circuitpython/ports/nrf/common-hal/audiobusio/I2SOut.c
Dan Halbert 189efdf074 Use NO_PIN, not 0, for PDMIn unset pins.
Also:
- Use NO_PIN, not 0xff for I2SOut (cosmetic fix only; no actual functional change)
- Add VOLTAGE_MONITOR/BATTERY, ACCELEROMETER_INTERRUPT pins for LED Glasses Driver
2021-09-10 13:18:29 -04:00

365 lines
13 KiB
C

/*
* This file is part of the MicroPython project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2019 Jeff Epler for Adafruit Industries
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <math.h>
#include <string.h>
#include "common-hal/microcontroller/Pin.h"
#include "common-hal/audiobusio/I2SOut.h"
#include "shared-bindings/audiobusio/I2SOut.h"
#include "shared-module/audiocore/__init__.h"
#include "supervisor/shared/tick.h"
#include "py/obj.h"
#include "py/runtime.h"
static audiobusio_i2sout_obj_t *instance;
struct { int16_t l, r;
} static_sample16 = {0x8000, 0x8000};
struct { uint8_t l1, r1, l2, r2;
} static_sample8 = {0x80, 0x80, 0x80, 0x80};
struct frequency_info { uint32_t RATIO;
uint32_t MCKFREQ;
int sample_rate;
float abserr;
};
struct ratio_info { uint32_t RATIO;
int16_t divisor;
bool can_16bit;
};
struct ratio_info ratios[] = {
{ I2S_CONFIG_RATIO_RATIO_32X, 32, true },
{ I2S_CONFIG_RATIO_RATIO_48X, 48, false },
{ I2S_CONFIG_RATIO_RATIO_64X, 64, true },
{ I2S_CONFIG_RATIO_RATIO_96X, 96, true },
{ I2S_CONFIG_RATIO_RATIO_128X, 128, true },
{ I2S_CONFIG_RATIO_RATIO_192X, 192, true },
{ I2S_CONFIG_RATIO_RATIO_256X, 256, true },
{ I2S_CONFIG_RATIO_RATIO_384X, 384, true },
{ I2S_CONFIG_RATIO_RATIO_512X, 512, true },
};
struct mclk_info { uint32_t MCKFREQ;
int divisor;
};
struct mclk_info mclks[] = {
{ I2S_CONFIG_MCKFREQ_MCKFREQ_32MDIV8, 8 },
{ I2S_CONFIG_MCKFREQ_MCKFREQ_32MDIV10, 10 },
{ I2S_CONFIG_MCKFREQ_MCKFREQ_32MDIV11, 11 },
{ I2S_CONFIG_MCKFREQ_MCKFREQ_32MDIV15, 15 },
{ I2S_CONFIG_MCKFREQ_MCKFREQ_32MDIV16, 16 },
{ I2S_CONFIG_MCKFREQ_MCKFREQ_32MDIV21, 21 },
{ I2S_CONFIG_MCKFREQ_MCKFREQ_32MDIV23, 23 },
{ I2S_CONFIG_MCKFREQ_MCKFREQ_32MDIV31, 31 },
{ I2S_CONFIG_MCKFREQ_MCKFREQ_32MDIV42, 42 },
{ I2S_CONFIG_MCKFREQ_MCKFREQ_32MDIV63, 63 },
{ I2S_CONFIG_MCKFREQ_MCKFREQ_32MDIV125, 125 },
};
static void calculate_ratio_info(uint32_t target_sample_rate, struct frequency_info *info,
int ratio_index, int mclk_index) {
info->RATIO = ratios[ratio_index].RATIO;
info->MCKFREQ = mclks[mclk_index].MCKFREQ;
info->sample_rate = 32000000
/ ratios[ratio_index].divisor / mclks[mclk_index].divisor;
info->abserr = fabsf(1.0f * target_sample_rate - info->sample_rate)
/ target_sample_rate;
}
void choose_i2s_clocking(audiobusio_i2sout_obj_t *self, uint32_t sample_rate) {
struct frequency_info best = {0, 0, 0, 1.0};
for (size_t ri = 0; ri < sizeof(ratios) / sizeof(ratios[0]); ri++) {
if (NRF_I2S->CONFIG.SWIDTH == I2S_CONFIG_SWIDTH_SWIDTH_16Bit
&& !ratios[ri].can_16bit) {
continue;
}
for (size_t mi = 0; mi < sizeof(mclks) / sizeof(mclks[0]); mi++) {
struct frequency_info info = {0, 0, 1.0};
calculate_ratio_info(sample_rate, &info, ri, mi);
if (info.abserr < best.abserr) {
best = info;
}
#ifdef DEBUG_CLOCKING
mp_printf(&mp_plat_print,
"RATIO=%3d MCKFREQ=%08x rate=%d abserr=%.4f\n",
info.RATIO, info.MCKFREQ, info.sample_rate,
(double)info.abserr);
#endif
}
}
NRF_I2S->CONFIG.RATIO = best.RATIO;
NRF_I2S->CONFIG.MCKFREQ = best.MCKFREQ;
self->sample_rate = best.sample_rate;
}
static void i2s_buffer_fill(audiobusio_i2sout_obj_t *self) {
void *buffer = self->buffers[self->next_buffer];
void *buffer_start = buffer;
NRF_I2S->TXD.PTR = (uintptr_t)buffer;
self->next_buffer = !self->next_buffer;
size_t bytesleft = self->buffer_length;
while (!self->paused && !self->stopping && bytesleft) {
if (self->sample_data == self->sample_end) {
uint32_t sample_buffer_length;
audioio_get_buffer_result_t get_buffer_result =
audiosample_get_buffer(self->sample, false, 0,
&self->sample_data, &sample_buffer_length);
self->sample_end = self->sample_data + sample_buffer_length;
if (get_buffer_result == GET_BUFFER_DONE) {
if (self->loop) {
audiosample_reset_buffer(self->sample, false, 0);
} else {
self->stopping = true;
break;
}
}
if (get_buffer_result == GET_BUFFER_ERROR || sample_buffer_length == 0) {
self->stopping = true;
break;
}
}
uint16_t bytecount = MIN(bytesleft, (size_t)(self->sample_end - self->sample_data));
if (self->samples_signed) {
memcpy(buffer, self->sample_data, bytecount);
} else if (self->bytes_per_sample == 2) {
uint16_t *bp = (uint16_t *)buffer;
uint16_t *be = (uint16_t *)(buffer + bytecount);
uint16_t *sp = (uint16_t *)self->sample_data;
for (; bp < be;) {
*bp++ = *sp++ + 0x8000;
}
} else {
uint8_t *bp = (uint8_t *)buffer;
uint8_t *be = (uint8_t *)(buffer + bytecount);
uint8_t *sp = (uint8_t *)self->sample_data;
for (; bp < be;) {
*bp++ = *sp++ + 0x80;
}
}
buffer += bytecount;
self->sample_data += bytecount;
bytesleft -= bytecount;
}
// Find the last frame of real audio data and replicate its samples until
// you have 32 bits worth, which is the fundamental unit of nRF I2S DMA
if (buffer != buffer_start) {
if (self->bytes_per_sample == 1 && self->channel_count == 1) {
// For 8-bit mono, 4 copies of the final sample are required
self->hold_value = 0x01010101 * *(uint8_t *)(buffer - 1);
} else if (self->bytes_per_sample == 2 && self->channel_count == 2) {
// For 16-bit stereo, 1 copy of the final sample is required
self->hold_value = *(uint32_t *)(buffer - 4);
} else {
// For 8-bit stereo and 16-bit mono, 2 copies of the final sample are required
self->hold_value = 0x00010001 * *(uint16_t *)(buffer - 2);
}
}
// Emulate pausing and stopping by filling the DMA buffer with copies of
// the last sample. This includes the case where this iteration of
// i2s_buffer_fill exhausted a non-looping sample.
if (self->paused || self->stopping) {
if (self->stopping) {
NRF_I2S->TASKS_STOP = 1;
self->playing = false;
}
uint32_t *bp = (uint32_t *)buffer;
uint32_t *be = (uint32_t *)(buffer + bytesleft);
for (; bp != be;) {
*bp++ = self->hold_value;
}
return;
}
}
void common_hal_audiobusio_i2sout_construct(audiobusio_i2sout_obj_t *self,
const mcu_pin_obj_t *bit_clock, const mcu_pin_obj_t *word_select,
const mcu_pin_obj_t *data, bool left_justified) {
if (instance) {
mp_raise_RuntimeError(translate("Device in use"));
}
instance = self;
claim_pin(bit_clock);
claim_pin(word_select);
claim_pin(data);
NRF_I2S->PSEL.SCK = self->bit_clock_pin_number = bit_clock->number;
NRF_I2S->PSEL.LRCK = self->word_select_pin_number = word_select->number;
NRF_I2S->PSEL.SDOUT = self->data_pin_number = data->number;
NRF_I2S->CONFIG.MODE = I2S_CONFIG_MODE_MODE_Master;
NRF_I2S->CONFIG.RXEN = I2S_CONFIG_RXEN_RXEN_Disabled;
NRF_I2S->CONFIG.TXEN = I2S_CONFIG_TXEN_TXEN_Enabled;
NRF_I2S->CONFIG.MCKEN = I2S_CONFIG_MCKEN_MCKEN_Enabled;
NRF_I2S->CONFIG.SWIDTH = I2S_CONFIG_SWIDTH_SWIDTH_16Bit;
NRF_I2S->CONFIG.ALIGN = I2S_CONFIG_ALIGN_ALIGN_Left;
NRF_I2S->CONFIG.FORMAT = left_justified ? I2S_CONFIG_FORMAT_FORMAT_Aligned
: I2S_CONFIG_FORMAT_FORMAT_I2S;
supervisor_enable_tick();
}
bool common_hal_audiobusio_i2sout_deinited(audiobusio_i2sout_obj_t *self) {
return self->data_pin_number == NO_PIN;
}
void common_hal_audiobusio_i2sout_deinit(audiobusio_i2sout_obj_t *self) {
if (common_hal_audiobusio_i2sout_deinited(self)) {
return;
}
NRF_I2S->TASKS_STOP = 1;
NRF_I2S->ENABLE = I2S_ENABLE_ENABLE_Disabled;
reset_pin_number(self->bit_clock_pin_number);
self->bit_clock_pin_number = NO_PIN;
reset_pin_number(self->word_select_pin_number);
self->word_select_pin_number = NO_PIN;
reset_pin_number(self->data_pin_number);
self->data_pin_number = NO_PIN;
instance = NULL;
supervisor_disable_tick();
}
void common_hal_audiobusio_i2sout_play(audiobusio_i2sout_obj_t *self,
mp_obj_t sample, bool loop) {
if (common_hal_audiobusio_i2sout_get_playing(self)) {
common_hal_audiobusio_i2sout_stop(self);
}
self->sample = sample;
self->loop = loop;
uint32_t sample_rate = audiosample_sample_rate(sample);
self->bytes_per_sample = audiosample_bits_per_sample(sample) / 8;
uint32_t max_buffer_length;
bool single_buffer, samples_signed;
audiosample_get_buffer_structure(sample, /* single channel */ true,
&single_buffer, &samples_signed, &max_buffer_length,
&self->channel_count);
self->single_buffer = single_buffer;
self->samples_signed = samples_signed;
NRF_I2S->CONFIG.SWIDTH = self->bytes_per_sample == 1
? I2S_CONFIG_SWIDTH_SWIDTH_8Bit
: I2S_CONFIG_SWIDTH_SWIDTH_16Bit;
NRF_I2S->CONFIG.CHANNELS = self->channel_count == 1
? I2S_CONFIG_CHANNELS_CHANNELS_Left
: I2S_CONFIG_CHANNELS_CHANNELS_Stereo;
choose_i2s_clocking(self, sample_rate);
/* Allocate buffers based on a maximum duration
* This duration was chosen empirically based on what would
* cause os.listdir('') to cause stuttering. It seems like a
* rather long time.
*/
enum { buffer_length_ms = 16 };
self->buffer_length = sample_rate * buffer_length_ms
* self->bytes_per_sample * self->channel_count / 1000;
self->buffer_length = (self->buffer_length + 3) & ~3;
self->buffers[0] = m_malloc(self->buffer_length, false);
self->buffers[1] = m_malloc(self->buffer_length, false);
audiosample_reset_buffer(self->sample, false, 0);
self->next_buffer = 0;
self->sample_data = self->sample_end = 0;
self->playing = true;
self->paused = false;
self->stopping = false;
i2s_buffer_fill(self);
NRF_I2S->RXTXD.MAXCNT = self->buffer_length / 4;
// Turn on the interrupt to the NVIC but not within the NVIC itself. This will wake the CPU and
// keep it awake until it is serviced without triggering an interrupt handler.
NRF_I2S->INTENSET = I2S_INTENSET_TXPTRUPD_Msk;
NRF_I2S->ENABLE = I2S_ENABLE_ENABLE_Enabled;
NRF_I2S->TASKS_START = 1;
i2s_background();
}
void common_hal_audiobusio_i2sout_pause(audiobusio_i2sout_obj_t *self) {
self->paused = true;
}
void common_hal_audiobusio_i2sout_resume(audiobusio_i2sout_obj_t *self) {
self->paused = false;
}
bool common_hal_audiobusio_i2sout_get_paused(audiobusio_i2sout_obj_t *self) {
return self->paused;
}
void common_hal_audiobusio_i2sout_stop(audiobusio_i2sout_obj_t *self) {
NRF_I2S->TASKS_STOP = 1;
self->stopping = true;
NRF_I2S->INTENCLR = I2S_INTENSET_TXPTRUPD_Msk;
}
bool common_hal_audiobusio_i2sout_get_playing(audiobusio_i2sout_obj_t *self) {
if (NRF_I2S->EVENTS_STOPPED) {
self->playing = false;
NRF_I2S->EVENTS_STOPPED = 0;
}
return self->playing;
}
void i2s_background(void) {
if (NVIC_GetPendingIRQ(I2S_IRQn) && NRF_I2S->EVENTS_TXPTRUPD) {
NRF_I2S->EVENTS_TXPTRUPD = 0;
NVIC_ClearPendingIRQ(I2S_IRQn);
if (instance) {
i2s_buffer_fill(instance);
} else {
NRF_I2S->TASKS_STOP = 1;
}
}
}
void i2s_reset(void) {
NRF_I2S->TASKS_STOP = 1;
NRF_I2S->INTENCLR = I2S_INTENSET_TXPTRUPD_Msk;
NRF_I2S->ENABLE = I2S_ENABLE_ENABLE_Disabled;
NRF_I2S->PSEL.MCK = 0xFFFFFFFF;
NRF_I2S->PSEL.SCK = 0xFFFFFFFF;
NRF_I2S->PSEL.LRCK = 0xFFFFFFFF;
NRF_I2S->PSEL.SDOUT = 0xFFFFFFFF;
NRF_I2S->PSEL.SDIN = 0xFFFFFFFF;
if (instance) {
supervisor_disable_tick();
}
instance = NULL;
}