circuitpython/ports/espressif/common-hal/audiobusio/__init__.c

234 lines
8.9 KiB
C

/*
* 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 <string.h>
#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(MP_ERROR_TEXT("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));
}
}