/* * 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 #include #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 "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; riCONFIG.SWIDTH == I2S_CONFIG_SWIDTH_SWIDTH_16Bit && !ratios[ri].can_16bit) { continue; } for (size_t mi=0; miCONFIG.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++) { *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++) { *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; } bool common_hal_audiobusio_i2sout_deinited(audiobusio_i2sout_obj_t* self) { return self->data_pin_number == 0xff; } void common_hal_audiobusio_i2sout_deinit(audiobusio_i2sout_obj_t* self) { if (common_hal_audiobusio_i2sout_deinited(self)) { return; } reset_pin_number(self->bit_clock_pin_number); self->bit_clock_pin_number = 0xff; reset_pin_number(self->word_select_pin_number); self->word_select_pin_number = 0xff; reset_pin_number(self->data_pin_number); self->data_pin_number = 0xff; instance = NULL; } 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 */ false, &single_buffer, &samples_signed, &max_buffer_length, &self->channel_count); self->single_buffer = single_buffer; self->samples_signed = samples_signed; 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->CONFIG.CHANNELS = self->channel_count == 1 ? I2S_CONFIG_CHANNELS_CHANNELS_Left : I2S_CONFIG_CHANNELS_CHANNELS_Stereo; NRF_I2S->RXTXD.MAXCNT = self->buffer_length / 4; 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; } 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 (NRF_I2S->EVENTS_TXPTRUPD) { NRF_I2S->EVENTS_TXPTRUPD = 0; if (instance) { i2s_buffer_fill(instance); } else { NRF_I2S->TASKS_STOP = 1; } } } void i2s_reset(void) { NRF_I2S->TASKS_STOP = 1; 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; instance = NULL; }