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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 "shared-bindings/microcontroller/__init__.h"
#include "bindings/rp2pio/StateMachine.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_AUDIOCORE
void audio_dma_reset(void) {
for (size_t channel = 0; channel < NUM_DMA_CHANNELS; channel++) {
if (MP_STATE_PORT(playing_audio)[channel] == NULL) {
continue;
}
audio_dma_stop(MP_STATE_PORT(playing_audio)[channel]);
}
}
STATIC size_t audio_dma_convert_samples(audio_dma_t *dma, uint8_t *input, uint32_t input_length, uint8_t *output, uint32_t output_length) {
#pragma GCC diagnostic push
#pragma GCC diagnostic ignored "-Wcast-align"
uint32_t output_length_used = input_length / dma->sample_spacing;
if (output_length_used > output_length) {
mp_raise_RuntimeError(translate("Internal audio buffer too small"));
}
uint32_t out_i = 0;
if (dma->sample_resolution <= 8 && dma->output_resolution > 8) {
// reading bytes, writing 16-bit words, so output buffer will be bigger.
output_length_used *= 2;
if (output_length_used > output_length) {
mp_raise_RuntimeError(translate("Internal audio buffer too small"));
}
// Correct "rail-to-rail" scaling of arbitrary-depth input to output
// requires more operations than this, but at least the vital 8- to
// 16-bit cases are correctly scaled now. Prior code was only
// considering 8-to-16 anyway, but had a slight DC offset in the
// result, so this is no worse off WRT supported resolutions.
uint16_t mul = ((1 << dma->output_resolution) - 1) / ((1 << dma->sample_resolution) - 1);
uint16_t offset = (1 << dma->output_resolution) / 2;
for (uint32_t i = 0; i < input_length; i += dma->sample_spacing) {
if (dma->signed_to_unsigned) {
((uint16_t *)output)[out_i] = (uint16_t)((((int8_t *)input)[i] + 0x80) * mul);
} else if (dma->unsigned_to_signed) {
((int16_t *)output)[out_i] = (int16_t)(((uint8_t *)input)[i] * mul - offset);
} else {
((uint16_t *)output)[out_i] = (uint16_t)(((uint8_t *)input)[i] * mul);
}
out_i += 1;
}
} else if (dma->sample_resolution <= 8 && dma->output_resolution <= 8) {
for (uint32_t i = 0; i < input_length; i += dma->sample_spacing) {
if (dma->signed_to_unsigned) {
((uint8_t *)output)[out_i] = ((int8_t *)input)[i] + 0x80;
} else if (dma->unsigned_to_signed) {
((int8_t *)output)[out_i] = ((uint8_t *)input)[i] - 0x80;
} else {
((uint8_t *)output)[out_i] = ((uint8_t *)input)[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 < input_length / 2; i += dma->sample_spacing) {
if (dma->signed_to_unsigned) {
((uint16_t *)output)[out_i] = ((int16_t *)input)[i] + 0x8000;
} else if (dma->unsigned_to_signed) {
((int16_t *)output)[out_i] = ((uint16_t *)input)[i] - 0x8000;
} else {
((uint16_t *)output)[out_i] = ((uint16_t *)input)[i];
}
if (dma->output_resolution < 16) {
if (dma->output_signed) {
((int16_t *)output)[out_i] = ((int16_t *)output)[out_i] >> shift;
} else {
((uint16_t *)output)[out_i] = ((uint16_t *)output)[out_i] >> shift;
}
}
out_i += 1;
}
} else {
// (dma->sample_resolution > 8 && dma->output_resolution <= 8)
// Not currently used, but might be in the future.
mp_raise_RuntimeError(translate("Audio conversion not implemented"));
}
#pragma GCC diagnostic pop
return output_length_used;
}
// buffer_idx is 0 or 1.
STATIC void audio_dma_load_next_block(audio_dma_t *dma, size_t buffer_idx) {
size_t dma_channel = dma->channel[buffer_idx];
audioio_get_buffer_result_t get_buffer_result;
uint8_t *sample_buffer;
uint32_t sample_buffer_length;
get_buffer_result = audiosample_get_buffer(dma->sample,
dma->single_channel_output, dma->audio_channel, &sample_buffer, &sample_buffer_length);
if (get_buffer_result == GET_BUFFER_ERROR) {
audio_dma_stop(dma);
return;
}
// Convert the sample format resolution and signedness, as necessary.
// The input sample buffer is what was read from a file, Mixer, or a raw sample buffer.
// The output buffer is one of the DMA buffers (passed in).
size_t output_length_used = audio_dma_convert_samples(
dma, sample_buffer, sample_buffer_length,
dma->buffer[buffer_idx], dma->buffer_length[buffer_idx]);
dma_channel_set_read_addr(dma_channel, dma->buffer[buffer_idx], false /* trigger */);
dma_channel_set_trans_count(dma_channel, output_length_used / dma->output_size, false /* trigger */);
if (get_buffer_result == GET_BUFFER_DONE) {
if (dma->loop) {
audiosample_reset_buffer(dma->sample, dma->single_channel_output, 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);
if (output_length_used == 0 &&
!dma_channel_is_busy(dma->channel[0]) &&
!dma_channel_is_busy(dma->channel[1])) {
// No data has been read, and both DMA channels have now finished, so it's safe to stop.
audio_dma_stop(dma);
dma->playing_in_progress = false;
}
}
}
}
// 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_output,
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 play because the DMA can't wrap to itself without the
// buffer being power of two aligned.
int dma_channel_0_maybe = dma_claim_unused_channel(false);
if (dma_channel_0_maybe < 0) {
return AUDIO_DMA_DMA_BUSY;
}
int dma_channel_1_maybe = dma_claim_unused_channel(false);
if (dma_channel_1_maybe < 0) {
dma_channel_unclaim((uint)dma_channel_0_maybe);
return AUDIO_DMA_DMA_BUSY;
}
dma->channel[0] = (uint8_t)dma_channel_0_maybe;
dma->channel[1] = (uint8_t)dma_channel_1_maybe;
dma->sample = sample;
dma->loop = loop;
dma->single_channel_output = single_channel_output;
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->output_resolution = output_resolution;
dma->sample_resolution = audiosample_bits_per_sample(sample);
dma->output_register_address = output_register_address;
audiosample_reset_buffer(sample, single_channel_output, audio_channel);
bool single_buffer; // True if data fits in one single buffer.
bool samples_signed;
uint32_t max_buffer_length;
audiosample_get_buffer_structure(sample, single_channel_output, &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->buffer[0] = (uint8_t *)m_realloc(dma->buffer[0], max_buffer_length);
dma->buffer_length[0] = max_buffer_length;
if (dma->buffer[0] == NULL) {
return AUDIO_DMA_MEMORY_ERROR;
}
if (!single_buffer) {
dma->buffer[1] = (uint8_t *)m_realloc(dma->buffer[1], max_buffer_length);
dma->buffer_length[1] = max_buffer_length;
if (dma->buffer[1] == 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_output && 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, 0);
if (!single_buffer) {
audio_dma_load_next_block(dma, 1);
}
// 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->buffer[0], // read address
1, // transaction count
false); // trigger
} else {
// Enable our DMA channels on DMA_IRQ_0 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->playing_in_progress = true;
dma_channel_start(dma->channel[0]);
return AUDIO_DMA_OK;
}
void audio_dma_stop(audio_dma_t *dma) {
// Disable our interrupts.
uint32_t channel_mask = 0;
if (dma->channel[0] < NUM_DMA_CHANNELS) {
channel_mask |= 1 << dma->channel[0];
}
if (dma->channel[1] < NUM_DMA_CHANNELS) {
channel_mask |= 1 << dma->channel[1];
}
dma_hw->inte0 &= ~channel_mask;
if (!dma_hw->inte0) {
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 (channel == NUM_DMA_CHANNELS) {
// Channel not in use.
continue;
}
dma_channel_config c = dma_channel_get_default_config(dma->channel[i]);
channel_config_set_enable(&c, false);
dma_channel_set_config(channel, &c, false /* trigger */);
if (dma_channel_is_busy(channel)) {
dma_channel_abort(channel);
}
dma_channel_set_read_addr(channel, NULL, false /* trigger */);
dma_channel_set_write_addr(channel, NULL, false /* trigger */);
dma_channel_set_trans_count(channel, 0, false /* trigger */);
dma_channel_unclaim(channel);
MP_STATE_PORT(playing_audio)[channel] = NULL;
dma->channel[i] = NUM_DMA_CHANNELS;
}
dma->playing_in_progress = false;
// 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_CH1_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.)
if (dma_channel_is_busy(dma->channel[0])) {
dma_hw->ch[dma->channel[1]].al1_ctrl |= DMA_CH1_CTRL_TRIG_EN_BITS;
dma_hw->ch[dma->channel[0]].al1_ctrl |= DMA_CH0_CTRL_TRIG_EN_BITS;
} else {
dma_hw->ch[dma->channel[0]].al1_ctrl |= DMA_CH0_CTRL_TRIG_EN_BITS;
dma_hw->ch[dma->channel[1]].al1_ctrl |= DMA_CH1_CTRL_TRIG_EN_BITS;
}
}
bool audio_dma_get_paused(audio_dma_t *dma) {
if (dma->channel[0] >= NUM_DMA_CHANNELS) {
return false;
}
uint32_t control = dma_hw->ch[dma->channel[0]].ctrl_trig;
return (control & DMA_CH0_CTRL_TRIG_EN_BITS) == 0;
}
uint32_t audio_dma_pause_all(void) {
uint32_t result = 0;
for (size_t channel = 0; channel < NUM_DMA_CHANNELS; channel++) {
audio_dma_t *dma = MP_STATE_PORT(playing_audio)[channel];
if (dma != NULL && !audio_dma_get_paused(dma)) {
audio_dma_pause(dma);
result |= (1 << channel);
}
}
return result;
}
void audio_dma_unpause_mask(uint32_t channel_mask) {
for (size_t channel = 0; channel < NUM_DMA_CHANNELS; channel++) {
audio_dma_t *dma = MP_STATE_PORT(playing_audio)[channel];
if (dma != NULL && (channel_mask & (1 << channel))) {
audio_dma_resume(dma);
}
}
}
void audio_dma_init(audio_dma_t *dma) {
dma->buffer[0] = NULL;
dma->buffer[1] = NULL;
dma->channel[0] = NUM_DMA_CHANNELS;
dma->channel[1] = NUM_DMA_CHANNELS;
}
void audio_dma_deinit(audio_dma_t *dma) {
m_free(dma->buffer[0]);
dma->buffer[0] = NULL;
m_free(dma->buffer[1]);
dma->buffer[1] = NULL;
}
bool audio_dma_get_playing(audio_dma_t *dma) {
if (dma->channel[0] == NUM_DMA_CHANNELS) {
return false;
}
return dma->playing_in_progress;
}
// WARN(tannewt): DO NOT print from here, or anything it calls. Printing calls
// background tasks such as this and causes a stack overflow.
// NOTE(dhalbert): I successfully printed from here while debugging.
// So it's possible, but be careful.
STATIC void dma_callback_fun(void *arg) {
audio_dma_t *dma = arg;
if (dma == NULL) {
return;
}
common_hal_mcu_disable_interrupts();
uint32_t channels_to_load_mask = dma->channels_to_load_mask;
dma->channels_to_load_mask = 0;
common_hal_mcu_enable_interrupts();
// Load the blocks for the requested channels.
uint32_t channel = 0;
while (channels_to_load_mask) {
if (channels_to_load_mask & 1) {
if (dma->channel[0] == channel) {
audio_dma_load_next_block(dma, 0);
}
if (dma->channel[1] == channel) {
audio_dma_load_next_block(dma, 1);
}
}
channels_to_load_mask >>= 1;
channel++;
}
}
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) {
continue;
}
// acknowledge interrupt early. Doing so late means that you could lose an
// interrupt if the buffer is very small and the DMA operation
// completed by the time callback_add() / dma_complete() returned. This
// affected PIO continuous write more than audio.
dma_hw->ints0 = mask;
if (MP_STATE_PORT(playing_audio)[i] != NULL) {
audio_dma_t *dma = MP_STATE_PORT(playing_audio)[i];
// Record all channels whose DMA has completed; they need loading.
dma->channels_to_load_mask |= mask;
background_callback_add(&dma->callback, dma_callback_fun, (void *)dma);
}
if (MP_STATE_PORT(background_pio)[i] != NULL) {
rp2pio_statemachine_obj_t *pio = MP_STATE_PORT(background_pio)[i];
rp2pio_statemachine_dma_complete(pio, i);
}
}
}
MP_REGISTER_ROOT_POINTER(mp_obj_t playing_audio[enum_NUM_DMA_CHANNELS]);
#endif
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