circuitpython/ports/raspberrypi/audio_dma.c

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/*
* This file is part of the MicroPython project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2021 Scott Shawcroft 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 "audio_dma.h"
#include "shared-bindings/audiocore/RawSample.h"
#include "shared-bindings/audiocore/WaveFile.h"
#include "supervisor/background_callback.h"
#include "py/mpstate.h"
#include "py/runtime.h"
#include "src/rp2_common/hardware_irq/include/hardware/irq.h"
#if CIRCUITPY_AUDIOPWMIO || CIRCUITPY_AUDIOBUSIO
#define AUDIO_DMA_CHANNEL_COUNT NUM_DMA_CHANNELS
void audio_dma_convert_signed(audio_dma_t* dma, uint8_t* buffer, uint32_t buffer_length,
uint8_t** output_buffer, uint32_t* output_buffer_length) {
if (dma->first_buffer_free) {
*output_buffer = dma->first_buffer;
} else {
*output_buffer = dma->second_buffer;
}
#pragma GCC diagnostic push
#pragma GCC diagnostic ignored "-Wcast-align"
if (dma->signed_to_unsigned ||
dma->unsigned_to_signed ||
dma->sample_spacing > 1 ||
(dma->sample_resolution != dma->output_resolution)) {
*output_buffer_length = buffer_length / dma->sample_spacing;
uint32_t out_i = 0;
if (dma->sample_resolution <= 8 && dma->output_resolution > 8) {
size_t shift = dma->output_resolution - dma->sample_resolution;
for (uint32_t i = 0; i < buffer_length; i += dma->sample_spacing) {
if (dma->signed_to_unsigned) {
((uint16_t*) *output_buffer)[out_i] = ((uint16_t) ((int8_t*) buffer)[i] + 0x80) << shift;
} else if (dma->unsigned_to_signed) {
((int16_t*) *output_buffer)[out_i] = ((int16_t) ((uint8_t*) buffer)[i] - 0x80) << shift;
} else {
((uint16_t*) *output_buffer)[out_i] = ((uint16_t) ((uint8_t*) buffer)[i]) << shift;
}
out_i += 1;
}
} else if (dma->sample_resolution <= 8 && dma->output_resolution <= 8) {
for (uint32_t i = 0; i < buffer_length; i += dma->sample_spacing) {
if (dma->signed_to_unsigned) {
((uint8_t*) *output_buffer)[out_i] = ((int8_t*) buffer)[i] + 0x80;
} else if (dma->unsigned_to_signed) {
((int8_t*) *output_buffer)[out_i] = ((uint8_t*) buffer)[i] - 0x80;
} else {
((uint8_t*) *output_buffer)[out_i] = ((uint8_t*) buffer)[i];
}
out_i += 1;
}
} else if (dma->sample_resolution > 8 && dma->output_resolution > 8) {
size_t shift = 16 - dma->output_resolution;
for (uint32_t i = 0; i < buffer_length / 2; i += dma->sample_spacing) {
if (dma->signed_to_unsigned) {
((uint16_t*) *output_buffer)[out_i] = ((int16_t*) buffer)[i] + 0x8000;
} else if (dma->unsigned_to_signed) {
((int16_t*) *output_buffer)[out_i] = ((uint16_t*) buffer)[i] - 0x8000;
} else {
((uint16_t*) *output_buffer)[out_i] = ((uint16_t*) buffer)[i];
}
if (dma->output_resolution < 16) {
if (dma->output_signed) {
((int16_t*) *output_buffer)[out_i] = ((int16_t*) *output_buffer)[out_i] >> shift;
} else {
((uint16_t*) *output_buffer)[out_i] = ((uint16_t*) *output_buffer)[out_i] >> shift;
}
}
out_i += 1;
}
}
} else {
*output_buffer = buffer;
*output_buffer_length = buffer_length;
}
#pragma GCC diagnostic pop
dma->first_buffer_free = !dma->first_buffer_free;
}
void audio_dma_load_next_block(audio_dma_t* dma) {
uint8_t dma_channel = dma->channel[1];
if (dma->first_channel_free) {
dma_channel = dma->channel[0];
}
dma->first_channel_free = !dma->first_channel_free;
uint8_t* output_buffer;
uint32_t output_buffer_length;
audioio_get_buffer_result_t get_buffer_result;
uint8_t* buffer;
uint32_t buffer_length;
get_buffer_result = audiosample_get_buffer(dma->sample,
dma->single_channel, dma->audio_channel, &buffer, &buffer_length);
if (get_buffer_result == GET_BUFFER_ERROR) {
audio_dma_stop(dma);
return;
}
audio_dma_convert_signed(dma, buffer, buffer_length, &output_buffer, &output_buffer_length);
// If we don't have an output buffer, save the pointer to first_buffer for use in the single
// buffer special case.
if (dma->first_buffer == NULL) {
dma->first_buffer = output_buffer;
}
dma_channel_set_trans_count(dma_channel, output_buffer_length / dma->output_size, false /* trigger */);
dma_channel_set_read_addr(dma_channel, output_buffer, false /* trigger */);
if (get_buffer_result == GET_BUFFER_DONE) {
if (dma->loop) {
audiosample_reset_buffer(dma->sample, dma->single_channel, dma->audio_channel);
} else {
// Set channel trigger to ourselves so we don't keep going.
dma_channel_hw_t* c = &dma_hw->ch[dma_channel];
c->al1_ctrl = (c->al1_ctrl & ~DMA_CH0_CTRL_TRIG_CHAIN_TO_BITS) | (dma_channel << DMA_CH0_CTRL_TRIG_CHAIN_TO_LSB);
}
}
}
// Playback should be shutdown before calling this.
audio_dma_result audio_dma_setup_playback(audio_dma_t* dma,
mp_obj_t sample,
bool loop,
bool single_channel,
uint8_t audio_channel,
bool output_signed,
uint8_t output_resolution,
uint32_t output_register_address,
uint8_t dma_trigger_source) {
// Use two DMA channels to because the DMA can't wrap to itself without the
// buffer being power of two aligned.
dma->channel[0] = dma_claim_unused_channel(false);
dma->channel[1] = dma_claim_unused_channel(false);
if (dma->channel[0] == NUM_DMA_CHANNELS || dma->channel[1] == NUM_DMA_CHANNELS) {
if (dma->channel[0] < NUM_DMA_CHANNELS) {
dma_channel_unclaim(dma->channel[0]);
}
return AUDIO_DMA_DMA_BUSY;
}
dma->sample = sample;
dma->loop = loop;
dma->single_channel = single_channel;
dma->audio_channel = audio_channel;
dma->signed_to_unsigned = false;
dma->unsigned_to_signed = false;
dma->output_signed = output_signed;
dma->sample_spacing = 1;
dma->first_channel_free = true;
dma->output_resolution = output_resolution;
dma->sample_resolution = audiosample_bits_per_sample(sample);
audiosample_reset_buffer(sample, single_channel, audio_channel);
bool single_buffer;
bool samples_signed;
uint32_t max_buffer_length;
audiosample_get_buffer_structure(sample, single_channel, &single_buffer, &samples_signed,
&max_buffer_length, &dma->sample_spacing);
// Check to see if we have to scale the resolution up.
if (dma->sample_resolution <= 8 && dma->output_resolution > 8) {
max_buffer_length *= 2;
}
if (output_signed != samples_signed ||
dma->sample_spacing > 1 ||
(dma->sample_resolution != dma->output_resolution)) {
max_buffer_length /= dma->sample_spacing;
dma->first_buffer = (uint8_t*) m_realloc(dma->first_buffer, max_buffer_length);
if (dma->first_buffer == NULL) {
return AUDIO_DMA_MEMORY_ERROR;
}
dma->first_buffer_free = true;
if (!single_buffer) {
dma->second_buffer = (uint8_t*) m_realloc(dma->second_buffer, max_buffer_length);
if (dma->second_buffer == NULL) {
return AUDIO_DMA_MEMORY_ERROR;
}
}
dma->signed_to_unsigned = !output_signed && samples_signed;
dma->unsigned_to_signed = output_signed && !samples_signed;
}
if (output_resolution > 8) {
dma->output_size = 2;
} else {
dma->output_size = 1;
}
// Transfer both channels at once.
if (!single_channel && audiosample_channel_count(sample) == 2) {
dma->output_size *= 2;
}
enum dma_channel_transfer_size dma_size = DMA_SIZE_8;
if (dma->output_size == 2) {
dma_size = DMA_SIZE_16;
} else if (dma->output_size == 4) {
dma_size = DMA_SIZE_32;
}
for (size_t i = 0; i < 2; i++) {
dma_channel_config c = dma_channel_get_default_config(dma->channel[i]);
channel_config_set_transfer_data_size(&c, dma_size);
channel_config_set_dreq(&c, dma_trigger_source);
channel_config_set_read_increment(&c, true);
channel_config_set_write_increment(&c, false);
// Chain to the other channel by default.
channel_config_set_chain_to(&c, dma->channel[(i + 1) % 2]);
dma_channel_set_config(dma->channel[i], &c, false /* trigger */);
dma_channel_set_write_addr(dma->channel[i], (void*) output_register_address, false /* trigger */);
}
// We keep the audio_dma_t for internal use and the sample as a root pointer because it
// contains the audiodma structure.
MP_STATE_PORT(playing_audio)[dma->channel[0]] = dma;
MP_STATE_PORT(playing_audio)[dma->channel[1]] = dma;
// Load the first two blocks up front.
audio_dma_load_next_block(dma);
if (!single_buffer) {
audio_dma_load_next_block(dma);
}
// Special case the DMA for a single buffer. It's commonly used for a single wave length of sound
// and may be short. Therefore, we use DMA chaining to loop quickly without involving interrupts.
// On the RP2040 we chain by having a second DMA writing to the config registers of the first.
// Read and write addresses change with DMA so we need to reset the read address back to the
// start of the sample.
if (single_buffer) {
dma_channel_config c = dma_channel_get_default_config(dma->channel[1]);
channel_config_set_transfer_data_size(&c, DMA_SIZE_32);
channel_config_set_dreq(&c, 0x3f); // dma as fast as possible
channel_config_set_read_increment(&c, false);
channel_config_set_write_increment(&c, false);
channel_config_set_chain_to(&c, dma->channel[1]); // Chain to ourselves so we stop.
dma_channel_configure(dma->channel[1], &c,
&dma_hw->ch[dma->channel[0]].al3_read_addr_trig, // write address
&dma->first_buffer, // read address
1, // transaction count
false); // trigger
} else {
// Enable our DMA channels on DMA0 to the CPU. This will wake us up when
// we're WFI.
dma_hw->inte0 |= (1 << dma->channel[0]) | (1 << dma->channel[1]);
irq_set_mask_enabled(1 << DMA_IRQ_0, true);
}
dma_channel_start(dma->channel[0]);
return AUDIO_DMA_OK;
}
void audio_dma_stop(audio_dma_t* dma) {
// Disable our interrupts.
dma_hw->inte0 &= ~((1 << dma->channel[0]) | (1 << dma->channel[1]));
irq_set_mask_enabled(1 << DMA_IRQ_0, false);
// Run any remaining audio tasks because we remove ourselves from
// playing_audio.
RUN_BACKGROUND_TASKS;
for (size_t i = 0; i < 2; i++) {
size_t channel = dma->channel[i];
if (dma_channel_is_busy(channel)) {
dma_channel_abort(channel);
}
dma_channel_unclaim(channel);
MP_STATE_PORT(playing_audio)[channel] = NULL;
dma->channel[i] = NUM_DMA_CHANNELS;
}
// Hold onto our buffers.
}
// To pause we simply stop the DMA. It is the responsibility of the output peripheral
// to hold the previous value.
void audio_dma_pause(audio_dma_t* dma) {
dma_hw->ch[dma->channel[0]].al1_ctrl &= ~DMA_CH0_CTRL_TRIG_EN_BITS;
dma_hw->ch[dma->channel[1]].al1_ctrl &= ~DMA_CH0_CTRL_TRIG_EN_BITS;
}
void audio_dma_resume(audio_dma_t* dma) {
// Always re-enable the non-busy channel first so it's ready to continue when the busy channel
// finishes and chains to it. (An interrupt could make the time between enables long.)
size_t first = 0;
size_t second = 1;
if (dma_channel_is_busy(dma->channel[0])) {
first = 1;
second = 0;
}
dma_hw->ch[dma->channel[first]].al1_ctrl |= DMA_CH0_CTRL_TRIG_EN_BITS;
dma_hw->ch[dma->channel[second]].al1_ctrl |= DMA_CH0_CTRL_TRIG_EN_BITS;
}
bool audio_dma_get_paused(audio_dma_t* dma) {
if (dma->channel[0] >= AUDIO_DMA_CHANNEL_COUNT) {
return false;
}
uint32_t control = dma_hw->ch[dma->channel[0]].ctrl_trig;
return (control & DMA_CH0_CTRL_TRIG_EN_BITS) == 0;
}
void audio_dma_init(audio_dma_t* dma) {
dma->first_buffer = NULL;
dma->second_buffer = NULL;
}
void audio_dma_deinit(audio_dma_t* dma) {
m_free(dma->first_buffer);
dma->first_buffer = NULL;
m_free(dma->second_buffer);
dma->second_buffer = NULL;
}
bool audio_dma_get_playing(audio_dma_t* dma) {
if (dma->channel[0] == NUM_DMA_CHANNELS) {
return false;
}
if (!dma_channel_is_busy(dma->channel[0]) &&
!dma_channel_is_busy(dma->channel[1])) {
audio_dma_stop(dma);
return false;
}
return true;
}
// WARN(tannewt): DO NOT print from here, or anything it calls. Printing calls
// background tasks such as this and causes a stack overflow.
STATIC void dma_callback_fun(void *arg) {
audio_dma_t* dma = arg;
if (dma == NULL) {
return;
}
audio_dma_load_next_block(dma);
}
void isr_dma_0(void) {
for (size_t i = 0; i < NUM_DMA_CHANNELS; i++) {
uint32_t mask = 1 << i;
if ((dma_hw->intr & mask) != 0 && MP_STATE_PORT(playing_audio)[i] != NULL) {
audio_dma_t* dma = MP_STATE_PORT(playing_audio)[i];
background_callback_add(&dma->callback, dma_callback_fun, (void*)dma);
dma_hw->ints0 = mask;
}
}
}
#endif