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Track more carefully which audio buffers to fill, based on interrupt channels

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This commit is contained in:
Dan Halbert 2021-08-10 22:00:09 -04:00
parent 2451c788f4
commit 24e61a7da8
4 changed files with 136 additions and 102 deletions
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185
ports/raspberrypi/audio_dma.c
View File

@ -28,6 +28,7 @@
#include "shared-bindings/audiocore/RawSample.h"
#include "shared-bindings/audiocore/WaveFile.h"
#include "shared-bindings/microcontroller/__init__.h"
#include "supervisor/background_callback.h"
#include "py/mpstate.h"
@ -47,71 +48,78 @@ void audio_dma_reset(void) {
}
}
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) {
size_t output_buffer_max_length;
if (dma->first_buffer_free) {
*output_buffer = dma->first_buffer;
output_buffer_max_length = dma->first_buffer_length;
} else {
*output_buffer = dma->second_buffer;
output_buffer_max_length = dma->second_buffer_length;
}
STATIC void audio_dma_convert_samples(
audio_dma_t *dma,
uint8_t *input, uint32_t input_length,
uint8_t *available_output_buffer, uint32_t available_output_buffer_length,
uint8_t **output, uint32_t *output_length) {
#pragma GCC diagnostic push
#pragma GCC diagnostic ignored "-Wcast-align"
// Check whether a conversion is necessary
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;
// Must convert.
// Write the conversion into the passed-in output buffer
*output = available_output_buffer;
*output_length = input_length / dma->sample_spacing;
if (*output_length > available_output_buffer_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 samples
*output_buffer_length = *output_buffer_length * 2;
if (*output_buffer_length > output_buffer_max_length) {
// reading bytes, writing 16-bit words, so output buffer will be bigger.
*output_length = *output_length * 2;
if (*output_length > available_output_buffer_length) {
mp_raise_RuntimeError(translate("Internal audio buffer too small"));
}
size_t shift = dma->output_resolution - dma->sample_resolution;
for (uint32_t i = 0; i < buffer_length; i += dma->sample_spacing) {
for (uint32_t i = 0; i < input_length; i += dma->sample_spacing) {
if (dma->signed_to_unsigned) {
((uint16_t *)*output_buffer)[out_i] = ((uint16_t)((int8_t *)buffer)[i] + 0x80) << shift;
((uint16_t *)*output)[out_i] = ((uint16_t)((int8_t *)input)[i] + 0x80) << shift;
} else if (dma->unsigned_to_signed) {
((int16_t *)*output_buffer)[out_i] = ((int16_t)((uint8_t *)buffer)[i] - 0x80) << shift;
((int16_t *)*output)[out_i] = ((int16_t)((uint8_t *)input)[i] - 0x80) << shift;
} else {
((uint16_t *)*output_buffer)[out_i] = ((uint16_t)((uint8_t *)buffer)[i]) << shift;
((uint16_t *)*output)[out_i] = ((uint16_t)((uint8_t *)input)[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) {
for (uint32_t i = 0; i < input_length; i += dma->sample_spacing) {
if (dma->signed_to_unsigned) {
((uint8_t *)*output_buffer)[out_i] = ((int8_t *)buffer)[i] + 0x80;
((uint8_t *)*output)[out_i] = ((int8_t *)input)[i] + 0x80;
} else if (dma->unsigned_to_signed) {
((int8_t *)*output_buffer)[out_i] = ((uint8_t *)buffer)[i] - 0x80;
((int8_t *)*output)[out_i] = ((uint8_t *)input)[i] - 0x80;
} else {
((uint8_t *)*output_buffer)[out_i] = ((uint8_t *)buffer)[i];
((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 < buffer_length / 2; i += dma->sample_spacing) {
for (uint32_t i = 0; i < input_length / 2; i += dma->sample_spacing) {
if (dma->signed_to_unsigned) {
((uint16_t *)*output_buffer)[out_i] = ((int16_t *)buffer)[i] + 0x8000;
((uint16_t *)*output)[out_i] = ((int16_t *)input)[i] + 0x8000;
} else if (dma->unsigned_to_signed) {
((int16_t *)*output_buffer)[out_i] = ((uint16_t *)buffer)[i] - 0x8000;
((int16_t *)*output)[out_i] = ((uint16_t *)input)[i] - 0x8000;
} else {
((uint16_t *)*output_buffer)[out_i] = ((uint16_t *)buffer)[i];
((uint16_t *)*output)[out_i] = ((uint16_t *)input)[i];
}
if (dma->output_resolution < 16) {
if (dma->output_signed) {
((int16_t *)*output_buffer)[out_i] = ((int16_t *)*output_buffer)[out_i] >> shift;
((int16_t *)*output)[out_i] = ((int16_t *)*output)[out_i] >> shift;
} else {
((uint16_t *)*output_buffer)[out_i] = ((uint16_t *)*output_buffer)[out_i] >> shift;
((uint16_t *)*output)[out_i] = ((uint16_t *)*output)[out_i] >> shift;
}
}
out_i += 1;
@ -122,65 +130,70 @@ void audio_dma_convert_signed(audio_dma_t *dma, uint8_t *buffer, uint32_t buffer
mp_raise_RuntimeError(translate("Audio conversion not implemented"));
}
} else {
*output_buffer = buffer;
*output_buffer_length = buffer_length;
// No conversion necessary. Designate the input buffer as the output buffer.
*output = input;
*output_length = input_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;
// channel_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];
uint8_t *output_buffer;
uint32_t output_buffer_length;
audioio_get_buffer_result_t get_buffer_result;
uint8_t *buffer;
uint32_t buffer_length;
uint8_t *sample_buffer;
uint32_t sample_buffer_length;
get_buffer_result = audiosample_get_buffer(dma->sample,
dma->single_channel_output, dma->audio_channel, &buffer, &buffer_length);
dma->single_channel_output, dma->audio_channel, &sample_buffer, &sample_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);
// Convert the sample format resolution and signedness, as necessary.
// The input sample buffer is what was read from a file or a raw sample buffer.
// The output buffer is one of the DMA buffers (passed in), or if no conversion was done,
// the original sample buffer (to save copying).
// 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;
}
// audio_dma_convert_samples() will write the converted samples into the given output
// buffer if necessary. If no conversion was needed, it will return the sample buffer
// as the output buffer.
uint8_t *output_buffer;
uint32_t output_buffer_length;
audio_dma_convert_samples(dma, sample_buffer, sample_buffer_length,
dma->buffer[buffer_idx], dma->buffer_length[buffer_idx],
&output_buffer, &output_buffer_length);
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 */);
dma_channel_set_trans_count(dma_channel, output_buffer_length / 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_buffer_length == 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;
} 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,
audio_dma_result audio_dma_setup_playback(
audio_dma_t *dma,
mp_obj_t sample,
bool loop,
bool single_channel_output,
@ -189,6 +202,7 @@ audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
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);
@ -213,14 +227,15 @@ audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
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);
dma->output_register_address = output_register_address;
audiosample_reset_buffer(sample, single_channel_output, audio_channel);
bool single_buffer;
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,
@ -236,17 +251,16 @@ audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
max_buffer_length /= dma->sample_spacing;
}
dma->first_buffer = (uint8_t *)m_realloc(dma->first_buffer, max_buffer_length);
dma->first_buffer_length = max_buffer_length;
if (dma->first_buffer == NULL) {
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;
}
dma->first_buffer_free = true;
if (!single_buffer) {
dma->second_buffer = (uint8_t *)m_realloc(dma->second_buffer, max_buffer_length);
dma->second_buffer_length = max_buffer_length;
if (dma->second_buffer == NULL) {
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;
}
}
@ -276,9 +290,11 @@ audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
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 */);
}
@ -288,9 +304,9 @@ audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
MP_STATE_PORT(playing_audio)[dma->channel[1]] = dma;
// Load the first two blocks up front.
audio_dma_load_next_block(dma);
audio_dma_load_next_block(dma, 0);
if (!single_buffer) {
audio_dma_load_next_block(dma);
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
@ -307,11 +323,11 @@ audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
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
&dma->buffer[0], // read address
1, // transaction count
false); // trigger
} else {
// Enable our DMA channels on DMA0 to the CPU. This will wake us up when
// 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);
@ -402,18 +418,19 @@ bool audio_dma_get_paused(audio_dma_t *dma) {
}
void audio_dma_init(audio_dma_t *dma) {
dma->first_buffer = NULL;
dma->second_buffer = NULL;
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->first_buffer);
dma->first_buffer = NULL;
m_free(dma->buffer[0]);
dma->buffer[0] = NULL;
m_free(dma->second_buffer);
dma->second_buffer = NULL;
m_free(dma->buffer[1]);
dma->buffer[1] = NULL;
}
bool audio_dma_get_playing(audio_dma_t *dma) {
@ -433,7 +450,25 @@ STATIC void dma_callback_fun(void *arg) {
return;
}
audio_dma_load_next_block(dma);
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) {
@ -441,6 +476,8 @@ void isr_dma_0(void) {
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];
// 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);
dma_hw->ints0 = mask;
}

9
ports/raspberrypi/audio_dma.h
View File

@ -43,15 +43,12 @@ typedef struct {
bool signed_to_unsigned;
bool unsigned_to_signed;
bool output_signed;
bool first_channel_free;
bool first_buffer_free;
bool playing_in_progress;
uint8_t output_resolution; // in bits
uint8_t sample_resolution; // in bits
uint8_t *first_buffer;
size_t first_buffer_length;
uint8_t *second_buffer;
size_t second_buffer_length;
uint8_t *buffer[2];
size_t buffer_length[2];
uint32_t channels_to_load_mask;
uint32_t output_register_address;
background_callback_t callback;
} audio_dma_t;

42
ports/raspberrypi/common-hal/audiopwmio/PWMAudioOut.c
View File

@ -157,27 +157,6 @@ void common_hal_audiopwmio_pwmaudioout_play(audiopwmio_pwmaudioout_obj_t *self,
tx_register += self->left_pwm.channel * sizeof(uint16_t);
}
audio_dma_result result = audio_dma_setup_playback(
&self->dma,
sample,
loop,
false, // single channel
0, // audio channel
false, // output signed
BITS_PER_SAMPLE,
(uint32_t)tx_register, // output register: PWM cc register
0x3b + pacing_timer); // data request line
if (result == AUDIO_DMA_DMA_BUSY) {
common_hal_audiopwmio_pwmaudioout_stop(self);
mp_raise_RuntimeError(translate("No DMA channel found"));
}
if (result == AUDIO_DMA_MEMORY_ERROR) {
common_hal_audiopwmio_pwmaudioout_stop(self);
mp_raise_RuntimeError(translate("Unable to allocate buffers for signed conversion"));
}
// OK! We got all of the resources we need and dma is ready.
self->pacing_timer = pacing_timer;
// Playback with two independent clocks. One is the sample rate which
@ -214,6 +193,27 @@ void common_hal_audiopwmio_pwmaudioout_play(audiopwmio_pwmaudioout_obj_t *self,
}
dma_hw->timer[pacing_timer] = best_numerator << 16 | best_denominator;
audio_dma_result result = audio_dma_setup_playback(
&self->dma,
sample,
loop,
false, // single channel
0, // audio channel
false, // output signed
BITS_PER_SAMPLE,
(uint32_t)tx_register, // output register: PWM cc register
0x3b + pacing_timer); // data request line
if (result == AUDIO_DMA_DMA_BUSY) {
common_hal_audiopwmio_pwmaudioout_stop(self);
mp_raise_RuntimeError(translate("No DMA channel found"));
}
if (result == AUDIO_DMA_MEMORY_ERROR) {
common_hal_audiopwmio_pwmaudioout_stop(self);
mp_raise_RuntimeError(translate("Unable to allocate buffers for signed conversion"));
}
// OK! We got all of the resources we need and dma is ready.
}
void common_hal_audiopwmio_pwmaudioout_stop(audiopwmio_pwmaudioout_obj_t *self) {

2
shared-module/audiocore/WaveFile.c
View File

@ -206,7 +206,7 @@ audioio_get_buffer_result_t audioio_wavefile_get_buffer(audioio_wavefile_obj_t *
}
if (need_more_data) {
uint16_t num_bytes_to_load = self->len;
uint32_t num_bytes_to_load = self->len;
if (num_bytes_to_load > self->bytes_remaining) {
num_bytes_to_load = self->bytes_remaining;
}