RP2040 PWMAudioOut: Release DMA channels after play has finished.

This commit is contained in:
Dan Halbert 2021-06-30 13:11:54 -04:00
parent 826e25989c
commit 33bbb8b1f4
19 changed files with 93 additions and 91 deletions

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@ -131,7 +131,7 @@ void audio_dma_load_next_block(audio_dma_t *dma) {
uint8_t *buffer;
uint32_t buffer_length;
audioio_get_buffer_result_t get_buffer_result =
audiosample_get_buffer(dma->sample, dma->single_channel, dma->audio_channel,
audiosample_get_buffer(dma->sample, dma->single_channel_output, dma->audio_channel,
&buffer, &buffer_length);
DmacDescriptor *descriptor = dma->second_descriptor;
@ -155,7 +155,7 @@ void audio_dma_load_next_block(audio_dma_t *dma) {
descriptor->SRCADDR.reg = ((uint32_t)output_buffer) + output_buffer_length;
if (get_buffer_result == GET_BUFFER_DONE) {
if (dma->loop) {
audiosample_reset_buffer(dma->sample, dma->single_channel, dma->audio_channel);
audiosample_reset_buffer(dma->sample, dma->single_channel_output, dma->audio_channel);
} else {
descriptor->DESCADDR.reg = 0;
}
@ -183,7 +183,7 @@ static void setup_audio_descriptor(DmacDescriptor *descriptor, uint8_t beat_size
audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
mp_obj_t sample,
bool loop,
bool single_channel,
bool single_channel_output,
uint8_t audio_channel,
bool output_signed,
uint32_t output_register_address,
@ -195,7 +195,7 @@ audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
dma->sample = sample;
dma->loop = loop;
dma->single_channel = single_channel;
dma->single_channel_output = single_channel_output;
dma->audio_channel = audio_channel;
dma->dma_channel = dma_channel;
dma->signed_to_unsigned = false;
@ -203,12 +203,12 @@ audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
dma->second_descriptor = NULL;
dma->spacing = 1;
dma->first_descriptor_free = true;
audiosample_reset_buffer(sample, single_channel, audio_channel);
audiosample_reset_buffer(sample, single_channel_output, audio_channel);
bool single_buffer;
bool samples_signed;
uint32_t max_buffer_length;
audiosample_get_buffer_structure(sample, single_channel, &single_buffer, &samples_signed,
audiosample_get_buffer_structure(sample, single_channel_output, &single_buffer, &samples_signed,
&max_buffer_length, &dma->spacing);
uint8_t output_spacing = dma->spacing;
if (output_signed != samples_signed) {
@ -254,12 +254,12 @@ audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
} else {
dma->beat_size = 1;
dma->bytes_per_sample = 1;
if (single_channel) {
if (single_channel_output) {
output_register_address += 1;
}
}
// Transfer both channels at once.
if (!single_channel && audiosample_channel_count(sample) == 2) {
if (!single_channel_output && audiosample_channel_count(sample) == 2) {
dma->beat_size *= 2;
}

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@ -42,7 +42,7 @@ typedef struct {
uint8_t beat_size;
uint8_t spacing;
bool loop;
bool single_channel;
bool single_channel_output;
bool signed_to_unsigned;
bool unsigned_to_signed;
bool first_buffer_free;
@ -72,16 +72,16 @@ void dma_free_channel(uint8_t channel);
// This sets everything up but doesn't start the timer.
// Sample is the python object for the sample to play.
// loop is true if we should loop the sample.
// single_channel is true if we only output a single channel. When false, all channels will be
// single_channel_output is true if we only output a single channel. When false, all channels will be
// output.
// audio_channel is the index of the channel to dma. single_channel must be false in this case.
// audio_channel is the index of the channel to dma. single_channel_output must be false in this case.
// output_signed is true if the dma'd data should be signed. False and it will be unsigned.
// output_register_address is the address to copy data to.
// dma_trigger_source is the DMA trigger source which cause another copy
audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
mp_obj_t sample,
bool loop,
bool single_channel,
bool single_channel_output,
uint8_t audio_channel,
bool output_signed,
uint32_t output_register_address,

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@ -129,7 +129,7 @@ void audio_dma_load_next_block(audio_dma_t *dma) {
uint8_t *buffer;
uint32_t buffer_length;
get_buffer_result = audiosample_get_buffer(dma->sample,
dma->single_channel, dma->audio_channel, &buffer, &buffer_length);
dma->single_channel_output, dma->audio_channel, &buffer, &buffer_length);
if (get_buffer_result == GET_BUFFER_ERROR) {
audio_dma_stop(dma);
@ -148,7 +148,7 @@ void audio_dma_load_next_block(audio_dma_t *dma) {
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);
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];
@ -161,13 +161,13 @@ void audio_dma_load_next_block(audio_dma_t *dma) {
audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
mp_obj_t sample,
bool loop,
bool single_channel,
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 because the DMA can't wrap to itself without the
// Use two DMA channels to play 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);
@ -180,7 +180,7 @@ audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
dma->sample = sample;
dma->loop = loop;
dma->single_channel = single_channel;
dma->single_channel_output = single_channel_output;
dma->audio_channel = audio_channel;
dma->signed_to_unsigned = false;
dma->unsigned_to_signed = false;
@ -189,12 +189,12 @@ audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
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);
audiosample_reset_buffer(sample, single_channel_output, audio_channel);
bool single_buffer;
bool samples_signed;
uint32_t max_buffer_length;
audiosample_get_buffer_structure(sample, single_channel, &single_buffer, &samples_signed,
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.
@ -227,10 +227,9 @@ audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
dma->output_size = 1;
}
// Transfer both channels at once.
if (!single_channel && audiosample_channel_count(sample) == 2) {
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;
@ -324,20 +323,19 @@ void audio_dma_stop(audio_dma_t *dma) {
// 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;
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.)
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[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;
}
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) {
@ -378,6 +376,8 @@ bool audio_dma_get_playing(audio_dma_t *dma) {
// 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) {

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@ -39,7 +39,7 @@ typedef struct {
uint8_t output_size;
uint8_t sample_spacing;
bool loop;
bool single_channel;
bool single_channel_output;
bool signed_to_unsigned;
bool unsigned_to_signed;
bool output_signed;
@ -66,16 +66,16 @@ void audio_dma_reset(void);
// This sets everything up but doesn't start the timer.
// Sample is the python object for the sample to play.
// loop is true if we should loop the sample.
// single_channel is true if we only output a single channel. When false, all channels will be
// single_channel_output is true if we only output a single channel. When false, all channels will be
// output.
// audio_channel is the index of the channel to dma. single_channel must be false in this case.
// audio_channel is the index of the channel to dma. single_channel_output must be false in this case.
// output_signed is true if the dma'd data should be signed. False and it will be unsigned.
// output_register_address is the address to copy data to.
// dma_trigger_source is the DMA trigger source which cause another copy
audio_dma_result audio_dma_setup_playback(audio_dma_t *dma,
mp_obj_t sample,
bool loop,
bool single_channel,
bool single_channel_output,
uint8_t audio_channel,
bool output_signed,
uint8_t output_resolution,

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@ -1,3 +1,4 @@
/*
* This file is part of the MicroPython project, http://micropython.org/
*
@ -36,10 +37,10 @@
// We don't bit pack because we'll only have two at most. Its better to save code size instead.
typedef struct {
mp_obj_base_t base;
bool left_justified;
rp2pio_statemachine_obj_t state_machine;
bool playing;
audio_dma_t dma;
bool left_justified;
bool playing;
} audiobusio_i2sout_obj_t;
void i2sout_reset(void);

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@ -96,11 +96,6 @@ void common_hal_audiopwmio_pwmaudioout_construct(audiopwmio_pwmaudioout_obj_t *s
mp_raise_RuntimeError(translate("All timers in use"));
}
claim_pin(left_channel);
if (right_channel != NULL) {
claim_pin(right_channel);
}
audio_dma_init(&self->dma);
self->pacing_timer = NUM_DMA_TIMERS;
@ -220,7 +215,10 @@ bool common_hal_audiopwmio_pwmaudioout_get_playing(audiopwmio_pwmaudioout_obj_t
if (!playing && self->pacing_timer < NUM_DMA_TIMERS) {
dma_hw->timer[self->pacing_timer] = 0;
self->pacing_timer = NUM_DMA_TIMERS;
audio_dma_stop(&self->dma);
}
return playing;
}

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@ -163,6 +163,8 @@ pwmout_result_t common_hal_pwmio_pwmout_construct(pwmio_pwmout_obj_t *self,
self->variable_frequency = variable_frequency;
self->duty_cycle = duty;
claim_pin(pin);
if (frequency == 0 || frequency > (common_hal_mcu_processor_get_frequency() / 2)) {
return PWMOUT_INVALID_FREQUENCY;
}

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@ -67,17 +67,17 @@ uint8_t common_hal_audioio_rawsample_get_channel_count(audioio_rawsample_obj_t *
}
void audioio_rawsample_reset_buffer(audioio_rawsample_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel) {
}
audioio_get_buffer_result_t audioio_rawsample_get_buffer(audioio_rawsample_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel,
uint8_t **buffer,
uint32_t *buffer_length) {
*buffer_length = self->len;
if (single_channel) {
if (single_channel_output) {
*buffer = self->buffer + (channel % self->channel_count) * (self->bits_per_sample / 8);
} else {
*buffer = self->buffer;
@ -85,13 +85,13 @@ audioio_get_buffer_result_t audioio_rawsample_get_buffer(audioio_rawsample_obj_t
return GET_BUFFER_DONE;
}
void audioio_rawsample_get_buffer_structure(audioio_rawsample_obj_t *self, bool single_channel,
void audioio_rawsample_get_buffer_structure(audioio_rawsample_obj_t *self, bool single_channel_output,
bool *single_buffer, bool *samples_signed,
uint32_t *max_buffer_length, uint8_t *spacing) {
*single_buffer = true;
*samples_signed = self->samples_signed;
*max_buffer_length = self->len;
if (single_channel) {
if (single_channel_output) {
*spacing = self->channel_count;
} else {
*spacing = 1;

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@ -44,14 +44,14 @@ typedef struct {
// These are not available from Python because it may be called in an interrupt.
void audioio_rawsample_reset_buffer(audioio_rawsample_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel);
audioio_get_buffer_result_t audioio_rawsample_get_buffer(audioio_rawsample_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel,
uint8_t **buffer,
uint32_t *buffer_length); // length in bytes
void audioio_rawsample_get_buffer_structure(audioio_rawsample_obj_t *self, bool single_channel,
void audioio_rawsample_get_buffer_structure(audioio_rawsample_obj_t *self, bool single_channel_output,
bool *single_buffer, bool *samples_signed,
uint32_t *max_buffer_length, uint8_t *spacing);

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@ -169,9 +169,9 @@ uint32_t audioio_wavefile_max_buffer_length(audioio_wavefile_obj_t *self) {
}
void audioio_wavefile_reset_buffer(audioio_wavefile_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel) {
if (single_channel && channel == 1) {
if (single_channel_output && channel == 1) {
return;
}
// We don't reset the buffer index in case we're looping and we have an odd number of buffer
@ -184,11 +184,11 @@ void audioio_wavefile_reset_buffer(audioio_wavefile_obj_t *self,
}
audioio_get_buffer_result_t audioio_wavefile_get_buffer(audioio_wavefile_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel,
uint8_t **buffer,
uint32_t *buffer_length) {
if (!single_channel) {
if (!single_channel_output) {
channel = 0;
}
@ -265,13 +265,14 @@ audioio_get_buffer_result_t audioio_wavefile_get_buffer(audioio_wavefile_obj_t *
return self->bytes_remaining == 0 ? GET_BUFFER_DONE : GET_BUFFER_MORE_DATA;
}
void audioio_wavefile_get_buffer_structure(audioio_wavefile_obj_t *self, bool single_channel,
void audioio_wavefile_get_buffer_structure(audioio_wavefile_obj_t *self, bool single_channel_output,
bool *single_buffer, bool *samples_signed,
uint32_t *max_buffer_length, uint8_t *spacing) {
*single_buffer = false;
// In WAV files, 8-bit samples are always unsigned, and larger samples are always signed.
*samples_signed = self->bits_per_sample > 8;
*max_buffer_length = 512;
if (single_channel) {
if (single_channel_output) {
*spacing = self->channel_count;
} else {
*spacing = 1;

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@ -57,14 +57,14 @@ typedef struct {
// These are not available from Python because it may be called in an interrupt.
void audioio_wavefile_reset_buffer(audioio_wavefile_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel);
audioio_get_buffer_result_t audioio_wavefile_get_buffer(audioio_wavefile_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel,
uint8_t **buffer,
uint32_t *buffer_length); // length in bytes
void audioio_wavefile_get_buffer_structure(audioio_wavefile_obj_t *self, bool single_channel,
void audioio_wavefile_get_buffer_structure(audioio_wavefile_obj_t *self, bool single_channel_output,
bool *single_buffer, bool *samples_signed,
uint32_t *max_buffer_length, uint8_t *spacing);

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@ -50,24 +50,24 @@ uint8_t audiosample_channel_count(mp_obj_t sample_obj) {
return proto->channel_count(MP_OBJ_TO_PTR(sample_obj));
}
void audiosample_reset_buffer(mp_obj_t sample_obj, bool single_channel, uint8_t audio_channel) {
void audiosample_reset_buffer(mp_obj_t sample_obj, bool single_channel_output, uint8_t audio_channel) {
const audiosample_p_t *proto = mp_proto_get_or_throw(MP_QSTR_protocol_audiosample, sample_obj);
proto->reset_buffer(MP_OBJ_TO_PTR(sample_obj), single_channel, audio_channel);
proto->reset_buffer(MP_OBJ_TO_PTR(sample_obj), single_channel_output, audio_channel);
}
audioio_get_buffer_result_t audiosample_get_buffer(mp_obj_t sample_obj,
bool single_channel,
bool single_channel_output,
uint8_t channel,
uint8_t **buffer, uint32_t *buffer_length) {
const audiosample_p_t *proto = mp_proto_get_or_throw(MP_QSTR_protocol_audiosample, sample_obj);
return proto->get_buffer(MP_OBJ_TO_PTR(sample_obj), single_channel, channel, buffer, buffer_length);
return proto->get_buffer(MP_OBJ_TO_PTR(sample_obj), single_channel_output, channel, buffer, buffer_length);
}
void audiosample_get_buffer_structure(mp_obj_t sample_obj, bool single_channel,
void audiosample_get_buffer_structure(mp_obj_t sample_obj, bool single_channel_output,
bool *single_buffer, bool *samples_signed,
uint32_t *max_buffer_length, uint8_t *spacing) {
const audiosample_p_t *proto = mp_proto_get_or_throw(MP_QSTR_protocol_audiosample, sample_obj);
proto->get_buffer_structure(MP_OBJ_TO_PTR(sample_obj), single_channel, single_buffer,
proto->get_buffer_structure(MP_OBJ_TO_PTR(sample_obj), single_channel_output, single_buffer,
samples_signed, max_buffer_length, spacing);
}

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@ -43,12 +43,12 @@ typedef uint32_t (*audiosample_sample_rate_fun)(mp_obj_t);
typedef uint8_t (*audiosample_bits_per_sample_fun)(mp_obj_t);
typedef uint8_t (*audiosample_channel_count_fun)(mp_obj_t);
typedef void (*audiosample_reset_buffer_fun)(mp_obj_t,
bool single_channel, uint8_t audio_channel);
bool single_channel_output, uint8_t audio_channel);
typedef audioio_get_buffer_result_t (*audiosample_get_buffer_fun)(mp_obj_t,
bool single_channel, uint8_t channel, uint8_t **buffer,
bool single_channel_output, uint8_t channel, uint8_t **buffer,
uint32_t *buffer_length);
typedef void (*audiosample_get_buffer_structure_fun)(mp_obj_t,
bool single_channel, bool *single_buffer,
bool single_channel_output, bool *single_buffer,
bool *samples_signed, uint32_t *max_buffer_length,
uint8_t *spacing);
@ -65,12 +65,12 @@ typedef struct _audiosample_p_t {
uint32_t audiosample_sample_rate(mp_obj_t sample_obj);
uint8_t audiosample_bits_per_sample(mp_obj_t sample_obj);
uint8_t audiosample_channel_count(mp_obj_t sample_obj);
void audiosample_reset_buffer(mp_obj_t sample_obj, bool single_channel, uint8_t audio_channel);
void audiosample_reset_buffer(mp_obj_t sample_obj, bool single_channel_output, uint8_t audio_channel);
audioio_get_buffer_result_t audiosample_get_buffer(mp_obj_t sample_obj,
bool single_channel,
bool single_channel_output,
uint8_t channel,
uint8_t **buffer, uint32_t *buffer_length);
void audiosample_get_buffer_structure(mp_obj_t sample_obj, bool single_channel,
void audiosample_get_buffer_structure(mp_obj_t sample_obj, bool single_channel_output,
bool *single_buffer, bool *samples_signed,
uint32_t *max_buffer_length, uint8_t *spacing);

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@ -94,7 +94,7 @@ bool common_hal_audiomixer_mixer_get_playing(audiomixer_mixer_obj_t *self) {
}
void audiomixer_mixer_reset_buffer(audiomixer_mixer_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel) {
for (uint8_t i = 0; i < self->voice_count; i++) {
common_hal_audiomixer_mixervoice_stop(self->voice[i]);
@ -280,11 +280,11 @@ static void mix_down_one_voice(audiomixer_mixer_obj_t *self,
}
audioio_get_buffer_result_t audiomixer_mixer_get_buffer(audiomixer_mixer_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel,
uint8_t **buffer,
uint32_t *buffer_length) {
if (!single_channel) {
if (!single_channel_output) {
channel = 0;
}
@ -351,13 +351,13 @@ audioio_get_buffer_result_t audiomixer_mixer_get_buffer(audiomixer_mixer_obj_t *
return GET_BUFFER_MORE_DATA;
}
void audiomixer_mixer_get_buffer_structure(audiomixer_mixer_obj_t *self, bool single_channel,
void audiomixer_mixer_get_buffer_structure(audiomixer_mixer_obj_t *self, bool single_channel_output,
bool *single_buffer, bool *samples_signed,
uint32_t *max_buffer_length, uint8_t *spacing) {
*single_buffer = false;
*samples_signed = self->samples_signed;
*max_buffer_length = self->len;
if (single_channel) {
if (single_channel_output) {
*spacing = self->channel_count;
} else {
*spacing = 1;

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@ -55,14 +55,14 @@ typedef struct {
// These are not available from Python because it may be called in an interrupt.
void audiomixer_mixer_reset_buffer(audiomixer_mixer_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel);
audioio_get_buffer_result_t audiomixer_mixer_get_buffer(audiomixer_mixer_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel,
uint8_t **buffer,
uint32_t *buffer_length); // length in bytes
void audiomixer_mixer_get_buffer_structure(audiomixer_mixer_obj_t *self, bool single_channel,
void audiomixer_mixer_get_buffer_structure(audiomixer_mixer_obj_t *self, bool single_channel_output,
bool *single_buffer, bool *samples_signed,
uint32_t *max_buffer_length, uint8_t *spacing);

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@ -295,9 +295,9 @@ bool audiomp3_mp3file_samples_signed(audiomp3_mp3file_obj_t *self) {
}
void audiomp3_mp3file_reset_buffer(audiomp3_mp3file_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel) {
if (single_channel && channel == 1) {
if (single_channel_output && channel == 1) {
return;
}
// We don't reset the buffer index in case we're looping and we have an odd number of buffer
@ -314,14 +314,14 @@ void audiomp3_mp3file_reset_buffer(audiomp3_mp3file_obj_t *self,
}
audioio_get_buffer_result_t audiomp3_mp3file_get_buffer(audiomp3_mp3file_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel,
uint8_t **bufptr,
uint32_t *buffer_length) {
if (!self->inbuf) {
return GET_BUFFER_ERROR;
}
if (!single_channel) {
if (!single_channel_output) {
channel = 0;
}
@ -363,13 +363,13 @@ audioio_get_buffer_result_t audiomp3_mp3file_get_buffer(audiomp3_mp3file_obj_t *
return GET_BUFFER_MORE_DATA;
}
void audiomp3_mp3file_get_buffer_structure(audiomp3_mp3file_obj_t *self, bool single_channel,
void audiomp3_mp3file_get_buffer_structure(audiomp3_mp3file_obj_t *self, bool single_channel_output,
bool *single_buffer, bool *samples_signed,
uint32_t *max_buffer_length, uint8_t *spacing) {
*single_buffer = false;
*samples_signed = true;
*max_buffer_length = self->frame_buffer_size;
if (single_channel) {
if (single_channel_output) {
*spacing = self->channel_count;
} else {
*spacing = 1;

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@ -58,14 +58,14 @@ typedef struct {
// These are not available from Python because it may be called in an interrupt.
void audiomp3_mp3file_reset_buffer(audiomp3_mp3file_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel);
audioio_get_buffer_result_t audiomp3_mp3file_get_buffer(audiomp3_mp3file_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel,
uint8_t **buffer,
uint32_t *buffer_length); // length in bytes
void audiomp3_mp3file_get_buffer_structure(audiomp3_mp3file_obj_t *self, bool single_channel,
void audiomp3_mp3file_get_buffer_structure(audiomp3_mp3file_obj_t *self, bool single_channel_output,
bool *single_buffer, bool *samples_signed,
uint32_t *max_buffer_length, uint8_t *spacing);

View File

@ -165,7 +165,7 @@ uint8_t common_hal_synthio_miditrack_get_channel_count(synthio_miditrack_obj_t *
}
void synthio_miditrack_reset_buffer(synthio_miditrack_obj_t *self,
bool single_channel, uint8_t channel) {
bool single_channel_output, uint8_t channel) {
self->next_span = 0;
}
@ -174,7 +174,7 @@ STATIC const uint16_t notes[] = {8372, 8870, 9397, 9956, 10548, 11175, 11840,
12544, 13290, 14080, 14917, 15804}; // 9th octave
audioio_get_buffer_result_t synthio_miditrack_get_buffer(synthio_miditrack_obj_t *self,
bool single_channel, uint8_t channel, uint8_t **buffer, uint32_t *buffer_length) {
bool single_channel_output, uint8_t channel, uint8_t **buffer, uint32_t *buffer_length) {
if (self->next_span >= self->total_spans) {
*buffer_length = 0;
@ -202,7 +202,7 @@ audioio_get_buffer_result_t synthio_miditrack_get_buffer(synthio_miditrack_obj_t
GET_BUFFER_DONE : GET_BUFFER_MORE_DATA;
}
void synthio_miditrack_get_buffer_structure(synthio_miditrack_obj_t *self, bool single_channel,
void synthio_miditrack_get_buffer_structure(synthio_miditrack_obj_t *self, bool single_channel_output,
bool *single_buffer, bool *samples_signed, uint32_t *max_buffer_length, uint8_t *spacing) {
*single_buffer = true;

View File

@ -49,16 +49,16 @@ typedef struct {
// These are not available from Python because it may be called in an interrupt.
void synthio_miditrack_reset_buffer(synthio_miditrack_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel);
audioio_get_buffer_result_t synthio_miditrack_get_buffer(synthio_miditrack_obj_t *self,
bool single_channel,
bool single_channel_output,
uint8_t channel,
uint8_t **buffer,
uint32_t *buffer_length); // length in bytes
void synthio_miditrack_get_buffer_structure(synthio_miditrack_obj_t *self, bool single_channel,
void synthio_miditrack_get_buffer_structure(synthio_miditrack_obj_t *self, bool single_channel_output,
bool *single_buffer, bool *samples_signed,
uint32_t *max_buffer_length, uint8_t *spacing);