Add audio output support!

This evolves the API from 2.x (and breaks it). Playback devices are now
separate from the samples themselves. This allows for greater playback
flexibility. Two sample sources are audioio.RawSample and audioio.WaveFile.
They can both be mono or stereo. They can be output to audioio.AudioOut or
audiobusio.I2SOut.

Internally, the dma tracking has changed from a TC counting block transfers
to an interrupt generated by the block event sent to the EVSYS. This reduces
the overhead of each DMA transfer so multiple can occure without using up TCs.

Fixes #652. Fixes #522. Huge progress on #263
This commit is contained in:
Scott Shawcroft 2018-03-12 16:09:13 -07:00
parent e311d17905
commit 28642ab10d
36 changed files with 2942 additions and 720 deletions

View File

@ -220,7 +220,10 @@ endif
SRC_ASF := $(addprefix asf4/$(CHIP_FAMILY)/, $(SRC_ASF))
SRC_C = \
audio_dma.c \
background.c \
clocks.c \
events.c \
fatfs_port.c \
flash_api.c \
mphalport.c \
@ -276,7 +279,7 @@ SRC_COMMON_HAL = \
neopixel_write/__init__.c \
os/__init__.c \
storage/__init__.c \
supervisor/__init__.c \
supervisor/__init__.c \
supervisor/Runtime.c \
time/__init__.c \
analogio/__init__.c \
@ -287,12 +290,11 @@ SRC_COMMON_HAL = \
pulseio/PulseOut.c \
pulseio/PWMOut.c \
usb_hid/__init__.c \
usb_hid/Device.c
# audiobusio/__init__.c \
audiobusio/PDMIn.c \
usb_hid/Device.c \
audioio/__init__.c \
audioio/AudioOut.c \
nvm/__init__.c \
# nvm/__init__.c \
audiobusio/PDMIn.c \
nvm/ByteArray.c \
touchio/__init__.c \
touchio/TouchIn.c \
@ -338,6 +340,8 @@ SRC_COMMON_HAL_EXPANDED = $(addprefix shared-bindings/, $(SRC_COMMON_HAL)) \
$(addprefix common-hal/, $(SRC_COMMON_HAL))
SRC_SHARED_MODULE = \
audioio/RawSample.c \
audioio/WaveFile.c \
bitbangio/__init__.c \
bitbangio/I2C.c \
bitbangio/OneWire.c \
@ -355,6 +359,22 @@ SRC_SHARED_MODULE = \
uheap/__init__.c \
ustack/__init__.c
ifeq ($(CHIP_FAMILY),samd21)
SRC_COMMON_HAL = $SRC_COMMON_HAL \
audiobusio/__init__.c \
audiobusio/I2SOut.c
endif
ifeq ($(CHIP_VARIANT),SAMD51G18)
SRC_COMMON_HAL = $SRC_COMMON_HAL \
audiobusio/__init__.c \
audiobusio/I2SOut.c
endif
ifeq ($(CHIP_VARIANT),SAMD51G19)
SRC_COMMON_HAL = $SRC_COMMON_HAL \
audiobusio/__init__.c \
audiobusio/I2SOut.c
endif
SRC_SHARED_MODULE_EXPANDED = $(addprefix shared-bindings/, $(SRC_SHARED_MODULE)) \
$(addprefix shared-module/, $(SRC_SHARED_MODULE))

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@ -0,0 +1,363 @@
/*
* This file is part of the MicroPython project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2018 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 "clocks.h"
#include "events.h"
#include "shared_dma.h"
#include "shared-bindings/audioio/RawSample.h"
#include "shared-bindings/audioio/WaveFile.h"
#include "py/mpstate.h"
uint32_t audiosample_sample_rate(mp_obj_t sample_obj) {
if (MP_OBJ_IS_TYPE(sample_obj, &audioio_rawsample_type)) {
audioio_rawsample_obj_t* sample = MP_OBJ_TO_PTR(sample_obj);
return sample->sample_rate;
}
if (MP_OBJ_IS_TYPE(sample_obj, &audioio_wavefile_type)) {
audioio_wavefile_obj_t* file = MP_OBJ_TO_PTR(sample_obj);
return file->sample_rate;
}
return 16000;
}
uint8_t audiosample_bits_per_sample(mp_obj_t sample_obj) {
if (MP_OBJ_IS_TYPE(sample_obj, &audioio_rawsample_type)) {
audioio_rawsample_obj_t* sample = MP_OBJ_TO_PTR(sample_obj);
return sample->bits_per_sample;
}
if (MP_OBJ_IS_TYPE(sample_obj, &audioio_wavefile_type)) {
audioio_wavefile_obj_t* file = MP_OBJ_TO_PTR(sample_obj);
return file->bits_per_sample;
}
return 8;
}
uint8_t audiosample_channel_count(mp_obj_t sample_obj) {
if (MP_OBJ_IS_TYPE(sample_obj, &audioio_rawsample_type)) {
audioio_rawsample_obj_t* sample = MP_OBJ_TO_PTR(sample_obj);
return sample->channel_count;
}
if (MP_OBJ_IS_TYPE(sample_obj, &audioio_wavefile_type)) {
audioio_wavefile_obj_t* file = MP_OBJ_TO_PTR(sample_obj);
return file->channel_count;
}
return 1;
}
void audiosample_reset_buffer(mp_obj_t sample_obj, bool single_channel, uint8_t audio_channel) {
if (MP_OBJ_IS_TYPE(sample_obj, &audioio_rawsample_type)) {
audioio_rawsample_obj_t* sample = MP_OBJ_TO_PTR(sample_obj);
audioio_rawsample_reset_buffer(sample, single_channel, audio_channel);
}
if (MP_OBJ_IS_TYPE(sample_obj, &audioio_wavefile_type)) {
audioio_wavefile_obj_t* file = MP_OBJ_TO_PTR(sample_obj);
audioio_wavefile_reset_buffer(file, single_channel, audio_channel);
}
}
bool audiosample_get_buffer(mp_obj_t sample_obj, bool single_channel, uint8_t channel, uint8_t** buffer, uint32_t* buffer_length) {
if (MP_OBJ_IS_TYPE(sample_obj, &audioio_rawsample_type)) {
audioio_rawsample_obj_t* sample = MP_OBJ_TO_PTR(sample_obj);
return audioio_rawsample_get_buffer(sample, single_channel, channel, buffer, buffer_length);
}
if (MP_OBJ_IS_TYPE(sample_obj, &audioio_wavefile_type)) {
audioio_wavefile_obj_t* file = MP_OBJ_TO_PTR(sample_obj);
return audioio_wavefile_get_buffer(file, single_channel, channel, buffer, buffer_length);
}
return true;
}
static void audiosample_get_buffer_structure(mp_obj_t sample_obj, bool single_channel,
bool* single_buffer, bool* samples_signed,
uint32_t* max_buffer_length, uint8_t* spacing) {
if (MP_OBJ_IS_TYPE(sample_obj, &audioio_rawsample_type)) {
audioio_rawsample_obj_t* sample = MP_OBJ_TO_PTR(sample_obj);
audioio_rawsample_get_buffer_structure(sample, single_channel, single_buffer,
samples_signed, max_buffer_length, spacing);
} else if (MP_OBJ_IS_TYPE(sample_obj, &audioio_wavefile_type)) {
audioio_wavefile_obj_t* file = MP_OBJ_TO_PTR(sample_obj);
audioio_wavefile_get_buffer_structure(file, single_channel, single_buffer, samples_signed,
max_buffer_length, spacing);
}
}
uint8_t find_free_audio_dma_channel(void) {
uint8_t channel;
for (channel = 0; channel < AUDIO_DMA_CHANNEL_COUNT; channel++) {
if (!dma_channel_enabled(channel)) {
return channel;
}
}
return channel;
}
audio_dma_t* audio_dma_state[AUDIO_DMA_CHANNEL_COUNT];
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,
uint8_t* output_spacing) {
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) {
*output_buffer_length = buffer_length / dma->spacing;
*output_spacing = 1;
uint32_t out_i = 0;
if (dma->bytes_per_sample == 1) {
for (uint32_t i = 0; i < buffer_length; i += dma->spacing) {
if (dma->signed_to_unsigned) {
((uint8_t*) *output_buffer)[out_i] = ((int8_t*) buffer)[i] + 0x80;
} else {
((int8_t*) *output_buffer)[out_i] = ((uint8_t*) buffer)[i] - 0x80;
}
out_i += 1;
}
} else if (dma->bytes_per_sample == 2) {
for (uint32_t i = 0; i < buffer_length / 2; i += dma->spacing) {
if (dma->signed_to_unsigned) {
((uint16_t*) *output_buffer)[out_i] = ((int16_t*) buffer)[i] + 0x8000;
} else {
((int16_t*) *output_buffer)[out_i] = ((uint16_t*) buffer)[i] - 0x8000;
}
out_i += 1;
}
}
} else {
*output_buffer = buffer;
*output_buffer_length = buffer_length;
*output_spacing = dma->spacing;
}
#pragma GCC diagnostic pop
dma->first_buffer_free = !dma->first_buffer_free;
}
void audio_dma_load_next_block(audio_dma_t* dma) {
uint8_t* buffer;
uint32_t buffer_length;
bool last_buffer = audiosample_get_buffer(dma->sample, dma->single_channel, dma->audio_channel,
&buffer, &buffer_length);
DmacDescriptor* descriptor = dma->second_descriptor;
if (dma->first_descriptor_free) {
descriptor = dma_descriptor(dma->dma_channel);
}
dma->first_descriptor_free = !dma->first_descriptor_free;
uint8_t* output_buffer;
uint32_t output_buffer_length;
uint8_t output_spacing;
audio_dma_convert_signed(dma, buffer, buffer_length, &output_buffer, &output_buffer_length,
&output_spacing);
descriptor->BTCNT.reg = output_buffer_length / dma->beat_size / output_spacing;
descriptor->SRCADDR.reg = ((uint32_t) output_buffer) + output_buffer_length;
if (last_buffer) {
if (dma->loop) {
audiosample_reset_buffer(dma->sample, dma->single_channel, dma->audio_channel);
} else {
descriptor->DESCADDR.reg = 0;
}
}
descriptor->BTCTRL.bit.VALID = true;
}
static void setup_audio_descriptor(DmacDescriptor* descriptor, uint8_t beat_size,
uint8_t spacing, uint32_t output_register_address) {
uint32_t beat_size_reg = DMAC_BTCTRL_BEATSIZE_BYTE;
if (beat_size == 2) {
beat_size_reg = DMAC_BTCTRL_BEATSIZE_HWORD;
} else if (beat_size == 4) {
beat_size_reg = DMAC_BTCTRL_BEATSIZE_WORD;
}
descriptor->BTCTRL.reg = beat_size_reg |
DMAC_BTCTRL_SRCINC |
DMAC_BTCTRL_EVOSEL_BLOCK |
DMAC_BTCTRL_STEPSIZE(spacing - 1) |
DMAC_BTCTRL_STEPSEL_SRC;
descriptor->DSTADDR.reg = output_register_address;
}
// 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,
uint32_t output_register_address,
uint8_t dma_trigger_source) {
uint8_t dma_channel = find_free_audio_dma_channel();
if (dma_channel >= AUDIO_DMA_CHANNEL_COUNT) {
return AUDIO_DMA_DMA_BUSY;
}
dma->sample = sample;
dma->loop = loop;
dma->single_channel = single_channel;
dma->audio_channel = audio_channel;
dma->dma_channel = dma_channel;
dma->signed_to_unsigned = false;
dma->unsigned_to_signed = false;
dma->second_descriptor = NULL;
dma->spacing = 1;
dma->first_descriptor_free = true;
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->spacing);
uint8_t output_spacing = dma->spacing;
if (output_signed != samples_signed) {
output_spacing = 1;
max_buffer_length /= dma->spacing;
dma->first_buffer = (uint8_t*) m_malloc(max_buffer_length, false);
if (dma->first_buffer == NULL) {
return AUDIO_DMA_MEMORY_ERROR;
}
dma->first_buffer_free = true;
if (!single_buffer) {
dma->second_buffer = (uint8_t*) m_malloc(max_buffer_length, false);
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;
}
dma->event_channel = 0xff;
if (!single_buffer) {
dma->second_descriptor = (DmacDescriptor*) m_malloc(sizeof(DmacDescriptor), false);
if (dma->second_descriptor == NULL) {
return AUDIO_DMA_MEMORY_ERROR;
}
// We're likely double buffering so set up the block interrupts.
turn_on_event_system();
dma->event_channel = find_sync_event_channel();
init_event_channel_interrupt(dma->event_channel, CORE_GCLK, EVSYS_ID_GEN_DMAC_CH_0 + dma_channel);
find_sync_event_channel();
// We keep the audio_dma_t for internal use and the sample as a root pointer because it
// contains the audiodma structure.
audio_dma_state[dma->dma_channel] = dma;
MP_STATE_PORT(playing_audio)[dma->dma_channel] = dma->sample;
}
dma->beat_size = 1;
dma->bytes_per_sample = 1;
if (audiosample_bits_per_sample(sample) == 16) {
dma->beat_size = 2;
dma->bytes_per_sample = 2;
}
// Transfer both channels at once.
if (!single_channel && audiosample_channel_count(sample) == 2) {
dma->beat_size *= 2;
}
DmacDescriptor* first_descriptor = dma_descriptor(dma_channel);
setup_audio_descriptor(first_descriptor, dma->beat_size, output_spacing, output_register_address);
if (single_buffer) {
first_descriptor->DESCADDR.reg = 0;
if (dma->loop) {
first_descriptor->DESCADDR.reg = (uint32_t) first_descriptor;
}
} else {
first_descriptor->DESCADDR.reg = (uint32_t) dma->second_descriptor;
setup_audio_descriptor(dma->second_descriptor, dma->beat_size, output_spacing, output_register_address);
dma->second_descriptor->DESCADDR.reg = (uint32_t) first_descriptor;
}
// Load the first two blocks up front.
audio_dma_load_next_block(dma);
if (!single_buffer) {
audio_dma_load_next_block(dma);
}
dma_configure(dma_channel, dma_trigger_source, true);
dma_enable_channel(dma_channel);
return AUDIO_DMA_OK;
}
void audio_dma_stop(audio_dma_t* dma) {
dma_disable_channel(dma->dma_channel);
disable_event_channel(dma->event_channel);
MP_STATE_PORT(playing_audio)[dma->dma_channel] = NULL;
dma->dma_channel = AUDIO_DMA_CHANNEL_COUNT;
}
void audio_dma_init(audio_dma_t* dma) {
dma->dma_channel = AUDIO_DMA_CHANNEL_COUNT;
}
void audio_dma_reset(void) {
for (uint8_t i = 0; i < AUDIO_DMA_CHANNEL_COUNT; i++) {
audio_dma_state[i] = NULL;
dma_disable_channel(i);
dma_descriptor(i)->BTCTRL.bit.VALID = false;
MP_STATE_PORT(playing_audio)[i] = NULL;
}
}
bool audio_dma_get_playing(audio_dma_t* dma) {
if (dma->dma_channel >= AUDIO_DMA_CHANNEL_COUNT) {
return false;
}
uint32_t status = dma_transfer_status(dma->dma_channel);
if ((status & DMAC_CHINTFLAG_TCMPL) != 0) {
audio_dma_stop(dma);
}
return status == 0;
}
// WARN(tannewt): DO NOT print from here. Printing calls background tasks such as this and causes a
// stack overflow.
void audio_dma_background(void) {
for (uint8_t i = 0; i < AUDIO_DMA_CHANNEL_COUNT; i++) {
audio_dma_t* dma = audio_dma_state[i];
if (dma == NULL) {
continue;
}
bool block_done = event_interrupt_active(dma->event_channel);
if (!block_done) {
continue;
}
audio_dma_load_next_block(dma);
}
}

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@ -0,0 +1,89 @@
/*
* This file is part of the MicroPython project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2018 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.
*/
#ifndef MICROPY_INCLUDED_ATMEL_SAMD_AUDIO_DMA_H
#define MICROPY_INCLUDED_ATMEL_SAMD_AUDIO_DMA_H
#include "extmod/vfs_fat_file.h"
#include "py/obj.h"
#include "shared-module/audioio/RawSample.h"
#include "shared-module/audioio/WaveFile.h"
typedef struct {
mp_obj_t sample;
uint8_t dma_channel;
uint8_t event_channel;
uint8_t audio_channel;
uint8_t bytes_per_sample;
uint8_t beat_size;
uint8_t spacing;
bool loop;
bool single_channel;
bool signed_to_unsigned;
bool unsigned_to_signed;
bool first_buffer_free;
uint8_t* first_buffer;
uint8_t* second_buffer;
bool first_descriptor_free;
DmacDescriptor* second_descriptor;
} audio_dma_t;
typedef enum {
AUDIO_DMA_OK,
AUDIO_DMA_DMA_BUSY,
AUDIO_DMA_MEMORY_ERROR,
} audio_dma_result;
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 audio_dma_init(audio_dma_t* dma);
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
// output.
// audio_channel is the index of the channel to dma. single_channel 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,
uint8_t audio_channel,
bool output_signed,
uint32_t output_register_address,
uint8_t dma_trigger_source);
void audio_dma_stop(audio_dma_t* dma);
bool audio_dma_get_playing(audio_dma_t* dma);
void audio_dma_background(void);
#endif // MICROPY_INCLUDED_ATMEL_SAMD_AUDIO_DMA_H

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@ -25,12 +25,12 @@
*/
#include "background.h"
// #include "common-hal/audioio/AudioOut.h"
#include "audio_dma.h"
#include "usb.h"
#include "usb_mass_storage.h"
void run_background_tasks(void) {
// audioout_background();
audio_dma_background();
usb_msc_background();
usb_cdc_background();
}

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@ -6,5 +6,5 @@ USB_MANUFACTURER = "Adafruit Industries LLC"
QSPI_FLASH_FILESYSTEM = 1
CHIP_VARIANT = SAMD51J19A
CHIP_VARIANT = SAMD51G19A
CHIP_FAMILY = samd51

165
ports/atmel-samd/clocks.c Normal file
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@ -0,0 +1,165 @@
/*
* This file is part of the MicroPython project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2018 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 "clocks.h"
#include "hpl_gclk_config.h"
#include "shared-bindings/microcontroller/__init__.h"
#include "py/runtime.h"
// TODO(tannewt): Should we have a way of sharing GCLKs based on their speed? Divisor doesn't
// gaurantee speed because it depends on the source.
uint8_t find_free_gclk(uint16_t divisor) {
if (divisor > (1 << 8)) {
if (gclk_enabled(1)) {
return 0xff;
}
return 1;
}
uint8_t first_8_bit = 2;
#ifdef SAMD21
first_8_bit = 3;
if (divisor <= (1 << 5) && !gclk_enabled(2)) {
return 2;
}
#endif
for (uint8_t i = first_8_bit; i < GCLK_GEN_NUM; i++) {
if (!gclk_enabled(i)) {
return i;
}
}
return 0xff;
}
bool gclk_enabled(uint8_t gclk) {
#ifdef SAMD51
return GCLK->GENCTRL[gclk].bit.GENEN;
#endif
#ifdef SAMD21
common_hal_mcu_disable_interrupts();
// Explicitly do a byte write so the peripheral knows we're just wanting to read the channel
// rather than write to it.
*((uint8_t*) &GCLK->GENCTRL.reg) = gclk;
while (GCLK->STATUS.bit.SYNCBUSY == 1) {}
bool enabled = GCLK->GENCTRL.bit.GENEN;
common_hal_mcu_enable_interrupts();
return enabled;
#endif
}
void disable_gclk(uint8_t gclk) {
#ifdef SAMD51
GCLK->GENCTRL[gclk].bit.GENEN = false;
while ((GCLK->SYNCBUSY.vec.GENCTRL & (1 << gclk)) != 0) {}
#endif
#ifdef SAMD21
GCLK->GENCTRL.reg = GCLK_GENCTRL_ID(gclk);
while (GCLK->STATUS.bit.SYNCBUSY == 1) {}
#endif
}
void connect_gclk_to_peripheral(uint8_t gclk, uint8_t peripheral) {
#ifdef SAMD21
GCLK->CLKCTRL.reg = GCLK_CLKCTRL_ID(peripheral) | GCLK_CLKCTRL_GEN(gclk) | GCLK_CLKCTRL_CLKEN;
#endif
#ifdef SAMD51
GCLK->PCHCTRL[peripheral].reg = GCLK_PCHCTRL_CHEN | GCLK_PCHCTRL_GEN(gclk);
while(GCLK->SYNCBUSY.reg != 0) {}
#endif
}
void disconnect_gclk_from_peripheral(uint8_t gclk, uint8_t peripheral) {
#ifdef SAMD21
GCLK->CLKCTRL.reg = GCLK_CLKCTRL_ID(peripheral) | GCLK_CLKCTRL_GEN(gclk);
#endif
#ifdef SAMD51
GCLK->PCHCTRL[peripheral].reg = 0;
#endif
}
void enable_clock_generator(uint8_t gclk, uint8_t source, uint16_t divisor) {
#ifdef SAMD21
GCLK->GENDIV.reg = GCLK_GENDIV_ID(gclk) | GCLK_GENDIV_DIV(divisor);
GCLK->GENCTRL.reg = GCLK_GENCTRL_ID(gclk) | GCLK_GENCTRL_SRC(source) | GCLK_GENCTRL_GENEN;
while (GCLK->STATUS.bit.SYNCBUSY != 0) {}
#endif
#ifdef SAMD51
GCLK->GENCTRL[gclk].reg = GCLK_GENCTRL_SRC(source) | GCLK_GENCTRL_DIV(divisor) | GCLK_GENCTRL_GENEN;
while ((GCLK->SYNCBUSY.vec.GENCTRL & (1 << gclk)) != 0) {}
#endif
}
void disable_clock_generator(uint8_t gclk) {
#ifdef SAMD21
GCLK->GENCTRL.reg = GCLK_GENCTRL_ID(gclk);
while (GCLK->STATUS.bit.SYNCBUSY != 0) {}
#endif
#ifdef SAMD51
GCLK->GENCTRL[gclk].reg = 0;
while ((GCLK->SYNCBUSY.vec.GENCTRL & (1 << gclk)) != 0) {}
#endif
}
void reset_gclks(void) {
// Never reset GCLK0 because its used for the core
#if CONF_GCLK_GEN_1_GENEN == 0
disable_gclk(1);
#endif
#if CONF_GCLK_GEN_2_GENEN == 0
disable_gclk(2);
#endif
#if CONF_GCLK_GEN_3_GENEN == 0
disable_gclk(3);
#endif
#if CONF_GCLK_GEN_4_GENEN == 0
disable_gclk(4);
#endif
#if CONF_GCLK_GEN_5_GENEN == 0
disable_gclk(5);
#endif
#if CONF_GCLK_GEN_6_GENEN == 0
disable_gclk(6);
#endif
#if CONF_GCLK_GEN_7_GENEN == 0
disable_gclk(7);
#endif
#ifdef SAMD51
#if CONF_GCLK_GEN_8_GENEN == 0
disable_gclk(8);
#endif
#if CONF_GCLK_GEN_9_GENEN == 0
disable_gclk(9);
#endif
#if CONF_GCLK_GEN_10_GENEN == 0
disable_gclk(10);
#endif
#if CONF_GCLK_GEN_11_GENEN == 0
disable_gclk(11);
#endif
#endif
}

55
ports/atmel-samd/clocks.h Normal file
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@ -0,0 +1,55 @@
/*
* This file is part of the MicroPython project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2018 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.
*/
#ifndef MICROPY_INCLUDED_ATMEL_SAMD_CLOCKS_H
#define MICROPY_INCLUDED_ATMEL_SAMD_CLOCKS_H
#include <stdbool.h>
#include <stdint.h>
#include "include/sam.h"
#ifdef SAMD51
#define CLOCK_48MHZ GCLK_GENCTRL_SRC_DFLL_Val
#endif
#ifdef SAMD21
#define CLOCK_48MHZ GCLK_GENCTRL_SRC_DFLL48M_Val
#endif
#define CORE_GCLK 0
uint8_t find_free_gclk(uint16_t divisor);
bool gclk_enabled(uint8_t gclk);
void reset_gclks(void);
void connect_gclk_to_peripheral(uint8_t gclk, uint8_t peripheral);
void disconnect_gclk_from_peripheral(uint8_t gclk, uint8_t peripheral);
void enable_clock_generator(uint8_t gclk, uint8_t source, uint16_t divisor);
void disable_clock_generator(uint8_t gclk);
#endif // MICROPY_INCLUDED_ATMEL_SAMD_CLOCKS_H

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@ -0,0 +1,376 @@
/*
* This file is part of the MicroPython project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2017 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 <stdint.h>
#include <string.h>
#include "extmod/vfs_fat_file.h"
#include "py/gc.h"
#include "py/mperrno.h"
#include "py/runtime.h"
#include "common-hal/audiobusio/I2SOut.h"
#include "shared-bindings/audiobusio/I2SOut.h"
#include "shared-bindings/audioio/RawSample.h"
#include "shared-bindings/microcontroller/Pin.h"
#include "atmel_start_pins.h"
#include "hal/include/hal_gpio.h"
#include "hpl/gclk/hpl_gclk_base.h"
#include "peripheral_clk_config.h"
#ifdef SAMD21
#include "hpl/pm/hpl_pm_base.h"
#endif
#include "audio_dma.h"
#include "clocks.h"
#include "events.h"
#include "samd21_pins.h"
#include "shared_dma.h"
#include "timers.h"
#ifdef SAMD21
#define SERCTRL(name) I2S_SERCTRL_ ## name
#endif
#ifdef SAMD51
#define SERCTRL(name) I2S_TXCTRL_ ## name
#endif
void i2sout_reset(void) {
// Make sure the I2S peripheral is running so we can see if the resources we need are free.
#ifdef SAMD51
// Connect the clock units to the 2mhz clock. It can't disable without it.
connect_gclk_to_peripheral(5, I2S_GCLK_ID_0);
connect_gclk_to_peripheral(5, I2S_GCLK_ID_1);
#endif
if (I2S->CTRLA.bit.ENABLE == 1) {
I2S->CTRLA.bit.ENABLE = 0;
while (I2S->SYNCBUSY.bit.ENABLE == 1) {}
}
// Make sure the I2S peripheral is running so we can see if the resources we need are free.
#ifdef SAMD51
// Connect the clock units to the 2mhz clock by default. They can't reset without it.
disconnect_gclk_from_peripheral(5, I2S_GCLK_ID_0);
disconnect_gclk_from_peripheral(5, I2S_GCLK_ID_1);
hri_mclk_clear_APBDMASK_I2S_bit(MCLK);
#endif
#ifdef SAMD21
_pm_disable_bus_clock(PM_BUS_APBC, I2S);
#endif
}
void common_hal_audiobusio_i2sout_construct(audiobusio_i2sout_obj_t* self,
const mcu_pin_obj_t* bit_clock, const mcu_pin_obj_t* word_select,
const mcu_pin_obj_t* data, bool left_justified) {
uint8_t serializer = 0xff;
uint8_t bc_clock_unit = 0xff;
uint8_t ws_clock_unit = 0xff;
#ifdef SAMD21
if (bit_clock == &pin_PA10
#ifdef PIN_PA20
|| bit_clock == &pin_PA20
#endif
) { // I2S SCK[0]
bc_clock_unit = 0;
}
#ifdef PIN_PB11
else if (bit_clock == &pin_PB11) { // I2S SCK[1]
bc_clock_unit = 1;
}
#endif
if (word_select == &pin_PA11
#ifdef PIN_PA21
|| word_select == &pin_PA21
#endif
) { // I2S FS[0]
ws_clock_unit = 0;
}
#ifdef PIN_PB12
else if (word_select == &pin_PB12) { // I2S FS[1]
ws_clock_unit = 1;
}
#endif
if (data == &pin_PA07 || data == &pin_PA19) { // I2S SD[0]
serializer = 0;
} else if (data == &pin_PA08
#ifdef PIN_PB16
|| data == &pin_PB16
#endif
) { // I2S SD[1]
serializer = 1;
}
#endif
#ifdef SAMD51
// Only clock unit 0 can be used for transmission.
if (bit_clock == &pin_PA10 || bit_clock == &pin_PB16) { // I2S SCK[0]
bc_clock_unit = 0;
}
if (word_select == &pin_PA09 || word_select == &pin_PA20) { // I2S FS[0]
ws_clock_unit = 0;
}
if (data == &pin_PA11 || data == &pin_PA21) { // I2S SDO
serializer = 0;
}
#endif
if (bc_clock_unit == 0xff) {
mp_raise_ValueError("Invalid bit clock pin");
}
if (ws_clock_unit == 0xff) {
mp_raise_ValueError("Invalid bit clock pin");
}
if (bc_clock_unit != ws_clock_unit) {
mp_raise_ValueError("Bit clock and word select must share a clock unit");
}
if (serializer == 0xff) {
mp_raise_ValueError("Invalid data pin");
}
self->clock_unit = ws_clock_unit;
self->serializer = serializer;
// Make sure the I2S peripheral is running so we can see if the resources we need are free.
#ifdef SAMD51
hri_mclk_set_APBDMASK_I2S_bit(MCLK);
// Connect the clock units to the 2mhz clock by default. They can't reset without it.
connect_gclk_to_peripheral(5, I2S_GCLK_ID_0);
connect_gclk_to_peripheral(5, I2S_GCLK_ID_1);
#endif
#ifdef SAMD21
_pm_enable_bus_clock(PM_BUS_APBC, I2S);
#endif
if (I2S->CTRLA.bit.ENABLE == 0) {
I2S->CTRLA.bit.SWRST = 1;
while (I2S->CTRLA.bit.SWRST == 1) {}
} else {
#ifdef SAMD21
if ((I2S->CTRLA.vec.SEREN & (1 << serializer)) != 0) {
mp_raise_RuntimeError("Serializer in use");
}
#endif
#ifdef SAMD51
if (I2S->CTRLA.bit.TXEN == 1) {
mp_raise_RuntimeError("Serializer in use");
}
#endif
}
#ifdef SAMD51
#define GPIO_I2S_FUNCTION GPIO_PIN_FUNCTION_J
#endif
#ifdef SAMD21
#define GPIO_I2S_FUNCTION GPIO_PIN_FUNCTION_G
#endif
assert_pin_free(bit_clock);
assert_pin_free(word_select);
assert_pin_free(data);
self->bit_clock = bit_clock;
self->word_select = word_select;
self->data = data;
claim_pin(bit_clock);
claim_pin(word_select);
claim_pin(data);
gpio_set_pin_function(self->bit_clock->pin, GPIO_I2S_FUNCTION);
gpio_set_pin_function(self->word_select->pin, GPIO_I2S_FUNCTION);
gpio_set_pin_function(self->data->pin, GPIO_I2S_FUNCTION);
self->left_justified = left_justified;
self->playing = false;
audio_dma_init(&self->dma);
}
bool common_hal_audiobusio_i2sout_deinited(audiobusio_i2sout_obj_t* self) {
return self->bit_clock == mp_const_none;
}
void common_hal_audiobusio_i2sout_deinit(audiobusio_i2sout_obj_t* self) {
if (common_hal_audiobusio_i2sout_deinited(self)) {
return;
}
reset_pin(self->bit_clock->pin);
self->bit_clock = mp_const_none;
reset_pin(self->word_select->pin);
self->word_select = mp_const_none;
reset_pin(self->data->pin);
self->data = mp_const_none;
}
void common_hal_audiobusio_i2sout_play(audiobusio_i2sout_obj_t* self,
mp_obj_t sample, bool loop) {
if (common_hal_audiobusio_i2sout_get_playing(self)) {
common_hal_audiobusio_i2sout_stop(self);
}
#ifdef SAMD21
if ((I2S->CTRLA.vec.CKEN & (1 << self->clock_unit)) == 1) {
mp_raise_RuntimeError("Clock unit in use");
}
#endif
uint8_t bits_per_sample = audiosample_bits_per_sample(sample);
// We always output stereo so output twice as many bits.
uint16_t bits_per_sample_output = bits_per_sample * 2;
uint16_t divisor = 48000000 / (bits_per_sample_output * audiosample_sample_rate(sample));
// Find a free GCLK to generate the MCLK signal.
uint8_t gclk = find_free_gclk(divisor);
if (gclk > GCLK_GEN_NUM) {
mp_raise_RuntimeError("Unable to find free GCLK");
}
self->gclk = gclk;
uint32_t clkctrl = I2S_CLKCTRL_MCKSEL_GCLK |
I2S_CLKCTRL_NBSLOTS(1) |
I2S_CLKCTRL_FSWIDTH_HALF;
if (self->left_justified) {
clkctrl |= I2S_CLKCTRL_BITDELAY_LJ;
} else {
clkctrl |= I2S_CLKCTRL_FSOUTINV | I2S_CLKCTRL_BITDELAY_I2S;
}
uint8_t channel_count = audiosample_channel_count(sample);
if (channel_count > 2) {
mp_raise_ValueError("Too many channels in sample.");
}
#ifdef SAMD21
uint32_t serctrl = (self->clock_unit << I2S_SERCTRL_CLKSEL_Pos) | SERCTRL(SERMODE_TX) | I2S_SERCTRL_TXSAME_SAME | I2S_SERCTRL_EXTEND_MSBIT | I2S_SERCTRL_TXDEFAULT_ONE | I2S_SERCTRL_SLOTADJ_LEFT;
#endif
#ifdef SAMD51
uint32_t serctrl = (self->clock_unit << I2S_RXCTRL_CLKSEL_Pos) | I2S_TXCTRL_TXSAME_SAME;
#endif
if (audiosample_channel_count(sample) == 1) {
serctrl |= SERCTRL(MONO_MONO);
} else {
serctrl |= SERCTRL(MONO_STEREO);
}
if (bits_per_sample == 8) {
serctrl |= SERCTRL(DATASIZE_8C);
clkctrl |= I2S_CLKCTRL_SLOTSIZE_8;
} else if (bits_per_sample == 16) {
serctrl |= SERCTRL(DATASIZE_16C);
clkctrl |= I2S_CLKCTRL_SLOTSIZE_16;
}
// Configure the I2S peripheral
I2S->CTRLA.bit.ENABLE = 0;
while (I2S->SYNCBUSY.bit.ENABLE == 1) {}
I2S->CLKCTRL[self->clock_unit].reg = clkctrl;
#ifdef SAMD21
I2S->SERCTRL[self->serializer].reg = serctrl;
#endif
#ifdef SAMD51
I2S->TXCTRL.reg = serctrl;
#endif
// The DFLL is always a 48mhz clock
enable_clock_generator(self->gclk, CLOCK_48MHZ, divisor);
connect_gclk_to_peripheral(self->gclk, I2S_GCLK_ID_0 + self->clock_unit);
I2S->CTRLA.bit.ENABLE = 1;
while (I2S->SYNCBUSY.bit.ENABLE == 1) {}
#ifdef SAMD21
uint32_t tx_register = (uint32_t) &I2S->DATA[self->serializer].reg;
uint8_t dmac_id = I2S_DMAC_ID_TX_0 + self->serializer;
#endif
#ifdef SAMD51
uint32_t tx_register = (uint32_t) &I2S->TXDATA.reg;
uint8_t dmac_id = I2S_DMAC_ID_TX_0;
#endif
audio_dma_result result = audio_dma_setup_playback(&self->dma, sample, loop, false, 0,
true /* output signed */, tx_register, dmac_id);
if (result == AUDIO_DMA_DMA_BUSY) {
common_hal_audiobusio_i2sout_stop(self);
mp_raise_RuntimeError("No DMA channel found");
} else if (result == AUDIO_DMA_MEMORY_ERROR) {
common_hal_audiobusio_i2sout_stop(self);
mp_raise_RuntimeError("Unable to allocate buffers for signed conversion");
}
I2S->INTFLAG.reg = I2S_INTFLAG_TXUR0 | I2S_INTFLAG_TXUR1;
I2S->CTRLA.vec.CKEN = 1 << self->clock_unit;
while ((I2S->SYNCBUSY.vec.CKEN & (1 << self->clock_unit)) != 0) {}
// Init the serializer after the clock. Otherwise, it will never enable because its unclocked.
#ifdef SAMD21
I2S->CTRLA.vec.SEREN = 1 << self->serializer;
while ((I2S->SYNCBUSY.vec.SEREN & (1 << self->serializer)) != 0) {}
#endif
#ifdef SAMD51
I2S->CTRLA.bit.TXEN = 1;
while (I2S->SYNCBUSY.bit.TXEN == 1) {}
#endif
self->playing = true;
}
void common_hal_audiobusio_i2sout_stop(audiobusio_i2sout_obj_t* self) {
audio_dma_stop(&self->dma);
#ifdef SAMD21
I2S->CTRLA.vec.SEREN &= ~(1 << self->serializer);
while ((I2S->SYNCBUSY.vec.SEREN & (1 << self->serializer)) != 0) {}
#endif
#ifdef SAMD51
I2S->CTRLA.bit.TXEN = 0;
while (I2S->SYNCBUSY.bit.TXEN == 1) {}
#endif
#ifdef SAMD21
if (self->clock_unit == 0) {
I2S->CTRLA.bit.CKEN0 = 0;
while (I2S->SYNCBUSY.bit.CKEN0 == 1) {}
} else {
I2S->CTRLA.bit.CKEN1 = 0;
while (I2S->SYNCBUSY.bit.CKEN1 == 1) {}
}
#endif
disconnect_gclk_from_peripheral(self->gclk, I2S_GCLK_ID_0 + self->clock_unit);
disable_clock_generator(self->gclk);
#ifdef SAMD51
connect_gclk_to_peripheral(5, I2S_GCLK_ID_0 + self->clock_unit);
#endif
self->playing = false;
}
bool common_hal_audiobusio_i2sout_get_playing(audiobusio_i2sout_obj_t* self) {
bool still_playing = audio_dma_get_playing(&self->dma);
if (self->playing && !still_playing) {
common_hal_audiobusio_i2sout_stop(self);
}
return still_playing;
}

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@ -0,0 +1,51 @@
/*
* This file is part of the MicroPython project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2017 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.
*/
#ifndef MICROPY_INCLUDED_ATMEL_SAMD_COMMON_HAL_AUDIOBUSIO_I2SOUT_H
#define MICROPY_INCLUDED_ATMEL_SAMD_COMMON_HAL_AUDIOBUSIO_I2SOUT_H
#include "common-hal/microcontroller/Pin.h"
#include "audio_dma.h"
#include "py/obj.h"
// 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;
const mcu_pin_obj_t *bit_clock;
const mcu_pin_obj_t *word_select;
const mcu_pin_obj_t *data;
uint8_t clock_unit;
uint8_t serializer;
uint8_t gclk;
bool playing;
audio_dma_t dma;
} audiobusio_i2sout_obj_t;
void i2sout_reset(void);
#endif // MICROPY_INCLUDED_ATMEL_SAMD_COMMON_HAL_AUDIOBUSIO_I2SOUT_H

View File

@ -37,7 +37,6 @@
#include "shared-bindings/audiobusio/PDMIn.h"
#include "shared-bindings/microcontroller/Pin.h"
#include "asf/sam0/drivers/port/port.h"
#include "samd21_pins.h"
#include "shared_dma.h"

View File

@ -28,8 +28,6 @@
#define MICROPY_INCLUDED_ATMEL_SAMD_COMMON_HAL_AUDIOBUSIO_AUDIOOUT_H
#include "common-hal/microcontroller/Pin.h"
#include "asf/sam0/drivers/i2s/i2s.h"
#include "asf/sam0/drivers/tc/tc.h"
#include "extmod/vfs_fat_file.h"
#include "py/obj.h"
@ -39,7 +37,6 @@ typedef struct {
const mcu_pin_obj_t *clock_pin;
const mcu_pin_obj_t *data_pin;
uint32_t frequency;
struct i2s_module i2s_instance;
uint8_t serializer;
uint8_t clock_unit;
uint8_t bytes_per_sample;

View File

@ -35,155 +35,128 @@
#include "shared-bindings/audioio/AudioOut.h"
#include "shared-bindings/microcontroller/Pin.h"
#include "asf/sam0/drivers/dac/dac.h"
#include "asf/sam0/drivers/dma/dma.h"
#include "asf/sam0/drivers/events/events.h"
#include "asf/sam0/drivers/port/port.h"
#include "asf/sam0/drivers/tc/tc.h"
#include "atmel_start_pins.h"
#include "hal/include/hal_gpio.h"
#include "hpl/gclk/hpl_gclk_base.h"
#include "peripheral_clk_config.h"
#ifdef SAMD21
#include "hpl/pm/hpl_pm_base.h"
#endif
#include "audio_dma.h"
#include "events.h"
#include "samd21_pins.h"
#include "shared_dma.h"
#undef ENABLE
// Shared with PWMOut
// TODO(tannewt): Factor these out so audioio can exist without PWMOut.
extern uint32_t target_timer_frequencies[TC_INST_NUM + TCC_INST_NUM];
extern uint8_t timer_refcount[TC_INST_NUM + TCC_INST_NUM];
extern const uint16_t prescaler[8];
// This timer is shared amongst all AudioOut objects under the assumption that
// the code is single threaded. The audioout_sample_timer, audioout_dac_instance,
// audioout_sample_event, and audioout_dac_event pointers live in
// MICROPY_PORT_ROOT_POINTERS so they don't get garbage collected.
// The AudioOut object is being currently played. Only it can pause the timer
// and change its frequency.
static audioio_audioout_obj_t* active_audioout;
static uint8_t refcount = 0;
struct wave_format_chunk {
uint16_t audio_format;
uint16_t num_channels;
uint32_t sample_rate;
uint32_t byte_rate;
uint16_t block_align;
uint16_t bits_per_sample;
uint16_t extra_params; // Assumed to be zero below.
};
#include "timers.h"
void audioout_reset(void) {
// Only reset DMA. PWMOut will reset the timer. Other code will reset the DAC.
refcount = 0;
MP_STATE_VM(audioout_sample_timer) = NULL;
MP_STATE_VM(audiodma_block_counter) = NULL;
MP_STATE_VM(audioout_dac_instance) = NULL;
if (MP_STATE_VM(audioout_sample_event) != NULL) {
events_detach_user(MP_STATE_VM(audioout_sample_event), EVSYS_ID_USER_DAC_START);
events_release(MP_STATE_VM(audioout_sample_event));
}
MP_STATE_VM(audioout_sample_event) = NULL;
if (MP_STATE_VM(audiodma_block_event) != NULL) {
events_release(MP_STATE_VM(audiodma_block_event));
}
MP_STATE_VM(audiodma_block_event) = NULL;
if (MP_STATE_VM(audioout_dac_event) != NULL) {
events_detach_user(MP_STATE_VM(audioout_dac_event), EVSYS_ID_USER_DMAC_CH_0);
events_release(MP_STATE_VM(audioout_dac_event));
}
MP_STATE_VM(audioout_dac_event) = NULL;
dma_abort_job(&audio_dma);
}
// WARN(tannewt): DO NOT print from here. It calls background tasks and causes a
// stack overflow.
void audioout_background(void) {
if (MP_STATE_VM(audiodma_block_counter) != NULL &&
active_audioout != NULL &&
active_audioout->second_buffer != NULL &&
active_audioout->last_loaded_block < tc_get_count_value(MP_STATE_VM(audiodma_block_counter))) {
uint8_t* buffer;
if (tc_get_count_value(MP_STATE_VM(audiodma_block_counter)) % 2 == 1) {
buffer = active_audioout->buffer;
} else {
buffer = active_audioout->second_buffer;
}
uint16_t num_bytes_to_load = active_audioout->len;
if (num_bytes_to_load > active_audioout->bytes_remaining) {
num_bytes_to_load = active_audioout->bytes_remaining;
}
UINT length_read;
f_read(&active_audioout->file->fp, buffer, num_bytes_to_load, &length_read);
active_audioout->bytes_remaining -= length_read;
active_audioout->last_loaded_block += 1;
if (active_audioout->bytes_remaining == 0) {
if (active_audioout->loop) {
// Loop back to the start of the file.
f_lseek(&active_audioout->file->fp, active_audioout->data_start);
active_audioout->bytes_remaining = active_audioout->file_length;
f_read(&active_audioout->file->fp, buffer, active_audioout->len - num_bytes_to_load, &length_read);
active_audioout->bytes_remaining -= length_read;
} else {
DmacDescriptor* descriptor = audio_dma.descriptor;
if (buffer == active_audioout->second_buffer) {
descriptor = active_audioout->second_descriptor;
}
descriptor->BTCNT.reg = length_read / active_audioout->bytes_per_sample;
descriptor->SRCADDR.reg = ((uint32_t) buffer) + length_read;
descriptor->DESCADDR.reg = 0;
}
}
if (active_audioout->bytes_per_sample == 2) {
// Undo twos complement.
for (uint16_t i = 0; i < length_read / 2; i++) {
buffer[2 * i + 1] ^= 0x80;
}
}
void common_hal_audioio_audioout_construct(audioio_audioout_obj_t* self,
const mcu_pin_obj_t* left_channel, const mcu_pin_obj_t* right_channel) {
#ifdef SAMD21
if (right_channel != NULL) {
mp_raise_ValueError("Right channel unsupported");
}
}
static void shared_construct(audioio_audioout_obj_t* self, const mcu_pin_obj_t* pin) {
assert_pin_free(pin);
// Configure the DAC to output on input event and to output an empty event
// that triggers the DMA to load the next sample.
MP_STATE_VM(audioout_dac_instance) = gc_alloc(sizeof(struct dac_module), false);
if (MP_STATE_VM(audioout_dac_instance) == NULL) {
mp_raise_msg(&mp_type_MemoryError, "");
if (left_channel != &pin_PA02) {
mp_raise_ValueError("Invalid pin");
}
struct dac_config config_dac;
dac_get_config_defaults(&config_dac);
config_dac.left_adjust = true;
config_dac.reference = DAC_REFERENCE_AVCC;
config_dac.clock_source = GCLK_GENERATOR_0;
enum status_code status = dac_init(MP_STATE_VM(audioout_dac_instance), DAC, &config_dac);
if (status != STATUS_OK) {
common_hal_audioio_audioout_deinit(self);
mp_raise_OSError(MP_EIO);
return;
assert_pin_free(left_channel);
claim_pin(left_channel);
#endif
#ifdef SAMD51
self->right_channel = NULL;
if (left_channel != &pin_PA02 && left_channel != &pin_PA05) {
mp_raise_ValueError("Invalid pin for left channel");
}
assert_pin_free(left_channel);
if (right_channel != NULL && right_channel != &pin_PA02 && right_channel != &pin_PA05) {
mp_raise_ValueError("Invalid pin for right channel");
}
if (right_channel == left_channel) {
mp_raise_ValueError("Cannot output both channels on the same pin");
}
claim_pin(left_channel);
if (right_channel != NULL) {
claim_pin(right_channel);
self->right_channel = right_channel;
gpio_set_pin_function(self->right_channel->pin, GPIO_PIN_FUNCTION_B);
audio_dma_init(&self->right_dma);
}
#endif
self->left_channel = left_channel;
gpio_set_pin_function(self->left_channel->pin, GPIO_PIN_FUNCTION_B);
audio_dma_init(&self->left_dma);
#ifdef SAMD51
hri_mclk_set_APBDMASK_DAC_bit(MCLK);
#endif
#ifdef SAMD21
_pm_enable_bus_clock(PM_BUS_APBC, DAC);
#endif
// SAMD21: This clock should be <= 12 MHz, per datasheet section 47.6.3.
// SAMD51: This clock should be <= 350kHz, per datasheet table 37-6.
_gclk_enable_channel(DAC_GCLK_ID, CONF_GCLK_DAC_SRC);
// There is a small chance the other output is being used by AnalogOut on the SAMD51 so
// only reset if the DAC is disabled.
if (DAC->CTRLA.bit.ENABLE == 0) {
DAC->CTRLA.bit.SWRST = 1;
while (DAC->CTRLA.bit.SWRST == 1) {}
}
struct dac_chan_config channel_config;
dac_chan_get_config_defaults(&channel_config);
dac_chan_set_config(MP_STATE_VM(audioout_dac_instance), DAC_CHANNEL_0, &channel_config);
dac_chan_enable(MP_STATE_VM(audioout_dac_instance), DAC_CHANNEL_0);
bool channel0_enabled = true;
#ifdef SAMD51
channel0_enabled = self->left_channel == &pin_PA02 || self->right_channel == &pin_PA02;
bool channel1_enabled = self->left_channel == &pin_PA05 || self->right_channel == &pin_PA05;
#endif
struct dac_events events_dac = { .generate_event_on_buffer_empty = true,
.on_event_start_conversion = true };
dac_enable_events(MP_STATE_VM(audioout_dac_instance), &events_dac);
if (channel0_enabled) {
#ifdef SAMD21
DAC->EVCTRL.reg |= DAC_EVCTRL_STARTEI;
DAC->CTRLB.reg = DAC_CTRLB_REFSEL_AVCC |
DAC_CTRLB_LEFTADJ |
DAC_CTRLB_EOEN;
#endif
#ifdef SAMD51
DAC->EVCTRL.reg |= DAC_EVCTRL_STARTEI0;
DAC->DACCTRL[0].reg = DAC_DACCTRL_CCTRL_CC1M |
DAC_DACCTRL_ENABLE |
DAC_DACCTRL_LEFTADJ;
DAC->CTRLB.reg = DAC_CTRLB_REFSEL_VREFPU;
#endif
}
#ifdef SAMD51
if (channel1_enabled) {
DAC->EVCTRL.reg |= DAC_EVCTRL_STARTEI1;
DAC->DACCTRL[1].reg = DAC_DACCTRL_CCTRL_CC1M |
DAC_DACCTRL_ENABLE |
DAC_DACCTRL_LEFTADJ;
DAC->CTRLB.reg = DAC_CTRLB_REFSEL_VREFPU;
}
#endif
// Figure out which timer we are using.
// Re-enable the DAC
DAC->CTRLA.bit.ENABLE = 1;
#ifdef SAMD21
while (DAC->STATUS.bit.SYNCBUSY == 1) {}
#endif
#ifdef SAMD51
while (DAC->SYNCBUSY.bit.ENABLE == 1) {}
#endif
// Use a timer to coordinate when DAC conversions occur.
Tc *t = NULL;
Tc *tcs[TC_INST_NUM] = TC_INSTS;
uint8_t tc_index = TC_INST_NUM;
for (uint8_t i = TC_INST_NUM; i > 0; i--) {
if (tcs[i - 1]->COUNT16.CTRLA.bit.ENABLE == 0) {
t = tcs[i - 1];
if (tc_insts[i - 1]->COUNT16.CTRLA.bit.ENABLE == 0) {
t = tc_insts[i - 1];
tc_index = i - 1;
break;
}
}
@ -192,234 +165,77 @@ static void shared_construct(audioio_audioout_obj_t* self, const mcu_pin_obj_t*
mp_raise_RuntimeError("All timers in use");
return;
}
MP_STATE_VM(audioout_sample_timer) = gc_alloc(sizeof(struct tc_module), false);
if (MP_STATE_VM(audioout_sample_timer) == NULL) {
common_hal_audioio_audioout_deinit(self);
mp_raise_msg(&mp_type_MemoryError, "");
}
self->tc_index = tc_index;
// Use the 48mhz clocks on both the SAMD21 and 51 because we will be going much slower.
uint8_t tc_gclk = 0;
#ifdef SAMD51
tc_gclk = 1;
#endif
turn_on_clocks(true, tc_index, tc_gclk);
// Don't bother setting the period. We set it before you playback anything.
struct tc_config config_tc;
tc_get_config_defaults(&config_tc);
config_tc.counter_size = TC_COUNTER_SIZE_16BIT;
config_tc.clock_prescaler = TC_CLOCK_PRESCALER_DIV1;
config_tc.wave_generation = TC_WAVE_GENERATION_MATCH_FREQ;
if (tc_init(MP_STATE_VM(audioout_sample_timer), t, &config_tc) != STATUS_OK) {
common_hal_audioio_audioout_deinit(self);
mp_raise_OSError(MP_EIO);
return;
};
struct tc_events events_tc;
events_tc.generate_event_on_overflow = true;
events_tc.on_event_perform_action = false;
events_tc.event_action = TC_EVENT_ACTION_OFF;
tc_enable_events(MP_STATE_VM(audioout_sample_timer), &events_tc);
tc_enable(MP_STATE_VM(audioout_sample_timer));
tc_stop_counter(MP_STATE_VM(audioout_sample_timer));
tc_set_enable(t, false);
tc_reset(t);
#ifdef SAMD51
t->COUNT16.WAVE.reg = TC_WAVE_WAVEGEN_MFRQ;
#endif
#ifdef SAMD21
t->COUNT16.CTRLA.bit.WAVEGEN = TC_CTRLA_WAVEGEN_MFRQ_Val;
#endif
t->COUNT16.EVCTRL.reg = TC_EVCTRL_OVFEO;
tc_set_enable(t, true);
t->COUNT16.CTRLBSET.reg = TC_CTRLBSET_CMD_STOP;
// Connect the timer overflow event, which happens at the target frequency,
// to the DAC conversion trigger.
MP_STATE_VM(audioout_sample_event) = gc_alloc(sizeof(struct events_resource), false);
if (MP_STATE_VM(audioout_sample_event) == NULL) {
common_hal_audioio_audioout_deinit(self);
mp_raise_msg(&mp_type_MemoryError, "");
}
struct events_config config;
events_get_config_defaults(&config);
// to the DAC conversion trigger(s).
#ifdef SAMD21
#define FIRST_TC_GEN_ID EVSYS_ID_GEN_TC3_OVF
#endif
#ifdef SAMD51
#define FIRST_TC_GEN_ID EVSYS_ID_GEN_TC0_OVF
#endif
uint8_t tc_gen_id = FIRST_TC_GEN_ID + 3 * tc_index;
uint8_t generator = EVSYS_ID_GEN_TC3_OVF;
if (t == TC4) {
generator = EVSYS_ID_GEN_TC4_OVF;
} else if (t == TC5) {
generator = EVSYS_ID_GEN_TC5_OVF;
#ifdef TC6
} else if (t == TC6) {
generator = EVSYS_ID_GEN_TC6_OVF;
#endif
#ifdef TC7
} else if (t == TC7) {
generator = EVSYS_ID_GEN_TC7_OVF;
#endif
}
turn_on_event_system();
config.generator = generator;
config.path = EVENTS_PATH_ASYNCHRONOUS;
if (events_allocate(MP_STATE_VM(audioout_sample_event), &config) != STATUS_OK ||
events_attach_user(MP_STATE_VM(audioout_sample_event), EVSYS_ID_USER_DAC_START) != STATUS_OK) {
common_hal_audioio_audioout_deinit(self);
mp_raise_OSError(MP_EIO);
return;
}
// Find a free event channel. We start at the highest channels because we only need and async
// path.
uint8_t channel = find_async_event_channel();
#ifdef SAMD51
connect_event_user_to_channel(EVSYS_ID_USER_DAC_START_1, channel);
#define EVSYS_ID_USER_DAC_START EVSYS_ID_USER_DAC_START_0
#endif
connect_event_user_to_channel(EVSYS_ID_USER_DAC_START, channel);
init_async_event_channel(channel, tc_gen_id);
// Connect the DAC to DMA
MP_STATE_VM(audioout_dac_event) = gc_alloc(sizeof(struct events_resource), false);
if (MP_STATE_VM(audioout_dac_event) == NULL) {
common_hal_audioio_audioout_deinit(self);
mp_raise_msg(&mp_type_MemoryError, "");
}
events_get_config_defaults(&config);
config.generator = EVSYS_ID_GEN_DAC_EMPTY;
config.path = EVENTS_PATH_ASYNCHRONOUS;
if (events_allocate(MP_STATE_VM(audioout_dac_event), &config) != STATUS_OK ||
events_attach_user(MP_STATE_VM(audioout_dac_event), EVSYS_ID_USER_DMAC_CH_0) != STATUS_OK) {
common_hal_audioio_audioout_deinit(self);
mp_raise_OSError(MP_EIO);
return;
}
self->tc_to_dac_event_channel = channel;
// Leave the DMA setup to the specific constructor.
}
void common_hal_audioio_audioout_construct_from_buffer(audioio_audioout_obj_t* self,
const mcu_pin_obj_t* pin,
uint16_t* buffer,
uint32_t len,
uint8_t bytes_per_sample) {
self->pin = pin;
if (pin != &pin_PA02) {
mp_raise_ValueError("Invalid pin");
}
if (refcount == 0) {
refcount++;
shared_construct(self, pin);
}
self->buffer = (uint8_t*) buffer;
self->second_buffer = NULL;
self->bytes_per_sample = bytes_per_sample;
self->len = len;
self->frequency = 8000;
}
void common_hal_audioio_audioout_construct_from_file(audioio_audioout_obj_t* self,
const mcu_pin_obj_t* pin,
pyb_file_obj_t* file) {
self->pin = pin;
if (pin != &pin_PA02) {
mp_raise_ValueError("Invalid pin");
}
if (refcount == 0) {
refcount++;
shared_construct(self, pin);
}
if (MP_STATE_VM(audiodma_block_counter) == NULL && !allocate_block_counter()) {
mp_raise_RuntimeError("Unable to allocate audio DMA block counter.");
}
// Load the wave
self->file = file;
uint8_t chunk_header[16];
f_rewind(&self->file->fp);
UINT bytes_read;
f_read(&self->file->fp, chunk_header, 16, &bytes_read);
if (bytes_read != 16 ||
memcmp(chunk_header, "RIFF", 4) != 0 ||
memcmp(chunk_header + 8, "WAVEfmt ", 8) != 0) {
mp_raise_ValueError("Invalid wave file");
}
uint32_t format_size;
f_read(&self->file->fp, &format_size, 4, &bytes_read);
if (bytes_read != 4 ||
format_size > sizeof(struct wave_format_chunk)) {
mp_raise_ValueError("Invalid format chunk size");
}
struct wave_format_chunk format;
f_read(&self->file->fp, &format, format_size, &bytes_read);
if (bytes_read != format_size) {
}
if (format.audio_format != 1 ||
format.num_channels > 1 ||
format.bits_per_sample > 16 ||
(format_size == 18 &&
format.extra_params != 0)) {
mp_raise_ValueError("Unsupported format");
}
// Get the frequency
self->frequency = format.sample_rate;
self->len = 512;
self->bytes_per_sample = format.bits_per_sample / 8;
// TODO(tannewt): Skip any extra chunks that occur before the data section.
uint8_t data_tag[4];
f_read(&self->file->fp, &data_tag, 4, &bytes_read);
if (bytes_read != 4 ||
memcmp((uint8_t *) data_tag, "data", 4) != 0) {
mp_raise_ValueError("Data chunk must follow fmt chunk");
}
uint32_t data_length;
f_read(&self->file->fp, &data_length, 4, &bytes_read);
if (bytes_read != 4) {
mp_raise_ValueError("Invalid file");
}
self->file_length = data_length;
self->data_start = self->file->fp.fptr;
// Try to allocate two buffers, one will be loaded from file and the other
// DMAed to DAC.
self->buffer = gc_alloc(self->len, false);
if (self->buffer == NULL) {
common_hal_audioio_audioout_deinit(self);
mp_raise_msg(&mp_type_MemoryError, "");
}
self->second_buffer = gc_alloc(self->len, false);
if (self->second_buffer == NULL) {
common_hal_audioio_audioout_deinit(self);
mp_raise_msg(&mp_type_MemoryError, "");
}
self->second_descriptor = gc_alloc(sizeof(DmacDescriptor), false);
if (self->second_descriptor == NULL) {
common_hal_audioio_audioout_deinit(self);
mp_raise_msg(&mp_type_MemoryError, "");
}
// Leave the DMA setup to playback.
}
bool common_hal_audioio_audioout_deinited(audioio_audioout_obj_t* self) {
return self->pin == mp_const_none;
return self->left_channel == mp_const_none;
}
void common_hal_audioio_audioout_deinit(audioio_audioout_obj_t* self) {
if (common_hal_audioio_audioout_deinited(self)) {
return;
}
refcount--;
if (refcount == 0) {
if (MP_STATE_VM(audioout_sample_timer) != NULL) {
tc_reset(MP_STATE_VM(audioout_sample_timer));
gc_free(MP_STATE_VM(audioout_sample_timer));
MP_STATE_VM(audioout_sample_timer) = NULL;
}
if (MP_STATE_VM(audioout_dac_instance) != NULL) {
dac_reset(MP_STATE_VM(audioout_dac_instance));
gc_free(MP_STATE_VM(audioout_dac_instance));
MP_STATE_VM(audioout_dac_instance) = NULL;
}
if (MP_STATE_VM(audioout_sample_event) != NULL) {
events_detach_user(MP_STATE_VM(audioout_sample_event), EVSYS_ID_USER_DAC_START);
events_release(MP_STATE_VM(audioout_sample_event));
gc_free(MP_STATE_VM(audioout_sample_event));
MP_STATE_VM(audioout_sample_event) = NULL;
}
if (MP_STATE_VM(audioout_dac_event) != NULL) {
events_release(MP_STATE_VM(audioout_dac_event));
gc_free(MP_STATE_VM(audioout_dac_event));
MP_STATE_VM(audioout_dac_event) = NULL;
}
reset_pin(self->pin->pin);
}
self->pin = mp_const_none;
disable_event_channel(self->tc_to_dac_event_channel);
reset_pin(self->left_channel->pin);
self->left_channel = mp_const_none;
#ifdef SAMD51
reset_pin(self->right_channel->pin);
self->right_channel = mp_const_none;
#endif
}
static void set_timer_frequency(uint32_t frequency) {
uint32_t system_clock = system_cpu_clock_get_hz();
static void set_timer_frequency(Tc* timer, uint32_t frequency) {
uint32_t system_clock = 48000000;
uint32_t new_top;
uint8_t new_divisor;
for (new_divisor = 0; new_divisor < 8; new_divisor++) {
@ -428,133 +244,91 @@ static void set_timer_frequency(uint32_t frequency) {
break;
}
}
uint8_t old_divisor = MP_STATE_VM(audioout_sample_timer)->hw->COUNT16.CTRLA.bit.PRESCALER;
uint8_t old_divisor = timer->COUNT16.CTRLA.bit.PRESCALER;
if (new_divisor != old_divisor) {
tc_disable(MP_STATE_VM(audioout_sample_timer));
MP_STATE_VM(audioout_sample_timer)->hw->COUNT16.CTRLA.bit.PRESCALER = new_divisor;
tc_enable(MP_STATE_VM(audioout_sample_timer));
}
while (tc_is_syncing(MP_STATE_VM(audioout_sample_timer))) {
/* Wait for sync */
}
MP_STATE_VM(audioout_sample_timer)->hw->COUNT16.CC[0].reg = new_top;
while (tc_is_syncing(MP_STATE_VM(audioout_sample_timer))) {
/* Wait for sync */
tc_set_enable(timer, false);
timer->COUNT16.CTRLA.bit.PRESCALER = new_divisor;
tc_set_enable(timer, true);
}
tc_wait_for_sync(timer);
timer->COUNT16.CC[0].reg = new_top;
tc_wait_for_sync(timer);
}
void common_hal_audioio_audioout_play(audioio_audioout_obj_t* self, bool loop) {
common_hal_audioio_audioout_get_playing(self);
// Shut down any active playback.
if (active_audioout != NULL) {
tc_stop_counter(MP_STATE_VM(audioout_sample_timer));
dma_abort_job(&audio_dma);
} else {
dac_enable(MP_STATE_VM(audioout_dac_instance));
void common_hal_audioio_audioout_play(audioio_audioout_obj_t* self,
mp_obj_t sample, bool loop) {
if (common_hal_audioio_audioout_get_playing(self)) {
common_hal_audioio_audioout_stop(self);
}
switch_audiodma_trigger(DAC_DMAC_ID_EMPTY);
struct dma_descriptor_config descriptor_config;
dma_descriptor_get_config_defaults(&descriptor_config);
if (self->bytes_per_sample == 2) {
descriptor_config.beat_size = DMA_BEAT_SIZE_HWORD;
} else {
descriptor_config.beat_size = DMA_BEAT_SIZE_BYTE;
audio_dma_result result = AUDIO_DMA_OK;
#ifdef SAMD21
result = audio_dma_setup_playback(&self->left_dma, sample, loop, true, 0,
false /* output unsigned */,
(uint32_t) &DAC->DATABUF.reg,
DAC_DMAC_ID_EMPTY);
#endif
#ifdef SAMD51
uint32_t left_channel_reg = (uint32_t) &DAC->DATABUF[0].reg;
uint8_t left_channel_trigger = DAC_DMAC_ID_EMPTY_0;
uint32_t right_channel_reg = 0;
uint8_t right_channel_trigger = 0;
if (self->left_channel == &pin_PA05) {
left_channel_reg = (uint32_t) &DAC->DATABUF[1].reg;
left_channel_trigger = DAC_DMAC_ID_EMPTY_1;
} else if (self->right_channel == &pin_PA05) {
right_channel_reg = (uint32_t) &DAC->DATABUF[1].reg;
right_channel_trigger = DAC_DMAC_ID_EMPTY_1;
}
descriptor_config.dst_increment_enable = false;
// Block transfer count is the number of beats per block (aka descriptor).
// In this case there are two bytes per beat so divide the length by two.
descriptor_config.block_transfer_count = self->len / self->bytes_per_sample;
descriptor_config.source_address = ((uint32_t)self->buffer + self->len);
descriptor_config.destination_address = ((uint32_t)&DAC->DATABUF.reg + 1);
descriptor_config.event_output_selection = DMA_EVENT_OUTPUT_BLOCK;
self->loop = loop;
if (self->second_buffer == NULL) {
if (loop) {
descriptor_config.next_descriptor_address = ((uint32_t)audio_dma.descriptor);
} else {
descriptor_config.next_descriptor_address = 0;
}
} else {
descriptor_config.next_descriptor_address = ((uint32_t)self->second_descriptor);
if (self->right_channel == &pin_PA02) {
right_channel_reg = (uint32_t) &DAC->DATABUF[0].reg;
right_channel_trigger = DAC_DMAC_ID_EMPTY_0;
}
dma_descriptor_create(audio_dma.descriptor, &descriptor_config);
if (self->second_buffer != NULL) {
// TODO(tannewt): Correctly set the end of this.
descriptor_config.block_transfer_count = self->len / self->bytes_per_sample;
descriptor_config.source_address = ((uint32_t)self->second_buffer + self->len);
descriptor_config.next_descriptor_address = ((uint32_t)audio_dma.descriptor);
dma_descriptor_create(self->second_descriptor, &descriptor_config);
self->last_loaded_block = 0;
self->bytes_remaining = self->file_length;
f_lseek(&self->file->fp, self->data_start);
// Seek to the start of the PCM.
UINT length_read;
f_read(&self->file->fp, self->buffer, self->len, &length_read);
self->bytes_remaining -= length_read;
if (self->bytes_per_sample == 2) {
// Undo twos complement.
for (uint16_t i = 0; i < length_read / 2; i++) {
self->buffer[2 * i + 1] ^= 0x80;
}
}
f_read(&self->file->fp, self->second_buffer, self->len, &length_read);
self->bytes_remaining -= length_read;
if (self->bytes_per_sample == 2) {
// Undo twos complement.
for (uint16_t i = 0; i < length_read / 2; i++) {
self->second_buffer[2 * i + 1] ^= 0x80;
}
result = audio_dma_setup_playback(&self->left_dma, sample, loop, true, 0,
false /* output unsigned */,
left_channel_reg,
left_channel_trigger);
if (right_channel_reg != 0 && result == AUDIO_DMA_OK) {
result = audio_dma_setup_playback(&self->right_dma, sample, loop, true, 1,
false /* output unsigned */,
right_channel_reg,
right_channel_trigger);
}
#endif
if (result != AUDIO_DMA_OK) {
audio_dma_stop(&self->left_dma);
#ifdef SAMD51
audio_dma_stop(&self->right_dma);
#endif
if (result == AUDIO_DMA_DMA_BUSY) {
mp_raise_RuntimeError("No DMA channel found");
} else if (result == AUDIO_DMA_MEMORY_ERROR) {
mp_raise_RuntimeError("Unable to allocate buffers for signed conversion");
}
}
active_audioout = self;
dma_start_transfer_job(&audio_dma);
if (MP_STATE_VM(audiodma_block_counter) != NULL) {
tc_start_counter(MP_STATE_VM(audiodma_block_counter));
}
set_timer_frequency(self->frequency);
tc_start_counter(MP_STATE_VM(audioout_sample_timer));
Tc* timer = tc_insts[self->tc_index];
set_timer_frequency(timer, audiosample_sample_rate(sample));
timer->COUNT16.CTRLBSET.reg = TC_CTRLBSET_CMD_RETRIGGER;
while (timer->COUNT16.STATUS.bit.STOP == 1) {}
self->playing = true;
}
void common_hal_audioio_audioout_stop(audioio_audioout_obj_t* self) {
if (active_audioout == self) {
if (MP_STATE_VM(audiodma_block_counter) != NULL) {
tc_stop_counter(MP_STATE_VM(audiodma_block_counter));
}
tc_stop_counter(MP_STATE_VM(audioout_sample_timer));
dma_abort_job(&audio_dma);
active_audioout = NULL;
dac_disable(MP_STATE_VM(audioout_dac_instance));
}
Tc* timer = tc_insts[self->tc_index];
timer->COUNT16.CTRLBSET.reg = TC_CTRLBSET_CMD_STOP;
audio_dma_stop(&self->left_dma);
#ifdef SAMD51
audio_dma_stop(&self->right_dma);
#endif
// FIXME(tannewt): Do we want to disable? What if we're sharing with an AnalogOut on the 51?
// dac_disable(MP_STATE_VM(audioout_dac_instance));
}
bool common_hal_audioio_audioout_get_playing(audioio_audioout_obj_t* self) {
if (!dma_is_busy(&audio_dma)) {
if (active_audioout != NULL) {
common_hal_audioio_audioout_stop(active_audioout);
}
active_audioout = NULL;
bool now_playing = audio_dma_get_playing(&self->left_dma);
if (self->playing && !now_playing) {
common_hal_audioio_audioout_stop(self);
}
return active_audioout == self;
}
void common_hal_audioio_audioout_set_frequency(audioio_audioout_obj_t* self,
uint32_t frequency) {
if (frequency == 0 || frequency > 350000) {
mp_raise_ValueError("Unsupported playback frequency");
}
self->frequency = frequency;
if (common_hal_audioio_audioout_get_playing(self)) {
set_timer_frequency(frequency);
}
}
uint32_t common_hal_audioio_audioout_get_frequency(audioio_audioout_obj_t* self) {
return self->frequency;
return now_playing;
}

View File

@ -29,28 +29,21 @@
#include "common-hal/microcontroller/Pin.h"
#include "extmod/vfs_fat_file.h"
#include "audio_dma.h"
#include "py/obj.h"
typedef struct {
mp_obj_base_t base;
const mcu_pin_obj_t *pin;
uint32_t frequency;
uint8_t* buffer;
const mcu_pin_obj_t *left_channel;
audio_dma_t left_dma;
#ifdef SAMD51
const mcu_pin_obj_t *right_channel;
audio_dma_t right_dma;
#endif
uint8_t tc_index;
// File playback specific:
uint8_t* second_buffer;
DmacDescriptor* second_descriptor;
uint32_t file_length; // In bytes
uint16_t data_start; // Where the data values start
uint8_t bytes_per_sample;
bool signed_samples;
uint16_t last_loaded_block;
uint32_t bytes_remaining;
bool loop;
uint32_t len;
pyb_file_obj_t* file;
uint8_t tc_to_dac_event_channel;
bool playing;
} audioio_audioout_obj_t;
void audioout_reset(void);

View File

@ -48,7 +48,6 @@ static uint32_t tc_periods[TC_INST_NUM];
uint32_t target_tcc_frequencies[TCC_INST_NUM];
uint8_t tcc_refcount[TCC_INST_NUM];
const uint16_t prescaler[8] = {1, 2, 4, 8, 16, 64, 256, 1024};
// This bitmask keeps track of which channels of a TCC are currently claimed.
#ifdef SAMD21

194
ports/atmel-samd/events.c Normal file
View File

@ -0,0 +1,194 @@
/*
* This file is part of the MicroPython project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2018 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 "events.h"
#include "clocks.h"
#include "py/runtime.h"
#ifdef SAMD21
#include "hpl/pm/hpl_pm_base.h"
#endif
#ifdef SAMD51
#include "hri/hri_mclk_d51.h"
#endif
void turn_on_event_system(void) {
#ifdef SAMD51
hri_mclk_set_APBBMASK_EVSYS_bit(MCLK);
#endif
#ifdef SAMD21
_pm_enable_bus_clock(PM_BUS_APBC, EVSYS);
#endif
}
void reset_event_system(void) {
#ifdef SAMD51
EVSYS->CTRLA.bit.SWRST = true;
hri_mclk_clear_APBBMASK_EVSYS_bit(MCLK);
#endif
#ifdef SAMD21
EVSYS->CTRL.bit.SWRST = true;
_pm_disable_bus_clock(PM_BUS_APBC, EVSYS);
#endif
}
static bool channel_free(int8_t channel) {
uint8_t generator;
#ifdef SAMD51
generator = EVSYS->Channel[channel].CHANNEL.bit.EVGEN;
#endif
#ifdef SAMD21
// Explicitly do a byte write so the peripheral knows we're just wanting to read the channel
// rather than write to it.
*((uint8_t*) &EVSYS->CHANNEL.reg) = channel;
generator = (EVSYS->CHANNEL.reg & EVSYS_CHANNEL_EVGEN_Msk) >> EVSYS_CHANNEL_EVGEN_Pos;
#endif
return generator == 0;
}
uint8_t find_async_event_channel(void) {
int8_t channel;
for (channel = EVSYS_CHANNELS - 1; channel > 0; channel--) {
if (channel_free(channel)) {
break;
}
}
if (channel < 0) {
mp_raise_RuntimeError("All event channels in use");
}
return channel;
}
#ifdef SAMD21
#define EVSYS_SYNCH_NUM EVSYS_CHANNELS
#endif
uint8_t find_sync_event_channel(void) {
int8_t channel;
for (channel = 0; channel < EVSYS_SYNCH_NUM; channel++) {
if (channel_free(channel)) {
break;
}
}
if (channel >= EVSYS_SYNCH_NUM) {
mp_raise_RuntimeError("All sync event channels in use");
}
return channel;
}
void disable_event_channel(uint8_t channel_number) {
#ifdef SAMD51
EVSYS->Channel[channel_number].CHANNEL.reg = 0;
#endif
#ifdef SAMD21
EVSYS->CHANNEL.reg = EVSYS_CHANNEL_CHANNEL(channel_number);
#endif
}
void disable_event_user(uint8_t user_number) {
#ifdef SAMD51
EVSYS->USER[user_number].reg = 0;
#endif
#ifdef SAMD21
EVSYS->USER.reg = EVSYS_USER_USER(user_number);
#endif
}
void connect_event_user_to_channel(uint8_t user, uint8_t channel) {
#ifdef SAMD51
EVSYS->USER[user].reg = channel + 1;
#endif
#ifdef SAMD21
EVSYS->USER.reg = EVSYS_USER_USER(user) | EVSYS_USER_CHANNEL(channel + 1);
#endif
}
void init_async_event_channel(uint8_t channel, uint8_t generator) {
#ifdef SAMD51
EVSYS->Channel[channel].CHANNEL.reg = EVSYS_CHANNEL_EVGEN(generator) | EVSYS_CHANNEL_PATH_ASYNCHRONOUS;
#endif
#ifdef SAMD21
EVSYS->CHANNEL.reg = EVSYS_CHANNEL_CHANNEL(channel) | EVSYS_CHANNEL_EVGEN(generator) | EVSYS_CHANNEL_PATH_ASYNCHRONOUS;
#endif
}
void init_event_channel_interrupt(uint8_t channel, uint8_t gclk, uint8_t generator) {
connect_gclk_to_peripheral(gclk, EVSYS_GCLK_ID_0 + channel);
#ifdef SAMD51
EVSYS->Channel[channel].CHANNEL.reg = EVSYS_CHANNEL_EVGEN(generator) |
EVSYS_CHANNEL_PATH_SYNCHRONOUS |
EVSYS_CHANNEL_EDGSEL_RISING_EDGE;
#endif
#ifdef SAMD21
EVSYS->CHANNEL.reg = EVSYS_CHANNEL_CHANNEL(channel) |
EVSYS_CHANNEL_EVGEN(generator) |
EVSYS_CHANNEL_PATH_RESYNCHRONIZED |
EVSYS_CHANNEL_EDGSEL_RISING_EDGE;
if (channel > 7) {
uint8_t value = 1 << (channel - 7);
EVSYS->INTFLAG.reg = EVSYS_INTFLAG_EVDp8(value) | EVSYS_INTFLAG_OVRp8(value);
EVSYS->INTENSET.reg = EVSYS_INTENSET_EVDp8(value) | EVSYS_INTENSET_OVRp8(value);
} else {
uint8_t value = 1 << channel;
EVSYS->INTFLAG.reg = EVSYS_INTFLAG_EVD(value) | EVSYS_INTFLAG_OVR(value);
EVSYS->INTENSET.reg = EVSYS_INTENSET_EVD(value) | EVSYS_INTENSET_OVR(value);
}
#endif
}
bool event_interrupt_active(uint8_t channel) {
bool active = false;
#ifdef SAMD21
if (channel > 7) {
uint8_t value = 1 << (channel - 7);
active = (EVSYS->INTFLAG.reg & EVSYS_INTFLAG_EVDp8(value)) != 0;
// Only clear if we know its active, otherwise there is the possibility it becomes active
// after we check but before we clear.
if (active) {
EVSYS->INTFLAG.reg = EVSYS_INTFLAG_EVDp8(value) | EVSYS_INTFLAG_OVRp8(value);
}
} else {
uint8_t value = 1 << channel;
active = (EVSYS->INTFLAG.reg & EVSYS_INTFLAG_EVD(value)) != 0;
if (active) {
EVSYS->INTFLAG.reg = EVSYS_INTFLAG_EVD(value) | EVSYS_INTFLAG_OVR(value);
}
}
#endif
#ifdef SAMD51
active = EVSYS->Channel[channel].CHINTFLAG.bit.EVD;
// Only clear if we know its active, otherwise there is the possibility it becomes after we
// check but before we clear.
if (active) {
EVSYS->Channel[channel].CHINTFLAG.reg = EVSYS_CHINTFLAG_EVD;
}
#endif
return active;
}

46
ports/atmel-samd/events.h Normal file
View File

@ -0,0 +1,46 @@
/*
* This file is part of the MicroPython project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2018 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.
*/
#ifndef MICROPY_INCLUDED_ATMEL_SAMD_EVENTS_H
#define MICROPY_INCLUDED_ATMEL_SAMD_EVENTS_H
#include <stdbool.h>
#include <stdint.h>
#include "include/sam.h"
void turn_on_event_system(void);
void reset_event_system(void);
uint8_t find_async_event_channel(void);
uint8_t find_sync_event_channel(void);
void disable_event_channel(uint8_t channel_number);
void disable_event_user(uint8_t user_number);
void connect_event_user_to_channel(uint8_t user, uint8_t channel);
void init_async_event_channel(uint8_t channel, uint8_t generator);
void init_event_channel_interrupt(uint8_t channel, uint8_t gclk, uint8_t generator);
bool event_interrupt_active(uint8_t channel);
#endif // MICROPY_INCLUDED_ATMEL_SAMD_EVENTS_H

View File

@ -195,12 +195,19 @@ extern const struct _mp_obj_module_t usb_hid_module;
// Scan gamepad every 32ms
#define CIRCUITPY_GAMEPAD_TICKS 0x1f
#if defined(__SAMD51G19A__) || defined(__SAMD51G18A__)
#define AUDIOBUSIO_MODULE
#else
#define AUDIOBUSIO_MODULE { MP_OBJ_NEW_QSTR(MP_QSTR_audiobusio), (mp_obj_t)&audiobusio_module },
#endif
#define EXTRA_BUILTIN_MODULES \
{ MP_OBJ_NEW_QSTR(MP_QSTR_audioio), (mp_obj_t)&audioio_module }, \
AUDIOBUSIO_MODULE \
{ MP_OBJ_NEW_QSTR(MP_QSTR_bitbangio), (mp_obj_t)&bitbangio_module }, \
{ MP_OBJ_NEW_QSTR(MP_QSTR_gamepad),(mp_obj_t)&gamepad_module }
// { MP_OBJ_NEW_QSTR(MP_QSTR_audioio), (mp_obj_t)&audioio_module },
// { MP_OBJ_NEW_QSTR(MP_QSTR_audiobusio), (mp_obj_t)&audiobusio_module },
#define EXPRESS_BOARD
#else
#define MICROPY_PY_BUILTINS_REVERSED (0)
#define MICROPY_PY_MICROPYTHON_MEM_INFO (0)
@ -253,15 +260,12 @@ extern const struct _mp_obj_module_t usb_hid_module;
#define MP_STATE_PORT MP_STATE_VM
#include "shared_dma.h"
#define MICROPY_PORT_ROOT_POINTERS \
const char *readline_hist[8]; \
vstr_t *repl_line; \
struct tc_module* audiodma_block_counter; \
struct events_resource* audiodma_block_event; \
struct tc_module* audioout_sample_timer; \
struct dac_module* audioout_dac_instance; \
struct events_resource* audioout_sample_event; \
struct events_resource* audioout_dac_event; \
mp_obj_t playing_audio[AUDIO_DMA_CHANNEL_COUNT]; \
FLASH_ROOT_POINTERS \
void run_background_tasks(void);

View File

@ -34,18 +34,10 @@
#include "shared-bindings/microcontroller/__init__.h"
// We allocate three DMA resources for the entire lifecycle of the board (not the
// vm) because the general_dma resource will be shared between the REPL and SPI
// flash. Both uses must block each other in order to prevent conflict.
COMPILER_ALIGNED(16) static DmacDescriptor dma_descriptors[3];
COMPILER_ALIGNED(16) static DmacDescriptor dma_descriptors[DMA_CHANNEL_COUNT];
// Don't use these directly. They are used by the DMA engine itself.
COMPILER_ALIGNED(16) static DmacDescriptor write_back_descriptors[3];
#define AUDIO_DMA_CHANNEL 0
#define SHARED_TX_CHANNEL 1
#define SHARED_RX_CHANNEL 2
COMPILER_ALIGNED(16) static DmacDescriptor write_back_descriptors[DMA_CHANNEL_COUNT];
#ifdef SAMD21
#define FIRST_SERCOM_RX_TRIGSRC 0x01
@ -56,51 +48,6 @@ COMPILER_ALIGNED(16) static DmacDescriptor write_back_descriptors[3];
#define FIRST_SERCOM_TX_TRIGSRC 0x05
#endif
// static void dma_configure_audio(uint8_t channel) {
// system_interrupt_enter_critical_section();
// /** Select the DMA channel and clear software trigger */
// DMAC->CHID.reg = DMAC_CHID_ID(channel);
// DMAC->CHCTRLA.reg &= ~DMAC_CHCTRLA_ENABLE;
// DMAC->CHCTRLA.reg = DMAC_CHCTRLA_SWRST;
// DMAC->SWTRIGCTRL.reg &= (uint32_t)(~(1 << channel));
// uint32_t event_output_enable = 0;
// if (output_event) {
// event_output_enable = DMAC_CHCTRLB_EVOE;
// }
// DMAC->CHCTRLB.reg = DMAC_CHCTRLB_LVL(DMA_PRIORITY_LEVEL_0) |
// DMAC_CHCTRLB_TRIGSRC(trigsrc) |
// DMAC_CHCTRLB_TRIGACT(DMA_TRIGGER_ACTION_BEAT) |
// event_output_enable;
// // config.peripheral_trigger = DAC_DMAC_ID_EMPTY;
// // config.trigger_action = DMA_TRIGGER_ACTION_BEAT;
// // config.event_config.input_action = DMA_EVENT_INPUT_TRIG;
// // config.event_config.event_output_enable = true;
// system_interrupt_leave_critical_section();
// }
void init_shared_dma(void) {
// Turn on the clocks
#ifdef SAMD51
MCLK->AHBMASK.reg |= MCLK_AHBMASK_DMAC;
#endif
#ifdef SAMD21
PM->AHBMASK.reg |= PM_AHBMASK_DMAC;
PM->APBBMASK.reg |= PM_APBBMASK_DMAC;
#endif
DMAC->CTRL.reg = DMAC_CTRL_SWRST;
DMAC->BASEADDR.reg = (uint32_t) dma_descriptors;
DMAC->WRBADDR.reg = (uint32_t) write_back_descriptors;
DMAC->CTRL.reg = DMAC_CTRL_DMAENABLE | DMAC_CTRL_LVLEN0;
// This allocates the lowest channel first so make sure the audio is first
// so it gets the highest priority.
// dma_configure_audio(0);
}
static uint8_t sercom_index(Sercom* sercom) {
#ifdef SAMD21
return ((uint32_t) sercom - (uint32_t) SERCOM0) / 0x400;
@ -115,7 +62,7 @@ static uint8_t sercom_index(Sercom* sercom) {
#endif
}
static void dma_configure(uint8_t channel_number, uint8_t trigsrc, bool output_event) {
void dma_configure(uint8_t channel_number, uint8_t trigsrc, bool output_event) {
#ifdef SAMD21
common_hal_mcu_disable_interrupts();
/** Select the DMA channel and clear software trigger */
@ -147,7 +94,7 @@ static void dma_configure(uint8_t channel_number, uint8_t trigsrc, bool output_e
#endif
}
static void enable_channel(uint8_t channel_number) {
void dma_enable_channel(uint8_t channel_number) {
#ifdef SAMD21
common_hal_mcu_disable_interrupts();
/** Select the DMA channel and clear software trigger */
@ -162,7 +109,38 @@ static void enable_channel(uint8_t channel_number) {
#endif
}
static uint8_t transfer_status(uint8_t channel_number) {
void dma_disable_channel(uint8_t channel_number) {
#ifdef SAMD21
common_hal_mcu_disable_interrupts();
/** Select the DMA channel and clear software trigger */
DMAC->CHID.reg = DMAC_CHID_ID(channel_number);
DMAC->CHCTRLA.bit.ENABLE = false;
common_hal_mcu_enable_interrupts();
#endif
#ifdef SAMD51
DmacChannel* channel = &DMAC->Channel[channel_number];
channel->CHCTRLA.bit.ENABLE = false;
#endif
}
bool dma_channel_enabled(uint8_t channel_number) {
#ifdef SAMD21
common_hal_mcu_disable_interrupts();
/** Select the DMA channel and clear software trigger */
DMAC->CHID.reg = DMAC_CHID_ID(channel_number);
bool enabled = DMAC->CHCTRLA.bit.ENABLE;
common_hal_mcu_enable_interrupts();
return enabled;
#endif
#ifdef SAMD51
DmacChannel* channel = &DMAC->Channel[channel_number];
return channel->CHCTRLA.bit.ENABLE;
#endif
}
uint8_t dma_transfer_status(uint8_t channel_number) {
#ifdef SAMD21
common_hal_mcu_disable_interrupts();
/** Select the DMA channel and clear software trigger */
@ -181,7 +159,6 @@ static uint8_t transfer_status(uint8_t channel_number) {
static bool channel_free(uint8_t channel_number) {
#ifdef SAMD21
common_hal_mcu_disable_interrupts();
/** Select the DMA channel and clear software trigger */
DMAC->CHID.reg = DMAC_CHID_ID(channel_number);
bool channel_free = DMAC->CHSTATUS.reg == 0;
common_hal_mcu_enable_interrupts();
@ -194,6 +171,29 @@ static bool channel_free(uint8_t channel_number) {
#endif
}
void init_shared_dma(void) {
// Turn on the clocks
#ifdef SAMD51
MCLK->AHBMASK.reg |= MCLK_AHBMASK_DMAC;
#endif
#ifdef SAMD21
PM->AHBMASK.reg |= PM_AHBMASK_DMAC;
PM->APBBMASK.reg |= PM_APBBMASK_DMAC;
#endif
DMAC->CTRL.reg = DMAC_CTRL_SWRST;
DMAC->BASEADDR.reg = (uint32_t) dma_descriptors;
DMAC->WRBADDR.reg = (uint32_t) write_back_descriptors;
DMAC->CTRL.reg = DMAC_CTRL_DMAENABLE | DMAC_CTRL_LVLEN0;
for (uint8_t i = 0; i < AUDIO_DMA_CHANNEL_COUNT; i++) {
dma_configure(i, 0, true);
}
}
// Do write and read simultaneously. If buffer_out is NULL, write the tx byte over and over.
// If buffer_out is a real buffer, ignore tx.
// DMAs buffer_out -> dest
@ -284,10 +284,10 @@ static int32_t shared_dma_transfer(void* peripheral,
// Start the RX job first so we don't miss the first byte. The TX job clocks
// the output.
if (rx_active) {
enable_channel(SHARED_RX_CHANNEL);
dma_enable_channel(SHARED_RX_CHANNEL);
}
if (tx_active) {
enable_channel(SHARED_TX_CHANNEL);
dma_enable_channel(SHARED_TX_CHANNEL);
}
@ -309,10 +309,10 @@ static int32_t shared_dma_transfer(void* peripheral,
// Channels cycle between Suspend -> Pending -> Busy and back while transfering. So, we check
// the channels transfer status for an error or completion.
if (rx_active) {
while ((transfer_status(SHARED_RX_CHANNEL) & 0x3) == 0) {}
while ((dma_transfer_status(SHARED_RX_CHANNEL) & 0x3) == 0) {}
}
if (tx_active) {
while ((transfer_status(SHARED_TX_CHANNEL) & 0x3) == 0) {}
while ((dma_transfer_status(SHARED_TX_CHANNEL) & 0x3) == 0) {}
}
if (sercom) {
@ -331,8 +331,8 @@ static int32_t shared_dma_transfer(void* peripheral,
}
}
if ((!rx_active || transfer_status(SHARED_RX_CHANNEL) == DMAC_CHINTFLAG_TCMPL) &&
(!tx_active || transfer_status(SHARED_TX_CHANNEL) == DMAC_CHINTFLAG_TCMPL)) {
if ((!rx_active || dma_transfer_status(SHARED_RX_CHANNEL) == DMAC_CHINTFLAG_TCMPL) &&
(!tx_active || dma_transfer_status(SHARED_TX_CHANNEL) == DMAC_CHINTFLAG_TCMPL)) {
return length;
}
return -2;
@ -362,75 +362,6 @@ int32_t qspi_dma_read(uint32_t address, uint8_t* buffer, uint32_t length) {
}
#endif
bool allocate_block_counter() {
// // Find a timer to count DMA block completions.
// Tc *t = NULL;
// Tc *tcs[TC_INST_NUM] = TC_INSTS;
// for (uint8_t i = TC_INST_NUM; i > 0; i--) {
// if (tcs[i - 1]->COUNT16.CTRLA.bit.ENABLE == 0) {
// t = tcs[i - 1];
// break;
// }
// }
// if (t == NULL) {
// return false;
// }
// MP_STATE_VM(audiodma_block_counter) = gc_alloc(sizeof(struct tc_module), false);
// if (MP_STATE_VM(audiodma_block_counter) == NULL) {
// return false;
// }
//
// // Don't bother setting the period. We set it before you playback anything.
// struct tc_config config_tc;
// tc_get_config_defaults(&config_tc);
// config_tc.counter_size = TC_COUNTER_SIZE_16BIT;
// config_tc.clock_prescaler = TC_CLOCK_PRESCALER_DIV1;
// if (tc_init(MP_STATE_VM(audiodma_block_counter), t, &config_tc) != STATUS_OK) {
// return false;
// };
//
// struct tc_events events_tc;
// events_tc.generate_event_on_overflow = false;
// events_tc.on_event_perform_action = true;
// events_tc.event_action = TC_EVENT_ACTION_INCREMENT_COUNTER;
// tc_enable_events(MP_STATE_VM(audiodma_block_counter), &events_tc);
//
// // Connect the timer overflow event, which happens at the target frequency,
// // to the DAC conversion trigger.
// MP_STATE_VM(audiodma_block_event) = gc_alloc(sizeof(struct events_resource), false);
// if (MP_STATE_VM(audiodma_block_event) == NULL) {
// return false;
// }
// struct events_config config;
// events_get_config_defaults(&config);
//
// uint8_t user = EVSYS_ID_USER_TC3_EVU;
// if (t == TC4) {
// user = EVSYS_ID_USER_TC4_EVU;
// } else if (t == TC5) {
// user = EVSYS_ID_USER_TC5_EVU;
// #ifdef TC6
// } else if (t == TC6) {
// user = EVSYS_ID_USER_TC6_EVU;
// #endif
// #ifdef TC7
// } else if (t == TC7) {
// user = EVSYS_ID_USER_TC7_EVU;
// #endif
// }
//
// config.generator = EVSYS_ID_GEN_DMAC_CH_0;
// config.path = EVENTS_PATH_ASYNCHRONOUS;
// if (events_allocate(MP_STATE_VM(audiodma_block_event), &config) != STATUS_OK ||
// events_attach_user(MP_STATE_VM(audiodma_block_event), user) != STATUS_OK) {
// return false;
// }
//
// tc_enable(MP_STATE_VM(audiodma_block_counter));
// tc_stop_counter(MP_STATE_VM(audiodma_block_counter));
return true;
}
void switch_audiodma_trigger(uint8_t trigger_dmac_id) {
//dma_configure(audio_dma.channel_id, trigger_dmac_id, true);
DmacDescriptor* dma_descriptor(uint8_t channel_number) {
return &dma_descriptors[channel_number];
}

View File

@ -32,6 +32,14 @@
#include "include/sam.h"
// We allocate DMA resources for the entire lifecycle of the board (not the
// vm) because the general_dma resource will be shared between the REPL and SPI
// flash. Both uses must block each other in order to prevent conflict.
#define AUDIO_DMA_CHANNEL_COUNT 3
#define DMA_CHANNEL_COUNT (AUDIO_DMA_CHANNEL_COUNT + 2)
#define SHARED_TX_CHANNEL (DMA_CHANNEL_COUNT - 2)
#define SHARED_RX_CHANNEL (DMA_CHANNEL_COUNT - 1)
volatile bool audio_dma_in_use;
void init_shared_dma(void);
@ -45,8 +53,11 @@ int32_t sercom_dma_write(Sercom* sercom, const uint8_t* buffer, uint32_t length)
int32_t sercom_dma_read(Sercom* sercom, uint8_t* buffer, uint32_t length, uint8_t tx);
int32_t sercom_dma_transfer(Sercom* sercom, const uint8_t* buffer_out, uint8_t* buffer_in, uint32_t length);
// Allocate a counter to track how far along we are in a DMA double buffer.
bool allocate_block_counter(void);
void switch_audiodma_trigger(uint8_t trigger_dmac_id);
void dma_configure(uint8_t channel_number, uint8_t trigsrc, bool output_event);
void dma_enable_channel(uint8_t channel_number);
void dma_disable_channel(uint8_t channel_number);
bool dma_channel_enabled(uint8_t channel_number);
uint8_t dma_transfer_status(uint8_t channel_number);
DmacDescriptor* dma_descriptor(uint8_t channel_number);
#endif // MICROPY_INCLUDED_ATMEL_SAMD_SHARED_DMA_H

View File

@ -45,11 +45,16 @@
#include "common-hal/analogio/AnalogIn.h"
#include "common-hal/analogio/AnalogOut.h"
#include "common-hal/audiobusio/PDMIn.h"
#include "common-hal/audiobusio/I2SOut.h"
#include "common-hal/audioio/AudioOut.h"
#include "common-hal/microcontroller/Pin.h"
#include "common-hal/pulseio/PulseIn.h"
#include "common-hal/pulseio/PulseOut.h"
#include "common-hal/pulseio/PWMOut.h"
#include "common-hal/usb_hid/Device.h"
#include "clocks.h"
#include "events.h"
#include "shared_dma.h"
#include "tick.h"
@ -192,15 +197,21 @@ void reset_port(void) {
sercom_instances[i]->SPI.CTRLA.bit.SWRST = 1;
}
// #ifdef EXPRESS_BOARD
// audioout_reset();
#ifdef EXPRESS_BOARD
audioout_reset();
#if !defined(__SAMD51G19A__) && !defined(__SAMD51G18A__)
i2sout_reset();
#endif
audio_dma_reset();
// touchin_reset();
// pdmin_reset();
// #endif
//pdmin_reset();
#endif
pulsein_reset();
pulseout_reset();
pwmout_reset();
reset_gclks();
analogin_reset();
#ifdef CIRCUITPY_GAMEPAD_TICKS
@ -209,6 +220,8 @@ void reset_port(void) {
analogout_reset();
reset_event_system();
reset_all_pins();
// Set up debugging pins after reset_all_pins().

View File

@ -39,6 +39,8 @@
#include "hri/hri_gclk_d51.h"
#endif
const uint16_t prescaler[8] = {1, 2, 4, 8, 16, 64, 256, 1024};
// This bitmask keeps track of which channels of a TCC are currently claimed.
#ifdef SAMD21
const uint8_t tcc_cc_num[3] = {4, 2, 2};
@ -73,8 +75,8 @@ const uint8_t tc_gclk_ids[TC_INST_NUM] = {TC0_GCLK_ID,
TC7_GCLK_ID,
#endif
};
const uint8_t tcc_gclk_ids[TCC_INST_NUM] = {TCC0_GCLK_ID,
TCC1_GCLK_ID,
const uint8_t tcc_gclk_ids[TCC_INST_NUM] = {TCC0_GCLK_ID,
TCC1_GCLK_ID,
TCC2_GCLK_ID,
#ifdef TCC3_GCLK_ID
TCC3_GCLK_ID,

View File

@ -28,6 +28,8 @@
#include "include/sam.h"
const uint16_t prescaler[8];
#ifdef SAMD21
const uint8_t tcc_cc_num[3];
const uint8_t tc_gclk_ids[TC_INST_NUM];

View File

@ -0,0 +1,233 @@
/*
* This file is part of the Micro Python project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2017 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 <stdint.h>
#include "lib/utils/context_manager_helpers.h"
#include "py/binary.h"
#include "py/objproperty.h"
#include "py/runtime.h"
#include "shared-bindings/microcontroller/Pin.h"
#include "shared-bindings/audiobusio/I2SOut.h"
#include "shared-bindings/util.h"
//| .. currentmodule:: audiobusio
//|
//| :class:`I2SOut` -- Output an I2S audio signal
//| ========================================================
//|
//| I2S is used to output an audio signal on an I2S bus.
//|
//| .. class:: I2SOut(bit_clock, word_select, data, *, left_justified)
//|
//| Create a I2SOut object associated with the given pins.
//|
//| :param ~microcontroller.Pin bit_clock: The bit clock (or serial clock) pin
//| :param ~microcontroller.Pin word_select: The word select (or left/right clock) pin
//| :param ~microcontroller.Pin data: The data pin
//| :param bool left_justified: True when data bits are aligned with the word select clock. False
//| when they are shifted by one to match classic I2S protocol.
//|
//| Simple 8ksps 440 Hz sine wave on `Metro M0 Express <https://www.adafruit.com/product/3505>`_
//| using `UDA1334 Breakout <https://www.adafruit.com/product/3678>`_::
//|
//| import audiobusio
//| import audioio
//| import board
//| import array
//| import time
//| import math
//|
//| # Generate one period of sine wav.
//| length = 8000 // 440
//| sine_wave = array.array("H", [0] * length)
//| for i in range(length):
//| sine_wave[i] = int(math.sin(math.pi * 2 * i / 18) * (2 ** 15) + 2 ** 15)
//|
//| sine_wave = audiobusio.RawSample(sine_wave, sample_rate=8000)
//| i2s = audiobusio.I2SOut(board.D1, board.D0, board.D9)
//| i2s.play(sine_wave, loop=True)
//| time.sleep(1)
//| i2s.stop()
//|
//| Playing a wave file from flash::
//|
//| import board
//| import audioio
//| import audiobusio
//| import digitalio
//|
//|
//| f = open("cplay-5.1-16bit-16khz.wav", "rb")
//| wav = audioio.WaveFile(f)
//|
//| a = audiobusio.I2SOut(board.D1, board.D0, board.D9)
//|
//| print("playing")
//| a.play(wav)
//| while a.playing:
//| pass
//| print("stopped")
//|
STATIC mp_obj_t audiobusio_i2sout_make_new(const mp_obj_type_t *type, size_t n_args, size_t n_kw, const mp_obj_t *pos_args) {
mp_arg_check_num(n_args, n_kw, 3, 4, true);
mp_map_t kw_args;
mp_map_init_fixed_table(&kw_args, n_kw, pos_args + n_args);
enum { ARG_bit_clock, ARG_word_select, ARG_data, ARG_left_justified };
static const mp_arg_t allowed_args[] = {
{ MP_QSTR_bit_clock, MP_ARG_OBJ | MP_ARG_REQUIRED },
{ MP_QSTR_word_select, MP_ARG_OBJ | MP_ARG_REQUIRED },
{ MP_QSTR_data, MP_ARG_OBJ | MP_ARG_REQUIRED },
{ MP_QSTR_left_justified, MP_ARG_OBJ | MP_ARG_KW_ONLY, {.u_bool = false} },
};
mp_arg_val_t args[MP_ARRAY_SIZE(allowed_args)];
mp_arg_parse_all(n_args, pos_args, &kw_args, MP_ARRAY_SIZE(allowed_args), allowed_args, args);
mp_obj_t bit_clock_obj = args[ARG_bit_clock].u_obj;
assert_pin(bit_clock_obj, false);
const mcu_pin_obj_t *bit_clock = MP_OBJ_TO_PTR(bit_clock_obj);
mp_obj_t word_select_obj = args[ARG_word_select].u_obj;
assert_pin(word_select_obj, false);
const mcu_pin_obj_t *word_select = MP_OBJ_TO_PTR(word_select_obj);
mp_obj_t data_obj = args[ARG_data].u_obj;
assert_pin(data_obj, false);
const mcu_pin_obj_t *data = MP_OBJ_TO_PTR(data_obj);
audiobusio_i2sout_obj_t *self = m_new_obj(audiobusio_i2sout_obj_t);
self->base.type = &audiobusio_i2sout_type;
common_hal_audiobusio_i2sout_construct(self, bit_clock, word_select, data, args[ARG_left_justified].u_bool);
return MP_OBJ_FROM_PTR(self);
}
//| .. method:: deinit()
//|
//| Deinitialises the I2SOut and releases any hardware resources for reuse.
//|
STATIC mp_obj_t audiobusio_i2sout_deinit(mp_obj_t self_in) {
audiobusio_i2sout_obj_t *self = MP_OBJ_TO_PTR(self_in);
common_hal_audiobusio_i2sout_deinit(self);
return mp_const_none;
}
STATIC MP_DEFINE_CONST_FUN_OBJ_1(audiobusio_i2sout_deinit_obj, audiobusio_i2sout_deinit);
//| .. method:: __enter__()
//|
//| No-op used by Context Managers.
//|
// Provided by context manager helper.
//| .. method:: __exit__()
//|
//| Automatically deinitializes the hardware when exiting a context. See
//| :ref:`lifetime-and-contextmanagers` for more info.
//|
STATIC mp_obj_t audiobusio_i2sout_obj___exit__(size_t n_args, const mp_obj_t *args) {
(void)n_args;
common_hal_audiobusio_i2sout_deinit(args[0]);
return mp_const_none;
}
STATIC MP_DEFINE_CONST_FUN_OBJ_VAR_BETWEEN(audiobusio_i2sout___exit___obj, 4, 4, audiobusio_i2sout_obj___exit__);
//| .. method:: play(sample, *, loop=False)
//|
//| Plays the sample once when loop=False and continuously when loop=True.
//| Does not block. Use `playing` to block.
//|
//| Sample must be an `audioio.WaveFile` or `audioio.RawSample`.
//|
//| The sample itself should consist of 8 bit or 16 bit samples.
//|
STATIC mp_obj_t audiobusio_i2sout_obj_play(size_t n_args, const mp_obj_t *pos_args, mp_map_t *kw_args) {
enum { ARG_sample, ARG_loop };
static const mp_arg_t allowed_args[] = {
{ MP_QSTR_sample, MP_ARG_OBJ | MP_ARG_REQUIRED },
{ MP_QSTR_loop, MP_ARG_BOOL | MP_ARG_KW_ONLY, {.u_bool = false} },
};
audiobusio_i2sout_obj_t *self = MP_OBJ_TO_PTR(pos_args[0]);
raise_error_if_deinited(common_hal_audiobusio_i2sout_deinited(self));
mp_arg_val_t args[MP_ARRAY_SIZE(allowed_args)];
mp_arg_parse_all(n_args - 1, pos_args + 1, kw_args, MP_ARRAY_SIZE(allowed_args), allowed_args, args);
mp_obj_t sample = args[ARG_sample].u_obj;
common_hal_audiobusio_i2sout_play(self, sample, args[ARG_loop].u_bool);
return mp_const_none;
}
MP_DEFINE_CONST_FUN_OBJ_KW(audiobusio_i2sout_play_obj, 1, audiobusio_i2sout_obj_play);
//| .. method:: stop()
//|
//| Stops playback.
//|
STATIC mp_obj_t audiobusio_i2sout_obj_stop(mp_obj_t self_in) {
audiobusio_i2sout_obj_t *self = MP_OBJ_TO_PTR(self_in);
raise_error_if_deinited(common_hal_audiobusio_i2sout_deinited(self));
common_hal_audiobusio_i2sout_stop(self);
return mp_const_none;
}
MP_DEFINE_CONST_FUN_OBJ_1(audiobusio_i2sout_stop_obj, audiobusio_i2sout_obj_stop);
//| .. attribute:: playing
//|
//| True when the audio sample is being output. (read-only)
//|
STATIC mp_obj_t audiobusio_i2sout_obj_get_playing(mp_obj_t self_in) {
audiobusio_i2sout_obj_t *self = MP_OBJ_TO_PTR(self_in);
raise_error_if_deinited(common_hal_audiobusio_i2sout_deinited(self));
return mp_obj_new_bool(common_hal_audiobusio_i2sout_get_playing(self));
}
MP_DEFINE_CONST_FUN_OBJ_1(audiobusio_i2sout_get_playing_obj, audiobusio_i2sout_obj_get_playing);
const mp_obj_property_t audiobusio_i2sout_playing_obj = {
.base.type = &mp_type_property,
.proxy = {(mp_obj_t)&audiobusio_i2sout_get_playing_obj,
(mp_obj_t)&mp_const_none_obj,
(mp_obj_t)&mp_const_none_obj},
};
STATIC const mp_rom_map_elem_t audiobusio_i2sout_locals_dict_table[] = {
// Methods
{ MP_ROM_QSTR(MP_QSTR_deinit), MP_ROM_PTR(&audiobusio_i2sout_deinit_obj) },
{ MP_ROM_QSTR(MP_QSTR___enter__), MP_ROM_PTR(&default___enter___obj) },
{ MP_ROM_QSTR(MP_QSTR___exit__), MP_ROM_PTR(&audiobusio_i2sout___exit___obj) },
{ MP_ROM_QSTR(MP_QSTR_play), MP_ROM_PTR(&audiobusio_i2sout_play_obj) },
{ MP_ROM_QSTR(MP_QSTR_stop), MP_ROM_PTR(&audiobusio_i2sout_stop_obj) },
// Properties
{ MP_ROM_QSTR(MP_QSTR_playing), MP_ROM_PTR(&audiobusio_i2sout_playing_obj) },
};
STATIC MP_DEFINE_CONST_DICT(audiobusio_i2sout_locals_dict, audiobusio_i2sout_locals_dict_table);
const mp_obj_type_t audiobusio_i2sout_type = {
{ &mp_type_type },
.name = MP_QSTR_I2SOut,
.make_new = audiobusio_i2sout_make_new,
.locals_dict = (mp_obj_dict_t*)&audiobusio_i2sout_locals_dict,
};

View File

@ -0,0 +1,45 @@
/*
* This file is part of the Micro Python project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2017, 2018 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.
*/
#ifndef MICROPY_INCLUDED_SHARED_BINDINGS_AUDIOBUSIO_I2SOUT_H
#define MICROPY_INCLUDED_SHARED_BINDINGS_AUDIOBUSIO_I2SOUT_H
#include "common-hal/audiobusio/I2SOut.h"
#include "common-hal/microcontroller/Pin.h"
extern const mp_obj_type_t audiobusio_i2sout_type;
void common_hal_audiobusio_i2sout_construct(audiobusio_i2sout_obj_t* self,
const mcu_pin_obj_t* bit_clock, const mcu_pin_obj_t* word_select, const mcu_pin_obj_t* data,
bool left_justified);
void common_hal_audiobusio_i2sout_deinit(audiobusio_i2sout_obj_t* self);
bool common_hal_audiobusio_i2sout_deinited(audiobusio_i2sout_obj_t* self);
void common_hal_audiobusio_i2sout_play(audiobusio_i2sout_obj_t* self, mp_obj_t sample, bool loop);
void common_hal_audiobusio_i2sout_stop(audiobusio_i2sout_obj_t* self);
bool common_hal_audiobusio_i2sout_get_playing(audiobusio_i2sout_obj_t* self);
#endif // MICROPY_INCLUDED_SHARED_BINDINGS_AUDIOBUSIO_I2SOUT_H

View File

@ -31,6 +31,7 @@
#include "shared-bindings/microcontroller/Pin.h"
#include "shared-bindings/audiobusio/__init__.h"
#include "shared-bindings/audiobusio/I2SOut.h"
#include "shared-bindings/audiobusio/PDMIn.h"
//| :mod:`audiobusio` --- Support for audio input and output over digital bus
@ -50,6 +51,7 @@
//| .. toctree::
//| :maxdepth: 3
//|
//| I2SOut
//| PDMIn
//|
//| All libraries change hardware state and should be deinitialized when they
@ -59,7 +61,8 @@
STATIC const mp_rom_map_elem_t audiobusio_module_globals_table[] = {
{ MP_ROM_QSTR(MP_QSTR___name__), MP_ROM_QSTR(MP_QSTR_audiobusio) },
{ MP_ROM_QSTR(MP_QSTR_PDMIn), MP_ROM_PTR(&audiobusio_pdmin_type) },
{ MP_ROM_QSTR(MP_QSTR_I2SOut), MP_ROM_PTR(&audiobusio_i2sout_type) },
//{ MP_ROM_QSTR(MP_QSTR_PDMIn), MP_ROM_PTR(&audiobusio_pdmin_type) },
};
STATIC MP_DEFINE_CONST_DICT(audiobusio_module_globals, audiobusio_module_globals_table);

View File

@ -32,6 +32,7 @@
#include "py/runtime.h"
#include "shared-bindings/microcontroller/Pin.h"
#include "shared-bindings/audioio/AudioOut.h"
#include "shared-bindings/audioio/RawSample.h"
#include "shared-bindings/util.h"
//| .. currentmodule:: audioio
@ -41,18 +42,13 @@
//|
//| AudioOut can be used to output an analog audio signal on a given pin.
//|
//| .. class:: AudioOut(pin, sample_source)
//| .. class:: AudioOut(left_channel, right_channel=None)
//|
//| Create a AudioOut object associated with the given pin. This allows you to
//| play audio signals out on the given pin. Sample_source must be a `bytes-like object <https://docs.python.org/3/glossary.html#term-bytes-like-object>`_.
//| Create a AudioOut object associated with the given pin(s). This allows you to
//| play audio signals out on the given pin(s).
//|
//| The sample itself should consist of 16 bit samples and be mono.
//| Microcontrollers with a lower output resolution will use the highest order
//| bits to output. For example, the SAMD21 has a 10 bit DAC that ignores the
//| lowest 6 bits when playing 16 bit samples.
//|
//| :param ~microcontroller.Pin pin: The pin to output to
//| :param bytes-like sample_source: The source of the sample
//| :param ~microcontroller.Pin left_channel: The pin to output the left channel to
//| :param ~microcontroller.Pin right_channel: The pin to output the right channel to
//|
//| Simple 8ksps 440 Hz sin wave::
//|
@ -68,8 +64,9 @@
//| for i in range(length):
//| sine_wave[i] = int(math.sin(math.pi * 2 * i / 18) * (2 ** 15) + 2 ** 15)
//|
//| sample = audioio.AudioOut(board.SPEAKER, sine_wave)
//| sample.play(loop=True)
//| dac = audioio.AudioOut(board.SPEAKER)
//| sine_wave = audioio.RawSample(sine_wave, mono=True, sample_rate=8000)
//| dac.play(sine_wave, loop=True)
//| time.sleep(1)
//| sample.stop()
//|
@ -83,41 +80,42 @@
//| speaker_enable = digitalio.DigitalInOut(board.SPEAKER_ENABLE)
//| speaker_enable.switch_to_output(value=True)
//|
//| f = open("cplay-5.1-16bit-16khz.wav", "rb")
//
//| a = audioio.AudioOut(board.A0, f)
//| wav = audioio.WaveFile("cplay-5.1-16bit-16khz.wav")
//| a = audioio.AudioOut(board.A0)
//|
//| print("playing")
//| a.play()
//| a.play(wav)
//| while a.playing:
//| pass
//| print("stopped")
//|
STATIC mp_obj_t audioio_audioout_make_new(const mp_obj_type_t *type, size_t n_args, size_t n_kw, const mp_obj_t *args) {
mp_arg_check_num(n_args, n_kw, 2, 2, true);
mp_obj_t pin_obj = args[0];
assert_pin(pin_obj, false);
const mcu_pin_obj_t *pin = MP_OBJ_TO_PTR(pin_obj);
// We explicitly don't check whether the pin is free because multiple
// AudioOuts may share it.
STATIC mp_obj_t audioio_audioout_make_new(const mp_obj_type_t *type, size_t n_args, size_t n_kw, const mp_obj_t *pos_args) {
mp_arg_check_num(n_args, n_kw, 1, 2, true);
mp_map_t kw_args;
mp_map_init_fixed_table(&kw_args, n_kw, pos_args + n_args);
enum { ARG_left_channel, ARG_right_channel };
static const mp_arg_t allowed_args[] = {
{ MP_QSTR_left_channel, MP_ARG_OBJ | MP_ARG_REQUIRED },
{ MP_QSTR_right_channel, MP_ARG_OBJ | MP_ARG_KW_ONLY, {.u_rom_obj = mp_const_none} },
};
mp_arg_val_t args[MP_ARRAY_SIZE(allowed_args)];
mp_arg_parse_all(n_args, pos_args, &kw_args, MP_ARRAY_SIZE(allowed_args), allowed_args, args);
mp_obj_t left_channel_obj = args[ARG_left_channel].u_obj;
assert_pin(left_channel_obj, false);
const mcu_pin_obj_t *left_channel_pin = MP_OBJ_TO_PTR(left_channel_obj);
mp_obj_t right_channel_obj = args[ARG_right_channel].u_obj;
const mcu_pin_obj_t *right_channel_pin = NULL;
if (right_channel_obj != mp_const_none) {
assert_pin(right_channel_obj, false);
right_channel_pin = MP_OBJ_TO_PTR(right_channel_obj);
}
// create AudioOut object from the given pin
audioio_audioout_obj_t *self = m_new_obj(audioio_audioout_obj_t);
self->base.type = &audioio_audioout_type;
mp_buffer_info_t bufinfo;
if (MP_OBJ_IS_TYPE(args[1], &fatfs_type_fileio)) {
common_hal_audioio_audioout_construct_from_file(self, pin, MP_OBJ_TO_PTR(args[1]));
} else if (mp_get_buffer(args[1], &bufinfo, MP_BUFFER_READ)) {
uint8_t bytes_per_sample = 1;
if (bufinfo.typecode == 'H') {
bytes_per_sample = 2;
} else if (bufinfo.typecode != 'B' && bufinfo.typecode != BYTEARRAY_TYPECODE) {
mp_raise_ValueError("sample_source buffer must be a bytearray or array of type 'H' or 'B'");
}
common_hal_audioio_audioout_construct_from_buffer(self, pin, ((uint16_t*)bufinfo.buf), bufinfo.len, bytes_per_sample);
} else {
mp_raise_TypeError("sample_source must be a file or bytes-like object");
}
common_hal_audioio_audioout_construct(self, left_channel_pin, right_channel_pin);
return MP_OBJ_FROM_PTR(self);
}
@ -152,30 +150,38 @@ STATIC mp_obj_t audioio_audioout_obj___exit__(size_t n_args, const mp_obj_t *arg
STATIC MP_DEFINE_CONST_FUN_OBJ_VAR_BETWEEN(audioio_audioout___exit___obj, 4, 4, audioio_audioout_obj___exit__);
//| .. method:: play(loop=False)
//| .. method:: play(sample, *, loop=False)
//|
//| Plays the sample once when loop=False and continuously when loop=True.
//| Does not block. Use `playing` to block.
//|
//| Sample must be an `audioio.WaveFile` or `audioio.RawSample`.
//|
//| The sample itself should consist of 16 bit samples. Microcontrollers with a lower output
//| resolution will use the highest order bits to output. For example, the SAMD21 has a 10 bit
//| DAC that ignores the lowest 6 bits when playing 16 bit samples.
//|
STATIC mp_obj_t audioio_audioout_obj_play(size_t n_args, const mp_obj_t *pos_args, mp_map_t *kw_args) {
enum { ARG_loop };
enum { ARG_sample, ARG_loop };
static const mp_arg_t allowed_args[] = {
{ MP_QSTR_loop, MP_ARG_BOOL, {.u_bool = false} },
{ MP_QSTR_sample, MP_ARG_OBJ | MP_ARG_REQUIRED },
{ MP_QSTR_loop, MP_ARG_BOOL | MP_ARG_KW_ONLY, {.u_bool = false} },
};
audioio_audioout_obj_t *self = MP_OBJ_TO_PTR(pos_args[0]);
raise_error_if_deinited(common_hal_audioio_audioout_deinited(self));
mp_arg_val_t args[MP_ARRAY_SIZE(allowed_args)];
mp_arg_parse_all(n_args - 1, pos_args + 1, kw_args, MP_ARRAY_SIZE(allowed_args), allowed_args, args);
common_hal_audioio_audioout_play(self, args[ARG_loop].u_bool);
mp_obj_t sample = args[ARG_sample].u_obj;
common_hal_audioio_audioout_play(self, sample, args[ARG_loop].u_bool);
return mp_const_none;
}
MP_DEFINE_CONST_FUN_OBJ_KW(audioio_audioout_play_obj, 1, audioio_audioout_obj_play);
//| .. method:: stop()
//|
//| Stops playback of this sample. If another sample is playing instead, it
//| won't be stopped.
//| Stops playback.
//|
STATIC mp_obj_t audioio_audioout_obj_stop(mp_obj_t self_in) {
audioio_audioout_obj_t *self = MP_OBJ_TO_PTR(self_in);
@ -187,7 +193,7 @@ MP_DEFINE_CONST_FUN_OBJ_1(audioio_audioout_stop_obj, audioio_audioout_obj_stop);
//| .. attribute:: playing
//|
//| True when the audio sample is being output. (read-only)
//| True when an audio sample is being output. (read-only)
//|
STATIC mp_obj_t audioio_audioout_obj_get_playing(mp_obj_t self_in) {
audioio_audioout_obj_t *self = MP_OBJ_TO_PTR(self_in);
@ -203,34 +209,6 @@ const mp_obj_property_t audioio_audioout_playing_obj = {
(mp_obj_t)&mp_const_none_obj},
};
//| .. attribute:: frequency
//|
//| 32 bit value that dictates how quickly samples are loaded into the DAC
//| in Hertz (cycles per second). When the sample is looped, this can change
//| the pitch output without changing the underlying sample.
//|
STATIC mp_obj_t audioio_audioout_obj_get_frequency(mp_obj_t self_in) {
audioio_audioout_obj_t *self = MP_OBJ_TO_PTR(self_in);
raise_error_if_deinited(common_hal_audioio_audioout_deinited(self));
return MP_OBJ_NEW_SMALL_INT(common_hal_audioio_audioout_get_frequency(self));
}
MP_DEFINE_CONST_FUN_OBJ_1(audioio_audioout_get_frequency_obj, audioio_audioout_obj_get_frequency);
STATIC mp_obj_t audioio_audioout_obj_set_frequency(mp_obj_t self_in, mp_obj_t frequency) {
audioio_audioout_obj_t *self = MP_OBJ_TO_PTR(self_in);
raise_error_if_deinited(common_hal_audioio_audioout_deinited(self));
common_hal_audioio_audioout_set_frequency(self, mp_obj_get_int(frequency));
return mp_const_none;
}
MP_DEFINE_CONST_FUN_OBJ_2(audioio_audioout_set_frequency_obj, audioio_audioout_obj_set_frequency);
const mp_obj_property_t audioio_audioout_frequency_obj = {
.base.type = &mp_type_property,
.proxy = {(mp_obj_t)&audioio_audioout_get_frequency_obj,
(mp_obj_t)&audioio_audioout_set_frequency_obj,
(mp_obj_t)&mp_const_none_obj},
};
STATIC const mp_rom_map_elem_t audioio_audioout_locals_dict_table[] = {
// Methods
{ MP_ROM_QSTR(MP_QSTR_deinit), MP_ROM_PTR(&audioio_audioout_deinit_obj) },
@ -241,7 +219,6 @@ STATIC const mp_rom_map_elem_t audioio_audioout_locals_dict_table[] = {
// Properties
{ MP_ROM_QSTR(MP_QSTR_playing), MP_ROM_PTR(&audioio_audioout_playing_obj) },
{ MP_ROM_QSTR(MP_QSTR_frequency), MP_ROM_PTR(&audioio_audioout_frequency_obj) },
};
STATIC MP_DEFINE_CONST_DICT(audioio_audioout_locals_dict, audioio_audioout_locals_dict_table);

View File

@ -29,21 +29,18 @@
#include "common-hal/audioio/AudioOut.h"
#include "common-hal/microcontroller/Pin.h"
#include "extmod/vfs_fat_file.h"
#include "shared-bindings/audioio/RawSample.h"
extern const mp_obj_type_t audioio_audioout_type;
void common_hal_audioio_audioout_construct_from_buffer(audioio_audioout_obj_t* self,
const mcu_pin_obj_t* pin, uint16_t* buffer, uint32_t len, uint8_t bytes_per_sample);
void common_hal_audioio_audioout_construct_from_file(audioio_audioout_obj_t* self,
const mcu_pin_obj_t* pin, pyb_file_obj_t* file);
// left_channel will always be non-NULL but right_channel may be for mono output.
void common_hal_audioio_audioout_construct(audioio_audioout_obj_t* self,
const mcu_pin_obj_t* left_channel, const mcu_pin_obj_t* right_channel);
void common_hal_audioio_audioout_deinit(audioio_audioout_obj_t* self);
bool common_hal_audioio_audioout_deinited(audioio_audioout_obj_t* self);
void common_hal_audioio_audioout_play(audioio_audioout_obj_t* self, bool loop);
void common_hal_audioio_audioout_play(audioio_audioout_obj_t* self, mp_obj_t sample, bool loop);
void common_hal_audioio_audioout_stop(audioio_audioout_obj_t* self);
bool common_hal_audioio_audioout_get_playing(audioio_audioout_obj_t* self);
uint32_t common_hal_audioio_audioout_get_frequency(audioio_audioout_obj_t* self);
void common_hal_audioio_audioout_set_frequency(audioio_audioout_obj_t* self, uint32_t frequency);
#endif // MICROPY_INCLUDED_SHARED_BINDINGS_AUDIOIO_AUDIOOUT_H

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@ -0,0 +1,183 @@
/*
* This file is part of the Micro Python project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2017 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 <stdint.h>
#include "lib/utils/context_manager_helpers.h"
#include "py/binary.h"
#include "py/objproperty.h"
#include "py/runtime.h"
#include "shared-bindings/microcontroller/Pin.h"
#include "shared-bindings/audioio/AudioOut.h"
#include "shared-bindings/util.h"
//| .. currentmodule:: audioio
//|
//| :class:`RawSample` -- A raw audio sample buffer
//| ========================================================
//|
//| An in-memory sound sample
//|
//| .. class:: RawSample(buffer, *, channel_count=1, sample_rate=8000)
//|
//| Create a RawSample based on the given buffer of signed values. If channel_count is more than
//| 1 then each channel's samples should rotate. In other words, for a two channel buffer, the
//| first sample will be for channel 1, the second sample will be for channel two, the third for
//| channel 1 and so on.
//|
//| :param array buffer: An `array.array` with samples
//| :param int channel_count: The number of channels in the buffer
//| :param int sample_rate: The desired playback sample rate
//|
//| Simple 8ksps 440 Hz sin wave::
//|
//| import audioio
//| import board
//| import array
//| import time
//| import math
//|
//| # Generate one period of sine wav.
//| length = 8000 // 440
//| sine_wave = array.array("h", [0] * length)
//| for i in range(length):
//| sine_wave[i] = int(math.sin(math.pi * 2 * i / 18) * (2 ** 15))
//|
//| dac = audioio.AudioOut(board.SPEAKER)
//| sine_wave = audioio.RawSample(sine_wave)
//| dac.play(sine_wave, loop=True)
//| time.sleep(1)
//| sample.stop()
//|
STATIC mp_obj_t audioio_rawsample_make_new(const mp_obj_type_t *type, size_t n_args, size_t n_kw, const mp_obj_t *pos_args) {
mp_arg_check_num(n_args, n_kw, 1, 2, true);
mp_map_t kw_args;
mp_map_init_fixed_table(&kw_args, n_kw, pos_args + n_args);
enum { ARG_buffer, ARG_channel_count, ARG_sample_rate };
static const mp_arg_t allowed_args[] = {
{ MP_QSTR_buffer, MP_ARG_OBJ | MP_ARG_REQUIRED },
{ MP_QSTR_channel_count, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 1 } },
{ MP_QSTR_sample_rate, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 8000} },
};
mp_arg_val_t args[MP_ARRAY_SIZE(allowed_args)];
mp_arg_parse_all(n_args, pos_args, &kw_args, MP_ARRAY_SIZE(allowed_args), allowed_args, args);
audioio_rawsample_obj_t *self = m_new_obj(audioio_rawsample_obj_t);
self->base.type = &audioio_rawsample_type;
mp_buffer_info_t bufinfo;
if (mp_get_buffer(args[ARG_buffer].u_obj, &bufinfo, MP_BUFFER_READ)) {
uint8_t bytes_per_sample = 1;
bool signed_samples = bufinfo.typecode == 'b' || bufinfo.typecode == 'h';
if (bufinfo.typecode == 'h' || bufinfo.typecode == 'H') {
bytes_per_sample = 2;
} else if (bufinfo.typecode != 'b' && bufinfo.typecode != 'B' && bufinfo.typecode != BYTEARRAY_TYPECODE) {
mp_raise_ValueError("sample_source buffer must be a bytearray or array of type 'h', 'H', 'b' or 'B'");
}
common_hal_audioio_rawsample_construct(self, ((uint8_t*)bufinfo.buf), bufinfo.len,
bytes_per_sample, signed_samples, args[ARG_channel_count].u_int,
args[ARG_sample_rate].u_int);
} else {
mp_raise_TypeError("buffer must be a bytes-like object");
}
return MP_OBJ_FROM_PTR(self);
}
//| .. method:: deinit()
//|
//| Deinitialises the AudioOut and releases any hardware resources for reuse.
//|
STATIC mp_obj_t audioio_rawsample_deinit(mp_obj_t self_in) {
audioio_rawsample_obj_t *self = MP_OBJ_TO_PTR(self_in);
common_hal_audioio_rawsample_deinit(self);
return mp_const_none;
}
STATIC MP_DEFINE_CONST_FUN_OBJ_1(audioio_rawsample_deinit_obj, audioio_rawsample_deinit);
//| .. method:: __enter__()
//|
//| No-op used by Context Managers.
//|
// Provided by context manager helper.
//| .. method:: __exit__()
//|
//| Automatically deinitializes the hardware when exiting a context. See
//| :ref:`lifetime-and-contextmanagers` for more info.
//|
STATIC mp_obj_t audioio_rawsample_obj___exit__(size_t n_args, const mp_obj_t *args) {
(void)n_args;
common_hal_audioio_rawsample_deinit(args[0]);
return mp_const_none;
}
STATIC MP_DEFINE_CONST_FUN_OBJ_VAR_BETWEEN(audioio_rawsample___exit___obj, 4, 4, audioio_rawsample_obj___exit__);
//| .. attribute:: sample_rate
//|
//| 32 bit value that dictates how quickly samples are played in Hertz (cycles per second).
//| When the sample is looped, this can change the pitch output without changing the underlying
//| sample. This will not change the sample rate of any active playback. Call ``play`` again to
//| change it.
//|
STATIC mp_obj_t audioio_rawsample_obj_get_sample_rate(mp_obj_t self_in) {
audioio_rawsample_obj_t *self = MP_OBJ_TO_PTR(self_in);
raise_error_if_deinited(common_hal_audioio_rawsample_deinited(self));
return MP_OBJ_NEW_SMALL_INT(common_hal_audioio_rawsample_get_sample_rate(self));
}
MP_DEFINE_CONST_FUN_OBJ_1(audioio_rawsample_get_sample_rate_obj, audioio_rawsample_obj_get_sample_rate);
STATIC mp_obj_t audioio_rawsample_obj_set_sample_rate(mp_obj_t self_in, mp_obj_t sample_rate) {
audioio_rawsample_obj_t *self = MP_OBJ_TO_PTR(self_in);
raise_error_if_deinited(common_hal_audioio_rawsample_deinited(self));
common_hal_audioio_rawsample_set_sample_rate(self, mp_obj_get_int(sample_rate));
return mp_const_none;
}
MP_DEFINE_CONST_FUN_OBJ_2(audioio_rawsample_set_sample_rate_obj, audioio_rawsample_obj_set_sample_rate);
const mp_obj_property_t audioio_rawsample_sample_rate_obj = {
.base.type = &mp_type_property,
.proxy = {(mp_obj_t)&audioio_rawsample_get_sample_rate_obj,
(mp_obj_t)&audioio_rawsample_set_sample_rate_obj,
(mp_obj_t)&mp_const_none_obj},
};
STATIC const mp_rom_map_elem_t audioio_rawsample_locals_dict_table[] = {
// Methods
{ MP_ROM_QSTR(MP_QSTR_deinit), MP_ROM_PTR(&audioio_rawsample_deinit_obj) },
{ MP_ROM_QSTR(MP_QSTR___enter__), MP_ROM_PTR(&default___enter___obj) },
{ MP_ROM_QSTR(MP_QSTR___exit__), MP_ROM_PTR(&audioio_rawsample___exit___obj) },
// Properties
{ MP_ROM_QSTR(MP_QSTR_sample_rate), MP_ROM_PTR(&audioio_rawsample_sample_rate_obj) },
};
STATIC MP_DEFINE_CONST_DICT(audioio_rawsample_locals_dict, audioio_rawsample_locals_dict_table);
const mp_obj_type_t audioio_rawsample_type = {
{ &mp_type_type },
.name = MP_QSTR_RawSample,
.make_new = audioio_rawsample_make_new,
.locals_dict = (mp_obj_dict_t*)&audioio_rawsample_locals_dict,
};

View File

@ -0,0 +1,45 @@
/*
* This file is part of the Micro Python project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2017 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.
*/
#ifndef MICROPY_INCLUDED_SHARED_BINDINGS_AUDIOIO_RAWSAMPLE_H
#define MICROPY_INCLUDED_SHARED_BINDINGS_AUDIOIO_RAWSAMPLE_H
#include "common-hal/audioio/AudioOut.h"
#include "common-hal/microcontroller/Pin.h"
#include "shared-module/audioio/RawSample.h"
extern const mp_obj_type_t audioio_rawsample_type;
void common_hal_audioio_rawsample_construct(audioio_rawsample_obj_t* self,
uint8_t* buffer, uint32_t len, uint8_t bytes_per_sample, bool samples_signed,
uint8_t channel_count, uint32_t sample_rate);
void common_hal_audioio_rawsample_deinit(audioio_rawsample_obj_t* self);
bool common_hal_audioio_rawsample_deinited(audioio_rawsample_obj_t* self);
uint32_t common_hal_audioio_rawsample_get_sample_rate(audioio_rawsample_obj_t* self);
void common_hal_audioio_rawsample_set_sample_rate(audioio_rawsample_obj_t* self, uint32_t sample_rate);
#endif // MICROPY_INCLUDED_SHARED_BINDINGS_AUDIOIO_RAWSAMPLE_H

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@ -0,0 +1,157 @@
/*
* This file is part of the Micro Python project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2018 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 <stdint.h>
#include "lib/utils/context_manager_helpers.h"
// #include "py/binary.h"
#include "py/objproperty.h"
#include "py/runtime.h"
// #include "shared-bindings/microcontroller/Pin.h"
#include "shared-bindings/audioio/WaveFile.h"
#include "shared-bindings/util.h"
//| .. currentmodule:: audioio
//|
//| :class:`WaveFile` -- Load a wave file for audio playback
//| ========================================================
//|
//| A .wav file prepped for audio playback
//|
//| .. class:: WaveFile(filename)
//|
//| Load a .wav file for playback with `audioio.AudioOut` or `audiobusio.I2SOut`.
//|
//| :param bytes-like file: Already opened wave file
//|
//| Playing a wave file from flash::
//|
//| import board
//| import audioio
//| import digitalio
//|
//| # Required for CircuitPlayground Express
//| speaker_enable = digitalio.DigitalInOut(board.SPEAKER_ENABLE)
//| speaker_enable.switch_to_output(value=True)
//|
//| data = open("cplay-5.1-16bit-16khz.wav", "rb")
//| wav = audioio.WaveFile(data)
//| a = audioio.AudioOut(board.A0)
//|
//| print("playing")
//| a.play(wav)
//| while a.playing:
//| pass
//| print("stopped")
//|
STATIC mp_obj_t audioio_wavefile_make_new(const mp_obj_type_t *type, size_t n_args, size_t n_kw, const mp_obj_t *args) {
mp_arg_check_num(n_args, n_kw, 1, 1, true);
audioio_wavefile_obj_t *self = m_new_obj(audioio_wavefile_obj_t);
self->base.type = &audioio_wavefile_type;
if (MP_OBJ_IS_TYPE(args[0], &fatfs_type_fileio)) {
common_hal_audioio_wavefile_construct(self, MP_OBJ_TO_PTR(args[0]));
} else {
mp_raise_TypeError("file must be a file opened in byte mode");
}
return MP_OBJ_FROM_PTR(self);
}
//| .. method:: deinit()
//|
//| Deinitialises the WaveFile and releases all memory resources for reuse.
//|
STATIC mp_obj_t audioio_wavefile_deinit(mp_obj_t self_in) {
audioio_wavefile_obj_t *self = MP_OBJ_TO_PTR(self_in);
common_hal_audioio_wavefile_deinit(self);
return mp_const_none;
}
STATIC MP_DEFINE_CONST_FUN_OBJ_1(audioio_wavefile_deinit_obj, audioio_wavefile_deinit);
//| .. method:: __enter__()
//|
//| No-op used by Context Managers.
//|
// Provided by context manager helper.
//| .. method:: __exit__()
//|
//| Automatically deinitializes the hardware when exiting a context. See
//| :ref:`lifetime-and-contextmanagers` for more info.
//|
STATIC mp_obj_t audioio_wavefile_obj___exit__(size_t n_args, const mp_obj_t *args) {
(void)n_args;
common_hal_audioio_wavefile_deinit(args[0]);
return mp_const_none;
}
STATIC MP_DEFINE_CONST_FUN_OBJ_VAR_BETWEEN(audioio_wavefile___exit___obj, 4, 4, audioio_wavefile_obj___exit__);
//| .. attribute:: sample_rate
//|
//| 32 bit value that dictates how quickly samples are loaded into the DAC
//| in Hertz (cycles per second). When the sample is looped, this can change
//| the pitch output without changing the underlying sample.
//|
STATIC mp_obj_t audioio_wavefile_obj_get_sample_rate(mp_obj_t self_in) {
audioio_wavefile_obj_t *self = MP_OBJ_TO_PTR(self_in);
raise_error_if_deinited(common_hal_audioio_wavefile_deinited(self));
return MP_OBJ_NEW_SMALL_INT(common_hal_audioio_wavefile_get_sample_rate(self));
}
MP_DEFINE_CONST_FUN_OBJ_1(audioio_wavefile_get_sample_rate_obj, audioio_wavefile_obj_get_sample_rate);
STATIC mp_obj_t audioio_wavefile_obj_set_sample_rate(mp_obj_t self_in, mp_obj_t sample_rate) {
audioio_wavefile_obj_t *self = MP_OBJ_TO_PTR(self_in);
raise_error_if_deinited(common_hal_audioio_wavefile_deinited(self));
common_hal_audioio_wavefile_set_sample_rate(self, mp_obj_get_int(sample_rate));
return mp_const_none;
}
MP_DEFINE_CONST_FUN_OBJ_2(audioio_wavefile_set_sample_rate_obj, audioio_wavefile_obj_set_sample_rate);
const mp_obj_property_t audioio_wavefile_sample_rate_obj = {
.base.type = &mp_type_property,
.proxy = {(mp_obj_t)&audioio_wavefile_get_sample_rate_obj,
(mp_obj_t)&audioio_wavefile_set_sample_rate_obj,
(mp_obj_t)&mp_const_none_obj},
};
STATIC const mp_rom_map_elem_t audioio_wavefile_locals_dict_table[] = {
// Methods
{ MP_ROM_QSTR(MP_QSTR_deinit), MP_ROM_PTR(&audioio_wavefile_deinit_obj) },
{ MP_ROM_QSTR(MP_QSTR___enter__), MP_ROM_PTR(&default___enter___obj) },
{ MP_ROM_QSTR(MP_QSTR___exit__), MP_ROM_PTR(&audioio_wavefile___exit___obj) },
// Properties
{ MP_ROM_QSTR(MP_QSTR_sample_rate), MP_ROM_PTR(&audioio_wavefile_sample_rate_obj) },
};
STATIC MP_DEFINE_CONST_DICT(audioio_wavefile_locals_dict, audioio_wavefile_locals_dict_table);
const mp_obj_type_t audioio_wavefile_type = {
{ &mp_type_type },
.name = MP_QSTR_WaveFile,
.make_new = audioio_wavefile_make_new,
.locals_dict = (mp_obj_dict_t*)&audioio_wavefile_locals_dict,
};

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@ -0,0 +1,44 @@
/*
* This file is part of the Micro Python project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2017 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.
*/
#ifndef MICROPY_INCLUDED_SHARED_BINDINGS_AUDIOIO_WAVEFILE_H
#define MICROPY_INCLUDED_SHARED_BINDINGS_AUDIOIO_WAVEFILE_H
#include "common-hal/audioio/AudioOut.h"
#include "common-hal/microcontroller/Pin.h"
#include "extmod/vfs_fat_file.h"
extern const mp_obj_type_t audioio_wavefile_type;
void common_hal_audioio_wavefile_construct(audioio_wavefile_obj_t* self,
pyb_file_obj_t* file);
void common_hal_audioio_wavefile_deinit(audioio_wavefile_obj_t* self);
bool common_hal_audioio_wavefile_deinited(audioio_wavefile_obj_t* self);
uint32_t common_hal_audioio_wavefile_get_sample_rate(audioio_wavefile_obj_t* self);
void common_hal_audioio_wavefile_set_sample_rate(audioio_wavefile_obj_t* self, uint32_t sample_rate);
#endif // MICROPY_INCLUDED_SHARED_BINDINGS_AUDIOIO_WAVEFILE_H

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@ -32,6 +32,7 @@
#include "shared-bindings/microcontroller/Pin.h"
#include "shared-bindings/audioio/__init__.h"
#include "shared-bindings/audioio/AudioOut.h"
#include "shared-bindings/audioio/WaveFile.h"
//| :mod:`audioio` --- Support for audio input and output
//| ======================================================
@ -48,6 +49,8 @@
//| :maxdepth: 3
//|
//| AudioOut
//| RawSample
//| WaveFile
//|
//| All classes change hardware state and should be deinitialized when they
//| are no longer needed if the program continues after use. To do so, either
@ -58,6 +61,8 @@
STATIC const mp_rom_map_elem_t audioio_module_globals_table[] = {
{ MP_ROM_QSTR(MP_QSTR___name__), MP_ROM_QSTR(MP_QSTR_audioio) },
{ MP_ROM_QSTR(MP_QSTR_AudioOut), MP_ROM_PTR(&audioio_audioout_type) },
{ MP_ROM_QSTR(MP_QSTR_RawSample), MP_ROM_PTR(&audioio_rawsample_type) },
{ MP_ROM_QSTR(MP_QSTR_WaveFile), MP_ROM_PTR(&audioio_wavefile_type) },
};
STATIC MP_DEFINE_CONST_DICT(audioio_module_globals, audioio_module_globals_table);

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@ -0,0 +1,94 @@
/*
* This file is part of the Micro Python project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2018 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 "shared-bindings/audioio/RawSample.h"
#include <stdint.h>
#include "shared-module/audioio/RawSample.h"
void common_hal_audioio_rawsample_construct(audioio_rawsample_obj_t* self,
uint8_t* buffer,
uint32_t len,
uint8_t bytes_per_sample,
bool samples_signed,
uint8_t channel_count,
uint32_t sample_rate) {
self->buffer = buffer;
self->bits_per_sample = bytes_per_sample * 8;
self->samples_signed = samples_signed;
self->len = len;
self->channel_count = channel_count;
self->sample_rate = sample_rate;
self->buffer_read = false;
}
void common_hal_audioio_rawsample_deinit(audioio_rawsample_obj_t* self) {
self->buffer = NULL;
}
bool common_hal_audioio_rawsample_deinited(audioio_rawsample_obj_t* self) {
return self->buffer == NULL;
}
uint32_t common_hal_audioio_rawsample_get_sample_rate(audioio_rawsample_obj_t* self) {
return self->sample_rate;
}
void common_hal_audioio_rawsample_set_sample_rate(audioio_rawsample_obj_t* self,
uint32_t sample_rate) {
self->sample_rate = sample_rate;
}
void audioio_rawsample_reset_buffer(audioio_rawsample_obj_t* self,
bool single_channel,
uint8_t channel) {
}
bool audioio_rawsample_get_buffer(audioio_rawsample_obj_t* self,
bool single_channel,
uint8_t channel,
uint8_t** buffer,
uint32_t* buffer_length) {
*buffer_length = self->len;
if (single_channel) {
*buffer = self->buffer + (channel % self->channel_count) * (self->bits_per_sample / 8);
} else {
*buffer = self->buffer;
}
return true;
}
void audioio_rawsample_get_buffer_structure(audioio_rawsample_obj_t* self, bool single_channel,
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) {
*spacing = self->channel_count;
} else {
*spacing = 1;
}
}

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@ -0,0 +1,57 @@
/*
* This file is part of the MicroPython project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2018 Scott Shawcroft
*
* 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.
*/
#ifndef MICROPY_INCLUDED_SHARED_MODULE_AUDIOIO_RAWSAMPLE_H
#define MICROPY_INCLUDED_SHARED_MODULE_AUDIOIO_RAWSAMPLE_H
#include "py/obj.h"
typedef struct {
mp_obj_base_t base;
uint8_t* buffer;
uint32_t len;
uint8_t bits_per_sample;
bool samples_signed;
uint8_t channel_count;
uint32_t sample_rate;
bool buffer_read;
} audioio_rawsample_obj_t;
// 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,
uint8_t channel);
bool audioio_rawsample_get_buffer(audioio_rawsample_obj_t* self,
bool single_channel,
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,
bool* single_buffer, bool* samples_signed,
uint32_t* max_buffer_length, uint8_t* spacing);
#endif // MICROPY_INCLUDED_SHARED_MODULE_AUDIOIO_RAWSAMPLE_H

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@ -0,0 +1,230 @@
/*
* This file is part of the Micro Python project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2018 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 "shared-bindings/audioio/WaveFile.h"
#include <stdint.h>
#include <string.h>
#include "py/runtime.h"
#include "shared-module/audioio/WaveFile.h"
struct wave_format_chunk {
uint16_t audio_format;
uint16_t num_channels;
uint32_t sample_rate;
uint32_t byte_rate;
uint16_t block_align;
uint16_t bits_per_sample;
uint16_t extra_params; // Assumed to be zero below.
};
void common_hal_audioio_wavefile_construct(audioio_wavefile_obj_t* self,
pyb_file_obj_t* file) {
// Load the wave
self->file = file;
uint8_t chunk_header[16];
f_rewind(&self->file->fp);
UINT bytes_read;
f_read(&self->file->fp, chunk_header, 16, &bytes_read);
if (bytes_read != 16 ||
memcmp(chunk_header, "RIFF", 4) != 0 ||
memcmp(chunk_header + 8, "WAVEfmt ", 8) != 0) {
mp_raise_ValueError("Invalid wave file");
}
uint32_t format_size;
f_read(&self->file->fp, &format_size, 4, &bytes_read);
if (bytes_read != 4 ||
format_size > sizeof(struct wave_format_chunk)) {
mp_raise_ValueError("Invalid format chunk size");
}
struct wave_format_chunk format;
f_read(&self->file->fp, &format, format_size, &bytes_read);
if (bytes_read != format_size) {
}
if (format.audio_format != 1 ||
format.num_channels > 2 ||
format.bits_per_sample > 16 ||
(format_size == 18 &&
format.extra_params != 0)) {
mp_raise_ValueError("Unsupported format");
}
// Get the sample_rate
self->sample_rate = format.sample_rate;
self->len = 512;
self->channel_count = format.num_channels;
self->bits_per_sample = format.bits_per_sample;
// TODO(tannewt): Skip any extra chunks that occur before the data section.
uint8_t data_tag[4];
f_read(&self->file->fp, &data_tag, 4, &bytes_read);
if (bytes_read != 4 ||
memcmp((uint8_t *) data_tag, "data", 4) != 0) {
mp_raise_ValueError("Data chunk must follow fmt chunk");
}
uint32_t data_length;
f_read(&self->file->fp, &data_length, 4, &bytes_read);
if (bytes_read != 4) {
mp_raise_ValueError("Invalid file");
}
self->file_length = data_length;
self->data_start = self->file->fp.fptr;
// Try to allocate two buffers, one will be loaded from file and the other
// DMAed to DAC.
self->buffer = m_malloc(self->len, false);
if (self->buffer == NULL) {
common_hal_audioio_wavefile_deinit(self);
mp_raise_msg(&mp_type_MemoryError, "");
}
self->second_buffer = m_malloc(self->len, false);
if (self->second_buffer == NULL) {
common_hal_audioio_wavefile_deinit(self);
mp_raise_msg(&mp_type_MemoryError, "");
}
}
void common_hal_audioio_wavefile_deinit(audioio_wavefile_obj_t* self) {
self->buffer = NULL;
}
bool common_hal_audioio_wavefile_deinited(audioio_wavefile_obj_t* self) {
return self->buffer == NULL;
}
uint32_t common_hal_audioio_wavefile_get_sample_rate(audioio_wavefile_obj_t* self) {
return self->sample_rate;
}
void common_hal_audioio_wavefile_set_sample_rate(audioio_wavefile_obj_t* self,
uint32_t sample_rate) {
self->sample_rate = sample_rate;
}
bool audioio_wavefile_samples_signed(audioio_wavefile_obj_t* self) {
return self->bits_per_sample > 8;
}
uint32_t audioio_wavefile_max_buffer_length(audioio_wavefile_obj_t* self) {
return 512;
}
void audioio_wavefile_reset_buffer(audioio_wavefile_obj_t* self,
bool single_channel,
uint8_t channel) {
if (single_channel && channel == 1) {
return;
}
// We don't reset the buffer index in case we're looping and we have an odd number of buffer
// loads
self->bytes_remaining = self->file_length;
f_lseek(&self->file->fp, self->data_start);
self->read_count = 0;
self->left_read_count = 0;
self->right_read_count = 0;
}
bool audioio_wavefile_get_buffer(audioio_wavefile_obj_t* self,
bool single_channel,
uint8_t channel,
uint8_t** buffer,
uint32_t* buffer_length) {
if (!single_channel) {
channel = 0;
}
uint32_t channel_read_count = self->left_read_count;
if (channel == 1) {
channel_read_count = self->right_read_count;
}
bool need_more_data = self->read_count == channel_read_count;
if (self->bytes_remaining == 0 && need_more_data) {
*buffer = NULL;
*buffer_length = 0;
return true;
}
if (need_more_data) {
uint16_t num_bytes_to_load = self->len;
if (num_bytes_to_load > self->bytes_remaining) {
num_bytes_to_load = self->bytes_remaining;
}
UINT length_read;
if (self->buffer_index % 2 == 1) {
*buffer = self->second_buffer;
} else {
*buffer = self->buffer;
}
f_read(&self->file->fp, *buffer, num_bytes_to_load, &length_read);
*buffer_length = length_read;
if (self->buffer_index % 2 == 1) {
self->second_buffer_length = length_read;
} else {
self->buffer_length = length_read;
}
self->bytes_remaining -= length_read;
self->buffer_index += 1;
self->read_count += 1;
}
uint32_t buffers_back = self->read_count - 1 - channel_read_count;
if ((self->buffer_index - buffers_back) % 2 == 0) {
*buffer = self->second_buffer;
*buffer_length = self->second_buffer_length;
} else {
*buffer = self->buffer;
*buffer_length = self->buffer_length;
}
if (channel == 0) {
self->left_read_count += 1;
} else if (channel == 1) {
self->right_read_count += 1;
*buffer = *buffer + self->bits_per_sample / 8;
}
return self->bytes_remaining == 0;
}
void audioio_wavefile_get_buffer_structure(audioio_wavefile_obj_t* self, bool single_channel,
bool* single_buffer, bool* samples_signed,
uint32_t* max_buffer_length, uint8_t* spacing) {
*single_buffer = false;
*samples_signed = self->bits_per_sample > 8;
*max_buffer_length = 512;
if (single_channel) {
*spacing = self->channel_count;
} else {
*spacing = 1;
}
}

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@ -0,0 +1,68 @@
/*
* This file is part of the MicroPython project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2018 Scott Shawcroft
*
* 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.
*/
#ifndef MICROPY_INCLUDED_SHARED_MODULE_AUDIOIO_WAVEFILE_H
#define MICROPY_INCLUDED_SHARED_MODULE_AUDIOIO_WAVEFILE_H
#include "py/obj.h"
typedef struct {
mp_obj_base_t base;
uint8_t* buffer;
uint32_t buffer_length;
uint8_t* second_buffer;
uint32_t second_buffer_length;
uint32_t file_length; // In bytes
uint16_t data_start; // Where the data values start
uint8_t bits_per_sample;
uint16_t buffer_index;
uint32_t bytes_remaining;
uint8_t channel_count;
uint16_t sample_rate;
uint32_t len;
pyb_file_obj_t* file;
uint32_t read_count;
uint32_t left_read_count;
uint32_t right_read_count;
} audioio_wavefile_obj_t;
// 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,
uint8_t channel);
bool audioio_wavefile_get_buffer(audioio_wavefile_obj_t* self,
bool single_channel,
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,
bool* single_buffer, bool* samples_signed,
uint32_t* max_buffer_length, uint8_t* spacing);
#endif // MICROPY_INCLUDED_SHARED_MODULE_AUDIOIO_WAVEFILE_H