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