/* * This file is part of the MicroPython project, http://micropython.org/ * * The MIT License (MIT) * * Copyright (c) 2020 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 "py/runtime.h" #include "common-hal/audiobusio/__init__.h" #include "bindings/espidf/__init__.h" #include "freertos/FreeRTOS.h" #include "freertos/task.h" #include "shared-module/audiocore/__init__.h" #define CIRCUITPY_BUFFER_COUNT 3 #define CIRCUITPY_BUFFER_SIZE 1023 #define CIRCUITPY_OUTPUT_SLOTS 2 static void i2s_fill_buffer(i2s_t *self) { if (self->next_buffer_size == 0) { // Error, no new buffer queued. return; } int16_t *output_buffer = (int16_t *)self->next_buffer; size_t output_buffer_size = self->next_buffer_size; const size_t bytes_per_output_frame = 4; size_t bytes_per_input_frame = self->channel_count * self->bytes_per_sample; if (!self->playing || self->paused || !self->sample || self->stopping) { memset(output_buffer, 0, self->next_buffer_size); self->next_buffer = NULL; self->next_buffer_size = 0; return; } while (!self->stopping && output_buffer_size > 0) { 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; } } size_t sample_bytecount = self->sample_end - self->sample_data; // The framecount is the minimum of space left in the output buffer or left in the incoming sample. size_t framecount = MIN(output_buffer_size / bytes_per_output_frame, sample_bytecount / bytes_per_input_frame); if (self->samples_signed && self->channel_count == 2) { if (self->bytes_per_sample == 2) { memcpy(output_buffer, self->sample_data, framecount * bytes_per_output_frame); } else { audiosample_convert_s8s_s16s(output_buffer, ((int8_t *)self->sample_data), framecount); } } else { if (self->samples_signed) { assert(self->channel_count == 1); if (self->bytes_per_sample == 1) { audiosample_convert_s8m_s16s(output_buffer, (int8_t *)(void *)self->sample_data, framecount); } else { audiosample_convert_s16m_s16s(output_buffer, (int16_t *)(void *)self->sample_data, framecount); } } else { if (self->channel_count == 1) { if (self->bytes_per_sample == 1) { audiosample_convert_u8m_s16s(output_buffer, (uint8_t *)(void *)self->sample_data, framecount); } else { audiosample_convert_u16m_s16s(output_buffer, (uint16_t *)(void *)self->sample_data, framecount); } } else { if (self->bytes_per_sample == 1) { audiosample_convert_u8s_s16s(output_buffer, (uint8_t *)(void *)self->sample_data, framecount); } else { audiosample_convert_u16s_s16s(output_buffer, (uint16_t *)(void *)self->sample_data, framecount); } } } } self->sample_data += framecount * bytes_per_input_frame; output_buffer += framecount * CIRCUITPY_OUTPUT_SLOTS; output_buffer_size -= framecount * bytes_per_output_frame; } self->next_buffer = NULL; self->next_buffer_size = 0; } static void i2s_callback_fun(void *self_in) { i2s_t *self = self_in; i2s_fill_buffer(self); } static bool i2s_event_interrupt(i2s_chan_handle_t handle, i2s_event_data_t *event, void *self_in) { i2s_t *self = self_in; self->underrun = self->underrun || self->next_buffer != NULL; self->next_buffer = *(int16_t **)event->data; self->next_buffer_size = event->size; background_callback_add(&self->callback, i2s_callback_fun, self_in); return false; } void port_i2s_allocate_init(i2s_t *self, bool left_justified) { i2s_chan_config_t chan_config = { .id = I2S_NUM_AUTO, .role = I2S_ROLE_MASTER, .dma_desc_num = CIRCUITPY_BUFFER_COUNT, .dma_frame_num = CIRCUITPY_BUFFER_SIZE, // in _frames_, so 1023 is 4092 bytes per dma buf which is the maximum }; esp_err_t err = i2s_new_channel(&chan_config, &self->handle, NULL); if (err == ESP_ERR_NOT_FOUND) { mp_raise_RuntimeError(translate("Peripheral in use")); } i2s_event_callbacks_t callbacks = { .on_recv = NULL, .on_recv_q_ovf = NULL, .on_sent = i2s_event_interrupt, .on_send_q_ovf = NULL, }; i2s_channel_register_event_callback(self->handle, &callbacks, self); } void port_i2s_deinit(i2s_t *self) { port_i2s_stop(self); i2s_del_channel(self->handle); self->handle = NULL; } void port_i2s_play(i2s_t *self, mp_obj_t sample, bool loop) { self->sample = sample; self->loop = loop; self->bytes_per_sample = audiosample_bits_per_sample(sample) / 8; self->channel_count = audiosample_channel_count(sample); bool single_buffer; bool samples_signed; uint32_t max_buffer_length; uint8_t spacing; audiosample_get_buffer_structure(sample, false, &single_buffer, &samples_signed, &max_buffer_length, &spacing); self->samples_signed = samples_signed; self->sample_data = self->sample_end = NULL; // We always output stereo so output twice as many bits. // uint16_t bits_per_sample_output = bits_per_sample * 2; audiosample_reset_buffer(self->sample, false, 0); uint32_t sample_rate = audiosample_sample_rate(sample); i2s_std_clk_config_t clk_config = I2S_STD_CLK_DEFAULT_CONFIG(sample_rate); CHECK_ESP_RESULT(i2s_channel_reconfig_std_clock(self->handle, &clk_config)); // preload the data self->playing = true; self->paused = false; self->stopping = false; // This will be slow but we can't rewind the underlying sample. So, we will // preload one frame at a time and drop the last sample that can't fit. // We cap ourselves at the max DMA set to prevent a sample drop if starting // fresh. uint32_t starting_frame; size_t bytes_loaded = 4; size_t preloaded = 0; while (bytes_loaded > 0 && preloaded < CIRCUITPY_BUFFER_SIZE * CIRCUITPY_BUFFER_COUNT) { self->next_buffer = &starting_frame; self->next_buffer_size = sizeof(starting_frame); i2s_fill_buffer(self); i2s_channel_preload_data(self->handle, &starting_frame, sizeof(uint32_t), &bytes_loaded); preloaded += 1; } // enable the channel i2s_channel_enable(self->handle); // The IDF will call us back when there is a free DMA buffer. } bool port_i2s_playing(i2s_t *self) { return self->playing && !self->stopping; } bool port_i2s_paused(i2s_t *self) { return self->paused; } void port_i2s_stop(i2s_t *self) { port_i2s_pause(self); self->sample = NULL; self->playing = false; self->stopping = false; } void port_i2s_pause(i2s_t *self) { if (!self->paused) { self->paused = true; CHECK_ESP_RESULT(i2s_channel_disable(self->handle)); } } void port_i2s_resume(i2s_t *self) { if (self->paused) { self->paused = false; CHECK_ESP_RESULT(i2s_channel_enable(self->handle)); } }