callback = NULL, - .cr2 = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event. */ - .dier = 0U -}; - static const dacsample_t dac_buffer[DAC_BUFFER_SIZE] = { // 256 values, max 4095 0x800,0x832,0x864,0x896,0x8c8,0x8fa,0x92c,0x95e, @@ -137,12 +135,12 @@ static const dacsample_t dac_buffer[DAC_BUFFER_SIZE] = { 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, - 0x0,0x1,0x2,0x6,0xa,0xf,0x16,0x1e, - 0x27,0x32,0x3d,0x4a,0x58,0x67,0x78,0x89, - 0x9c,0xb0,0xc5,0xdb,0xf2,0x10a,0x123,0x13e, + 0x159,0x13e,0x123,0x10a,0xf2, 0xdb, 0xc5, 0xb0, + 0x9c, 0x89, 0x78, 0x67, 0x58, 0x4a, 0x3d, 0x32, + 0x27, 0x1e, 0x16, 0xf, 0xa, 0x6, 0x2, 0x1, + 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, @@ -152,7 +150,7 @@ static const dacsample_t dac_buffer[DAC_BUFFER_SIZE] = { static const dacsample_t dac_buffer_triangle[DAC_BUFFER_SIZE] = { // 256 values, max 4095 - 0x20,0x40,0x60,0x80,0xa0,0xc0,0xe0,0x100, + 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, @@ -183,64 +181,68 @@ static const dacsample_t dac_buffer_triangle[DAC_BUFFER_SIZE] = { 0x3e0,0x3c0,0x3a0,0x380,0x360,0x340,0x320,0x300, 0x2e0,0x2c0,0x2a0,0x280,0x260,0x240,0x220,0x200, 0x1e0,0x1c0,0x1a0,0x180,0x160,0x140,0x120,0x100, - 0xe0,0xc0,0xa0,0x80,0x60,0x40,0x20,0x0 + 0xe0, 0xc0, 0xa0, 0x80, 0x60, 0x40, 0x20, 0x0 }; -// static const dacsample_t dac_buffer_square[DAC_BUFFER_SIZE] = { -// // First half is max, second half is 0 -// [0 ... DAC_BUFFER_SIZE/2-1] = DAC_SAMPLE_MAX, -// [DAC_BUFFER_SIZE/2 ... DAC_BUFFER_SIZE -1] = 0, -// }; +static const dacsample_t dac_buffer_square[DAC_BUFFER_SIZE] = { + // First half is max, second half is 0 + [0 ... DAC_BUFFER_SIZE/2-1] = DAC_SAMPLE_MAX, + [DAC_BUFFER_SIZE/2 ... DAC_BUFFER_SIZE -1] = 0, +}; -dacsample_t dac_buffer_lr[DAC_BUFFER_SIZE] = { DAC_OFF_VALUE }; +static dacsample_t dac_buffer_empty[DAC_BUFFER_SIZE] = { DAC_OFF_VALUE }; float dac_if[8] = {0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0}; -/* - * DAC streaming callback. +/** + * DAC streaming callback. Does all of the main computing for sound synthesis. */ -static void end_cb1(DACDriver * dacp, dacsample_t * samples, size_t rows) { +static void dac_end(DACDriver * dacp, dacsample_t * sample_p, size_t sample_count) { (void)dacp; (void)dac_buffer; + // (void)dac_buffer_triangle; + (void)dac_buffer_square; uint8_t working_voices = voices; if (working_voices > DAC_VOICES_MAX) working_voices = DAC_VOICES_MAX; - for (uint8_t s = 0; s < rows; s++) { + for (uint8_t s = 0; s < sample_count; s++) { if (working_voices > 0) { uint16_t sample_sum = 0; for (uint8_t i = 0; i < working_voices; i++) { - dac_if[i] = dac_if[i] + ((frequencies[i]*(float)DAC_BUFFER_SIZE)/(float)DAC_SAMPLE_RATE*1.5); + dac_if[i] = dac_if[i] + ((frequencies[i]*DAC_BUFFER_SIZE)/DAC_SAMPLE_RATE); // Needed because % doesn't work with floats // 0.5 less than the size because we use round() later - while(dac_if[i] >= (DAC_BUFFER_SIZE - 0.5)) + while (dac_if[i] >= (DAC_BUFFER_SIZE)) dac_if[i] = dac_if[i] - DAC_BUFFER_SIZE; + uint16_t dac_i = (uint16_t)dac_if[i]; // Wavetable generation/lookup - // sine - // sample_sum += dac_buffer[(uint16_t)round(dac_if[i])] / working_voices; - // triangle wave (5 voices) - sample_sum += dac_buffer_triangle[(uint16_t)round(dac_if[i])] / working_voices; - // rising triangle (4 voices) + // SINE + sample_sum += dac_buffer[dac_i] / working_voices / 3; + // TRIANGLE + sample_sum += dac_buffer_triangle[dac_i] / working_voices / 3; + // RISING TRIANGLE // sample_sum += (uint16_t)round((dac_if[i] * DAC_SAMPLE_MAX) / DAC_BUFFER_SIZE / working_voices ); - // square (max 5 voices) + // SQUARE // sample_sum += ((dac_if[i] > (DAC_BUFFER_SIZE / 2)) ? DAC_SAMPLE_MAX / working_voices: 0); + sample_sum += dac_buffer_square[dac_i] / working_voices / 3; } - samples[s] = sample_sum; + sample_p[s] = sample_sum; } else { - samples[s] = DAC_OFF_VALUE; + sample_p[s] = DAC_OFF_VALUE; } } if (playing_notes) { - note_position += rows; + note_position += sample_count; - // end of the note - if ((note_position >= (note_length*420))) { + // End of the note - 35 is arbitary here, but gets us close to AVR's timing + if ((note_position >= (note_length*DAC_SAMPLE_RATE/35))) { stop_note((*notes_pointer)[current_note][0]); current_note++; if (current_note >= notes_count) { @@ -255,34 +257,52 @@ static void end_cb1(DACDriver * dacp, dacsample_t * samples, size_t rows) { envelope_index = 0; note_length = ((*notes_pointer)[current_note][1] / 4) * (((float)note_tempo) / 100); - note_position = note_position - (note_length*420); - // note_position = 0; + // Skip forward in the next note's length if we've over shot the last, so + // the overall length of the song is the same + note_position = note_position - (note_length*DAC_SAMPLE_RATE/35); } } - } -/* - * DAC error callback. - */ -static void error_cb1(DACDriver *dacp, dacerror_t err) { +static void dac_error(DACDriver *dacp, dacerror_t err) { (void)dacp; (void)err; - chSysHalt("DAC failure"); + chSysHalt("DAC failure. halp"); } -static const DACConfig dac1cfg1 = { +static const GPTConfig gpt6cfg1 = { + .frequency = 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 = DAC_SAMPLE_MAX, .datamode = DAC_DHRM_12BIT_RIGHT }; -static const DACConversionGroup dacgrpcfg1 = { +/** + * @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 = end_cb1, - .error_cb = error_cb1, - .trigger = DAC_TRG(0) + .end_cb = dac_end, + .error_cb = dac_error, + .trigger = DAC_TRG(0b000) }; void audio_init() { @@ -304,20 +324,20 @@ void audio_init() { #endif #endif // ARM EEPROM - palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG ); - // palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG ); - palSetPadMode(GPIOA, 4, PAL_MODE_OUTPUT_PUSHPULL ); - palSetPad(GPIOA, 4); - // dacStart(&DACD1, &dac1cfg1); - // dacStartConversion(&DACD1, &dacgrpcfg1, dac_buffer_lr, 1); - dacStart(&DACD2, &dac1cfg1); - dacStartConversion(&DACD2, &dacgrpcfg1, dac_buffer_lr, DAC_BUFFER_SIZE); +#if defined(A4_AUDIO) + palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG ); + dacStart(&DACD1, &dac_conf); + dacStartConversion(&DACD1, &dac_conv_cfg, dac_buffer_empty, DAC_BUFFER_SIZE); +#endif +#if defined(A5_AUDIO) + palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG ); + dacStart(&DACD2, &dac_conf); + dacStartConversion(&DACD2, &dac_conv_cfg, dac_buffer_empty, DAC_BUFFER_SIZE); +#endif - gptStart(&GPTD6, &gpt7cfg1); + gptStart(&GPTD6, &gpt6cfg1); gptStartContinuous(&GPTD6, 2U); - // gptStart(&GPTD7, &gpt7cfg1); - // gptStartContinuous(&GPTD7, 2U); audio_initialized = true; @@ -337,13 +357,10 @@ void stop_all_notes() { } voices = 0; - gptStopTimer(&GPTD8); - playing_notes = false; playing_note = false; frequency = 0; frequency_alt = 0; - volume = 0; for (uint8_t i = 0; i < 8; i++) { @@ -382,7 +399,6 @@ void stop_note(float freq) { if (voices == 0) { frequency = 0; frequency_alt = 0; - volume = 0; playing_note = false; } } @@ -428,6 +444,7 @@ void play_note(float freq, int vol) { if (freq > 0) { envelope_index = 0; frequencies[voices] = freq; + dac_if[voices] = 0.0f; volumes[voices] = vol; voices++; } @@ -450,7 +467,6 @@ void play_notes(float (*np)[][2], uint16_t n_count, bool n_repeat) { notes_count = n_count; notes_repeat = n_repeat; - place = 0; current_note = 0; note_length = ((*notes_pointer)[current_note][1] / 4) * (((float)note_tempo) / 100); diff --git a/quantum/stm32/mcuconf.h b/quantum/stm32/mcuconf.h index 36f8ca2252a..b84ccc225a3 100644 --- a/quantum/stm32/mcuconf.h +++ b/quantum/stm32/mcuconf.h @@ -141,8 +141,8 @@ #define STM32_GPT_USE_TIM3 FALSE #define STM32_GPT_USE_TIM4 FALSE #define STM32_GPT_USE_TIM6 TRUE -#define STM32_GPT_USE_TIM7 TRUE -#define STM32_GPT_USE_TIM8 TRUE +#define STM32_GPT_USE_TIM7 FALSE +#define STM32_GPT_USE_TIM8 FALSE #define STM32_GPT_TIM1_IRQ_PRIORITY 7 #define STM32_GPT_TIM2_IRQ_PRIORITY 7 #define STM32_GPT_TIM3_IRQ_PRIORITY 7